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

[alsa-devel] CA0132 Issues with Alienware
by Antonio Vázquez Blanco 03 Jan '14

03 Jan '14

Dear Alsa developing members, I'm sorry to contact you via this mailing list even if I don't know if it used for this, but I felt I had no other option when after various attempts I saw the bug tracker was down and various members of the project didn't answer my emails. My computer has three jack ports on one side (Microphone, headphone and earphones). The problem is that Alsa currently doesn't have jack detection on earphones jack plug, so I'm stuck with using only left audio on the headphone jack. I would really like to be able to use earphones plug which in my hardware is node 0x0f. Although I know C programming I don't have the knowledge to do this myself. Thanks in advance. Your sincerely, Antonio Vázquez Blanco. -- If comunications require exchange of sensible data, please consider using my PGP public key<https://drive.google.com/file/d/0Bw0Vql7Olv4LMVlrVlJNMkVjUUU/edit?usp=shari…>for data encryption. Si la comunicación requiere el intercambio de información sensible, por favor, considere usar mi clave PGP pública<https://drive.google.com/file/d/0Bw0Vql7Olv4LMVlrVlJNMkVjUUU/edit?usp=shari…>para encriptar los datos.

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[alsa-devel] /dev/snd/timer failing on Raspberry Pi
by David Preece 02 Jan '14

02 Jan '14

Hi, As per subject, the dmix plugin is failing on Arch and a Raspberry Pi with a poll blocking around the file descriptor for /dev/snd/timer. Could it be related to the RPi not having a real time clock? There are copious details on the RPi forums... http://www.raspberrypi.org/forum/viewtopic.php?f=53&t=64936 Thanks, Dave

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[alsa-devel] [PATCH] ASoC: generic: add generic compound card with DT support
by Jean-Francois Moine 02 Jan '14

