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

[alsa-devel] [PATCH] ASoC: arizona: Add default case to silence build warning
by Charles Keepax 04 Jul '13

04 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index eb5c80d..bcf739d 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -233,6 +233,9 @@ int arizona_init_gpio(struct snd_soc_codec *codec) switch (arizona->type) { case WM5110: snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity"); + break; + default: + break; } snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity"); -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: wm8962: fix NULL pdata pointer
by Shawn Guo 04 Jul '13

04 Jul '13

There is an error in merge commit 384b834 on conflict resolution which causes the following NULL pdata pointer bug. wm8962 0-001a: customer id 0 revision D Unable to handle kernel NULL pointer dereference at virtual address 00000004 pgd = 80004000 [00000004] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.10.0+ #1 task: bf870000 ti: bf874000 task.ti: bf874000 PC is at wm8962_probe+0x134/0x6c8 LR is at regmap_unlock_mutex+0x10/0x14 pc : [<80452100>] lr : [<80304cf4>] psr: a0000113 sp : bf875c98 ip : 00000000 fp : bf875cd4 r10: 00000000 r9 : bfb1830c r8 : 80779bc4 r7 : 00000000 r6 : 00000001 r5 : bfbac010 r4 : bfb33e00 r3 : 80304ce4 r2 : 00000000 r1 : 00000001 r0 : fffffffb Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c53c7d Table: 1000404a DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0xbf874238) Stack: (0xbf875c98 to 0xbf876000) ... Fix the error by assigning pdata a correct pointer. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b1dc7d4..e2de9ec 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_pdata *pdata = &wm8962->pdata; int i, trigger, irq_pol; bool dmicclk, dmicdat; -- 1.7.9.5

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[alsa-devel] [PATCH 00/11] HDSPM: RME AIO/RayDAT extensions and cleanups
by Adrian Knoth 04 Jul '13

04 Jul '13

Hi! Here's a series of patches that most importantly add support for AIO's Analog Expansion Boards (AEBs) and all the missing controls to adjust card settings (output levels, optical/coaxial etc.). I also refactored a couple of functions to clean up the code and fixed the S/PDIF sync/frequency reporting. Cheers Adrian Knoth (11): ALSA: hdspm - Add missing defines for RME AIO and RayDAT ALSA: hdspm - Introduce hdspm_is_raydat_or_aio() ALSA: hdspm - Augment HDSPM_TOGGLE_SETTING for AIO/RayDAT ALSA: hdspm - Drop duplicate code in hdspm_set_system_clock_mode() ALSA: hdspm - Add S/PDIF and WCK48 controls for RME RayDAT ALSA: hdspm - Add S/PDIF, XLR, WCK48 and ADAT-in controls for RME AIO cards ALSA: hdspm - Refactor ENUMERATED_CTL_INFO into function ALSA: hdspm - Introduce generic AIO tristate control ALSA: hdspm - Enable AD/DA/PH gains and S/PDIF-Input select on AIO ALSA: hdspm - Add support for AEBs on RME AIO ALSA: hdspm - Fix S/PDIF Sync status and frequency on RME AIO sound/pci/rme9652/hdspm.c | 254 ++++++++++++++++++++++++++++++++++++---------- 1 file changed, 199 insertions(+), 55 deletions(-) -- 1.8.3.1

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[alsa-devel] [RFC] [patch 1/1] imx-sgtl5000: return E_PROBE_DEFER if ssi/codec not found
by Arnaud Patard 04 Jul '13

04 Jul '13

If the ssi or codec drivers are not loaded (for instance, because spi or i2c bus drivers are not loaded), returning -EINVAL will for people to unload and then reload the module to get sound working. Returning E_PROBE_DEFER will mitigate this. Signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org> Index: linux/sound/soc/fsl/imx-sgtl5000.c =================================================================== --- linux.orig/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:33:01.000000000 +0200 +++ linux/sound/soc/fsl/imx-sgtl5000.c 2013-05-16 09:35:14.000000000 +0200 @@ -113,13 +113,13 @@ static int imx_sgtl5000_probe(struct pla ssi_pdev = of_find_device_by_node(ssi_np); if (!ssi_pdev) { dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto fail; } codec_dev = of_find_i2c_device_by_node(codec_np); if (!codec_dev) { dev_err(&pdev->dev, "failed to find codec platform device\n"); - return -EINVAL; + return -EPROBE_DEFER; } data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);

