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

[alsa-devel] [PATCH] fix format type mismatch in sound/pci/hda/patch_sigmatel.c
by Toralf Förster 21 May '14

21 May '14

Signed-off-by: Toralf Förster <toralf.foerster(a)gmx.de> --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 75515b4..7f40a15 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -795,7 +795,7 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) } while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { - if (sscanf(dev->name, "HP_Mute_LED_%d_%x", + if (sscanf(dev->name, "HP_Mute_LED_%u_%x", &spec->gpio_led_polarity, &spec->gpio_led) == 2) { unsigned int max_gpio; @@ -808,7 +808,7 @@ static int find_mute_led_cfg(struct hda_codec *codec, int default_polarity) spec->vref_mute_led_nid = spec->gpio_led; return 1; } - if (sscanf(dev->name, "HP_Mute_LED_%d", + if (sscanf(dev->name, "HP_Mute_LED_%u", &spec->gpio_led_polarity) == 1) { set_hp_led_gpio(codec); return 1; -- 1.9.3

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Re: [alsa-devel] [PATCH] ALSA: hda - Disable AA-loopback on ALC668
by Takashi Iwai 21 May '14

21 May '14

At Tue, 20 May 2014 18:58:26 +0200, Gabriele Mazzotta wrote: > > On Tuesday 20 May 2014 18:06:53, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 20 May 2014 17:59:30 +0200, > > Gabriele Mazzotta wrote: > > > > > > On Thursday 15 May 2014 21:56:23, Gabriele Mazzotta <gabriele.mzt(a)gmail.com> > > > wrote: > > > > On Wednesday 14 May 2014 16:31:13, Gabriele Mazzotta > > > <gabriele.mzt(a)gmail.com> wrote: > > > > I'm sorry, I was wrong. This patch does not help. I'm still trying to > > > > understand what temporarily fixed the problem, I was quite sure this patch > > > did > > > > it, but all of the sudden the noise came back. I'm still trying to find the > > > > real cause, but I doubt I'll succeed. > > > > > > > > Regards, > > > > Gabriele > > > I have to correct myself again, the first patch I submitted is working, I'm > > > sorry. However, it seems that disabling the AA-loopback path is not necessary, > > > muting the input 0x0b in node 0x0c (as it was suggested here: > > > https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1252733/comments…) is enough to get rid of the white noise. > > > > This reminds me of a similar problem reported on Sony VAIO. > > https://bugzilla.novell.com/show_bug.cgi?id=873704 > > > > The codec is different, it's ALC275, though. > > > > > What is the proper way to fix this problem? > > > > Just to be sure: does the mic input itself work correctly? > > Unfortunately, I cannot test it. > > > Maybe the easiest fix would be disabling the aamix, indeed. > > But you shouldn't apply it generically to ALC668. I'm sure that it's > > pretty specific to machines, not codec chips. So, create a fixup > > function that matches with your machine instead. > > Yes, you are probably right. All I know is that this bug affect the > XPS13, XPS12 and XPS15. > > Is the following patch OK? > --- > > > ALSA: hda - White noise fix for XPS13 9333 > > Disable the AA-loopback path to get rid of the constant white noise > that can be heard when headphones are used. > > Signed-off-by: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> Looks good, so applied now. Thanks. Takashi > --- > sound/pci/hda/patch_realtek.c | 19 ++++++++++++++++++- > 1 file changed, 18 insertions(+), 1 deletion(-) > > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 49e884f..a2c4374 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -3894,6 +3894,16 @@ static void alc_fixup_no_shutup(struct hda_codec *codec, > } > } > > +static void alc_fixup_disable_aamix(struct hda_codec *codec, > + const struct hda_fixup *fix, int action) > +{ > + if (action == HDA_FIXUP_ACT_PRE_PROBE) { > + struct alc_spec *spec = codec->spec; > + /* Disable AA-loopback as it causes white noise */ > + spec->gen.mixer_nid = 0; > + } > +} > + > static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, > const struct hda_fixup *fix, int action) > { > @@ -5313,6 +5323,7 @@ enum { > ALC662_FIXUP_BASS_1A, > ALC662_FIXUP_BASS_CHMAP, > ALC668_FIXUP_AUTO_MUTE, > + ALC668_FIXUP_DELL_DISABLE_AAMIX, > }; > > static const struct hda_fixup alc662_fixups[] = { > @@ -5479,6 +5490,12 @@ static const struct hda_fixup alc662_fixups[] = { > .type = HDA_FIXUP_FUNC, > .v.func = alc_fixup_inv_dmic_0x12, > }, > + [ALC668_FIXUP_DELL_DISABLE_AAMIX] = { > + .type = HDA_FIXUP_FUNC, > + .v.func = alc_fixup_disable_aamix, > + .chained = true, > + .chain_id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE > + }, > [ALC668_FIXUP_AUTO_MUTE] = { > .type = HDA_FIXUP_FUNC, > .v.func = alc_fixup_auto_mute_via_amp, > @@ -5539,7 +5556,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { > SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), > SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > - SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > + SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_DISABLE_AAMIX), > SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), > SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), > SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > -- > 2.0.0.rc2 >

