Alsa-devel August 2015

alsa-devel@alsa-project.org

122 participants
313 discussions

[alsa-devel] [PATCH v2 1/3] ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format
by Lars-Peter Clausen 29 Jan '16

29 Jan '16

The DAI link format should be specified for the whole link rather than just one component on the link. So move the format specification for the HDMI audio link from the CPU component to the link itself. Since the sh-mobile-hdmi DAI driver doesn't implement the set_fmt() callback in this case there is no functional difference between only specifying the the format for the CPU side or for the whole link, but the later it will allow us to remove support for just specifying the format for one component. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes since v1. --- arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 6d949f1..75de26c 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -1040,9 +1040,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "sh_mobile_hdmi-hifi", -- 1.8.0

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[alsa-devel] [PATCH 0/4] ASoC: fix saif record unwork issue
by Dong Aisheng 16 Jan '16

16 Jan '16

This patch series mainly fixes the saif record unwork issue. The root cause consists of: 1) clk_prepare and clk_enable un-balanced 2) missed setting a proper initial clock rate for saif in dt 3) sgtl5000 dapm widget kcontrol does not work So can not test record on mx28evk. 4) missed setting clkmux mode for saif in dt With this patch series it can fix sgtl5000 widget kcontrol unwork and saif record unwork issue. BTW, just let people konw, i still met a few overruns during record, that's another issue and will be fixed in separte patches later. Patch based on sound-2.6/for-3.6. Dong Aisheng (4): ASoC: mxs-saif: fix clock prepare and enable unbalance issue ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe ARM: mx28evk: add missed clkmux select for saif arch/arm/mach-mxs/mach-mxs.c | 3 +++ sound/soc/codecs/sgtl5000.c | 2 -- sound/soc/mxs/mxs-saif.c | 24 ++++++++++++++++++++++++ 3 files changed, 27 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH 00/12] dw-hdmi development
by Russell King - ARM Linux 07 Jan '16

07 Jan '16

This sub-series is a mixture of development: * Removing the incorrect pixel repetition configuration code * Preventing pixel-doubled modes from being used * Adding interlaced video support * Implementing the sink_is_hdmi/sink_has_audio flags I suggested a few months ago * Only enabling audio support if the sink indicates it has audio * Avoiding double-enabling the HDMI interface * Fixing the mis-leading name of "dw_hdmi_phy_enable_power" * Adding connector mode forcing (important if your monitor bounces RXSENSE and HPD signals while in low-power mode.) * Improving the HDMI enable/disabling on sink status For review (and testing if people feel like it). Acks/tested-bys etc welcome. It applies on top of my drm-dwhdmi-devel branch, which is waiting for David Airlie to pull (see pull request on dri-devel, 15th July.) drivers/gpu/drm/bridge/dw_hdmi.c | 275 ++++++++++++++++++++++++++++++--------- drivers/gpu/ipu-v3/ipu-dc.c | 18 ++- drivers/gpu/ipu-v3/ipu-di.c | 129 +++++++++--------- 3 files changed, 291 insertions(+), 131 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

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[alsa-devel] [PATCH 1/2] ALSA: usb-audio: Add a volume scale quirk for AudioQuest DragonFly
by Anssi Hannula 14 Dec '15