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[alsa-devel] [PATCH 10/11] ALSA: hdspm - Add support for AEBs on RME AIO
by Adrian Knoth 04 Jul '13

04 Jul '13

AIO cards allow to use AEB (Analogue Expansion Boards) to add four input and/or output channels. This patch adds the necessary code to detect and enable the additional I/O channels. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 111713c..585397b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -648,7 +648,8 @@ static char *texts_ports_aio_in_ss[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", - "ADAT.7", "ADAT.8" + "ADAT.7", "ADAT.8", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ss[] = { @@ -657,14 +658,16 @@ static char *texts_ports_aio_out_ss[] = { "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", "ADAT.7", "ADAT.8", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_ds[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ds[] = { @@ -672,14 +675,16 @@ static char *texts_ports_aio_out_ds[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_qs[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_qs[] = { @@ -687,7 +692,8 @@ static char *texts_ports_aio_out_qs[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aes32[] = { @@ -764,8 +770,8 @@ static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in, */ 10, 11, /* spdif in */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */ - -1, -1, - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -779,7 +785,8 @@ static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -792,7 +799,8 @@ static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 14, 16, 18, /* adat in */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -807,7 +815,7 @@ static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 14, 16, 18, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -821,7 +829,8 @@ static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 16, /* adat in */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -836,7 +845,8 @@ static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 16, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -6602,10 +6612,6 @@ static int snd_hdspm_create(struct snd_card *card, break; case AIO: - if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { - snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n"); - } - hdspm->ss_in_channels = AIO_IN_SS_CHANNELS; hdspm->ds_in_channels = AIO_IN_DS_CHANNELS; hdspm->qs_in_channels = AIO_IN_QS_CHANNELS; @@ -6613,6 +6619,20 @@ static int snd_hdspm_create(struct snd_card *card, hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS; hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS; + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { + snd_printk(KERN_INFO "HDSPM: AEB input board found\n"); + hdspm->ss_in_channels += 4; + hdspm->ds_in_channels += 4; + hdspm->qs_in_channels += 4; + } + + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) { + snd_printk(KERN_INFO "HDSPM: AEB output board found\n"); + hdspm->ss_out_channels += 4; + hdspm->ds_out_channels += 4; + hdspm->qs_out_channels += 4; + } + hdspm->channel_map_out_ss = channel_map_aio_out_ss; hdspm->channel_map_out_ds = channel_map_aio_out_ds; hdspm->channel_map_out_qs = channel_map_aio_out_qs; -- 1.8.3.1

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[alsa-devel] [PATCH 1/1] ASoC: Samsung: Remove redundant comment
by Sachin Kamat 04 Jul '13

04 Jul '13

There is a typo in the filename (i2c mentioned instead of i2s). However, this is a redundant piece of information. Delete it altogether. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/s3c-i2s-v2.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 20e98d1..e5e81b1 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -1,6 +1,4 @@ -/* sound/soc/samsung/s3c-i2c-v2.c - * - * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. +/* ALSA Soc Audio Layer - I2S core for newer Samsung SoCs. * * Copyright (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory(a)wolfsonmicro.com -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: add ak4554 driver
by Kuninori Morimoto 04 Jul '13

