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

[alsa-devel] [PATCH] ASoC: core: Remove dependencies on GENERIC_HARDIRQS
by Mark Brown 06 Sep '13

06 Sep '13

From: Mark Brown <broonie(a)linaro.org> All architectures now use the generic IRQ subsystem so this is redundant. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/samsung/Kconfig | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 15106c0..b33b45d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -107,7 +107,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8782 select SND_SOC_WM8804 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C - select SND_SOC_WM8903 if I2C && GENERIC_HARDIRQS + select SND_SOC_WM8903 if I2C select SND_SOC_WM8904 if I2C select SND_SOC_WM8940 if I2C select SND_SOC_WM8955 if I2C diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1b0cccd..d446c6b 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -63,7 +63,7 @@ config SND_SOC_SAMSUNG_SMDK_WM8580 config SND_SOC_SAMSUNG_SMDK_WM8994 tristate "SoC I2S Audio support for WM8994 on SMDK" depends on SND_SOC_SAMSUNG - depends on I2C=y && GENERIC_HARDIRQS + depends on I2C=y select MFD_WM8994 select SND_SOC_WM8994 select SND_SAMSUNG_I2S -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: kirkwood: change the compatible string of the kirkwood-i2s driver
by Thomas Petazzoni 06 Sep '13

06 Sep '13

The compatible string of the kirkwood-i2s driver was chosen as "marvell,mvebu-audio". Using such a compatible string is not a good idea, since "mvebu" is the name of a large family of SOCs, in which new, unknown SOCs will be coming in the future. It is therefore impossible to know what will be evolutions of this hardware block in the next generations of the SOCs. For this reason, the recommandation for compatible strings of on-SOCs devices has always been to use the name of the oldest SOC that has the hardware block. New SOCs that have an exactly compatible hardware block can reference it using the same compatible string. See [1], [2] and [3] for various cases were this suggestion was made, including from Rob Herring, a Device Tree binding maintainer. As an example, there are already small differences between current generations: * On Kirkwood, only one interrupt is used for audio. * On Dove, two interrupts are used, one for audio data and one for error reporting. In the near future, I'll be adding audio support to Armada 370, which allows has the same hardware block (but maybe with minor variants). Therefore, this patch changes the driver to accept "marvell,kirkwood-audio" and "marvell,dove-audio" as compatible strings instead of the too-generic "marvell,mvebu-audio". The reason for the two different compatible strings is the difference in the number of interrupts used by the two SOCs for audio. This Device Tree binding has never been part of a Linux kernel stable release so far, so it can be changed now without breaking backward compatibility. [1] http://lists.infradead.org/pipermail/linux-mtd/2012-March/040417.html [2] http://lists.infradead.org/pipermail/linux-arm-kernel/2013-April/161065.html [3] http://lists.infradead.org/pipermail/linux-arm-kernel/2012-March/087702.html Signed-off-by: Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> --- Documentation/devicetree/bindings/sound/mvebu-audio.txt | 12 ++++++++---- sound/soc/kirkwood/kirkwood-i2s.c | 3 ++- 2 files changed, 10 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt index 7e5fd37..f0062c5 100644 --- a/Documentation/devicetree/bindings/sound/mvebu-audio.txt +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -2,13 +2,17 @@ Required properties: -- compatible: "marvell,mvebu-audio" +- compatible: + "marvell,kirkwood-audio" for Kirkwood platforms + "marvell,dove-audio" for Dove platforms - reg: physical base address of the controller and length of memory mapped region. -- interrupts: list of two irq numbers. - The first irq is used for data flow and the second one is used for errors. +- interrupts: + with "marvell,kirkwood-audio", the audio interrupt + with "marvell,dove-audio", a list of two interrupts, the first for + the data flow, and the second for errors. - clocks: one or two phandles. The first one is mandatory and defines the internal clock. @@ -21,7 +25,7 @@ Required properties: Example: i2s1: audio-controller@b4000 { - compatible = "marvell,mvebu-audio"; + compatible = "marvell,dove-audio"; reg = <0xb4000 0x2210>; interrupts = <21>, <22>; clocks = <&gate_clk 13>; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 7fce340..0f3d73d 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -559,7 +559,8 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) #ifdef CONFIG_OF static struct of_device_id mvebu_audio_of_match[] = { - { .compatible = "marvell,mvebu-audio" }, + { .compatible = "marvell,kirkwood-audio" }, + { .compatible = "marvell,dove-audio" }, { } }; MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); -- 1.8.1.2

