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

[alsa-devel] [PATCHv6][ 1/4] ASoC: eukrea-tlv320: Add DT support.
by Denis Carikli 24 Oct '13

24 Oct '13

Cc: Rob Herring <rob.herring(a)calxeda.com> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Sascha Hauer <kernel(a)pengutronix.de> Cc: linux-arm-kernel(a)lists.infradead.org Cc: Eric Bénard <eric(a)eukrea.com> Signed-off-by: Denis Carikli <denis(a)eukrea.com> --- .../devicetree/bindings/sound/eukrea-tlv320.txt | 23 ++++ sound/soc/fsl/Kconfig | 5 +- sound/soc/fsl/eukrea-tlv320.c | 135 ++++++++++++++++---- 3 files changed, 140 insertions(+), 23 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt diff --git a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt new file mode 100644 index 0000000..8791037 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -0,0 +1,23 @@ +Audio complex for Eukrea boards with tlv320aic23 codec. + +Required properties: +- compatible : "eukrea,eukrea-tlv320" +- model : The user-visible name of this sound complex. +- ssi-controller : The phandle of the SSI controller. +- audio-codec : The phandle of the tlv320aic23 audio codec. +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). +- mux-ext-port : The external port of the i.MX audio muxer. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + + sound { + compatible = "eukrea,eukrea-tlv320"; + model = "imx51-eukrea-tlv320aic23"; + ssi-controller = <&ssi2>; + fsl,audio-codec = <&tlv320aic23>; + mux-int-port = <2>; + mux-ext-port = <3>; + }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index b7ab71f..9c3cd64 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -161,12 +161,15 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || OF depends on I2C select SND_SOC_TLV320AIC23 select SND_SOC_IMX_PCM_FIQ select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9a4a0ca..92710cf 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -17,6 +17,8 @@ #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,49 +28,64 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 +struct eukrea_tlv320 { + char of; +}; + static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct eukrea_tlv320 *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int 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) { - pr_err("%s: failed set cpu dai format\n", __func__); - return ret; + /* fsl_ssi lacks the set_fmt ops. */ + if (!priv->of) { + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + dev_err(cpu_dai->dev, + "Failed to set the cpu dai format\n"); + 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) { - pr_err("%s: failed set codec dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec dai format.\n"); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret) { - pr_err("%s: failed setting codec sysclk\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec dai sysclk.\n"); return ret; } - snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); - ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, - SND_SOC_CLOCK_IN); - if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); - return ret; - } + /* fsl_ssi lacks the set_sysclk ops */ + if (!priv->of) { + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(cpu_dai->dev, "Can't set CPU system clock IMX_SSP_SYS_CLK.\n"); + return ret; + } + } return 0; } @@ -80,7 +97,6 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, @@ -97,8 +113,56 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct platform_device *ssi_pdev; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np; + struct eukrea_tlv320 *priv; + + priv = devm_kzalloc(&pdev->dev, + sizeof(struct eukrea_tlv320), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; - if (machine_is_eukrea_cpuimx27()) { + if (np) { + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, + "ssi-controller missing or invalid.\n"); + ret = -ENODEV; + goto err; + } + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + eukrea_tlv320_dai.platform_of_node = ssi_np; + priv->of = 1; + } else { + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + } + + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -115,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -130,14 +198,29 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,eukrea-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } eukrea_tlv320.dev = &pdev->dev; + snd_soc_card_set_drvdata(&eukrea_tlv320, priv); ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -149,13 +232,21 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,eukrea-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, - .remove = eukrea_tlv320_remove,}; + .remove = eukrea_tlv320_remove, +}; module_platform_driver(eukrea_tlv320_driver); -- 1.7.9.5

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[alsa-devel] [GIT PULL] [00/29] playback-only DICE driver
by Clemens Ladisch 24 Oct '13

