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[alsa-devel] [PATCH 1/2] ALSA: Add a helper to add a new attribute group to card
by Takashi Iwai 05 Feb '15

05 Feb '15

For assigning sysfs entries for a card device from the driver, introduce a new helper function, snd_card_add_dev_attr(). In this way, we can avoid the possible race between the device registration and the sysfs addition / removal. The driver can pass a new attribute group to add freely. This has to be called before snd_card_register(). Currently, up to two extra groups can be added. More than that, it'll return an error. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/core.h | 3 +++ sound/core/init.c | 31 +++++++++++++++++++++++++------ 2 files changed, 28 insertions(+), 6 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 58882bfacdd7..da5748289968 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -132,6 +132,7 @@ struct snd_card { struct completion *release_completion; struct device *dev; /* device assigned to this card */ struct device card_dev; /* cardX object for sysfs */ + const struct attribute_group *dev_groups[4]; /* assigned sysfs attr */ bool registered; /* card_dev is registered? */ #ifdef CONFIG_PM @@ -262,6 +263,8 @@ void snd_card_set_id(struct snd_card *card, const char *id); int snd_card_register(struct snd_card *card); int snd_card_info_init(void); int snd_card_info_done(void); +int snd_card_add_dev_attr(struct snd_card *card, + const struct attribute_group *group); int snd_component_add(struct snd_card *card, const char *component); int snd_card_file_add(struct snd_card *card, struct file *file); int snd_card_file_remove(struct snd_card *card, struct file *file); diff --git a/sound/core/init.c b/sound/core/init.c index 96194599e82e..35419054821c 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -181,7 +181,7 @@ void snd_device_initialize(struct device *dev, struct snd_card *card) EXPORT_SYMBOL_GPL(snd_device_initialize); static int snd_card_do_free(struct snd_card *card); -static const struct attribute_group *card_dev_attr_groups[]; +static const struct attribute_group card_dev_attr_group; static void release_card_device(struct device *dev) { @@ -269,7 +269,8 @@ int snd_card_new(struct device *parent, int idx, const char *xid, card->card_dev.parent = parent; card->card_dev.class = sound_class; card->card_dev.release = release_card_device; - card->card_dev.groups = card_dev_attr_groups; + card->card_dev.groups = card->dev_groups; + card->dev_groups[0] = &card_dev_attr_group; err = kobject_set_name(&card->card_dev.kobj, "card%d", idx); if (err < 0) goto __error; @@ -700,14 +701,32 @@ static struct attribute *card_dev_attrs[] = { NULL }; -static struct attribute_group card_dev_attr_group = { +static const struct attribute_group card_dev_attr_group = { .attrs = card_dev_attrs, }; -static const struct attribute_group *card_dev_attr_groups[] = { - &card_dev_attr_group, - NULL +/** + * snd_card_add_dev_attr - Append a new sysfs attribute group to card + * @card: card instance + * @group: attribute group to append + */ +int snd_card_add_dev_attr(struct snd_card *card, + const struct attribute_group *group) +{ + int i; + + /* loop for (arraysize-1) here to keep NULL at the last entry */ + for (i = 0; i < ARRAY_SIZE(card->dev_groups) - 1; i++) { + if (!card->dev_groups[i]) { + card->dev_groups[i] = group; + return 0; + } + } + + dev_err(card->dev, "Too many groups assigned\n"); + return -ENOSPC; }; +EXPORT_SYMBOL_GPL(snd_card_add_dev_attr); /** * snd_card_register - register the soundcard -- 2.2.2

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[alsa-devel] ASoC updates for v3.20
by Mark Brown 05 Feb '15