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Re: [alsa-devel] [PATCH 4/5 v2] ARM: shmobile: bockw: add R-Car sound support (PIO)
by Simon Horman 06 Sep '13

06 Sep '13

[ CC Mark Brown, alsa-devel ] On Thu, Sep 05, 2013 at 02:57:51PM +0900, Simon Horman wrote: > On Mon, Aug 26, 2013 at 01:52:35AM -0700, Kuninori Morimoto wrote: > > This patch enables R-Car sound, > > AK4643 (CN19) and AK4554 (CN20/CN21) codec chip > > on Bock-W. > > > > But, it supports PIO transfer only at this point. > > User can check sound settings (Dip-switch/PFC etc) > > via this patch, but will get under/over flow error > > when playback/capture. > > Because PIO transfer via SSI will be interrupted > > "sampling rate" times per 1 second. > > > > DMA transfer will be supported when HPB-DMAC was > > enabled on r8a7778. > > > > You will notice strange ALSA sound card HW > > numbering on Bock-W board. > > This came from AK4554 strange format on playback/capture. > > The format on playback/capture is same on "normal" codec chip, > > but AK4554 was different. > > Because of that, AK4554 playback/capture are > > registered as a different sound card. > > > > Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > > --- > > v1 -> v2 > > Thanks, I have queued this up for v3.13. > > There was a bit of diff-noise, please make sure I have applied it > correctly. The patch I applied is below: Unfortunately I didn't read your cover email carefully before queuing this up. I now see that it has dependencies that aren't present in v3.11-rc and thus I have temporarily dropped it. It seems to me that the dependencies could be provided by the topic/car branch of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git. Morimoto-san, is this correct? If so, Mark, would it be possible for you to provide topic/car or similar as stable branch for me to use as a base? My intention is to queue these up for v3.13.

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[alsa-devel] Roland Sound Canvas SD-50
by Keith A. Milner 05 Sep '13

05 Sep '13

I saw a recent discussion on Alsa-users about the Roland SD-50 which seemed to tail off inconclusively. Does anyone know if this was found to work or not? I ask because I have the option of picking one up (relatively) cheaply. It looks like an interesting little box. But if there isn't a working ALSA solution yet, I might be tempted to pick it up anyway and try to get it working, although I may need some help with that. Cheers, Keith

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[alsa-devel] Documentation Request
by Carl Canuck 05 Sep '13

05 Sep '13

Hello, I've recently begun the work of porting a Windows application to LInux/ALSA. I think ALSA and it's drivers are quite good, but the documentation and example code are terribly lacking, especially for one of the most important use-cases of all: the full on digital audio workstation. I humbly request that a full_duplex_audio_with_midi.c file be added to the example code with the following parameters: 1. Full duplex audio, just simply memcpy'ing the input to the output would suffice 2. Accepts MIDI input; it doesn't have to do anything useful with the data, just printing the event parameters to stdout would suffice. 3. At run time, the application lists all available audio devices and allows the user to select the device, then make settings such as buffer size, sample rate, bit depth and whether to use mmap based on the capabilities of the chosen device 4. At run time, allows optionally selecting a MIDI device. 5. Copious code comments that makes no assumption that the person reading knows anything about the ALSA API or it's constructs, invocations and methodologies. This could either be an interactive CLI application written in C, or even a full GUI application using Qt or GTK. This could then be used as a shell for writing more complex applications, instead of each developer having to reinvent the wheel by trying to figure out how to combine about 4 or so of the other example applications into a single app. /Carl

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[alsa-devel] Constraints and hw_params with highly configurable HW
by Uli Franke 05 Sep '13