24 Oct '13

Hi Takashi, please pull this driver for FireWire DACs based on the DICE chip family. While the driver supports only PCM playback at the moment, it fills a gap left by FFADO, which supports only full- duplex devices. Thanks, Clemens ---------------------------------------------------------------- The following changes since commit 6e4664525b1db28f8c4e1130957f70a94c19213e: Linux 3.11 (2013-09-02 13:46:10 -0700) are available in the git repository at: git://git.alsa-project.org/alsa-kprivate.git dice-driver-playback-only for you to fetch changes up to b20be8de1b3972ccf9af72850b045214faa8d830: ALSA: dice: restrict the driver to playback-only devices (2013-10-20 22:07:57 +0200) ---------------------------------------------------------------- Clemens Ladisch (28): ALSA: add DICE driver ALSA: dice: optimize bus reset handling ALSA: dice: allow all sample rates ALSA: dice: reduce noisy logging ALSA: dice, firewire-lib: add blocking mode ALSA: dice: fix hang when unplugging a running device ALSA: dice: implement hwdep device ALSA: dice: clear device lock when closing hwdep device ALSA: firewire: introduce amdtp_out_stream_running() ALSA: dice: reorganize interface definitions ALSA: dice: fix device detection for other vendors ALSA: dice: support dual-wire stream format at 192 kHz ALSA: dice: optimize reading of consecutive registers ALSA: firewire: extend snd_fw_transaction() ALSA: dice: avoid superflous write at bus reset ALSA: dice: remove 10s period length limit ALSA: dice: remove superfluous field ALSA: dice: make amdtp_rates[] const ALSA: dice: get clock capabilities ALSA: dice: allow notifications during initialization ALSA: dice: get rate-dependent parameters ALSA: dice: dynamic sample rate selection ALSA: dice: check clock change timeout ALSA: dice: add a proc file to show device information ALSA: dice: document quadlet alignment ALSA: dice: dice_proc_read: remove wrong typecast ALSA: dice: fix detection of Weiss devices ALSA: dice: restrict the driver to playback-only devices Stefan Richter (1): ALSA: dice: fix locking Documentation/ioctl/ioctl-number.txt | 1 + include/uapi/sound/Kbuild | 1 + include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 51 ++ sound/firewire/Kconfig | 15 + sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 209 +++-- sound/firewire/amdtp.h | 46 +- sound/firewire/cmp.c | 50 +- sound/firewire/dice-interface.h | 371 +++++++++ sound/firewire/dice.c | 1494 ++++++++++++++++++++++++++++++++++ sound/firewire/fcp.c | 2 +- sound/firewire/isight.c | 43 +- sound/firewire/lib.c | 24 +- sound/firewire/lib.h | 7 +- sound/firewire/scs1x.c | 8 +- sound/firewire/speakers.c | 16 +- 17 files changed, 2192 insertions(+), 151 deletions(-) create mode 100644 include/uapi/sound/firewire.h create mode 100644 sound/firewire/dice-interface.h create mode 100644 sound/firewire/dice.c

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[alsa-devel] [PATCH 1/2] ALSA: hda - Add support of ALC255 codecs
by Takashi Iwai 24 Oct '13

24 Oct '13

From: Kailang Yang <kailang(a)realtek.com> It's just another variant of ALC269 & co. Signed-off-by: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52c26d3..1c4f8c5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2540,6 +2540,7 @@ enum { ALC269_TYPE_ALC283, ALC269_TYPE_ALC284, ALC269_TYPE_ALC286, + ALC269_TYPE_ALC255, }; /* @@ -2565,6 +2566,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC282: case ALC269_TYPE_ALC283: case ALC269_TYPE_ALC286: + case ALC269_TYPE_ALC255: ssids = alc269_ssids; break; default: @@ -4102,6 +4104,9 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0286: spec->codec_variant = ALC269_TYPE_ALC286; break; + case 0x10ec0255: + spec->codec_variant = ALC269_TYPE_ALC255; + break; } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { @@ -4814,6 +4819,7 @@ static int patch_alc680(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, + { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, -- 1.8.4

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[alsa-devel] [PATCH] ALSA: hda - Fix mute LED on HP laptops in runtime suspend
by Takashi Iwai 24 Oct '13

24 Oct '13

When HP laptops with mute and mic-record LEDs go to runtime suspend, these LEDs are turned on forcibly no matter whether GPIO pis are on or off. This strange behavior seems triggered by resetting the HD-audio bus link at azx_rutime_suspend(). So, just add a new hda_bus flag to avoid the link reset at runtime suspend and set it for these HP machines. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_intel.c | 3 ++- sound/pci/hda/patch_sigmatel.c | 4 +++- 3 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 7aa9870..77db694 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -698,6 +698,7 @@ struct hda_bus { unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ + unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */ int primary_dig_out_type; /* primary digital out PCM type */ }; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6e61a01..a0a06f7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2986,7 +2986,8 @@ static int azx_runtime_suspend(struct device *dev) STATESTS_INT_MASK); azx_stop_chip(chip); - azx_enter_link_reset(chip); + if (!chip->bus->avoid_link_reset) + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index fba0cef..69a549a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2091,8 +2091,10 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, { struct sigmatel_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ + codec->bus->avoid_link_reset = 1; + } } static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, -- 1.8.4