02 Jan '14

Some audio cards are built from different hardware components. When such compound cards don't need specific code, this driver creates them with the required DAI links and routes from a DT. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- This code was first developped on the generic simple card, but its recent DT extension cannot be easily extended again to support compound cards as the one in the Cubox. Note also that the example relies on a proposed patch of mine aiming to render the codec name / OF node optional in DAI links (http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070082.h…). --- .../devicetree/bindings/sound/compound-card.txt | 95 ++++++++++++ sound/soc/generic/Kconfig | 6 + sound/soc/generic/Makefile | 2 + sound/soc/generic/compound-card.c | 247 +++++++++++++++++++++++++ 4 file changed, 350 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/compound-card.txt b/Documentation/devicetree/bindings/sound/compound-card.txt new file mode 100644 index 0000000..554a796 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/compound-card.txt @@ -0,0 +1,95 @@ +Device-Tree bindings for compound audio card + +Compound audio card describes the links between the different parts +of an audio card built from different hardware components. + +Required properties: + - compatible: should be "compound-audio-card" + - audio-controller: phandle of the audio controller + +Optional properties: + - routes: list of couple of strings (sink, source) + +Required subnodes: + - link: DAI link subnode + At least one link must be specified. + +Required link subnode properties: + - link-name: names of the DAI link and of the stream + - cpu-dai-name: name of the CPU or CODEC DAI + An empty string indicates that the CPU DAI is + the same as the audio controller. + - codec-dai-name: name of the CODEC DAI + +Optional link subnode properties: + - audio-codec or codec-name: phandle or name of the CODEC + in case the codec-dai-name is not unique + - format: DAI format. One of: + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" + - front-end or back-end: present if the DAI link describes resp. + a front-end CPU DAI or a back-end CODEC DAI + - playback or capture: present if the DAI link is used for + playback or capture only + +Example node: + +The audio subsystem found in the SolidRun Cubox is composed of: + +- an audio controller which outputs playback streams on either one or both + of I2S and S/PDIF, + +- a HDMI transmitter which can get its input from either I2S or S/PDIF, + +- an optical S/PDIF connector. + + sound { + compatible = "compound-audio-card"; + audio-controller = <&audio1>; + routes = + "I2S Playback", "System Playback", + "SPDIF Playback", "System Playback", + "hdmi-out", "I2S Playback", + "hdmi-out", "SPDIF Playback", + "spdif-out", "SPDIF Playback"; + link@0 { + link-name = "System audio"; + front-end; + cpu-dai-name = "mvebu-audio"; + codec-dai-name = "snd-soc-dummy-dai"; + format = "i2s"; + playback; + }; + link@1 { + link-name = "hdmi-i2s"; + back-end; + cpu-dai-name = "i2s"; + codec-dai-name = "i2s-hifi"; + playback; + }; + link@2 { + link-name = "hdmi-spdif"; + back-end; + cpu-dai-name = "spdif"; + codec-dai-name = "spdif-hifi"; + playback; + }; + link@3 { + link-name = "spdif"; + back-end; + cpu-dai-name = "spdif"; + codec-dai-name = "dit-hifi"; + playback; + }; + }; + + hdmi_codec: hdmi-codec { + compatible = "nxp,tda998x-codec"; + audio-ports = <0x03>, <0x04>; + }; + + spdif_codec: spdif-codec { + compatible = "linux,spdif-dit"; + }; + + diff --git a/sound/soc/generic/Kconfig b/sound/soc/generic/Kconfig index 610f612..f2678ae 100644 --- a/sound/soc/generic/Kconfig +++ b/sound/soc/generic/Kconfig @@ -2,3 +2,9 @@ config SND_SIMPLE_CARD tristate "ASoC Simple sound card support" help This option enables generic simple sound card support + +config SND_COMPOUND_CARD + tristate "ASoC Compound sound card support" + depends on OF + help + This option enables the generic compound sound card support. diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 9c3b246..434cb79 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,3 +1,5 @@ snd-soc-simple-card-objs := simple-card.o +snd-soc-compound-card-objs := compound-card.o obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_COMPOUND_CARD) += snd-soc-compound-card.o diff --git a/sound/soc/generic/compound-card.c b/sound/soc/generic/compound-card.c new file mode 100644 index 0000000..f57e159 --- /dev/null +++ b/sound/soc/generic/compound-card.c @@ -0,0 +1,247 @@ +/* + * ASoC compound sound card support + * + * Copyright (C) 2013 Jean-Francois Moine <moinejf(a)free.fr> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/of.h> +#include <sound/soc.h> + +// link types +#define LINK_FE 0 /* front-end */ +#define LINK_BE 1 /* back-end */ +#define LINK_CC 2 /* (controller or codec) to codec */ + +struct compound_card_data { + struct snd_soc_card snd_card; + struct snd_soc_dai_link *snd_links; +}; + +/* get a DAI link */ +static int get_link(struct compound_card_data *priv, + struct snd_soc_dai_link *link, + struct device_node *ctrl_np, + struct device_node *np) +{ + struct device_node *codec_np; + const char *cpu_dai_name; + int type, ret; + + /* optional link-name */ + of_property_read_string(np, "link-name", &link->name); + link->stream_name = link->name; + + /* link type */ + if (of_property_read_bool(np, "front-end")) + type = LINK_FE; + else if (of_property_read_bool(np, "back-end")) + type = LINK_BE; + else + type = LINK_CC; + + /* cpu-dai-name */ + ret = of_property_read_string(np, "cpu-dai-name", + &cpu_dai_name); + if (ret) { + dev_err(priv->snd_card.dev, + "No valid cpu-dai-name found\n"); + return ret; + } + + /* an empty cpu-dai-name forces the audio-controller */ + if (cpu_dai_name[0] == '\0') + link->cpu_of_node = ctrl_np; + else + link->cpu_dai_name = cpu_dai_name; + + /* the codec may be omitted or specified by name or phandle */ + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) + link->codec_of_node = codec_np; + else + of_property_read_string(np, "codec-name", + &link->codec_name); + + /* codec-dai-name */ + ret = of_property_read_string(np, "codec-dai-name", + &link->codec_dai_name); + if (ret) { + dev_err(priv->snd_card.dev, + "Failed to parse codec-dai-name\n"); + return ret; + } + + if (type == LINK_BE) { + link->no_pcm = 1; + /* don't set the platform, otherwise, the platform functions + * are called many times */ + } else { + link->platform_of_node = ctrl_np; + if (type == LINK_FE) + link->dynamic = 1; + } + + /* DAI format */ + link->dai_fmt = snd_soc_of_parse_daifmt(np, NULL) & + SND_SOC_DAIFMT_FORMAT_MASK; + + if (of_property_read_bool(np, "playback")) + link->playback_only = 1; + else if (of_property_read_bool(np, "capture")) + link->capture_only = 1; + + return 0; +} + +static int compound_card_dt_probe(struct platform_device *pdev, + struct compound_card_data *priv) +{ + struct device_node *np, *ctrl_np, *link_np; + struct snd_soc_dai_link *link; + int ret, num_links; + + np = pdev->dev.of_node; + + ctrl_np = of_parse_phandle(np, "audio-controller", 0); + if (!ctrl_np) { + dev_err(&pdev->dev, "No valid audio-controller phandle found\n"); + return -EINVAL; + } + + /* get the optional routes */ + if (of_property_count_strings(np, "routes")) + snd_soc_of_parse_audio_routing(&priv->snd_card, "routes"); + + priv->snd_card.name = "ASoC compound sound card"; + + /* check the number of links */ + num_links = 0; + for_each_child_of_node(np, link_np) + num_links++; + + if (num_links == 0) { + dev_err(&pdev->dev, "No DAI link\n"); + return -EINVAL; + } + + link = devm_kzalloc(&pdev->dev, + sizeof(struct snd_soc_dai_link) * num_links, + GFP_KERNEL); + if (!link) { + ret = -ENOMEM; + goto err; + } + priv->snd_links = link; + priv->snd_card.num_links = num_links; + for_each_child_of_node(np, link_np) { + ret = get_link(priv, link, ctrl_np, link_np); + if (ret) + goto err; + link++; + } + + return 0; + +err: + if (link_np) + of_node_put(link_np); + if (ctrl_np) + of_node_put(ctrl_np); + for (num_links = 0, link = priv->snd_links; + num_links < priv->snd_card.num_links; + num_links++, link++) { + struct device_node *codec_np; + + codec_np = (struct device_node *) link->codec_of_node; + if (!codec_np) + break; + of_node_put(codec_np); + } + return ret; +} + +static int compound_card_dt_remove(struct platform_device *pdev, + struct compound_card_data *priv) +{ + struct device_node *codec_np; + struct snd_soc_dai_link *link; + int num_links; + + if (!priv->snd_links) + return 0; + if (priv->snd_links->platform_of_node) + of_node_put((struct device_node *) + priv->snd_links->platform_of_node); + for (num_links = 0, link = priv->snd_links; + num_links < priv->snd_card.num_links; + num_links++, link++) { + codec_np = (struct device_node *) link->codec_of_node; + if (!codec_np) + break; + of_node_put(codec_np); + } + return 0; +} + +static int compound_card_probe(struct platform_device *pdev) +{ + struct compound_card_data *priv; + int ret; + + priv = devm_kzalloc(&pdev->dev, sizeof *priv, GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->snd_card.owner = THIS_MODULE; + priv->snd_card.dev = &pdev->dev; + + ret = compound_card_dt_probe(pdev, priv); + + if (ret) + return ret; + + priv->snd_card.dai_link = priv->snd_links; + + dev_set_drvdata(&pdev->dev, priv); + + return snd_soc_register_card(&priv->snd_card); +} + +static int compound_card_remove(struct platform_device *pdev) +{ + struct compound_card_data *priv = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_card(&priv->snd_card); + + compound_card_dt_remove(pdev, priv); + + return 0; +} + +static const struct of_device_id compound_card_of_dev_ids[] = { + { .compatible = "compound-audio-card", }, + {} +}; +MODULE_DEVICE_TABLE(of, compound_card_of_dev_ids); + +static struct platform_driver compound_card = { + .driver = { + .name = "asoc-compound-card", + .owner = THIS_MODULE, + .of_match_table = compound_card_of_dev_ids, + }, + .probe = compound_card_probe, + .remove = compound_card_remove, +}; + +module_platform_driver(compound_card); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("ASoC Compound Sound Card"); +MODULE_AUTHOR("Jean-Francois Moine <moinejf(a)free.fr>"); -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] [PATCH] ASoC: ssm2518: Fix off-by-one error by ffs()
by Takashi Iwai 02 Jan '14

