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

[alsa-devel] [PATCH] ASoC: simple-card: fix __asoc_simple_card_dai_init
by Xiubo Li 16 Jan '14

16 Jan '14

If the CPU/CODEC DAI set_sysclk() is not support, the -ENOTSUPP will returnd. Here do the check like set_fmt(). Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/generic/simple-card.c | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5..b6014fe 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -19,22 +19,27 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) { - int ret = 0; + int ret; daifmt |= set->fmt; - if (daifmt) + if (daifmt) { ret = snd_soc_dai_set_fmt(dai, daifmt); - - if (ret == -ENOTSUPP) { - dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n"); - ret = 0; + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_fmt error\n"); + return ret; + } } - if (!ret && set->sysclk) + if (set->sysclk) { ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_sysclk error\n"); + return ret; + } + } - return ret; + return 0; } static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) -- 1.8.4

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[alsa-devel] [PATCH 0/3] ASoC: tlv320aic32x4 bugfixes
by Markus Pargmann 16 Jan '14

16 Jan '14

Hi, Another round of tlv320aic32x4 bugfixes. I will send some more non-bugfix patches for this codec next week including the DT compatibles we talked about. Regards, Markus Markus Pargmann (3): ASoC: tlv320aic32x4: Fix regmap range_min ASoC: tlv320aic32x4: Fix mono playback ASoC: tlv320aic32x4: Fix MICPGA input configuration sound/soc/codecs/tlv320aic32x4.c | 30 ++++++++++++++++++------------ sound/soc/codecs/tlv320aic32x4.h | 3 +++ 2 files changed, 21 insertions(+), 12 deletions(-) -- 1.8.5.2

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[alsa-devel] [PATCH] ASoC: core: Return -ENOTSUPP from set_sysclk() if no operation provided
by Mark Brown 15 Jan '14

15 Jan '14

From: Mark Brown <broonie(a)linaro.org> Make it easier for generic code to work with set_sysclk() by distinguishing between the operation not being supported and an error as is done for other operations like set_dai_fmt() Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a128f0cd752..fe1df50805a3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3485,7 +3485,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, freq, dir); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); @@ -3506,7 +3506,7 @@ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, return codec->driver->set_sysclk(codec, clk_id, source, freq, dir); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk); -- 1.8.5.2

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Re: [alsa-devel] Surround sound not working on ALC887 based motherboard
by Raymond Yau 15 Jan '14

15 Jan '14

2014/1/15 Olivier Crête <olivier.crete(a)ocrete.ca> > Hi, > > > > > I have a Asus P5G41T-M LX Plus motherboard which is ICH7 + > > > Realtek > > > ALC887-VD, alsa only lets me access the first 2 channels > > > (stereo), but > > > not the 6 other ones. It works fine in Windows. > > > > > > Trying to run this after killing pulseaudio fails: > > > > > > $ aplay -D surround40:0 -f S16_LE -r 44100 -c 4 /dev/zero -v > > > Playing raw data '/dev/zero' : Signed 16 bit Little Endian, > > > Rate 44100 Hz, Channels 4 > > > aplay: set_params:1239: Channels count non available > > > > > > > > > Output from alsa-info: > > > > http://www.alsa-project.org/db/?f=fe378d28f0dbb5a427b005d20866a762fd37d31a > > > > > > > > post output of pactl list or pulseaudio verbose log > > > > http://www.ocrete.ca/surround/pactl-list.txt > > http://www.ocrete.ca/surround/pulseaudio-vv.txt > > > > > you have to change channel mode from 2ch to 6ch to retask the input > > > jacks as output if your motherboard only have three audio jacks at > > > rear panel > > > > How would one do that ? Pulseaudio only offers Stereo profiles. > amixer -c0 set "Channel Mode" "6ch" > > > do you mean pulseaudio cannot probe the surroun40 or surround51 > > profile ? > > Yes, but I can't use it with aplay even without pulseaudio running. > > > Simple mixer control 'Channel Mode',0 > > Capabilities: enum > > Items: '2ch' '4ch' '6ch' > > Item0: '2ch' > > > For some reason didn't get your last email, but I saw I saw it on the > archive.. Thanks for your solution, it worked! Definitely something > PulseAudio should expose better. do it really work since rear mic become active port even when it had been retasked as output ? Ports: analog-input-front-mic: Front Microphone (priority: 8500, not available) analog-input-rear-mic: Rear Microphone (priority: 8200, available) analog-input-linein: Line In (priority: 8100, available) Active Port: analog-input-rear-mic