14 Dec '15

AudioQuest DragonFly DAC reports a volume control range of 0..50 (0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which is obviously incorrect and causes software using the dB information in e.g. volume sliders to have a massive volume difference in 100..102% range. The actual volume mapping seems to be neither linear volume nor linear dB scale, but instead quite close to the cubic mapping e.g. alsamixer uses, with a range of -53...0 dB. Add a quirk for DragonFly to use a custom dB mapping, based on my measurements, using a 10-item range TLV (so it still fits in alsa-lib MAX_TLV_RANGE_SIZE). Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the range is 0..50, so if this gets fixed/changed in later HW revisions it will no longer be applied. Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> Cc: <stable(a)vger.kernel.org> --- sound/usb/mixer_quirks.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 337c317ead6f..39d7f34e44e6 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -37,6 +37,7 @@ #include <sound/control.h> #include <sound/hwdep.h> #include <sound/info.h> +#include <sound/tlv.h> #include "usbaudio.h" #include "mixer.h" @@ -1733,6 +1734,38 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer) return 0; } +static void snd_dragonfly_quirk_db_scale(struct usb_mixer_interface *mixer) +{ + struct usb_mixer_elem_list *list; + struct usb_mixer_elem_info *cval; + static const int unit_id = 7; + + /* approximation using 10 ranges based on output measurement on hw v1.2 */ + static const DECLARE_TLV_DB_RANGE(scale, + 0, 1, TLV_DB_MINMAX_ITEM(-5300, -4970), + 2, 5, TLV_DB_MINMAX_ITEM(-4710, -4160), + 6, 7, TLV_DB_MINMAX_ITEM(-3884, -3710), + 8, 14, TLV_DB_MINMAX_ITEM(-3443, -2560), + 15, 16, TLV_DB_MINMAX_ITEM(-2475, -2324), + 17, 19, TLV_DB_MINMAX_ITEM(-2228, -2031), + 20, 26, TLV_DB_MINMAX_ITEM(-1910, -1393), + 27, 31, TLV_DB_MINMAX_ITEM(-1322, -1032), + 32, 40, TLV_DB_MINMAX_ITEM(-968, -490), + 41, 50, TLV_DB_MINMAX_ITEM(-441, 0), + ); + + for (list = mixer->id_elems[unit_id]; list; list = list->next_id_elem) { + cval = (struct usb_mixer_elem_info *)list; + if (cval->control == UAC_FU_VOLUME && + cval->min == 0 && cval->max == 50) { + usb_audio_info(mixer->chip, "applying DragonFly dB scale quirk\n"); + list->kctl->tlv.p = scale; + list->kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ; + list->kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } + } +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1810,6 +1843,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x800c): /* Focusrite Scarlett 18i20 */ err = snd_scarlett_controls_create(mixer); break; + + case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ + snd_dragonfly_quirk_db_scale(mixer); + break; } return err; -- 2.3.8

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[alsa-devel] [PATCH 0/4 v5] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 27 Nov '15

27 Nov '15

Hopefully last version? :-) * Added commit text about duplicate events (patch 4/4) * Added locks in bind/unbind on i915 side (patch 2/4) * Fixed docbook comments in i915 struct (patch 1/4) * Removed port_mst_streaming parameter (patch 1/4) * Added Reviewed-by - 1 & 2 by Jani, 3 & 4 by Takashi. Hopefully this was correct, otherwise feel free to adjust yourself before committing. Both Jani and Takashi seem okay with this going into 4.3. Nobody has said which tree you prefer to take it through, so how about Takashi merging it? David Henningsson (4): drm/i915: Add audio pin sense / ELD callback drm/i915: Call audio pin/ELD notify function ALSA: hda - allow codecs to access the i915 pin/ELD callback ALSA: hda - Wake the codec up on pin/ELD notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 27 ++++++++++++++++++++++++--- include/drm/i915_component.h | 16 ++++++++++++++++ include/sound/hda_i915.h | 7 +++++++ sound/hda/hdac_i915.c | 10 ++++++++++ sound/pci/hda/patch_hdmi.c | 22 +++++++++++++++++++++- 6 files changed, 79 insertions(+), 4 deletions(-) -- 1.9.1

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[alsa-devel] [RFC PATCH 02/10] ASoC: Define soc_init_dai_link() to wrap link intialization.
by mengdong.lin＠intel.com 25 Nov '15