02 Jan '14

ffs() returns the bit position from 1, while the ssm2158 driver code assumes it being 0-based. Also, the bit mask computation of the two channel slots are incorrect; it must have worked just casually. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I found it casully while checking ab8500 bug. The patch is untested, so please review/test first, and give Cc to stable if needed. sound/soc/codecs/ssm2518.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 95aed552139a..cc8debce752f 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -549,13 +549,13 @@ static int ssm2518_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, right_slot = 0; } else { /* We assume the left channel < right channel */ - left_slot = ffs(tx_mask); - tx_mask &= ~(1 << tx_mask); + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); if (tx_mask == 0) { right_slot = left_slot; } else { - right_slot = ffs(tx_mask); - tx_mask &= ~(1 << tx_mask); + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); } } -- 1.8.5.2

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Re: [alsa-devel] [PATCH 08/19] ASoC: kirkwood: Don't set unused struct snd_pcm_hardware fields
by Jean-Francois Moine 02 Jan '14

02 Jan '14

> On 12/20/2013 08:13 PM, Jean-Francois Moine wrote: > > On Fri, 20 Dec 2013 18:18:49 +0100 > > Lars-Peter Clausen <lars(a)metafoo.de> wrote: > > > >> On 12/20/2013 07:05 PM, Jean-Francois Moine wrote: > >>>> diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c > >>>> index 4af1936..aac22fc 100644 > >>>> --- a/sound/soc/kirkwood/kirkwood-dma.c > >>>> +++ b/sound/soc/kirkwood/kirkwood-dma.c > >>> [snip] > >>>> @@ -43,12 +33,6 @@ static struct snd_pcm_hardware kirkwood_dma_snd_hw = { > >>>> SNDRV_PCM_INFO_MMAP_VALID | > >>>> SNDRV_PCM_INFO_BLOCK_TRANSFER | > >>>> SNDRV_PCM_INFO_PAUSE), > >>>> - .formats = KIRKWOOD_FORMATS, > >>>> - .rates = KIRKWOOD_RATES, > >>>> - .rate_min = 8000, > >>>> - .rate_max = 384000, > >>>> - .channels_min = 1, > >>>> - .channels_max = 8, > >>>> .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES, > >>>> .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, > >>>> .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, > >>> > >>> Lars, > >>> > >>> You removed too many things. The 'formats' field is mandatory. > >> > >> No it is not. While snd_soc_set_runtime_hwparams() uses it it is later > >> overwritten again in soc_pcm_init_runtime_hw(). > > > > I have a DPCM system and soc_pcm_init_runtime_hw() is not called > > (either 'dynamic' or 'no_pcm' is set in the DAI links). > > Ok, I see dpcm_set_fe_runtime() does things slightly different. Well, I advanced a bit with DPCM, and I have a problem with this field. In the driver, the front-end DAI is the audio controller and the back-ends DAIs are the HDMI and SPDIF outputs. These back-end DAIs have different rates and formats, as have the audio controller outputs (I2S and SPDIF). So, I used intermediate DAIs which represent the audio controller outputs, and they are described in the DAI links: link 0 (FE): audio controller <-> dummy DAI link 1 (BE): i2s audio controller output <-> HDMI output link 2 (BE): spdif audio controller output <-> HDMI output link 3 (BE): spdif audio controller output <-> SPDIF output Without any patch in the core, the rates and formats are always the rates and formats of the audio controller (FE). This is due to the 'goto dynamic' in soc_pcm_open(): as the back-ends are linked to real DAIs, the rate and format constraints must be checked. So, as a temporary patch, I replaced: if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) goto dynamic; by: if (rtd->dai_link->dynamic) goto dynamic; (indeed, this will not work if the back-end is linked to the dummy DAI) and, I get the correct rates and formats in the runtime hardware parameters. But these values are lost: - on DMA open (FE pcm open), when the driver calls snd_soc_set_runtime_hwparams() (loss of the formats), and - on dpcm_set_fe_runtime() call (loss of the rates). The first problem can be fixed in the audio controller by a hack, saving /restoring the formats on calling snd_soc_set_runtime_hwparams(), and the second problem is easily fixed moving dpcm_set_fe_runtime() at the beginning of dpcm_fe_dai_startup(). Are these good solutions? -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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[alsa-devel] hda problem on Gigabyte GA-Z87X-UD3H motherboard
by Guillaume Desclaux 02 Jan '14

