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[alsa-devel] [PATCH] ASoC: wm8996: Call _POST_PMU callback for CPVDD
by Mark Brown 22 Jan '12

22 Jan '12

We should be allowing a 5ms delay after the charge pump is started in order to ensure it has finished ramping. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8e8f8d1..390ccdd 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1120,7 +1120,8 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), -- 1.7.7.3

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[alsa-devel] [PATCH 1/3] ASoC: Provide REGULATOR_SUPPLY widget type
by Mark Brown 22 Jan '12

22 Jan '12

Modern devices allow systems to enable and disable individual supplies on the device, allowing additional power saving by switching off regulators which power portions of the device which are not currently in use. Add a new SND_SOC_DAPM_REGULATOR_SUPPLY widget type factoring out the code for managing such widgets from individual drivers. The widget name will be used as the supply name when requesting the regulator from the regulator API. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 9 ++++++++ sound/soc/soc-dapm.c | 51 ++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 58 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index d26a9b7..bfa0d3c 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -243,6 +243,10 @@ { .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \ .shift = wshift, .invert = winvert, .event = wevent, \ .event_flags = wflags} +#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay) \ +{ .id = snd_soc_dapm_regulator_supply, .name = wname, \ + .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ @@ -322,6 +326,8 @@ struct snd_soc_dapm_context; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_regulator_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -411,6 +417,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_regulator_supply, /* external regulator */ snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ @@ -465,6 +472,8 @@ struct snd_soc_dapm_widget { struct list_head list; struct snd_soc_dapm_context *dapm; + void *priv; /* widget specific data */ + /* dapm control */ short reg; /* negative reg = no direct dapm */ unsigned char shift; /* bits to shift */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 541bcbf..131508a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -40,6 +40,7 @@ #include <linux/jiffies.h> #include <linux/debugfs.h> #include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -55,6 +56,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, + [snd_soc_dapm_regulator_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, @@ -90,6 +92,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, + [snd_soc_dapm_regulator_supply] = 11, [snd_soc_dapm_supply] = 11, [snd_soc_dapm_post] = 12, }; @@ -352,6 +355,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_micbias: case snd_soc_dapm_vmid: case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_hp: @@ -680,8 +684,13 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) DAPM_UPDATE_STAT(widget, path_checks); - if (widget->id == snd_soc_dapm_supply) + switch (widget->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: return 0; + default: + break; + } switch (widget->id) { case snd_soc_dapm_adc: @@ -745,8 +754,13 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) DAPM_UPDATE_STAT(widget, path_checks); - if (widget->id == snd_soc_dapm_supply) + switch (widget->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: return 0; + default: + break; + } /* active stream ? */ switch (widget->id) { @@ -828,6 +842,19 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_reg_event); +/* + * Handler for regulator supply widget. + */ +int dapm_regulator_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) + return regulator_enable(w->priv); + else + return regulator_disable_deferred(w->priv, w->shift); +} +EXPORT_SYMBOL_GPL(dapm_regulator_event); + static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) { if (w->power_checked) @@ -1308,6 +1335,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, } switch (w->id) { case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -1411,6 +1439,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) */ switch (w->id) { case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1769,6 +1798,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -2007,6 +2037,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_pre: case snd_soc_dapm_post: case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: list_add(&path->list, &dapm->card->paths); @@ -2673,10 +2704,25 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w; size_t name_len; + int ret; if ((w = dapm_cnew_widget(widget)) == NULL) return -ENOMEM; + switch (w->id) { + case snd_soc_dapm_regulator_supply: + w->priv = devm_regulator_get(dapm->dev, w->name); + if (IS_ERR(w->priv)) { + ret = PTR_ERR(w->priv); + dev_err(dapm->dev, "Failed to request %s: %d\n", + w->name, ret); + return ret; + } + break; + default: + break; + } + name_len = strlen(widget->name) + 1; if (dapm->codec && dapm->codec->name_prefix) name_len += 1 + strlen(dapm->codec->name_prefix); @@ -2722,6 +2768,7 @@ int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_supply: + case snd_soc_dapm_regulator_supply: w->power_check = dapm_supply_check_power; break; default: -- 1.7.7.3

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[alsa-devel] [PATCH 1/2] ASoC: wm8988: Convert to table based DAPM and control init
by Mark Brown 22 Jan '12

22 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8988.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index ab52963d..40aebaf 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -317,7 +317,7 @@ static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = { SND_SOC_DAPM_INPUT("RINPUT2"), }; -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route wm8988_dapm_routes[] = { { "Left Line Mux", "Line 1", "LINPUT1" }, { "Left Line Mux", "Line 2", "LINPUT2" }, @@ -743,7 +743,6 @@ static int wm8988_resume(struct snd_soc_codec *codec) static int wm8988_probe(struct snd_soc_codec *codec) { struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; int ret = 0; ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8988->control_type); @@ -767,12 +766,6 @@ static int wm8988_probe(struct snd_soc_codec *codec) wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, wm8988_snd_controls, - ARRAY_SIZE(wm8988_snd_controls)); - snd_soc_dapm_new_controls(dapm, wm8988_dapm_widgets, - ARRAY_SIZE(wm8988_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -791,6 +784,13 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8988 = { .reg_cache_size = ARRAY_SIZE(wm8988_reg), .reg_word_size = sizeof(u16), .reg_cache_default = wm8988_reg, + + .controls = wm8988_snd_controls, + .num_controls = ARRAY_SIZE(wm8988_snd_controls), + .dapm_widgets = wm8988_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8988_dapm_widgets), + .dapm_routes = wm8988_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8988_dapm_routes), }; #if defined(CONFIG_SPI_MASTER) -- 1.7.7.3

1 1

[alsa-devel] [PATCH 1/5] ASoC: wm8985: Convert to devm_kzalloc()
by Mark Brown 21 Jan '12

21 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8985.c | 10 ++-------- 1 files changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index c0c86b3..e62a4c5 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1079,7 +1079,7 @@ static int __devinit wm8985_spi_probe(struct spi_device *spi) struct wm8985_priv *wm8985; int ret; - wm8985 = kzalloc(sizeof *wm8985, GFP_KERNEL); + wm8985 = devm_kzalloc(&spi->dev, sizeof *wm8985, GFP_KERNEL); if (!wm8985) return -ENOMEM; @@ -1088,15 +1088,12 @@ static int __devinit wm8985_spi_probe(struct spi_device *spi) ret = snd_soc_register_codec(&spi->dev, &soc_codec_dev_wm8985, &wm8985_dai, 1); - if (ret < 0) - kfree(wm8985); return ret; } static int __devexit wm8985_spi_remove(struct spi_device *spi) { snd_soc_unregister_codec(&spi->dev); - kfree(spi_get_drvdata(spi)); return 0; } @@ -1117,7 +1114,7 @@ static __devinit int wm8985_i2c_probe(struct i2c_client *i2c, struct wm8985_priv *wm8985; int ret; - wm8985 = kzalloc(sizeof *wm8985, GFP_KERNEL); + wm8985 = devm_kzalloc(&i2c->dev, sizeof *wm8985, GFP_KERNEL); if (!wm8985) return -ENOMEM; @@ -1126,15 +1123,12 @@ static __devinit int wm8985_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8985, &wm8985_dai, 1); - if (ret < 0) - kfree(wm8985); return ret; } static __devexit int wm8985_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.7.3

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[alsa-devel] [PATCH 0/4] ASoC: Do not use core to handle msbits constraint for TI drivers
by Peter Ujfalusi 21 Jan '12

21 Jan '12

Hi, The implementation in the core applies the msbits constraint to all sample size. These codecs, and cpu dai drivers only have msbits constraint when they are configured with 32 bit sample size. Apply the msbits constraint in the drivers for 32 bit samples only. Regards, Peter --- Peter Ujfalusi (4): Revert "ASoC: tlv320dac33: Use core to set the msbits constraint" Revert "ASoC: twl4030: Use core to set the msbits constraint" Revert "ASoC: omap-dmic: Use core to set the msbits constraint" ASoC: omap-mcpdm: Do not use core to set msbit constraint sound/soc/codecs/tlv320dac33.c | 3 ++- sound/soc/codecs/twl4030.c | 7 +++---- sound/soc/omap/omap-dmic.c | 7 ++++--- sound/soc/omap/omap-mcpdm.c | 4 ++-- 4 files changed, 11 insertions(+), 10 deletions(-) -- 1.7.8.3

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[alsa-devel] [PATCH] - HDA-Models.txt: Update model names
by Dave Lentz 21 Jan '12

21 Jan '12

Update Conexant model names and add Realtek ALC665. Signed-off-by: Dave Lentz <dtl131 at gmail.com> --- Documentation/sound/alsa/HD-Audio-Models.txt | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-)

1 0

[alsa-devel] [PATCH v2] ASoC: Add external amplifier controls for Visstrim_M10.
by Javier Martin 20 Jan '12

20 Jan '12

Visstrim_M10 has an external class D amplifier. This patch provides support for controlling the 4 possible gain levels and per channel muting. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- Changes since v1: - Specify amplifier gains using tlv array. - Check range values in amplifier controls. --- sound/soc/imx/mx27vis-aic32x4.c | 80 +++++++++++++++++++++++++++++++++++++++ 1 files changed, 80 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index d37e23c..155899c 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -25,16 +25,37 @@ #include <linux/moduleparam.h> #include <linux/device.h> #include <linux/i2c.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/tlv.h> #include <asm/mach-types.h> #include <mach/audmux.h> +#include <mach/iomux-mx27.h> #include "../codecs/tlv320aic32x4.h" #include "imx-ssi.h" +#define MX27VIS_AMP_GAIN 0 +#define MX27VIS_AMP_MUTE 1 + +#define MX27VIS_PIN_G0 (GPIO_PORTF + 9) +#define MX27VIS_PIN_G1 (GPIO_PORTF + 8) +#define MX27VIS_PIN_SDL (GPIO_PORTE + 5) +#define MX27VIS_PIN_SDR (GPIO_PORTF + 7) + +static int mx27vis_amp_gain; +static int mx27vis_amp_mute; + +static const int mx27vis_amp_pins[] = { + MX27VIS_PIN_G0 | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_G1 | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_SDL | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_SDR | GPIO_GPIO | GPIO_OUT, +}; + static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -74,8 +95,60 @@ static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { .hw_params = mx27vis_aic32x4_hw_params, }; +static int mx27vis_amp_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int value = ucontrol->value.integer.value[0]; + unsigned int reg = mc->reg; + int max = mc->max; + + if (value > max) + return -EINVAL; + + switch (reg) { + case MX27VIS_AMP_GAIN: + gpio_set_value(MX27VIS_PIN_G0, value & 1); + gpio_set_value(MX27VIS_PIN_G1, value >> 1); + mx27vis_amp_gain = value; + break; + case MX27VIS_AMP_MUTE: + gpio_set_value(MX27VIS_PIN_SDL, value & 1); + gpio_set_value(MX27VIS_PIN_SDR, value >> 1); + mx27vis_amp_mute = value; + break; + } + return 0; +} + +static int mx27vis_amp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + switch (reg) { + case MX27VIS_AMP_GAIN: + ucontrol->value.integer.value[0] = mx27vis_amp_gain; + break; + case MX27VIS_AMP_MUTE: + ucontrol->value.integer.value[0] = mx27vis_amp_mute; + break; + } + return 0; +} + +/* From 6dB to 24dB in steps of 6dB */ +static const DECLARE_TLV_DB_SCALE(mx27vis_amp_tlv, 600, 600, 0); + static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = { SOC_DAPM_PIN_SWITCH("External Mic"), + SOC_SINGLE_EXT_TLV("LO Ext Boost", MX27VIS_AMP_GAIN, 0, 3, 0, + mx27vis_amp_get, mx27vis_amp_set, mx27vis_amp_tlv), + SOC_DOUBLE_EXT("LO Ext Mute Switch", MX27VIS_AMP_MUTE, 0, 1, 1, 0, + mx27vis_amp_get, mx27vis_amp_set), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { @@ -146,6 +219,13 @@ static int __init mx27vis_aic32x4_init(void) MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) ); + ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, + ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); + if (ret) { + printk(KERN_ERR "ASoC: unable to setup gpios\n"); + platform_device_put(mx27vis_aic32x4_snd_device); + } + return ret; } -- 1.7.0.4

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Re: [alsa-devel] [PATCH] ASoC: mxs: Fix mxs-saif timeout
by Mark Brown 20 Jan '12

20 Jan '12

On Thu, Jan 19, 2012 at 10:23:22AM -0200, Fabio Estevam wrote: > On a mx28evk board the following errors happens on mxs-sgtl5000 probe: Applied, thanks!

1 0

[alsa-devel] [PATCH] ASoC: Enable line out VMID buffer for single ended line outputs in wm_hubs
by Mark Brown 20 Jan '12

20 Jan '12

For optimal performance the single ended line outputs require that the line output VMID buffer be enabled. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2a61094..9ccc416 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -613,6 +613,8 @@ SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("LINEOUT_VMID_BUF", WM8993_ANTIPOP1, 7, 0, NULL, 0), + SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0, @@ -834,9 +836,11 @@ static const struct snd_soc_dapm_route lineout1_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout1_se_routes[] = { + { "LINEOUT1N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" }, + { "LINEOUT1P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" }, @@ -853,9 +857,11 @@ static const struct snd_soc_dapm_route lineout2_diff_routes[] = { }; static const struct snd_soc_dapm_route lineout2_se_routes[] = { + { "LINEOUT2N Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" }, { "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" }, + { "LINEOUT2P Mixer", NULL, "LINEOUT_VMID_BUF" }, { "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" }, { "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" }, -- 1.7.7.3

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[alsa-devel] Improve hw_params documentation -- details of constraint resolution
by Joerg-Cyril.Hoehle＠t-systems.com 20 Jan '12

20 Jan '12

Hi, Andrew Eikum recently improved the (still much lacking) ALSA documentation -- thanks Andrew: + * When a parameter is set or restricted using a snd_pcm_hw_params_set* + * function, all of the other ranges will be updated to exclude as many + * impossible configurations as possible. Attempting to set a parameter + * outside of its acceptable range will result in the function failing + * and an error code being returned. I'd like to check whether that's exactly what's happening. 1. Consider the following scenario: set_hw_params_period_near([in,out]10ms,[out]dir) yields 16ms and direction=0 <=> "exact match" set_hw_params_buffer_size(X) set_hw_params() So far, I thought that if ALSA replied with direction=0, then that corner stone of the configuration space was fixed. OTOH, if constraint resolutions occurs at the time of set_hw_params() *again and finally*, ALSA may realize that the particularly large or small buffer size X is not reachable with the 16ms period, whereas a 50ms or 5ms period may allow buffer size X. So what is the outcome? a) 16ms period and error from set_hw_params() because the constraints don't fit? b) revised 5 or 50ms period with buffer size X, because 5 or 50ms is a near period? c) 16ms period, error from set_buffer_size and success from set_hw_params as ALSA chooses an arbitrary buffer size (typically 4-8 periods)? I believe c) is implemented. The documentation may lead one to believe b) happens. Behaviour b) may better suit what users expect, but compatibility likely imposes c). Any "official" statement? 2. + * Attempting to set a parameter + * outside of its acceptable range will result in the function failing + * and an error code being returned. That does not seem to apply to the set_*_near() functions. set_hw_params_period_time(2ms) may fail because it's too small for the device at hand, whereas set_hw_params_period_time_near(2ms) causes the smallest supported period to be chosen. Does this need clarification or is using *_near not "attempting to set a parameter"? *_min and *_max appear to behave likewise: they set lower and upper bounds without checking whether these are reachable. I think the current behaviour, namely not checking whether near, min or max fall into the "acceptable range", is useful. Thank you for your help, Jörg Höhle

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Re: [alsa-devel] [PATCH] snd-hda-intel: better Alienware M17x R3 quirk
by Takashi Iwai 20 Jan '12

20 Jan '12

At Thu, 19 Jan 2012 22:08:50 +0100, Albert Pool wrote: > > I have been told that this way the rear headphone connector is > working as well; with model=alienware only laptop speakers work. > The subsystem of both controller and codec is 1028:0490. > > Signed-off-by: Albert Pool <albertpool(a)solcon.nl> Thanks, applied now. Takashi > --- > sound/pci/hda/patch_sigmatel.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c > index 87e684f..4071354 100644 > --- a/sound/pci/hda/patch_sigmatel.c > +++ b/sound/pci/hda/patch_sigmatel.c > @@ -1608,7 +1608,7 @@ static const struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = { > SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x043a, > "Alienware M17x", STAC_ALIENWARE_M17X), > SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, > - "Alienware M17x", STAC_ALIENWARE_M17X), > + "Alienware M17x R3", STAC_DELL_EQ), > {} /* terminator */ > }; > > -- > 1.7.4.1 > >

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[alsa-devel] [PATCH v2 0/2] patch set about the MXS-DMA
by Huang Shijie 20 Jan '12

20 Jan '12

patch 1: move the mxs dma header to a more common place. Beside the mx23/mx28, the apbh-dma is also used by MX50 and MX6Q. So move the mxs dma header file to a more common place, and rename it to mxs-dma.h patch 2: rewrite the last parameter of mxs_dma_prep_slave_sg(). In the new GPMI version, some hardware behavior changes. The WAIT4END bit should be set in the middle one of the ECC READ page DMA chain. The DMA chain should be set like this: +-----+ +-----+ +-----+ | cmd | ------------> | cmd | ------------------> | cmd | +-----+ +-----+ +-----+ ^ ^ | | | | set WAIT4END here too set WAIT4END here So rewrite the last parameter of mxs_dma_prep_slave_sg(), use the dma_ctrl_flags to set the proper DMA bit for DMA Command Structure. And change the relative drivers with the dma_ctrl_flags. TEST: Tested the mxs-mmc driver and gpmi-nand drive with the two patches in MX28 board. v1 --> v2: [1] squash the trial patches into single patch. [2] use the dma_ctrl_flags. Huang Shijie (2): mxs-dma : move the mxs dma.h to a more common place mxs-dma : rewrite the last parameter of mxs_dma_prep_slave_sg() drivers/dma/mxs-dma.c | 34 +++++++++++++++++--- drivers/mmc/host/mxs-mmc.c | 12 +++--- drivers/mtd/nand/gpmi-nand/gpmi-lib.c | 28 +++++++++++++--- drivers/mtd/nand/gpmi-nand/gpmi-nand.h | 6 +++- drivers/mtd/nand/gpmi-nand/gpmi-regs.h | 2 + .../include/mach/dma.h => include/linux/mxs-dma.h | 0 sound/soc/mxs/mxs-pcm.c | 2 +- sound/soc/mxs/mxs-pcm.h | 2 +- sound/soc/mxs/mxs-saif.c | 2 +- 9 files changed, 67 insertions(+), 21 deletions(-) rename arch/arm/mach-mxs/include/mach/dma.h => include/linux/mxs-dma.h (100%) -- 1.7.3.2

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[alsa-devel] acer ferrari one hda realtek - attempt to improve forte mic IC support
by Alban Browaeys 19 Jan '12

19 Jan '12

A lot of reports about aspire one issues lead me to think the mic IC processing is the same as on this acer ferrari one 200. Looking at alsa-driver-aspire_one-v1.09 assembly I see that it setup the bdl entries to point to a capture buffer which it then split in two buffers : namely mic and speaker . This split happens by moving the first 32 bits to mic , the next 32 to speaker , then to mic and so forth. azx_pcm_hw formats is set to S16 LE. I was wondering about the the 32 bits step between mic and speaker buffer and ended up on azx_pcm_preapre where for the capture case snd_hda_calc_stream_format is hardcoded a number of channels of 4. My issue is I attempted the same trick and it seems I still miss something . Ie the azx_dev->format passed in azx_setup_controller to SD_FORMAT registry is incremented by 3 namely 0x33 . Though I still miss half of each samples same as when I was passing runtime->channels ie 2 and 0x31. The codec is ALC272X ATI SB. Best regards, Alban Browaeys

1 0

[alsa-devel] [alsa-lib PATCH] Document strdup allocation in snd_card_get_name family
by Andrew Eikum 19 Jan '12

19 Jan '12

This was spotted by Jörg Höhle on the wine-patches ML. Signed-off-by: Andrew Eikum <aeikum(a)codeweavers.com> --- I noticed my documentation patches haven't been making their way into alsa-lib.git. Am I submitting these wrong? I added the SOB line above, although I notice not all patches that get into ALSA have it. On the other hand, alsa-lib.git hasn't been changed since November so perhaps my patches are sitting in someone's tree waiting to be pushed :) Please let me know if I can help get these in. src/control/cards.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-)

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[alsa-devel] [PATCH v6] ASoC: Apply msbits constraint for sample size bigger than the msbits
by Peter Ujfalusi 19 Jan '12

19 Jan '12

In order to avoid confusing the applications with msbits bigger than the selected sample size apply the msbits constraint only to sample size bigger than the requested msbits. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Let's try this way. I think this implementation is quete clear for the first look to understand what it is doing. Regards, Peter sound/soc/soc-pcm.c | 23 +++++++++++++++-------- 1 files changed, 15 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8bb1793..d6fdde2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -68,7 +68,7 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, * like the DAC/ADC resolution to use but there isn't right now. */ static int sample_sizes[] = { - 8, 16, 24, 32, + 32, 24, 16, 8, }; static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, @@ -85,13 +85,20 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, return; for (i = 0; i < ARRAY_SIZE(sample_sizes); i++) { - ret = snd_pcm_hw_constraint_msbits(substream->runtime, - 0, sample_sizes[i], - bits); - if (ret != 0) - dev_warn(dai->dev, - "Failed to set MSB %d/%d: %d\n", - bits, sample_sizes[i], ret); + if (sample_sizes[i] > bits) { + ret = snd_pcm_hw_constraint_msbits(substream->runtime, + 0, sample_sizes[i], bits); + if (ret != 0) + dev_warn(dai->dev, + "Failed to set MSB %d/%d: %d\n", + bits, sample_sizes[i], ret); + } else { + /* + * The remaining sample sizes are smaller than the + * requested msbits, we can bail out from the loop. + */ + break; + } } } -- 1.7.8.3

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[alsa-devel] [PATCH] ASoC: Don't tell applications about msbits unless we're ignoring input
by Mark Brown 19 Jan '12

19 Jan '12

On the off chance that an application both pays attention and gets confused. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 77c40ee..9f6e55b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -85,9 +85,11 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, return; for (i = 0; i < ARRAY_SIZE(sample_sizes); i++) { - ret = snd_pcm_hw_constraint_msbits(substream->runtime, - 0, sample_sizes[i], - bits); + if (bits >= sample_sizes) + continue; + + ret = snd_pcm_hw_constraint_msbits(substream->runtime, 0, + sample_sizes[i], bits); if (ret != 0) dev_warn(dai->dev, "Failed to set MSB %d/%d: %d\n", -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: Add external amplifier controls for Visstrim_M10.
by Javier Martin 19 Jan '12

19 Jan '12

Visstrim_M10 has an external class D amplifier. This patch provides support for controlling the 4 possible gain levels and per channel muting. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/mx27vis-aic32x4.c | 72 +++++++++++++++++++++++++++++++++++++++ 1 files changed, 72 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index d37e23c..47a7bf9 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -25,16 +25,36 @@ #include <linux/moduleparam.h> #include <linux/device.h> #include <linux/i2c.h> +#include <linux/gpio.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> #include <mach/audmux.h> +#include <mach/iomux-mx27.h> #include "../codecs/tlv320aic32x4.h" #include "imx-ssi.h" +#define MX27VIS_AMP_GAIN 0 +#define MX27VIS_AMP_MUTE 1 + +#define MX27VIS_PIN_G0 (GPIO_PORTF + 9) +#define MX27VIS_PIN_G1 (GPIO_PORTF + 8) +#define MX27VIS_PIN_SDL (GPIO_PORTE + 5) +#define MX27VIS_PIN_SDR (GPIO_PORTF + 7) + +static int mx27vis_amp_gain; +static int mx27vis_amp_mute; + +static const int mx27vis_amp_pins[] = { + MX27VIS_PIN_G0 | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_G1 | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_SDL | GPIO_GPIO | GPIO_OUT, + MX27VIS_PIN_SDR | GPIO_GPIO | GPIO_OUT, +}; + static int mx27vis_aic32x4_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -74,8 +94,53 @@ static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { .hw_params = mx27vis_aic32x4_hw_params, }; +static int mx27vis_amp_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int value = ucontrol->value.integer.value[0]; + unsigned int reg = mc->reg; + + switch (reg) { + case MX27VIS_AMP_GAIN: + gpio_set_value(MX27VIS_PIN_G0, value & 1); + gpio_set_value(MX27VIS_PIN_G1, value >> 1); + mx27vis_amp_gain = value; + break; + case MX27VIS_AMP_MUTE: + gpio_set_value(MX27VIS_PIN_SDL, value & 1); + gpio_set_value(MX27VIS_PIN_SDR, value >> 1); + mx27vis_amp_mute = value; + break; + } + return 0; +} + +static int mx27vis_amp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + switch (reg) { + case MX27VIS_AMP_GAIN: + ucontrol->value.integer.value[0] = mx27vis_amp_gain; + break; + case MX27VIS_AMP_MUTE: + ucontrol->value.integer.value[0] = mx27vis_amp_mute; + break; + } + return 0; +} + static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = { SOC_DAPM_PIN_SWITCH("External Mic"), + SOC_SINGLE_EXT("LO Ext Boost", MX27VIS_AMP_GAIN, 0, 3, 0, + mx27vis_amp_get, mx27vis_amp_set), + SOC_DOUBLE_EXT("LO Ext Mute Switch", MX27VIS_AMP_MUTE, 0, 1, 1, 0, + mx27vis_amp_get, mx27vis_amp_set), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { @@ -146,6 +211,13 @@ static int __init mx27vis_aic32x4_init(void) MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) ); + ret = mxc_gpio_setup_multiple_pins(mx27vis_amp_pins, + ARRAY_SIZE(mx27vis_amp_pins), "MX27VIS_AMP"); + if (ret) { + printk(KERN_ERR "ASoC: unable to setup gpios\n"); + platform_device_put(mx27vis_aic32x4_snd_device); + } + return ret; } -- 1.7.0.4

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[alsa-devel] [PATCH] Fix bint check version check
by Ozan Çağlayan 19 Jan '12

19 Jan '12

It seems that the bint stuff is not a part of 3.2 kernel as it's pulled by Linus only a few days ago, so increment version check to 3.3.0: commit 69116f279a9eaf4c540934269342d9149538fc79 Author: Rusty Russell <rusty(a)rustcorp.com.au> Date: Fri Jan 13 09:32:17 2012 +1030 module_param: avoid bool abuse, add bint for special cases. Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- include/adriver.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/include/adriver.h b/include/adriver.h index a6e6cbc..9bc8ca8 100644 --- a/include/adriver.h +++ b/include/adriver.h @@ -830,7 +830,7 @@ static inline unsigned long msecs_to_jiffies(const unsigned int m) #endif #endif -#if LINUX_VERSION_CODE <= KERNEL_VERSION(3, 2, 0) +#if LINUX_VERSION_CODE < KERNEL_VERSION(3, 3, 0) #include <linux/moduleparam.h> #ifndef param_get_bint #define bint bool -- 1.7.7.1

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[alsa-devel] [PATCH v5] ASoC: Apply msbits constraint for sample size bigger than the msbits
by Peter Ujfalusi 19 Jan '12

19 Jan '12

In order to avoid confusing the applications with msbits bigger than the selected sample size apply the msbits constraint only to sample seze bigger than the requested msbits. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, Converted to use while loop. We run the loop as long as the sample size is bigger than the requested msbits. Most of the drivers require 24/32 configuration, so not point of looping for smaller sample sizes. Regards, Peter sound/soc/soc-pcm.c | 17 +++++++++-------- 1 files changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8bb1793..ef41a39 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -68,13 +68,14 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, * like the DAC/ADC resolution to use but there isn't right now. */ static int sample_sizes[] = { - 8, 16, 24, 32, + 32, 24, 16, 8, }; static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - int ret, i, bits; + int ret, bits; + int i = 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) bits = dai->driver->playback.sig_bits; @@ -84,14 +85,14 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, if (!bits) return; - for (i = 0; i < ARRAY_SIZE(sample_sizes); i++) { - ret = snd_pcm_hw_constraint_msbits(substream->runtime, - 0, sample_sizes[i], - bits); + /* Apply constraint only for sample size bigger than requested msbits */ + while (sample_sizes[i] > bits && i < ARRAY_SIZE(sample_sizes)) { + ret = snd_pcm_hw_constraint_msbits(substream->runtime, 0, + sample_sizes[i], bits); if (ret != 0) - dev_warn(dai->dev, - "Failed to set MSB %d/%d: %d\n", + dev_warn(dai->dev, "Failed to set MSB %d/%d: %d\n", bits, sample_sizes[i], ret); + i++; } } -- 1.7.8.3

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Re: [alsa-devel] [PATCH 01/10] MXS-DMA : move the mxs-dma.h to a more common place
by Shawn Guo 19 Jan '12

19 Jan '12

On Thu, Jan 19, 2012 at 02:15:58PM +0800, Huang Shijie wrote: > Move the header to a more common place. > The DMA engine is not only used in mx23/mx28, but also used > in mx50/mx6q. > It will also be used in the future chips. > > Rename it to mxs-dma.h > > Signed-off-by: Huang Shijie <b32955(a)freescale.com> > --- > arch/arm/mach-mxs/include/mach/dma.h | 28 ---------------------------- > include/linux/mxs-dma.h | 28 ++++++++++++++++++++++++++++ > 2 files changed, 28 insertions(+), 28 deletions(-) > delete mode 100644 arch/arm/mach-mxs/include/mach/dma.h > create mode 100644 include/linux/mxs-dma.h > I'm wondering if it would be better to create folder include/linux/fsl for mxs-dma.h accommodation, so that any headers that need to shared between mxs and imx later can find a place. Considering that Freescale has many IPs shared between MPCxxx (PowerPC) and i.MX (ARM) families, it could also be a good place for any headers that need to be shared between these two product lines from Freescale. -- Regards, Shawn

1 0

[alsa-devel] [PATCH] ASoC: Disable register synchronisation for low frequency WM8996 SYSCLK
by Mark Brown 19 Jan '12

19 Jan '12

With a low frequency SYSCLK and a fast I2C clock register synchronisation may occasionally take too long to take effect, causing I/O issues. Disable synchronisation in order to avoid any issues. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 4 ++++ sound/soc/codecs/wm8996.h | 4 ++++ 2 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 691ae4e..8e8f8d1 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2007,6 +2007,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int lfclk = 0; int ratediv = 0; + int sync = WM8996_REG_SYNC; int src; int old; @@ -2051,6 +2052,7 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, case 32000: case 32768: lfclk = WM8996_LFCLK_ENA; + sync = 0; break; default: dev_warn(codec->dev, "Unsupported clock rate %dHz\n", @@ -2064,6 +2066,8 @@ static int wm8996_set_sysclk(struct snd_soc_dai *dai, WM8996_SYSCLK_SRC_MASK | WM8996_SYSCLK_DIV_MASK, src << WM8996_SYSCLK_SRC_SHIFT | ratediv); snd_soc_update_bits(codec, WM8996_CLOCKING_1, WM8996_LFCLK_ENA, lfclk); + snd_soc_update_bits(codec, WM8996_CONTROL_INTERFACE_1, + WM8996_REG_SYNC, sync); snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_1, WM8996_SYSCLK_ENA, old); diff --git a/sound/soc/codecs/wm8996.h b/sound/soc/codecs/wm8996.h index 0fde643..de9ac3e 100644 --- a/sound/soc/codecs/wm8996.h +++ b/sound/soc/codecs/wm8996.h @@ -1567,6 +1567,10 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, /* * R257 (0x101) - Control Interface (1) */ +#define WM8996_REG_SYNC 0x8000 /* REG_SYNC */ +#define WM8996_REG_SYNC_MASK 0x8000 /* REG_SYNC */ +#define WM8996_REG_SYNC_SHIFT 15 /* REG_SYNC */ +#define WM8996_REG_SYNC_WIDTH 1 /* REG_SYNC */ #define WM8996_AUTO_INC 0x0004 /* AUTO_INC */ #define WM8996_AUTO_INC_MASK 0x0004 /* AUTO_INC */ #define WM8996_AUTO_INC_SHIFT 2 /* AUTO_INC */ -- 1.7.7.3

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[alsa-devel] Realtek model parser problem with the new jack detection
by David Henningsson 19 Jan '12

19 Jan '12

Hi Takashi, I'm troubleshooting a problem with the Realtek quirk/model parsers. The problem is basically that add_jack_kctls call snd_hda_jack_detect_enable, which in turn overwrites the current unsol tag. E g, first the model parser sets a pin to AC_USRSP_EN | ALC_HP_EVENT, then comes snd_hda_jack_detect_enable and does AC_USRSP_EN | jack->tag. I'm not sure of the best way to resolve this; either keep the current tag (and mark all jacks as dirty on an unsol event), or read back the current tag and set it as jack->action (and then always use the jack table). What do you think? Example alsa-info for this is in http://www.alsa-project.org/db/?f=5cc88967e9da547975924a06c948ece00f1f0263 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] Suggestion for change in HD-Audio-Models.txt documentation
by Dave Lentz 19 Jan '12

19 Jan '12

See attachment. Conexant model names have been updated.

2 1

Re: [alsa-devel] [PATCH 01/10] MXS-DMA : move the mxs-dma.h to a more common place
by Wolfram Sang 19 Jan '12

19 Jan '12

On Thu, Jan 19, 2012 at 02:15:58PM +0800, Huang Shijie wrote: > Move the header to a more common place. > The DMA engine is not only used in mx23/mx28, but also used > in mx50/mx6q. > It will also be used in the future chips. > > Rename it to mxs-dma.h > > Signed-off-by: Huang Shijie <b32955(a)freescale.com> > --- > arch/arm/mach-mxs/include/mach/dma.h | 28 ---------------------------- > include/linux/mxs-dma.h | 28 ++++++++++++++++++++++++++++ > 2 files changed, 28 insertions(+), 28 deletions(-) > delete mode 100644 arch/arm/mach-mxs/include/mach/dma.h > create mode 100644 include/linux/mxs-dma.h Please use -M with git format-patch, so we can better see that it is a pure rename. I'd also suggest to squash patches up to 05/10 into this one, (collecting all the proper acks from maintainers) since the changes are trivial and it will reduce dependencies and temporary build-failures. Regards, Wolfram -- Pengutronix e.K. | Wolfram Sang | Industrial Linux Solutions | http://www.pengutronix.de/ |

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[alsa-devel] [PATCH] ASoC: add driver for max9768 amplifier
by Wolfram Sang 19 Jan '12

19 Jan '12

Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> --- include/sound/max9768.h | 23 ++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9768.c | 244 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 273 insertions(+), 0 deletions(-) create mode 100644 include/sound/max9768.h create mode 100644 sound/soc/codecs/max9768.c diff --git a/include/sound/max9768.h b/include/sound/max9768.h new file mode 100644 index 0000000..f8f1639 --- /dev/null +++ b/include/sound/max9768.h @@ -0,0 +1,23 @@ +/* + * Platform data for MAX9768 + * Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K. + * same licence as the driver + */ + +#ifndef __SOUND_MAX9768_PDATA_H__ +#define __SOUND_MAX9768_PDATA_H__ + +/** + * struct max9768_pdata - optional platform specific MAX9768 configuration + * @shdn_gpio: GPIO to SHDN pin. If not valid, pin must be hardwired HIGH + * @mute_gpio: GPIO to MUTE pin. If not valid, control for mute won't be added + * @flags: configuration flags, e.g. set classic PWM mode. + */ +struct max9768_pdata { + int shdn_gpio; + int mute_gpio; + unsigned flags; +#define MAX9768_FLAG_CLASSIC_PWM (1 << 0) +}; + +#endif /* __SOUND_MAX9768_PDATA_H__*/ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7c205e7..a861014 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DFBMCS320 select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C + select SND_SOC_MAX9768 if I2C select SND_SOC_MAX98088 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C @@ -425,6 +426,9 @@ config SND_SOC_WM9713 config SND_SOC_LM4857 tristate +config SND_SOC_MAX9768 + tristate + config SND_SOC_MAX9877 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index de80781..cf3555b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -25,6 +25,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o +snd-soc-max9768-objs := max9768.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o @@ -129,6 +130,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o +obj-$(CONFIG_SND_SOC_MAX9768) += snd-soc-max9768.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c new file mode 100644 index 0000000..bd57b45 --- /dev/null +++ b/sound/soc/codecs/max9768.c @@ -0,0 +1,244 @@ +/* + * MAX9768 AMP driver + * + * Copyright (C) 2011, 2012 by Wolfram Sang, Pengutronix e.K. + * based on lm4857.c by Lars-Peter Clausen et al. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; version 2 of the License. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/max9768.h> + +/* The register offsets in the cache array */ +#define MAX9768_VOL 0 +#define MAX9768_CTRL 3 + +/* Commands */ +#define MAX9768_CTRL_PWM 0x15 +#define MAX9768_CTRL_FILTERLESS 0x16 + +struct max9768 { + struct i2c_client *client; + int mute_gpio; + int shdn_gpio; + u32 flags; +}; + +static const uint8_t max9768_default_regs[] = { + 0x00, 0x00, 0x00, MAX9768_CTRL_FILTERLESS, +}; + +static int max9768_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + uint8_t data; + int ret; + + ret = snd_soc_cache_write(codec, reg, value); + if (ret < 0) + return ret; + + data = (reg << 6) | value; + ret = i2c_smbus_write_byte(codec->control_data, data); + if (ret) + dev_err(codec->dev, "Failed to write register: %d\n", ret); + + return ret; +} + +static unsigned int max9768_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + unsigned int val; + int ret; + + ret = snd_soc_cache_read(codec, reg, &val); + + return ret ?: val; +} + + +static int max9768_get_gpio(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); + int val = gpio_get_value_cansleep(max9768->mute_gpio); + + ucontrol->value.integer.value[0] = !val; + + return 0; +} + +static int max9768_set_gpio(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); + + gpio_set_value_cansleep(max9768->mute_gpio, !ucontrol->value.integer.value[0]); + + return 0; +} + +static const unsigned int volume_tlv[] = { + TLV_DB_RANGE_HEAD(43), + 0, 0, TLV_DB_SCALE_ITEM(-16150, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-9280, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-9030, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(-8680, 0, 0), + 4, 4, TLV_DB_SCALE_ITEM(-8430, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(-8080, 0, 0), + 6, 6, TLV_DB_SCALE_ITEM(-7830, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(-7470, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(-7220, 0, 0), + 9, 9, TLV_DB_SCALE_ITEM(-6870, 0, 0), + 10, 10, TLV_DB_SCALE_ITEM(-6620, 0, 0), + 11, 11, TLV_DB_SCALE_ITEM(-6270, 0, 0), + 12, 12, TLV_DB_SCALE_ITEM(-6020, 0, 0), + 13, 13, TLV_DB_SCALE_ITEM(-5670, 0, 0), + 14, 14, TLV_DB_SCALE_ITEM(-5420, 0, 0), + 15, 17, TLV_DB_SCALE_ITEM(-5060, 250, 0), + 18, 18, TLV_DB_SCALE_ITEM(-4370, 0, 0), + 19, 19, TLV_DB_SCALE_ITEM(-4210, 0, 0), + 20, 20, TLV_DB_SCALE_ITEM(-3960, 0, 0), + 21, 21, TLV_DB_SCALE_ITEM(-3760, 0, 0), + 22, 22, TLV_DB_SCALE_ITEM(-3600, 0, 0), + 23, 23, TLV_DB_SCALE_ITEM(-3340, 0, 0), + 24, 24, TLV_DB_SCALE_ITEM(-3150, 0, 0), + 25, 25, TLV_DB_SCALE_ITEM(-2980, 0, 0), + 26, 26, TLV_DB_SCALE_ITEM(-2720, 0, 0), + 27, 27, TLV_DB_SCALE_ITEM(-2520, 0, 0), + 28, 30, TLV_DB_SCALE_ITEM(-2350, 190, 0), + 31, 31, TLV_DB_SCALE_ITEM(-1750, 0, 0), + 32, 34, TLV_DB_SCALE_ITEM(-1640, 100, 0), + 35, 37, TLV_DB_SCALE_ITEM(-1310, 110, 0), + 38, 39, TLV_DB_SCALE_ITEM(-990, 100, 0), + 40, 40, TLV_DB_SCALE_ITEM(-710, 0, 0), + 41, 41, TLV_DB_SCALE_ITEM(-600, 0, 0), + 42, 42, TLV_DB_SCALE_ITEM(-500, 0, 0), + 43, 43, TLV_DB_SCALE_ITEM(-340, 0, 0), + 44, 44, TLV_DB_SCALE_ITEM(-190, 0, 0), + 45, 45, TLV_DB_SCALE_ITEM(-50, 0, 0), + 46, 46, TLV_DB_SCALE_ITEM(50, 0, 0), + 47, 50, TLV_DB_SCALE_ITEM(120, 40, 0), + 51, 57, TLV_DB_SCALE_ITEM(290, 50, 0), + 58, 58, TLV_DB_SCALE_ITEM(650, 0, 0), + 59, 62, TLV_DB_SCALE_ITEM(700, 60, 0), + 63, 63, TLV_DB_SCALE_ITEM(950, 0, 0), +}; + +static const struct snd_kcontrol_new max9768_volume[] = { + SOC_SINGLE_TLV("Playback Volume", MAX9768_VOL, 0, 63, 0, volume_tlv), +}; + +static const struct snd_kcontrol_new max9768_mute[] = { + SOC_SINGLE_BOOL_EXT("Mute Switch", 0, max9768_get_gpio, max9768_set_gpio), +}; + +static int max9768_probe(struct snd_soc_codec *codec) +{ + struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = max9768->client; + + if (max9768->flags & MAX9768_FLAG_CLASSIC_PWM) { + ret = snd_soc_write(codec, MAX9768_CTRL, MAX9768_CTRL_PWM); + if (ret) + return ret; + } + + ret = snd_soc_add_controls(codec, max9768_volume, + ARRAY_SIZE(max9768_volume)); + if (ret) + return ret; + + if (max9768->mute_gpio >= 0) { + ret = snd_soc_add_controls(codec, max9768_mute, + ARRAY_SIZE(max9768_mute)); + if (ret) + return ret; + } + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_max9768 = { + .write = max9768_write, + .read = max9768_read, + .probe = max9768_probe, + .reg_cache_size = ARRAY_SIZE(max9768_default_regs), + .reg_word_size = sizeof(uint8_t), + .reg_cache_default = max9768_default_regs, +}; + +static int __devinit max9768_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct max9768 *max9768; + struct max9768_pdata *pdata = client->dev.platform_data; + int err; + + max9768 = devm_kzalloc(&client->dev, sizeof(*max9768), GFP_KERNEL); + if (!max9768) + return -ENOMEM; + + if (pdata) { + /* Mute on powerup to avoid clicks */ + err = gpio_request_one(pdata->mute_gpio, GPIOF_INIT_HIGH, "MAX9768 Mute"); + max9768->mute_gpio = err ?: pdata->mute_gpio; + + /* Activate chip, enabling I2C */ + err = gpio_request_one(pdata->shdn_gpio, GPIOF_INIT_HIGH, "MAX9768 Shutdown"); + max9768->shdn_gpio = err ?: pdata->shdn_gpio; + + max9768->flags = pdata->flags; + } else { + max9768->shdn_gpio = -EINVAL; + max9768->mute_gpio = -EINVAL; + } + + max9768->client = client; + i2c_set_clientdata(client, max9768); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_max9768, NULL, 0); +} + +static int __devexit max9768_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max9768_i2c_id[] = { + { "max9768", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9768_i2c_id); + +static struct i2c_driver max9768_i2c_driver = { + .driver = { + .name = "max9768", + .owner = THIS_MODULE, + }, + .probe = max9768_i2c_probe, + .remove = __devexit_p(max9768_i2c_remove), + .id_table = max9768_i2c_id, +}; +module_i2c_driver(max9768_i2c_driver); + +MODULE_AUTHOR("Wolfram Sang <w.sang(a)pengutronix.de>"); +MODULE_DESCRIPTION("ASoC MAX9768 amplifier driver"); +MODULE_LICENSE("GPL v2"); -- 1.7.8.3