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[alsa-devel] [PATCH] ASoC: samsung: Add 'mclk' handling for Snow sound-card driver
by Tushar Behera 21 May '14

21 May '14

For audio to work on Snow board, we need the codec master clock 'mclk' to be properly configured. Currently XCLKOUT is configured as 'mclk' for codec chip and it is required to be clocked at 24MHz for the codec to work properly. Add appropriate clock handling within Snow sound-card driver to set 'mclk' to operate at 24MHz. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- Documentation/devicetree/bindings/sound/snow.txt | 7 ++++++ sound/soc/samsung/snow.c | 26 +++++++++++++++++++++- 2 files changed, 32 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index 678b191..94dd29c 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -7,6 +7,10 @@ Required properties: - samsung,i2s-controller: The phandle of the Samsung I2S controller - samsung,audio-codec: The phandle of the audio codec +Optional properties: +- clocks: The phandle of the master clock to audio codec +- clock-names: Should be "mclk" + Example: sound { @@ -14,4 +18,7 @@ sound { samsung,i2s-controller = <&i2s0>; samsung,audio-codec = <&max98095>; + + clocks = <&pmu_system_controller>; + clock-names = "mclk"; }; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 014c177..c910739 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -15,13 +15,18 @@ #include <linux/platform_device.h> #include <linux/of.h> #include <linux/of_device.h> +#include <linux/clk.h> #include <sound/soc.h> #include "i2s.h" +/* Desired clock rate for codec mclk */ #define FIN_PLL_RATE 24000000 +static struct clk *mclk; +static unsigned long mclk_rate; + static struct snd_soc_dai_link snow_dai[] = { { .name = "Primary", @@ -41,7 +46,7 @@ static int snow_late_probe(struct snd_soc_card *card) /* Set the MCLK rate for the codec */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, - FIN_PLL_RATE, SND_SOC_CLOCK_IN); + mclk_rate, SND_SOC_CLOCK_IN); if (ret < 0) return ret; @@ -68,6 +73,16 @@ static int snow_probe(struct platform_device *pdev) struct device_node *i2s_node, *codec_node; int i, ret; + /* The codec MCLK should be clocked at 24MHz */ + mclk = devm_clk_get(&pdev->dev, "mclk"); + if (!IS_ERR(mclk)) { + clk_prepare_enable(mclk); + clk_set_rate(mclk, FIN_PLL_RATE); + mclk_rate = clk_get_rate(mclk); + } else { + mclk_rate = FIN_PLL_RATE; + } + i2s_node = of_parse_phandle(pdev->dev.of_node, "samsung,i2s-controller", 0); if (!i2s_node) { @@ -101,6 +116,14 @@ static int snow_probe(struct platform_device *pdev) return ret; } +static int snow_remove(struct platform_device *pdev) +{ + if (!IS_ERR(mclk)) + clk_disable_unprepare(mclk); + + return 0; +} + static const struct of_device_id snow_of_match[] = { { .compatible = "google,snow-audio-max98090", }, { .compatible = "google,snow-audio-max98095", }, @@ -115,6 +138,7 @@ static struct platform_driver snow_driver = { .of_match_table = snow_of_match, }, .probe = snow_probe, + .remove = snow_remove, }; module_platform_driver(snow_driver); -- 1.7.9.5