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[alsa-devel] [PATCH] Change the prototype of soc_widget_read
by Arun Shamanna Lakshmi 15 Jan '14

15 Jan '14

soc_widget_read API returns the register data and it is possible that a register can contain 0xffffffff at any point of time. In such cases snd_soc_widget_write is not called after the read operation inside snd_soc_update_bits_locked API. Thus, change the prototype of soc_widget_read to return only the error code and pass the register data through the pointer argument. Arun Shamanna Lakshmi (1): ASoC: dapm: Change prototype of soc_widget_read sound/soc/soc-dapm.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) -- 1.7.9.5

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[alsa-devel] [PATCH 0/2] ASoC: codec tlv320aic32x4 fixes
by Markus Pargmann 15 Jan '14

15 Jan '14

Hi, Here are two fixes for the tlv320aic32x4 codec. The first patch fixes a regmap bug introduced in 3.13. The second is not as important, it adds the codec to the compatible list in the binding documentation. Regards, Markus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: codec doc, tlv320aic3x: Add tlv320aic32x4 as compatible Documentation/devicetree/bindings/sound/tlv320aic3x.txt | 1 + sound/soc/codecs/tlv320aic32x4.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) -- 1.8.5.2

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[alsa-devel] [PATCH v3 0/6] dma: Indicate residue granularity in dma_slave_caps
by Lars-Peter Clausen 14 Jan '14

14 Jan '14

New in v3: * Expanded the kernel doc documentation for the dma_residue_granularity struct a bit New in v2: * Dropped byte level granularity as discussed during v1 review * Added support for auto-detecting the NO_RESIDUE flag on the compat path in the generic ASoC dmaengine PCM driver * Updated a few ASoC drivers to take advantage of the new API Updated original cover text from v1 below: This patch series introduces a new field to the dma_slave_caps struct which exposes the granularity with which the reported residue is updated, updates the pl330 DMA driver to set this field and lets the generic dmaengine PCM audio driver check that field. The reason for introducing the field is that for audio we need to know the granularity with which the reported residue is updated. The uncertainty of the ALSA period pointer depends on the granularity of the reported residue. E.g. if residue is only updated once every period this means that the position in the buffer from/to which the hardware is currently reading/writing could be anywhere between the reported PCM pointer and the reported PCM pointer plus one period size (plus some extra margin for interrupt latency, etc). Certain algorithms (e.g. pulseaudio's timer based scheduling) only work correctly if the uncertainty of the PCM pointer is low enough. Exposing this information from the DMA driver makes it possible for applications to switch to alternative modes of operation which also work with a high granularity. The patch series introduces four levels of granularity, which I think will cover most DMA cores. I'd like some feedback though if anybody can think of a DMA controller for which none of the levels would be correct. The four levels are: * DESCRIPTOR: The DMA channel is only able to tell whether a descriptor has been completed or not, which means residue reporting is not supported by this channel. The residue field of the dma_tx_state field will always be 0. * SEGMENT: The DMA channel updates the residue field after each successfully completed segment of the transfer (For cyclic transfers this is after each period). This is typically implemented by having the hardware generate an interrupt after each transferred segment and then the driver updates the outstanding residue by the size of the segment. Another possibility is if the hardware supports SG and the segment descriptor has a field which gets set by the hardware after the segment has been completed. The driver then counts the number of segments which do not have the flag set to compute the residue. * BURST: The DMA channel updates the residue field after each transferred burst. This is typically only supported if the hardware has a progress register of some sort (E.g. a register with the current read/write address or a register with the amount of bursts/beats/bytes that have been transferred or still need to be transferred). - Lars Lars-Peter Clausen (6): dma: Indicate residue granularity in dma_slave_caps dma: pl330: Set residue_granularity ASoC: generic-dmaengine-pcm: Check NO_RESIDUE flag at runtime ASoC: generic-dmaengine-pcm: Check DMA residue granularity ASoC: axi-{spdif,i2s}: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag ASoC: samsung: Remove SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag drivers/dma/pl330.c | 1 + include/linux/dmaengine.h | 28 +++++++++++++++ sound/soc/adi/axi-i2s.c | 3 +- sound/soc/adi/axi-spdif.c | 3 +- sound/soc/samsung/dmaengine.c | 1 - sound/soc/soc-generic-dmaengine-pcm.c | 66 ++++++++++++++++++++++------------- 6 files changed, 73 insertions(+), 29 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH resend] ASoC: simple-card: fix one bug to writing to the platform data
by Xiubo Li 14 Jan '14