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[alsa-devel] [PATCH 0/8] treewide: Remove OOM message after input_alloc_device
by Joe Perches 24 Oct '13

24 Oct '13

Joe Perches (8): Documentation: Remove OOM message after input_allocate_device cell: Remove OOM message after input_allocate_device hid: Remove OOM message after input_allocate_device input: Remove OOM message after input_allocate_device media: Remove OOM message after input_allocate_device platform:x86: Remove OOM message after input_allocate_device staging: Remove OOM message after input_allocate_device sound: Remove OOM message after input_allocate_device Documentation/input/input-programming.txt | 1 - arch/powerpc/platforms/cell/cbe_powerbutton.c | 1 - drivers/hid/hid-input.c | 1 - drivers/hid/hid-picolcd_core.c | 5 ++--- drivers/input/joystick/as5011.c | 2 -- drivers/input/joystick/db9.c | 1 - drivers/input/joystick/gamecon.c | 4 +--- drivers/input/joystick/turbografx.c | 1 - drivers/input/joystick/walkera0701.c | 1 - drivers/input/keyboard/amikbd.c | 4 +--- drivers/input/keyboard/davinci_keyscan.c | 1 - drivers/input/keyboard/gpio_keys.c | 1 - drivers/input/keyboard/lpc32xx-keys.c | 1 - drivers/input/keyboard/max7359_keypad.c | 1 - drivers/input/keyboard/mcs_touchkey.c | 1 - drivers/input/keyboard/mpr121_touchkey.c | 1 - drivers/input/keyboard/nomadik-ske-keypad.c | 1 - drivers/input/keyboard/opencores-kbd.c | 1 - drivers/input/keyboard/pmic8xxx-keypad.c | 1 - drivers/input/keyboard/pxa27x_keypad.c | 1 - drivers/input/keyboard/pxa930_rotary.c | 1 - drivers/input/keyboard/qt1070.c | 1 - drivers/input/keyboard/qt2160.c | 1 - drivers/input/keyboard/sh_keysc.c | 1 - drivers/input/keyboard/tc3589x-keypad.c | 1 - drivers/input/keyboard/tnetv107x-keypad.c | 1 - drivers/input/keyboard/w90p910_keypad.c | 1 - drivers/input/misc/88pm80x_onkey.c | 1 - drivers/input/misc/88pm860x_onkey.c | 1 - drivers/input/misc/arizona-haptics.c | 4 +--- drivers/input/misc/atlas_btns.c | 4 +--- drivers/input/misc/da9052_onkey.c | 1 - drivers/input/misc/da9055_onkey.c | 4 +--- drivers/input/misc/ideapad_slidebar.c | 1 - drivers/input/misc/ims-pcu.c | 7 +------ drivers/input/misc/kxtj9.c | 4 +--- drivers/input/misc/max8997_haptic.c | 1 - drivers/input/misc/mc13783-pwrbutton.c | 4 +--- drivers/input/misc/mpu3050.c | 1 - drivers/input/misc/pcf8574_keypad.c | 1 - drivers/input/misc/pm8xxx-vibrator.c | 1 - drivers/input/misc/pmic8xxx-pwrkey.c | 1 - drivers/input/misc/pwm-beeper.c | 1 - drivers/input/misc/twl4030-pwrbutton.c | 4 +--- drivers/input/misc/twl6040-vibra.c | 1 - drivers/input/mouse/appletouch.c | 4 +--- drivers/input/mouse/bcm5974.c | 4 +--- drivers/input/mouse/cyapa.c | 4 +--- drivers/input/mouse/inport.c | 1 - drivers/input/mouse/logibm.c | 1 - drivers/input/mouse/pc110pad.c | 1 - drivers/input/mouse/pxa930_trkball.c | 1 - drivers/input/tablet/aiptek.c | 5 +---- drivers/input/tablet/gtco.c | 1 - drivers/input/touchscreen/88pm860x-ts.c | 1 - drivers/input/touchscreen/atmel_mxt_ts.c | 1 - drivers/input/touchscreen/atmel_tsadcc.c | 1 - drivers/input/touchscreen/bu21013_ts.c | 1 - drivers/input/touchscreen/cyttsp4_core.c | 2 -- drivers/input/touchscreen/da9034-ts.c | 1 - drivers/input/touchscreen/edt-ft5x06.c | 1 - drivers/input/touchscreen/eeti_ts.c | 5 +---- drivers/input/touchscreen/htcpen.c | 1 - drivers/input/touchscreen/intel-mid-touch.c | 1 - drivers/input/touchscreen/lpc32xx_ts.c | 1 - drivers/input/touchscreen/mcs5000_ts.c | 1 - drivers/input/touchscreen/migor_ts.c | 1 - drivers/input/touchscreen/mk712.c | 1 - drivers/input/touchscreen/pixcir_i2c_ts.c | 1 - drivers/input/touchscreen/s3c2410_ts.c | 1 - drivers/input/touchscreen/ti_am335x_tsc.c | 1 - drivers/input/touchscreen/tnetv107x-ts.c | 1 - drivers/media/rc/imon.c | 8 ++------ drivers/media/usb/em28xx/em28xx-input.c | 4 +--- drivers/media/usb/pwc/pwc-if.c | 1 - drivers/platform/x86/asus-laptop.c | 5 ++--- drivers/platform/x86/eeepc-laptop.c | 4 +--- drivers/platform/x86/ideapad-laptop.c | 4 +--- drivers/platform/x86/intel_mid_powerbtn.c | 4 +--- drivers/platform/x86/panasonic-laptop.c | 5 +---- drivers/platform/x86/thinkpad_acpi.c | 1 - drivers/platform/x86/topstar-laptop.c | 4 +--- drivers/platform/x86/toshiba_acpi.c | 4 +--- drivers/staging/cptm1217/clearpad_tm1217.c | 2 -- drivers/staging/iio/adc/mxs-lradc.c | 4 +--- drivers/staging/ste_rmi4/synaptics_i2c_rmi4.c | 2 -- sound/pci/hda/hda_beep.c | 4 +--- 87 files changed, 29 insertions(+), 152 deletions(-) -- 1.8.1.2.459.gbcd45b4.dirty