3 6

Re: [alsa-devel] [PATCH 00/10] patch set about the MXS-DMA
by Shawn Guo 19 Jan '12

19 Jan '12

On Thu, Jan 19, 2012 at 02:15:57PM +0800, Huang Shijie wrote: > This patch set does two things about the MXS-DMA: > [1] patch 1 ~ patch 5: move the mxs dma header to a more common place. > Beside the mx23/mx28, the APBH-DMA is also used by MX50 and MX6Q. > So move the mxs dma header file to a more common place, and rename > it to mxs-dma.h > For the sake of bitsec, these 5 patches need to be one. > [2] patch 6 ~ patch 10: rewrite the last parameter of mxs_dma_prep_slave_sg(). For the sake of bitsec, at least patches #7 and #8 need to be one patch. That said, if I apply the series and then check out the commit at patch #7, you need to all mxs-dma client drivers, mxs-mmc, gpmi do not break. > In the new GPMI version, some hardware behavior changes. > The WAIT4END bit should be set in the middle one of the ECC READ page DMA chain. > The DMA chain should be set like this: > > +-----+ +-----+ +-----+ > | cmd | ------------> | cmd | ------------------> | cmd | > +-----+ +-----+ +-----+ > ^ ^ > | | > | | > set WAIT4END here too set WAIT4END here > > Please read the comment in patch "MXS-DMA : add more flags for MXS-DMA". > > Huang Shijie (10): > MXS-DMA : move the mxs-dma.h to a more common place > MXS-DMA : change the header > MXS-MMC : change the DMA header file > MTD/GPMI : change the DMA header file > ASoc : change the DMA header file > MXS-DMA : add more flags for MXS-DMA > MXS-DMA : change the last parameter of mxs_dma_prep_slave_sg() > MXS-MMC : use the new DMA flags > MTD/GPMI : add a new field `gpmi_version` > MTD/GPMI : change the code for new DMA interface > Please try to make these subject prefix consistent with the existing ones in the tree. You can get them from 'git log' output. For example "mxs-dma: ..." is better than "MXS-DMA : ...". -- Regards, Shawn

2 1

Re: [alsa-devel] [PATCH 06/10] MXS-DMA : add more flags for MXS-DMA
by Russell King - ARM Linux 19 Jan '12

19 Jan '12

On Thu, Jan 19, 2012 at 02:16:03PM +0800, Huang Shijie wrote: > [1] Background : > The GPMI does ECC read page operation with a DMA chain consist of three DMA > Command Structures. The middle one of the chain is used to enable the BCH, > and read out the NAND page. > > The WAIT4END(wait for command end) is a comunication signal between > the GPMI and MXS-DMA. > > [2] The current DMA code sets the WAIT4END bit at the last one, such as: > > +-----+ +-----+ +-----+ > | cmd | ------------> | cmd | ------------------> | cmd | > +-----+ +-----+ +-----+ > ^ > | > | > set WAIT4END here > > This chain works fine in the mx23/mx28. > > [3] But in the new GPMI version (used in MX50/MX60), the WAIT4END bit should > be set not only at the last DMA Command Structure, > but also at the middle one, such as: > > +-----+ +-----+ +-----+ > | cmd | ------------> | cmd | ------------------> | cmd | > +-----+ +-----+ +-----+ > ^ ^ > | | > | | > set WAIT4END here too set WAIT4END here > > If we do not set WAIT4END, the BCH maybe stall in "ECC reading page" state. > In the next ECC write page operation, a DMA-timeout occurs. > This has been catched in the MX6Q board. > > In order to fix the bug, we should let the driver to > set the proper DMA flags in the DMA command structrues. > > So add the new flags for MXS-DMA. > The driver can use these flags to control the DMA in a more flexible way. > > Signed-off-by: Huang Shijie <b32955(a)freescale.com> > --- > include/linux/mxs-dma.h | 26 ++++++++++++++++++++++++++ > 1 files changed, 26 insertions(+), 0 deletions(-) > > diff --git a/include/linux/mxs-dma.h b/include/linux/mxs-dma.h > index 203d7c4..3ef73b8 100644 > --- a/include/linux/mxs-dma.h > +++ b/include/linux/mxs-dma.h > @@ -11,6 +11,32 @@ > > #include <linux/dmaengine.h> > > +/* > + * The drivers use these flags for ->device_prep_slave_sg() : > + * [1] If there is only one DMA command in the DMA chain, the code should be: > + * ...... > + * ->device_prep_slave_sg(MXS_DMA_F_WAIT4END); > + * ...... > + * [2] If there are two DMA commands in the DMA chain, the code should be > + * ...... > + * ->device_prep_slave_sg(0); > + * ...... > + * ->device_prep_slave_sg(MXS_DMA_F_LASTONE); > + * ...... > + * [3] If there are more than two DMA commands in the DMA chain, the code > + * should be: > + * ...... > + * ->device_prep_slave_sg(0); // First > + * ...... > + * ->device_prep_slave_sg(MXS_DMA_F_APPEND [| MXS_DMA_F_WAIT4END]); > + * ...... > + * ->device_prep_slave_sg(MXS_DMA_F_LASTONE); // Last > + */ > +#define MXS_DMA_F_APPEND (1 << 0) > +#define MXS_DMA_F_WAIT4END (1 << 1) > + > +#define MXS_DMA_F_LASTONE (MXS_DMA_F_APPEND | MXS_DMA_F_WAIT4END) Err, the 'flags' argument to device_prep_slave_sg() is supposed to be from the set of enum dma_ctrl_flags. What this means is that your MXS_DMA_F_APPEND is equivalent to DMA_PREP_INTERRUPT, and MXS_DMA_F_WAIT4END is equivalent to DMA_CTRL_ACK. What this does is make your drivers completely dependent on your DMA engine implementation. That's not a good idea when devices get reused in different SoCs. If you need to supply extra flags which aren't in the dma_ctrl_flags, at least make sure that they're using different bits. For bonus points, also have your driver _check_ the DMA engine it's connected to before it passes these flags.

1 0

[alsa-devel] [PATCH] ASoC: Convert blackfin directory to module_platform_driver
by Axel Lin 19 Jan '12

19 Jan '12

Factor out some boilerplate code. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-ac97.c | 13 +------------ sound/soc/blackfin/bf5xx-i2s-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-i2s.c | 13 +------------ sound/soc/blackfin/bf5xx-tdm-pcm.c | 12 +----------- sound/soc/blackfin/bf5xx-tdm.c | 12 +----------- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +----------- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +----------- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +----------- 9 files changed, 9 insertions(+), 101 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 56815c1..fcff583 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -475,17 +475,7 @@ static struct platform_driver bf5xx_pcm_driver = { .remove = __devexit_p(bf5xx_soc_platform_remove), }; -static int __init snd_bf5xx_pcm_init(void) -{ - return platform_driver_register(&bf5xx_pcm_driver); -} -module_init(snd_bf5xx_pcm_init); - -static void __exit snd_bf5xx_pcm_exit(void) -{ - platform_driver_unregister(&bf5xx_pcm_driver); -} -module_exit(snd_bf5xx_pcm_exit); +module_platform_driver(bf5xx_pcm_driver); MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 6d21625..f4e9dc4 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -375,18 +375,7 @@ static struct platform_driver asoc_bfin_ac97_driver = { .remove = __devexit_p(asoc_bfin_ac97_remove), }; -static int __init bfin_ac97_init(void) -{ - return platform_driver_register(&asoc_bfin_ac97_driver); -} -module_init(bfin_ac97_init); - -static void __exit bfin_ac97_exit(void) -{ - platform_driver_unregister(&asoc_bfin_ac97_driver); -} -module_exit(bfin_ac97_exit); - +module_platform_driver(asoc_bfin_ac97_driver); MODULE_AUTHOR("Roy Huang"); MODULE_DESCRIPTION("AC97 driver for ADI Blackfin"); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 7565e15..6ec3d41 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -314,17 +314,7 @@ static struct platform_driver bfin_i2s_pcm_driver = { .remove = __devexit_p(bfin_i2s_soc_platform_remove), }; -static int __init snd_bfin_i2s_pcm_init(void) -{ - return platform_driver_register(&bfin_i2s_pcm_driver); -} -module_init(snd_bfin_i2s_pcm_init); - -static void __exit snd_bfin_i2s_pcm_exit(void) -{ - platform_driver_unregister(&bfin_i2s_pcm_driver); -} -module_exit(snd_bfin_i2s_pcm_exit); +module_platform_driver(bfin_i2s_pcm_driver); MODULE_AUTHOR("Cliff Cai"); MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 00cc3e0..a400633 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -288,18 +288,7 @@ static struct platform_driver bfin_i2s_driver = { }, }; -static int __init bfin_i2s_init(void) -{ - return platform_driver_register(&bfin_i2s_driver); -} - -static void __exit bfin_i2s_exit(void) -{ - platform_driver_unregister(&bfin_i2s_driver); -} - -module_init(bfin_i2s_init); -module_exit(bfin_i2s_exit); +module_platform_driver(bfin_i2s_driver); /* Module information */ MODULE_AUTHOR("Cliff Cai"); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index c95cc03..4406f9a 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -339,17 +339,7 @@ static struct platform_driver bfin_tdm_driver = { .remove = __devexit_p(bf5xx_soc_platform_remove), }; -static int __init snd_bfin_tdm_init(void) -{ - return platform_driver_register(&bfin_tdm_driver); -} -module_init(snd_bfin_tdm_init); - -static void __exit snd_bfin_tdm_exit(void) -{ - platform_driver_unregister(&bfin_tdm_driver); -} -module_exit(snd_bfin_tdm_exit); +module_platform_driver(bfin_tdm_driver); MODULE_AUTHOR("Barry Song"); MODULE_DESCRIPTION("ADI Blackfin TDM PCM DMA module"); diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index a822d1e..bf65e0e 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -314,17 +314,7 @@ static struct platform_driver bfin_tdm_driver = { }, }; -static int __init bfin_tdm_init(void) -{ - return platform_driver_register(&bfin_tdm_driver); -} -module_init(bfin_tdm_init); - -static void __exit bfin_tdm_exit(void) -{ - platform_driver_unregister(&bfin_tdm_driver); -} -module_exit(bfin_tdm_exit); +module_platform_driver(bfin_tdm_driver); /* Module information */ MODULE_AUTHOR("Barry Song"); diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c index 8df2a3b..85ed39a 100644 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ b/sound/soc/blackfin/bfin-eval-adau1373.c @@ -184,17 +184,7 @@ static struct platform_driver bfin_eval_adau1373_driver = { .remove = __devexit_p(bfin_eval_adau1373_remove), }; -static int __init bfin_eval_adau1373_init(void) -{ - return platform_driver_register(&bfin_eval_adau1373_driver); -} -module_init(bfin_eval_adau1373_init); - -static void __exit bfin_eval_adau1373_exit(void) -{ - platform_driver_unregister(&bfin_eval_adau1373_driver); -} -module_exit(bfin_eval_adau1373_exit); +module_platform_driver(bfin_eval_adau1373_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c index e5550ac..1a88fe9 100644 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ b/sound/soc/blackfin/bfin-eval-adau1701.c @@ -121,17 +121,7 @@ static struct platform_driver bfin_eval_adau1701_driver = { .remove = __devexit_p(bfin_eval_adau1701_remove), }; -static int __init bfin_eval_adau1701_init(void) -{ - return platform_driver_register(&bfin_eval_adau1701_driver); -} -module_init(bfin_eval_adau1701_init); - -static void __exit bfin_eval_adau1701_exit(void) -{ - platform_driver_unregister(&bfin_eval_adau1701_driver); -} -module_exit(bfin_eval_adau1701_exit); +module_platform_driver(bfin_eval_adau1701_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver"); diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c index 897cfa6..0bc995f 100644 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ b/sound/soc/blackfin/bfin-eval-adav80x.c @@ -157,17 +157,7 @@ static struct platform_driver bfin_eval_adav80x_driver = { .id_table = bfin_eval_adav80x_ids, }; -static int __init bfin_eval_adav80x_init(void) -{ - return platform_driver_register(&bfin_eval_adav80x_driver); -} -module_init(bfin_eval_adav80x_init); - -static void __exit bfin_eval_adav80x_exit(void) -{ - platform_driver_unregister(&bfin_eval_adav80x_driver); -} -module_exit(bfin_eval_adav80x_exit); +module_platform_driver(bfin_eval_adav80x_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver"); -- 1.7.5.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: Make WM5100 interrupt path use regmap directly
by Mark Brown 18 Jan '12

18 Jan '12

This will allow us to move the interrupt allocation out of the ASoC part of the driver and simplifies the locking by removing any reliance in the bulk of the interrupt path on the big CODEC lock. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 187 ++++++++++++++++++++++++++------------------- 1 files changed, 107 insertions(+), 80 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index b22b425..02a759d 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -51,6 +51,7 @@ struct wm5100_fll { /* codec private data */ struct wm5100_priv { + struct device *dev; struct regmap *regmap; struct snd_soc_codec *codec; @@ -891,48 +892,48 @@ static int wm5100_dbvdd_ev(struct snd_soc_dapm_widget *w, } } -static void wm5100_log_status3(struct snd_soc_codec *codec, int val) +static void wm5100_log_status3(struct wm5100_priv *wm5100, int val) { if (val & WM5100_SPK_SHUTDOWN_WARN_EINT) - dev_crit(codec->dev, "Speaker shutdown warning\n"); + dev_crit(wm5100->dev, "Speaker shutdown warning\n"); if (val & WM5100_SPK_SHUTDOWN_EINT) - dev_crit(codec->dev, "Speaker shutdown\n"); + dev_crit(wm5100->dev, "Speaker shutdown\n"); if (val & WM5100_CLKGEN_ERR_EINT) - dev_crit(codec->dev, "SYSCLK underclocked\n"); + dev_crit(wm5100->dev, "SYSCLK underclocked\n"); if (val & WM5100_CLKGEN_ERR_ASYNC_EINT) - dev_crit(codec->dev, "ASYNCCLK underclocked\n"); + dev_crit(wm5100->dev, "ASYNCCLK underclocked\n"); } -static void wm5100_log_status4(struct snd_soc_codec *codec, int val) +static void wm5100_log_status4(struct wm5100_priv *wm5100, int val) { if (val & WM5100_AIF3_ERR_EINT) - dev_err(codec->dev, "AIF3 configuration error\n"); + dev_err(wm5100->dev, "AIF3 configuration error\n"); if (val & WM5100_AIF2_ERR_EINT) - dev_err(codec->dev, "AIF2 configuration error\n"); + dev_err(wm5100->dev, "AIF2 configuration error\n"); if (val & WM5100_AIF1_ERR_EINT) - dev_err(codec->dev, "AIF1 configuration error\n"); + dev_err(wm5100->dev, "AIF1 configuration error\n"); if (val & WM5100_CTRLIF_ERR_EINT) - dev_err(codec->dev, "Control interface error\n"); + dev_err(wm5100->dev, "Control interface error\n"); if (val & WM5100_ISRC2_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ISRC2 underclocked\n"); + dev_err(wm5100->dev, "ISRC2 underclocked\n"); if (val & WM5100_ISRC1_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ISRC1 underclocked\n"); + dev_err(wm5100->dev, "ISRC1 underclocked\n"); if (val & WM5100_FX_UNDERCLOCKED_EINT) - dev_err(codec->dev, "FX underclocked\n"); + dev_err(wm5100->dev, "FX underclocked\n"); if (val & WM5100_AIF3_UNDERCLOCKED_EINT) - dev_err(codec->dev, "AIF3 underclocked\n"); + dev_err(wm5100->dev, "AIF3 underclocked\n"); if (val & WM5100_AIF2_UNDERCLOCKED_EINT) - dev_err(codec->dev, "AIF2 underclocked\n"); + dev_err(wm5100->dev, "AIF2 underclocked\n"); if (val & WM5100_AIF1_UNDERCLOCKED_EINT) - dev_err(codec->dev, "AIF1 underclocked\n"); + dev_err(wm5100->dev, "AIF1 underclocked\n"); if (val & WM5100_ASRC_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ASRC underclocked\n"); + dev_err(wm5100->dev, "ASRC underclocked\n"); if (val & WM5100_DAC_UNDERCLOCKED_EINT) - dev_err(codec->dev, "DAC underclocked\n"); + dev_err(wm5100->dev, "DAC underclocked\n"); if (val & WM5100_ADC_UNDERCLOCKED_EINT) - dev_err(codec->dev, "ADC underclocked\n"); + dev_err(wm5100->dev, "ADC underclocked\n"); if (val & WM5100_MIXER_UNDERCLOCKED_EINT) - dev_err(codec->dev, "Mixer underclocked\n"); + dev_err(wm5100->dev, "Mixer underclocked\n"); } static int wm5100_post_ev(struct snd_soc_dapm_widget *w, @@ -940,16 +941,17 @@ static int wm5100_post_ev(struct snd_soc_dapm_widget *w, int event) { struct snd_soc_codec *codec = w->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); int ret; ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_3); ret &= WM5100_SPK_SHUTDOWN_WARN_STS | WM5100_SPK_SHUTDOWN_STS | WM5100_CLKGEN_ERR_STS | WM5100_CLKGEN_ERR_ASYNC_STS; - wm5100_log_status3(codec, ret); + wm5100_log_status3(wm5100, ret); ret = snd_soc_read(codec, WM5100_INTERRUPT_RAW_STATUS_4); - wm5100_log_status4(codec, ret); + wm5100_log_status4(wm5100, ret); return 0; } @@ -2200,55 +2202,59 @@ static int wm5100_dig_vu[] = { WM5100_DAC_DIGITAL_VOLUME_6R, }; -static void wm5100_set_detect_mode(struct snd_soc_codec *codec, int the_mode) +static void wm5100_set_detect_mode(struct wm5100_priv *wm5100, int the_mode) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); struct wm5100_jack_mode *mode = &wm5100->pdata.jack_modes[the_mode]; BUG_ON(the_mode >= ARRAY_SIZE(wm5100->pdata.jack_modes)); gpio_set_value_cansleep(wm5100->pdata.hp_pol, mode->hp_pol); - snd_soc_update_bits(codec, WM5100_ACCESSORY_DETECT_MODE_1, - WM5100_ACCDET_BIAS_SRC_MASK | - WM5100_ACCDET_SRC, - (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | - mode->micd_src << WM5100_ACCDET_SRC_SHIFT); - snd_soc_update_bits(codec, WM5100_MISC_CONTROL, - WM5100_HPCOM_SRC, - mode->micd_src << WM5100_HPCOM_SRC_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_ACCESSORY_DETECT_MODE_1, + WM5100_ACCDET_BIAS_SRC_MASK | + WM5100_ACCDET_SRC, + (mode->bias << WM5100_ACCDET_BIAS_SRC_SHIFT) | + mode->micd_src << WM5100_ACCDET_SRC_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_MISC_CONTROL, + WM5100_HPCOM_SRC, + mode->micd_src << WM5100_HPCOM_SRC_SHIFT); wm5100->jack_mode = the_mode; - dev_dbg(codec->dev, "Set microphone polarity to %d\n", + dev_dbg(wm5100->dev, "Set microphone polarity to %d\n", wm5100->jack_mode); } -static void wm5100_micd_irq(struct snd_soc_codec *codec) +static void wm5100_micd_irq(struct wm5100_priv *wm5100) { - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); - int val; + unsigned int val; + int ret; - val = snd_soc_read(codec, WM5100_MIC_DETECT_3); + ret = regmap_read(wm5100->regmap, WM5100_MIC_DETECT_3, &val); + if (ret != 0) { + dev_err(wm5100->dev, "Failed to read micropone status: %d\n", + ret); + return; + } - dev_dbg(codec->dev, "Microphone event: %x\n", val); + dev_dbg(wm5100->dev, "Microphone event: %x\n", val); if (!(val & WM5100_ACCDET_VALID)) { - dev_warn(codec->dev, "Microphone detection state invalid\n"); + dev_warn(wm5100->dev, "Microphone detection state invalid\n"); return; } /* No accessory, reset everything and report removal */ if (!(val & WM5100_ACCDET_STS)) { - dev_dbg(codec->dev, "Jack removal detected\n"); + dev_dbg(wm5100->dev, "Jack removal detected\n"); wm5100->jack_mic = false; wm5100->jack_detecting = true; snd_soc_jack_report(wm5100->jack, 0, SND_JACK_LINEOUT | SND_JACK_HEADSET | SND_JACK_BTN_0); - snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - WM5100_ACCDET_RATE_MASK); + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + WM5100_ACCDET_RATE_MASK); return; } @@ -2258,7 +2264,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) */ if (val & 0x400) { if (wm5100->jack_detecting) { - dev_dbg(codec->dev, "Microphone detected\n"); + dev_dbg(wm5100->dev, "Microphone detected\n"); wm5100->jack_mic = true; snd_soc_jack_report(wm5100->jack, SND_JACK_HEADSET, @@ -2266,11 +2272,11 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) /* Increase poll rate to give better responsiveness * for buttons */ - snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - 5 << WM5100_ACCDET_RATE_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 5 << WM5100_ACCDET_RATE_SHIFT); } else { - dev_dbg(codec->dev, "Mic button up\n"); + dev_dbg(wm5100->dev, "Mic button up\n"); snd_soc_jack_report(wm5100->jack, 0, SND_JACK_BTN_0); } @@ -2283,7 +2289,7 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) * plain headphones. */ if (wm5100->jack_detecting && (val & 0x3f8)) { - wm5100_set_detect_mode(codec, !wm5100->jack_mode); + wm5100_set_detect_mode(wm5100, !wm5100->jack_mode); return; } @@ -2293,20 +2299,20 @@ static void wm5100_micd_irq(struct snd_soc_codec *codec) */ if (val & 0x3fc) { if (wm5100->jack_mic) { - dev_dbg(codec->dev, "Mic button detected\n"); + dev_dbg(wm5100->dev, "Mic button detected\n"); snd_soc_jack_report(wm5100->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); } else if (wm5100->jack_detecting) { - dev_dbg(codec->dev, "Headphone detected\n"); + dev_dbg(wm5100->dev, "Headphone detected\n"); snd_soc_jack_report(wm5100->jack, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); /* Increase the detection rate a bit for * responsiveness. */ - snd_soc_update_bits(codec, WM5100_MIC_DETECT_1, - WM5100_ACCDET_RATE_MASK, - 7 << WM5100_ACCDET_RATE_SHIFT); + regmap_update_bits(wm5100->regmap, WM5100_MIC_DETECT_1, + WM5100_ACCDET_RATE_MASK, + 7 << WM5100_ACCDET_RATE_SHIFT); } } } @@ -2319,7 +2325,7 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) wm5100->jack = jack; wm5100->jack_detecting = true; - wm5100_set_detect_mode(codec, 0); + wm5100_set_detect_mode(wm5100, 0); /* Slowest detection rate, gives debounce for initial * detection */ @@ -2358,52 +2364,70 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) static irqreturn_t wm5100_irq(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct wm5100_priv *wm5100 = data; irqreturn_t status = IRQ_NONE; - int irq_val; + unsigned int irq_val, mask_val; + int ret; - irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3); - if (irq_val < 0) { - dev_err(codec->dev, "Failed to read IRQ status 3: %d\n", - irq_val); + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_3, &irq_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ status 3: %d\n", + ret); irq_val = 0; } - irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_3_MASK); - snd_soc_write(codec, WM5100_INTERRUPT_STATUS_3, irq_val); + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_3_MASK, + &mask_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ mask 3: %d\n", + ret); + mask_val = 0xffff; + } + + irq_val &= ~mask_val; + + regmap_write(wm5100->regmap, WM5100_INTERRUPT_STATUS_3, irq_val); if (irq_val) status = IRQ_HANDLED; - wm5100_log_status3(codec, irq_val); + wm5100_log_status3(wm5100, irq_val); if (irq_val & WM5100_FLL1_LOCK_EINT) { - dev_dbg(codec->dev, "FLL1 locked\n"); + dev_dbg(wm5100->dev, "FLL1 locked\n"); complete(&wm5100->fll[0].lock); } if (irq_val & WM5100_FLL2_LOCK_EINT) { - dev_dbg(codec->dev, "FLL2 locked\n"); + dev_dbg(wm5100->dev, "FLL2 locked\n"); complete(&wm5100->fll[1].lock); } if (irq_val & WM5100_ACCDET_EINT) - wm5100_micd_irq(codec); + wm5100_micd_irq(wm5100); - irq_val = snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4); - if (irq_val < 0) { - dev_err(codec->dev, "Failed to read IRQ status 4: %d\n", - irq_val); + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_4, &irq_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ status 4: %d\n", + ret); irq_val = 0; } - irq_val &= ~snd_soc_read(codec, WM5100_INTERRUPT_STATUS_4_MASK); + + ret = regmap_read(wm5100->regmap, WM5100_INTERRUPT_STATUS_4_MASK, + &mask_val); + if (ret < 0) { + dev_err(wm5100->dev, "Failed to read IRQ mask 4: %d\n", + ret); + mask_val = 0xffff; + } + + irq_val &= ~mask_val; if (irq_val) status = IRQ_HANDLED; - snd_soc_write(codec, WM5100_INTERRUPT_STATUS_4, irq_val); + regmap_write(wm5100->regmap, WM5100_INTERRUPT_STATUS_4, irq_val); - wm5100_log_status4(codec, irq_val); + wm5100_log_status4(wm5100, irq_val); return status; } @@ -2612,11 +2636,12 @@ static int wm5100_probe(struct snd_soc_codec *codec) if (irq_flags & (IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING)) ret = request_threaded_irq(i2c->irq, NULL, - wm5100_edge_irq, - irq_flags, "wm5100", codec); + wm5100_edge_irq, irq_flags, + "wm5100", wm5100); else ret = request_threaded_irq(i2c->irq, NULL, wm5100_irq, - irq_flags, "wm5100", codec); + irq_flags, "wm5100", + wm5100); if (ret != 0) { dev_err(codec->dev, "Failed to request IRQ %d: %d\n", @@ -2679,7 +2704,7 @@ static int wm5100_probe(struct snd_soc_codec *codec) err_gpio: if (i2c->irq) - free_irq(i2c->irq, codec); + free_irq(i2c->irq, wm5100); for (i = 0; i < ARRAY_SIZE(wm5100->adsp1); i++) wm_adsp1_exit(wm5100->adsp1[i]); @@ -2698,7 +2723,7 @@ static int wm5100_remove(struct snd_soc_codec *codec) } if (i2c->irq) - free_irq(i2c->irq, codec); + free_irq(i2c->irq, wm5100); for (i = 0; i < ARRAY_SIZE(wm5100->adsp1); i++) wm_adsp1_exit(wm5100->adsp1[i]); @@ -2758,6 +2783,8 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, if (wm5100 == NULL) return -ENOMEM; + wm5100->dev = &i2c->dev; + wm5100->regmap = regmap_init_i2c(i2c, &wm5100_regmap); if (IS_ERR(wm5100->regmap)) { ret = PTR_ERR(wm5100->regmap); -- 1.7.7.3

1 1

[alsa-devel] [PATCH] ASoC: Remove redundant set_bias_level() from WM5100 remove()
by Mark Brown 18 Jan '12

18 Jan '12

The framework should bring the device down before it calls the driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index fca7f0a..b22b425 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2693,8 +2693,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) struct i2c_client *i2c = to_i2c_client(codec->dev); int i; - wm5100_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm5100->pdata.hp_pol) { gpio_free(wm5100->pdata.hp_pol); } -- 1.7.7.3

1 0

[alsa-devel] [PATCH 1/2] ASoC: tlv320aic32x4: always enable dividers
by Wolfram Sang 18 Jan '12

18 Jan '12

Dividers (such as MDAC) are always needed, independent of the codec being I2S master or slave. Needed on a custom board where the codec has to be slave. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> --- sound/soc/codecs/tlv320aic32x4.c | 102 +++++++++++++++++--------------------- 1 files changed, 46 insertions(+), 56 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index eb401ef..3806cb6 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -60,7 +60,6 @@ struct aic32x4_rate_divs { struct aic32x4_priv { u32 sysclk; - s32 master; u8 page_no; void *control_data; u32 power_cfg; @@ -369,7 +368,6 @@ static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u8 iface_reg_1; u8 iface_reg_2; u8 iface_reg_3; @@ -384,11 +382,9 @@ static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - aic32x4->master = 1; iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER; break; case SND_SOC_DAIFMT_CBS_CFS: - aic32x4->master = 0; break; default: printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n"); @@ -526,64 +522,58 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute) static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - switch (level) { case SND_SOC_BIAS_ON: - if (aic32x4->master) { - /* Switch on PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, AIC32X4_PLLEN); - - /* Switch on NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, AIC32X4_NDACEN); - - /* Switch on MDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_MDAC, - AIC32X4_MDACEN, AIC32X4_MDACEN); - - /* Switch on NADC Divider */ - snd_soc_update_bits(codec, AIC32X4_NADC, - AIC32X4_NADCEN, AIC32X4_NADCEN); - - /* Switch on MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, AIC32X4_MADCEN); - - /* Switch on BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, AIC32X4_BCLKEN); - } + /* Switch on PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, AIC32X4_PLLEN); + + /* Switch on NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, AIC32X4_NDACEN); + + /* Switch on MDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, AIC32X4_MDACEN); + + /* Switch on NADC Divider */ + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, AIC32X4_NADCEN); + + /* Switch on MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, AIC32X4_MADCEN); + + /* Switch on BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, AIC32X4_BCLKEN); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (aic32x4->master) { - /* Switch off PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, 0); - - /* Switch off NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, 0); - - /* Switch off MDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_MDAC, - AIC32X4_MDACEN, 0); - - /* Switch off NADC Divider */ - snd_soc_update_bits(codec, AIC32X4_NADC, - AIC32X4_NADCEN, 0); - - /* Switch off MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, 0); - - /* Switch off BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 0); - } + /* Switch off PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); + + /* Switch off NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); + + /* Switch off MDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, 0); + + /* Switch off NADC Divider */ + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, 0); + + /* Switch off MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); + + /* Switch off BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); break; case SND_SOC_BIAS_OFF: break; -- 1.7.8.3

3 7

[alsa-devel] [PATCH] ASoC: da7210: Add support for PLL and SRM
by Ashish Chavan 18 Jan '12

18 Jan '12

Hi, Current DA7210 driver does support PLL mode fully. It uses fixed value of input master clock and PLL mode is enabled and disabled based on the sampling frequency being used for playback or recording. It also doesn't support Sample Rate Measurement feature of DA7210 hardware. This patch adds full support for PLL and SRM. Basically following three modes of operation are possible for DA7210 hardware, (1) I2S SLAVE mode with PLL bypassed (2) I2S SLAVE mode with PLL enabled (3) I2S Master mode with PLL enabled This patch adds support for all three modes. Also, in case of SLAVE mode with PLL, it supports SRM (Sample Rate Measurement) feature of the chip. This patch has been tested on DA7210 EVB with Samsung SMDK6410 board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 293 ++++++++++++++++++++++++++++++++++++++------- 1 files changed, 250 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index ab38e93..3945d65 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -145,7 +145,12 @@ #define DA7210_DAI_EN (1 << 7) /*PLL_DIV3 bit fields */ +#define DA7210_PLL_DIV_L_MASK (0xF << 0) +#define DA7210_MCLK_RANGE_MASK (0x30 << 0) +#define DA7210_MCLK_RANGE_32768_HZ (0 << 4) #define DA7210_MCLK_RANGE_10_20_MHZ (1 << 4) +#define DA7210_MCLK_RANGE_20_40_MHZ (2 << 4) +#define DA7210_MCLK_RANGE_40_80_MHZ (3 << 4) #define DA7210_PLL_BYP (1 << 6) /* PLL bit fields */ @@ -161,12 +166,16 @@ #define DA7210_PLL_FS_48000 (0xB << 0) #define DA7210_PLL_FS_88200 (0xE << 0) #define DA7210_PLL_FS_96000 (0xF << 0) +#define DA7210_MCLK_DET_EN (0x1 << 5) +#define DA7210_MCLK_SRM_EN (0x1 << 6) #define DA7210_PLL_EN (0x1 << 7) /* SOFTMUTE bit fields */ #define DA7210_RAMP_EN (1 << 6) /* CONTROL bit fields */ +#define DA7210_REG_EN (1 << 0) +#define DA7210_BIAS_EN (1 << 2) #define DA7210_NOISE_SUP_EN (1 << 3) /* IN_GAIN bit fields */ @@ -205,6 +214,47 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* PLL Master mode dividers */ +static const u8 da7210_pll_master_div[] = { + 0xE8, 0x6C, 0x2, /* MCLK=12Mhz Fs=44.1Khz */ + 0xF3, 0x12, 0x7, /* MCLK=12Mhz Fs=48Khz */ + 0xDF, 0x28, 0xC, /* MCLK=13Mhz Fs=44.1Khz */ + 0xE8, 0xFD, 0x5, /* MCLK=13Mhz Fs=48Khz */ + 0xDB, 0x0A, 0xD, /* MCLK=13.5Mhz Fs=44.1Khz */ + 0xE4, 0x82, 0x3, /* MCLK=13.5Mhz Fs=48Khz */ + 0xD4, 0x5A, 0x2, /* MCLK=14.4Mhz Fs=44.1Khz */ + 0xDD, 0x3A, 0x0, /* MCLK=14.4Mhz Fs=48Khz */ + 0xBB, 0x43, 0x9, /* MCLK=19.2Mhz Fs=44.1Khz */ + 0xC1, 0xEB, 0x8, /* MCLK=19.2Mhz Fs=48Khz */ + 0xB9, 0x6D, 0xA, /* MCLK=19.68Mhz Fs=44.1Khz */ + 0xBF, 0xEC, 0x0, /* MCLK=19.68Mhz Fs=48Khz */ + 0xB8, 0xFB, 0xB, /* MCLK=19.8Mhz Fs=44.1Khz */ + 0xBF, 0x70, 0x0, /* MCLK=19.8Mhz Fs=48Khz */ +}; + +/* PLL SRM mode dividers */ +static const u8 da7210_pll_srm_div[] = { + 0xED, 0xBF, 0x5, /* MCLK=12Mhz */ + 0xE4, 0x13, 0x0, /* MCLK=13Mhz */ + 0xDF, 0xC6, 0x8, /* MCLK=13.5Mhz */ + 0xD8, 0xCA, 0x1, /* MCLK=14.4Mhz */ + 0xBE, 0x97, 0x9, /* MCLK=19.2Mhz */ + 0xBC, 0xAC, 0xD, /* MCLK=19.68Mhz */ + 0xBC, 0x35, 0xE, /* MCLK=19.8Mhz */ +}; + +/* Constants for MCLK range */ +enum da7210_mclk_range { + MCLK_32768_HZ, + MCLK_10_20_MHZ, + MCLK_20_40_MHZ, + MCLK_40_80_MHZ, +}; + +enum clk_src { + DA7210_CLKSRC_MCLK +}; + #define DA7210_VERSION "0.0.1" /* @@ -627,6 +677,8 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Codec private data */ struct da7210_priv { enum snd_soc_control_type control_type; + unsigned int mclk_rate; + int master; }; /* @@ -673,8 +725,9 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; - u32 fs, bypass; + u32 fs; /* set DAI source to Left and Right ADC */ snd_soc_write(codec, DA7210_DAI_SRC_SEL, @@ -707,43 +760,33 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, switch (params_rate(params)) { case 8000: fs = DA7210_PLL_FS_8000; - bypass = DA7210_PLL_BYP; break; case 11025: fs = DA7210_PLL_FS_11025; - bypass = 0; break; case 12000: fs = DA7210_PLL_FS_12000; - bypass = DA7210_PLL_BYP; break; case 16000: fs = DA7210_PLL_FS_16000; - bypass = DA7210_PLL_BYP; break; case 22050: fs = DA7210_PLL_FS_22050; - bypass = 0; break; case 32000: fs = DA7210_PLL_FS_32000; - bypass = DA7210_PLL_BYP; break; case 44100: fs = DA7210_PLL_FS_44100; - bypass = 0; break; case 48000: fs = DA7210_PLL_FS_48000; - bypass = DA7210_PLL_BYP; break; case 88200: fs = DA7210_PLL_FS_88200; - bypass = 0; break; case 96000: fs = DA7210_PLL_FS_96000; - bypass = DA7210_PLL_BYP; break; default: return -EINVAL; @@ -753,8 +796,15 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs); - snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass); + if (da7210->mclk_rate) { + /* PLL mode, disable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, 0); + } else { + /* PLL bypass mode, enable PLL bypass */ + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, + DA7210_PLL_BYP); + } /* Enable active mode */ snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, DA7210_SC_MST_EN); @@ -768,6 +818,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) { struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); u32 dai_cfg1; u32 dai_cfg3; @@ -776,9 +827,11 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: + da7210->master = 1; dai_cfg1 |= DA7210_DAI_MODE_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: + da7210->master = 0; dai_cfg1 |= DA7210_DAI_MODE_SLAVE; break; default: @@ -830,10 +883,190 @@ static int da7210_mute(struct snd_soc_dai *dai, int mute) #define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) +static int da7210_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case DA7210_CLKSRC_MCLK: + switch (freq) { + case 12000000: + case 13000000: + case 13500000: + case 14400000: + case 19200000: + case 19680000: + case 19800000: + da7210->mclk_rate = freq; + return 0; + default: + dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", + freq); + return -EINVAL; + } + break; + default: + dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id); + return -EINVAL; + } +} + +/** + * da7210_set_dai_clkdiv: Set codec input clock divider + * @param codec_dai : pointer to codec DAI + * @param div_id : DA7210 has only one clk_div, so div_id is always zero + * @param div : Divider value (configured via control reg MCLK_RANGE) + * @return int : Zero for success, negative error code for error + * + */ +static int da7210_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mclk_range = 0; + + /* + * Select proper MCLK range in PLL DIV3 register. + * Bits[4:5] decides this. + * Bit[5:4] MCLK Range + * 0:0 - 32.768 KHz + * 0:1 - 10 to 20 MHz + * 1:0 - 20 to 40 MHz + * 1:1 - 40 to 80 MHz + */ + switch (div) { + case MCLK_32768_HZ: + mclk_range = DA7210_MCLK_RANGE_32768_HZ; + break; + case MCLK_10_20_MHZ: + mclk_range = DA7210_MCLK_RANGE_10_20_MHZ; + break; + case MCLK_20_40_MHZ: + mclk_range = DA7210_MCLK_RANGE_20_40_MHZ; + break; + case MCLK_40_80_MHZ: + mclk_range = DA7210_MCLK_RANGE_40_80_MHZ; + break; + default: + dev_err(codec_dai->dev, "Unsupported MCLK range %d\n", div); + return -EINVAL; + } + snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_MCLK_RANGE_MASK, + mclk_range); + return 0; +} + +/** + * da7210_set_dai_pll :Configure the codec PLL + * @param codec_dai : pointer to codec DAI + * @param pll_id : da7210 has only one pll, so pll_id is always zero + * @param fref : MCLK frequency, should be < 20MHz + * @param fout : FsDM value, Refer page 44 & 45 of datasheet + * @return int : Zero for success, negative error code for error + * + * Note: Supported PLL input frequencies are 12MHz, 13MHz, 13.5MHz, 14.4MHz, + * 19.2MHz, 19.6MHz and 19.8MHz + */ +static int da7210_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int fref, unsigned int fout) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); + + u8 pll_div1, pll_div2, pll_div3; + int pll_div_offset = 0, fsdm_offset = 0, mclk_offset = 0; + + switch (fref) { + case 12000000: + mclk_offset = 0; + break; + case 13000000: + mclk_offset = 1; + break; + case 13500000: + mclk_offset = 2; + break; + case 14400000: + mclk_offset = 3; + break; + case 19200000: + mclk_offset = 4; + break; + case 19680000: + mclk_offset = 5; + break; + case 19800000: + mclk_offset = 6; + break; + default: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", + fref); + return -EINVAL; + } + + switch (fout) { + case 2822400: + fsdm_offset = 0; + break; + case 3072000: + fsdm_offset = 1; + break; + default: + dev_err(codec_dai->dev, "Unsupported PLL input frequency %d\n", + fout); + return -EINVAL; + } + + if (da7210->master) { + /* In PLL master mode, use Master mode PLL dividers */ + pll_div_offset = (mclk_offset * 2 + fsdm_offset) * 3; + pll_div1 = da7210_pll_master_div[pll_div_offset]; + pll_div2 = da7210_pll_master_div[pll_div_offset + 1]; + pll_div3 = da7210_pll_master_div[pll_div_offset + 2]; + + } else { + /* In PLL slave mode, use SRM mode PLL dividers */ + pll_div_offset = (mclk_offset * 3); + pll_div1 = da7210_pll_srm_div[pll_div_offset]; + pll_div2 = da7210_pll_srm_div[pll_div_offset + 1]; + pll_div3 = da7210_pll_srm_div[pll_div_offset + 2]; + } + + /* Disable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0); + /* Write PLL dividers */ + snd_soc_write(codec, DA7210_PLL_DIV1, pll_div1); + snd_soc_write(codec, DA7210_PLL_DIV2, pll_div2); + snd_soc_update_bits(codec, DA7210_PLL_DIV3, + DA7210_PLL_DIV_L_MASK, pll_div3); + + if (da7210->master) { + /* In master mode, no need to enable SRM */ + snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, + DA7210_PLL_EN); + } else { + /* In slave mode, enable SRM and PLL */ + snd_soc_update_bits(codec, DA7210_PLL, + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN), + (DA7210_PLL_EN | DA7210_MCLK_SRM_EN | + DA7210_MCLK_DET_EN)); + } + /* Enable active mode */ + snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, + DA7210_SC_MST_EN); + return 0; +} + /* DAI operations */ static const struct snd_soc_dai_ops da7210_dai_ops = { .hw_params = da7210_hw_params, .set_fmt = da7210_set_dai_fmt, + .set_sysclk = da7210_set_dai_sysclk, + .set_clkdiv = da7210_set_dai_clkdiv, + .set_pll = da7210_set_dai_pll, .digital_mute = da7210_mute, }; @@ -872,24 +1105,13 @@ static int da7210_probe(struct snd_soc_codec *codec) return ret; } - /* FIXME - * - * This driver use fixed value here - * And below settings expects MCLK = 12.288MHz - * - * When you select different MCLK, please check... - * DA7210_PLL_DIV1 val - * DA7210_PLL_DIV2 val - * DA7210_PLL_DIV3 val - * DA7210_PLL_DIV3 :: DA7210_MCLK_RANGExxx - */ + da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ + da7210->master = 0; /* This will be set from set_fmt() */ + + /* Enable internal regulator & bias current */ + snd_soc_write(codec, DA7210_CONTROL, DA7210_REG_EN | DA7210_BIAS_EN); - /* - * make sure that DA7210 use bypass mode before start up - */ snd_soc_write(codec, DA7210_STARTUP1, 0); - snd_soc_write(codec, DA7210_PLL_DIV3, - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); /* * ADC settings @@ -967,21 +1189,6 @@ static int da7210_probe(struct snd_soc_codec *codec) /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); - /* - * If 48kHz sound came, it use bypass mode, - * and when it is 44.1kHz, it use PLL. - * - * This time, this driver sets PLL always ON - * and controls bypass/PLL mode by switching - * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. - * see da7210_hw_params - */ - snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ - snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); - snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | - DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); - snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); - /* As suggested by Dialog */ snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); -- 1.7.1

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Re: [alsa-devel] [PATCH 4/4] snd-usb-6fire: add analog input volume control
by Torsten Schenk 18 Jan '12

18 Jan '12

> > static struct __devinitdata snd_kcontrol_new elements[] = { > > @@ -442,6 +497,15 @@ > > .get = usb6fire_control_digital_thru_get, > > .put = usb6fire_control_digital_thru_put > > }, > > + { > > + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > > + .name = "Input 1/2 Capture Volume", > > + .index = 0, > > If this is the only volume element, you don't need to add "1/2", no? The problem is, that there is only an input control for 2 out of 4 analog input channels since channel 3 and 4 use a different ADC and there doesn't seem to be a volume control on that chip. Since this control only changes volume of channel 1 and 2, I wanted to have an indication about that. Thanks, Torsten

2 2

[alsa-devel] [PATCH v4 0/5] ASoC: updates for core support for msbits constraints
by Peter Ujfalusi 18 Jan '12

18 Jan '12

Hi Mark, Changes from v3: - no need to initialize ret to zero in soc_pcm_apply_msb. The function is void. - created against for-3.4 branch (leaving out already applied patches) Intro mail for v1: I would prefer to have the msbits constraints to be applied to only sample sizes bigger than the requested msbits from dais. I'm resending your patch plus the modification which will make the possible. I have also converted the drivers I know have this constraint. If you prefer, you can squash the first two patch. Regards, Peter --- Peter Ujfalusi (5): ASoC: Apply msbits constraint for sample size bigger than the msbits ASoC: omap-dmic: Use core to set the msbits constraint ASoC: omap-mcpdm: Set 24msbits constraint ASoC: tlv320dac33: Use core to set the msbits constraint ASoC: twl4030: Use core to set the msbits constraint sound/soc/codecs/tlv320dac33.c | 3 +-- sound/soc/codecs/twl4030.c | 7 ++++--- sound/soc/omap/omap-dmic.c | 7 +++---- sound/soc/omap/omap-mcpdm.c | 2 ++ sound/soc/soc-pcm.c | 4 +++- 5 files changed, 13 insertions(+), 10 deletions(-) -- 1.7.8.3

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Re: [alsa-devel] [PATCH v2] ASoC: sgtl5000: Print revision number in hex
by Mark Brown 18 Jan '12

18 Jan '12

On Mon, Jan 16, 2012 at 05:32:21PM -0200, Fabio Estevam wrote: > Throughout the sgtl5000 driver source code and also in the sgtl5000 datasheet > the revision code is shown in hexadecimal. > > Print it hex format, for consistency. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix wrong register name in restore
by Mark Brown 18 Jan '12

18 Jan '12

On Wed, Jan 18, 2012 at 01:58:07PM +0800, zengzm.kernel(a)gmail.com wrote: > From: Zeng Zhaoming <zengzm.kernel(a)gmail.com> > > Correct SGTL5000_CHIP_CLK_CTRL to SGTL5000_CHIP_REF_CTRL in > sgtl5000_restore_regs(), and add comment to explain the > restore order. Applied, thanks.