14 Jan '14

It's a bug that writing to the platform data directly, for it should be constant. So just copy it before writing. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Resend this patch. sound/soc/generic/simple-card.c | 40 +++++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 19 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5528dd6..53395f5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,9 +9,10 @@ * published by the Free Software Foundation. */ #include <linux/clk.h> +#include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> -#include <linux/module.h> +#include <linux/string.h> #include <sound/simple_card.h> static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, @@ -190,36 +191,37 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device_node *np = pdev->dev.of_node; struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; + int ret; cinfo = NULL; of_cpu = NULL; of_codec = NULL; of_platform = NULL; + + cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); + if (!cinfo) + return -ENOMEM; + if (np && of_device_is_available(np)) { - 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, - &of_platform); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - return ret; - } - } else { - return -ENOMEM; + cinfo->snd_card.dev = dev; + + ret = asoc_simple_card_parse_of(np, cinfo, dev, + &of_cpu, + &of_codec, + &of_platform); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; } } else { - cinfo = pdev->dev.platform_data; - if (!cinfo) { + if (!dev->platform_data) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; + memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); + cinfo->snd_card.dev = dev; } if (!cinfo->name || -- 1.8.4

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[alsa-devel] [PATCH v2 1/5] ASoC: pcm: Properly initialize hw->rate_max
by Lars-Peter Clausen 14 Jan '14

14 Jan '14

If none of the components (CODEC or CPU DAI) sets a maximum sample rate we'll end up with the rate_max field of the runtime hardware set to 0. (Note that it is still possible for the components to constrain the supported sample rates using other methods, e.g. setting a list constraint) If rate_max is 0 this means that the sound card doesn't support any rates at all, which is not the desired result. So initialize rate_max to UINT_MAX. For symmetry reasons also set rate_min to 0. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes in v2 --- sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 313adba..b6e0c32 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -252,6 +252,9 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + hw->rate_min = 0; + hw->rate_max = UINT_MAX; + snd_pcm_limit_hw_rates(runtime); hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); -- 1.8.0

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[alsa-devel] [PATCH] ASoC: simple-card: simplify code
by Jean-Francois Moine 14 Jan '14