05 Sep '13

Hi everybody We're currently working on some neat enhancements for the still experimental snd_dice driver originally initiated by Clemens Ladisch. To list some of the new and already working features: * firmware loading, backed by a user-space command line utility * generic ALSA control support for mostly all device specific settings for all our (i.e. Weiss Engineering) DICE based products and last but not least * capture support * slave mode (external clock source support). I'm currently debugging and fine-tuning things and that's where some questions arose. The initial implementation of Clemens tried to fetch all possible configurations of a device during probe in order to have all combinations of rates and channels at hand when it comes to formulate the constraints set within the ALSA pcm substream open callback. That's a nice approach but will fail with a lot of devices as their configurations can (and mostly will) be volatile. Our devices for instance feature several switches, options and alike which have heavy influence on the channel layout and sample rates and combination of both. I consulted Takashi's documentation on constraints which suggests to "use the source" but I couldn't find any similar situation within other drivers (in case I should have missed something, just let me know of course). Therefore I had the idea to define some maximum constraints and let the ALSA hw_params callback fail in case some settings or combinations are determined to be not supported during its invocation. Is this viable? Could this be/is this consistent with the specs? If yes, would EINVAL an appropriate error code to return? Regards Uli

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[alsa-devel] Recall: [PATCH] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell
by Lin, Mengdong 05 Sep '13

05 Sep '13

Lin, Mengdong would like to recall the message, "[PATCH] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell".

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[alsa-devel] [PATCH] ALSA: hda - unmute pin amplifier in infoframe setup for Haswell
by mengdong.lin＠intel.com 05 Sep '13

05 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> When Gfx driver reconnects a port and transcoder, the pin amplifier will be muted. To enable sound, the pin amp need to be unmuted. This patch - moves pin amp unmuting from stream preparing to hdmi_setup_audio_infoframe(). So if port:transcoder reconnection happens during stream playback, the ELDV unsol event can stil trigger pin's amp unmuting when re-setting up audio info frame. - remove reading pin amp status before unmuting for speed-up, since pin amp should always be unmuted. - rename haswell_verify_pin_D0() to haswell_verify_pin_cvt_D0(), since the convertor power state is also fixed here. - define is_haswell() to replace checking Haswell vendor ID everywhere. This patch is mostly based on suggestion of David Henningsson. Cc: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9a58893..471c158 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -44,6 +44,8 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +#define is_haswell(codec) ((codec)->vendor_id == 0x80862807) + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -894,6 +896,11 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; + if (is_haswell(codec)) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + eld = &per_pin->sink_eld; if (!eld->monitor_present) return; @@ -1033,10 +1040,10 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, +static void haswell_verify_pin_cvt_D0(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t nid) { - int pwr, lamp, ramp; + int pwr; /* For Haswell, the converter 1/2 may keep in D3 state after bootup, * thus pins could only choose converter 0 for use. Make sure the @@ -1052,25 +1059,6 @@ static void haswell_verify_pin_D0(struct hda_codec *codec, pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; snd_printd("Haswell HDMI audio: Power for pin 0x%x is now D%d\n", nid, pwr); } - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - if (lamp != ramp) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT | lamp); - - lamp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_LEFT | AC_AMP_GET_OUTPUT); - ramp = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_RIGHT | AC_AMP_GET_OUTPUT); - snd_printd("Haswell HDMI audio: Mute after set on pin 0x%x: [0x%x 0x%x]\n", nid, lamp, ramp); - } } /* @@ -1087,8 +1075,8 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int pinctl; int new_pinctl = 0; - if (codec->vendor_id == 0x80862807) - haswell_verify_pin_D0(codec, cvt_nid, pin_nid); + if (is_haswell(codec)) + haswell_verify_pin_cvt_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, @@ -1227,7 +1215,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_config_cvts(codec, pin_idx, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1358,14 +1346,10 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) /* Haswell-specific workaround: re-setup when the transcoder is * changed during the stream playback */ - if (codec->vendor_id == 0x80862807 && - eld->eld_valid && !old_eld_valid && per_pin->setup) { - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_OUT_UNMUTE); + if (is_haswell(codec) && + eld->eld_valid && !old_eld_valid && per_pin->setup) hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); - } } mutex_unlock(&pin_eld->lock); @@ -1405,7 +1389,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) return 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) intel_haswell_fixup_connect_list(codec, pin_nid); pin_idx = spec->num_pins; @@ -2014,7 +1998,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); } @@ -2025,7 +2009,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { codec->patch_ops.set_power_state = haswell_set_power_state; codec->dp_mst = true; } -- 1.8.1.2