25 Nov '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Define soc_init_dai_link() to wrap link initialization, to reuse it later by snd_soc_instantiate_card() when adding new DAI links from topology in component probing phase. Move static func snd_soc_init_multicodec(), so that it can be reused by soc_init_dai_link(). This saves adding a function declaration for snd_soc_init_multicodec(). Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fb78674..e9e8956 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1141,6 +1141,100 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } } +static int snd_soc_init_multicodec(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + /* Legacy codec/codec_dai link is a single entry in multicodec */ + if (dai_link->codec_name || dai_link->codec_of_node || + dai_link->codec_dai_name) { + dai_link->num_codecs = 1; + + dai_link->codecs = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai_link_component), + GFP_KERNEL); + if (!dai_link->codecs) + return -ENOMEM; + + dai_link->codecs[0].name = dai_link->codec_name; + dai_link->codecs[0].of_node = dai_link->codec_of_node; + dai_link->codecs[0].dai_name = dai_link->codec_dai_name; + } + + if (!dai_link->codecs) { + dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); + return -EINVAL; + } + + return 0; +} + +static int soc_init_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *link) +{ + int i, ret; + + ret = snd_soc_init_multicodec(card, link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init multicodec\n"); + return ret; + } + + for (i = 0; i < link->num_codecs; i++) { + /* + * Codec must be specified by 1 of name or OF node, + * not both or neither. + */ + if (!!link->codecs[i].name == + !!link->codecs[i].of_node) { + dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* Codec DAI name must be specified */ + if (!link->codecs[i].dai_name) { + dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } + } + + /* + * Platform may be specified by either name or OF node, but + * can be left unspecified, and a dummy platform will be used. + */ + if (link->platform_name && link->platform_of_node) { + dev_err(card->dev, + "ASoC: Both platform name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + + /* + * CPU device may be specified by either name or OF node, but + * can be left unspecified, and will be matched based on DAI + * name alone.. + */ + if (link->cpu_name && link->cpu_of_node) { + dev_err(card->dev, + "ASoC: Neither/both cpu name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* + * At least one of CPU DAI name or CPU device name/node must be + * specified + */ + if (!link->cpu_dai_name && + !(link->cpu_name || link->cpu_of_node)) { + dev_err(card->dev, + "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + + return 0; +} + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { @@ -2363,33 +2457,6 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, } EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute); -static int snd_soc_init_multicodec(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_link) -{ - /* Legacy codec/codec_dai link is a single entry in multicodec */ - if (dai_link->codec_name || dai_link->codec_of_node || - dai_link->codec_dai_name) { - dai_link->num_codecs = 1; - - dai_link->codecs = devm_kzalloc(card->dev, - sizeof(struct snd_soc_dai_link_component), - GFP_KERNEL); - if (!dai_link->codecs) - return -ENOMEM; - - dai_link->codecs[0].name = dai_link->codec_name; - dai_link->codecs[0].of_node = dai_link->codec_of_node; - dai_link->codecs[0].dai_name = dai_link->codec_dai_name; - } - - if (!dai_link->codecs) { - dev_err(card->dev, "ASoC: DAI link has no CODECs\n"); - return -EINVAL; - } - - return 0; -} - /** * snd_soc_register_card - Register a card with the ASoC core * @@ -2398,7 +2465,7 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card, */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, j, ret; + int i, ret; struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) @@ -2407,63 +2474,11 @@ int snd_soc_register_card(struct snd_soc_card *card) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai_link *link = &card->dai_link[i]; - ret = snd_soc_init_multicodec(card, link); + ret = soc_init_dai_link(card, link); if (ret) { - dev_err(card->dev, "ASoC: failed to init multicodec\n"); - return ret; - } - - for (j = 0; j < link->num_codecs; j++) { - /* - * Codec must be specified by 1 of name or OF node, - * not both or neither. - */ - if (!!link->codecs[j].name == - !!link->codecs[j].of_node) { - dev_err(card->dev, "ASoC: Neither/both codec name/of_node are set for %s\n", - link->name); - return -EINVAL; - } - /* Codec DAI name must be specified */ - if (!link->codecs[j].dai_name) { - dev_err(card->dev, "ASoC: codec_dai_name not set for %s\n", - link->name); - return -EINVAL; - } - } - - /* - * Platform may be specified by either name or OF node, but - * can be left unspecified, and a dummy platform will be used. - */ - if (link->platform_name && link->platform_of_node) { - dev_err(card->dev, - "ASoC: Both platform name/of_node are set for %s\n", + dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); - return -EINVAL; - } - - /* - * CPU device may be specified by either name or OF node, but - * can be left unspecified, and will be matched based on DAI - * name alone.. - */ - if (link->cpu_name && link->cpu_of_node) { - dev_err(card->dev, - "ASoC: Neither/both cpu name/of_node are set for %s\n", - link->name); - return -EINVAL; - } - /* - * At least one of CPU DAI name or CPU device name/node must be - * specified - */ - if (!link->cpu_dai_name && - !(link->cpu_name || link->cpu_of_node)) { - dev_err(card->dev, - "ASoC: Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", - link->name); - return -EINVAL; + return ret; } } -- 1.9.1

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[alsa-devel] [PATCH v1 1/1] sst: replace custom implementation of readq / writeq
by Andy Shevchenko 24 Nov '15