02 Jan '14

Hi, I've reported an issue with snd-hda-intel with my Gigabyte GA-Z87X-UD3H motherboard on Launchpad but it seems not the best place to that See :https://bugs.launchpad.net/bugs/1248116 In "Volume control PulseAudio", I can see pulseaudio changing the audio output "Internal Audio Analogic Stereo". The port switch rapidly from "Analogic output" to "Analogic headphones" making a jerky sound. You can find the trace.log file here : https://bugs.launchpad.net/ubuntu/+source/pulseaudio/+bug/1248116/+attachme… It seems that there are many hda_power_count 1/ This problem appears with the front connector on the motherboard connected to a HDA front panel on the case (I've double-checked the connection) 2/ The problem still appears when the front connector on the motherboard is not connected at all to the case ! 3/ This bug seems quite common with Gigabyte motherboard : If you search with Google "Gigabyte Z87X hda" you will find other descriptions of this same strange behavior : https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1242302 https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1067434 If I disable jack detection by specifying a hint for early patching (jack_detect = no), the sound is cristal clear ! If there is anything I can do to help you understand what happens, don't hesitate to ask. Best regards Guillaume

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[alsa-devel] [PATCH] ASoC: simple-card: cleanup asoc_simple_card_parse_of() code
by Xiubo Li 02 Jan '14

02 Jan '14

Since there is only one common format via "simple-audio-card,format" for simple card driver(CPU & CODEC DAI), there is no need to do the INV mask. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d340320..8189e68 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -135,7 +135,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, /* get CPU/CODEC common format via simple-audio-card,format */ info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & - (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + SND_SOC_DAIFMT_FORMAT_MASK; ret = asoc_simple_card_get_component(node, info); if (ret) -- 1.8.4

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[alsa-devel] [PATCH] ASoC: simple-card: Add DAPM routes parse from device tree
by Xiubo Li 02 Jan '14