14 Jan '14

This patch - removes the fields of the platform data which are of no use to the non-DT platform callers, - uses a new private structure to handle all the sound card information, - simplifies the code and make easier a possible multi-DAI links extension. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- Xiubo, I also removed 'of_device_is_available' which seems really useless: the module is not probed when the DT status is not "okay". --- include/sound/simple_card.h | 4 -- sound/soc/generic/simple-card.c | 178 ++++++++++----------- 2 files changed, 88 insertions(+), 94 deletions(-) diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 6c74527..e1ac996 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -29,10 +29,6 @@ struct asoc_simple_card_info { unsigned int daifmt; struct asoc_simple_dai cpu_dai; struct asoc_simple_dai codec_dai; - - /* used in simple-card.c */ - struct snd_soc_dai_link snd_link; - struct snd_soc_card snd_card; }; #endif /* __SIMPLE_CARD_H */ diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 5528dd6..580900e 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -14,6 +14,13 @@ #include <linux/module.h> #include <sound/simple_card.h> +struct simple_card_data { + struct snd_soc_card snd_card; + unsigned int daifmt; + struct asoc_simple_dai cpu_dai; + struct asoc_simple_dai codec_dai; +}; + static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, struct asoc_simple_dai *set, unsigned int daifmt) @@ -38,18 +45,18 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; + unsigned int daifmt = priv->daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &priv->codec_dai, daifmt); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &priv->cpu_dai, daifmt); if (ret < 0) return ret; @@ -59,7 +66,8 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, struct asoc_simple_dai *dai, - struct device_node **node) + struct device_node **node, + const char **name) { struct clk *clk; int ret; @@ -73,7 +81,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, return -ENODEV; /* get dai->name */ - ret = snd_soc_of_get_dai_name(np, &dai->name); + ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) goto parse_error; @@ -117,23 +125,21 @@ parse_error: } static int asoc_simple_card_parse_of(struct device_node *node, - struct asoc_simple_card_info *info, - struct device *dev, - struct device_node **of_cpu, - struct device_node **of_codec, - struct device_node **of_platform) + struct simple_card_data *priv, + struct device *dev) { + struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; struct device_node *np; char *name; int ret; /* get CPU/CODEC common format via simple-audio-card,format */ - info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + priv->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&priv->snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -144,8 +150,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,cpu"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->cpu_dai, - of_cpu); + &priv->cpu_dai, + (struct device_node **) + &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); if (ret < 0) return ret; @@ -154,109 +162,99 @@ static int asoc_simple_card_parse_of(struct device_node *node, np = of_get_child_by_name(node, "simple-audio-card,codec"); if (np) ret = asoc_simple_card_sub_parse_of(np, - &info->codec_dai, - of_codec); + &priv->codec_dai, + (struct device_node **) + &dai_link->codec_of_node, + &dai_link->codec_dai_name); if (ret < 0) return ret; /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(info->cpu_dai.name) + - strlen(info->codec_dai.name) + 2, + strlen(dai_link->cpu_dai_name) + + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); - sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); - info->name = info->card = name; + sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); + priv->snd_card.name = name; + dai_link->name = dai_link->stream_name = name; /* simple-card assumes platform == cpu */ - *of_platform = *of_cpu; + dai_link->platform_of_node = dai_link->cpu_of_node; - dev_dbg(dev, "card-name : %s\n", info->card); - dev_dbg(dev, "platform : %04x\n", info->daifmt); + dev_dbg(dev, "card-name : %s\n", name); + dev_dbg(dev, "platform : %04x\n", priv->daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", - info->cpu_dai.name, - info->cpu_dai.fmt, - info->cpu_dai.sysclk); + dai_link->cpu_dai_name, + priv->cpu_dai.fmt, + priv->cpu_dai.sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", - info->codec_dai.name, - info->codec_dai.fmt, - info->codec_dai.sysclk); + dai_link->codec_dai_name, + priv->codec_dai.fmt, + priv->codec_dai.sysclk); return 0; } static int asoc_simple_card_probe(struct platform_device *pdev) { + struct simple_card_data *priv; + struct snd_soc_dai_link *dai_link; struct asoc_simple_card_info *cinfo; struct device_node *np = pdev->dev.of_node; - struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; - cinfo = NULL; - of_cpu = NULL; - of_codec = NULL; - of_platform = NULL; - if (np && of_device_is_available(np)) { - 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, - &of_platform); - if (ret < 0) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "parse error %d\n", ret); - return ret; - } - } else { - return -ENOMEM; - } - } else { - cinfo = pdev->dev.platform_data; - if (!cinfo) { - dev_err(dev, "no info for asoc-simple-card\n"); + priv = devm_kzalloc(dev, + sizeof(*priv) + sizeof(*dai_link), + GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->snd_card.owner = THIS_MODULE; + priv->snd_card.dev = &pdev->dev; + dai_link = (struct snd_soc_dai_link *) (priv + 1); + priv->snd_card.dai_link = dai_link; + priv->snd_card.num_links = 1; + + cinfo = pdev->dev.platform_data; + if (cinfo) { + if (!cinfo->name || + !cinfo->card || + !cinfo->codec_dai.name || + !cinfo->codec || + !cinfo->platform || + !cinfo->cpu_dai.name) { + dev_err(dev, "insufficient asoc_simple_card_info settings\n"); return -EINVAL; } - cinfo->snd_card.dev = &pdev->dev; - } - - if (!cinfo->name || - !cinfo->card || - !cinfo->codec_dai.name || - !(cinfo->codec || of_codec) || - !(cinfo->platform || of_platform) || - !(cinfo->cpu_dai.name || of_cpu)) { - dev_err(dev, "insufficient asoc_simple_card_info settings\n"); + dai_link->name = cinfo->name; + dai_link->stream_name = cinfo->name; + dai_link->cpu_dai_name = cinfo->cpu_dai.name; + dai_link->platform_name = cinfo->platform; + dai_link->codec_name = cinfo->codec; + dai_link->codec_dai_name = cinfo->codec_dai.name; + memcpy(&priv->cpu_dai, &cinfo->cpu_dai, + sizeof(priv->cpu_dai)); + memcpy(&priv->codec_dai, &cinfo->codec_dai, + sizeof(priv->codec_dai)); + dai_link->init = asoc_simple_card_dai_init; + } else if (np) { + int ret; + + ret = asoc_simple_card_parse_of(np, priv, dev); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "parse error %d\n", ret); + return ret; + } + } else { + dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - /* - * init snd_soc_dai_link - */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; - cinfo->snd_link.init = asoc_simple_card_dai_init; - - /* - * init snd_soc_card - */ - cinfo->snd_card.name = cinfo->card; - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; - - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + snd_soc_card_set_drvdata(&priv->snd_card, priv); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); + return devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); } static const struct of_device_id asoc_simple_of_match[] = { -- Ken ar c'hentañ | ** Breizh ha Linux atav! ** Jef | http://moinejf.free.fr/

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