24 Nov '15

The readq() and writeq() helpers are available in the asm-generic/io-64-nonatomic-hi-lo.h and asm-generic/io-64-nonatomic-lo-hi.h headers. Replace custom implementation by the generic helpers. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/atom/sst/sst_pvt.c | 16 ++++++---------- sound/soc/intel/common/sst-dsp.c | 9 ++++----- 2 files changed, 10 insertions(+), 15 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_pvt.c b/sound/soc/intel/atom/sst/sst_pvt.c index adb32fe..6cc222d 100644 --- a/sound/soc/intel/atom/sst/sst_pvt.c +++ b/sound/soc/intel/atom/sst/sst_pvt.c @@ -31,7 +31,10 @@ #include <sound/pcm.h> #include <sound/soc.h> #include <sound/compress_driver.h> + +#include <asm-generic/io-64-nonatomic-lo-hi.h> #include <asm/platform_sst_audio.h> + #include "../sst-mfld-platform.h" #include "sst.h" #include "../../common/sst-dsp.h" @@ -49,25 +52,18 @@ u32 sst_shim_read(void __iomem *addr, int offset) u64 sst_reg_read64(void __iomem *addr, int offset) { - u64 val = 0; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - - return val; + return readq(addr + offset); } int sst_shim_write64(void __iomem *addr, int offset, u64 value) { - memcpy_toio(addr + offset, &value, sizeof(value)); + writeq(value, addr + offset); return 0; } u64 sst_shim_read64(void __iomem *addr, int offset) { - u64 val = 0; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - return val; + return readq(addr + offset); } void sst_set_fw_state_locked( diff --git a/sound/soc/intel/common/sst-dsp.c b/sound/soc/intel/common/sst-dsp.c index a627236..5274482 100644 --- a/sound/soc/intel/common/sst-dsp.c +++ b/sound/soc/intel/common/sst-dsp.c @@ -22,6 +22,8 @@ #include <linux/io.h> #include <linux/delay.h> +#include <asm-generic/io-64-nonatomic-lo-hi.h> + #include "sst-dsp.h" #include "sst-dsp-priv.h" @@ -43,16 +45,13 @@ EXPORT_SYMBOL_GPL(sst_shim32_read); void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value) { - memcpy_toio(addr + offset, &value, sizeof(value)); + writeq(value, addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_write64); u64 sst_shim32_read64(void __iomem *addr, u32 offset) { - u64 val; - - memcpy_fromio(&val, addr + offset, sizeof(val)); - return val; + return readq(addr + offset); } EXPORT_SYMBOL_GPL(sst_shim32_read64); -- 2.5.0

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[alsa-devel] [PATCH 00/13] dmaengine + omap drivers: support fro deferred probing
by Peter Ujfalusi 20 Nov '15

20 Nov '15

Hi, Vinod: as I promised: https://lkml.org/lkml/2015/5/8/80 With this series it is possible to put omap-dma or edma to even late_initcall and the drivers will figure out the load order fine(ish). We need to add dma_request_slave_channel_compat_reason() which is the equivalent of dma_request_slave_channel_compat() but returning with error codes in case of failure instead of NULL pointer. The rest of the series just converts the OMAP/daVinci drivers to use this new function to get the channel(s) and to handle the deferred probing. I did not moved the omap-dma, edma or ti-dma-crossbar from arch_initcall. If I do so UART will only comes up after the DMA driver is loaded since we are using, or going to use 8250 with DAM support. This delays the kernel messages. Other issue is the MMC/SD cards. On board with eMMC and SD card slot the mmcblk0/1 might get swapped due to different probe order for the MMC/SD drivers. For example in omap5-uevm: 1. omap-dma in arch_initcall the SD card is mmcblk1 (4809c000.mmc) and eMMC is mmcblk0 (480b4000.mmc) 2. omap-dma in late_initcall the SD card is mmcblk0 (4809c000.mmc) and eMMC is mmcblk1 (480b4000.mmc) Because in case 1 the 4809c000.mmc got deferred by missing regulator so 480b4000.mmc got registered first. In case 2 both deferring because of DMA and at the end 4809c000.mmc get registered first. So far I have not found a way to bind mmcblk0/1 to a specific node... Regards, Peter --- Peter Ujfalusi (13): dmaengine: of_dma: Correct return code for of_dma_request_slave_channel in case !CONFIG_OF dmaengine: Introduce dma_request_slave_channel_compat_reason() serial: 8250_dma: Support for deferred probing when requesting DMA channels mmc: omap_hsmmc: No need to check DMA channel validity at module remove mmc: omap_hsmmc: Support for deferred probing when requesting DMA channels mmc: omap: Support for deferred probing when requesting DMA channels mmc: davinci_mmc: Support for deferred probing when requesting DMA channels crypto: omap-aes - Support for deferred probing when requesting DMA channels crypto: omap-des - Support for deferred probing when requesting DMA channels crypto: omap-sham - Support for deferred probing when requesting DMA channel spi: omap2-mcspi: Support for deferred probing when requesting DMA channels [media] omap3isp: Support for deferred probing when requesting DMA channel ASoC: omap-pcm: Switch to use dma_request_slave_channel_compat_reason() drivers/crypto/omap-aes.c | 38 ++++++++++++++++--------------- drivers/crypto/omap-des.c | 38 ++++++++++++++++--------------- drivers/crypto/omap-sham.c | 15 ++++++++---- drivers/media/platform/omap3isp/isphist.c | 12 +++++++--- drivers/mmc/host/davinci_mmc.c | 26 ++++++++++++--------- drivers/mmc/host/omap.c | 20 ++++++++++++---- drivers/mmc/host/omap_hsmmc.c | 28 ++++++++++------------- drivers/spi/spi-omap2-mcspi.c | 36 +++++++++++++++++------------ drivers/tty/serial/8250/8250_dma.c | 18 +++++++-------- include/linux/dmaengine.h | 22 ++++++++++++++++++ include/linux/of_dma.h | 2 +- sound/soc/omap/omap-pcm.c | 16 ++++++++----- 12 files changed, 164 insertions(+), 107 deletions(-) -- 2.3.5

