September 2015 - Alsa-devel - mailman.alsa-project.org

[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

2 1

[alsa-devel] [PATCH V1 0/3] fsl esai/spdif/asrc need spba clock
by Shengjiu Wang 04 Nov '15

04 Nov '15

fsl esai/spdif/asrc need spba clock Shengjiu Wang (3): ASoC: fsl_esai: spba clock is needed by esai device ASoC: fsl_spdif: spba clk is needed by spdif device ASoC: fsl_asrc: spba clock is needed by asrc device Documentation/devicetree/bindings/sound/fsl,asrc.txt | 2 ++ Documentation/devicetree/bindings/sound/fsl,esai.txt | 2 ++ Documentation/devicetree/bindings/sound/fsl,spdif.txt | 2 ++ sound/soc/fsl/fsl_asrc.c | 12 ++++++++++++ sound/soc/fsl/fsl_asrc.h | 1 + sound/soc/fsl/fsl_esai.c | 13 +++++++++++++ sound/soc/fsl/fsl_spdif.c | 15 +++++++++++++++ 7 files changed, 47 insertions(+) -- 1.9.1

4 12

[alsa-devel] [PATCH 00/10] Add support for Imagination Technologies audio controllers
by Damien Horsley 04 Nov '15

04 Nov '15

From: "Damien.Horsley" <Damien.Horsley(a)imgtec.com> Add drivers and binding documents for the following Imagination Technologies audio controllers: I2S Input Controller I2S Output Controller Parallel Output Controller SPDIF Input Controller SPDIF Output Controller These controllers are used in Pistachio SoC Damien.Horsley (10): ASoC: img: Add binding document for I2S input controller ASoC: img: Add driver for I2S input controller ASoC: img: Add binding document for I2S output controller ASoC: img: Add driver for I2S output controller ASoC: img: Add binding document for parallel output controller ASoC: img: Add driver for parallel output controller ASoC: img: Add binding document for SPDIF input controller ASoC: img: Add driver for SPDIF input controller ASoC: img: Add binding document for SPDIF output controller ASoC: img: Add driver for SPDIF output controller .../devicetree/bindings/sound/img,i2s-in.txt | 45 ++ .../devicetree/bindings/sound/img,i2s-out.txt | 49 ++ .../devicetree/bindings/sound/img,parallel-out.txt | 44 ++ .../devicetree/bindings/sound/img,spdif-in.txt | 41 ++ .../devicetree/bindings/sound/img,spdif-out.txt | 44 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/img/Kconfig | 44 ++ sound/soc/img/Makefile | 5 + sound/soc/img/img-i2s-in.c | 506 +++++++++++++ sound/soc/img/img-i2s-out.c | 555 ++++++++++++++ sound/soc/img/img-parallel-out.c | 372 ++++++++++ sound/soc/img/img-spdif-in.c | 797 +++++++++++++++++++++ sound/soc/img/img-spdif-out.c | 432 +++++++++++ 14 files changed, 2936 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,i2s-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,parallel-out.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-in.txt create mode 100644 Documentation/devicetree/bindings/sound/img,spdif-out.txt create mode 100644 sound/soc/img/Kconfig create mode 100644 sound/soc/img/Makefile create mode 100644 sound/soc/img/img-i2s-in.c create mode 100644 sound/soc/img/img-i2s-out.c create mode 100644 sound/soc/img/img-parallel-out.c create mode 100644 sound/soc/img/img-spdif-in.c create mode 100644 sound/soc/img/img-spdif-out.c -- 2.1.4

3 15

[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

2 3

[alsa-devel] [PATCH v2- ALSA 1/3] HDA - Allow hda_jack_tbl to use another nid for pin sense
by Raymond Yau 28 Oct '15