05 Feb '15

The following changes since commit e36f014edff70fc02b3d3d79cead1d58f289332e: Linux 3.19-rc7 (2015-02-01 20:07:21 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.20-2 for you to fetch changes up to 16ca41c6cc904d7eb9e92afc5f576e375c314f90: Merge remote-tracking branches 'asoc/topic/wm8960', 'asoc/topic/wm8988' and 'asoc/topic/xtfpga' into asoc-next (2015-02-04 20:57:26 +0000) ---------------------------------------------------------------- ASoC: Updates for v3.20 More updates for v3.20: - Lots of refactoring from Lars-Peter Clausen, moving drivers to more data driven initialization and rationalizing a lot of DAPM usage. - Much improved handling of CDCLK clocks on Samsung I2S controllers. - Lots of driver specific cleanups and feature improvements. - CODEC support for TI PCM514x and TLV320AIC3104 devices. - Board support for Tegra systems with Realtek RT5677. ---------------------------------------------------------------- Alexander Morozov (1): ASoC: wm8904: add dt ids table Alexandre Belloni (2): ASoC: atmel: sam9g20_wm8731: remove useless include Asoc: sam9x5_wm8731: depend on ARCH_AT91 instead of SOC_AT91SAM9X5 Anatol Pomozov (3): ASoC: tegra: Add platform driver for rt5677 audio codec ASoC: ts3a227e: Remap keys to match Android headset specification ASoC: ts3a227e: Add dts property that allows to specify micbias voltage Andrew Jackson (9): ASoC: dwc: Remove unnecessary debug messages and tests ASoC: dwc: Reorder code in preparation for DT support ASoC: dwc: Switch to managed clock resource ASoC: dwc: Read I2S block configuration from registers ASoC: dwc: Prepare clock before use ASoC: dwc: Register components with managed interface ASoC: dwc: Add documentation for I2S DT ASoC: dwc: Add devicetree support for Designware I2S ASoC: dwc: Remove unnecessary conditional compilation Arnd Bergmann (3): ASoC: davinci: fix DM365_EVM codec selection ASoC: rt5677: fix SPI dependency ASoC: pxa: make TTC DKB tristate Bard Liao (6): ASoC: rt5670: set platform data by dmi ASoC: rt5670: Add runtime PM support ASoC: rt5670: Keep sysclk on if JD func is used ASoC: rt5670: redefine ASRC control registers 0x84 and 0x85 ASoC: rt286: Fix capture volume setting issue ASoC: rt5645: Add rt5650 codec support Bo Shen (7): ASoC: atmel_ssc_dai: fix start event for I2S mode ASoC: atmel_ssc_dai: remove clock pin comments ASoC: wm8904: add new compatible string ASoC: atmel_ssc_dai: fix the setting for DSP mode ASoC: atmel_ssc_dai: refactor the startup and shutdown ASoC: atmel_ssc_dai: only clean PDC when using PDC ASoC: atmel-pcm-dma: won't check direction when configure dma Charles Keepax (2): ASoC: arizona: Add delay for output disable ASoC: arizona: Coalesce output power up delays Dan Carpenter (1): ASoC: Intel: remove an unused struct member Eric Nelson (1): ASoC: sgtl5000: add delay before first I2C access Fabio Estevam (10): ASoC: fsl_spdif: Use dev_name() for registering the irq ASoC: fsl_asrc: Use dev_name() for registering the irq ASoC: mxs-saif: Do not store the irq number in the private structure ASoC: fsl_ssi: Make error message concise ASoC: fsl_asrc: Make error message concise ASoC: fsl_sai: Make error message concise ASoC: fsl_esai: Make error message concise ASoC: fsl_spdif: Make error message concise ASoC: fsl_ssi: Change irq type to 'int' ASoC: fsl: imx-spdif: Set the card owner field Fang, Yang A (1): ASoC: Intel: Add support rt5645 in sst driver Filip Brozovic (1): ASoC: sgtl5000: Use shift mask when setting codec mode Jarkko Nikula (2): ASoC: pcm: Fix vague codec and cpu DAI prepare error messages ASoC: rt5640: Add RT5642 ACPI ID for Intel Baytrail Jianqun Xu (1): ASoC: rockchip: i2s: set TDL and RDL to 16 samples Jie Yang (8): ASoC: Intel: Store the entry_point read from FW file ASoC: Intel: Always enable DRAM block for FW dump ASoC: Intel: Add stream direction for pcm-module map ASoC: Intel: Split hsw_pcm_data for playback and capture ASoC: Intel: initial stream_hw_id to invalid value ASoC: Intel: Used lock version to update shim registers MAINTAINERS: ASoC: add maintainer for