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[alsa-devel] [PATCH v3 0/7] ASoC: Add support in core to set msbits constraint
by Peter Ujfalusi 18 Jan '12

18 Jan '12

Hi Mark, Changes from v2: - initialize ret to zero in soc_pcm_apply_msb I would prefer to have the msbits constraints to be applied to only sample sizes bigger than the requested msbits from dais. I'm resending your patch plus the modification which will make the possible. I have also converted the drivers I know have this constraint. If you prefer, you can squash the first two patch. Regards, Peter --- Mark Brown (2): ASoC: Allow drivers to specify how many bits are significant on a DAI ASoC: 24 bits are significant on the WM8996 audio interfaces Peter Ujfalusi (5): ASoC: Apply msbits constraint for sample size bigger than the msbits ASoC: omap-dmic: Use core to set the msbits constraint ASoC: omap-mcpdm: Set 24msbits constraint ASoC: tlv320dac33: Use core to set the msbits constraint ASoC: twl4030: Use core to set the msbits constraint include/sound/soc.h | 1 + sound/soc/codecs/tlv320dac33.c | 3 +-- sound/soc/codecs/twl4030.c | 7 ++++--- sound/soc/codecs/wm8996.c | 4 ++++ sound/soc/omap/omap-dmic.c | 7 +++---- sound/soc/omap/omap-mcpdm.c | 2 ++ sound/soc/soc-pcm.c | 39 +++++++++++++++++++++++++++++++++++++++ 7 files changed, 54 insertions(+), 9 deletions(-) -- 1.7.8.3

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[alsa-devel] [PATCH v2 0/7] ASoC: Add support in core to set msbits constraint
by Peter Ujfalusi 18 Jan '12

18 Jan '12

Hi Mark, I would prefer to have the msbits constraints to be applied to only sample sizes bigger than the requested msbits from dais. I'm resending your patch plus the modification which will make the possible. I have also converted the drivers I know have this constraint. If you prefer, you can squash the first two patch. Regards, Peter --- Mark Brown (2): ASoC: Allow drivers to specify how many bits are significant on a DAI ASoC: 24 bits are significant on the WM8996 audio interfaces Peter Ujfalusi (5): ASoC: Apply msbits constraint for sample size bigger than the msbits ASoC: omap-dmic: Use core to set the msbits constraint ASoC: omap-mcpdm: Set 24msbits constraint ASoC: tlv320dac33: Use core to set the msbits constraint ASoC: twl4030: Use core to set the msbits constraint include/sound/soc.h | 1 + sound/soc/codecs/tlv320dac33.c | 3 +-- sound/soc/codecs/twl4030.c | 7 ++++--- sound/soc/codecs/wm8996.c | 4 ++++ sound/soc/omap/omap-dmic.c | 7 +++---- sound/soc/omap/omap-mcpdm.c | 2 ++ sound/soc/soc-pcm.c | 38 ++++++++++++++++++++++++++++++++++++++ 7 files changed, 53 insertions(+), 9 deletions(-) -- 1.7.8.3

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[alsa-devel] [PATCH 0/7] extract a simple dmaengine library from shdma.c
by Guennadi Liakhovetski 18 Jan '12

18 Jan '12

This patch series has been triggered by a recent series by Shimoda-san http://www.spinics.net/lists/linux-sh/index.html#09895 In a follow up to that thread it has been decided to extract a common library from the current shdma driver to be re-used by other dmaengine drivers. Primarily this library provides a flexible transfer descriptor management for hardware, that does not dupport scatter-gather lists natively. In such cases this library can be used to split transfer requests into smaller chunks, up to the size, supported by the specific hardware, enter them onto a linked list, thack their completion and call user callbacks. This is a first shot, aimed to be used by the SUDMAC driver. Once it is confirmed, that this version can be conveniently used with it, I'll do some cosmetic clean up, notably, add / improve documentation:-) Thanks Guennadi --- Guennadi Liakhovetski, Ph.D. Freelance Open-Source Software Developer http://www.open-technology.de/

1 1

Re: [alsa-devel] [PATCH v2] ASoC: sgtl5000: Fix wrong register name in restore
by Dong Aisheng-B29396 18 Jan '12

18 Jan '12

> -----Original Message----- > From: zengzm.kernel(a)gmail.com [mailto:zengzm.kernel@gmail.com] > Sent: Wednesday, January 18, 2012 1:58 PM > To: alsa-devel(a)alsa-project.org > Cc: Dong Aisheng-B29396; broonie(a)opensource.wolfsonmicro.com; > julia.lawall(a)lip6.fr; festevam(a)gmail.com; w.sang(a)pengutronix.de; > shawn.guo(a)linaro.org; lrg(a)ti.com; zengzm.kernel(a)gmail.com > Subject: [PATCH v2] ASoC: sgtl5000: Fix wrong register name in restore > Importance: High > > From: Zeng Zhaoming <zengzm.kernel(a)gmail.com> > > Correct SGTL5000_CHIP_CLK_CTRL to SGTL5000_CHIP_REF_CTRL in > sgtl5000_restore_regs(), and add comment to explain the restore order. > > Reported-by: Julia Lawall <julia.lawall(a)lip6.fr> > Signed-off-by: Zeng Zhaoming <zengzm.kernel(a)gmail.com> > --- > sound/soc/codecs/sgtl5000.c | 17 +++++++++++++---- > 1 files changed, 13 insertions(+), 4 deletions(-) > Acked-by: Dong Aisheng <dong.aisheng(a)linaro.org> Regards Dong Aisheng

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[alsa-devel] Hardware-specific IOCTLs
by Ivan Bublikov 18 Jan '12

18 Jan '12

Message body Hello All, My question is about how to allocate a new hardware-specific IOCTL code. For example, the HDA driver consists of common HDA code and vendor-specific codes for various HDA codecs. As an HDA codec vendor, I want to be able to control some functionality in my codec, that is not present in codecs from other vendors. For this, I want to implement a special IOCTL in my vendor-specific code. I understand that I can register an IOCTL handler using snd_ctl_register_ioctl. My difficulty is how to define a code for my IOCTL and avoid a conflict with the common code in the future. Looking at header files, I see common "ALSA" audio IOCTLs starting with letter 'A', and "HWDEP" IOCTLs starting with letter 'H'. Some of H-IOCTLs are defined in asound.h, starting at offset 0, and some in hda_hwdep.h and sb16_csp.h starting at offset 0x10. My guess is that the latter start at 0x10 to give some expansion room for H-IOCTLs in asound.h...? I think I need some offset range in H-IOCTLs or a new IOCTL family with another letter, officially reserved for such codec-specific IOCTLs. Any suggestions? Best regards, Ivan

2 3

[alsa-devel] ASoC updates for 3.3
by Mark Brown 17 Jan '12

17 Jan '12

The following changes since commit 29abceb67f8a230da806db4ed73899595bd2ae76: Merge branch 'for-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into topic/asoc (2012-01-12 09:48:20 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 All small, easily reviewed fixes. Javier Martin (1): ASoC: Fix DMA channel leak in imx-pcm-dma-mx2 driver. Liam Girdwood (2): ASoC: dapm - Fix check for codec context in dapm_power_widgets(). ASoC: core - Free platform DAPM context at platform removal. Mark Brown (1): ASoC: Wait for WM8993 FLL to stabilise Zeng Zhaoming (1): ASoC: sgtl5000: update author email address sound/soc/codecs/sgtl5000.c | 2 +- sound/soc/codecs/wm8993.c | 6 ++++++ sound/soc/imx/imx-pcm-dma-mx2.c | 12 +++++++----- sound/soc/soc-core.c | 4 ++++ sound/soc/soc-dapm.c | 2 +- 5 files changed, 19 insertions(+), 7 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: Say we can't read WM5100 ID register
by Mark Brown 17 Jan '12

17 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 0e38a24..fca7f0a 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2839,7 +2839,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, ret = regmap_read(wm5100->regmap, WM5100_SOFTWARE_RESET, &reg); if (ret < 0) { - dev_err(&i2c->dev, "Failed to read ID register\n"); + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); goto err_reset; } switch (reg) { -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: When releasing WM5100 put /RESET into reset
by Mark Brown 17 Jan '12

17 Jan '12

Reset is active low, make sure we leave it asserted when release the device. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 9995d14..0e38a24 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2901,7 +2901,7 @@ static __devinit int wm5100_i2c_probe(struct i2c_client *i2c, err_reset: wm5100_free_gpio(i2c); if (wm5100->pdata.reset) { - gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_set_value_cansleep(wm5100->pdata.reset, 0); gpio_free(wm5100->pdata.reset); } err_ldo: @@ -2934,7 +2934,7 @@ static __devexit int wm5100_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); wm5100_free_gpio(client); if (wm5100->pdata.reset) { - gpio_set_value_cansleep(wm5100->pdata.reset, 1); + gpio_set_value_cansleep(wm5100->pdata.reset, 0); gpio_free(wm5100->pdata.reset); } if (wm5100->pdata.ldo_ena) { -- 1.7.7.3

1 0

[alsa-devel] [PATCH 1/2] ASoC: Allow drivers to specify how many bits are significant on a DAI
by Mark Brown 17 Jan '12

17 Jan '12

Most devices accept data in formats that don't correspond directly to their internal format. ALSA allows us to set a msbits constraint which tells userspace about this in case it finds it useful (for example, in order to avoid wasting effort dithering bits that will be ignored when raising the sample size of data) so provide a mechanism for drivers to specify the number of bits that are actually significant on a DAI and add the appropriate constraints along with all the others. This is done slightly awkwardly as the constraint is specified per sample size - we loop over every possible sample size, including ones that the device doesn't support and including ones that have fewer bits than are actually used, but this is harmless as the upper layers do the right thing in these cases. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 36 ++++++++++++++++++++++++++++++++++++ 2 files changed, 37 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0a56767..346a458 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -519,6 +519,7 @@ struct snd_soc_pcm_stream { unsigned int rate_max; /* max rate */ unsigned int channels_min; /* min channels */ unsigned int channels_max; /* max channels */ + unsigned int sig_bits; /* number of bits of content */ }; /* SoC audio ops */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cdc860a..8bb1793 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -63,6 +63,39 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, } /* + * List of sample sizes that might go over the bus for parameter + * application. There ought to be a wildcard sample size for things + * like the DAC/ADC resolution to use but there isn't right now. + */ +static int sample_sizes[] = { + 8, 16, 24, 32, +}; + +static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret, i, bits; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + bits = dai->driver->playback.sig_bits; + else + bits = dai->driver->capture.sig_bits; + + if (!bits) + return; + + for (i = 0; i < ARRAY_SIZE(sample_sizes); i++) { + ret = snd_pcm_hw_constraint_msbits(substream->runtime, + 0, sample_sizes[i], + bits); + if (ret != 0) + dev_warn(dai->dev, + "Failed to set MSB %d/%d: %d\n", + bits, sample_sizes[i], ret); + } +} + +/* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls * startup for the cpu DAI, platform, machine and codec DAI. @@ -187,6 +220,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto config_err; } + soc_pcm_apply_msb(substream, codec_dai); + soc_pcm_apply_msb(substream, cpu_dai); + /* Symmetry only applies if we've already got an active stream. */ if (cpu_dai->active) { ret = soc_pcm_apply_symmetry(substream, cpu_dai); -- 1.7.7.3

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[alsa-devel] sound devices USBPre2 gets 'cannot submit syncpipe for urb 0' error when run in usb2 mode
by alexander 17 Jan '12

17 Jan '12

I've understood that linux infact does have support for uac2 so this device should work in usb2 mode but currently doesn't. I've made an alsa bug report with more info. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5512

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[alsa-devel] [PATCH 1/2] ASoC: Wait for WM8993 FLL to stabilise
by Mark Brown 17 Jan '12

17 Jan '12

Ensure the FLL is locked before we return from set_fll(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 36562ed..69f5491 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -542,6 +542,12 @@ static int _wm8993_set_fll(struct snd_soc_codec *codec, int fll_id, int source, /* Enable the FLL */ snd_soc_write(codec, WM8993_FLL_CONTROL_1, reg1 | WM8993_FLL_ENA); + /* Both overestimates */ + if (Fref < 1000000) + msleep(3); + else + msleep(1); + dev_dbg(codec->dev, "FLL enabled at %dHz->%dHz\n", Fref, Fout); wm8993->fll_fref = Fref; -- 1.7.7.3

1 1

Re: [alsa-devel] [Alsa-user] Read and write with different output sizes
by Elias Kokkinis 17 Jan '12

17 Jan '12

Dear all, after some further research into the matter, it seems I have to do some asynchronous capture/playback with ALSA. I have searched for examples and found this site: http://alumnos.elo.utfsm.cl/~yanez/alsa-sample-programs/ where two example programs are included. microphone.c and speaker.c which use almost all available input/output modes in ALSA. The interesting thing is that you can run both programs concatenated since they write and read to the standard output and create a simple audio loop-back. I have tested different configurations and the configuration I want works with direct and asynchronous read/write with mmap and read/write with poll. Now I have taken the bits and pieces of those to programs and I have tried to create a single program that will read and write with different buffer sizes. I can't seem to be able to "reroute" the write from the standard output [these programs use write() and read()] to a buffer, or the read from the standard input to a buffer. Any ideas? Comments? I thank everyone for their time, Elias On 16 Jan 2012, at 12:53, Elias Kokkinis wrote: > Hello all, > > I am posting a question I previously posted in alsa-devel, but did not get the expected feedback. > > I am using alsa drivers in an OMAP3530 processor to do some audio processing. I am implementing a typical speech enhancement algorithm with overlap-add. I use a window length of 2048 samples and an overlap of 1024 samples. That means I need to use an input (capture) buffer of 2048 and an output (playback) buffer of 1024. Can this be configured in alsa? Can the playback and capture streams have different buffer sizes, but operate at the same sample rate (e.g. 44100Hz)? > > Best regards, > Elias > ------------------------------------------------------------------------------ > RSA(R) Conference 2012 > Mar 27 - Feb 2 > Save $400 by Jan. 27 > Register now! > http://p.sf.net/sfu/rsa-sfdev2dev2 > _______________________________________________ > Alsa-user mailing list > Alsa-user(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user

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[alsa-devel] [PATCH 0/2] ASoC: omap-mcbsp: CPU version check changes
by Peter Ujfalusi 17 Jan '12

17 Jan '12

Hello, The following patches changes the way we check CPU versions to enable different features (FIFO, and some register differencies). In this way we are not going to need to change (and bloat) the checks for future versions of OMAP. Regards, Peter --- Peter Ujfalusi (2): ASoC: omap-mcbsp: Simplify checks for McBSP IP with FIFO ASoC: omap-mcbsp: Use future proof OMAP version checks for configuration sound/soc/omap/omap-mcbsp.c | 15 ++++++++------- 1 files changed, 8 insertions(+), 7 deletions(-) -- 1.7.8.3

3 5

[alsa-devel] QasTools 0.17.0 released
by Sebastian H. 17 Jan '12

17 Jan '12

QasTools version 0.17.0 is available. QasTools is a collection of dektop applications for ALSA. Changes: * all: setting names for state save/restore changed/regrouped * all: smart main window sizeHint() to get a useable window size after leaving fullscreen * all: new common info dialog - uses QListView page selection intead of tabs * shared: device selection style tweaks: selected item now emphasized by a thin frame * shared: device selection style tweaks: Rounded selection area * shared: use QSplitter instead of QDockwidget for device selection view. It couldn't be detached anyway and size restoring works more reliable. * qasconfig: settings directory changed to shared ~/.config/qastools/ (was ~/.config/qasconfig/) * qasconfig: new basic command line options (--version, --help, etc.) * qashctl: new basic command line options (--version, --help, etc.) * qasmixer: new application icon Homepage with more information http://xwmw.org/qastools Project page http://sourceforge.net/projects/qastools/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] [PATCH] ASoC: Make WM8993 I2C usage unconditional
by Mark Brown 17 Jan '12

17 Jan '12

The WM8993 only supports I2C so don't ifdef the I2C support in the driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index b64fb91..36562ed 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1654,7 +1654,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8993 = { .set_bias_level = wm8993_set_bias_level, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8993_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1758,27 +1757,22 @@ static struct i2c_driver wm8993_i2c_driver = { .remove = __devexit_p(wm8993_i2c_remove), .id_table = wm8993_i2c_id, }; -#endif static int __init wm8993_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8993_i2c_driver); if (ret != 0) { pr_err("WM8993: Unable to register I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8993_modinit); static void __exit wm8993_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8993_i2c_driver); -#endif } module_exit(wm8993_exit); -- 1.7.7.3

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[alsa-devel] Unable to see ADC output data with ADAV801
by Wayne Tams 17 Jan '12

17 Jan '12

Hi, I am attempting to add ADAV801 codec support to my 2.6.35 kernel to be used with an ARM processor. I can see that there is a mainlined adav80x.c driver that appears at kernel version 3.1, I have had a quick look at back-porting the driver but this looks as though it could get quite messy. However, I did come across an older ADAV801 driver which looks a lot simpler, but I cannot find it in previous mainlined kernel versions, http://www.spinics.net/lists/alsa-devel/msg36465.html, so I thought I would use this as a basis for my own driver. _From what I understand from the datasheet and the way the older driver has configured the registers on the ADAV801 I should be able to see some data on the record outputs, however this is not the case. Could anyone confirm whether or not this driver dated August 2010 is OK to use? Or perhaps some advice on areas to look at that may be preventing the CODEC from manipulating and outputting the digital conversion. Best regards Wayne

1 0

[alsa-devel] [PATCH] ASoC: Fix DMA channel leak in imx-pcm-dma-mx2 driver.
by Javier Martin 17 Jan '12

17 Jan '12

"snd_imx_pcm_hw_params" callback can be called several times by the user (i.e. OSS emulation) leading to a DMA channel leak. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/imx-pcm-dma-mx2.c | 12 +++++++----- 1 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 43fdc24..d2bb1cb 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -88,11 +88,13 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, iprtd->dma_data.dma_request = dma_params->dma; /* Try to grab a DMA channel */ - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); - iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); - if (!iprtd->dma_chan) - return -EINVAL; + if (!iprtd->dma_chan) { + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + iprtd->dma_chan = dma_request_channel(mask, filter, iprtd); + if (!iprtd->dma_chan) + return -EINVAL; + } switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: -- 1.7.0.4

2 1

[alsa-devel] [PATCH] ASoC: Add WM8962 DAC and ADC L/R swap support
by Mark Brown 17 Jan '12

17 Jan '12

Simple switches since there's no per-channel control. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 11125c9..e35237a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2223,6 +2223,8 @@ SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8962_LEFT_DAC_VOLUME, WM8962_RIGHT_DAC_VOLUME, 1, 127, 0, digital_tlv), SOC_SINGLE("DAC High Performance Switch", WM8962_ADC_DAC_CONTROL_2, 0, 1, 0), +SOC_SINGLE("DAC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 5, 1, 0), +SOC_SINGLE("ADC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 8, 1, 0), SOC_SINGLE("ADC High Performance Switch", WM8962_ADDITIONAL_CONTROL_1, 5, 1, 0), -- 1.7.7.3

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[alsa-devel] [PATCH] sound: memalloc.c: Fix build warnings
by Fabio Estevam 16 Jan '12

16 Jan '12

Fix the following build warnings: sound/core/memalloc.c: In function ‘snd_mem_proc_write’: sound/core/memalloc.c:426: warning: ‘vendor’ may be used uninitialized in this function sound/core/memalloc.c:426: warning: ‘device’ may be used uninitialized in this function sound/core/memalloc.c:426: warning: ‘size’ may be used uninitialized in this function sound/core/memalloc.c:426: warning: ‘buffers’ may be used uninitialized in this function sound/core/memalloc.c:427: warning: ‘mask’ may be used uninitialized in this function Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/core/memalloc.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 6915692..7dbad80 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -423,8 +423,11 @@ static ssize_t snd_mem_proc_write(struct file *file, const char __user * buffer, return count; if (strcmp(token, "add") == 0) { char *endp; - int vendor, device, size, buffers; - long mask; + int vendor = 0; + int device = 0; + int size = 0; + int buffers = 0; + long mask = 0; int i, alloced; struct pci_dev *pci; -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: Optimise performance of WM8904 ADC 128fs OSR mode
by Mark Brown 16 Jan '12

16 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8904.c | 31 ++++++++++++++++++++++++++++++- sound/soc/codecs/wm8904.h | 11 +++++++++++ 2 files changed, 41 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index f2a740d..14afc11 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -304,6 +304,7 @@ static bool wm8904_readable_register(struct device *dev, unsigned int reg) case WM8904_EQ23: case WM8904_EQ24: case WM8904_CONTROL_INTERFACE_TEST_1: + case WM8904_ADC_TEST_0: case WM8904_ANALOGUE_OUTPUT_BIAS_0: case WM8904_FLL_NCO_TEST_0: case WM8904_FLL_NCO_TEST_1: @@ -569,6 +570,29 @@ static const char *hpf_mode_text[] = { static const struct soc_enum hpf_mode = SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text); +static int wm8904_adc_osr_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int val; + int ret; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + if (ucontrol->value.integer.value[0]) + val = 0; + else + val = WM8904_ADC_128_OSR_TST_MODE | WM8904_ADC_BIASX1P5; + + snd_soc_update_bits(codec, WM8904_ADC_TEST_0, + WM8904_ADC_128_OSR_TST_MODE | WM8904_ADC_BIASX1P5, + val); + + return ret; +} + static const struct snd_kcontrol_new wm8904_adc_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8904_ADC_DIGITAL_VOLUME_LEFT, WM8904_ADC_DIGITAL_VOLUME_RIGHT, 1, 119, 0, digital_tlv), @@ -585,7 +609,12 @@ SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0, SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0), SOC_ENUM("High Pass Filter Mode", hpf_mode), -SOC_SINGLE("ADC 128x OSR Switch", WM8904_ANALOGUE_ADC_0, 0, 1, 0), +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC 128x OSR Switch", + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, + .put = wm8904_adc_osr_put, + .private_value = SOC_SINGLE_VALUE(WM8904_ANALOGUE_ADC_0, 0, 1, 0), +}, }; static const char *drc_path_text[] = { diff --git a/sound/soc/codecs/wm8904.h b/sound/soc/codecs/wm8904.h index 9e8c841..c29a0e8 100644 --- a/sound/soc/codecs/wm8904.h +++ b/sound/soc/codecs/wm8904.h @@ -123,6 +123,7 @@ #define WM8904_EQ23 0x9C #define WM8904_EQ24 0x9D #define WM8904_CONTROL_INTERFACE_TEST_1 0xA1 +#define WM8904_ADC_TEST_0 0xC6 #define WM8904_ANALOGUE_OUTPUT_BIAS_0 0xCC #define WM8904_FLL_NCO_TEST_0 0xF7 #define WM8904_FLL_NCO_TEST_1 0xF8 @@ -1557,6 +1558,16 @@ #define WM8904_USER_KEY_WIDTH 1 /* USER_KEY */ /* + * R198 (0xC6) - ADC Test 0 + */ +#define WM8904_ADC_128_OSR_TST_MODE 0x0004 /* ADC_128_OSR_TST_MODE */ +#define WM8904_ADC_128_OSR_TST_MODE_SHIFT 2 /* ADC_128_OSR_TST_MODE */ +#define WM8904_ADC_128_OSR_TST_MODE_WIDTH 1 /* ADC_128_OSR_TST_MODE */ +#define WM8904_ADC_BIASX1P5 0x0001 /* ADC_BIASX1P5 */ +#define WM8904_ADC_BIASX1P5_SHIFT 0 /* ADC_BIASX1P5 */ +#define WM8904_ADC_BIASX1P5_WIDTH 1 /* ADC_BIASX1P5 */ + +/* * R204 (0xCC) - Analogue Output Bias 0 */ #define WM8904_PGA_BIAS_MASK 0x0070 /* PGA_BIAS - [6:4] */ -- 1.7.7.3

1 0

[alsa-devel] [PATCH 2/2] ASoC: sgtl5000: update author email address
by zengzm.kernel＠gmail.com 16 Jan '12

16 Jan '12

From: Zeng Zhaoming <zengzm.kernel(a)gmail.com> Update MODULE_AUTHOR email address. Signed-off-by: Zeng Zhaoming <zengzm.kernel(a)gmail.com> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 71df1fe..a87e9ee 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1464,5 +1464,5 @@ static void __exit sgtl5000_exit(void) module_exit(sgtl5000_exit); MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver"); -MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng(a)freescale.com>"); +MODULE_AUTHOR("Zeng Zhaoming <zengzm.kernel(a)gmail.com>"); MODULE_LICENSE("GPL"); -- 1.7.6.1.385.gb7fcd0

2 1

[alsa-devel] [PATCH] ASoC: core - Free platform DAPM context at platform removal.
by Liam Girdwood 16 Jan '12

16 Jan '12

Fix platform removal by freeing the platform DAPM resources and remove it from the DAPM list. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3986520..b5ecf6d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -907,6 +907,10 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) if (err < 0) printk(KERN_ERR "asoc: failed to remove %s\n", platform->name); } + + /* Make sure all DAPM widgets are freed */ + snd_soc_dapm_free(&platform->dapm); + platform->probed = 0; list_del(&platform->card_list); module_put(platform->dev->driver->owner); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: dapm - Fix check for codec context in dapm_power_widgets().
by Liam Girdwood 16 Jan '12

16 Jan '12

Fixes a NULL pointer dereference in dapm_power_widgets() if the dapm context has no codec. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3ad1f59..1f55ded 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1426,7 +1426,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) dapm->target_bias_level = SND_SOC_BIAS_ON; break; case SND_SOC_DAPM_STREAM_STOP: - if (dapm->codec->active) + if (dapm->codec && dapm->codec->active) dapm->target_bias_level = SND_SOC_BIAS_ON; else dapm->target_bias_level = SND_SOC_BIAS_STANDBY; -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: wm8983: Remove useless snd_kcontrol
by Dimitris Papastamos 16 Jan '12

16 Jan '12

This must be a leftover from a previous driver. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8983.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index 17f04ec..7053880 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -249,9 +249,6 @@ static const char *eq5_cutoff_text[] = { static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5, eq5_cutoff_text); -static const char *speaker_mode_text[] = { "Class A/B", "Class D" }; -static const SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text); - static const char *depth_3d_text[] = { "Off", "6.67%", @@ -369,8 +366,6 @@ static const struct snd_kcontrol_new wm8983_snd_controls[] = { SOC_SINGLE_TLV("EQ5 Volume", WM8983_EQ5_HIGH_SHELF, 0, 24, 1, eq_tlv), SOC_ENUM("3D Depth", depth_3d), - - SOC_ENUM("Speaker Mode", speaker_mode) }; static const struct snd_kcontrol_new left_out_mixer[] = { -- 1.7.8.3

2 1

[alsa-devel] [PATCH] ALSA: HDA: Fix internal microphone on Dell Studio 16 XPS 1645
by David Henningsson 16 Jan '12

16 Jan '12

More than one user reports that changing the model from "both" to "dmic" makes their Internal Mic work. Cc: stable(a)kernel.org Tested-by: Martin Ling <martin-launchpad(a)earth.li> BugLink: https://bugs.launchpad.net/bugs/795823 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 87e684f..3556408 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1596,7 +1596,7 @@ static const struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02bd, "Dell Studio 1557", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02fe, - "Dell Studio XPS 1645", STAC_DELL_M6_BOTH), + "Dell Studio XPS 1645", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0413, "Dell Studio 1558", STAC_DELL_M6_DMIC), {} /* terminator */ -- 1.7.5.4

2 1

[alsa-devel] [PATCH] alsa: export compress headers
by Vinod Koul 16 Jan '12

16 Jan '12

Export compress_offload.h and compress_params.h for userland to use Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- include/sound/Kbuild | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/include/sound/Kbuild b/include/sound/Kbuild index 802947f..6df30ed 100644 --- a/include/sound/Kbuild +++ b/include/sound/Kbuild @@ -6,3 +6,5 @@ header-y += hdsp.h header-y += hdspm.h header-y += sb16_csp.h header-y += sfnt_info.h +header-y += compress_params.h +header-y += compress_offload.h -- 1.7.0.4

3 2

[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: Fix wrong register name in restore
by zengzm.kernel＠gmail.com 16 Jan '12

16 Jan '12

From: Zeng Zhaoming <zengzm.kernel(a)gmail.com> Correct SGTL5000_CHIP_CLK_CTRL to SGTL5000_CHIP_REF_CTRL in sgtl5000_restore_regs(), and add comment to explain the restore order. Signed-off-by: Zeng Zhaoming <zengzm.kernel(a)gmail.com> --- sound/soc/codecs/sgtl5000.c | 11 +++++++++-- 1 files changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index fc9b127..71df1fe 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -987,12 +987,12 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) /* restore regular registers */ for (reg = 0; reg <= SGTL5000_CHIP_SHORT_CTRL; reg += 2) { - /* this regs depends on the others */ + /* These regs should restore in particular order */ if (reg == SGTL5000_CHIP_ANA_POWER || reg == SGTL5000_CHIP_CLK_CTRL || reg == SGTL5000_CHIP_LINREG_CTRL || reg == SGTL5000_CHIP_LINE_OUT_CTRL || - reg == SGTL5000_CHIP_CLK_CTRL) + reg == SGTL5000_CHIP_REF_CTRL) continue; snd_soc_write(codec, reg, cache[reg]); @@ -1005,6 +1005,13 @@ static int sgtl5000_restore_regs(struct snd_soc_codec *codec) /* * restore power and other regs according * to set_power() and set_clock() + * the order of restore is: + * 1. SGTL5000_CHIP_CLK_CTRL MCLK_FREQ bits (1:0) should be restore after + * SGTL5000_CHIP_ANA_POWER PLL bits set + * 2. SGTL5000_CHIP_LINREG_CTRL should be set before + * SGTL5000_CHIP_ANA_POWER LINREG_D restored + * 3. SGTL5000_CHIP_REF_CTRL controls Analog Ground Voltage, + * I prefer to resotre it after SGTL5000_CHIP_ANA_POWER restored */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, cache[SGTL5000_CHIP_LINREG_CTRL]); -- 1.7.6.1.385.gb7fcd0

3 2

[alsa-devel] [PATCH v3] ASoC: jz4740: Convert qi_lb60 to use snd_soc_register_card()
by Axel Lin 15 Jan '12

15 Jan '12

Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- v2: Fix a typo in v1. v3: Free the card before the GPIOs since the card uses the GPIOs arch/mips/jz4740/board-qi_lb60.c | 6 ++++ sound/soc/jz4740/qi_lb60.c | 56 +++++++++++++++++-------------------- 2 files changed, 32 insertions(+), 30 deletions(-) diff --git a/arch/mips/jz4740/board-qi_lb60.c b/arch/mips/jz4740/board-qi_lb60.c index 639e3ce..9a91fe9 100644 --- a/arch/mips/jz4740/board-qi_lb60.c +++ b/arch/mips/jz4740/board-qi_lb60.c @@ -418,6 +418,11 @@ static struct platform_device qi_lb60_charger_device = { }, }; +/* audio */ +static struct platform_device qi_lb60_audio_device = { + .name = "qi-lb60-audio", + .id = -1, +}; static struct platform_device *jz_platform_devices[] __initdata = { &jz4740_udc_device, @@ -434,6 +439,7 @@ static struct platform_device *jz_platform_devices[] __initdata = { &qi_lb60_gpio_keys, &qi_lb60_pwm_beeper, &qi_lb60_charger_device, + &qi_lb60_audio_device, }; static void __init board_gpio_setup(void) diff --git a/sound/soc/jz4740/qi_lb60.c b/sound/soc/jz4740/qi_lb60.c index 0097c3b..e8aaff1 100644 --- a/sound/soc/jz4740/qi_lb60.c +++ b/sound/soc/jz4740/qi_lb60.c @@ -91,56 +91,52 @@ static struct snd_soc_card qi_lb60 = { .num_dapm_routes = ARRAY_SIZE(qi_lb60_routes), }; -static struct platform_device *qi_lb60_snd_device; - static const struct gpio qi_lb60_gpios[] = { { QI_LB60_SND_GPIO, GPIOF_OUT_INIT_LOW, "SND" }, { QI_LB60_AMP_GPIO, GPIOF_OUT_INIT_LOW, "AMP" }, }; -static int __init qi_lb60_init(void) +static int __devinit qi_lb60_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &qi_lb60; int ret; - qi_lb60_snd_device = platform_device_alloc("soc-audio", -1); - - if (!qi_lb60_snd_device) - return -ENOMEM; - ret = gpio_request_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); - if (ret) { - pr_err("qi_lb60 snd: Failed to request gpios: %d\n", ret); - goto err_device_put; - } + if (ret) + return ret; - platform_set_drvdata(qi_lb60_snd_device, &qi_lb60); + card->dev = &pdev->dev; - ret = platform_device_add(qi_lb60_snd_device); + ret = snd_soc_register_card(card); if (ret) { - pr_err("qi_lb60 snd: Failed to add snd soc device: %d\n", ret); - goto err_unset_pdata; + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); } - - return 0; - -err_unset_pdata: - platform_set_drvdata(qi_lb60_snd_device, NULL); -/*err_gpio_free_array:*/ - gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); -err_device_put: - platform_device_put(qi_lb60_snd_device); - return ret; } -module_init(qi_lb60_init); -static void __exit qi_lb60_exit(void) +static int __devexit qi_lb60_remove(struct platform_device *pdev) { - platform_device_unregister(qi_lb60_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); gpio_free_array(qi_lb60_gpios, ARRAY_SIZE(qi_lb60_gpios)); + return 0; } -module_exit(qi_lb60_exit); + +static struct platform_driver qi_lb60_driver = { + .driver = { + .name = "qi-lb60-audio", + .owner = THIS_MODULE, + }, + .probe = qi_lb60_probe, + .remove = __devexit_p(qi_lb60_remove), +}; + +module_platform_driver(qi_lb60_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_DESCRIPTION("ALSA SoC QI LB60 Audio support"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:qi-lb60-audio"); -- 1.7.5.4

3 4

[alsa-devel] [PATCH] ASoC: omap-mcpdm: Set constraint for 24 msb
by Peter Ujfalusi 15 Jan '12

15 Jan '12

OMAP McPDM supports 24 msb in the 32 bit sample. Make sure that applications get this information. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcpdm.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 0e25df4..50d758a 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -265,6 +265,8 @@ static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream, mutex_lock(&mcpdm->mutex); + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); + if (!dai->active) { /* Enable watch dog for ES above ES 1.0 to avoid saturation */ if (omap_rev() != OMAP4430_REV_ES1_0) { -- 1.7.8.3

2 1

[alsa-devel] [PATCH] alsamixer: Confirm that a drastic volume change is intentional.
by Kusanagi Kouichi 14 Jan '12

14 Jan '12

Home key has been disabled because it might at least surprise a user unless volume is already nearly full. Not only home but 9, 8 and so on might also surprise a user. So instead of disableing key, show confirmation dialog for drastic volume change. Signed-off-by: Kusanagi Kouichi <slash(a)ac.auone-net.jp> --- alsamixer/mixer_widget.c | 52 ++++++++++++++++++++++++++++++++++++++------ alsamixer/volume_mapping.c | 3 ++ 2 files changed, 48 insertions(+), 7 deletions(-) diff --git a/alsamixer/mixer_widget.c b/alsamixer/mixer_widget.c index caaf777..d01a6ec 100644 --- a/alsamixer/mixer_widget.c +++ b/alsamixer/mixer_widget.c @@ -207,6 +207,7 @@ static void show_help(void) _("Up/Down Change volume"), _("+ - Change volume"), _("Page Up/Dn Change volume in big steps"), + _("Home Set volume to 100%"), _("End Set volume to 0%"), _("0-9 Set volume to 0%-90%"), _("Q W E Increase left/both/right volumes"), @@ -305,9 +306,10 @@ static void change_enum_relative(struct control *control, int delta) snd_mixer_selem_set_enum_item(control->elem, i, new_index); } -static void change_volume_to_percent(struct control *control, int value, unsigned int channels) +static int change_volume_to_percent(struct control *control, int value, unsigned int channels) { int (*set_func)(snd_mixer_elem_t *, snd_mixer_selem_channel_id_t, double, int); + int err = 0; if (!(control->flags & HAS_VOLUME_1)) channels = LEFT; @@ -316,9 +318,10 @@ static void change_volume_to_percent(struct control *control, int value, unsigne else set_func = set_normalized_capture_volume; if (channels & LEFT) - set_func(control->elem, control->volume_channels[0], value / 100.0, 0); - if (channels & RIGHT) - set_func(control->elem, control->volume_channels[1], value / 100.0, 0); + err = set_func(control->elem, control->volume_channels[0], value / 100.0, 0); + if (channels & RIGHT && err <= 0) + err = set_func(control->elem, control->volume_channels[1], value / 100.0, 0); + return err; } static double clamp_volume(double v) @@ -371,7 +374,44 @@ static void change_control_to_percent(int value, unsigned int channels) if (control->flags & TYPE_ENUM) change_enum_to_percent(control, value); else - change_volume_to_percent(control, value, channels); + if (change_volume_to_percent(control, -value, channels)) { + char buf[50]; + const char * const text[] = { buf, " (Y/n)" }; + const struct widget *active_widget; + + snprintf(buf, sizeof buf, "Do you really change volume to %d%%?", value); + show_text(text, 2, ""); + active_widget = get_active_widget(); + wtimeout(active_widget->window, -1); + while (1) { + switch (wgetch(active_widget->window)) { +#ifdef KEY_RESIZE + case KEY_RESIZE: + window_size_changed(); + continue; +#endif + case 'Y': + case 'y': + case '\n': + case KEY_ENTER: + change_volume_to_percent(control, value, channels); + break; + case 'N': + case 'n': + case 27: + case KEY_CANCEL: + case KEY_CLOSE: + case KEY_EXIT: + break; + default: + continue; + } + break; + } + active_widget->handle_key(KEY_ENTER); + } else + change_volume_to_percent(control, value, channels); + display_controls(); } @@ -536,12 +576,10 @@ static void on_handle_key(int key) case KEY_NPAGE: change_control_relative(-5, LEFT | RIGHT); break; -#if 0 case KEY_BEG: case KEY_HOME: change_control_to_percent(100, LEFT | RIGHT); break; -#endif case KEY_LL: case KEY_END: change_control_to_percent(0, LEFT | RIGHT); diff --git a/alsamixer/volume_mapping.c b/alsamixer/volume_mapping.c index 1c0d7c4..bd160ed 100644 --- a/alsamixer/volume_mapping.c +++ b/alsamixer/volume_mapping.c @@ -133,6 +133,9 @@ static int set_normalized_volume(snd_mixer_elem_t *elem, double min_norm; int err; + if (volume < 0) + return -volume - get_normalized_volume(elem, channel, ctl_dir) >= 0.5; + err = get_dB_range[ctl_dir](elem, &min, &max); if (err < 0 || min >= max) { err = get_raw_range[ctl_dir](elem, &min, &max); -- 1.7.8.3