28 Oct '15

- add sense_nid to __snd_hda_jack_add_kctl for pin sense - update slave pin sense when master pin sense change - enable unsolicited event when widcap support UNSOL Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 366efbf..d81d65f 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -19,6 +19,7 @@ #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" +#include "hda_generic.h" /** * is_jack_detectable - Check whether the given pin is jack-detectable @@ -157,7 +158,14 @@ static void jack_detect_update(struct hda_codec *codec, if (jack->phantom_jack) jack->pin_sense = AC_PINSENSE_PRESENCE; else - jack->pin_sense = read_pin_sense(codec, jack->nid); + jack->pin_sense = read_pin_sense(codec, + jack->sense_nid ? jack->sense_nid : jack->nid); + + if (jack->slave_nid) { + struct hda_jack_tbl *slave = snd_hda_jack_tbl_get(codec, + jack->slave_nid); + slave->pin_sense = jack->pin_sense; + } /* A gating jack indicates the jack is invalid if gating is unplugged */ if (jack->gating_jack && !snd_hda_jack_detect(codec, jack->gating_jack)) @@ -269,11 +277,13 @@ snd_hda_jack_detect_enable_callback(struct hda_codec *codec, hda_nid_t nid, jack->jack_detect = 1; if (codec->jackpoll_interval > 0) return callback; /* No unsol if we're polling instead */ - err = snd_hda_codec_write_cache(codec, nid, 0, + if (is_jack_detectable(codec, nid)) { + err = snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | jack->tag); - if (err < 0) - return ERR_PTR(err); + if (err < 0) + return ERR_PTR(err); + } return callback; } EXPORT_SYMBOL_GPL(snd_hda_jack_detect_enable_callback); @@ -384,7 +394,7 @@ static void hda_free_jack_priv(struct snd_jack *jack) * will have the given name and index. */ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, - const char *name, bool phantom_jack) + const char *name, bool phantom_jack, hda_nid_t sense_nid) { struct hda_jack_tbl *jack; int err, state, type; @@ -403,6 +413,7 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, jack->phantom_jack = !!phantom_jack; jack->type = type; + jack->sense_nid = sense_nid; jack->jack->private_data = jack; jack->jack->private_free = hda_free_jack_priv; state = snd_hda_jack_detect(codec, nid); @@ -422,7 +433,7 @@ static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid, const char *name) { - return __snd_hda_jack_add_kctl(codec, nid, name, false); + return __snd_hda_jack_add_kctl(codec, nid, name, false, nid); } EXPORT_SYMBOL_GPL(snd_hda_jack_add_kctl); @@ -451,7 +462,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, if (phantom_jack) /* Example final name: "Internal Mic Phantom Jack" */ strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); - err = __snd_hda_jack_add_kctl(codec, nid, name, phantom_jack); + err = __snd_hda_jack_add_kctl(codec, nid, name, phantom_jack, nid); if (err < 0) return err; diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index 387d309..ef3d162 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -41,6 +41,8 @@ struct hda_jack_tbl { hda_nid_t gated_jack; /* gated is dependent on this jack */ int type; struct snd_jack *jack; + hda_nid_t sense_nid; + hda_nid_t slave_nid; }; struct hda_jack_tbl * -- 2.5.0

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[alsa-devel] [PATCH v2 0/4] Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 26 Oct '15

26 Oct '15

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver as present on the A10, A20 and A31 SoC boards. For now just the SPDIF transmitter has been tested on a Mele A2000. In order for this patch set to be functional we require audio clock patches which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery This is version 2 of the patch series. After Mark's initial comments I realised that I shouldn't of blindly copied the register set-up from the original SDK that was released. It doesn't help also that the datasheet that finally emerged lacked a serious amount of detail. If anybody does recognise the IP used for the block then please speak out. In short, much of the code that was in the first release is no longer here. I'm still unsure about the machine driver setup. Simple card seems a bit overkill and what I've added is pretty much a copy of what is done in imx-spdif. I'm not planning on arguing the case for my method so if you can advise then I'm all ears. Thanks in advance, CK --- Changes since v1: - Moved sunxi-machine-spdif.c to seperate patch - replaced sunxi in naming scheme with sun4i in the sun4i-spdif driver. - split tx controller into seperate enable/disable functions - moved setclk and setfmt functionality into hw params - added support for mono signals. - cleaned up probe clock set up. - removed all writes to transmit status registers. - removed of_id - removed power management code. - Added support for more rates. --- Marcus Cooper (4): dt-bindings: add sunxi SPDIF transceiver bindings dt-binding: Add sunxi S/PDIF machine driver ASoC: sunxi: Add S/PDIF machine driver. ASOC: sunxi: Add support for the spdif block .../devicetree/bindings/sound/sunxi,spdif.txt | 49 ++ .../bindings/sound/sunxi-audio-spdif.txt | 36 ++ sound/soc/sunxi/Kconfig | 12 + sound/soc/sunxi/Makefile | 4 + sound/soc/sunxi/sun4i-spdif.c | 612 +++++++++++++++++++++ sound/soc/sunxi/sunxi-machine-spdif.c | 108 ++++ 6 files changed, 821 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/sunxi-audio-spdif.txt create mode 100644 sound/soc/sunxi/sun4i-spdif.c create mode 100644 sound/soc/sunxi/sunxi-machine-spdif.c -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl_sai: fix no frame clk in master mode
by Zidan Wang 23 Oct '15

23 Oct '15

After several open/close sai test with ctrl+c, there will be I/O error. The SAI can't work anymore, can't recover. There will be no frame clock. With adding the software reset in trigger stop, the issue can be fixed. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c737f1..6a947dc 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -504,6 +504,13 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_FR, FSL_SAI_CSR_FR); + + /* Software Reset for both Tx and Rx */ + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + /* Clear SR bit to finish the reset */ + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); } break; default: -- 1.9.1

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