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[alsa-devel] Support for getting "manufacturer" from snd_seq_client_info and/or snd_ctl_card_info
by Adam Goode 21 May '14

21 May '14

Hi, I am doing a bit of work on the Alsa Web MIDI implementation in Chrome: https://code.google.com/p/chromium/issues/detail?id=344410 Web MIDI likes to know the "manufacturer" field as part of its API: http://webaudio.github.io/web-midi-api/#attributes-3 The current implementation in Chrome uses rawmidi, and then tries to parse out the manufacturer from the longname of the card. This works reasonably well, but is brittle and results in a lot of code. I am working on a seq implementation for Chrome, to replace the rawmidi implementation. I can retain the manufacturer extraction hack on the card, but seq requires yet another hack to guess the card for a particular seq client. Any suggestions on how to extract the manufacturer more easily? The best solution would be to add a manufacturer (or vendor) field to the snd_seq_client_info ioctl struct. This would likely require a new ioctl, since there is only 64 bytes reserved in the current struct. I am guessing this is not too likely to be accepted. (I would want to also add the same manufacturer field to the snd_ctl_card_info.) The next best solution would be to add a card field to snd_seq_client_info, which could be optionally populated with the card that is associated to the client. A completely different solution would be to add all this info to sysfs which is easier to extend dynamically and won't require any ioctl changes at all. (I kind of like this solution the best, but it would be a change from the ioctl-based mechanisms done so far.) Ultimately, if I am stuck guessing and parsing, that will work as long as the underlying implementation doesn't change. But I would rather have a more reliable solution going forward. Any thoughts? Thanks, Adam Goode

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[alsa-devel] [PATCH] ASoC: neo1973_wm8753: Automatically disconnected non-connected pins
by Lars-Peter Clausen 21 May '14

21 May '14

The DAPM routes for this board are complete, hence we can let the core take care of disconnecting non-connected pins rather than doing it manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/neo1973_wm8753.c | 8 +------- 1 file changed, 1 insertion(+), 7 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index b080033..9b4a09f 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -271,15 +271,8 @@ static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = rtd->card; - /* set up NC codec pins */ - snd_soc_dapm_nc_pin(&codec->dapm, "OUT3"); - snd_soc_dapm_nc_pin(&codec->dapm, "OUT4"); - snd_soc_dapm_nc_pin(&codec->dapm, "LINE1"); - snd_soc_dapm_nc_pin(&codec->dapm, "LINE2"); - /* set endpoints to default off mode */ snd_soc_dapm_disable_pin(&card->dapm, "GSM Line Out"); snd_soc_dapm_disable_pin(&card->dapm, "GSM Line In"); @@ -355,6 +348,7 @@ static struct snd_soc_card neo1973 = { .num_dapm_widgets = ARRAY_SIZE(neo1973_wm8753_dapm_widgets), .dapm_routes = neo1973_wm8753_routes, .num_dapm_routes = ARRAY_SIZE(neo1973_wm8753_routes), + .fully_routed = true, }; static struct platform_device *neo1973_snd_device; -- 1.8.0

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[alsa-devel] [PATCH] ASoC: samsung: Allow setting OP_CLK of the IIS Multi Audio Interface
by Sylwester Nawrocki 21 May '14