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Re: [alsa-devel] [PATCH] ASoC: tpa6130a2: Add device tree support
by Mark Brown 24 Oct '13

24 Oct '13

On Wed, Oct 23, 2013 at 02:03:28PM +0200, Sebastian Reichel wrote: > Add device tree support to tpa6130a2 driver and document the > bindings. Applied, thanks.

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[alsa-devel] [PATCH] ALSA: hda - Fix unbalanced runtime PM refcount after S3/S4
by Takashi Iwai 24 Oct '13

24 Oct '13

When a machine goes to S3/S4 after power-save is enabled, the runtime PM refcount might be incorrectly decreased because the power-down triggered soon after resume assumes that the controller was already powered up, and issues the pm_notify down. This patch fixes the incorrect pm_notify call simply by checking the current value properly. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5b6c4e3..748c6a9 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4864,8 +4864,8 @@ static void hda_power_work(struct work_struct *work) spin_unlock(&codec->power_lock); state = hda_call_codec_suspend(codec, true); - codec->pm_down_notified = 0; - if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { + if (!codec->pm_down_notified && + !bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { codec->pm_down_notified = 1; hda_call_pm_notify(bus, false); } -- 1.8.4

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[alsa-devel] [PATCH v2 00/11] Fix AM335x-evm analog audio support
by jsarha＠ti.com 23 Oct '13