2 2

[alsa-devel] [PATCH 0/2] cuse: implement mmap/munmap
by Miklos Szeredi 13 Jan '12

13 Jan '12

These patches implement mmap support for CUSE (character device in userspace). The immediate application for this is to get old games requiring OSS/mmap support to work with OSS Proxy, written by Tejun. So firstly this is a call for testers; please try this out and report any problems. Three pieces are needed: kernel: git://git.kernel.org/pub/scm/linux/kernel/git/mszeredi/fuse.git for-next fuse: git://fuse.git.sourceforge.net/gitroot/fuse/fuse master osspd: git://fuse.git.sourceforge.net/gitroot/fuse/osspd master Comments about the interface and implementation in general (not related to OSS proxy) are very welcome as well. Thanks, Miklos --- Miklos Szeredi (1): fuse: create fuse_conn_operations Tejun Heo (1): cuse: implement memory mapping --- fs/fuse/cuse.c | 426 +++++++++++++++++++++++++++++++++++++++++++++++++- fs/fuse/dev.c | 155 ++----------------- fs/fuse/fuse_i.h | 35 ++++- fs/fuse/inode.c | 167 +++++++++++++++++++- include/linux/fuse.h | 25 +++ 5 files changed, 656 insertions(+), 152 deletions(-)

3 4

[alsa-devel] Read and write different buffer sizes
by Elias Kokkinis 13 Jan '12

13 Jan '12

Hello all, I would like to ask if it is possible in ALSA to define different buffer sizes in the input and output streams. I am using PCM sound and I want to read 2048 samples, do some processing involving overlap-add and then keep 1024 of them and put them to the output. And so on... So I want to have the same sample rate and everything but different buffer sizes. Is this possible? I have tried an implementation of this, but I think the output for some reason still outputs 2048 samples and the audio is choppy, like having 1024 samples of audio and 1024 of silence. I don't know if I am making myself clear here, maybe it's the 10 hours of work up to now, but anyway any help/comments would be greatly appreciated. Elias Kokkinis

2 2

Re: [alsa-devel] Trying to understand alsa
by Andrew Eikum 13 Jan '12

13 Jan '12

On Thu, Jan 12, 2012 at 11:16:47PM +0000, Jonathan Andrews wrote: > On Thu, 2012-01-12 at 16:12 -0600, Andrew Eikum wrote: > > On Thu, Jan 12, 2012 at 09:44:11PM +0000, Jonathan Andrews wrote: > > > I have an application that works using 512 sample packets of 22050Hz 16 > > > bit mono audio. The 'receiver' takes many audio streams from a network > > > via UDP, at the moment it pipes them into pulse. > > > > > > Can alsa buffer audio. At the moment every time I and set an audio > > > buffer size I get a negative response from > > > snd_pcm_hw_params_set_buffer_size . I'm somewhat confused about the > > > units alsa uses ... > > > > > > > You don't want to over-specify your requirements. You require a buffer > > size of "at least" 3 * 512 frames. So use set_buffer_size_min(). > > I cant find any reference to "set_buffer_size_min" in the ALSA API > documentation I have or the link you provided ? > snd_pcm_hw_params_set_buffer_time_near() is the closest and that seems > to take an argument in useconds. > ? > http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html > The doxygen docs are split strangely. The function is documented here: http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___h_w___params.… I find keeping the alsa-lib source on hand for quick grepping is often helpful. Andrew

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Re: [alsa-devel] Dell N5110 HDA External Speaker is mute
by Takashi Iwai 13 Jan '12

13 Jan '12

At Thu, 12 Jan 2012 11:51:12 +0000, Cefn Hoile wrote: > > Thanks, Takashi. > > I don't know what tools and invocations to use test out the GPIO > values you describe. I think you're saying I shouldn't bother toggling > things through HDA Analyzer as there's nothing obviously wrong in that > part of the configuration (as reported by alsa-info.sh which is > attached as requested). > > I noticed in HD-Audio.txt a section which mentions a similar problem > and IDT codecs specifically (below). Is there an IRC or mailing list > where people could talk me through the basics of trying different > GPIO/EAPD, Power Enable or Pin Sense manipulations. I wouldn't know > how, yet. If I succeed I'll report back. > > Hopefully this will be worth it, as I think the N5110 is a pretty > popular model with a common card, and probably a bunch of Dell > hardware could get fixed if it could be figured out. You can change the GPIO setup via hda-verb on the fly. For example, to set GPIO bit 0, try hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x01 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x01 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x01 where the output direction is assumed, and only GPIO 0 is set. For GPIO bit 1, hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x02 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIR 0x02 hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x02 and 0x04 for GPIO bit 2. Your device have up to three GPIO bits. If none of above doesn't any change, try to set DATA 0 instead of 1, and/or DIR 0 instead of 1. Takashi > QUOTING FROM HD-Audio.txt >>>>>>> > > If you are masochistic enough to debug the driver problem, note the > following: > > - The speaker (and the headphone, too) output often requires the > external amplifier. This can be set usually via EAPD verb or a > certain GPIO. If the codec pin supports EAPD, you have a better > chance via SET_EAPD_BTL verb (0x70c). On others, GPIO pin (mostly > it's either GPIO0 or GPIO1) may turn on/off EAPD. > - Some Realtek codecs require special vendor-specific coefficients to > turn on the amplifier. See patch_realtek.c. > - IDT codecs may have extra power-enable/disable controls on each > analog pin. See patch_sigmatel.c. > - Very rare but some devices don't accept the pin-detection verb until > triggered. Issuing GET_PIN_SENSE verb (0xf09) may result in the > codec-communication stall. Some examples are found in > patch_realtek.c. > > <<<<<<<< > > On 12 January 2012 11:25, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 12 Jan 2012 10:56:24 +0000, > > Cefn Hoile wrote: > >> > >> In case anything has changed in my config since the original info run, > >> here is the same script run right now... > >> > >> http://www.alsa-project.org/db/?f=d0a1fd23339878916b611c88e64e4868b45445ce > > > > Could you rather attach? Then it'll be archived and searchable. > > > > Anyway, I don't see any obvious problem in the output. > > If the pin assignment is correct, the routing and amp setups should be > > OK with the current status. > > > > Maybe it's a missing GPIO setup. Try to adjust GPIO mask, direction > > and value bits. > > > > > > Takashi

2 7

[alsa-devel] [PATCH] Alsa-utils: Fix typo causing some volumes not to be initialized
by David Henningsson 13 Jan '12

13 Jan '12

When I was reading the source code, I stumbled over this typo causing Center, LFE, and Surround speakers not to be initialized. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- alsactl/init/default | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/alsactl/init/default b/alsactl/init/default index af64cb1..5fb0113 100644 --- a/alsactl/init/default +++ b/alsactl/init/default @@ -124,7 +124,7 @@ CTL{name}="Front Speaker Playback Switch",CTL{do_search}=="1", \ CTL{values}="on" CTL{reset}="mixer" -CTL{name}="Front Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" +CTL{name}="Surround Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" # if master volume control is present, turn speaker volume to max ENV{has_pmaster_vol}=="true",CTL{write}=="0dB",GOTO="" ENV{has_pmaster_vol}=="true",CTL{write}=="100%",GOTO="" @@ -134,7 +134,7 @@ CTL{name}="Surround Speaker Playback Switch",CTL{do_search}=="1", \ CTL{values}="on" CTL{reset}="mixer" -CTL{name}="Front Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" +CTL{name}="Center Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" # if master volume control is present, turn speaker volume to max ENV{has_pmaster_vol}=="true",CTL{write}=="0dB",GOTO="" ENV{has_pmaster_vol}=="true",CTL{write}=="100%",GOTO="" @@ -144,7 +144,7 @@ CTL{name}="Center Speaker Playback Switch",CTL{do_search}=="1", \ CTL{values}="on" CTL{reset}="mixer" -CTL{name}="Front Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" +CTL{name}="LFE Speaker Playback Volume",PROGRAM!="__ctl_search",GOTO="" # if master volume control is present, turn speaker volume to max ENV{has_pmaster_vol}=="true",CTL{write}=="0dB",GOTO="" ENV{has_pmaster_vol}=="true",CTL{write}=="100%",GOTO="" -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ALSA: HDA: Use LPIB position fix for Macbook Pro 7, 1
by David Henningsson 13 Jan '12

13 Jan '12

Several users have reported "choppy" audio under the 3.2 kernel, and that changing position_fix to 1 has resolved their problem. The chip is an nVidia Corporation MCP89 High Definition Audio, [10de:0d94] (rev a2). Cc: stable(a)kernel.org (v3.2+) BugLink: https://bugs.launchpad.net/bugs/909419 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bd447b6..2e1c063 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2500,6 +2500,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS 1101HA", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), + SND_PCI_QUIRK(0x10de, 0xcb89, "Macbook Pro 7,1", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), -- 1.7.5.4

2 3

[alsa-devel] duplicated test in sound/soc/codecs/sgtl5000.c
by Julia Lawall 13 Jan '12

13 Jan '12

There is a duplicated test in the function sgtl5000_restore_regs in the file sound/soc/codecs/sgtl5000.c: /* this regs depends on the others */ if (reg == SGTL5000_CHIP_ANA_POWER || reg == SGTL5000_CHIP_CLK_CTRL || reg == SGTL5000_CHIP_LINREG_CTRL || reg == SGTL5000_CHIP_LINE_OUT_CTRL || reg == SGTL5000_CHIP_CLK_CTRL) continue; I don't know if some constant other than SGTL5000_CHIP_CLK_CTRL should be tested, or if the second test should just be deleted. julia

4 5

[alsa-devel] Todays GIT system of alsa
by Donn Washburn 13 Jan '12

13 Jan '12

Hey Group; I wish to let you know that todays GIT set of files fixed openSuSEs 11.4 and 12.1 problem that I have been having. That was a loss of pots/adjustments and mixed mute controls on kmix, gmixer and pavucontrol. All of mixers work. The system required snd-hda-intel module. The system is a ASUS M4A87TD Mother board and sound VIA VT1818 High Definition Audio 8 Channel Codec. lsmod | less snd_pcm_oss 42358 0 snd_mixer_oss 18292 1 snd_pcm_oss snd_seq 49875 0 snd_seq_device 14530 1 snd_seq snd_hda_intel 29539 10 snd_hda_codec 105907 3 snd_hda_codec_hdmi,snd_hda_codec_via,snd_hda_intel snd_hwdep 13604 1 snd_hda_codec snd_pcm 81432 7 snd_pcm_oss,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_timer 25525 5 snd_seq,snd_pcm snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_codec_hdmi, snd_hda-codec_via,snd_hda_soundcore 15165 1 snd Thanks again -- 73 de Donn Washburn 307 Savoy Street Email:" n5xwb(a)comcast.net " Sugar Land, TX 77478 LL# 1.281.242.3256 Ham Callsign N5XWB HAMs : " n5xwb(a)arrl.net " VoIP via Skype:n5xwbg BMWMOA #:4146 Ambassador " http://counter.li.org " #279316

1 0

[alsa-devel] [PATCH] ASoC: codecs: wm9090: Fix build warnings due to unused variables
by Fabio Estevam 13 Jan '12

13 Jan '12

Fix the following build warnings: sound/soc/codecs/wm9090.c: In function ‘wm9090_set_bias_level’: sound/soc/codecs/wm9090.c:496: warning: unused variable ‘ret’ sound/soc/codecs/wm9090.c:496: warning: unused variable ‘i’ sound/soc/codecs/wm9090.c:495: warning: unused variable ‘reg_cache’ sound/soc/codecs/wm9090.c: In function ‘wm9090_i2c_remove’: sound/soc/codecs/wm9090.c:658: warning: unused variable ‘wm9090’ Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/wm9090.c | 5 ----- 1 files changed, 0 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 41ebe0d..acac7d6 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -492,9 +492,6 @@ static int wm9090_add_controls(struct snd_soc_codec *codec) static int wm9090_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 *reg_cache = codec->reg_cache; - int i, ret; - switch (level) { case SND_SOC_BIAS_ON: break; @@ -655,8 +652,6 @@ static int wm9090_i2c_probe(struct i2c_client *i2c, static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) { - struct wm9090_priv *wm9090 = i2c_get_clientdata(i2c); - snd_soc_unregister_codec(&i2c->dev); return 0; -- 1.7.1

2 1

Re: [alsa-devel] Trying to understand alsa
by Jonathan Andrews 13 Jan '12

13 Jan '12

On Thu, 2012-01-12 at 16:12 -0600, Andrew Eikum wrote: > On Thu, Jan 12, 2012 at 09:44:11PM +0000, Jonathan Andrews wrote: > > I have an application that works using 512 sample packets of 22050Hz 16 > > bit mono audio. The 'receiver' takes many audio streams from a network > > via UDP, at the moment it pipes them into pulse. > > > > Can alsa buffer audio. At the moment every time I and set an audio > > buffer size I get a negative response from > > snd_pcm_hw_params_set_buffer_size . I'm somewhat confused about the > > units alsa uses ... > > > > You don't want to over-specify your requirements. You require a buffer > size of "at least" 3 * 512 frames. So use set_buffer_size_min(). I cant find any reference to "set_buffer_size_min" in the ALSA API documentation I have or the link you provided ? snd_pcm_hw_params_set_buffer_time_near() is the closest and that seems to take an argument in useconds. ? http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html > Otherwise ALSA will try to set exactly that buffer size, which can > fail. > > Check the function signatures for units. Notice that > set_buffer_size*() all take snd_pcm_uframes_t, that is, the number of > frames you want to store. In ALSA terms, a "frame" is a set of a > single sample for every channel. Since you have mono audio, a frame > and a sample are actually the same unit (for 16 bit stereo audio, > 1 frame = 2 samples = 32 bits). > > > What I want to do is tell ALSA to hold a buffer of 3 of my packets (3 x > > 1024Bytes, thats 512 x 16 bit samples) while I feed extra packets (1K > > Byte, 512 samples per buffer) in for playback. The packets are arriving > > at roughly the correct rate, I just need a buffer to iron out any > > jitter in network transmit, do I have to do this myself ? > > > > Can somebody help by telling me which numbers I push into which places > > to make it work ? > > > > At the moment I get i keep getting a broken pipe, if I underrun how can > > I make it just wait for me ? > > > > If a packet arrives very late (and one will, eventually), you will > underrun. That's unavoidable. You can check for SND_PCM_STATE_XRUN > from snd_pcm_state(). It's undocumented, but you need to call > snd_pcm_avail_update() first to get an accurate reading from > snd_pcm_state(). When an underrun occurs, recover with > snd_pcm_recover() and then start writing data again. Ok, thanks. > If a packet arrives early, you'll need to check that the ALSA buffer > isn't full (see snd_pcm_avail_update()), and store it within your > application to write later if it is full. > > There's some mostly-accurate information here: > http://0pointer.de/blog/projects/guide-to-sound-apis.html > As with all ALSA documentation, it is confusing and often incorrect, > but it's probably the most helpful document I've found. Many thanks for the help, Jon

1 1

[alsa-devel] Trying to understand alsa
by Jonathan Andrews 13 Jan '12

13 Jan '12

> If a packet arrives very late (and one will, eventually), you will > underrun. That's unavoidable. You can check for SND_PCM_STATE_XRUN > from snd_pcm_state(). It's undocumented, but you need to call > snd_pcm_avail_update() first to get an accurate reading from > snd_pcm_state(). When an underrun occurs, recover with > snd_pcm_recover() and then start writing data again. > Ok, this my first attempt at recovery, still haven't solved buffering problem so it underruns for each and every write. Does this look roughly correct ? Thanks, Jon #ifdef ALSA // snd_pcm_writen - writes non interleaved (mono) samples // bufs an explicit void* pointer type set to point at real sample buffer "audio" // broken pipe indicates buffer underrun numrecs=sizeof(audio)/2; // 1k buffer, 512 samples //printf("%d \n", snd_pcm_wait(playback_handle[activeslot],1)); err = snd_pcm_writen (playback_handle[activeslot], &bufs, numrecs); if (err<numrecs) { fflush(stdout); if (err>0) { fprintf (stderr, "snd_pcm_writen: wrote only %d\n",err); fflush(stderr); } else { if (err == EPIPE) { errb=snd_pcm_recover(playback_handle[activeslot],err,0); // error passed is resullt of writen if (errb<0) { fprintf(stderr,"Failed to recover from underrun\n"); fflush(stderr); exit(1); } } else { fprintf (stderr, "snd_pcm_writen: write to audio interface failed (%s) (%d)\n", snd_strerror (err),err); fflush(stdout); fflush(stderr); exit (1); } } } #endif

1 0

[alsa-devel] Trying to understand alsa
by Jonathan Andrews 13 Jan '12

13 Jan '12

On Thu, 2012-01-12 at 21:50 +0000, Fons Adriaensen wrote: > On Thu, Jan 12, 2012 at 09:44:11PM +0000, Jonathan Andrews wrote: > > > What I want to do is tell ALSA to hold a buffer of 3 of my packets (3 x > > 1024Bytes, thats 512 x 16 bit samples) while I feed extra packets (1K > > Byte, 512 samples per buffer) in for playback. The packets are arriving > > at roughly the correct rate, I just need a buffer to iron out any > > jitter in network transmit, do I have to do this myself ? > > Unless data is arriving on average at *exactly* the sample > rate of the ALSA device your buffering will sooner or later > underrun or overrun. ALSA itself can't handle this use case, > it requires adaptive resampling and it's not easy. I'm assuming alsa is holding 3k bytes for me and im writing 1k byte chunks, so if the audio has not been refreshed by 3 chunks (70ms) then I should throw it away as its no longer 'real time' At the moment i would happy just to throw away packets that would cause an overrun. I just wish i could convince alsa to buffer for me .... ho hum ... with pulse I just did this to set up 'some sort of buffer length' with 'some sort of pre buffering before play' interval. Its a bit hacky at the moment but seems to work. pa_buffer_attr paattr; int buflen=sizeof(audio); paattr.tlength = buflen * 4; // Playback only, target length of the buffer (was *2) paattr.minreq = buflen; paattr.fragsize = buflen; paattr.prebuf = buflen * 6; // How much data pulse holds before it starts playing (was *4) paattr.maxlength = buflen * 8; // playback pipeline N samples deep (was *4) Im looking for alsa equivalent. Thanks, Jon

1 0

[alsa-devel] Trying to understand alsa
by Jonathan Andrews 13 Jan '12

13 Jan '12

Hi people, i'm trying to understand alsa.... I have an application that works using 512 sample packets of 22050Hz 16 bit mono audio. The 'receiver' takes many audio streams from a network via UDP, at the moment it pipes them into pulse. Can alsa buffer audio. At the moment every time I and set an audio buffer size I get a negative response from snd_pcm_hw_params_set_buffer_size . I'm somewhat confused about the units alsa uses ... What I want to do is tell ALSA to hold a buffer of 3 of my packets (3 x 1024Bytes, thats 512 x 16 bit samples) while I feed extra packets (1K Byte, 512 samples per buffer) in for playback. The packets are arriving at roughly the correct rate, I just need a buffer to iron out any jitter in network transmit, do I have to do this myself ? Can somebody help by telling me which numbers I push into which places to make it work ? At the moment I get i keep getting a broken pipe, if I underrun how can I make it just wait for me ? I only code in plain old C at the moment. Many thanks for any help/advice anyone can offer. Jon

3 2

[alsa-devel] [PATCH RESEND] ASoC: omap: Add .owner to struct snd_soc_card
by Axel Lin 12 Jan '12

12 Jan '12

Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/am3517evm.c | 1 + sound/soc/omap/ams-delta.c | 1 + sound/soc/omap/igep0020.c | 1 + sound/soc/omap/n810.c | 1 + sound/soc/omap/omap3evm.c | 1 + sound/soc/omap/omap3pandora.c | 1 + sound/soc/omap/omap4-hdmi-card.c | 1 + sound/soc/omap/osk5912.c | 1 + sound/soc/omap/overo.c | 1 + sound/soc/omap/rx51.c | 1 + sound/soc/omap/sdp3430.c | 1 + sound/soc/omap/sdp4430.c | 1 + sound/soc/omap/zoom2.c | 1 + 13 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index c1cd4a0..add4866 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -107,6 +107,7 @@ static struct snd_soc_dai_link am3517evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_am3517evm = { .name = "am3517evm", + .owner = THIS_MODULE, .dai_link = &am3517evm_dai, .num_links = 1, diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index a04a433..3e523a7 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -597,6 +597,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = { /* Audio card driver */ static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", + .owner = THIS_MODULE, .dai_link = &ams_delta_dai_link, .num_links = 1, .set_bias_level = ams_delta_set_bias_level, diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index 591fbf8..ccae58a 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link igep2_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_igep2 = { .name = "igep2", + .owner = THIS_MODULE, .dai_link = &igep2_dai, .num_links = 1, }; diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index fc6209b..597be41 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -289,6 +289,7 @@ static struct snd_soc_dai_link n810_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_n810 = { .name = "N810", + .owner = THIS_MODULE, .dai_link = &n810_dai, .num_links = 1, diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 6857895..071fcb0 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -70,6 +70,7 @@ static struct snd_soc_dai_link omap3evm_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_omap3evm = { .name = "omap3evm", + .owner = THIS_MODULE, .dai_link = &omap3evm_dai, .num_links = 1, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 7605c37..07794bd 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -233,6 +233,7 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { /* SoC card */ static struct snd_soc_card snd_soc_card_omap3pandora = { .name = "omap3pandora", + .owner = THIS_MODULE, .dai_link = omap3pandora_dai, .num_links = ARRAY_SIZE(omap3pandora_dai), }; diff --git a/sound/soc/omap/omap4-hdmi-card.c b/sound/soc/omap/omap4-hdmi-card.c index 52d471c..28d689b 100644 --- a/sound/soc/omap/omap4-hdmi-card.c +++ b/sound/soc/omap/omap4-hdmi-card.c @@ -74,6 +74,7 @@ static struct snd_soc_dai_link omap4_hdmi_dai = { static struct snd_soc_card snd_soc_omap4_hdmi = { .name = "OMAP4HDMI", + .owner = THIS_MODULE, .dai_link = &omap4_hdmi_dai, .num_links = 1, }; diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 351ec9d..d859b59 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -108,6 +108,7 @@ static struct snd_soc_dai_link osk_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_osk = { .name = "OSK5912", + .owner = THIS_MODULE, .dai_link = &osk_dai, .num_links = 1, diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index c3550ae..2ee889c 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -72,6 +72,7 @@ static struct snd_soc_dai_link overo_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_card_overo = { .name = "overo", + .owner = THIS_MODULE, .dai_link = &overo_dai, .num_links = 1, }; diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index ad16db5..fada6ef 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -383,6 +383,7 @@ static struct snd_soc_codec_conf rx51_codec_conf[] = { /* Audio card */ static struct snd_soc_card rx51_sound_card = { .name = "RX-51", + .owner = THIS_MODULE, .dai_link = rx51_dai, .num_links = ARRAY_SIZE(rx51_dai), .aux_dev = rx51_aux_dev, diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index e8fbf8e..2c85066 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -213,6 +213,7 @@ static struct snd_soc_dai_link sdp3430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp3430 = { .name = "SDP3430", + .owner = THIS_MODULE, .dai_link = sdp3430_dai, .num_links = ARRAY_SIZE(sdp3430_dai), diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 2735fa0..175ba9a 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -226,6 +226,7 @@ static struct snd_soc_dai_link sdp4430_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_sdp4430 = { .name = "SDP4430", + .owner = THIS_MODULE, .dai_link = sdp4430_dai, .num_links = ARRAY_SIZE(sdp4430_dai), diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 7641a7f..981616d 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -157,6 +157,7 @@ static struct snd_soc_dai_link zoom2_dai[] = { /* Audio machine driver */ static struct snd_soc_card snd_soc_zoom2 = { .name = "Zoom2", + .owner = THIS_MODULE, .dai_link = zoom2_dai, .num_links = ARRAY_SIZE(zoom2_dai), -- 1.7.5.4

3 2

[alsa-devel] Dell N5110 HDA External Speaker is mute
by Cefn Hoile 12 Jan '12

12 Jan '12

I'd like to report an issue experienced with the latest Oneiric, Dell N5110, and IDT STAC92HD87 which has been explored through quite a few steps already, and seems to have bottomed out to a set of pins/controls/flags which are not being properly managed in order to toggle speaker activation when a headphone is inserted. So far, I haven't heard sound from my speakers at all, although within Alsamixer, the Speaker volume is visibly toggled from 100% to MM (mute) on headphone insertion. I've tried every model against snd-hda-intel, many different other options, and started to explore the HDA Analyzer, which includes controls named like this... Node 0x13 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Device: name="STAC92xx Analog", type="Audio", device=0 Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=63 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=1, idx=0, ofs=0 Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 Amp-Out vals: [0x7f 0x7f] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples ...but toggling these checkboxes within the HDA Analyzer gui doesn't trigger any sound from the speakers either. It could be a rich combination of the very many checkboxes and sliders which are revealed by the Analyzer, but I don't really know where to start. The bug information downstream is here... https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/914290 I'd really welcome suggestions for what else to try and happy to invest time exploring options to try and make ALSA better still. Cefn http://cefn.com

2 5

Re: [alsa-devel] [PATCH 2/4] snd-usb-6fire: add individual volume control for analog channels
by Torsten Schenk 12 Jan '12

12 Jan '12

Sorry, somehow my message got corrupted in my last post. I will search the problem later and reply again. Thanks, Torsten

1 0

Re: [alsa-devel] [PATCH 2/4] snd-usb-6fire: add individual volume control for analog channels
by Torsten Schenk 12 Jan '12

12 Jan '12

On Thu, 12 Jan 2012 07:42:19 +0100Takashi Iwai <tiwai(a)suse.de> wrote: > At Thu, 12 Jan 2012 00:15:56 +0100,> Torsten Schenk wrote:> > > > /*> > - * calculated with $value\[i\] = 128 \cdot sqrt[3]{\frac{i}{128}}$> > - * this is done because the linear values cause rapid degredation> > - * of volume in the uppermost region.> > - */> > -static const u8 log_volume_table[128] = {> > - 0x00, 0x19, 0x20, 0x24, 0x28, 0x2b, 0x2e, 0x30, 0x32, 0x34,> > - 0x36, 0x38, 0x3a, 0x3b, 0x3d, 0x3e, 0x40, 0x41, 0x42, 0x43,> > - 0x44, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4c, 0x4d, 0x4e,> > - 0x4e, 0x4f, 0x50, 0x51, 0x52, 0x53, 0x53, 0x54, 0x55, 0x56,> > - 0x56, 0x57, 0x58, 0x58, 0x59, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c,> > - 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x62, 0x62,> > - 0x63, 0x63, 0x64, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x68,> > - 0x68, 0x69, 0x69, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c,> > - 0x6d, 0x6d, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71,> > - 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75,> > - 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79,> > - 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c,> > - 0x7d, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f };> > So, with the new implementation, you don't need the linear -> dB> conversion?I decided to change the volume slider behaviour because of theadditional information I got on the DAC. On the other hand it seemsreasonable to do that in a separate patch. I could do so in my nextsubmit. > > > .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,> > - .info = usb6fire_control_master_vol_info,> > - .get = usb6fire_control_master_vol_get,> > - .put = usb6fire_control_master_vol_put> > + .info = usb6fire_control_output_vol_info,> > + .get = usb6fire_control_output_vol_get,> > + .put = usb6fire_control_output_vol_put,> > + .private_value = 1> > },> > {> > .iface = SNDRV_CTL_ELEM_IFACE_MIXER,> > + .name = "Output 5/6 Playback Volume",> > + .index = 0,> > + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,> > + .info = usb6fire_control_output_vol_info,> > + .get = usb6fire_control_output_vol_get,> > + .put = usb6fire_control_output_vol_put,> > + .private_value = 2> > + }> > +};> > Add the terminator in this patch. I know you fixed in the last patch> but, In general, try not to introduce a bug in each commit. Otherwise> it'll make bisection difficult.That's right, I've overseen this problem in the first place andstumbled upon it after patch 2 and 3 have already been sent. Thanks,Torsten

1 0

[alsa-devel] overrun in case of record
by Rajeev kumar 12 Jan '12

12 Jan '12

Hello Mark I am testing my audio driver on spear platform.During recording I am getting two issues. 1. Overrun issue 2. pcm_read error I am using arecord for capturing data. #arecord -f dat > t overrun!!! (at least 32.949 ms long) Overrun issue: The issue is only reproducible if we try to record for more than 40 sec.The strange part is that Instead of overrun condition generated, it will record complete data for 10sec.(This is verified with the number of bytes recorded). So could you please let me know whether this is an error condition or just a warning. pcm_read error: For this I have seen a bug is locked in the bugzila(*Bug 472469* <show_bug.cgi?id=472469> -arecord: input/output error ). and the status is closed saying insufficient data. Could you please let me know what is the meaning of this bug? Is it like, data is not available in the buffer and pcm_read from the user space fails? But why this situation will arise? Best Regards Rajeev

4 15

[alsa-devel] ASoC updates for 3.3
by Mark Brown 12 Jan '12

12 Jan '12

The following changes since commit d6b24507972341e5babdca8e25448325c9137232: Merge branch 'for-3.3' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc into topic/asoc (2012-01-11 15:30:53 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 The MFD patch is an incremental patch on top of the WM8994 changes being carried by ASoC this merge window. Axel Lin (2): ASoC: Fix build dependency for SND_ATMEL_SOC_SSC ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC Joachim Eastwood (1): ASoC: check for substream not channels_min in pcm engines Mark Brown (3): ASoC: Fix idma build after update for channel count check mfd: Put WM8994 into cache only mode when suspending ASoC: Dynamically allocate the rtd device for a non-empty release() drivers/mfd/wm8994-core.c | 2 + include/sound/soc.h | 6 ++-- sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel-pcm.c | 5 +-- sound/soc/blackfin/bf5xx-ac97-pcm.c | 5 +-- sound/soc/blackfin/bf5xx-i2s-pcm.c | 5 +-- sound/soc/blackfin/bf5xx-tdm-pcm.c | 5 +-- sound/soc/davinci/davinci-pcm.c | 5 +-- sound/soc/ep93xx/ep93xx-pcm.c | 5 +-- sound/soc/jz4740/jz4740-pcm.c | 5 +-- sound/soc/kirkwood/kirkwood-dma.c | 5 +-- sound/soc/mid-x86/sst_platform.c | 5 +-- sound/soc/omap/omap-pcm.c | 5 +-- sound/soc/samsung/dma.c | 5 +-- sound/soc/samsung/idma.c | 4 +- sound/soc/soc-core.c | 44 +++++++++++++++++++--------------- sound/soc/soc-dapm.c | 3 +- sound/soc/tegra/tegra_pcm.c | 5 +-- 18 files changed, 58 insertions(+), 63 deletions(-)

1 0

[alsa-devel] [PATCH] ASoC: Use regmap update bits operation for drivers using regmap
by Mark Brown 12 Jan '12

12 Jan '12

If a driver is using regmap directly ensure that we're coherent with non-ASoC register updates by using the regmap API directly to do our read/modify/write cycles. This will bypass the ASoC cache but drivers using regmap directly should not be using the ASoC cache. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 25 +++++++++++++++---------- sound/soc/soc-dapm.c | 27 +++++++++++++++++---------- sound/soc/soc-io.c | 1 + 4 files changed, 34 insertions(+), 20 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0992dff..fde163b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -559,6 +559,7 @@ struct snd_soc_codec { unsigned int ac97_created:1; /* Codec has been created by SoC */ unsigned int sysfs_registered:1; /* codec has been sysfs registered */ unsigned int cache_init:1; /* codec cache has been initialized */ + unsigned int using_regmap:1; /* using regmap access */ u32 cache_only; /* Suppress writes to hardware */ u32 cache_sync; /* Cache needs to be synced to hardware */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3986520..07ab062e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1865,23 +1865,28 @@ EXPORT_SYMBOL_GPL(snd_soc_bulk_write_raw); int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, unsigned int mask, unsigned int value) { - int change; + bool change; unsigned int old, new; int ret; - ret = snd_soc_read(codec, reg); - if (ret < 0) - return ret; - - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = snd_soc_write(codec, reg, new); + if (codec->using_regmap) { + ret = regmap_update_bits_check(codec->control_data, reg, + mask, value, &change); + } else { + ret = snd_soc_read(codec, reg); if (ret < 0) return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) + ret = snd_soc_write(codec, reg, new); } + if (ret < 0) + return ret; + return change; } EXPORT_SYMBOL_GPL(snd_soc_update_bits); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3ad1f59..3b8eaa8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -197,21 +197,28 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, unsigned short reg, unsigned int mask, unsigned int value) { - int change; + bool change; unsigned int old, new; int ret; - ret = soc_widget_read(w, reg); - if (ret < 0) - return ret; - - old = ret; - new = (old & ~mask) | (value & mask); - change = old != new; - if (change) { - ret = soc_widget_write(w, reg, new); + if (w->codec && w->codec->using_regmap) { + ret = regmap_update_bits_check(w->codec->control_data, + reg, mask, value, &change); + if (ret != 0) + return ret; + } else { + ret = soc_widget_read(w, reg); if (ret < 0) return ret; + + old = ret; + new = (old & ~mask) | (value & mask); + change = old != new; + if (change) { + ret = soc_widget_write(w, reg, new); + if (ret < 0) + return ret; + } } return change; diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index c8610cb..39ba507 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -140,6 +140,7 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ + codec->using_regmap = true; break; default: -- 1.7.7.3

1 0

[alsa-devel] [PATCH 1/2] ASoC: Add WM8962 DAC and ADC L/R swap support
by Mark Brown 12 Jan '12

12 Jan '12

Simple switches since there's no per-channel control. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 11125c9..e35237a 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2223,6 +2223,8 @@ SOC_DOUBLE_R_TLV("Sidetone Volume", WM8962_DAC_DSP_MIXING_1, SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8962_LEFT_DAC_VOLUME, WM8962_RIGHT_DAC_VOLUME, 1, 127, 0, digital_tlv), SOC_SINGLE("DAC High Performance Switch", WM8962_ADC_DAC_CONTROL_2, 0, 1, 0), +SOC_SINGLE("DAC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 5, 1, 0), +SOC_SINGLE("ADC L/R Swap Switch", WM8962_AUDIO_INTERFACE_0, 8, 1, 0), SOC_SINGLE("ADC High Performance Switch", WM8962_ADDITIONAL_CONTROL_1, 5, 1, 0), -- 1.7.7.3

1 1

[alsa-devel] [PATCH 4/4] snd-usb-6fire: add analog input volume control
by Torsten Schenk 12 Jan '12

12 Jan '12

- Add a stereo volume control for analog input channel pair 1/2. - Add missing end-of-list entry at end of list for array mute_elements and vol_elements Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-12 00:19:40.680844042 +0100 +++ b/sound/usb/6fire/control.c 2012-01-12 00:23:21.013835655 +0100 @@ -76,6 +76,17 @@ comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); } +static void usb6fire_control_input_vol_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + int i; + + if (comm_rt) + for (i = 0; i < 2; i++) + comm_rt->write8(comm_rt, 0x12, 0x1c + i, + rt->input_vol[i] & 0x3f); +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -257,6 +268,48 @@ return 0; } +static int usb6fire_control_input_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = -15; + uinfo->value.integer.max = 15; + return 0; +} + +static int usb6fire_control_input_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + int changed = 0; + + if (rt->input_vol[0] != ucontrol->value.integer.value[0]) { + rt->input_vol[0] = ucontrol->value.integer.value[0]; + changed = 1; + } + if (rt->input_vol[1] != ucontrol->value.integer.value[1]) { + rt->input_vol[1] = ucontrol->value.integer.value[1]; + changed = 1; + } + + if (changed) + usb6fire_control_input_vol_update(rt); + + return changed; +} + +static int usb6fire_control_input_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = rt->input_vol[0]; + ucontrol->value.integer.value[1] = rt->input_vol[1]; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -378,7 +431,8 @@ .get = usb6fire_control_output_vol_get, .put = usb6fire_control_output_vol_put, .private_value = 2 - } + }, + {} }; static struct __devinitdata snd_kcontrol_new mute_elements[] = { @@ -411,7 +465,8 @@ .get = usb6fire_control_output_mute_get, .put = usb6fire_control_output_mute_put, .private_value = 2 - } + }, + {} }; static struct __devinitdata snd_kcontrol_new elements[] = { @@ -442,6 +497,15 @@ .get = usb6fire_control_digital_thru_get, .put = usb6fire_control_digital_thru_put }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 1/2 Capture Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_input_vol_info, + .get = usb6fire_control_input_vol_get, + .put = usb6fire_control_input_vol_put + }, {} }; @@ -505,6 +569,7 @@ usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); usb6fire_control_output_mute_update(rt); + usb6fire_control_input_vol_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-11 19:54:50.651190007 +0100 +++ b/sound/usb/6fire/control.h 2012-01-11 23:44:43.212950616 +0100 @@ -45,6 +45,7 @@ bool usb_streaming; u8 output_vol[6]; u8 output_mute; + s8 input_vol[2]; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

2 1

[alsa-devel] [PATCH 2/4] snd-usb-6fire: add individual volume control for analog channels
by Torsten Schenk 12 Jan '12

12 Jan '12

Add a stereo volume control for every analog output pair 1/2, 3/4, 5/6. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-09 22:33:01.512182294 +0100 +++ b/sound/usb/6fire/control.c 2012-01-12 00:10:48.981232196 +0100 @@ -7,6 +7,10 @@ * Created: Jan 01, 2011 * Copyright: (C) Torsten Schenk * + * Thanks to: + * - Holger Ruckdeschel: he found out how to control individual channel + * volumes and introduced mute switch + * * 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 @@ -24,26 +28,6 @@ static char *line_phono_texts[2] = { "Line", "Phono" }; /* - * calculated with $value\[i\] = 128 \cdot sqrt[3]{\frac{i}{128}}$ - * this is done because the linear values cause rapid degredation - * of volume in the uppermost region. - */ -static const u8 log_volume_table[128] = { - 0x00, 0x19, 0x20, 0x24, 0x28, 0x2b, 0x2e, 0x30, 0x32, 0x34, - 0x36, 0x38, 0x3a, 0x3b, 0x3d, 0x3e, 0x40, 0x41, 0x42, 0x43, - 0x44, 0x46, 0x47, 0x48, 0x49, 0x4a, 0x4b, 0x4c, 0x4d, 0x4e, - 0x4e, 0x4f, 0x50, 0x51, 0x52, 0x53, 0x53, 0x54, 0x55, 0x56, - 0x56, 0x57, 0x58, 0x58, 0x59, 0x5a, 0x5b, 0x5b, 0x5c, 0x5c, - 0x5d, 0x5e, 0x5e, 0x5f, 0x60, 0x60, 0x61, 0x61, 0x62, 0x62, - 0x63, 0x63, 0x64, 0x65, 0x65, 0x66, 0x66, 0x67, 0x67, 0x68, - 0x68, 0x69, 0x69, 0x6a, 0x6a, 0x6b, 0x6b, 0x6c, 0x6c, 0x6c, - 0x6d, 0x6d, 0x6e, 0x6e, 0x6f, 0x6f, 0x70, 0x70, 0x70, 0x71, - 0x71, 0x72, 0x72, 0x73, 0x73, 0x73, 0x74, 0x74, 0x75, 0x75, - 0x75, 0x76, 0x76, 0x77, 0x77, 0x77, 0x78, 0x78, 0x78, 0x79, - 0x79, 0x7a, 0x7a, 0x7a, 0x7b, 0x7b, 0x7b, 0x7c, 0x7c, 0x7c, - 0x7d, 0x7d, 0x7d, 0x7e, 0x7e, 0x7e, 0x7f, 0x7f }; - -/* * data that needs to be sent to device. sets up card internal stuff. * values dumped from windows driver and filtered by trial'n'error. */ @@ -58,7 +42,7 @@ { 0x22, 0x03, 0x00 }, { 0x20, 0x03, 0x08 }, { 0x22, 0x04, 0x00 }, { 0x20, 0x04, 0x08 }, { 0x22, 0x05, 0x01 }, { 0x20, 0x05, 0x08 }, { 0x22, 0x04, 0x01 }, { 0x12, 0x04, 0x00 }, { 0x12, 0x05, 0x00 }, - { 0x12, 0x0d, 0x78 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, + { 0x12, 0x0d, 0x38 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, { 0x12, 0x23, 0x00 }, { 0x12, 0x06, 0x02 }, { 0x12, 0x03, 0x00 }, { 0x12, 0x02, 0x00 }, { 0x22, 0x03, 0x01 }, { 0 } /* TERMINATING ENTRY */ @@ -73,16 +57,15 @@ DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; -static void usb6fire_control_master_vol_update(struct control_runtime *rt) +static void usb6fire_control_output_vol_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; - if (comm_rt) { - /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 0x7f - - log_volume_table[rt->master_vol]); - /* unmute */ - comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); - } + int i; + + if (comm_rt) + for (i = 0; i < 6; i++) + comm_rt->write8(comm_rt, 0x12, 0x0f + i, + 180 - rt->output_vol[i]); } static void usb6fire_control_line_phono_update(struct control_runtime *rt) @@ -141,7 +124,6 @@ return ret; /* disable digital inputs and outputs */ - /* TODO: use spdif_x to enable/disable digital channels */ ret = comm_rt->write16(comm_rt, 0x02, 0x03, 0x00, 0x00); if (ret < 0) return ret; @@ -164,34 +146,61 @@ return -EINVAL; } -static int usb6fire_control_master_vol_info(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = 2; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 127; + uinfo->value.integer.max = 180; return 0; } -static int usb6fire_control_master_vol_put(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; int changed = 0; - if (rt->master_vol != ucontrol->value.integer.value[0]) { - rt->master_vol = ucontrol->value.integer.value[0]; - usb6fire_control_master_vol_update(rt); + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return 0; + } + + if (rt->output_vol[ch0] != ucontrol->value.integer.value[0]) { + rt->output_vol[ch0] = ucontrol->value.integer.value[0]; changed = 1; } + if (rt->output_vol[ch1] != ucontrol->value.integer.value[1]) { + rt->output_vol[ch1] = ucontrol->value.integer.value[1]; + changed = 1; + } + + if (changed) + usb6fire_control_output_vol_update(rt); + return changed; } -static int usb6fire_control_master_vol_get(struct snd_kcontrol *kcontrol, +static int usb6fire_control_output_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct control_runtime *rt = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = rt->master_vol; + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX "Invalid channel in volume control."); + return 0; + } + + ucontrol->value.integer.value[0] = rt->output_vol[ch0]; + ucontrol->value.integer.value[1] = rt->output_vol[ch1]; + return 0; } @@ -286,18 +295,42 @@ return 0; } -static struct __devinitdata snd_kcontrol_new elements[] = { +static struct __devinitdata snd_kcontrol_new vol_elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 1/2 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 0 + }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Volume", + .name = "Output 3/4 Playback Volume", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = usb6fire_control_master_vol_info, - .get = usb6fire_control_master_vol_get, - .put = usb6fire_control_master_vol_put + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 1 }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 5/6 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 2 + } +}; + +static struct __devinitdata snd_kcontrol_new elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line/Phono Capture Route", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -326,6 +359,39 @@ {} }; +static int usb6fire_control_add_virtual( + struct control_runtime *rt, + struct snd_card *card, + char *name, + struct snd_kcontrol_new *elems) +{ + int ret; + int i; + struct snd_kcontrol *vmaster = snd_ctl_make_virtual_master(name, NULL); + struct snd_kcontrol *control; + + if (!vmaster) + return -ENOMEM; + ret = snd_ctl_add(card, vmaster); + if (ret < 0) + return ret; + + i = 0; + while (elems[i].name) { + control = snd_ctl_new1(&elems[i], rt); + if (!control) + return -ENOMEM; + ret = snd_ctl_add(card, control); + if (ret < 0) + return ret; + ret = snd_ctl_add_slave(vmaster, control); + if (ret < 0) + return ret; + i++; + } + return 0; +} + int __devinit usb6fire_control_init(struct sfire_chip *chip) { int i; @@ -351,9 +417,17 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); - usb6fire_control_master_vol_update(rt); + usb6fire_control_output_vol_update(rt); usb6fire_control_streaming_update(rt); + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Volume", vol_elements); + if (ret) { + kfree(rt); + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + return ret; + } + i = 0; while (elements[i].name) { ret = snd_ctl_add(chip->card, snd_ctl_new1(&elements[i], rt)); diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 22:32:56.088002206 +0100 +++ b/sound/usb/6fire/control.h 2012-01-09 22:36:07.698544501 +0100 @@ -43,7 +43,7 @@ bool line_phono_switch; bool digital_thru_switch; bool usb_streaming; - u8 master_vol; + u8 output_vol[6]; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