21 May '14

This patch adds support for setting source clock of the "Core CLK" of the IIS Multi Audio Interface. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 4 ++++ sound/soc/samsung/i2s.h | 1 + 2 files changed, 5 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 048ead9..ae02811 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -451,6 +451,10 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, u32 mod = readl(i2s->addr + I2SMOD); switch (clk_id) { + case SAMSUNG_I2S_OPCLK: + mod &= ~MOD_OPCLK_MASK; + mod |= dir; + break; case SAMSUNG_I2S_CDCLK: /* Shouldn't matter in GATING(CLOCK_IN) mode */ if (dir == SND_SOC_CLOCK_IN) diff --git a/sound/soc/samsung/i2s.h b/sound/soc/samsung/i2s.h index 7966afc..21ff24e 100644 --- a/sound/soc/samsung/i2s.h +++ b/sound/soc/samsung/i2s.h @@ -18,5 +18,6 @@ #define SAMSUNG_I2S_RCLKSRC_0 0 #define SAMSUNG_I2S_RCLKSRC_1 1 #define SAMSUNG_I2S_CDCLK 2 +#define SAMSUNG_I2S_OPCLK 3 #endif /* __SND_SOC_SAMSUNG_I2S_H */ -- 1.7.9.5

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[alsa-devel] [PATCH 15/15] ASoC: pxa: remove mach header dependency
by Xia Kaixu 21 May '14

21 May '14

From: Arnd Bergmann <arnd(a)arndb.de> As we are moving the mmp platform towards multiplatform support, we have to stop including platform header files. This changes the pxa-ssp sound driver file to no longer depend on mach/hardware.h and mach/dma.h. The code using the definitions from those headers is actually gone already, the only thing that was still being used was the pxa_dma_desc typedef, which we can easily work around by using the normal 'struct pxa_dma_desc' name. The pxa2xx-dma driver still uses this header, so we include it explicitly there, which is ok because that is only used on pxa, not on mmp. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Xia Kaixu <kaixu.xia(a)linaro.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Eric Miao <eric.y.miao(a)gmail.com> Cc: Russell King <linux(a)arm.linux.org.uk> Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org --- sound/arm/pxa2xx-pcm.c | 2 ++ sound/arm/pxa2xx-pcm.h | 3 +-- sound/soc/pxa/pxa-ssp.c | 2 -- sound/soc/pxa/pxa2xx-pcm.c | 2 ++ 4 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index e6c727b..83be8e3 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -14,6 +14,8 @@ #include <linux/dma-mapping.h> #include <linux/dmaengine.h> +#include <mach/dma.h> + #include <sound/core.h> #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h index 2a8fc08..0033098 100644 --- a/sound/arm/pxa2xx-pcm.h +++ b/sound/arm/pxa2xx-pcm.h @@ -9,12 +9,11 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ -#include <mach/dma.h> struct pxa2xx_runtime_data { int dma_ch; struct snd_dmaengine_dai_dma_data *params; - pxa_dma_desc *dma_desc_array; + struct pxa_dma_desc *dma_desc_array; dma_addr_t dma_desc_array_phys; }; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a3119a0..9b19ee7 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -34,8 +34,6 @@ #include <sound/pxa2xx-lib.h> #include <sound/dmaengine_pcm.h> -#include <mach/hardware.h> - #include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index d58b09f..42f2f01 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -15,6 +15,8 @@ #include <linux/dmaengine.h> #include <linux/of.h> +#include <mach/dma.h> + #include <sound/core.h> #include <sound/soc.h> #include <sound/pxa2xx-lib.h> -- 1.7.9.5

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[alsa-devel] [PATCH] Fix for possible null pointer dereference in dma.c
by Rickard Strandqvist 21 May '14