23 Oct '13

From: Jyri Sarha <jsarha(a)ti.com> The first version of patches can been found here: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-September/066178.… Changes for v2 - Dropped our "ASoC: davinci-mcasp: Add pinctrl support" since driver core is taking care of this now. - Cleanup am33xx audio build - Add regulators to tlv320aic3x DT binding document - Remove dm365-voice-codec-audio DT support as it has never been tested an probably does not work - Add output pins and Line In connector to davinci-evm-audio DT binding doc - Remove asp_chan_q and ram_chan_q properties from mcasp DT node in DT mode mcasp is hardcoded to event queue 0 (highest priority) - Add pins to tlv320aic3x DT bindings document. If I misunderstood Marks comment and this patch is not needed, then just leave it out Changes based on TI internal discussions - Move system clock rate logic away from from evm_hw_params soc-op - Remove unnecesary #if defined(CONFIG_OF) from davinci-evm.c - Make dma property DT binding document more exact - Add only "dma" reg location instead of separate "dma-tx" and "dma-rx" - Primarily look for "mpu" reg property, but fall back to index 0 if not found - Remove interrupt property from mcasp DT node as it is not used - Remove #address-cells and #size-cells mcasp properties as they are not needed This set of patches fixes the basic audio support for am335x-evm. It should also be relatively simple to add the necessary nodes to relevant dts files to get BeagleBone + AudioCape and am335x-evmsk working too. The patch set depends on following patches: [PATCH v11 4/8] ARM: dts: add AM33XX EDMA support https://lkml.org/lkml/2013/6/18/49 [PATCH v11 5/8] ARM: dts: add AM33XX SPI DMA support https://lkml.org/lkml/2013/6/18/55 [PATCH v2] ARM: EDMA: Fix clearing of unused list for DT DMA resources https://lkml.org/lkml/2013/7/22/441 I have tried my best not to break the existing support for older davinci boards, but since I do not have those boards I can not be sure. Some commit comments refer to a dmaengine based davinci audio implementation which is planned for but nothing has been done yet. Best regards, Jyri Darren Etheridge (1): ARM/dts: am335x-evm: Add audio support for am335x-evm.dts Hebbar, Gururaja (2): ASoC: davinci-evm: Add device tree binding ASoC: davinci: Add support for AM33xx SoC Audio Jyri Sarha (7): ASoC: davinci-evm: Move sysclk logic away from evm_hw_params ASoC: davinci-mcasp: Add DMA register locations to DT ASoC: davinci-mcasp: Extract DMA channels directly from DT ASoC: davinci-mcasp: Remove interrupt property from DT bindings doc ASoC: tlv320aic3x: Add regulators to DT bindings document ASoC: tlv320aic3x: Add codec pins to DT bindings document ARM/dts: am33xx: mcasp: Add new dma register location to reg-property Pantelis Antoniou (1): ARM/dts: am33xx: Add mcasp0 and mcasp1 device tree entries .../bindings/sound/davinci-evm-audio.txt | 58 ++++++ .../bindings/sound/davinci-mcasp-audio.txt | 15 +- .../devicetree/bindings/sound/tlv320aic3x.txt | 26 +++ arch/arm/boot/dts/am335x-evm.dts | 56 ++++++ arch/arm/boot/dts/am33xx.dtsi | 25 +++ include/linux/platform_data/davinci_asp.h | 2 + sound/soc/davinci/Kconfig | 18 +- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 184 +++++++++++++++++--- sound/soc/davinci/davinci-mcasp.c | 106 +++++++---- 10 files changed, 428 insertions(+), 63 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

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[alsa-devel] [PATCH 1/3] ASoC: samsung: Provide helper for DMA init
by Mark Brown 23 Oct '13