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[alsa-devel] [PATCH] ARM: sun7i: Fix HDMI Audio driver's Alsa interface and do 8CH IEC60958 output
by thomas schorpp 13 Nov '15

13 Nov '15

Version 0.2 Users doc: All You need to get a HDMI- Pulseaudio Server on Your sunxi board to output Multichannel PCM is this line in default.pa, ttable pcms in asound.conf get ignored: +load-module module-alsa-sink device="hw:1,0" channels=8 channel_map="front-left,front-right,lfe,front-center,rear-left,rear-right,side-left,side-right" ### Automatically load driver modules depending on the hardware available .ifexists module-udev-detect.so load-module module-udev-detect Probe with speaker-test, You may need other channel mapping than I. If you run PA in system mode please read the security documentation before, see PA log for link. y tom

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[alsa-devel] [PATCH] ASoC: rt5677: handle reset_pin logical state correctly
by Anatol Pomozov 29 Oct '15

29 Oct '15

According to the datasheet RESET is active low pin, i.e. system goes to reset state when pin is low. Handle logic state correctly - set reset_pin to logical 0 at boot time, and set it to logical 1 when we need to reset the chip. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- sound/soc/codecs/rt5677.c | 8 ++++---- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index f070789..1b3c13d 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -18,7 +18,7 @@ Required properties: Optional properties: - realtek,pow-ldo2-gpio : The GPIO that controls the CODEC's POW_LDO2 pin. -- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. +- realtek,reset-gpio : The GPIO that controls the CODEC's RESET pin. Active low. - realtek,in1-differential - realtek,in2-differential diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 3b46b35..433652f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4766,7 +4766,7 @@ static int rt5677_remove(struct snd_soc_codec *codec) if (rt5677->pow_ldo2) gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); return 0; } @@ -4783,7 +4783,7 @@ static int rt5677_suspend(struct snd_soc_codec *codec) if (rt5677->pow_ldo2) gpiod_set_value_cansleep(rt5677->pow_ldo2, 0); if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 0); + gpiod_set_value_cansleep(rt5677->reset_pin, 1); } return 0; @@ -4797,7 +4797,7 @@ static int rt5677_resume(struct snd_soc_codec *codec) if (rt5677->pow_ldo2) gpiod_set_value_cansleep(rt5677->pow_ldo2, 1); if (rt5677->reset_pin) - gpiod_set_value_cansleep(rt5677->reset_pin, 1); + gpiod_set_value_cansleep(rt5677->reset_pin, 0); if (rt5677->pow_ldo2 || rt5677->reset_pin) msleep(10); @@ -5142,7 +5142,7 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, rt5677->pow_ldo2 = 0; } rt5677->reset_pin = devm_gpiod_get_optional(&i2c->dev, - "realtek,reset", GPIOD_OUT_HIGH); + "realtek,reset", GPIOD_OUT_LOW); if (IS_ERR(rt5677->reset_pin)) { ret = PTR_ERR(rt5677->reset_pin); dev_err(&i2c->dev, "Failed to request RESET: %d\n", ret); -- 2.4.3.573.g4eafbef

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Alsa-devel August 2015