2 1

[alsa-devel] [PATCH v2] ASoC: Route Mic Bias in Visstrim_M10 board.
by Javier Martin 12 Jan '12

12 Jan '12

Visstrim_M10 board uses an external microphone that can be enabled/disabled by the user --- Changes since v1: - Change "audio_map" to "aic32x4_dapm_routes". - Use snd_soc_card fields to register controls, routes and widgets. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/mx27vis-aic32x4.c | 24 ++++++++++++++++++++++++ 1 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 054110b..30adbe1 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -74,6 +74,24 @@ static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { .hw_params = mx27vis_aic32x4_hw_params, }; +static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = { + SOC_DAPM_PIN_SWITCH("External Mic"), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_MIC("External Mic", NULL), +}; + +static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { + {"Mic Bias", NULL, "External Mic"}, + {"IN1_R", NULL, "Mic Bias"}, + {"IN2_R", NULL, "Mic Bias"}, + {"IN3_R", NULL, "Mic Bias"}, + {"IN1_L", NULL, "Mic Bias"}, + {"IN2_L", NULL, "Mic Bias"}, + {"IN3_L", NULL, "Mic Bias"}, +}; + static struct snd_soc_dai_link mx27vis_aic32x4_dai = { .name = "tlv320aic32x4", .stream_name = "TLV320AIC32X4", @@ -88,6 +106,12 @@ static struct snd_soc_card mx27vis_aic32x4 = { .name = "visstrim_m10-audio", .dai_link = &mx27vis_aic32x4_dai, .num_links = 1, + .controls = mx27vis_aic32x4_controls, + .num_controls = ARRAY_SIZE(mx27vis_aic32x4_controls), + .dapm_widgets = aic32x4_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), + .dapm_routes = aic32x4_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; static struct platform_device *mx27vis_aic32x4_snd_device; -- 1.7.0.4

2 1

[alsa-devel] [PATCH 3/4] snd-usb-6fire: add mute controls for analog outputs
by Torsten Schenk 12 Jan '12

12 Jan '12

Add an individual mute control for every analog output channel. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-12 00:10:48.981232196 +0100 +++ b/sound/usb/6fire/control.c 2012-01-12 00:19:40.680844042 +0100 @@ -68,6 +68,14 @@ 180 - rt->output_vol[i]); } +static void usb6fire_control_output_mute_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + + if (comm_rt) + comm_rt->write8(comm_rt, 0x12, 0x0e, ~rt->output_mute); +} + static void usb6fire_control_line_phono_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; @@ -204,6 +212,51 @@ return 0; } +static int usb6fire_control_output_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + u8 old = rt->output_mute; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + rt->output_mute &= ~((1 << ch0) | (1 << ch1)); + rt->output_mute |= ucontrol->value.integer.value[0] << ch0; + rt->output_mute |= ucontrol->value.integer.value[1] << ch1; + + if (rt->output_mute != old) + usb6fire_control_output_mute_update(rt); + + return rt->output_mute != old; +} + +static int usb6fire_control_output_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + ucontrol->value.integer.value[0] = 1 & (rt->output_mute >> ch0); + ucontrol->value.integer.value[1] = 1 & (rt->output_mute >> ch1); + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -328,6 +381,39 @@ } }; +static struct __devinitdata snd_kcontrol_new mute_elements[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 1/2 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 3/4 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + .private_value = 1 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 5/6 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_mute_get, + .put = usb6fire_control_output_mute_put, + .private_value = 2 + } +}; + static struct __devinitdata snd_kcontrol_new elements[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -418,13 +504,21 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); usb6fire_control_output_vol_update(rt); + usb6fire_control_output_mute_update(rt); usb6fire_control_streaming_update(rt); ret = usb6fire_control_add_virtual(rt, chip->card, "Master Playback Volume", vol_elements); if (ret) { + snd_printk(KERN_ERR PREFIX "cannot add control.\n"); kfree(rt); + return ret; + } + ret = usb6fire_control_add_virtual(rt, chip->card, + "Master Playback Switch", mute_elements); + if (ret) { snd_printk(KERN_ERR PREFIX "cannot add control.\n"); + kfree(rt); return ret; } diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 22:36:07.698544501 +0100 +++ b/sound/usb/6fire/control.h 2012-01-11 19:54:50.651190007 +0100 @@ -44,6 +44,7 @@ bool digital_thru_switch; bool usb_streaming; u8 output_vol[6]; + u8 output_mute; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

1 0

[alsa-devel] [PATCH 1/4] snd-usb-6fire: remove driver version information
by Torsten Schenk 12 Jan '12

12 Jan '12

Remove unused driver version information from the individual files. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> diff -Nur a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c --- a/sound/usb/6fire/chip.c 2012-01-09 21:55:55.863579333 +0100 +++ b/sound/usb/6fire/chip.c 2012-01-09 22:33:49.857786986 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify @@ -29,7 +28,7 @@ #include <sound/initval.h> MODULE_AUTHOR("Torsten Schenk <torsten.schenk(a)zoho.com>"); -MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.0"); +MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver"); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); diff -Nur a/sound/usb/6fire/chip.h b/sound/usb/6fire/chip.h --- a/sound/usb/6fire/chip.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/chip.h 2012-01-09 22:33:38.469409108 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c --- a/sound/usb/6fire/comm.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/comm.c 2012-01-09 22:33:11.620517902 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h --- a/sound/usb/6fire/comm.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/comm.h 2012-01-09 22:33:16.900693132 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/common.h b/sound/usb/6fire/common.h --- a/sound/usb/6fire/common.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/common.h 2012-01-09 22:33:24.636949983 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/control.c 2012-01-09 22:33:01.512182294 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/control.h 2012-01-09 22:32:56.088002206 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c --- a/sound/usb/6fire/firmware.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/firmware.c 2012-01-09 22:33:30.933158868 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c --- a/sound/usb/6fire/midi.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/midi.c 2012-01-09 22:35:07.356357430 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h --- a/sound/usb/6fire/midi.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/midi.h 2012-01-09 22:33:58.866085888 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c --- a/sound/usb/6fire/pcm.c 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/pcm.c 2012-01-09 22:34:04.546274285 +0100 @@ -5,7 +5,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify diff -Nur a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h --- a/sound/usb/6fire/pcm.h 2012-01-09 21:55:55.000000000 +0100 +++ b/sound/usb/6fire/pcm.h 2012-01-09 22:34:07.798382278 +0100 @@ -3,7 +3,6 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify

1 0

[alsa-devel] [PATCH] ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
by Liam Girdwood 11 Jan '12

11 Jan '12

The DL1 PDM interface adds a little gain depending on the output device. Add a method to retrieve the gain value for machine driver usage. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/codecs/twl6040.c | 23 +++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 1 + 2 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4ae6b82..3b00ee7 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -33,6 +33,7 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> @@ -1042,6 +1043,28 @@ int twl6040_get_hs_step_size(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(twl6040_get_hs_step_size); +int twl6040_get_dl1_gain(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + + if (snd_soc_dapm_get_pin_status(dapm, "EP")) + return -1; /* -1dB */ + + if (snd_soc_dapm_get_pin_status(dapm, "HSOR") || + snd_soc_dapm_get_pin_status(dapm, "HSOL")) { + + u8 val = snd_soc_read(codec, TWL6040_REG_HSLCTL); + if (val & TWL6040_HSDACMODE) + /* HSDACL in LP mode */ + return -8; /* -8dB */ + else + /* HSDACL in HP mode */ + return -1; /* -1dB */ + } + return 0; /* 0dB */ +} +EXPORT_SYMBOL_GPL(twl6040_get_dl1_gain); + static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* Capture gains */ SOC_DOUBLE_TLV("Capture Preamplifier Volume", diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index 0028699..b580c9d 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -39,5 +39,6 @@ void twl6040_hs_jack_detect(struct snd_soc_codec *codec, int twl6040_get_clk_id(struct snd_soc_codec *codec); int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim); int twl6040_get_hs_step_size(struct snd_soc_codec *codec); +int twl6040_get_dl1_gain(struct snd_soc_codec *codec); #endif /* End of __TWL6040_H__ */ -- 1.7.5.4

2 5

[alsa-devel] [FIXED PATCH - ice1724 1/1] ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
by pavel.hofman＠ivitera.com 11 Jan '12

11 Jan '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Add the capture pcm only if there is at least one ADC configured in the SYSCONF register. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/sound/pci/ice1712/envy24ht.h b/sound/pci/ice1712/envy24ht.h index a0c5e00..4ca33a8 100644 --- a/sound/pci/ice1712/envy24ht.h +++ b/sound/pci/ice1712/envy24ht.h @@ -66,6 +66,7 @@ enum { #define VT1724_CFG_CLOCK384 0x40 /* 16.9344Mhz, 44.1kHz*384 */ #define VT1724_CFG_MPU401 0x20 /* MPU401 UARTs */ #define VT1724_CFG_ADC_MASK 0x0c /* one, two or one and S/PDIF, stereo ADCs */ +#define VT1724_CFG_ADC_NONE 0x0c /* no ADCs */ #define VT1724_CFG_DAC_MASK 0x03 /* one, two, three, four stereo DACs */ #define VT1724_REG_AC97_CFG 0x05 /* byte */ diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 9e18d3a..e797823 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1117,14 +1117,21 @@ static struct snd_pcm_ops snd_vt1724_capture_pro_ops = { static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device) { struct snd_pcm *pcm; - int err; + int capt, err; - err = snd_pcm_new(ice->card, "ICE1724", device, 1, 1, &pcm); + if ((ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_ADC_MASK) == + VT1724_CFG_ADC_NONE) + capt = 0; + else + capt = 1; + err = snd_pcm_new(ice->card, "ICE1724", device, 1, capt, &pcm); if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_vt1724_playback_pro_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_vt1724_capture_pro_ops); + if (capt) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_vt1724_capture_pro_ops); pcm->private_data = ice; pcm->info_flags = 0; -- 1.7.0.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: hdspm - Refactor serial number to avoid code duplication
by Adrian Knoth 11 Jan '12

11 Jan '12

The serial number is used multiple times in hdspm.c. Since it belongs to the card, let's store it in struct hdspm and refer to it whenever necessary. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d623451..1609253 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -941,6 +941,8 @@ struct hdspm { cycles_t last_interrupt; + unsigned int serial; + struct hdspm_peak_rms peak_rms; }; @@ -4694,7 +4696,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n", (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF, - (hdspm_read(hdspm, HDSPM_midiStatusIn0)>>8) & 0xFFFFFF); + hdspm->serial); snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n", hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase); @@ -6266,8 +6268,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, hdspm_version.card_type = hdspm->io_type; strncpy(hdspm_version.cardname, hdspm->card_name, sizeof(hdspm_version.cardname)); - hdspm_version.serial = (hdspm_read(hdspm, - HDSPM_midiStatusIn0)>>8) & 0xFFFFFF; + hdspm_version.serial = hdspm->serial; hdspm_version.firmware_rev = hdspm->firmware_rev; hdspm_version.addons = 0; if (hdspm->tco) @@ -6866,12 +6867,14 @@ static int __devinit snd_hdspm_probe(struct pci_dev *pci, } if (hdspm->io_type != MADIface) { + hdspm->serial = (hdspm_read(hdspm, + HDSPM_midiStatusIn0)>>8) & 0xFFFFFF; sprintf(card->shortname, "%s_%x", hdspm->card_name, - (hdspm_read(hdspm, HDSPM_midiStatusIn0)>>8) & 0xFFFFFF); + hdspm->serial); sprintf(card->longname, "%s S/N 0x%x at 0x%lx, irq %d", hdspm->card_name, - (hdspm_read(hdspm, HDSPM_midiStatusIn0)>>8) & 0xFFFFFF, + hdspm->serial, hdspm->port, hdspm->irq); } else { sprintf(card->shortname, "%s", hdspm->card_name); -- 1.7.7.3

2 7

[alsa-devel] Questions on the ALSA user lib interfaces.
by naveen440＠comcast.net 11 Jan '12

11 Jan '12

Hello, I am new to ALSA interfaces and i wanted to understand some of the interfaces clearly to make sure that i write the application correctly. Can someone provide with some information on the usage of the following interfaces? What is the usage model for using them. 1. Config interfaces (snd_config_top() and the rest of them). Are they to be used to configure the ALSA userlib? Some example code or pseudo code on this topic would be useful. 2. Higher level control interfaces (snd_hctl_ and the rest of them). Again some example code explaining the usage would be useful. 3. Setup controls (snd_sctl and the rest of them). Again some usage scenario's on this context would be useful to know. Thanks in advance. Naveen

1 0

[alsa-devel] [PATCH - ice1724 1/3] ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
by pavel.hofman＠ivitera.com 10 Jan '12

10 Jan '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Add the capture pcm only if there is at least one ADC configured in the SYSCONF register. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 9e18d3a..e797823 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1117,14 +1117,21 @@ static struct snd_pcm_ops snd_vt1724_capture_pro_ops = { static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device) { struct snd_pcm *pcm; - int err; + int capt, err; - err = snd_pcm_new(ice->card, "ICE1724", device, 1, 1, &pcm); + if ((ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_ADC_MASK) == + VT1724_CFG_ADC_NONE) + capt = 0; + else + capt = 1; + err = snd_pcm_new(ice->card, "ICE1724", device, 1, capt, &pcm); if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_vt1724_playback_pro_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_vt1724_capture_pro_ops); + if (capt) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_vt1724_capture_pro_ops); pcm->private_data = ice; pcm->info_flags = 0; -- 1.7.0.4

1 2

[alsa-devel] [PATCH] ASoC: Make WM8962 I2C usage unconditional
by Mark Brown 10 Jan '12

10 Jan '12

We only support I2C so no need to ifdef. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 14 +------------- 1 files changed, 1 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e65c063..11125c9 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -4187,7 +4187,6 @@ static const struct regmap_config wm8962_regmap = { .cache_type = REGCACHE_RBTREE, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8962_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -4316,27 +4315,16 @@ static struct i2c_driver wm8962_i2c_driver = { .remove = __devexit_p(wm8962_i2c_remove), .id_table = wm8962_i2c_id, }; -#endif static int __init wm8962_modinit(void) { - int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&wm8962_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register WM8962 I2C driver: %d\n", - ret); - } -#endif - return 0; + return i2c_add_driver(&wm8962_i2c_driver); } module_init(wm8962_modinit); static void __exit wm8962_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8962_i2c_driver); -#endif } module_exit(wm8962_exit); -- 1.7.7.3

1 0

[alsa-devel] [PATCH] ASoC: Return proper error if i2c_add_driver fails
by Axel Lin 10 Jan '12

10 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8523.c | 4 ++-- sound/soc/codecs/wm8711.c | 4 ++-- sound/soc/codecs/wm8737.c | 4 ++-- sound/soc/codecs/wm8962.c | 4 ++-- sound/soc/codecs/wm8991.c | 4 ++-- 5 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 7fea2c3..84b2ddc 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -559,7 +559,7 @@ static struct i2c_driver wm8523_i2c_driver = { static int __init wm8523_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8523_i2c_driver); if (ret != 0) { @@ -567,7 +567,7 @@ static int __init wm8523_modinit(void) ret); } #endif - return 0; + return ret; } module_init(wm8523_modinit); diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 0b76d1d..727fb57 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -494,7 +494,7 @@ static struct i2c_driver wm8711_i2c_driver = { static int __init wm8711_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8711_i2c_driver); if (ret != 0) { @@ -509,7 +509,7 @@ static int __init wm8711_modinit(void) ret); } #endif - return 0; + return ret; } module_init(wm8711_modinit); diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index ff95e62c..9a19a44 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -728,7 +728,7 @@ static struct spi_driver wm8737_spi_driver = { static int __init wm8737_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8737_i2c_driver); if (ret != 0) { @@ -743,7 +743,7 @@ static int __init wm8737_modinit(void) ret); } #endif - return 0; + return ret; } module_init(wm8737_modinit); diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 296de4e..30bc788 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -4288,7 +4288,7 @@ static struct i2c_driver wm8962_i2c_driver = { static int __init wm8962_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8962_i2c_driver); if (ret != 0) { @@ -4296,7 +4296,7 @@ static int __init wm8962_modinit(void) ret); } #endif - return 0; + return ret; } module_init(wm8962_modinit); diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 7ee40da..d877487 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1402,13 +1402,13 @@ static struct i2c_driver wm8991_i2c_driver = { static int __init wm8991_modinit(void) { - int ret; + int ret = 0; ret = i2c_add_driver(&wm8991_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8991 I2C driver: %d\n", ret); } - return 0; + return ret; } module_init(wm8991_modinit); -- 1.7.5.4

2 1

[alsa-devel] [PATCH RESEND] ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
by Axel Lin 10 Jan '12

10 Jan '12

commit 739be96 "ASoC: Fix build dependency for SND_ATMEL_SOC_SSC" introduces below build warnings: drivers/misc/Kconfig:212:error: recursive dependency detected! drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC SND_ATMEL_SOC_SSC needs ATMEL_SSC to pass compilation. This patch remove the "select ATMEL_SSC" from SND_ATMEL_SOC_SSC to avoid above warnings. And then ensures all the machine drivers that select SND_ATMEL_SOC_SSC need to depend on ATMEL_SSC. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/atmel/Kconfig | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a4d6742..72b09cf 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -9,7 +9,6 @@ config SND_ATMEL_SOC config SND_ATMEL_SOC_SSC tristate depends on SND_ATMEL_SOC - select ATMEL_SSC help Say Y or M if you want to add support for codecs the ATMEL SSC interface. You will also needs to select the individual @@ -27,7 +26,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" - depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_SSC select SND_SOC_TLV320AIC23 help -- 1.7.5.4

2 1

[alsa-devel] MXS: How to configure SGTL5000 ASoC as master?
by Lauri Hintsala 10 Jan '12

10 Jan '12

Hi, SGTL5000 ASoC is hard coded to be slave in sound/soc/mxs/mxs-sgtl5000.c. In our HW design mclk of ASoC is sourced from external oscillator. Because of our design ASoC should be configured as master. In that case master clock of SAIF should be disabled. Is there any plan to have configuration for mxs-sgtl5000 driver? What is preferred way to solve this issue? BR, Lauri Hintsala p.s. I resent this mail because I wasn't subscriber of alsa-devel mailing list.

2 2

[alsa-devel] [PATCH] ASoC: utils - Add support for a dummy codec driver.
by Liam Girdwood 10 Jan '12

10 Jan '12

This is useful to create dummy codec devices where we need to have some DAI links without a real Codec. e.g. could be used to represent dumb FM, MODEM, etc This is also used by dynamic PCM for DAI links that have no codec. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-utils.c | 20 +++++++++++++++++++- 1 files changed, 19 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 4220bb0..329122d 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -89,14 +89,32 @@ static struct snd_soc_platform_driver dummy_platform = { .ops = &dummy_dma_ops, }; +static struct snd_soc_codec_driver dummy_codec; +static struct snd_soc_dai_driver dummy_dai = { + .name = "snd-soc-dummy-dai", +}; + static __devinit int snd_soc_dummy_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &dummy_platform); + int ret; + + ret = snd_soc_register_codec(&pdev->dev, &dummy_codec, &dummy_dai, 1); + if (ret < 0) + return ret; + + ret = snd_soc_register_platform(&pdev->dev, &dummy_platform); + if (ret < 0) { + snd_soc_unregister_codec(&pdev->dev); + return ret; + } + + return ret; } static __devexit int snd_soc_dummy_remove(struct platform_device *pdev) { snd_soc_unregister_platform(&pdev->dev); + snd_soc_unregister_codec(&pdev->dev); return 0; } -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
by Axel Lin 10 Jan '12

10 Jan '12

commit 739be96 "ASoC: Fix build dependency for SND_ATMEL_SOC_SSC" introduces below build warnings: drivers/misc/Kconfig:212:error: recursive dependency detected! drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC SND_ATMEL_SOC_SSC needs ATMEL_SSC to pass compilation. This patch removes the "select ATMEL_SSC" from SND_ATMEL_SOC_SSC to avoid above warnings. And then ensures all the machine drivers that select SND_ATMEL_SOC_SSC need to depend on ATMEL_SSC. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/atmel/Kconfig | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a4d6742..72b09cf 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -9,7 +9,6 @@ config SND_ATMEL_SOC config SND_ATMEL_SOC_SSC tristate depends on SND_ATMEL_SOC - select ATMEL_SSC help Say Y or M if you want to add support for codecs the ATMEL SSC interface. You will also needs to select the individual @@ -27,7 +26,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" - depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_SSC select SND_SOC_TLV320AIC23 help -- 1.7.5.4

2 4

[alsa-devel] [PATCH] ASoC: twl6040 - add method to query HS DC offset step size in mV
by Liam Girdwood 10 Jan '12

10 Jan '12

Provide a method for mach drivers to query the HS DC offset step size in mV. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/codecs/twl6040.c | 13 +++++++++++++ sound/soc/codecs/twl6040.h | 1 + 2 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3376e6f..4ae6b82 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1029,6 +1029,19 @@ int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim) } EXPORT_SYMBOL_GPL(twl6040_get_trim_value); +int twl6040_get_hs_step_size(struct snd_soc_codec *codec) +{ + struct twl6040 *twl6040 = codec->control_data; + + if (twl6040_get_revid(twl6040) < TWL6040_REV_ES1_2) + /* For ES under ES_1.3 HS step is 2 mV */ + return 2; + else + /* For ES_1.3 HS step is 1 mV */ + return 1; +} +EXPORT_SYMBOL_GPL(twl6040_get_hs_step_size); + static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* Capture gains */ SOC_DOUBLE_TLV("Capture Preamplifier Volume", diff --git a/sound/soc/codecs/twl6040.h b/sound/soc/codecs/twl6040.h index a83277b..0028699 100644 --- a/sound/soc/codecs/twl6040.h +++ b/sound/soc/codecs/twl6040.h @@ -38,5 +38,6 @@ void twl6040_hs_jack_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, int report); int twl6040_get_clk_id(struct snd_soc_codec *codec); int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim); +int twl6040_get_hs_step_size(struct snd_soc_codec *codec); #endif /* End of __TWL6040_H__ */ -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Fix build dependency for SND_AT91_SOC_AFEB9260
by Axel Lin 10 Jan '12

10 Jan '12

Make SND_AT91_SOC_AFEB9260 depend on ATMEL_SSC to fix below build errors: LD .tmp_vmlinux1 sound/built-in.o: In function `atmel_ssc_remove': sound/soc/atmel/atmel_ssc_dai.c:713: undefined reference to `ssc_free' sound/built-in.o: In function `atmel_ssc_probe': sound/soc/atmel/atmel_ssc_dai.c:700: undefined reference to `ssc_request' sound/built-in.o: In function `atmel_ssc_set_audio': sound/soc/atmel/atmel_ssc_dai.c:845: undefined reference to `ssc_request' sound/soc/atmel/atmel_ssc_dai.c:851: undefined reference to `ssc_free' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch replaces "[PATCH] ASoC: Fix build dependency for SND_ATMEL_SOC_SSC". sound/soc/atmel/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1fcc81..72b09cf 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -26,7 +26,7 @@ config SND_AT91_SOC_SAM9G20_WM8731 config SND_AT91_SOC_AFEB9260 tristate "SoC Audio support for AFEB9260 board" - depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC + depends on ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_SSC select SND_SOC_TLV320AIC23 help -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Fix recursive dependency for SND_ATMEL_SOC_SSC
by Fabio Estevam 10 Jan '12

10 Jan '12

commit 739be96 (ASoC: Fix build dependency for SND_ATMEL_SOC_SSC) introduced the following recursive dependency: drivers/misc/Kconfig:212:error: recursive dependency detected! drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC Fix it by moving ATMEL_SSC to the 'depends' line. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- This applies against linux-next and doing a 'make randconfig' triggers the error. sound/soc/atmel/Kconfig | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a4d6742..509b152 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -8,8 +8,7 @@ config SND_ATMEL_SOC config SND_ATMEL_SOC_SSC tristate - depends on SND_ATMEL_SOC - select ATMEL_SSC + depends on SND_ATMEL_SOC && ATMEL_SSC help Say Y or M if you want to add support for codecs the ATMEL SSC interface. You will also needs to select the individual -- 1.7.1

2 1

[alsa-devel] [PATCH v2] ASoC: Fix recursive dependency for SND_AT91_SOC_SAM9G20_WM8731
by Fabio Estevam 10 Jan '12

10 Jan '12

commit 739be96 (ASoC: Fix build dependency for SND_ATMEL_SOC_SSC) introduced the following recursive dependency: drivers/misc/Kconfig:212:error: recursive dependency detected! drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC Fix it by moving ATMEL_SSC to the 'depends' line. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- This applies against linux-next and doing a 'make randconfig' triggers the error. Change since v1: - Get rid of the recursive errors by fixing SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a4d6742..bb49b7d 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -17,8 +17,7 @@ config SND_ATMEL_SOC_SSC config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" - depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC && \ - AT91_PROGRAMMABLE_CLOCKS + depends on ARCH_AT91SAM9G20 && AT91_PROGRAMMABLE_CLOCKS select SND_ATMEL_SOC_SSC select SND_SOC_WM8731 help -- 1.7.1

2 1

[alsa-devel] Public ridicule due to sound/soc/soc-core.c abuse of the driver model
by Greg KH 10 Jan '12

10 Jan '12

Hi Frank, Liam, and sound maintainers. It was recently pointed out to me that the sound/soc/soc-core.c is flagrantly abusing the driver model by providing "empty" release functions, just to keep the kernel from complaining: /* stop no dev release warning */ static void soc_ac97_device_release(struct device *dev){} .... static void rtd_release(struct device *dev) {} As per the documentation in the kernel tree[1], I now get to publicly mock you for thinking you know better than the driver model. Come on people, do you think that I wrote the code in the kernel that produces those errors just for fun? It was telling you to fix your code by providing a function to free the structure that is being released, not to try to trick the kernel because you think you know better. The kernel was trying to help you out here, to get the programming model correct, in a place that was commonly misunderstood. But, by you trying to be "smart", you just proved that your code was incorrect and buggy. Please fix this for the 3.3 kernel release. Yes, I know it's been there for a number of years, but that still doesn't make it correct. Especially as stuff like this tends to get cut-and-pasted over time. thanks, greg k-h [1] Documentation/kobject.txt

3 14

[alsa-devel] [PATCH] ASoC: core - Improve card registration error messaging for large DAI links.
by Liam Girdwood 10 Jan '12

10 Jan '12

Print out the offending DAI link entry when a naming error occurs. Makes thing easier to debug for machines with a large number of DAI links. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index acbb960..e1d8433 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2854,7 +2854,8 @@ int snd_soc_register_card(struct snd_soc_card *card) */ if (!!link->codec_name == !!link->codec_of_node) { dev_err(card->dev, - "Neither/both codec name/of_node are set\n"); + "Neither/both codec name/of_node are set for %s\n", + link->name); return -EINVAL; } @@ -2864,7 +2865,7 @@ int snd_soc_register_card(struct snd_soc_card *card) */ if (link->platform_name && link->platform_of_node) { dev_err(card->dev, - "Both platform name/of_node are set\n"); + "Both platform name/of_node are set for %s\n", link->name); return -EINVAL; } @@ -2874,7 +2875,8 @@ int snd_soc_register_card(struct snd_soc_card *card) */ if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { dev_err(card->dev, - "Neither/both cpu_dai name/of_node are set\n"); + "Neither/both cpu_dai name/of_node are set for %s\n", + link->name); return -EINVAL; } } -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Dynamically allocate the rtd device for a non-empty release
by Mark Brown 09 Jan '12

09 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Greg, you asked for this to be included in 3.3 but I really don't see any reason for doing so - this isn't exactly new code and there aren't any practical issues being reported. Is there any actual rush? include/sound/soc.h | 6 +++--- sound/soc/soc-core.c | 34 +++++++++++++++++++++------------- 2 files changed, 24 insertions(+), 16 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a7425e9..0a56767 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -862,7 +862,7 @@ struct snd_soc_card { /* SoC machine DAI configuration, glues a codec and cpu DAI together */ struct snd_soc_pcm_runtime { - struct device dev; + struct device *dev; struct snd_soc_card *card; struct snd_soc_dai_link *dai_link; struct mutex pcm_mutex; @@ -954,12 +954,12 @@ static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platfo static inline void snd_soc_pcm_set_drvdata(struct snd_soc_pcm_runtime *rtd, void *data) { - dev_set_drvdata(&rtd->dev, data); + dev_set_drvdata(rtd->dev, data); } static inline void *snd_soc_pcm_get_drvdata(struct snd_soc_pcm_runtime *rtd) { - return dev_get_drvdata(&rtd->dev); + return dev_get_drvdata(rtd->dev); } static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index dcc4b62..9d8a631 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -884,9 +884,9 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(&rtd->dev, &dev_attr_pmdown_time); - device_remove_file(&rtd->dev, &dev_attr_codec_reg); - device_unregister(&rtd->dev); + device_remove_file(rtd->dev, &dev_attr_pmdown_time); + device_remove_file(rtd->dev, &dev_attr_codec_reg); + device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1061,7 +1061,10 @@ err_probe: return ret; } -static void rtd_release(struct device *dev) {} +static void rtd_release(struct device *dev) +{ + kfree(dev); +} static int soc_post_component_init(struct snd_soc_card *card, struct snd_soc_codec *codec, @@ -1104,11 +1107,16 @@ static int soc_post_component_init(struct snd_soc_card *card, /* register the rtd device */ rtd->codec = codec; - rtd->dev.parent = card->dev; - rtd->dev.release = rtd_release; - rtd->dev.init_name = name; + + rtd->dev = kzalloc(sizeof(struct device), GFP_KERNEL); + if (!rtd->dev) + return -ENOMEM; + device_initialize(rtd->dev); + rtd->dev->parent = card->dev; + rtd->dev->release = rtd_release; + rtd->dev->init_name = name; mutex_init(&rtd->pcm_mutex); - ret = device_register(&rtd->dev); + ret = device_add(rtd->dev); if (ret < 0) { dev_err(card->dev, "asoc: failed to register runtime device: %d\n", ret); @@ -1117,14 +1125,14 @@ static int soc_post_component_init(struct snd_soc_card *card, rtd->dev_registered = 1; /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(&rtd->dev); + ret = snd_soc_dapm_sys_add(rtd->dev); if (ret < 0) dev_err(codec->dev, "asoc: failed to add codec dapm sysfs entries: %d\n", ret); /* add codec sysfs entries */ - ret = device_create_file(&rtd->dev, &dev_attr_codec_reg); + ret = device_create_file(rtd->dev, &dev_attr_codec_reg); if (ret < 0) dev_err(codec->dev, "asoc: failed to add codec sysfs files: %d\n", ret); @@ -1213,7 +1221,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (ret) return ret; - ret = device_create_file(&rtd->dev, &dev_attr_pmdown_time); + ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); if (ret < 0) printk(KERN_WARNING "asoc: failed to add pmdown_time sysfs\n"); @@ -1311,8 +1319,8 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(&rtd->dev, &dev_attr_codec_reg); - device_unregister(&rtd->dev); + device_remove_file(rtd->dev, &dev_attr_codec_reg); + device_del(rtd->dev); rtd->dev_registered = 0; } -- 1.7.7.3

2 2

Re: [alsa-devel] linux-next: Tree for Jan 9 (sound/soc vs. drivers/misc)
by Randy Dunlap 09 Jan '12

09 Jan '12

On 01/08/2012 10:04 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20120106: > > This tree is still broken for sparc64 defconfig (at least). AFAICT, all builds (on X86_32 or X86_64) have this kconfig warning: drivers/misc/Kconfig:212:error: recursive dependency detected! drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC -- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

1 0

[alsa-devel] [PATCH] ASoC: Route Mic Bias in Visstrim_M10 board.
by Javier Martin 09 Jan '12

09 Jan '12

Visstrim_M10 board uses an external microphone that can be enabled/disabled by the user. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> --- sound/soc/imx/mx27vis-aic32x4.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 38 insertions(+), 0 deletions(-) diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c index 054110b..7e55a99 100644 --- a/sound/soc/imx/mx27vis-aic32x4.c +++ b/sound/soc/imx/mx27vis-aic32x4.c @@ -74,6 +74,43 @@ static struct snd_soc_ops mx27vis_aic32x4_snd_ops = { .hw_params = mx27vis_aic32x4_hw_params, }; +static const struct snd_kcontrol_new mx27vis_aic32x4_controls[] = { + SOC_DAPM_PIN_SWITCH("External Mic"), +}; + +static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { + SND_SOC_DAPM_MIC("External Mic", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"Mic Bias", NULL, "External Mic"}, + {"IN1_R", NULL, "Mic Bias"}, + {"IN2_R", NULL, "Mic Bias"}, + {"IN3_R", NULL, "Mic Bias"}, + {"IN1_L", NULL, "Mic Bias"}, + {"IN2_L", NULL, "Mic Bias"}, + {"IN3_L", NULL, "Mic Bias"}, +}; + +static int mx27vis_aic32x4_dai_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + int err = 0; + + snd_soc_dapm_new_controls(dapm, aic32x4_dapm_widgets, + ARRAY_SIZE(aic32x4_dapm_widgets)); + + err = snd_soc_add_controls(codec, mx27vis_aic32x4_controls, + ARRAY_SIZE(mx27vis_aic32x4_controls)); + if (err < 0) + return err; + + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + static struct snd_soc_dai_link mx27vis_aic32x4_dai = { .name = "tlv320aic32x4", .stream_name = "TLV320AIC32X4", @@ -82,6 +119,7 @@ static struct snd_soc_dai_link mx27vis_aic32x4_dai = { .codec_name = "tlv320aic32x4.0-0018", .cpu_dai_name = "imx-ssi.0", .ops = &mx27vis_aic32x4_snd_ops, + .init = mx27vis_aic32x4_dai_init, }; static struct snd_soc_card mx27vis_aic32x4 = { -- 1.7.0.4

2 1

[alsa-devel] hdsp: potential oops in snd_hdsp_info_pref_sync_ref()
by Dan Carpenter 09 Jan '12

09 Jan '12

My static check complains that there is a potential oops here but I'm not sure what the correct fix is. sound/pci/rme9652/hdsp.c 2631 switch (hdsp->io_type) { 2632 case Digiface: 2633 case H9652: 2634 uinfo->value.enumerated.items = 6; 2635 break; 2636 case Multiface: 2637 uinfo->value.enumerated.items = 4; 2638 break; 2639 case H9632: 2640 uinfo->value.enumerated.items = 3; 2641 break; 2642 default: 2643 uinfo->value.enumerated.items = 0; ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ We set this to zero here. 2644 break; 2645 } 2646 2647 if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ enumerated.item is unsigned int so it's always greater or equal to zero. 2648 uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; We set it to UINT_MAX here. 2649 strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Reading UINT_MAX beyond the end of the array here will probably oops. 2650 return 0; There is a similar issue in snd_asihpi_cmode_info(). regards, dan carpenter

2 1

[alsa-devel] patch for jack plugin
by Fons Adriaensen 09 Jan '12

09 Jan '12

The following patch will make snd_pcm_poll_descriptors_revents() return the correct value when used on a jack plugin device. The current version returns POLLIN regardless of capture or playback. --- pcm_jack_orig.c 2011-12-29 16:42:07.000000000 +0100 +++ pcm_jack.c 2011-12-29 16:51:18.000000000 +0100 @@ -77,17 +77,18 @@ return 0; } -static int snd_pcm_jack_poll_revents(snd_pcm_ioplug_t *io ATTRIBUTE_UNUSED, +static int snd_pcm_jack_poll_revents(snd_pcm_ioplug_t *io, struct pollfd *pfds, unsigned int nfds, unsigned short *revents) { static char buf[1]; assert(pfds && nfds == 1 && revents); - read(pfds[0].fd, buf, 1); - - *revents = pfds[0].revents; + *revents = pfds[0].revents & ~(POLLIN | POLLOUT); + if (pfds[0].revents & POLLIN) { + *revents |= (io->stream == SND_PCM_STREAM_PLAYBACK) ? POLLOUT : POLLIN; + } return 0; } Ciao, -- FA Vor uns liegt ein weites Tal, die Sonne scheint - ein Glitzerstrahl.

2 1

[alsa-devel] imac27 12,2 (2011) model for patch_cirrus.c
by Jérémy Lal 09 Jan '12

09 Jan '12

Hi, it'd be glad to have patch_cirrus.c recognize model=imac27_122, (i'm adding a model because it's harder to auto-detect it), please find attached patch, tested on linux 3.2-rc7. I don't know anything about pin configs, just that it works with the ones in the patch. Note that i tried with model=auto (no sound) and model=imac27 (front speakers not muted when headphones are plugged in, and surround speakers not properly detected). Regards, Jérémy.