04 Jul '13

ak4554 is very simple DA/AD converter which has no setting register. But, playback format is SND_SOC_DAIFMT_RIGHT_J, and, capture format is SND_SOC_DAIFMT_LEFT_J on same bit clock, LR clock. Because of that, snd_soc_dai_driver consists from "playback only" and "capture only" here, and it has software base symmetric_rates check. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4554.c | 171 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 176 insertions(+) create mode 100644 sound/soc/codecs/ak4554.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..b4ec4ea 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -196,6 +196,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4554 + tristate + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..e601a47 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4554-objs := ak4554.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -136,6 +137,7 @@ obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c new file mode 100644 index 0000000..277f216 --- /dev/null +++ b/sound/soc/codecs/ak4554.c @@ -0,0 +1,171 @@ +/* + * ak4554.c + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + * + * 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 <sound/soc.h> + +/* + * ak4554 is very simple DA/AD converter which has no setting register. + * But, playback format is SND_SOC_DAIFMT_RIGHT_J, + * and, capture format is SND_SOC_DAIFMT_LEFT_J + * on same bit clock, LR clock. + * Because of that, snd_soc_dai_driver consists from + * "playback only" and "capture only" here. + */ + +struct ak4554_priv { + int rate; + int usrcnt; +}; + +static int ak4554_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + int dai_is_play = (dai->id == 0); + int fmt_is_play = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + dev_err(dai->dev, "can't be clock master\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + fmt_is_play = 1; + break; + case SND_SOC_DAIFMT_LEFT_J: + fmt_is_play = 0; + break; + } + + if (dai_is_play != fmt_is_play) { + dev_err(dai->dev, "format error\n"); + return -EINVAL; + } + + return 0; +} + +static int ak4554_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct ak4554_priv *priv = dev_get_drvdata(dai->dev); + + if ((priv->usrcnt > 0) && + (priv->rate != params_rate(params))) { + dev_err(dai->dev, "asymmetric rate\n"); + return -EIO; + } + + priv->usrcnt++; + priv->rate = params_rate(params); + + return 0; +} + +static void ak4554_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct ak4554_priv *priv = dev_get_drvdata(dai->dev); + + priv->usrcnt--; + + if (priv->usrcnt <= 0) + priv->rate = 0; +} + +static const struct snd_soc_dai_ops ak4554_dai_ops = { + .set_fmt = ak4554_dai_set_fmt, + .hw_params = ak4554_dai_hw_params, + .shutdown = ak4554_dai_shutdown, +}; + +struct snd_soc_dai_driver ak4554_dai[] = { + { + .name = "ak4554-hifi-playback", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &ak4554_dai_ops, + }, { + .name = "ak4554-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &ak4554_dai_ops, + } +}; +EXPORT_SYMBOL_GPL(ak4554_dai); + +static int ak4554_probe(struct snd_soc_codec *codec) +{ + dev_info(codec->dev, "probed\n"); + return 0; +} + +static int ak4554_remove(struct snd_soc_codec *codec) +{ + dev_info(codec->dev, "removed\n"); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { + .probe = ak4554_probe, + .remove = ak4554_remove, +}; + +static int ak4554_soc_probe(struct platform_device *pdev) +{ + struct ak4554_priv *priv; + + priv = devm_kzalloc(&pdev->dev, sizeof(struct ak4554_priv), + GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_ak4554, + ak4554_dai, ARRAY_SIZE(ak4554_dai)); +} + +static int ak4554_soc_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ak4554_driver = { + .driver = { + .name = "ak4554-adc-dac", + .owner = THIS_MODULE, + }, + .probe = ak4554_soc_probe, + .remove = ak4554_soc_remove, +}; +module_platform_driver(ak4554_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SoC AK4554 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 1.7.9.5

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[alsa-devel] [PATCH 0/2] HDSPM - Documentation and TCO on AES(32)
by Adrian Knoth 04 Jul '13

04 Jul '13

Hi! Here are two patches that I've received from Martin. The first only adds comments to the driver to document the various register settings. Might be useful for other developers. The second patch adds TCO support on AES(32) cards. I took the liberty to polish Martin's work to reflect the kernel coding style. It still contains plenty of long-line warnings from checkpatch.pl, but I could live with this. There are also a few (five?) whitespace fixes and a typo fix which are technically not related to TCO support. I could manually separate them into two commits, but maybe it's acceptable as it is? Cheers Adrian Knoth (1): ALSA: hdspm - Add TCO support for RME AES(32) Martin Dausel (1): ALSA: hdspm - Added some comments and control register documentation sound/pci/rme9652/hdspm.c | 438 +++++++++++++++++++++++++++++++++------------- 1 file changed, 317 insertions(+), 121 deletions(-) -- 1.8.3.1