02 Jan '14

Parses a simple DAPM route table from device tree. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Documentation/devicetree/bindings/sound/simple-card.txt | 13 +++++++++++-- sound/soc/generic/simple-card.c | 11 +++++++++-- 2 files changed, 20 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 769a346..2ee80c76 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -9,8 +9,13 @@ Required properties: Optional properties: - simple-audio-card,format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" + "i2s", "right_j", "left_j" , "dsp_a" + "dsp_b", "ac97", "pdm", "msb", "lsb" +- simple-audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. + Required subnodes: - simple-audio-card,cpu : CPU sub-node @@ -38,6 +43,10 @@ Example: sound { compatible = "simple-audio-card"; simple-audio-card,format = "left_j"; + simple-audio-routing = + "MIC_IN", "Mic Jack", + "Headphone Jack", "HP_OUT", + "Ext Spk", "LINE_OUT"; simple-audio-card,cpu { sound-dai = <&sh_fsi2 0>; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 1d87db8..643c9ed 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -116,12 +116,18 @@ static int asoc_simple_card_parse_of(struct device_node *node, { struct device_node *np; char *name; - int ret = 0; + int ret; /* get CPU/CODEC common format via simple-audio-card,format */ info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + /* DAPM routes */ + ret = snd_soc_of_parse_audio_routing(&info->snd_card, + "simple-audio-routing"); + if (ret) + return ret; + /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); @@ -185,6 +191,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); if (cinfo) { int ret; + cinfo->snd_card.dev = &pdev->dev; ret = asoc_simple_card_parse_of(np, cinfo, dev, &of_cpu, &of_codec, @@ -196,6 +203,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) } } } else { + cinfo->snd_card.dev = &pdev->dev; cinfo = pdev->dev.platform_data; } @@ -235,7 +243,6 @@ static int asoc_simple_card_probe(struct platform_device *pdev) cinfo->snd_card.owner = THIS_MODULE; cinfo->snd_card.dai_link = &cinfo->snd_link; cinfo->snd_card.num_links = 1; - cinfo->snd_card.dev = &pdev->dev; return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); } -- 1.8.4

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[alsa-devel] [PATCH] ASoC: add snd_soc_of_parse_master_clkdir() for dt
by Xiubo Li 02 Jan '14

02 Jan '14

This patch adds snd_soc_of_parse_master_clkdir() and supports below style on dt: [prefix]master-clkdir; If this property is absent 0 will be returned, 1 otherwise. 0 : SND_SOC_CLOCK_IN 1 : SND_SOC_CLOCK_OUT Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- I'm not very sure whether this patch is need for dt. For the CPU and CODEC devices' sysclk directions maybe different of the same DAI link. include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 19 +++++++++++++++++++ 2 files changed, 21 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5a049d9..1d6396a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1170,6 +1170,8 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname); int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); +unsigned int snd_soc_of_parse_master_clkdir(struct device_node *np, + const char *prefix); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix); int snd_soc_of_get_dai_name(struct device_node *of_node, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index be88df5..faf102f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4472,6 +4472,25 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_routing); +unsigned int snd_soc_of_parse_master_clkdir(struct device_node *np, + const char *prefix) +{ + char prop[128]; + + if (!prefix) + prefix = ""; + + /* + * check "[prefix]master-clkdir" + * true means SND_SOC_CLOCK_OUT + * false means SND_SOC_CLOCK_IN + */ + snprintf(prop, sizeof(prop), "%smaster-clkdir", prefix); + + return !!of_property_read_bool(np, prop); +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_master_clkdir); + unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix) { -- 1.8.4

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[alsa-devel] [PATCH] ASoC: simple-card: Add device's module clock selection.
by Xiubo Li 02 Jan '14

02 Jan '14

Try to get the device's module clock if the dt has no clocks and system-clock-frequency properties. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- a.out | 0 sound/soc/generic/simple-card.c | 21 ++++++++++++++++++--- 2 files changed, 18 insertions(+), 3 deletions(-) create mode 100644 a.out diff --git a/a.out b/a.out new file mode 100644 index 0000000..e69de29 diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6ce9745..c0d92813 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -90,14 +90,29 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * dai->sysclk come from * "clocks = <&xxx>" (if system has common clock) * or "system-clock-frequency = <xxx>" + * or device's module clock. */ - clk = of_clk_get(np, 0); - if (IS_ERR(clk)) + if (of_property_read_bool(np, "clocks")) { + clk = of_clk_get(np, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + + dai->sysclk = clk_get_rate(clk); + } else if (of_property_read_bool(np, "system-clock-frequency")) { of_property_read_u32(np, "system-clock-frequency", &dai->sysclk); - else + } else { + clk = of_clk_get(*node, 0); + if (IS_ERR(clk)) { + ret = PTR_ERR(clk); + goto parse_error; + } + dai->sysclk = clk_get_rate(clk); + } ret = 0; -- 1.8.4

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