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[alsa-devel] [PATCH v2 1/2] ALSA: hda - define is_haswell() to check if a display audio codec is Haswell
by mengdong.lin＠intel.com 04 Sep '13

04 Sep '13

From: Mengdong Lin <mengdong.lin(a)intel.com> To apply Haswell specific fixings, this patch defines is_haswell() to check whether a display audio codec is Haswell, to avoid explicitly checking Haswell vendor ID everywhere. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 9a58893..974b55c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -44,6 +44,8 @@ static bool static_hdmi_pcm; module_param(static_hdmi_pcm, bool, 0644); MODULE_PARM_DESC(static_hdmi_pcm, "Don't restrict PCM parameters per ELD info"); +#define is_haswell(codec) ((codec)->vendor_id == 0x80862807) + struct hdmi_spec_per_cvt { hda_nid_t cvt_nid; int assigned; @@ -1087,7 +1089,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int pinctl; int new_pinctl = 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_verify_pin_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { @@ -1227,7 +1229,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, mux_idx); /* configure unused pins to choose other converters */ - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) haswell_config_cvts(codec, pin_idx, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); @@ -1358,7 +1360,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) /* Haswell-specific workaround: re-setup when the transcoder is * changed during the stream playback */ - if (codec->vendor_id == 0x80862807 && + if (is_haswell(codec) && eld->eld_valid && !old_eld_valid && per_pin->setup) { snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1405,7 +1407,7 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) return 0; - if (codec->vendor_id == 0x80862807) + if (is_haswell(codec)) intel_haswell_fixup_connect_list(codec, pin_nid); pin_idx = spec->num_pins; @@ -2014,7 +2016,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) codec->spec = spec; hdmi_array_init(spec, 4); - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { intel_haswell_enable_all_pins(codec, true); intel_haswell_fixup_enable_dp12(codec); } @@ -2025,7 +2027,7 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) { + if (is_haswell(codec)) { codec->patch_ops.set_power_state = haswell_set_power_state; codec->dp_mst = true; } -- 1.8.1.2

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[alsa-devel] [PATCH 1/2] ASoC: core: Implement devm_snd_soc_register_component()
by Mark Brown 04 Sep '13

04 Sep '13

From: Mark Brown <broonie(a)linaro.org> Since with the wider use of devres many drivers are now only calling snd_soc_unregister_component() in their remove functions providing a managed version will save a reasonable amount of code. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 3 +++ sound/soc/Makefile | 2 +- sound/soc/soc-devres.c | 52 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 56 insertions(+), 1 deletion(-) create mode 100644 sound/soc/soc-devres.c diff --git a/include/sound/soc.h b/include/sound/soc.h index a72af63..f126676 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -381,6 +381,9 @@ void snd_soc_unregister_codec(struct device *dev); int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, unsigned int reg); diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 61a64d2..8b9e701 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c new file mode 100644 index 0000000..13fe86f --- /dev/null +++ b/sound/soc/soc-devres.c @@ -0,0 +1,52 @@ +/* + * soc-devres.c -- ALSA SoC Audio Layer devres functions + * + * Copyright (C) 2013 Linaro Ltd + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> + +static void devm_component_release(struct device *dev, void *res) +{ + snd_soc_unregister_component(*(struct device **)res); +} + +/** + * devm_snd_soc_register_component - resource managed component registration + * @dev: Device used to manage component + * @cmpnt_drv: Component driver + * @dai_drv: DAI driver + * @num_dai: Number of DAIs to register + * + * Register a component with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_component); -- 1.8.4.rc3

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