2 9

[alsa-devel] TWL4030 low-latency
by Edgar Berdahl 09 Jan '12

09 Jan '12

Dear Alsa Devel List, I would like to send some feedback on the TWL4030 driver when used on the Beagle Board. We have tested it under Ubuntu and Angstrom, and we find it does not work well at low latencies (e.g. input-to-output in 5ms, 10ms, or 20ms). Perhaps it was never tested extensively at these latencies. I thought I would send you this information in case you would look into it, or at least pass it on to the relevant contact. I can also run further tests if someone informs me what I should try. 1) To get the drivers to run at particularly low latencies (e.g. 20ms), it is necessary to operate the drivers in NON-realtime mode. It would interest me to know whether this means that the realtime functionality of the Beagle Board xM hardware or OS is hampered, or whether the TWL4030 driver is not operating up to specification. (If instead the TWL4030 driver is operating in realtime mode, then Jack stops running when an unrelated low-latency USB-serial device is opened.) 2) Some audio software (e.g. Pure Data, ChucK, etc.) will not connect to the TWL4030 ALSA drivers directly, instead they will only start when jack intermediates between the audio software and the TWL4030 ALSA drivers. 3) In some situations, the TWL4030 drivers appear not to start using the requested settings. For example, when we start jack requesting two periods per buffer, it is apparently getting initialized at eight periods per buffer, see the final lines below: Calling $ jackd -r -dalsa -dhw:0 -p128 -n2 -i2 -o2 -s -S -r on the Beagle under Ubuntu 10.10 and Angstrom demo image results in the following jackd 0.118.0 Copyright 2001-2009 Paul Davis, Stephane Letz, Jack O'Quinn, Torben Hohn and ot. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support. loading driver .. apparent rate = 44100 creating alsa driver ... hw:0|hw:0|128|2|44100|2|2|nomon|swmeter|soft-mode|16bit control device hw:0 configuring for 44100Hz, period = 128 frames (2.9 ms), buffer = 2 periods ALSA: final selected sample format for capture: 16bit little-endian ALSA: use **8** periods for capture ALSA: final selected sample format for playback: 16bit little-endian ALSA: use **8** periods for playback Thanks very much for listening! Best, Edgar Berdahl Lecturer Center for Computer Research in Music and Acoustics (CCRMA) Stanford University http://ccrma.stanford.edu/courses/250a

6 8

[alsa-devel] [PATCH v2 4/4] ASoC: OMAP: ams-delta: drop .set_bias_level callback
by Janusz Krzysztofik 09 Jan '12

09 Jan '12

This functionality has just been implemented in the cx20442 codec driver, no need to keep it here duplicated. Once done, remove the no longer used AMS_DELTA_LATCH2_MODEM_NRESET symbol from the board header file and a call to the regulator_toggle() helper function from the old API wrapper found in the board file. While being at it, simplify the way the modem .pm callback handles the regulator, and drop that no longer used helper function and its related consumer setup elements completely. Depends on patch 3/4 "ARM: OMAP1: ams-delta: update the modem to use regulator API" Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Changes against initial version: * don't move consumer setup elements, now named to indicated their modem related purpose, down the file, * don't track the regulator enavble/disable state, compare new target power state with the old one instead; thanks to Mark Brown for suggesting this simplification, * actually drop all references to AMS_DELTA_LATCH2_MODEM_NRESET. arch/arm/mach-omap1/board-ams-delta.c | 42 +++----------------- arch/arm/plat-omap/include/plat/board-ams-delta.h | 1 - sound/soc/omap/ams-delta.c | 32 ---------------- 3 files changed, 7 insertions(+), 68 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index 3389237..704fee6 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -18,7 +18,6 @@ #include <linux/input.h> #include <linux/interrupt.h> #include <linux/leds.h> -#include <linux/mutex.h> #include <linux/platform_device.h> #include <linux/regulator/consumer.h> #include <linux/regulator/fixed.h> @@ -291,35 +290,8 @@ static struct platform_device modem_nreset_device = { struct modem_private_data { struct regulator *regulator; - struct { - struct mutex lock; - bool enabled; - } consumer; }; -static int regulator_toggle(struct modem_private_data *priv, bool enable) -{ - int err = 0; - - mutex_lock(&priv->consumer.lock); - if (IS_ERR(priv->regulator)) { - err = PTR_ERR(priv->regulator); - } else if (enable) { - if (!priv->consumer.enabled) { - err = regulator_enable(priv->regulator); - priv->consumer.enabled = true; - } - } else { - if (priv->consumer.enabled) { - err = regulator_disable(priv->regulator); - priv->consumer.enabled = false; - } - } - mutex_unlock(&priv->consumer.lock); - - return err; -} - static struct modem_private_data modem_priv; void ams_delta_latch_write(int base, int ngpio, u16 mask, u16 value) @@ -330,8 +302,6 @@ void ams_delta_latch_write(int base, int ngpio, u16 mask, u16 value) for (; bit < ngpio; bit++, bitpos = bitpos << 1) { if (!(mask & bitpos)) continue; - else if (base + bit == AMS_DELTA_GPIO_PIN_MODEM_NRESET) - regulator_toggle(&modem_priv, (value & bitpos) != 0); else gpio_set_value(base + bit, (value & bitpos) != 0); } @@ -530,10 +500,16 @@ static void modem_pm(struct uart_port *port, unsigned int state, unsigned old) { struct modem_private_data *priv = port->private_data; + if (IS_ERR(priv->regulator)) + return; + if (state == old) return; - regulator_toggle(priv, state == 0); + if (state == 0) + regulator_enable(priv->regulator); + else if (old == 0) + regulator_disable(priv->regulator); } static struct plat_serial8250_port ams_delta_modem_ports[] = { @@ -593,7 +569,6 @@ static int __init late_init(void) gpio_direction_input(AMS_DELTA_GPIO_PIN_MODEM_IRQ); /* Initialize the modem_nreset regulator consumer before use */ - mutex_init(&modem_priv.consumer.lock); modem_priv.regulator = ERR_PTR(-ENODEV); ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, @@ -606,9 +581,6 @@ static int __init late_init(void) /* * Once the modem device is registered, the modem_nreset * regulator can be requested on behalf of that device. - * In addition to the modem .pm callback, that regulator - * is still used via the ams_delta_latch_write() wrapper - * by the ASoC driver until updated. */ modem_priv.regulator = regulator_get(&ams_delta_modem_device.dev, "RESET#"); diff --git a/arch/arm/plat-omap/include/plat/board-ams-delta.h b/arch/arm/plat-omap/include/plat/board-ams-delta.h index 027e79e..ad6f865 100644 --- a/arch/arm/plat-omap/include/plat/board-ams-delta.h +++ b/arch/arm/plat-omap/include/plat/board-ams-delta.h @@ -30,7 +30,6 @@ #define AMD_DELTA_LATCH2_SCARD_RSTIN 0x0400 #define AMD_DELTA_LATCH2_SCARD_CMDVCC 0x0800 -#define AMS_DELTA_LATCH2_MODEM_NRESET 0x1000 #define AMS_DELTA_LATCH2_MODEM_CODEC 0x2000 #define AMS_DELTA_GPIO_PIN_KEYBRD_DATA 0 diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 0462c7e..d213d04 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -426,29 +426,6 @@ static struct snd_soc_ops ams_delta_ops = { }; -/* Board specific codec bias level control */ -static int ams_delta_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - switch (level) { - case SND_SOC_BIAS_ON: - case SND_SOC_BIAS_PREPARE: - case SND_SOC_BIAS_STANDBY: - if (card->dapm.bias_level == SND_SOC_BIAS_OFF) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, - AMS_DELTA_LATCH2_MODEM_NRESET); - break; - case SND_SOC_BIAS_OFF: - if (card->dapm.bias_level != SND_SOC_BIAS_OFF) - ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, - 0); - } - card->dapm.bias_level = level; - - return 0; -} - /* Digital mute implemented using modem/CPU multiplexer. * Shares hardware with codec config pulse generation */ static bool ams_delta_muted = 1; @@ -512,9 +489,6 @@ static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd) ams_delta_ops.shutdown = ams_delta_shutdown; } - /* Set codec bias level */ - ams_delta_set_bias_level(card, dapm, SND_SOC_BIAS_STANDBY); - /* Add hook switch - can be used to control the codec from userspace * even if line discipline fails */ ret = snd_soc_jack_new(rtd->codec, "hook_switch", @@ -597,7 +571,6 @@ static struct snd_soc_card ams_delta_audio_card = { .name = "AMS_DELTA", .dai_link = &ams_delta_dai_link, .num_links = 1, - .set_bias_level = ams_delta_set_bias_level, }; /* Module init/exit */ @@ -646,11 +619,6 @@ static void __exit ams_delta_module_exit(void) ARRAY_SIZE(ams_delta_hook_switch_gpios), ams_delta_hook_switch_gpios); - /* Keep modem power on */ - ams_delta_set_bias_level(&ams_delta_audio_card, - &ams_delta_audio_card.rtd[0].codec->dapm, - SND_SOC_BIAS_STANDBY); - platform_device_unregister(cx20442_platform_device); platform_device_unregister(ams_delta_audio_platform_device); } -- 1.7.3.4

3 2

[alsa-devel] [PATCH - ice1724 1/4] ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
by pavel.hofman＠ivitera.com 09 Jan '12

09 Jan '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Add the capture pcm only if there is at least one ADC configured in the SYSCONF register. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/pci/ice1712/ice1724.c b/pci/ice1712/ice1724.c index 4353e76..2a0be2c 100644 --- a/pci/ice1712/ice1724.c +++ b/pci/ice1712/ice1724.c @@ -1117,14 +1117,18 @@ static struct snd_pcm_ops snd_vt1724_capture_pro_ops = { static int __devinit snd_vt1724_pcm_profi(struct snd_ice1712 *ice, int device) { struct snd_pcm *pcm; - int err; + int capt, err; - err = snd_pcm_new(ice->card, "ICE1724", device, 1, 1, &pcm); + capt = (~ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_ADC_MASK) ? + 1 : 0; + err = snd_pcm_new(ice->card, "ICE1724", device, 1, capt, &pcm); if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_vt1724_playback_pro_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_vt1724_capture_pro_ops); + if (capt) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_vt1724_capture_pro_ops); pcm->private_data = ice; pcm->info_flags = 0; -- 1.7.0.4

4 15

[alsa-devel] [PATCH] Revert "ASoC: Fix build dependency for SND_ATMEL_SOC_SSC"
by Axel Lin 09 Jan '12

09 Jan '12

This reverts commit 739be96ab83755e10fd0c2b6a34c8a73254527f7 because it introduces below build warnings: drivers/misc/Kconfig:212:error: recursive dependency detected! drivers/misc/Kconfig:212: symbol ATMEL_SSC is selected by SND_ATMEL_SOC_SSC sound/soc/atmel/Kconfig:9: symbol SND_ATMEL_SOC_SSC is selected by SND_AT91_SOC_SAM9G20_WM8731 sound/soc/atmel/Kconfig:18: symbol SND_AT91_SOC_SAM9G20_WM8731 depends on ATMEL_SSC Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/atmel/Kconfig | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index a4d6742..d1fcc81 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -9,7 +9,6 @@ config SND_ATMEL_SOC config SND_ATMEL_SOC_SSC tristate depends on SND_ATMEL_SOC - select ATMEL_SSC help Say Y or M if you want to add support for codecs the ATMEL SSC interface. You will also needs to select the individual -- 1.7.5.4

2 4

[alsa-devel] [PATCH] ALSA au88x0: Fix IRQ fifo error and channels swap of 4 channels playback
by Raymond Yau 09 Jan '12

09 Jan '12

[PATCH ] ALSA au88x0 - Fix IRQ fifo error and channels swap of 4 channels playback - Fix IRQ fifo error when playing stereo by set stereo flag of fifo control This also fix the swap of front and rear channels on au8830 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1840 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1726

3 5

[alsa-devel] [PATCH] ASoC: introduce SOC_SINGLE_S8_TLV macro
by Patrick Lai 09 Jan '12

09 Jan '12

ASoC currently has macro for double 8-bit signed TLV. Enhance info,get,put callback functions of SOC_DOUBLE_S8_TLV for single single 8-bit signed TLV. Signed-off-by: Patrick Lai <plai(a)codeaurora.org> --- include/sound/soc.h | 10 ++++++++++ sound/soc/soc-core.c | 28 ++++++++++++++++++++-------- 2 files changed, 30 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 11cfb59..558e9ff 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -55,6 +55,16 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \ + .put = snd_soc_put_volsw_s8, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .min = xmin, .max = xmax, \ + .platform_max = xmax} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5d3685..3d1ba5f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2399,13 +2399,15 @@ int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; int platform_max; int min = mc->min; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; if (!mc->platform_max) mc->platform_max = mc->max; platform_max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; + uinfo->count = shift == rshift ? 1 : 2; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max - min; return 0; @@ -2428,13 +2430,16 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; int min = mc->min; int val = snd_soc_read(codec, reg); ucontrol->value.integer.value[0] = - ((signed char)(val & 0xff))-min; - ucontrol->value.integer.value[1] = - ((signed char)((val >> 8) & 0xff))-min; + ((signed char)((val >> shift) & 0xff))-min; + if (shift != rshift) + ucontrol->value.integer.value[1] = + ((signed char)((val >> rshift) & 0xff))-min; return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_s8); @@ -2455,13 +2460,20 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; int min = mc->min; - unsigned int val; + unsigned int val, val2, val_mask; - val = (ucontrol->value.integer.value[0]+min) & 0xff; - val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; + val = ((ucontrol->value.integer.value[0]+min) & 0xff) << shift; + val_mask = 0xff << shift; + if (shift != rshift) { + val2 = (ucontrol->value.integer.value[1]+min) & 0xff; + val |= val2 << rshift; + val_mask |= 0xff << rshift; + } - return snd_soc_update_bits_locked(codec, reg, 0xffff, val); + return snd_soc_update_bits_locked(codec, reg, val_mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); -- 1.7.3.3 -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum. --

3 5

[alsa-devel] [PATCH] ASoC: Fix idma build after update for channel count check
by Mark Brown 09 Jan '12

09 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/idma.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c index 3ba6aba..c227c31 100644 --- a/sound/soc/samsung/idma.c +++ b/sound/soc/samsung/idma.c @@ -398,6 +398,7 @@ static int idma_new(struct snd_soc_pcm_runtime *rtd) if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = preallocate_idma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); + } return ret; } -- 1.7.7.3

3 2

Re: [alsa-devel] Not able to restart audio after hibernation
by Mark Brown 09 Jan '12

09 Jan '12

On Mon, Jan 09, 2012 at 08:46:45AM +0530, Shiraz Hashim wrote: > On Sat, Jan 07, 2012 at 02:32:05AM +0800, Mark Brown wrote: > > This suggests that you have a bug in your suspend and resume > > implementation, possibly outside of this driver. > The driver works absolutely fine in case of suspend to ram where the > power domains (which includes the I2S controller) is switched off. That doesn't mean everything is OK, for example the clock tree or DMA controller might not be preserving state properly. > As the soc-core.c misses hibernation call backs of dev_pm_ops, we were > wondering whether it has been tested before on some other platform. Not to my knowledge, but I don't see any reason why anything more than the trivial implementation should really be needed except for the general pointlessness of resuming after hibernating.

1 0

[alsa-devel] cannot submit syncpipe, usb2 audio
by alexander 08 Jan '12

08 Jan '12

|I asked in alsa-users about the state of usb2 audio class compliance some time ago but I never got a definitive answer.. I've done some digging on my own and found someone had a similar issue with the same error about a year ago. This is what dmesg says when I connect the device in usb2 mode, the last row is the error when I try to use it. [ 2384.749399] usb 1-2: USB disconnect, device number 2 [ 2390.729825] usb 1-2: new high speed USB device number 3 using ehci_hcd [ 2390.863654] generic-usb 0003:0926:0202.0003: hiddev0,hidraw0: USB HID v1.01 Device [Sound Devices USBPre2] on usb-0000:00:02.1-2/input3 [ 2418.388135] cannot submit syncpipe for urb 0, error -90: internal error Here are the lsusb -v as usb1.1 http://pastebin.com/Jj6mXHxJ and here as usb2 http://pastebin.com/8uCBAkNb I've looked at them and the only thing that differs is the|bInterval, wich is 1 in usb1.1 and 4 in usb2

1 1

[alsa-devel] [PATCH] ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range()
by Xi Wang 08 Jan '12

08 Jan '12

A malicious USB device may feed in carefully crafted min/max/res values, so that the inner loop in parse_uac2_sample_rate_range() could run for a long time or even never terminate, e.g., given max = INT_MAX. Also nr_rates could be a large integer, which causes an integer overflow in the subsequent call to kmalloc() in parse_audio_format_rates_v2(). Thus, kmalloc() would allocate a smaller buffer than expected, leading to a memory corruption. To exploit the two vulnerabilities, an attacker needs physical access to the machine to plug in a malicious USB device. This patch makes two changes. 1) The type of "rate" is changed to unsigned int, so that the loop could stop once "rate" is larger than INT_MAX. 2) Limit nr_rates to avoid overflow. Signed-off-by: Xi Wang <xi.wang(a)gmail.com> --- sound/usb/format.c | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index 89421d1..a99de67 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -226,7 +226,7 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, int min = combine_quad(&data[2 + 12 * i]); int max = combine_quad(&data[6 + 12 * i]); int res = combine_quad(&data[10 + 12 * i]); - int rate; + unsigned int rate; if ((max < 0) || (min < 0) || (res < 0) || (max < min)) continue; @@ -253,6 +253,9 @@ static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets, fp->rates |= snd_pcm_rate_to_rate_bit(rate); nr_rates++; + /* avoid overflow */ + if (nr_rates == KMALLOC_MAX_SIZE / sizeof(int)) + break; /* avoid endless loop */ if (res == 0) -- 1.7.5.4

2 6

[alsa-devel] ASoC updates for 3.3
by Mark Brown 08 Jan '12

08 Jan '12

The following changes since commit 32696af13724aaf7651d1cf95bc1a7a8af97a5c8: ASoC: Convert corgi to table based DAPM and control init (2011-12-30 10:44:45 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 and are the last fixes for 3.3. Ashish Chavan (1): ASoC: da7210: Add support for line input and mic Axel Lin (28): ASoC: Convert cs42l51 to devm_kzalloc() ASoC: Convert max98088 to devm_kzalloc() ASoC: Convert max98095 to devm_kzalloc() ASoC: Convert max9850 to devm_kzalloc() ASoC: Convert rt5631 to devm_kzalloc() ASoC: Convert ssm2602 to devm_kzalloc() ASoC: Convert sta32x to devm_kzalloc() ASoC: Convert tlv320aic23 to devm_kzalloc() ASoC: Convert tlv320aic26 to devm_kzalloc() ASoC: Convert tlv320aic3x to devm_kzalloc() ASoC: Convert alc5623 to devm_kzalloc() ASoC: Convert cs4270 to devm_kzalloc() ASoC: Convert tlv320dac33 to devm_kzalloc() ASoC: Convert tpa6130a2 to devm_kzalloc() ASoC: Fix return value of ak4641_pcm_set_dai_fmt() ASoC: Fix return value of wm8962_gpio_direction_out() ASoC: Use dai_fmt in saarb machine driver ASoC: Use dai_fmt in spitz machine driver ASoC: Use dai_fmt in z2 machine driver ASoC: Convert saarb to table based DAPM init ASoC: Convert spitz to table based DAPM and control init ASoC: Convert tavorevb3 to table based DAPM init ASoC: Convert z2 to table based DAPM init ASoC: Fix return value of wm8903_gpio_direction_in() and wm8903_gpio_direction_out() ASoC: Use dai_fmt in tavorevb3 machine driver ASoC: Fix return value of wm8580_set_sysclk() ASoC: pxa: Convert corgi to use snd_soc_register_card() ASoC: sta32x: Optimize the array work to find rate_min and rate_max Julia Lawall (9): ASoC: davinci-vcif.c: use devm_ functions ASoC: davinci-mcasp.c: use devm_ functions ASoC: davinci-i2s.c: use devm_ functions ASoC: psc-i2s.c: use devm_ functions ASoC: psc-ac97.c: use devm_ functions ASoC: i2sc.c: use devm_ functions ASoC: dma.c: use devm_ functions ASoC: dbdma2.c: use devm_ functions ASoC: ac97c.c: use devm_ functions Sangsu Park (1): ASoC: soc-pcm: Allocate PCM operations dynamically to support multiple DAIs arch/arm/mach-pxa/corgi.c | 9 ++++ sound/soc/au1x/ac97c.c | 40 ++++++------------- sound/soc/au1x/dbdma2.c | 14 ++----- sound/soc/au1x/dma.c | 12 +----- sound/soc/au1x/i2sc.c | 45 ++++++--------------- sound/soc/au1x/psc-ac97.c | 41 +++++++------------- sound/soc/au1x/psc-i2s.c | 42 +++++++------------- sound/soc/codecs/ak4641.c | 7 +++- sound/soc/codecs/alc5623.c | 11 +---- sound/soc/codecs/cs4270.c | 6 +-- sound/soc/codecs/cs42l51.c | 8 +--- sound/soc/codecs/da7210.c | 77 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98088.c | 6 +-- sound/soc/codecs/max98095.c | 7 +-- sound/soc/codecs/max9850.c | 6 +-- sound/soc/codecs/rt5631.c | 7 +-- sound/soc/codecs/ssm2602.c | 12 ++---- sound/soc/codecs/sta32x.c | 14 ++---- sound/soc/codecs/tlv320aic23.c | 5 +-- sound/soc/codecs/tlv320aic26.c | 8 +--- sound/soc/codecs/tlv320aic3x.c | 5 +-- sound/soc/codecs/tlv320dac33.c | 6 +-- sound/soc/codecs/tpa6130a2.c | 5 +-- sound/soc/codecs/wm8580.c | 8 +++- sound/soc/codecs/wm8903.c | 18 +++++++-- sound/soc/codecs/wm8962.c | 10 +++- sound/soc/davinci/davinci-i2s.c | 39 ++++++------------ sound/soc/davinci/davinci-mcasp.c | 40 ++++++------------- sound/soc/davinci/davinci-vcif.c | 14 ++----- sound/soc/pxa/corgi.c | 43 +++++++++++---------- sound/soc/pxa/saarb.c | 23 ++++------- sound/soc/pxa/spitz.c | 37 +++++------------- sound/soc/pxa/tavorevb3.c | 24 ++++-------- sound/soc/pxa/z2.c | 28 ++++---------- sound/soc/soc-pcm.c | 38 +++++++++---------- 35 files changed, 313 insertions(+), 402 deletions(-)

2 1

[alsa-devel] [PATCH] - ALSA au88x0 - XTALK fixes
by Raymond Yau 08 Jan '12

08 Jan '12

[PATCH ALSA 1/2] Au88x0 - Xtalk - fix write/read of eq and xt instates - fix write/read of eq and xt instates [PATCH ALSA 2/2] Au88x0 - Fix Xtalk's constants] - Fix XtalkGainsDefault, XtalkGains1Chn - Fix XtalkWideCoefsLeftEQ, XtalkWideCoefsRightEQ - Fix XtlakWideCoefsLeftXT, XtalkWideCoefsRightXT

2 1

[alsa-devel] [PATCH - ice1724 1/1] ALSA: ice1724 - Check for ac97 to avoid kernel oops
by pavel.hofman＠ivitera.com 08 Jan '12

08 Jan '12

From: Pavel Hofman <pavel.hofman(a)ivitera.com> Cards with identical PCI ids but no AC97 config in EEPROM do not have the ac97 field initialized. We must check for this case to avoid kernel oops. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/pci/ice1712/amp.c b/pci/ice1712/amp.c index e328cfb..e525da2 100644 --- a/pci/ice1712/amp.c +++ b/pci/ice1712/amp.c @@ -68,8 +68,11 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice) static int __devinit snd_vt1724_amp_add_controls(struct snd_ice1712 *ice) { - /* we use pins 39 and 41 of the VT1616 for left and right read outputs */ - snd_ac97_write_cache(ice->ac97, 0x5a, snd_ac97_read(ice->ac97, 0x5a) & ~0x8000); + if (ice->ac97) + /* we use pins 39 and 41 of the VT1616 for left and right + read outputs */ + snd_ac97_write_cache(ice->ac97, 0x5a, + snd_ac97_read(ice->ac97, 0x5a) & ~0x8000); return 0; } -- 1.7.0.4

2 1

Re: [alsa-devel] ALSA regression in recent kernels: Intel HDA/ALC892 sound glitches
by Takashi Iwai 08 Jan '12

08 Jan '12

At Sun, 8 Jan 2012 09:49:27 +0000 (GMT), Artem S. Tashkinov wrote: > > > On Jan 8, 2012, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Thu, 5 Jan 2012 11:29:40 +0000 (GMT), > > Artem S. Tashkinov wrote: > > > > > > Hello, > > > > > > Since kernel 3.0.11 I have quite terrible sound artifacts - random pops, clicks and > > > short pauses (~500ms). Linux kernel 3.0.0 is *bug free*. So, some some of patches in > > > kernel *3.0.11* broke the driver, the bug is, of course, present in kernels 3.1 and 3.2. > > > > Could you give alsa-info.sh output to check more details? > > Also, any chance to identify which 3.0.x release breaks the things? > > alsa-info output was attached to my original mail posted into LKML, please check out > you e-mail client. Ah, OK, I'll check it later. I'm still on vacation so I can't check in details. > There are so few changes in Intel/Realtek HDA ALSA code between 3.0 and 3.0.11 > I thought you could identify the culprit just by looking at the list. Well, it's not too easy since the problem isn't deterministic. You can check the patches with ALSA: hda (or capital) and ALSA: pcm prefix. > ALSA: fm801: Fix double free in case of error in tuner detection > ALSA: fm801: Gracefully handle failure of tuner auto-detect > ALSA: HDA: Add new revision for ALC662 > ALSA: hda - Add position_fix quirk for Dell Inspiron 1010 > ALSA: hda - Add support for 92HD65 / 92HD66 family of codecs > ALSA: HDA: Add support for IDT 92HD93 > ALSA: HDA: Cirrus - fix "Surround Speaker" volume control name > ALSA: HDA: conexant support for Lenovo T520/W520 > ALSA: hda - Disable power-widget control for IDT 92HD83/93 as default > ALSA: hda - Don't add elements of other codecs to vmaster slave > ALSA: hda - Fix ADC input-amp handling for Cx20549 codec > ALSA: HDA: No power nids on 92HD93 > ALSA: hda/realtek - Avoid bogus HP-pin assignment > ALSA: hda/realtek - Fix auto-mute with HP+LO configuration > ALSA: hda/realtek - Skip invalid digital out pins > ALSA: hda - Remove bad code for IDT 92HD83 family patch > ALSA: pcm - fix race condition in wait_for_avail() > ALSA: ua101: fix crash when unplugging > ALSA: usb-audio: Check for possible chip NULL pointer before clearing probing flag > ALSA: usb-audio - Check the dB-range validity in the later read, too > ALSA: usb-audio - clear chip->probing on error exit > ALSA: usb-audio - Fix the missing volume quirks at delayed init > > If you have no ideas about which patch could have broken things, I will try to figure it out > by bisecting. That'll be helpful. thanks, Takashi

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Re: [alsa-devel] ALSA regression in recent kernels: Intel HDA/ALC892 sound glitches
by Takashi Iwai 08 Jan '12

08 Jan '12

At Thu, 5 Jan 2012 11:29:40 +0000 (GMT), Artem S. Tashkinov wrote: > > Hello, > > Since kernel 3.0.11 I have quite terrible sound artifacts - random pops, clicks and > short pauses (~500ms). Linux kernel 3.0.0 is *bug free*. So, some some of patches in > kernel *3.0.11* broke the driver, the bug is, of course, present in kernels 3.1 and 3.2. Could you give alsa-info.sh output to check more details? Also, any chance to identify which 3.0.x release breaks the things? thanks, Takashi

1 0

[alsa-devel] [PATCH] hda_intel: Add Oaktrail identifiers
by Alan Cox 08 Jan '12

08 Jan '12

From: Li Peng <pengi.li(a)intel.com> Oaktrail has 0x8086, 0x080a - AZX_DRIVER_SCH Taken from the Meego patches for Oaktrail Signed-off-by: Li Peng <peng.li(a)intel.com> Signed-off-by: Alan Cox <alan(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c2f79e6..b339eb5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2972,6 +2972,9 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */ + { PCI_DEVICE(0x8086, 0x080a), + .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | + AZX_DCAPS_BUFSIZE}, /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC |

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[alsa-devel] [PATCH] ALSA: HDA: Remove Poulsbo position fix quirks
by David Henningsson 08 Jan '12

08 Jan '12

Now that we have changed the poulsbo chip to use LPIB position fix, we can remove the individual machine quirks that do the same thing. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8e232ab..0852e20 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2493,12 +2493,10 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1028, 0x02c6, "Dell Inspiron 1010", POS_FIX_LPIB), SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB), - SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS 1101HA", POS_FIX_LPIB), SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB), SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB), SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), -- 1.7.5.4

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[alsa-devel] [PATCH 1/3] ALSA: HDA: Fix master control for Cirrus Logic 421X
by David Henningsson 08 Jan '12

08 Jan '12

The control name "HP/Speakers" is non-standard, and since there is only one DAC on this chip there is no need for a virtual master anyway. Cc: stable(a)kernel.org Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_cirrus.c | 13 +------------ 1 files changed, 1 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index acfb645..9139558 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -1779,30 +1779,19 @@ static int build_cs421x_output(struct hda_codec *codec) struct auto_pin_cfg *cfg = &spec->autocfg; struct snd_kcontrol *kctl; int err; - char *name = "HP/Speakers"; + char *name = "Master"; fix_volume_caps(codec, dac); - if (!spec->vmaster_sw) { - err = add_vmaster(codec, dac); - if (err < 0) - return err; - } err = add_mute(codec, name, 0, HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); if (err < 0) return err; - err = snd_ctl_add_slave(spec->vmaster_sw, kctl); - if (err < 0) - return err; err = add_volume(codec, name, 0, HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); if (err < 0) return err; - err = snd_ctl_add_slave(spec->vmaster_vol, kctl); - if (err < 0) - return err; if (cfg->speaker_outs) { err = snd_hda_ctl_add(codec, 0, -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Optimise performance of WM8904 ADC 128fs OSR mode
by Mark Brown 08 Jan '12

08 Jan '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8904.c | 31 ++++++++++++++++++++++++++++++- sound/soc/codecs/wm8904.h | 11 +++++++++++ 2 files changed, 41 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 31d2e41..e3bd0eb 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -304,6 +304,7 @@ static bool wm8904_readable_register(struct device *dev, unsigned int reg) case WM8904_EQ23: case WM8904_EQ24: case WM8904_CONTROL_INTERFACE_TEST_1: + case WM8904_ADC_TEST_0: case WM8904_ANALOGUE_OUTPUT_BIAS_0: case WM8904_FLL_NCO_TEST_0: case WM8904_FLL_NCO_TEST_1: @@ -569,6 +570,29 @@ static const char *hpf_mode_text[] = { static const struct soc_enum hpf_mode = SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text); +static int wm8904_adc_osr_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int val; + int ret; + + ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + + if (ucontrol->value.integer.value[0]) + val = WM8904_ADC_128_OSR_TST_MODE | WM8904_ADC_BIASX1P5; + else + val = 0; + + snd_soc_update_bits(codec, WM8904_ADC_TEST_0, + WM8904_ADC_128_OSR_TST_MODE | WM8904_ADC_BIASX1P5, + val); + + return ret; +} + static const struct snd_kcontrol_new wm8904_adc_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8904_ADC_DIGITAL_VOLUME_LEFT, WM8904_ADC_DIGITAL_VOLUME_RIGHT, 1, 119, 0, digital_tlv), @@ -585,7 +609,12 @@ SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0, SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0), SOC_ENUM("High Pass Filter Mode", hpf_mode), -SOC_SINGLE("ADC 128x OSR Switch", WM8904_ANALOGUE_ADC_0, 0, 1, 0), +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC 128x OSR Switch", + .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, + .put = wm8904_adc_osr_put, + .private_value = SOC_SINGLE_VALUE(WM8904_ANALOGUE_ADC_0, 0, 1, 0), +}, }; static const char *drc_path_text[] = { diff --git a/sound/soc/codecs/wm8904.h b/sound/soc/codecs/wm8904.h index 9e8c841..c29a0e8 100644 --- a/sound/soc/codecs/wm8904.h +++ b/sound/soc/codecs/wm8904.h @@ -123,6 +123,7 @@ #define WM8904_EQ23 0x9C #define WM8904_EQ24 0x9D #define WM8904_CONTROL_INTERFACE_TEST_1 0xA1 +#define WM8904_ADC_TEST_0 0xC6 #define WM8904_ANALOGUE_OUTPUT_BIAS_0 0xCC #define WM8904_FLL_NCO_TEST_0 0xF7 #define WM8904_FLL_NCO_TEST_1 0xF8 @@ -1557,6 +1558,16 @@ #define WM8904_USER_KEY_WIDTH 1 /* USER_KEY */ /* + * R198 (0xC6) - ADC Test 0 + */ +#define WM8904_ADC_128_OSR_TST_MODE 0x0004 /* ADC_128_OSR_TST_MODE */ +#define WM8904_ADC_128_OSR_TST_MODE_SHIFT 2 /* ADC_128_OSR_TST_MODE */ +#define WM8904_ADC_128_OSR_TST_MODE_WIDTH 1 /* ADC_128_OSR_TST_MODE */ +#define WM8904_ADC_BIASX1P5 0x0001 /* ADC_BIASX1P5 */ +#define WM8904_ADC_BIASX1P5_SHIFT 0 /* ADC_BIASX1P5 */ +#define WM8904_ADC_BIASX1P5_WIDTH 1 /* ADC_BIASX1P5 */ + +/* * R204 (0xCC) - Analogue Output Bias 0 */ #define WM8904_PGA_BIAS_MASK 0x0070 /* PGA_BIAS - [6:4] */ -- 1.7.7.3

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[alsa-devel] TI Davinci expert wanted McASP/ALSA
by David Smith 08 Jan '12

08 Jan '12

Hi, I am beginning to start on an audio project for the TI DM8148/EVM. I intend to 1). extend the current McASP/PCM drivers to support 4 transmit serializes to support four stereo pairs of 24-bit 96kHz in DIT mode - output on McASP2 (for 7.1 audio digital) 2). extend the current McASP/PCM drivers to support 4 transmit serializes to support four stereo pairs of 24-bit 96kHz in i2s mode - output on McASP2 (for 7.1 audio digital) 3). make relevant changes to EDMA etc. 4). make sure McASP2/3 hardware registers are set up correctly 5). add relevant support to a new ALSA driver and present McASP2 and McASP3 as two separate 7.1 capable sound cards. I am looking for some help (paid or otherwise. Anyone interested? David.

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Re: [alsa-devel] [PATCH] asoc: replace 0xffffffff with DMA_BIT_MASK macro
by Mark Brown 07 Jan '12

07 Jan '12

On Sun, Jan 01, 2012 at 02:43:03AM +0100, manabian(a)gmail.com wrote: > From: Joachim Eastwood <joachim.eastwood(a)jotron.com> Applied, thanks.

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Re: [alsa-devel] [PATCH] asoc: use proper defines for stream directions in pcm engines
by Mark Brown 07 Jan '12

07 Jan '12

On Sun, Jan 01, 2012 at 02:14:24AM +0100, manabian(a)gmail.com wrote: > From: Joachim Eastwood <joachim.eastwood(a)jotron.com> Applied, thanks.

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Re: [alsa-devel] [PATCH] asoc: check for substream not channels_min in pcm engines
by Mark Brown 07 Jan '12

07 Jan '12

On Sun, Jan 01, 2012 at 01:58:44AM +0100, manabian(a)gmail.com wrote: > From: Joachim Eastwood <joachim.eastwood(a)jotron.com> > > This is a follow up on 53dea36c70c1857 which fixes the other affected > pcm engines. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] asoc: replace 0xffffffff with DMA_BIT_MASK macro
by Mark Brown 07 Jan '12

07 Jan '12

On Sun, Jan 01, 2012 at 02:43:03AM +0100, manabian(a)gmail.com wrote: > From: Joachim Eastwood <joachim.eastwood(a)jotron.com> > > Signed-off-by: Joachim Eastwood <joachim.eastwood(a)jotron.com> I'm assuming that you are actually Joachim Eastwood and there's some git misconfiguration going on here? If you're not Jochim you need to sign off the patch yourself as well. Please also use subject lines corresponding to the subsystem.

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Re: [alsa-devel] [PATCH v8 2/2] davinci: CQ93VC: remove machine specific header file inclusion from codec driver
by Mark Brown 07 Jan '12

07 Jan '12

On Wed, Jan 04, 2012 at 12:47:49PM +0530, Manjunath Hadli wrote: > remove unnecessary inclusion of machine specific header > file mach/dm365.h from cq93vc.c voice codec driver > which comes in the way of platform code consolidation. *Always* CC maintainers on patches. If you send me a copy directly this looks OK.

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[alsa-devel] Not able to restart audio after hibernation
by Rajeev kumar 07 Jan '12

07 Jan '12

Hello Mark, I am testing my audio driver on spear platform with the linux kernel 2.6.37. During testing I put my system in hibernation mode and found that there is no sound after resume. The steps I am using for hibernation are 1. Have audio player on, select some audio, (using aplay utils) 2. Hibernate machine 3. Resume from hibernate After resume it seems that there is no data available in the buffer to play, as application is giving pcm_write error and exit. While going through the suspend/resume implementation in the alsa framework, get to know that feature to support hibernation is missing. The suspend/resume implementation for audio is with dev_pm_ops which is present in snd-core.c . If we look into the structure, we have only suspend, resume and poweroff feature present in it. To support hibernation we need freeze,thaw and restore also. Please correct me if I am wrong? Just for experiment I modified the structure and added freeze, thaw and restore functionality, but the result is same. The new modified structure looks like static const struct dev_pm_ops soc_pm_ops = { .suspend = soc_suspend, .resume = soc_resume, .freeze = soc_suspend, .thaw = soc_resume, .restore = soc_resume, .poweroff = soc_poweroff, } Just for info audio is working fine in case of suspend-to-ram. Best Regards ~Rajeev

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[alsa-devel] [RFC][PATCH] ARM: NUC900: Rename nuc900-audio platform device to nuc900-ac97
by Axel Lin 06 Jan '12

06 Jan '12

This change ensures the platform device name matches nuc900-ac97 platform driver name. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi ZongShun, I don't have this hardware, I'd appreciate if you can test this patch. This patch is against linux-next 20120103. Thanks, Axel arch/arm/mach-w90x900/clksel.c | 2 +- arch/arm/mach-w90x900/cpu.c | 2 +- arch/arm/mach-w90x900/dev.c | 6 +++--- arch/arm/mach-w90x900/mfp.c | 2 +- sound/soc/nuc900/nuc900-ac97.c | 2 +- 5 files changed, 7 insertions(+), 7 deletions(-) diff --git a/arch/arm/mach-w90x900/clksel.c b/arch/arm/mach-w90x900/clksel.c index 3de4a52..06d867dc 100644 --- a/arch/arm/mach-w90x900/clksel.c +++ b/arch/arm/mach-w90x900/clksel.c @@ -48,7 +48,7 @@ static void clock_source_select(const char *dev_id, unsigned int clkval) offset = ATAOFFSET; else if (strcmp(dev_id, "nuc900-lcd") == 0) offset = LCDOFFSET; - else if (strcmp(dev_id, "nuc900-audio") == 0) + else if (strcmp(dev_id, "nuc900-ac97") == 0) offset = AUDOFFSET; else offset = CPUOFFSET; diff --git a/arch/arm/mach-w90x900/cpu.c b/arch/arm/mach-w90x900/cpu.c index 604e1db..9a06619 100644 --- a/arch/arm/mach-w90x900/cpu.c +++ b/arch/arm/mach-w90x900/cpu.c @@ -79,7 +79,7 @@ static DEFINE_CLK(timer4, 23); static struct clk_lookup nuc900_clkregs[] = { DEF_CLKLOOK(&clk_lcd, "nuc900-lcd", NULL), - DEF_CLKLOOK(&clk_audio, "nuc900-audio", NULL), + DEF_CLKLOOK(&clk_audio, "nuc900-ac97", NULL), DEF_CLKLOOK(&clk_fmi, "nuc900-fmi", NULL), DEF_CLKLOOK(&clk_ms, "nuc900-fmi", "MS"), DEF_CLKLOOK(&clk_sd, "nuc900-fmi", "SD"), diff --git a/arch/arm/mach-w90x900/dev.c b/arch/arm/mach-w90x900/dev.c index 5b0c38a..78110be 100644 --- a/arch/arm/mach-w90x900/dev.c +++ b/arch/arm/mach-w90x900/dev.c @@ -501,8 +501,8 @@ static struct resource nuc900_ac97_resource[] = { }; -struct platform_device nuc900_device_audio = { - .name = "nuc900-audio", +struct platform_device nuc900_device_ac97 = { + .name = "nuc900-ac97", .id = -1, .num_resources = ARRAY_SIZE(nuc900_ac97_resource), .resource = nuc900_ac97_resource, @@ -523,7 +523,7 @@ static struct platform_device *nuc900_public_dev[] __initdata = { &nuc900_device_emc, &nuc900_device_spi, &nuc900_device_wdt, - &nuc900_device_audio, + &nuc900_device_ac97, }; /* Provide adding specific CPU platform devices API */ diff --git a/arch/arm/mach-w90x900/mfp.c b/arch/arm/mach-w90x900/mfp.c index 9dd7461..c58d142 100644 --- a/arch/arm/mach-w90x900/mfp.c +++ b/arch/arm/mach-w90x900/mfp.c @@ -155,7 +155,7 @@ void mfp_set_groupg(struct device *dev, const char *subname) } else if (strcmp(dev_id, "nuc900-i2c1") == 0) { mfpen &= ~(GPIOG2TO3); mfpen |= ENI2C1;/*enable i2c1*/ - } else if (strcmp(dev_id, "nuc900-audio") == 0) { + } else if (strcmp(dev_id, "nuc900-ac97") == 0) { mfpen &= ~(GPIOG22TO23); mfpen |= ENAC97;/*enable AC97*/ } else if (strcmp(dev_id, "nuc900-mmc-port1") == 0) { diff --git a/sound/soc/nuc900/nuc900-ac97.c b/sound/soc/nuc900/nuc900-ac97.c index 45d11dd..946020a 100644 --- a/sound/soc/nuc900/nuc900-ac97.c +++ b/sound/soc/nuc900/nuc900-ac97.c @@ -366,7 +366,7 @@ static int __devinit nuc900_ac97_drvprobe(struct platform_device *pdev) goto out3; /* enbale ac97 multifunction pin */ - mfp_set_groupg(nuc900_audio->dev, "nuc900-audio"); + mfp_set_groupg(nuc900_audio->dev, NULL); return 0; -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Fix build dependency for SND_ATMEL_SOC_SSC
by Axel Lin 06 Jan '12

06 Jan '12

Make SND_ATMEL_SOC_SSC select ATMEL_SSC to fix below build errors: LD .tmp_vmlinux1 sound/built-in.o: In function `atmel_ssc_remove': sound/soc/atmel/atmel_ssc_dai.c:713: undefined reference to `ssc_free' sound/built-in.o: In function `atmel_ssc_probe': sound/soc/atmel/atmel_ssc_dai.c:700: undefined reference to `ssc_request' sound/built-in.o: In function `atmel_ssc_set_audio': sound/soc/atmel/atmel_ssc_dai.c:845: undefined reference to `ssc_request' sound/soc/atmel/atmel_ssc_dai.c:851: undefined reference to `ssc_free' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch is against linux-next 20120103. sound/soc/atmel/Kconfig | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index d1fcc81..a4d6742 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -9,6 +9,7 @@ config SND_ATMEL_SOC config SND_ATMEL_SOC_SSC tristate depends on SND_ATMEL_SOC + select ATMEL_SSC help Say Y or M if you want to add support for codecs the ATMEL SSC interface. You will also needs to select the individual -- 1.7.5.4

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[alsa-devel] eeprom params for ice1724 module - recommendation?
by Pavel Hofman 05 Jan '12

05 Jan '12

Hi, I have a generic ice1724 card ooAoo SQ210a (e.g. http://forum.lowyat.net/topic/851912/all ). The card has identical PCI identification with Chaintec AV710, no DAC controls (the DAC is hardwired in pin-control mode with no GPIOs attached), no identification string in eeprom and of course incorrect eeprom values. Since the PCI ids overlap with the Chaintec, I cannot define correct eeprom_data for this specific card in the driver. Furthermore I would not be surprised if the other ooAoo cards with different setups shared the same PCI ids. I can skip the Chaintec config by specifying an unknown model in snd-ice1724 param, e.g. modprobe snd-ice1724 model=aooa I would like to be able to specify correct eeprom values for the model too. That would be simple by adding several new module params - eeep2-sysconf, eeep2-aclink, eeep2-spdif etc. However, I have another ice1724 card properly supported by the driver (Infrasonic Quartet). I can use modprobe snd-ice1724 model=quartet,aooa and everything works correctly. But if I wanted to define eeep2-xxx params for the aooa, I would have to define the same params for the quartet too which I find superfluous since they are already correctly in the driver. Or I could use a string type instead of int and use something like eeep2-sysconf=default,47 Or I could "encode" the eeep2 params into the model name, such as modprobe snd-ice1724 model="quartet,aooa(eeep2-sysconf=47;eeep2-spdif=20)" Please which version would you recommend (i.e. accept as a patch)? Thank you for your expert advice. Best regards, Pavel.