21 May '14

There is otherwise a risk of a possible null pointer dereference. Was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/samsung/dma.c | 10 ++++++---- 1 fil ändrad, 6 tillägg(+), 4 borttagningar(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index dc09b71..b1f6757 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -115,17 +115,19 @@ static void dma_enqueue(struct snd_pcm_substream *substream) static void audio_buffdone(void *data) { struct snd_pcm_substream *substream = data; - struct runtime_data *prtd = substream->runtime->private_data; + struct runtime_data *prtd = NULL; pr_debug("Entered %s\n", __func__); - if (prtd->state & ST_RUNNING) { + if(substream) + prtd = substream->runtime->private_data; + + if (prtd && prtd->state & ST_RUNNING) { prtd->dma_pos += prtd->dma_period; if (prtd->dma_pos >= prtd->dma_end) prtd->dma_pos = prtd->dma_start; - if (substream) - snd_pcm_period_elapsed(substream); + snd_pcm_period_elapsed(substream); spin_lock(&prtd->lock); if (!samsung_dma_has_circular()) { -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: ad1980: Replace goto loop with do-while loop
by Lars-Peter Clausen 21 May '14

21 May '14

Using a proper do-while loop here instead of a open-coded goto loop is both cleaner and shorter. Also fixes the following warnings from smatch: sound/soc/codecs/ad1980.c:213 ad1980_reset() info: loop could be replaced with if statement. sound/soc/codecs/ad1980.c:212 ad1980_reset() info: ignoring unreachable code. sound/soc/codecs/ad1980.c:215 ad1980_reset() info: ignoring unreachable code. While we are at it also change retry_cnt to unsigned int, using u16 for a on-stack loop counter doesn't make that much sense. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1980.c | 39 +++++++++++++++++++-------------------- 1 file changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 34d965a..304d300 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -189,28 +189,27 @@ static struct snd_soc_dai_driver ad1980_dai = { static int ad1980_reset(struct snd_soc_codec *codec, int try_warm) { - u16 retry_cnt = 0; + unsigned int retry_cnt = 0; -retry: - if (try_warm && soc_ac97_ops->warm_reset) { - soc_ac97_ops->warm_reset(codec->ac97); - if (ac97_read(codec, AC97_RESET) == 0x0090) - return 1; - } - - soc_ac97_ops->reset(codec->ac97); - /* Set bit 16slot in register 74h, then every slot will has only 16 - * bits. This command is sent out in 20bit mode, in which case the - * first nibble of data is eaten by the addr. (Tag is always 16 bit)*/ - ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900); - - if (ac97_read(codec, AC97_RESET) != 0x0090) - goto err; - return 0; + do { + if (try_warm && soc_ac97_ops->warm_reset) { + soc_ac97_ops->warm_reset(codec->ac97); + if (ac97_read(codec, AC97_RESET) == 0x0090) + return 1; + } -err: - while (retry_cnt++ < 10) - goto retry; + soc_ac97_ops->reset(codec->ac97); + /* + * Set bit 16slot in register 74h, then every slot will has only + * 16 bits. This command is sent out in 20bit mode, in which + * case the first nibble of data is eaten by the addr. (Tag is + * always 16 bit) + */ + ac97_write(codec, AC97_AD_SERIAL_CFG, 0x9900); + + if (ac97_read(codec, AC97_RESET) == 0x0090) + return 0; + } while (retry_cnt++ < 10); printk(KERN_ERR "AD1980 AC97 reset failed\n"); return -EIO; -- 1.8.0

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[alsa-devel] [PATCH] ALSA: hda/tegra - Fix MODULE_DEVICE_TABLE typo.
by Dylan Reid 20 May '14

20 May '14

I missed a rename during the review process. Fix the MODULE_DEVICE_TABLE to match the structure. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/hda_tegra.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index e472bc8..086fd06 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -496,7 +496,7 @@ static const struct of_device_id hda_tegra_match[] = { { .compatible = "nvidia,tegra30-hda" }, {}, }; -MODULE_DEVICE_TABLE(of, tegra_platform_hda_match); +MODULE_DEVICE_TABLE(of, hda_tegra_match); static int hda_tegra_probe(struct platform_device *pdev) { -- 1.8.1.3.605.g02339dd

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