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[alsa-devel] [PATCH v3 1/2] mfd: arizona: Add GPIO control register bit definitions
by Charles Keepax 04 Jul '13

04 Jul '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- include/linux/mfd/arizona/gpio.h | 96 ++++++++++++++++++++++++++++++++++++++ 1 files changed, 96 insertions(+), 0 deletions(-) create mode 100644 include/linux/mfd/arizona/gpio.h diff --git a/include/linux/mfd/arizona/gpio.h b/include/linux/mfd/arizona/gpio.h new file mode 100644 index 0000000..d2146bb --- /dev/null +++ b/include/linux/mfd/arizona/gpio.h @@ -0,0 +1,96 @@ +/* + * GPIO configuration for Arizona devices + * + * Copyright 2013 Wolfson Microelectronics. PLC. + * + * Author: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> + * + * 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. + */ + +#ifndef _ARIZONA_GPIO_H +#define _ARIZONA_GPIO_H + +#define ARIZONA_GP_FN_TXLRCLK 0x00 +#define ARIZONA_GP_FN_GPIO 0x01 +#define ARIZONA_GP_FN_IRQ1 0x02 +#define ARIZONA_GP_FN_IRQ2 0x03 +#define ARIZONA_GP_FN_OPCLK 0x04 +#define ARIZONA_GP_FN_FLL1_OUT 0x05 +#define ARIZONA_GP_FN_FLL2_OUT 0x06 +#define ARIZONA_GP_FN_PWM1 0x08 +#define ARIZONA_GP_FN_PWM2 0x09 +#define ARIZONA_GP_FN_SYSCLK_UNDERCLOCKED 0x0A +#define ARIZONA_GP_FN_ASYNCCLK_UNDERCLOCKED 0x0B +#define ARIZONA_GP_FN_FLL1_LOCK 0x0C +#define ARIZONA_GP_FN_FLL2_LOCK 0x0D +#define ARIZONA_GP_FN_FLL1_CLOCK_OK 0x0F +#define ARIZONA_GP_FN_FLL2_CLOCK_OK 0x10 +#define ARIZONA_GP_FN_HEADPHONE_DET 0x12 +#define ARIZONA_GP_FN_MIC_DET 0x13 +#define ARIZONA_GP_FN_WSEQ_STATUS 0x15 +#define ARIZONA_GP_FN_CIF_ADDRESS_ERROR 0x16 +#define ARIZONA_GP_FN_ASRC1_LOCK 0x1A +#define ARIZONA_GP_FN_ASRC2_LOCK 0x1B +#define ARIZONA_GP_FN_ASRC_CONFIG_ERROR 0x1C +#define ARIZONA_GP_FN_DRC1_SIGNAL_DETECT 0x1D +#define ARIZONA_GP_FN_DRC1_ANTICLIP 0x1E +#define ARIZONA_GP_FN_DRC1_DECAY 0x1F +#define ARIZONA_GP_FN_DRC1_NOISE 0x20 +#define ARIZONA_GP_FN_DRC1_QUICK_RELEASE 0x21 +#define ARIZONA_GP_FN_DRC2_SIGNAL_DETECT 0x22 +#define ARIZONA_GP_FN_DRC2_ANTICLIP 0x23 +#define ARIZONA_GP_FN_DRC2_DECAY 0x24 +#define ARIZONA_GP_FN_DRC2_NOISE 0x25 +#define ARIZONA_GP_FN_DRC2_QUICK_RELEASE 0x26 +#define ARIZONA_GP_FN_MIXER_DROPPED_SAMPLE 0x27 +#define ARIZONA_GP_FN_AIF1_CONFIG_ERROR 0x28 +#define ARIZONA_GP_FN_AIF2_CONFIG_ERROR 0x29 +#define ARIZONA_GP_FN_AIF3_CONFIG_ERROR 0x2A +#define ARIZONA_GP_FN_SPK_TEMP_SHUTDOWN 0x2B +#define ARIZONA_GP_FN_SPK_TEMP_WARNING 0x2C +#define ARIZONA_GP_FN_UNDERCLOCKED 0x2D +#define ARIZONA_GP_FN_OVERCLOCKED 0x2E +#define ARIZONA_GP_FN_DSP_IRQ1 0x35 +#define ARIZONA_GP_FN_DSP_IRQ2 0x36 +#define ARIZONA_GP_FN_ASYNC_OPCLK 0x3D +#define ARIZONA_GP_FN_BOOT_DONE 0x44 +#define ARIZONA_GP_FN_DSP1_RAM_READY 0x45 +#define