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[alsa-devel] [PATCH] ASoC: Fix mic bias handling in TI tlv320aic32x codec.
by Javier Martin 05 Jan '12

05 Jan '12

tlv320aic32x doesn't have a dedicated MIC input, it has a MICBIAS pin instead, which is used to provide power to an external microphone. This means one can enable this MICBIAS pin for any input configuration of INX_L and INX_R. For this reason it's not possible to add this as a DAPM event since we lack the possibility to associate it to a DAPM route. So we use platform data totell the device whether we are connecting a microphone or not. --- sound/soc/codecs/tlv320aic32x4.c | 6 +++--- sound/soc/codecs/tlv320aic32x4.h | 1 + 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e93b9d1..f260e60 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -208,7 +208,6 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { ARRAY_SIZE(right_input_mixer_controls)), SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), - SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), @@ -659,8 +658,9 @@ static int aic32x4_probe(struct snd_soc_codec *codec) /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { - snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN | - AIC32X4_MICBIAS_2075V); + snd_soc_write(codec, AIC32X4_MICBIAS, + AIC32X4_MICBIAS_ON | AIC32X4_MICBIAS_LDOIN | + AIC32X4_MICBIAS_2075V); } if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) { snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE); diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index aae2b24..7b9ed7e 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -113,6 +113,7 @@ #define AIC32X4_PLLCLKIN 0x03 +#define AIC32X4_MICBIAS_ON 0x40 #define AIC32X4_MICBIAS_LDOIN 0x08 #define AIC32X4_MICBIAS_2075V 0x60 -- 1.7.0.4

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[alsa-devel] tlv32aic3x, rx-51, regulator
by Peter Meerwald 05 Jan '12

05 Jan '12

Hello, the tlv320aic3x codec driver requests a number of supplies/regulators (IOVDD, DVDD, AVDD, DRVDD), these seem mandatory for the codec driver to operate and somehow tied to the rx51 platform I want to use the tlv320aic3x codec driver with some other board where I don't care about supplies/regulators -- how is this going to work? do I need to have dummy supplies just to keep tlv320aic3x happy? shouldn't the ai3x_supply_names be something that is optional and passed in via platform_data? regards, p. -- Peter Meerwald +43-664-2444418 (mobile)

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[alsa-devel] [alsa-lib PATCH 2/4] Search-and-replace improve documentation language
by Andrew Eikum 05 Jan '12

05 Jan '12

--- src/pcm/pcm.c | 146 ++++++++++++++++++++++++++------------------------------- 1 files changed, 67 insertions(+), 79 deletions(-)

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Re: [alsa-devel] [PATCH] ASoC: Add support for TI LM49453 Audio codec
by Mark Brown 05 Jan '12

05 Jan '12

On Wed, Jan 04, 2012 at 03:11:03AM -0800, Reddy, MR Swami wrote: > On Wednesday, January 04, 2012 2:48 AM, Mark Brown wrote: Your mail client appears to be very broken, it's reflowed all my text to remove line breaks which does nothing for legibility. > >> + SND_SOC_DAPM_OUT_DRV("Sidetone DAC HPR Playback", > >> + LM49453_P0_STN_SEL_REG, 1, 0, NULL, 0), > >What exactly do these register bits do? The use of OUT_DRV widgets looks very suspicious for a sidetone, an output driver would usually be a power > >amplifier or transducer driver so I suspect the power sequencing will be all wrong. It looks like these are mixer input controls which wouldn't > >normally be widgets at all. > The above selects the Sidetone onto left and right HP DAC output. Sidetone is not a mixer control. What makes you say that the sidetone is not a mixer? That's generally exactly what a sidetone is - mixing an input into an output, usually with a lot of attenuation. What you're describing above just sounds like a normal sidetone, why have you done these as output driver widgets? > >> + { "AMIC1Bias", NULL, "AMIC1"}, > >> + { "Mic1_Input Capture Route", "MIC1", "AMIC1Bias"}, > >Your CODEC driver should not be connecting the microphone biases, the connections are board specific. > The above Micbias is internal to codec. Please advice. You're saying that the CODEC has a MEMS microphone built into it? > >That memset is just going to mask warnings, it shouldn't be needed. > OK. Generally it's OK to just make updates in new versions without explicitly commenting on each one, this makes it much easier to see areas of your reply that are signifigant. > >> + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, val); > >> + snd_soc_write(codec, LM49453_P0_DAC_OT_CLK_DIV_REG, val); > >> + snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, val); > >...and frankly I've no idea what this code is doing, I'd expect a clock divider configuration to vary depending on the >input clock as well as the sample rate. > The 'val' is the clock divider value based on the sampling rate. If these are clock dividers shouldn't they also depend on the input clock? > >> + case 19000000: > >> + val = 0; > >> + break; > >> + case 48000: > >> + case 32576: > >> + val = 1; > >> + break; > >> + default: > >> + return -EINVAL; > >> + } > >> + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), > >> + val); > >I don't think this does what you think it does, your mask and value don't overlap at all. > In the above _update_bits(), the LM49453_P0_PMC_SETUP_REG register's 4th bit is set, based on the 'val'. Please let me know, if something wrong there. Please read what I wrote above. > >> + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(0), > >> + (!!mute << 0)); > >> + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1), > >> + (!!mute << 1)); > >Why are there two separate update bits operations here? > For muting the HP left and right channels using BIT(0) and BIT(1). That doesn't answer my question, my question was why there are two *separate* update_bits() operations.

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[alsa-devel] Does ASoC support sound card with multiple interfaces?
by Ashish Chavan 05 Jan '12

05 Jan '12

Hi, I have a codec with multiple TX and RX channels. In ASoC tree, I have seen examples where these kind of codecs are implemented as multichannel codecs. Instead I want it to be treated as card with multiple interfaces, e.g. Instead of one interface with eight TX channels, I need it to behave it as card with four stereo playback interfaces. The primary aim for doing this is that multiple applications should be able to play/record simultaneously. AFAIK, In case of a multichannel card, if it is opened by one application for stereo playback, there is no way to use rest of the unused channels by other applications. Is it easily possible to achieve what I am trying to? Any pointers to example driver source will greatly help! BTW all three i.e., platform, machine as well as codec driver are to be implemented. I am aware of the fact that this requirement may affect all three. Thanks in advance, -- Ashish || GNU FAN ||

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[alsa-devel] [alsa-lib PATCH 4/4] Improve snd_device_name_hint documentation
by Andrew Eikum 04 Jan '12

04 Jan '12

--- src/control/namehint.c | 26 ++++++++++++++++---------- 1 files changed, 16 insertions(+), 10 deletions(-)

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[alsa-devel] [alsa-lib PATCH 3/4] Improve hw_params documentation
by Andrew Eikum 04 Jan '12

04 Jan '12

--- include/pcm.h | 17 ++++++++++++++++- src/pcm/pcm.c | 10 +++++++++- 2 files changed, 25 insertions(+), 2 deletions(-)

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[alsa-devel] [alsa-lib PATCH 1/4] Remove docs for snd_names_list and snd_names_list_free
by Andrew Eikum 04 Jan '12

04 Jan '12

No need to have misleading docs for functions that do nothing. --- Resending the whole sequence. New patch 4/4 improves the documentation for the name hints functions. src/names.c | 13 ++----------- 1 files changed, 2 insertions(+), 11 deletions(-)

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[alsa-devel] FW: Your message to Alsa-devel awaits moderator approval
by Reddy, MR Swami 04 Jan '12

04 Jan '12

Hi, Iam getting the below email, when I post patch to alsa-devel mailing list. This happens from yesterday onwards (ie 3rd Jan 2012). Do I need to take any action from my side? Plz confirm. Thanks Swami -----Original Message----- From: alsa-devel-bounces(a)alsa-project.org [mailto:alsa-devel-bounces@alsa-project.org] Sent: Wednesday, January 04, 2012 4:41 PM To: Reddy, MR Swami Subject: Your message to Alsa-devel awaits moderator approval Your mail to 'Alsa-devel' with the subject RE: [PATCH] ASoC: Add support for TI LM49453 Audio codec Is being held until the list moderator can review it for approval. The reason it is being held: Post by non-member to a members-only list Either the message will get posted to the list, or you will receive notification of the moderator's decision. If you would like to cancel this posting, please visit the following URL: http://mailman.alsa-project.org/mailman/confirm/alsa-devel/ab4c2673e9de3c4e…

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[PATCH] ALSA: HDA: Use LPIB position fix for Oaktrail
by David Henningsson 04 Jan '12

04 Jan '12

According to the thread on alsa-devel, the LPIB method is to prefer for Oaktrail controller chip. Reference: http://mailman.alsa-project.org/pipermail/alsa-devel/2012-January/047800.ht… Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bd447b6..8e232ab 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2997,7 +2997,7 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Poulsbo */ { PCI_DEVICE(0x8086, 0x080a), .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | - AZX_DCAPS_BUFSIZE}, + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_LPIB }, /* Oaktrail */ /* ICH */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH | AZX_DCAPS_OLD_SSYNC | -- 1.7.5.4 --------------050407000008040100030201--

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[alsa-devel] Join play of pcm in multiple playback devices
by Irfan shaikh 03 Jan '12

03 Jan '12

Hello All, I have written an application which supports join play of PCM Audio. i.e Capture and write to playback device 1 and playback device 2 simultaneously. Below code is just to explain the above scenario : /######################/ while(1) { retval=snd_pcm_readi(pAlsaConfig->pcm_handle, buffer, pAlsaConfig->alsaBufferSize); snd_pcm_writei(pAlsaConfig->pcm_handle1, buffer,retval); snd_pcm_writei(pAlsaConfig->pcm_handle2, buffer,retval); } /######################/ In above case, renderer 2 lags behind the renderer 1 i.e there is delay between two audio devices when actually heard. Please help me if some better implementation for join play. I have seen some thing as snd_pcm_link(). But i was not clear with its usage, while doing snd_pcm_writei ??? Thanks and regards, Irfan

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[alsa-devel] [PATCH] ASoC: sta32x: Optimize the array work to find rate_min and rate_max
by Axel Lin 03 Jan '12

03 Jan '12

For a given ir and fs, there is at most one possible match for the case mclk_ratios[ir][j].ratio * fs == freq. Thus we can break from the inner loop once a match is found. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/sta32x.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index fbd1450..7db6fa5 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -522,6 +522,7 @@ static int sta32x_set_dai_sysclk(struct snd_soc_dai *codec_dai, rate_min = fs; if (fs > rate_max) rate_max = fs; + break; } } } -- 1.7.5.4

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Re: [alsa-devel] [PATCH] ASoC: Add support for TI LM49453 Audio codec
by Mark Brown 03 Jan '12

03 Jan '12

On Tue, Jan 03, 2012 at 05:35:26AM -0800, Reddy, MR Swami wrote: > ASoC: Add support for TI LM49453 Audio codec. So, overall there's quite a few issues here. Many of them are fairly minor but there's a few big ones. The biggest issue is that this code won't compile, you need to submit code against development versions of the kernel. There's also quite a few errors with the use of snd_soc_update_bits() that make me wonder how well tested this code is. > +/* Register default values for LM49453 driver. */ > +static const u8 lm49453_reg_defs[LM49453_MAX_REGISTER + 1] = { > + [0x60] = 0x01, > + [0x67] = 0x01, This (and pretty much all I2C/SPI devices) really should be using regmap directly - the register map is pretty sparse so it would benefit from the rbtree cache. > +struct lm49453_priv { > + enum snd_soc_control_type control_type; Not needed since you only support I2C. > +static const struct snd_kcontrol_new lm49453_headset_left_mixer[] = { > +SOC_DAPM_SINGLE("Port1_1 HPL", LM49453_P0_DACHPL1_REG, 0, 1, 0), These should all have " Switch" at the end of the name so the UIs know how to handle them (similarly for all your other mixer controls). > +static const struct snd_kcontrol_new lm49453_snd_controls[] = { > + SOC_ENUM("Mic2Mode Capture Route", lm49453_mic2mode_enum), > + SOC_ENUM("DMIC12 Capture Route", lm49453_dmic12_cfg_enum), > + SOC_ENUM("DMIC34 Capture Route", lm49453_dmic34_cfg_enum), These three look like they should be DAPM? > + SOC_SINGLE_TLV("Mic1 Capture Volume", LM49453_P0_ADC_LEVELL_REG, > + 0, 6, 0, digital_tlv), > + SOC_SINGLE_TLV("Mic2 Capture Volume", LM49453_P0_ADC_LEVELR_REG, > + 0, 6, 0, digital_tlv), The register names suggest that these are really just left and right channels on the ADC, do they really only affect the microphone paths? > + SOC_SINGLE_TLV("PORT1_1_RX_LVL", LM49453_P0_PORT1_RX_LVL1_REG, > + 0, 3, 0, port_tlv), All these should have Volume at the end of their name so that UIs know how to display them. > + SOC_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, > + 0, 0x3F, 0, digital_tlv), > + SOC_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, > + 0, 0x3F, 0, digital_tlv), These sidetone controls all look like they could be combined into stereo controls - is there a pressing reason why are they done as mono? If there is a comment would probably help. > + SND_SOC_DAPM_INPUT("PORT1_1_RX"), > + SND_SOC_DAPM_INPUT("PORT1_2_RX"), > + SND_SOC_DAPM_INPUT("PORT1_3_RX"), > + SND_SOC_DAPM_INPUT("PORT1_4_RX"), > + SND_SOC_DAPM_INPUT("PORT1_5_RX"), > + SND_SOC_DAPM_INPUT("PORT1_6_RX"), > + SND_SOC_DAPM_INPUT("PORT1_7_RX"), > + SND_SOC_DAPM_INPUT("PORT1_8_RX"), > + SND_SOC_DAPM_INPUT("PORT2_1_RX"), > + SND_SOC_DAPM_INPUT("PORT2_2_RX"), INPUT widgets should generally be line inputs... looking at the DAPM routes these are actually digital widgets floating inside the CODEC. It's not clear why they're represented though as they don't have any control associated with them. If you just need widgets to make the routing easier then use PGAs, Inputs get treated specially in power control and are likely to confuse things. > + SND_SOC_DAPM_MICBIAS("AMIC1Bias", LM49453_P0_MICL_REG, 6, 0), > + SND_SOC_DAPM_MICBIAS("AMIC2Bias", LM49453_P0_MICR_REG, 6, 0), Don't use MICBIAS widgets, use SUPPLY widgets - MICBIAS widgets will (eventually) be phased out as SUPPLY widgets do the same job but are easier to use in machine drivers. > + SND_SOC_DAPM_SUPPLY("DMIC1supply", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_SUPPLY("DMIC2supply", SND_SOC_NOPM, 0, 0, NULL, 0), > + > + SND_SOC_DAPM_SUPPLY("Headset Rail", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_SUPPLY("Speaker Rail", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_SUPPLY("Earpiece Rail", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_SUPPLY("Haptic Rail", SND_SOC_NOPM, 0, 0, NULL, 0), None of these widgets will do anything, they don't have any power bits or callbacks associated with them so won't do anything. Just delete them. > + /* Port1 TX controls */ > + SND_SOC_DAPM_OUT_DRV("P1_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_3_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_4_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_5_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_6_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_7_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P1_8_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + > + /* Port2 TX controls */ > + SND_SOC_DAPM_OUT_DRV("P2_1_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("P2_2_TX", SND_SOC_NOPM, 0, 0, NULL, 0), > + > + SND_SOC_DAPM_MIXER("Port1 Playback", LM49453_P0_AUDIO_PORT1_BASIC_REG, > + 0, 0, NULL, 0), > + SND_SOC_DAPM_MIXER("Port2 Playback", LM49453_P0_AUDIO_PORT2_BASIC_REG, > + 0, 0, NULL, 0), Similar issues to the capture versions of these apply here. > + /* Sidetone controls */ > + SND_SOC_DAPM_OUT_DRV("Sidetone DAC HPL Playback", > + LM49453_P0_STN_SEL_REG, 0, 0, NULL, 0), > + SND_SOC_DAPM_OUT_DRV("Sidetone DAC HPR Playback", > + LM49453_P0_STN_SEL_REG, 1, 0, NULL, 0), What exactly do these register bits do? The use of OUT_DRV widgets looks very suspicious for a sidetone, an output driver would usually be a power amplifier or transducer driver so I suspect the power sequencing will be all wrong. It looks like these are mixer input controls which wouldn't normally be widgets at all. > + { "Port1_8 Mixer", NULL, "PORT1_8_RX_LVL" }, > + { "Port1_8 Mixer", NULL, "ADC Left" }, > + { "Port1_8 Mixer", NULL, "ADC Right" }, > + { "Port1_8 Mixer", NULL, "DMIC1 Left" }, > + { "Port1_8 Mixer", NULL, "DMIC1 Right" }, > + { "Port1_8 Mixer", NULL, "DMIC2 Left" }, > + { "Port1_8 Mixer", NULL, "DMIC2 Right" }, All this looks pretty odd, do the mixers really have no control over their inputs and just mix every possible input in? > + { "AMIC1Bias", NULL, "AMIC1"}, > + { "Mic1_Input Capture Route", "MIC1", "AMIC1Bias"}, Your CODEC driver should not be connecting the microphone biases, the connections are board specific. > +/* PLL P Values array */ > +int pll_p[] = { 143, /* For 8 fs p val */ > + 123, /* For 11.2 fs p val*/ > + 71, /* For 16 fs p val*/ > + 61, /* For 22.1 fs p val*/ > + 47, /* For 24 fs p val*/ > + 35, /* For 32 fs p val*/ > + 30, /* For 44.1 fs p val*/ > + 23, /* For 48 fs p val*/ > + 23 /* For 96fs p val*/ > +}; > + switch (lm49453->fs_rate) { > + case 8000: > + pll_div->p1 = pll_p[0]; > + break; Eew, no. Why are you indexing into the table with magic numbers from a switch statement to assign constants? That's not not really legible. Either put the frequencies into the table and look up in the table or put the constants directly into the switch statement. > +static int lm49453_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, > + int source, unsigned int freq_in, > + unsigned int freq_out) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); > + struct pll_state *state; > + struct _pll_div pll_div; > + unsigned int pwr_mask; > + int ret; > + > + memset(&pll_div, 0, sizeof(pll_div)); That memset is just going to mask warnings, it shouldn't be needed. > + /* Setting DAC clock dividers based on substream sample rate. */ > + switch (lm49453->fs_rate) { > + case 8000: > + case 16000: > + case 32000: > + case 24000: > + case 48000: > + val = 256; > + break; > + case 11025: > + case 22050: > + case 44100: > + val = 216; > + break; > + case 96000: > + val = 127; > + break; > + default: > + return -EINVAL; > + } val really isn't a meaningful variable name... > + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, val); > + snd_soc_write(codec, LM49453_P0_DAC_OT_CLK_DIV_REG, val); > + snd_soc_write(codec, LM49453_P0_DAC_HP_CLK_DIV_REG, val); ...and frankly I've no idea what this code is doing, I'd expect a clock divider configuration to vary depending on the input clock as well as the sample rate. > + switch (lm49453->sysclk) { > + case 12288000: > + case 26000000: > + case 19000000: > + val = 0; > + break; > + case 48000: > + case 32576: > + val = 1; > + break; > + default: > + return -EINVAL; > + } > + > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, BIT(4), val); I don't think this does what you think it does, your mask and value don't overlap at all. Have you tested? The whole thing looks very suspicous, running audio directly off a 32.768kHz clock would be unusual. I rather suspect this ought not to be done in hw_params() at all but instead should be in set_sysclk(). > +static int lm49453_set_dai_clkdiv(struct snd_soc_dai *codec_dai, > + int div_id, int div) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + > + switch (div_id) { > + case LM49453_PMC_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_PMC_CLK_DIV_REG, div); > + break; > + case LM49453_HSDET_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_HSDET_CLK_DIV_REG, div); > + break; > + case LM49453_DMIC_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_DMIC_CLK_DIV_REG, div); > + break; > + case LM49453_ADC_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_ADC_CLK_DIV_REG, div); > + break; > + case LM49453_DAC_CLK_DIV: > + snd_soc_write(codec, LM49453_P0_DAC_OT_CLK_DIV_REG, div); > + break; Again, not overburned with documentation here but I rather suspect that the driver ought to be able to figure out how to configure most of these dividers by itself given the SYSCLK and current sample rate. > + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { > + case SND_SOC_DAIFMT_CBS_CFS: > + aif_val = !(LM49453_P0_AUDIO_PORT1_BASIC_FMT_BIT); > + break; Use ~ if that's what you mean, though frankly I'm just guessing. > + case SND_SOC_DAIFMT_CBS_CFM: > + aif_val = LM49453_P0_AUDIO_PORT1_BASIC_FMT_BIT; > + break; > + case SND_SOC_DAIFMT_CBM_CFS: > + aif_val = ~LM49453_P0_AUDIO_PORT1_BASIC_FMT_BIT; > + break; > + case SND_SOC_DAIFMT_CBM_CFM: > + aif_val = LM49453_P0_AUDIO_PORT1_BASIC_FMT_BIT; > + break; You appear to be setting the same setting the same register value for CBS_CFM as for CBM_CFM. Are you sure the device can support all these clocking modes? > + snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, > + aif_val, aif_val); Another error with update_bits(). > +static int lm49453_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, > + unsigned int freq, int dir) > +{ > + struct snd_soc_codec *codec = dai->codec; > + struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); > + > + switch (freq) { > + case 12288000: /* Platform System Clock */ > + case 26000000: /* 26 MHz */ > + case 19200000: /* 19.2 MHz */ > + case 48000: /* 48 KHz */ > + case 32576: /* 32.576 KHz */ The first comment is specific to your board, the others are blindingly obvious - I'd suggest removing them all. > + lm49453->sysclk = freq; > + return 0; > + } > + return -EINVAL; Move this into the default: in the switch for legibility. > +static int lm49453_startup(struct snd_pcm_substream *substream, > + struct snd_soc_dai *codec_dai) > +{ > + unsigned int startup_mask; > + struct snd_soc_codec *codec = codec_dai->codec; > + dev_dbg(codec->dev, "lm49453_startup\n"); > + > + startup_mask = LM49453_P0_PMC_SETUP_CHIP_EN | > + LM49453_P0_PMC_SETUP_PLL_EN; > + snd_soc_update_bits(codec, LM49453_P0_PMC_SETUP_REG, startup_mask, > + startup_mask); CHIP_EN should be handled by DAPM or set_bias_level(), PLL_EN looks like it should be done by set_pll() but possibly also by DAPM. > +static int lm49453_hp_mute(struct snd_soc_dai *dai, int mute) > +{ > + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(0), > + (!!mute << 0)); > + snd_soc_update_bits(dai->codec, LM49453_P0_DAC_DSP_REG, BIT(1), > + (!!mute << 1)); Why are there two separate update bits operations here? > + case SND_SOC_BIAS_STANDBY: > + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { > + /* Power up with Headset active */ Why are you doing this? DAPM should be taking care of enabling outputs... > + reg = snd_soc_read(codec, LM49453_P0_PMC_SETUP_REG); > + reg &= ~(LM49453_P0_PMC_SETUP_CHIP_EN); > + snd_soc_write(codec, LM49453_P0_PMC_SETUP_REG, reg); > + } Use _update_bits(). > + case SND_SOC_BIAS_OFF: > + reg = snd_soc_read(codec, LM49453_P0_PMC_SETUP_REG); > + snd_soc_write(codec, LM49453_P0_PMC_SETUP_REG, > + (reg | LM49453_P0_PMC_SETUP_CHIP_EN)); _update_bits() if this should be here at all. > + break; > + } Indentation. > +int lm49453_headset_detected(struct snd_soc_codec *codec) > +{ > + u8 val = 0; > + val = snd_soc_read(codec, LM49453_P0_HSD_IRQ1_REG); > + return (val >> 6) & 1; > +} > +EXPORT_SYMBOL_GPL(lm49453_headset_detected); Don't invent your own API for accessory detection - use the standard mechanisms we've got for reporting events. > +static int lm49453_probe(struct snd_soc_codec *codec) > +{ > + int ret = 0; > + int reg; > + > + codec->dapm.idle_bias_off = 1; Set this in the snd_soc_codec_driver structure. > + .reg_word_size = sizeof(u8), > + .reg_cache_default = lm49453_reg_defs, > + .compress_type = SND_SOC_RBTREE_COMPRESSION, This won't compile with current kernels, advanced cache types are no longer supported in ASoC. You should use regmap directly. > +#define LM49453_CLKSRC_MCLK 1 > +#define LM49453_CLKSRC_PLL_1 2 > +#define LM49453_CLKSRC_PLL_2 3 > +#define LM49453_CLKSRC_PORT1_RX 4 > +#define LM49453_CLKSRC_PORT2_RX 5 > +#define LM49453_CLKSRC_DAC_HP 6 > +#define LM49453_CLKSRC_DAC_OT 7 At no point are any of these referenced in the code... > +#define LM49453_CLKSRC_ADC 12 > +#define LM49453_CLKSRC_DMIC1 13 > +#define LM49453_CLKSRC_DMIC2 14 Really? > +/* Clock divider Id's */ ' shouldn't be used for plurals.

1 0

[alsa-devel] [PATCH v2] ASoC: soc-pcm: Allocate PCM operations dynamically to support multiple DAIs
by Sangsu Park 03 Jan '12

03 Jan '12

The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch uses runtime's pcm_ops instead of global pcm_ops for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. --- sound/soc/soc-pcm.c | 38 ++++++++++++++++++-------------------- 1 files changed, 18 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec..cdc860a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -598,17 +598,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -616,10 +605,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops = &rtd->ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_pcm_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->pointer = soc_pcm_pointer; + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -643,20 +641,20 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->pointer = platform->driver->ops->pointer; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); -- 1.7.1

3 5

[alsa-devel] [PATCH] s3c24xx_simtec: fix typo
by Peter Meerwald 03 Jan '12

03 Jan '12

From: Peter Meerwald <p.meerwald(a)bct-electronic.com> --- sound/soc/samsung/s3c24xx_simtec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_simtec.c b/sound/soc/samsung/s3c24xx_simtec.c index a253bcc..dcef50c 100644 --- a/sound/soc/samsung/s3c24xx_simtec.c +++ b/sound/soc/samsung/s3c24xx_simtec.c @@ -343,7 +343,7 @@ int __devinit simtec_audio_core_probe(struct platform_device *pdev, snd_dev = platform_device_alloc("soc-audio", -1); if (!snd_dev) { - dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + dev_err(&pdev->dev, "failed to alloc soc-audio device\n"); ret = -ENOMEM; goto err_gpio; } -- 1.7.4.1

1 0

[alsa-devel] [PATCH] ALSA: HDA: Fix typo for ALC269VB_FIXUP_DMIC
by David Henningsson 03 Jan '12

03 Jan '12

This fixup is not actually used, so in practice this is just a cosmetic fix. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d24adbd..5e82acf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5320,7 +5320,7 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, - [ALC269_FIXUP_DMIC] = { + [ALC269VB_FIXUP_DMIC] = { .type = ALC_FIXUP_PINS, .v.pins = (const struct alc_pincfg[]) { { 0x12, 0x99a3092f }, /* int-mic */ -- 1.7.5.4

1 0

[alsa-devel] [PATCH v3] ASoC: soc-pcm: Allocate PCM operations dynamically to support multiple DAIs
by Sangsu Park 03 Jan '12

03 Jan '12

The original code does not cover the case that two DAIs(CPU) have different ASoC core PCM operations(like mmap, pointer...). Currently we have only one global soc_pcm_ops for ASoC core PCM operation. When two DAIs have different pointer functions, second DAI's pointer function is set for both first DAI and second DAI in case of original code. This patch uses runtime's pcm_ops instead of global pcm_ops for each DAIs. So each DAIs can have different ASoC core PCM operations. This is needed to support multiple DAIs. Signed-off-by: Sangsu Park <sangsu4u.park(a)samsung.com> --- sound/soc/soc-pcm.c | 38 ++++++++++++++++++-------------------- 1 files changed, 18 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8aa7cec..cdc860a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -598,17 +598,6 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) return offset; } -/* ASoC PCM operations */ -static struct snd_pcm_ops soc_pcm_ops = { - .open = soc_pcm_open, - .close = soc_pcm_close, - .hw_params = soc_pcm_hw_params, - .hw_free = soc_pcm_hw_free, - .prepare = soc_pcm_prepare, - .trigger = soc_pcm_trigger, - .pointer = soc_pcm_pointer, -}; - /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { @@ -616,10 +605,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_pcm_ops *soc_pcm_ops = &rtd->ops; struct snd_pcm *pcm; char new_name[64]; int ret = 0, playback = 0, capture = 0; + soc_pcm_ops->open = soc_pcm_open; + soc_pcm_ops->close = soc_pcm_close; + soc_pcm_ops->hw_params = soc_pcm_hw_params; + soc_pcm_ops->hw_free = soc_pcm_hw_free; + soc_pcm_ops->prepare = soc_pcm_prepare; + soc_pcm_ops->trigger = soc_pcm_trigger; + soc_pcm_ops->pointer = soc_pcm_pointer; + /* check client and interface hw capabilities */ snprintf(new_name, sizeof(new_name), "%s %s-%d", rtd->dai_link->stream_name, codec_dai->name, num); @@ -643,20 +641,20 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = pcm; pcm->private_data = rtd; if (platform->driver->ops) { - soc_pcm_ops.mmap = platform->driver->ops->mmap; - soc_pcm_ops.pointer = platform->driver->ops->pointer; - soc_pcm_ops.ioctl = platform->driver->ops->ioctl; - soc_pcm_ops.copy = platform->driver->ops->copy; - soc_pcm_ops.silence = platform->driver->ops->silence; - soc_pcm_ops.ack = platform->driver->ops->ack; - soc_pcm_ops.page = platform->driver->ops->page; + soc_pcm_ops->mmap = platform->driver->ops->mmap; + soc_pcm_ops->pointer = platform->driver->ops->pointer; + soc_pcm_ops->ioctl = platform->driver->ops->ioctl; + soc_pcm_ops->copy = platform->driver->ops->copy; + soc_pcm_ops->silence = platform->driver->ops->silence; + soc_pcm_ops->ack = platform->driver->ops->ack; + soc_pcm_ops->page = platform->driver->ops->page; } if (playback) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, soc_pcm_ops); if (capture) - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, soc_pcm_ops); if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); -- 1.7.1

1 0

Re: [alsa-devel] It looks like snd-soc-rx51 only works as built-in, not as a module
by Mark Brown 02 Jan '12

02 Jan '12

On Mon, Jan 02, 2012 at 08:33:03PM +0200, Felipe Contreras wrote: > On Sat, Dec 31, 2011 at 2:59 AM, Mark Brown > > Which is relevant because...? To repeat, the ordering of module loading > > is totally ortogonal as any ordering of modules is supported. You > > appear to be totally ignoring what I am writing here. > One thing how things _should_ be, and another is how things actually > _are_. Right now snd-soc-rx51 doesn't work if it's loaded before > snd_soc_tlv320aic3x. Period. Can you provide logs showing this (with debug logging enabled in soc-core)? If this is happening it's a very serious issue, though I am rather surprised that nobody else has noticed such an issue since the usual instantiation order would have the card appear before the CODEC (due to the fact that the card is normally a platform device) and the structure of the code makes it difficult to imagine a failure path. > That means that *right now*, snd-soc-rx51 works fine with udev, > because snd_soc_tlv320aic3x will be loaded at boot time. Now, if > snd-soc-rx51 is converted and loaded automatically by udev as well, > the issues might appear again. Not if we look at and resolve the actual issue. Once again, any instantation ordering should be supported. I really don't understand why you keep on talking about fixing the module ordering,

1 0

[alsa-devel] [PATCH v8 3/3 RESEND] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 02 Jan '12

02 Jan '12

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Added DAPM support for AUX1, AUX2 and INPGA (Moved here from DAPM patch) Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 77 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 77 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 62e6a9c..ab38e93 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -181,9 +181,14 @@ /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) +#define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ @@ -234,9 +239,19 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); /* ADC and DAC high pass filter f0 value */ static const char * const da7210_hpf_cutoff_txt[] = { @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), + /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, @@ -421,26 +447,42 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), }; /* In Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), }; /* Out Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), }; /* Out Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; /* Mono Mixer */ static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), }; @@ -451,14 +493,23 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), /* Input PGAs */ SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + /* Input Mixers */ SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixl_controls[0], @@ -514,12 +565,21 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Input path */ {"Mic Left", NULL, "MICL"}, {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, {"In Mixer Left", "Mic Left Switch", "Mic Left"}, {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, {"In Mixer Right", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, {"INPGA Left", NULL, "In Mixer Left"}, {"ADC Left", NULL, "INPGA Left"}, @@ -528,9 +588,20 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"ADC Right", NULL, "INPGA Right"}, /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, @@ -887,6 +958,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

2 8

[alsa-devel] [PATCH 1/3] ASoC: Use dai_fmt in corgi machine driver
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/corgi.c | 14 ++------------ 1 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 5e5004a..5ff6dac 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -142,18 +142,6 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, break; } - /* set codec DAI configuration */ - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - - /* set cpu DAI configuration */ - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk, SND_SOC_CLOCK_IN); @@ -311,6 +299,8 @@ static struct snd_soc_dai_link corgi_dai = { .platform_name = "pxa-pcm-audio", .codec_name = "wm8731.0-001b", .init = corgi_wm8731_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, .ops = &corgi_ops, }; -- 1.7.5.4

3 6

[alsa-devel] [PATCH v2 2/4] ASoC: cx20442: add bias control over a platform provided regulator
by Janusz Krzysztofik 02 Jan '12

02 Jan '12

Now that a regulator device for controlling the codec chip reset state over a platform agnostic regulator API is available on the only board using this driver so far, extend the driver with a bias control function which will request virtual power to the codec chip from that virtual regulator, and will supersede the present implementation existing at the sound card level. Thanks to the regulator sharing mechanism, both the old (the sound card) and the new (the codec) implementations should coexist smoothly until the sound card file is updated. For this to work as expected, update the sound card .set_bias_level callback to not touch codec->dapm.bias_level. While extending the cx20442 structure, drop unused control_type member. Created against linxu-3.2-rc6, tested on top of patch 1/4 "ARM: OMAP1: ams-delta: set up a regulator over the modem reset GPIO pin". Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Changes against initial version: * don't track the regulator enable/disable state, compare new bias level against the old one instead; thanks to Mark Brown for suggesting this change, * update the Amstrad Delta sound card file to not touch the codec bias level for the above change to work as expected, * display a warning if getting the regulator fails. sound/soc/codecs/cx20442.c | 48 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/omap/ams-delta.c | 8 ++---- 2 files changed, 49 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index bc7067d..941c527 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -16,6 +16,7 @@ #include <linux/tty.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/regulator/consumer.h> #include <sound/core.h> #include <sound/initval.h> @@ -25,8 +26,8 @@ struct cx20442_priv { - enum snd_soc_control_type control_type; void *control_data; + struct regulator *por; }; #define CX20442_PM 0x0 @@ -324,6 +325,38 @@ static struct snd_soc_dai_driver cx20442_dai = { }, }; +static int cx20442_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct cx20442_priv *cx20442 = snd_soc_codec_get_drvdata(codec); + int err = 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level != SND_SOC_BIAS_STANDBY) + break; + if (IS_ERR(cx20442->por)) + err = PTR_ERR(cx20442->por); + else + err = regulator_enable(cx20442->por); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level != SND_SOC_BIAS_PREPARE) + break; + if (IS_ERR(cx20442->por)) + err = PTR_ERR(cx20442->por); + else + err = regulator_disable(cx20442->por); + break; + default: + break; + } + if (!err) + codec->dapm.bias_level = level; + + return err; +} + static int cx20442_codec_probe(struct snd_soc_codec *codec) { struct cx20442_priv *cx20442; @@ -331,9 +364,13 @@ static int cx20442_codec_probe(struct snd_soc_codec *codec) cx20442 = kzalloc(sizeof(struct cx20442_priv), GFP_KERNEL); if (cx20442 == NULL) return -ENOMEM; - snd_soc_codec_set_drvdata(codec, cx20442); + cx20442->por = regulator_get(codec->dev, "POR"); + if (IS_ERR(cx20442->por)) + dev_warn(codec->dev, "failed to get the regulator"); cx20442->control_data = NULL; + + snd_soc_codec_set_drvdata(codec, cx20442); codec->hw_write = NULL; codec->card->pop_time = 0; @@ -350,6 +387,12 @@ static int cx20442_codec_remove(struct snd_soc_codec *codec) tty_hangup(tty); } + if (!IS_ERR(cx20442->por)) { + /* should be already in STANDBY, hence disabled */ + regulator_put(cx20442->por); + } + + snd_soc_codec_set_drvdata(codec, NULL); kfree(cx20442); return 0; } @@ -359,6 +402,7 @@ static const u8 cx20442_reg; static struct snd_soc_codec_driver cx20442_codec_dev = { .probe = cx20442_codec_probe, .remove = cx20442_codec_remove, + .set_bias_level = cx20442_set_bias_level, .reg_cache_default = &cx20442_reg, .reg_cache_size = 1, .reg_word_size = sizeof(u8), diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index be81bc7..0462c7e 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -431,22 +431,20 @@ static int ams_delta_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_codec *codec = card->rtd->codec; - switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + if (card->dapm.bias_level == SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, AMS_DELTA_LATCH2_MODEM_NRESET); break; case SND_SOC_BIAS_OFF: - if (codec->dapm.bias_level != SND_SOC_BIAS_OFF) + if (card->dapm.bias_level != SND_SOC_BIAS_OFF) ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET, 0); } - codec->dapm.bias_level = level; + card->dapm.bias_level = level; return 0; } -- 1.7.3.4

2 1

[alsa-devel] [PATCH] ASoC: Fix return value of wm8903_gpio_direction_in() and wm8903_gpio_direction_out()
by Axel Lin 02 Jan '12

02 Jan '12

We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm8903.c | 18 ++++++++++++++---- 1 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d88b727..c91fb2f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1777,13 +1777,18 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK; val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | WM8903_GP1_DIR; - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) @@ -1803,13 +1808,18 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; + int ret; mask = WM8903_GP1_FN_MASK | WM8903_GP1_DIR_MASK | WM8903_GP1_LVL_MASK; val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | (value << WM8903_GP2_LVL_SHIFT); - return snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, + mask, val); + if (ret < 0) + return ret; + + return 0; } static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: Convert saarb to table based DAPM init
by Axel Lin 02 Jan '12

02 Jan '12

Also remove a unused ret variable to silence the build warning. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/saarb.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index b2be225..c34146b 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -51,7 +51,7 @@ static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { }; /* saarb machine audio map */ -static const struct snd_soc_dapm_route audio_map[] = { +static const struct snd_soc_dapm_route saarb_audio_map[] = { {"Headset Stereophone", NULL, "HS1"}, {"Headset Stereophone", NULL, "HS2"}, @@ -121,17 +121,17 @@ static struct snd_soc_card snd_soc_card_saarb = { .owner = THIS_MODULE, .dai_link = saarb_dai, .num_links = ARRAY_SIZE(saarb_dai), + + .dapm_widgets = saarb_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(saarb_dapm_widgets), + .dapm_routes = saarb_audio_map, + .num_dapm_routes = ARRAY_SIZE(saarb_audio_map), }; static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - snd_soc_dapm_new_controls(dapm, saarb_dapm_widgets, - ARRAY_SIZE(saarb_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); /* connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); -- 1.7.5.4

2 4

[alsa-devel] [PATCH 1/4] ASoC: Use dai_fmt in saarb machine driver
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/saarb.c | 11 ++--------- 1 files changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c index 2e21712..b2be225 100644 --- a/sound/soc/pxa/saarb.c +++ b/sound/soc/pxa/saarb.c @@ -92,15 +92,6 @@ static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret < 0) - return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); return ret; @@ -119,6 +110,8 @@ static struct snd_soc_dai_link saarb_dai[] = { .platform_name = "pxa-pcm-audio", .codec_name = "88pm860x-codec", .init = saarb_pm860x_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &saarb_i2s_ops, }, }; -- 1.7.5.4

3 8

[alsa-devel] [PATCH 1/3] ASoC: Fix return value of ak4641_pcm_set_dai_fmt()
by Axel Lin 02 Jan '12

02 Jan '12

We can't just pass back the return value of snd_soc_update_bits() as it will be 1 if a bit changed rather than zero. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4641.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 266ebea..c4d165a 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -339,6 +339,7 @@ static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; u8 btif; + int ret; /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -358,7 +359,11 @@ static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - return snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); + ret = snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif); + if (ret < 0) + return ret; + + return 0; } static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, -- 1.7.5.4

2 3

Re: [alsa-devel] [PATCH 0/10] use devm_ functions
by Mark Brown 02 Jan '12

02 Jan '12

On Thu, Dec 29, 2011 at 05:51:19PM +0100, Julia Lawall wrote: > The following is a semantic patch (http://coccinelle.lip6.fr/) that > introduces uses of various devm_ functions. The main strategy is Applied all except the jz4740 patch which Axel already did, thanks.