23 Oct '13

From: Mark Brown <broonie(a)linaro.org> In preparation for using the dmaengine helpers in ASoC rather than the dmaengine wrappers for the Samsung API wrap the configuration of dma_data. The dmaengine code expects different data to that used by the legacy API. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/samsung/ac97.c | 51 +++++++++++++++--------------------------------- sound/soc/samsung/dma.c | 8 ++++++++ sound/soc/samsung/dma.h | 3 +++ sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/pcm.c | 18 +++++++++-------- 5 files changed, 38 insertions(+), 44 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 2acf987..bd09410 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -221,24 +221,6 @@ static struct snd_ac97_bus_ops s3c_ac97_ops = { .reset = s3c_ac97_cold_reset, }; -static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct s3c_dma_params *dma_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c_ac97_pcm_out; - else - dma_data = &s3c_ac97_pcm_in; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - - return 0; -} - static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -279,21 +261,6 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return -ENODEV; - else - snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); - - return 0; -} - static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -329,15 +296,27 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops s3c_ac97_dai_ops = { - .hw_params = s3c_ac97_hw_params, .trigger = s3c_ac97_trigger, }; static const struct snd_soc_dai_ops s3c_ac97_mic_dai_ops = { - .hw_params = s3c_ac97_hw_mic_params, .trigger = s3c_ac97_mic_trigger, }; +static int s3c_ac97_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, &s3c_ac97_pcm_out, &s3c_ac97_pcm_in); + + return 0; +} + +static int s3c_ac97_mic_dai_probe(struct snd_soc_dai *dai) +{ + samsung_asoc_init_dma_data(dai, NULL, &s3c_ac97_mic_in); + + return 0; +} + static struct snd_soc_dai_driver s3c_ac97_dai[] = { [S3C_AC97_DAI_PCM] = { .name = "samsung-ac97", @@ -354,6 +333,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_dai_probe, .ops = &s3c_ac97_dai_ops, }, [S3C_AC97_DAI_MIC] = { @@ -365,6 +345,7 @@ static struct snd_soc_dai_driver s3c_ac97_dai[] = { .channels_max = 1, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .probe = s3c_ac97_mic_dai_probe, .ops = &s3c_ac97_mic_dai_ops, }, }; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index fe2748b..ee23194 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -441,6 +441,14 @@ static struct snd_soc_platform_driver samsung_asoc_platform = { .pcm_free = dma_free_dma_buffers, }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture) +{ + snd_soc_dai_init_dma_data(dai, playback, capture); +} +EXPORT_SYMBOL_GPL(samsung_asoc_init_dma_data); + int samsung_asoc_dma_platform_register(struct device *dev) { return snd_soc_register_platform(dev, &samsung_asoc_platform); diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 0e86315..fb09a1c 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -22,6 +22,9 @@ struct s3c_dma_params { char *ch_name; }; +void samsung_asoc_init_dma_data(struct snd_soc_dai *dai, + struct s3c_dma_params *playback, + struct s3c_dma_params *capture); int samsung_asoc_dma_platform_register(struct device *dev); void samsung_asoc_dma_platform_unregister(struct device *dev); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2c4d250..9cd0a2a 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -963,7 +963,7 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); - snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + samsung_asoc_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); if (other) { other->addr = i2s->addr; diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index e54256f..6a5e4bf 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -275,7 +275,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(rtd->cpu_dai); - struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -284,13 +283,6 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = pcm->dma_playback; - else - dma_data = pcm->dma_capture; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); - /* Strictly check for sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -461,10 +453,20 @@ static const struct snd_soc_dai_ops s3c_pcm_dai_ops = { .set_fmt = s3c_pcm_set_fmt, }; +static int s3c_pcm_dai_probe(struct snd_soc_dai *dai) +{ + struct s3c_pcm_info *pcm = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, pcm->dma_playback, pcm->dma_capture); + + return 0; +} + #define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 #define S3C_PCM_DAI_DECLARE \ .symmetric_rates = 1, \ + .probe = s3c_pcm_dai_probe, \ .ops = &s3c_pcm_dai_ops, \ .playback = { \ .channels_min = 2, \ -- 1.8.4.rc3

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[alsa-devel] [PATCH RESEND] ASoC: dont call dapm_sync while reporting jack always
by Vinod Koul 23 Oct '13

23 Oct '13

While reporting the jack status snd_soc_jack_report() invokes snd_soc_dapm_sync() always. This should be required when we have pins associated with jack and reporting enables or disables these. So add a check for this case Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Sorry previous one had some non upstream cruft sound/soc/soc-jack.c | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index c9f7953..9fd4ed8 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -65,6 +65,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) struct snd_soc_codec *codec; struct snd_soc_dapm_context *dapm; struct snd_soc_jack_pin *pin; + unsigned int sync = 0; int enable; trace_snd_soc_jack_report(jack, mask, status); @@ -92,12 +93,16 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) snd_soc_dapm_enable_pin(dapm, pin->pin); else snd_soc_dapm_disable_pin(dapm, pin->pin); + + /* we need to sync for this case only */ + sync = 1; } /* Report before the DAPM sync to help users updating micbias status */ blocking_notifier_call_chain(&jack->notifier, jack->status, jack); - snd_soc_dapm_sync(dapm); + if (sync) + snd_soc_dapm_sync(dapm); snd_jack_report(jack->jack, jack->status); -- 1.7.0.4

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