ARIZONA_GP_FN_SYSCLK_ENA_STATUS 0x4B +#define ARIZONA_GP_FN_ASYNCCLK_ENA_STATUS 0x4C + +#define ARIZONA_GPN_DIR 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_MASK 0x8000 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_SHIFT 15 /* GPN_DIR */ +#define ARIZONA_GPN_DIR_WIDTH 1 /* GPN_DIR */ +#define ARIZONA_GPN_PU 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_MASK 0x4000 /* GPN_PU */ +#define ARIZONA_GPN_PU_SHIFT 14 /* GPN_PU */ +#define ARIZONA_GPN_PU_WIDTH 1 /* GPN_PU */ +#define ARIZONA_GPN_PD 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_MASK 0x2000 /* GPN_PD */ +#define ARIZONA_GPN_PD_SHIFT 13 /* GPN_PD */ +#define ARIZONA_GPN_PD_WIDTH 1 /* GPN_PD */ +#define ARIZONA_GPN_LVL 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_MASK 0x0800 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_SHIFT 11 /* GPN_LVL */ +#define ARIZONA_GPN_LVL_WIDTH 1 /* GPN_LVL */ +#define ARIZONA_GPN_POL 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_MASK 0x0400 /* GPN_POL */ +#define ARIZONA_GPN_POL_SHIFT 10 /* GPN_POL */ +#define ARIZONA_GPN_POL_WIDTH 1 /* GPN_POL */ +#define ARIZONA_GPN_OP_CFG 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_MASK 0x0200 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_SHIFT 9 /* GPN_OP_CFG */ +#define ARIZONA_GPN_OP_CFG_WIDTH 1 /* GPN_OP_CFG */ +#define ARIZONA_GPN_DB 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_MASK 0x0100 /* GPN_DB */ +#define ARIZONA_GPN_DB_SHIFT 8 /* GPN_DB */ +#define ARIZONA_GPN_DB_WIDTH 1 /* GPN_DB */ +#define ARIZONA_GPN_FN_MASK 0x007F /* GPN_DB */ +#define ARIZONA_GPN_FN_SHIFT 0 /* GPN_DB */ +#define ARIZONA_GPN_FN_WIDTH 7 /* GPN_DB */ + +#endif -- 1.7.2.5

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[alsa-devel] ASoC: wm8962: NULL pdata pointer
by Shawn Guo 04 Jul '13

04 Jul '13

Mark, I'm running next-20130703 and seeing the following NULL pointer bug. Unable to handle kernel NULL pointer dereference at virtual address 00000004 pgd = 80004000 [00000004] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.10.0-next-20130703 #6 task: bf870000 ti: bf874000 task.ti: bf874000 PC is at wm8962_probe+0x140/0x6b4 ... It seems to me that it's caused by a merge error in 384b834. And the change below can remove the problem. Shawn ---8<---- diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index b1dc7d4..e2de9ec 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3377,7 +3377,7 @@ static int wm8962_probe(struct snd_soc_codec *codec) { int ret; struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); - struct wm8962_pdata *pdata = dev_get_platdata(codec->dev); + struct wm8962_pdata *pdata = &wm8962->pdata; int i, trigger, irq_pol; bool dmicclk, dmicdat;

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