1 0

[alsa-devel] [PATCH 01/14] ASoC: Convert alc5623 to devm_kzalloc()
by Axel Lin 02 Jan '12

02 Jan '12

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/alc5623.c | 11 +++-------- 1 files changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 6a9b621..3feee56 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -1022,7 +1022,8 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); - alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL); + alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), + GFP_KERNEL); if (alc5623 == NULL) return -ENOMEM; @@ -1044,7 +1045,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, alc5623_dai.name = "alc5623-hifi"; break; default: - kfree(alc5623); return -EINVAL; } @@ -1053,20 +1053,15 @@ static int alc5623_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); - if (ret != 0) { + if (ret != 0) dev_err(&client->dev, "Failed to register codec: %d\n", ret); - kfree(alc5623); - } return ret; } static int alc5623_i2c_remove(struct i2c_client *client) { - struct alc5623_priv *alc5623 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - kfree(alc5623); return 0; } -- 1.7.5.4

3 17

[alsa-devel] Non-cacheable memory to alsa ring buffer
by Babu, Ramesh 02 Jan '12

02 Jan '12

Hi How can I allocate non-cacheable region for ring buffer? I have an issue, where my DMA is not fetching right data from Memory. I suspect that caching might be issue here. I am using the below api to allocate memory: retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), RING_MIN_BUFFER, RING_MAX_BUFFER); Where, RING_MIN_BUFFER = RING_MAX_BUFFER = (8*1024) Thanks Ramesh

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[alsa-devel] [PATCH v8 3/3 RESEND] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 02 Jan '12

02 Jan '12

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- Changes since v7: - Added DAPM support for AUX1, AUX2 and INPGA (Moved here from DAPM patch) Changes since v2: - Removed static enable of mic and aux, as now DAPM will take care of that Changes since v1: - Removed explicit setting of default gains - Removed control to set mic bias voltage --- sound/soc/codecs/da7210.c | 77 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 77 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 0ebcbd5..ff084a6 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -181,9 +181,14 @@ /* AUX1_L bit fields */ #define DA7210_AUX1_L_VOL (0x3F << 0) +#define DA7210_AUX1_L_EN (1 << 7) /* AUX1_R bit fields */ #define DA7210_AUX1_R_VOL (0x3F << 0) +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) /* Minimum INPGA and AUX1 volume to enable noise suppression */ #define DA7210_INPGA_MIN_VOL_NS 0x0A /* 10.5dB */ @@ -234,9 +239,19 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); /* ADC and DAC high pass filter f0 value */ static const char const *da7210_hpf_cutoff_txt[] = { @@ -344,6 +359,17 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), + /* DAC Equalizer controls */ SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, @@ -421,26 +447,42 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = { SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0), SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_INMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_L, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_INMIX_L, 4, 1, 0), }; /* In Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = { SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0), SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_INMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_INMIX_R, 3, 1, 0), + SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_INMIX_R, 4, 1, 0), }; /* Out Mixer Left */ static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = { + SOC_DAPM_SINGLE("Aux1 Left Switch", DA7210_OUTMIX_L, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_L, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_L, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_L, 3, 1, 0), SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0), }; /* Out Mixer Right */ static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = { + SOC_DAPM_SINGLE("Aux1 Right Switch", DA7210_OUTMIX_R, 0, 1, 0), + SOC_DAPM_SINGLE("Aux2 Switch", DA7210_OUTMIX_R, 1, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUTMIX_R, 2, 1, 0), + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUTMIX_R, 3, 1, 0), SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0), }; /* Mono Mixer */ static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = { + SOC_DAPM_SINGLE("INPGA Right Switch", DA7210_OUT2, 3, 1, 0), + SOC_DAPM_SINGLE("INPGA Left Switch", DA7210_OUT2, 4, 1, 0), SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0), SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0), }; @@ -451,14 +493,23 @@ static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + SND_SOC_DAPM_INPUT("AUX2"), /* Input PGAs */ SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Left", DA7210_STARTUP3, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux1 Right", DA7210_STARTUP3, 3, 1, NULL, 0), + SND_SOC_DAPM_PGA("Aux2 Mono", DA7210_STARTUP3, 4, 1, NULL, 0), SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0), + /* MICBIAS */ + SND_SOC_DAPM_SUPPLY("Mic Bias", DA7210_MIC_L, 6, 0, NULL, 0), + /* Input Mixers */ SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0, &da7210_dapm_inmixl_controls[0], @@ -514,12 +565,21 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { /* Input path */ {"Mic Left", NULL, "MICL"}, {"Mic Right", NULL, "MICR"}, + {"Aux1 Left", NULL, "AUX1L"}, + {"Aux1 Right", NULL, "AUX1R"}, + {"Aux2 Mono", NULL, "AUX2"}, {"In Mixer Left", "Mic Left Switch", "Mic Left"}, {"In Mixer Left", "Mic Right Switch", "Mic Right"}, + {"In Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"In Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Left", "Outmix Left Switch", "Out Mixer Left"}, {"In Mixer Right", "Mic Right Switch", "Mic Right"}, {"In Mixer Right", "Mic Left Switch", "Mic Left"}, + {"In Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"In Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"In Mixer Right", "Outmix Right Switch", "Out Mixer Right"}, {"INPGA Left", NULL, "In Mixer Left"}, {"ADC Left", NULL, "INPGA Left"}, @@ -528,9 +588,20 @@ static const struct snd_soc_dapm_route da7210_audio_map[] = { {"ADC Right", NULL, "INPGA Right"}, /* Output path */ + {"Out Mixer Left", "Aux1 Left Switch", "Aux1 Left"}, + {"Out Mixer Left", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Left", "INPGA Left Switch", "INPGA Left"}, + {"Out Mixer Left", "INPGA Right Switch", "INPGA Right"}, {"Out Mixer Left", "DAC Left Switch", "DAC Left"}, + + {"Out Mixer Right", "Aux1 Right Switch", "Aux1 Right"}, + {"Out Mixer Right", "Aux2 Switch", "Aux2 Mono"}, + {"Out Mixer Right", "INPGA Right Switch", "INPGA Right"}, + {"Out Mixer Right", "INPGA Left Switch", "INPGA Left"}, {"Out Mixer Right", "DAC Right Switch", "DAC Right"}, + {"Mono Mixer", "INPGA Right Switch", "INPGA Right"}, + {"Mono Mixer", "INPGA Left Switch", "INPGA Left"}, {"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"}, {"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"}, @@ -887,6 +958,12 @@ static int da7210_probe(struct snd_soc_codec *codec) snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_AUX1_R_EN); + /* Enable Aux2 */ + snd_soc_write(codec, DA7210_AUX2, DA7210_AUX2_EN); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] USB-Audio and XHCI (USB 3)
by Aurélien Leblond 01 Jan '12

01 Jan '12

Hi everyone, I was wondering if anyone managed to run a USB 2 soundcard on a USB3 (XHCI) port? I'm running the kernel 3.2 rc7. With the M-Audio Fast Track Ultra (USB 2): The output of dmesg is this: [ 475.094595] usb 3-1: new high-speed USB device number 2 using xhci_hcd [ 475.108605] usb 3-1: config 1 interface 3 altsetting 0 bulk endpoint 0x7 has invalid maxpacket 8 [ 475.108615] usb 3-1: config 1 interface 3 altsetting 0 bulk endpoint 0x87 has invalid maxpacket 8 [ 475.108958] xhci_hcd 0000:04:00.0: WARN: short transfer on control ep [ 475.109328] xhci_hcd 0000:04:00.0: WARN: short transfer on control ep [ 475.109879] xhci_hcd 0000:04:00.0: WARN: short transfer on control ep [ 475.110564] xhci_hcd 0000:04:00.0: ERROR: unexpected command completion code 0x11. [ 475.110585] usb 3-1: can't set config #1, error -22 The snd-usb-audio module is not probed automatically, and if I modprobe it manually, no sound card is detected. The error message is described in this bug: https://bugzilla.novell.com/show_bug.cgi?id=679944 but if I understand well, kernel wise, everything is "ok" and the issue lies in Alsa The output of lsusb is: Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0020 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 003: ID 050d:0307 Belkin Components Bus 002 Device 003: ID 0408:1af1 Quanta Computer, Inc. Bus 001 Device 004: ID 046d:c51b Logitech, Inc. V220 Cordless Optical Mouse for Notebooks Bus 003 Device 002: ID 0763:2080 Midiman M-Audio RunTime DFU The Edirol UA-25Ex (USB 1) works fine, I get a few warnings in dmesg but the sound card is detected and functional... Let me know if I should provide any more info, or outputs, anything that can help. Kind Regards, Aurélien

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[alsa-devel] [intel-hda] ALC663 question.
by Andrew 01 Jan '12

01 Jan '12

Hello, I'm trying to get things working correctly for ASUS N75SF laptop and it looks like I'm somewhat stuck. The laptop has an ALC663 codec, 2 front speakers and an external subwoofer. The 'auto' settings on 3.1.6 kernel provided me no meens of enabling the subwoofer, the automute switch didn't work either. I threw on a patch with a quirk, that fixed most of the things except one: Whatever I do, I get lfe to output front-right channel. I followed alc663_asus_21jd_clfe example, but still that only gives me a volume control of that mixer. I'm not very familiar with alsa/hda codebase, so I might be missing some trivia. Does the code have to register a pcm for each dac? If so, the problem is in alc_build_pcms, since it looks like it only uses dac_nids[0]. Setting spec->multiout.max_channels in init code does not help. my current n75sf_setup code is as follows: --- cut --- struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic_pin = 0x18; spec->int_mic_pin = 0x19; spec->auto_mic = 1; --- cut --- _From the codec's point of view, there are 3 dacs, nids 0x02 0x03 Either of these can serve stereo to front or headphone plug via mixers nid 0x4 dac is the for lfe, only the second channel is wired. pin 0x16 (lfe) can only get stream from 0x4, which gets identical data as dac 0x2 This is the point I do not know where to look for the bug. So far, my patch adds: * Proper automuting toggle switch (subwoofer now gets disabled as well, when headphones are plugged). * Volume control for LFE channel (the stream is still identical to front right) * Channel mode switcher (2ch, front only, 3ch - front + subwoofer enabled) * (Mostly) working Headphone pin mode selection enum. Things I still want to fix before sending in the patch: * 5ch mode with headphone out configured as line-out for the rear speakers, (Needs some things filled out and some testing as well) * Toggle 2ch/3ch mode on subwoofer insertion (Still needs some investigation and testing) * Auto-mic and external mic need testing. Regards, Andrew

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[alsa-devel] Period time/size in adaptive USB - alignment to nrpacks?
by Pavel Hofman 01 Jan '12

01 Jan '12

Hi, I have been analyzing USB URBs in wireshark. Output stream, adaptive mode, 16 bits, 48kHz, stereo. Each URB is issued always at nrpacks milliseconds. Each URB transfers 48 samples * 2 bytes * 2 channels * nrpacks milliseconds of bytes. The last URB of period keeps the same pace (nrpacks milliseconds), however holds only the remaining number of bytes. This creates ripples in the data stream bitrate, possibly leading to increased jitter of the PLL'd clock in the USB receiver. When period size is aligned to multiples of samplerate/1000 * audio frame bytes * nrpacks, all URBs carry constant number of bytes. I am wondering if for the xx.000 Hz samplerates it made sense to offer only these aligned multiples in snd_pcm_hw_params_set_period_time_near. Thank you for your expert opinion. Best regards, Pavel.

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[alsa-devel] ASoC updates for 3.3
by Mark Brown 31 Dec '11

31 Dec '11

The following changes since commit c5cf4dbc7f804bb4ff02a065b927bd8688204253: ASoC: Add trivial pm_runtime usage to Samsung DAI drivers (2011-12-23 12:08:32 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.3 Axel Lin (8): ASoC: Convert da7210 to devm_kzalloc() ASoC: Convert lm4857 to devm_kzalloc() ASoC: Convert uda1380 to devm_kzalloc() ASoC: Convert 88pm860x-codec to devm_kzalloc() ASoC: Fix build error in sound/soc/kirkwood/kirkwood-i2s.c ASoC: Use dai_fmt in kirkwood-openrd machine driver ASoC: Use dai_fmt in kirkwood-t5325 machine driver ASoC: Convert kirkwood-t5325 to table based DAPM init Joachim Eastwood (1): ASoC: wm8776: add missing break in sample size switch Mark Brown (1): Merge branch 'for-3.2' into for-3.3 sound/soc/codecs/88pm860x-codec.c | 7 ++----- sound/soc/codecs/da7210.c | 7 ++----- sound/soc/codecs/lm4857.c | 13 ++----------- sound/soc/codecs/uda1380.c | 6 ++---- sound/soc/codecs/wm8776.c | 1 + sound/soc/kirkwood/kirkwood-i2s.c | 3 +-- sound/soc/kirkwood/kirkwood-openrd.c | 14 ++------------ sound/soc/kirkwood/kirkwood-t5325.c | 24 +++++++----------------- sound/soc/kirkwood/kirkwood.h | 1 - 9 files changed, 19 insertions(+), 57 deletions(-)

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[alsa-devel] [PATCH 0/6] snd-usb endpoint rework, version 6
by Daniel Mack 31 Dec '11

31 Dec '11

Fortunately, the effects reported by Aurélien were easy to fix, and indeed just regressions from previous version of the patch set. Those are now out of the way, so here's version 6. One thing that has also been added here is a sixth patch that adds some documentation to endpoint.c to explain the concept of the new implementation. Thanks, and please have more test results coming. Daniel Daniel Mack (6): ALSA: snd-usb: add snd_usb_audio-wide mutex ALSA: snd-usb: implement new endpoint streaming model ALSA: snd-usb: switch over to new endpoint streaming logic ALSA: snd-usb: remove old streaming logic ALSA: snd-usb: add support for implicit feedback ALSA: snd-usb: add some documentation sound/usb/card.c | 10 +- sound/usb/card.h | 62 ++ sound/usb/endpoint.c | 1602 ++++++++++++++++++++++++++++---------------------- sound/usb/endpoint.h | 32 +- sound/usb/pcm.c | 440 ++++++++++++--- sound/usb/stream.c | 31 +- sound/usb/usbaudio.h | 2 + 7 files changed, 1381 insertions(+), 798 deletions(-) -- 1.7.7.4

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Re: [alsa-devel] [PATCH 0/6] snd-usb endpoint rework, version 6
by Grant Diffey 31 Dec '11

31 Dec '11

> Yes, any kind of USB audio device is good. > > Thanks for heling test! :) > > > So sadly not all is well in my world the logitech is only registering as a mono device for some reason. nevyn@cetacea:~$ speaker-test -c 2 -D hw:Headset speaker-test 1.0.24.2 Playback device is hw:Headset Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Channels count (2) not available for playbacks: Invalid argument Setting of hwparams failed: Invalid argument it's clearly stereo. lsusb -vv attached for reference h

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[alsa-devel] [PATCH 1/6] ASoC: atmel: Add .owner to struct snd_soc_card
by Axel Lin 31 Dec '11

31 Dec '11

Missed .owner of struct snd_soc_card will prevent the module from being removed from underneath its users. Reported-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/atmel/sam9g20_wm8731.c | 1 + sound/soc/atmel/snd-soc-afeb9260.c | 1 + 2 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 0377c54..c883514 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -189,6 +189,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { static struct snd_soc_card snd_soc_at91sam9g20ek = { .name = "AT91SAMG20-EK", + .owner = THIS_MODULE, .dai_link = &at91sam9g20ek_dai, .num_links = 1, .set_bias_level = at91sam9g20ek_set_bias_level, diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c index d427e92..4ca667d 100644 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ b/sound/soc/atmel/snd-soc-afeb9260.c @@ -135,6 +135,7 @@ static struct snd_soc_dai_link afeb9260_dai = { /* Audio machine driver */ static struct snd_soc_card snd_soc_machine_afeb9260 = { .name = "AFEB9260", + .owner = THIS_MODULE, .dai_link = &afeb9260_dai, .num_links = 1, }; -- 1.7.5.4

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Re: [alsa-devel] It looks like snd-soc-rx51 only works as built-in, not as a module
by Mark Brown 31 Dec '11

31 Dec '11

On Fri, Dec 30, 2011 at 07:34:29PM +0000, Måns Rullgård wrote: > Måns Rullgård <mans(a)mansr.com> writes: > > In which tree? I haven't seen anything of the kind. The random sampling of drivers I looked at seemed to be doing something sensible. > That would be this one: > http://article.gmane.org/gmane.linux.alsa.devel/89052 Oh, well. Liam is a bit lossy sometimes. You really need to repost patches if they're worthwhile, don't just let them rot.

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Re: [alsa-devel] It looks like snd-soc-rx51 only works as built-in, not as a module
by Mark Brown 31 Dec '11

31 Dec '11

On Fri, Dec 30, 2011 at 10:00:55PM +0200, Felipe Contreras wrote: > On Thu, Dec 29, 2011 at 11:54 PM, Mark Brown > > On Thu, Dec 29, 2011 at 11:51:11PM +0200, Felipe Contreras wrote: > >> >> > That should not be required, the modules should be loadable in any > >> >> > order. > >> Yes, but udev loads snd_soc_tlv320aic3x, not snd-soc-rx51. > > That is compltelely orthogonal to what you were saying above about the > > ordering of module loading. > Not really, because if both are compiled as modules, > snd_soc_tlv320aic3x will always be loaded first (automatically by > udev). Which is relevant because...? To repeat, the ordering of module loading is totally ortogonal as any ordering of modules is supported. You appear to be totally ignoring what I am writing here. > > The reason the driver is not loaded automatically is that the OMAP > > machine drivers have not been converted to platform devices. > Well, if that's the case then there's a real issue. There doesn't seem > to be a MODULE_DEPENDS(), or anything like that. A MODULE_DEPENDS() would also be irrelevant here. > snd-soc-rx51 seems to depend on snd_soc_tlv320aic3x through > codec_dai_name, and codec_name, and so on. I don't know how this > dependency is supposed to work out. All the drivers should load via the normal driver instantiation mechainsms and when they're all there they'll get matched together.

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Re: [alsa-devel] [PATCH 0/6] snd-usb endpoint rework, version 6
by Daniel Mack 30 Dec '11

30 Dec '11

On 12/30/2011 04:16 PM, Felix Homann wrote: > Hi Daniel, > > 2011/12/30 Daniel Mack <zonque(a)gmail.com>: >> >> Any chance that more people who've been waiting for support of the FTU >> can use the lazy time between the years and quickly test this set? > > Could you please put a combined patch on your github place? Done: https://gist.github.com/1168715 > Should it be OK to use the alsa-driver/alsa-kernel update approach or > would I need to build a complete kernel image? Take Linus' tree, merge Takashi's tree (from git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git master) and then apply the gist. Daniel

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Re: [alsa-devel] It looks like snd-soc-rx51 only works as built-in, not as a module
by Mark Brown 30 Dec '11

30 Dec '11

On Thu, Dec 29, 2011 at 11:51:11PM +0200, Felipe Contreras wrote: > On Thu, Dec 29, 2011 at 11:27 PM, Mark Brown > > On Thu, Dec 29, 2011 at 11:25:50PM +0200, Felipe Contreras wrote: > >> On Thu, Dec 29, 2011 at 8:37 PM, Mark Brown > >> > On Sat, Dec 10, 2011 at 05:59:57PM +0200, Felipe Contreras wrote: > >> >> It was not working for me, but it seems the problem was related to > >> >> mdev/udev; snd_soc_tlv320aic3x has to be loaded before snd-soc-rx51. > >> > That should not be required, the modules should be loadable in any > >> > order. > >> Are you sure? I recall having a similar discussion with Rusell King, > >> and the conclusion is that certain modules are supposed to be loaded > >> by udev at boot time, and it seems snd_soc_tlv320aic3x is one of them. > > I'm absolutely positive. ASoC supports loading the modules in any > > order. udev is supposed to load anything autoloadable at boot time, > > including I2C devices, but that's orthogonal to the order in which > > things actually get loaded - udev can randomly reorder things if it > > feels like it. > Yes, but udev loads snd_soc_tlv320aic3x, not snd-soc-rx51. That is compltelely orthogonal to what you were saying above about the ordering of module loading. The reason the driver is not loaded automatically is that the OMAP machine drivers have not been converted to platform devices.

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[alsa-devel] pcm timer resolution out of range (rate = 0, period_size = 1), Division by zero in kernel
by Appalayagari Sreedhar 30 Dec '11

30 Dec '11

Hi, We are working with 2.6.32 kernel. we are working with ALSA model codec driver for conexant chipset. we are using OSS emulation for audio playback and recording. we are not setting the fragmenttaion block using OSS ioctl function (SNDCTL_DSP_SETFRAGMENT) , assuming that it will be taken care by the ALSA framework based on the sampling rate, bits per sample etc. we are facing the below issue randomly. ALSA sound/core/pcm_timer.c:71: -calling VIDIOC_pcm timer resolution out of range (rate = 0, period_size = 1) Division by zero in kernel. We tried to program the fragmentation block after the sampling rate, bits per sample etc are configured. it is working fine for 5 or 6 iterations for recording and playback. after that the audio quality is gone bad. the audio is not smooth and it is sharp. could you please help to resolve this problem. Thank you, Sreedhar.

2 3

[alsa-devel] [PATCH v2 3/4] ARM: OMAP1: ams-delta: update the modem to use regulator API
by Janusz Krzysztofik 30 Dec '11

30 Dec '11

After the CX20442 codec driver already takes care of enabling the codec power for itself, but before dropping the old bias control method from the Amstrad Delta ASoC sound card file, which in fact keeps the modem power always on, even after the ASoC device close for now, extend the modem setup with a power management callback, which toggles the regulator up to the modem's needs, reusing the previously set up regulator consumer for this. Also, drop the MODEM_NRESET pin setup from the modem initialization procedure, as this operation was already ineffective since patch 1/4, and not needed because the regulator is set up as initially enabled. Depends on patch 1/4 "ARM: OMAP1: ams-delta: set up a regulator over the modem reset GPIO pin" to apply cleanly, and requires patch 2/4 "ASoC: cx20442: add bias control over a platform provided regulator" for the sound card / codec bundle to still work. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- No functional changes against initial version. arch/arm/mach-omap1/board-ams-delta.c | 22 +++++++++++++++++----- 1 files changed, 17 insertions(+), 5 deletions(-) diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index 074d3b5..3389237 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -526,6 +526,16 @@ static void __init ams_delta_init(void) omap_writew(omap_readw(ARM_RSTCT1) | 0x0004, ARM_RSTCT1); } +static void modem_pm(struct uart_port *port, unsigned int state, unsigned old) +{ + struct modem_private_data *priv = port->private_data; + + if (state == old) + return; + + regulator_toggle(priv, state == 0); +} + static struct plat_serial8250_port ams_delta_modem_ports[] = { { .membase = IOMEM(MODEM_VIRT), @@ -536,6 +546,8 @@ static struct plat_serial8250_port ams_delta_modem_ports[] = { .iotype = UPIO_MEM, .regshift = 1, .uartclk = BASE_BAUD * 16, + .pm = modem_pm, + .private_data = &modem_priv, }, { }, }; @@ -584,9 +596,8 @@ static int __init late_init(void) mutex_init(&modem_priv.consumer.lock); modem_priv.regulator = ERR_PTR(-ENODEV); - ams_delta_latch2_write( - AMS_DELTA_LATCH2_MODEM_NRESET | AMS_DELTA_LATCH2_MODEM_CODEC, - AMS_DELTA_LATCH2_MODEM_NRESET | AMS_DELTA_LATCH2_MODEM_CODEC); + ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, + AMS_DELTA_LATCH2_MODEM_CODEC); err = platform_device_register(&ams_delta_modem_device); if (err) @@ -595,8 +606,9 @@ static int __init late_init(void) /* * Once the modem device is registered, the modem_nreset * regulator can be requested on behalf of that device. - * The regulator is used via ams_delta_latch_write() - * by the modem and ASoC drivers until updated. + * In addition to the modem .pm callback, that regulator + * is still used via the ams_delta_latch_write() wrapper + * by the ASoC driver until updated. */ modem_priv.regulator = regulator_get(&ams_delta_modem_device.dev, "RESET#"); -- 1.7.3.4

1 0

[alsa-devel] [PATCH v2 1/4] ARM: OMAP1: ams-delta: set up regulator over modem reset GPIO pin
by Janusz Krzysztofik 30 Dec '11

30 Dec '11

The Amstrad Delta on-board latch2 bit named MODEM_NRESET, now available as a GPIO pin AMS_DELTA_GPIO_PIN_NMODEM_RESET, is used to power up/down (bring into/out of a reset state) two distinct on-board devices simultaneously: the modem, and the voice codec. As a consequence, that bit is, or can be, manipulated concurrently by two drivers, or their platform provided hooks. Instead of updating those drivers to use the gpiolib API as a new method of controlling the MODEM_NRESET pin state, like it was done to other drivers accessing latch2 pins, and still being vulnerable to potential concurrency conflicts, or trying to solve that sharing issue with a custom piece of code, set up a fixed regulator device on top of that GPIO pin, with the intention of updating both drivers to manipulate that regulator, not the GPIO pin directly. Before the ASoC driver is updated and the modem platform data expanded with a power management callback for switching its power, the ams_delta_latch_write() function, which still provides the old API for accessing latch2 functionality from not updated drivers, is modified to toggle the regulator instead of the MODEM_NRESET GPIO pin. A helper function provided for balancing the regulator enable/disable operations, together with the consumer data needed for tracking the regulator state, will be removed once the drivers are updated. Depends on patch series "ARM: OMAP1: ams-delta: replace custom I/O with GPIO". Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- Changes against initial version: * rename consumer setup elements to match their final, modem only related purpose, * initialize the regulator pointer and mutex before first use, then omit testing that pointer against NULL value. arch/arm/mach-omap1/Kconfig | 2 + arch/arm/mach-omap1/board-ams-delta.c | 104 +++++++++++++++++++++++++++++++-- 2 files changed, 100 insertions(+), 6 deletions(-) diff --git a/arch/arm/mach-omap1/Kconfig b/arch/arm/mach-omap1/Kconfig index 5b1edba..4b6a774 100644 --- a/arch/arm/mach-omap1/Kconfig +++ b/arch/arm/mach-omap1/Kconfig @@ -157,6 +157,8 @@ config MACH_AMS_DELTA select FIQ select GPIO_GENERIC_PLATFORM select LEDS_GPIO_REGISTER + select REGULATOR + select REGULATOR_FIXED_VOLTAGE help Support for the Amstrad E3 (codename Delta) videophone. Say Y here if you have such a device. diff --git a/arch/arm/mach-omap1/board-ams-delta.c b/arch/arm/mach-omap1/board-ams-delta.c index 673cf21..074d3b5 100644 --- a/arch/arm/mach-omap1/board-ams-delta.c +++ b/arch/arm/mach-omap1/board-ams-delta.c @@ -18,7 +18,11 @@ #include <linux/input.h> #include <linux/interrupt.h> #include <linux/leds.h> +#include <linux/mutex.h> #include <linux/platform_device.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/fixed.h> +#include <linux/regulator/machine.h> #include <linux/serial_8250.h> #include <linux/export.h> @@ -237,11 +241,6 @@ static struct gpio latch_gpios[] __initconst = { .label = "scard_cmdvcc", }, { - .gpio = AMS_DELTA_GPIO_PIN_MODEM_NRESET, - .flags = GPIOF_OUT_INIT_LOW, - .label = "modem_nreset", - }, - { .gpio = AMS_DELTA_GPIO_PIN_MODEM_CODEC, .flags = GPIOF_OUT_INIT_LOW, .label = "modem_codec", @@ -258,6 +257,71 @@ static struct gpio latch_gpios[] __initconst = { }, }; +static struct regulator_consumer_supply modem_nreset_consumers[] __initconst = { + REGULATOR_SUPPLY("RESET#", "serial8250.1"), + REGULATOR_SUPPLY("POR", "cx20442-codec"), +}; + +static struct regulator_init_data modem_nreset_data __initconst = { + .constraints = { + .valid_ops_mask = REGULATOR_CHANGE_STATUS, + .boot_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(modem_nreset_consumers), + .consumer_supplies = modem_nreset_consumers, +}; + +static struct fixed_voltage_config modem_nreset_config __initconst = { + .supply_name = "modem_nreset", + .microvolts = 3300000, + .gpio = AMS_DELTA_GPIO_PIN_MODEM_NRESET, + .startup_delay = 25000, + .enable_high = 1, + .enabled_at_boot = 1, + .init_data = &modem_nreset_data, +}; + +static struct platform_device modem_nreset_device = { + .name = "reg-fixed-voltage", + .id = -1, + .dev = { + .platform_data = &modem_nreset_config, + }, +}; + +struct modem_private_data { + struct regulator *regulator; + struct { + struct mutex lock; + bool enabled; + } consumer; +}; + +static int regulator_toggle(struct modem_private_data *priv, bool enable) +{ + int err = 0; + + mutex_lock(&priv->consumer.lock); + if (IS_ERR(priv->regulator)) { + err = PTR_ERR(priv->regulator); + } else if (enable) { + if (!priv->consumer.enabled) { + err = regulator_enable(priv->regulator); + priv->consumer.enabled = true; + } + } else { + if (priv->consumer.enabled) { + err = regulator_disable(priv->regulator); + priv->consumer.enabled = false; + } + } + mutex_unlock(&priv->consumer.lock); + + return err; +} + +static struct modem_private_data modem_priv; + void ams_delta_latch_write(int base, int ngpio, u16 mask, u16 value) { int bit = 0; @@ -266,7 +330,10 @@ void ams_delta_latch_write(int base, int ngpio, u16 mask, u16 value) for (; bit < ngpio; bit++, bitpos = bitpos << 1) { if (!(mask & bitpos)) continue; - gpio_set_value(base + bit, (value & bitpos) != 0); + else if (base + bit == AMS_DELTA_GPIO_PIN_MODEM_NRESET) + regulator_toggle(&modem_priv, (value & bitpos) != 0); + else + gpio_set_value(base + bit, (value & bitpos) != 0); } } EXPORT_SYMBOL(ams_delta_latch_write); @@ -496,6 +563,12 @@ static int __init late_init(void) platform_add_devices(late_devices, ARRAY_SIZE(late_devices)); + err = platform_device_register(&modem_nreset_device); + if (err) { + pr_err("Couldn't register the modem regulator device\n"); + return err; + } + omap_cfg_reg(M14_1510_GPIO2); ams_delta_modem_ports[0].irq = gpio_to_irq(AMS_DELTA_GPIO_PIN_MODEM_IRQ); @@ -507,6 +580,10 @@ static int __init late_init(void) } gpio_direction_input(AMS_DELTA_GPIO_PIN_MODEM_IRQ); + /* Initialize the modem_nreset regulator consumer before use */ + mutex_init(&modem_priv.consumer.lock); + modem_priv.regulator = ERR_PTR(-ENODEV); + ams_delta_latch2_write( AMS_DELTA_LATCH2_MODEM_NRESET | AMS_DELTA_LATCH2_MODEM_CODEC, AMS_DELTA_LATCH2_MODEM_NRESET | AMS_DELTA_LATCH2_MODEM_CODEC); @@ -514,8 +591,23 @@ static int __init late_init(void) err = platform_device_register(&ams_delta_modem_device); if (err) goto gpio_free; + + /* + * Once the modem device is registered, the modem_nreset + * regulator can be requested on behalf of that device. + * The regulator is used via ams_delta_latch_write() + * by the modem and ASoC drivers until updated. + */ + modem_priv.regulator = regulator_get(&ams_delta_modem_device.dev, + "RESET#"); + if (IS_ERR(modem_priv.regulator)) { + err = PTR_ERR(modem_priv.regulator); + goto unregister; + } return 0; +unregister: + platform_device_unregister(&ams_delta_modem_device); gpio_free: gpio_free(AMS_DELTA_GPIO_PIN_MODEM_IRQ); return err; -- 1.7.3.4

1 0

[alsa-devel] [PATCH v2 0/4] Amstrad Delta: access MODEM_RESET GPIO pin over a regulator
by Janusz Krzysztofik 30 Dec '11

30 Dec '11

The Amstrad Delta on-board latch2 bit named MODEM_NRESET, now available as a GPIO pin AMS_DELTA_GPIO_PIN_NMODEM_RESET, is used to power up/down (bring into/out of a reset state) two distinct on-board devices simultaneously: the modem, and the voice codec. As a consequence, that bit is, or can be, manipulated concurrently by two drivers, or their platform provided hooks. Instead of updating those drivers to use the gpiolib API as a new method of controlling the MODEM_NRESET pin state, like it was done to other drivers accessing latch2 pins, and still being vulnerable to potential concurrency conflicts, or trying to solve that sharing issue with a custom piece of code, set up a fixed regulator device on top of that GPIO pin and update both drivers to manipulate that regulator, not the GPIO pin directly. Changes against initial version: * in both the codec and the modem callbacks, don't track the regulator enable/disable state, compare new targert bias level (the codec case) or power state (the modem case) with the old value instead; thanks to Mark Brown who suggested this solution, * a few other minor changes, mostly stylistic. Janusz Krzysztofik (4): ARM: OMAP1: ams-delta: set up regulator over modem reset GPIO pin ASoC: cx20442: add bias control over a platform provided regulator ARM: OMAP1: ams-delta: update the modem to use regulator API ASoC: OMAP: ams-delta: drop .set_bias_level callback arch/arm/mach-omap1/Kconfig | 2 + arch/arm/mach-omap1/board-ams-delta.c | 94 +++++++++++++++++++-- arch/arm/plat-omap/include/plat/board-ams-delta.h | 1 - sound/soc/codecs/cx20442.c | 48 ++++++++++- sound/soc/omap/ams-delta.c | 34 -------- 5 files changed, 133 insertions(+), 46 deletions(-) -- 1.7.3.4

1 0

[alsa-devel] [PATCH 1/3] ASoC: pxa: Convert poodle to use snd_soc_register_card()
by Axel Lin 30 Dec '11

30 Dec '11

Use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- arch/arm/mach-pxa/poodle.c | 6 ++++++ sound/soc/pxa/poodle.c | 42 +++++++++++++++++++++++------------------- 2 files changed, 29 insertions(+), 19 deletions(-) diff --git a/arch/arm/mach-pxa/poodle.c b/arch/arm/mach-pxa/poodle.c index 69036e4..744baee 100644 --- a/arch/arm/mach-pxa/poodle.c +++ b/arch/arm/mach-pxa/poodle.c @@ -158,6 +158,11 @@ static struct scoop_pcmcia_config poodle_pcmcia_config = { EXPORT_SYMBOL(poodle_scoop_device); +static struct platform_device poodle_audio_device = { + .name = "poodle-audio", + .id = -1, +}; + /* LoCoMo device */ static struct resource locomo_resources[] = { [0] = { @@ -407,6 +412,7 @@ static struct platform_device sharpsl_rom_device = { static struct platform_device *devices[] __initdata = { &poodle_locomo_device, &poodle_scoop_device, + &poodle_audio_device, &sharpsl_nand_device, &sharpsl_rom_device, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 4c29bc1..c9e24bf 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -281,22 +281,18 @@ static struct snd_soc_dai_link poodle_dai = { }; /* poodle audio machine driver */ -static struct snd_soc_card snd_soc_poodle = { +static struct snd_soc_card poodle = { .name = "Poodle", .dai_link = &poodle_dai, .num_links = 1, .owner = THIS_MODULE, }; -static struct platform_device *poodle_snd_device; - -static int __init poodle_init(void) +static int __devinit poodle_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &poodle; int ret; - if (!machine_is_poodle()) - return -ENODEV; - locomo_gpio_set_dir(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_AMP_ON, 0); /* should we mute HP at startup - burning power ?*/ @@ -305,28 +301,36 @@ static int __init poodle_init(void) locomo_gpio_set_dir(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 0); - poodle_snd_device = platform_device_alloc("soc-audio", -1); - if (!poodle_snd_device) - return -ENOMEM; - - platform_set_drvdata(poodle_snd_device, &snd_soc_poodle); - ret = platform_device_add(poodle_snd_device); + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); if (ret) - platform_device_put(poodle_snd_device); - + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); return ret; } -static void __exit poodle_exit(void) +static int __devexit poodle_remove(struct platform_device *pdev) { - platform_device_unregister(poodle_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_init(poodle_init); -module_exit(poodle_exit); +static struct platform_driver poodle_driver = { + .driver = { + .name = "poodle-audio", + .owner = THIS_MODULE, + }, + .probe = poodle_probe, + .remove = __devexit_p(poodle_remove), +}; + +module_platform_driver(poodle_driver); /* Module information */ MODULE_AUTHOR("Richard Purdie"); MODULE_DESCRIPTION("ALSA SoC Poodle"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:poodle-audio"); -- 1.7.5.4

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Re: [alsa-devel] It looks like snd-soc-rx51 only works as built-in, not as a module
by Mark Brown 29 Dec '11

29 Dec '11

On Thu, Dec 29, 2011 at 11:25:50PM +0200, Felipe Contreras wrote: > On Thu, Dec 29, 2011 at 8:37 PM, Mark Brown > > On Sat, Dec 10, 2011 at 05:59:57PM +0200, Felipe Contreras wrote: > >> It was not working for me, but it seems the problem was related to > >> mdev/udev; snd_soc_tlv320aic3x has to be loaded before snd-soc-rx51. > > That should not be required, the modules should be loadable in any > > order. > Are you sure? I recall having a similar discussion with Rusell King, > and the conclusion is that certain modules are supposed to be loaded > by udev at boot time, and it seems snd_soc_tlv320aic3x is one of them. I'm absolutely positive. ASoC supports loading the modules in any order. udev is supposed to load anything autoloadable at boot time, including I2C devices, but that's orthogonal to the order in which things actually get loaded - udev can randomly reorder things if it feels like it.

1 0

Re: [alsa-devel] It looks like snd-soc-rx51 only works as built-in, not as a module
by Mark Brown 29 Dec '11

29 Dec '11

On Sat, Dec 10, 2011 at 05:59:57PM +0200, Felipe Contreras wrote: > It was not working for me, but it seems the problem was related to > mdev/udev; snd_soc_tlv320aic3x has to be loaded before snd-soc-rx51. That should not be required, the modules should be loadable in any order.

1 0

[alsa-devel] [PATCH] ASoC: Remove unused label from wm8994 probe()
by Mark Brown 29 Dec '11

29 Dec '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a993690..71472b6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3759,7 +3759,7 @@ err_irq: wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR, codec); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, codec); wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, codec); -err: + return ret; } -- 1.7.7.3

1 0

[alsa-devel] [PATCH 1/7] ASoC: Make WM8978 I2C usage unconditional
by Mark Brown 29 Dec '11

29 Dec '11

The driver only supports I2C. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8978.c | 6 ------ 1 files changed, 0 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 85d514d..0b1f7ad 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -1001,7 +1001,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8978 = { .reg_cache_default = wm8978_reg, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm8978_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1043,27 +1042,22 @@ static struct i2c_driver wm8978_i2c_driver = { .remove = __devexit_p(wm8978_i2c_remove), .id_table = wm8978_i2c_id, }; -#endif static int __init wm8978_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&wm8978_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8978 I2C driver: %d\n", ret); } -#endif return ret; } module_init(wm8978_modinit); static void __exit wm8978_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8978_i2c_driver); -#endif } module_exit(wm8978_exit); -- 1.7.7.3

1 6

[alsa-devel] [PATCH 01/10] ASoC: Convert 88pm860x-codec to devm_kzalloc()
by Axel Lin 28 Dec '11

28 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/88pm860x-codec.c | 7 ++----- 1 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 99ca53c..9fd3b68 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1430,7 +1430,8 @@ static int __devinit pm860x_codec_probe(struct platform_device *pdev) struct resource *res; int i, ret; - pm860x = kzalloc(sizeof(struct pm860x_priv), GFP_KERNEL); + pm860x = devm_kzalloc(&pdev->dev, sizeof(struct pm860x_priv), + GFP_KERNEL); if (pm860x == NULL) return -ENOMEM; @@ -1459,17 +1460,13 @@ static int __devinit pm860x_codec_probe(struct platform_device *pdev) out: platform_set_drvdata(pdev, NULL); - kfree(pm860x); return -EINVAL; } static int __devexit pm860x_codec_remove(struct platform_device *pdev) { - struct pm860x_priv *pm860x = platform_get_drvdata(pdev); - snd_soc_unregister_codec(&pdev->dev); platform_set_drvdata(pdev, NULL); - kfree(pm860x); return 0; } -- 1.7.5.4

2 14

[alsa-devel] [PATCH] ASoC: Convert sgtl500 to use devm_kzalloc()
by Fabio Estevam 28 Dec '11

28 Dec '11

Convert sgtl500 codec driver to use devm_kzalloc(). Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 14 +++----------- 1 files changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index c189663..32f491d 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1402,7 +1402,8 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, struct sgtl5000_priv *sgtl5000; int ret; - sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL); + sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), + GFP_KERNEL); if (!sgtl5000) return -ENOMEM; @@ -1410,22 +1411,13 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &sgtl5000_driver, &sgtl5000_dai, 1); - if (ret) { - dev_err(&client->dev, "Failed to register codec: %d\n", ret); - kfree(sgtl5000); - return ret; - } - - return 0; + return ret; } static __devexit int sgtl5000_i2c_remove(struct i2c_client *client) { - struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - kfree(sgtl5000); return 0; } -- 1.7.1

2 2

[alsa-devel] [PATCH] ASoC: mxs: Add .owner to struct snd_soc_card
by Axel Lin 28 Dec '11

28 Dec '11

Add missing .owner of struct snd_soc_card. This prevents the module from being removed from underneath its users. Signed-off-by: Lothar Waßmann <LW(a)KARO-electronics.de> Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- Hi Mark, I found Lothar's patch is not merged now. Not sure if you still have Lothar's patch. If not, you can apply this one. Regards, Axel sound/soc/mxs/mxs-sgtl5000.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 259278f..60f052b 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -105,6 +105,7 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { static struct snd_soc_card mxs_sgtl5000 = { .name = "mxs_sgtl5000", + .owner = THIS_MODULE, .dai_link = mxs_sgtl5000_dai, .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), }; -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Fix voltage units in dev_err message
by Fabio Estevam 28 Dec '11

28 Dec '11

vdda, vddio and vddd are voltages expressed in milivolts (mV), so use the proper annotation. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index bbcf921..c189663 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1077,7 +1077,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec) /* according to datasheet, maximum voltage of supplies */ if (vdda > 3600 || vddio > 3600 || vddd > 1980) { dev_err(codec->dev, - "exceed max voltage vdda %dmv vddio %dma vddd %dma\n", + "exceed max voltage vdda %dmV vddio %dmV vddd %dmV\n", vdda, vddio, vddd); return -EINVAL; -- 1.7.1

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[alsa-devel] ASoC update for 3.2
by Mark Brown 28 Dec '11

28 Dec '11

The following changes since commit 497d496598f411cb7f0690725a0e8b9029cebdb9: ASoC: Fix hx4700 error handling to free gpios if snd_soc_register_card fails (2011-12-12 12:58:42 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-3.2 Joachim Eastwood (1): ASoC: wm8776: add missing break in sample size switch sound/soc/codecs/wm8776.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-)

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[alsa-devel] [PATCH 1/3] ASoC: Use dai_fmt in kirkwood-openrd machine driver
by Axel Lin 28 Dec '11

28 Dec '11

Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/kirkwood/kirkwood-openrd.c | 14 ++------------ 1 files changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 8a5a3dd..55d2ed3 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -26,18 +26,7 @@ static int openrd_client_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int ret; - unsigned int freq, fmt; - - fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS; - ret = snd_soc_dai_set_fmt(cpu_dai, fmt); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_fmt(codec_dai, fmt); - if (ret < 0) - return ret; + unsigned int freq; switch (params_rate(params)) { default: @@ -69,6 +58,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { .platform_name = "kirkwood-pcm-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, .ops = &openrd_client_ops, }, }; -- 1.7.5.4

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[alsa-devel] [PATCH v2] ASoC: Fix build error in sound/soc/kirkwood/kirkwood-i2s.c
by Axel Lin 28 Dec '11

28 Dec '11

Since commit db33f4de "ARM: Orion: Remove address map info from all platform data structures", the dram is removed from struct kirkwood_asoc_platform_data. This patch fixes below build error: CC sound/soc/kirkwood/kirkwood-i2s.o sound/soc/kirkwood/kirkwood-i2s.c: In function 'kirkwood_i2s_dev_probe': sound/soc/kirkwood/kirkwood-i2s.c:444: error: 'struct kirkwood_asoc_platform_data' has no member named 'dram' sound/soc/kirkwood/kirkwood-i2s.c:450: error: 'struct kirkwood_asoc_platform_data' has no member named 'dram' make[3]: *** [sound/soc/kirkwood/kirkwood-i2s.o] Error 1 make[2]: *** [sound/soc/kirkwood] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Andrew Lunn <andrew(a)lunn.ch> Cc: Nicolas Pitre <nico(a)fluxnic.net> --- v2: the dram field of struct kirkwood_dma_data is not being used now. Thus also remove it in this patch. This patch is against linux-next. sound/soc/kirkwood/kirkwood-i2s.c | 3 +-- sound/soc/kirkwood/kirkwood.h | 1 - 2 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index f6bb211..3cb9aa4 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -441,13 +441,12 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) goto err_ioremap; } - if (!data || !data->dram) { + if (!data) { dev_err(&pdev->dev, "no platform data ?!\n"); err = -EINVAL; goto err_ioremap; } - priv->dram = data->dram; priv->burst = data->burst; return snd_soc_register_dai(&pdev->dev, &kirkwood_i2s_dai); diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index bb6e6a5..9047436 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -123,7 +123,6 @@ struct kirkwood_dma_data { void __iomem *io; int irq; int burst; - struct mbus_dram_target_info *dram; }; #endif -- 1.7.5.4

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