Intel BDW/HSW ASoC driver ASoC: Intel: initial scalar variable ba Jyri Sarha (3): ASoC: simple-card: Enable and disable DAI clocks as needed ASoC: tlv320aic3x: Add support for tlv320aic3104 ASoC: tlv320aic3x: Fix bad comment before intercon_extra_3104 definition Krzysztof Kozlowski (8): ASoC: alc5632: Constify struct regmap_config and snd_soc_codec_driver ASoC: cs35l32: Constify struct regmap_config and snd_soc_codec_driver ASoC: cs42l52: Constify struct regmap_config and snd_soc_codec_driver ASoC: cs42l56: Constify struct regmap_config and snd_soc_codec_driver ASoC: cs42l73: Constify struct regmap_config and snd_soc_codec_driver ASoC: wm8804: Constify struct regmap_config and snd_soc_codec_driver ASoC: wm8988: Constify struct regmap_config and snd_soc_codec_driver ASoC: wm8804: Constify struct regmap_config and snd_soc_codec_driver Kuninori Morimoto (19): ASoC: rsnd: add recovery support for under/over flow error on SRC ASoC: rsnd: remove unsed .rate ASoC: rsnd: tidyup rsnd_io_to_xxx() macro declaration position ASoC: rsnd: rsnd_mod controls clock ASoC: rsnd: rename rsnd_dai_get() to rsnd_rdai_get() ASoC: rsnd: rename rsnd_dai_is_clk_master() to rsnd_rdai_is_clk_master() ASoC: rsnd: remove unused rsnd_dai_id() ASoC: rsnd: remove rsnd_dai_get_platform_info() macro ASoC: rsnd: add rsnd_io_to_rdai() ASoC: rsnd: replace rsnd_dai_is_play() to rsnd_io_is_play() ASoC: rsnd: remove unsed rdai parameter from each function ASoC: rsnd: replace rdai to priv in each function ASoC: rsnd: save priv in struct rsnd_dai ASoC: rsnd: error meesage indicates its port ASoC: rsnd: ignore DIRQ when DMA transfer ASoC: rsnd: tidyup DMA start/stop timing on SSI ASoC: rsnd: tidyup DVC/SRC/SSI order ASoC: rsnd: call missing snd_ctl_remove() ASoC: rsnd: call missing snd_soc_unregiter_component/platform() Lad, Prabhakar (1): ASoC: ts3a227e: fix sparse warning Lars-Peter Clausen (109): ASoC: wm8750: Make VREF a VMID instead of OUTPUT widget ASoC: wm8750: Cleanup manual bias level transitions ASoC: dapm: Don't mark MICBIAS widgets as auto non-connected ASoC: dapm: Simplify fully route card handling ASoC: dmaengine: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: au1x: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: nuc900: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: rcar: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: sh: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: txx9: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: broadwell: Drop unnecessary snd_soc_dapm_enable() calls ASoC: byt-rt5640: Fix snd_soc_dapm_ignore_suspend() calls ASoC: byt-rt5640: Register microphone routes with the card DAPM context ASoC: intel: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: sam9g20_wm8731: Use static DAI format setup ASoC: db1200: Use static DAI format setup ASoC: eureka-tlv320: Use static DAI format setup ASoC: mx27vis-aci32x4: Use static DAI format setup ASoC: wm1133-ev1: Use static DAI format setup ASoC: mxs-sgtl5000: Use static DAI format setup ASoC: ams-delta: Use static DAI format setup ASoC: raumfeld: Use static DAI format setup ASoC: zylonite: Use static DAI format setup ASoC: goni_wm8994: Use static DAI format setup ASoC: h1940_uda1380: Use static DAI format setup ASoC: jive_wm8750: Use static DAI format setup ASoC: neo1973_wm8753: Use static DAI format setup ASoC: rx1950_uda1380: Use static DAI format setup ASoC: s3c24xx_simtec: Use static DAI format setup ASoC: s3c24xx_uda134x: Use static DAI format setup ASoC: smartq_wm8987: Use static DAI format setup ASoC: smdk_wm8580: Use static DAI format setup ASoC: smdk_wm8570pcm: Use static DAI format setup ASoC: smdk_wm8994pcm: Use static DAI format setup ASoC: migor: Use static DAI format setup ASoC: Add helper function for changing the DAI link format ASoC: omap-twl4030: Use snd_soc_runtime_set_dai_fmt() ASoC: mop500_ab8500: Use snd_soc_runtime_set_dai_fmt() ASoC: Export snd_soc_runtime_set_dai_fmt() ASoC: ttc-dkb: Remove unnecessary snd_soc_dapm_disable_pin() calls ASoC: corgi: Automatically disconnect non-connected pins ASoC: e740: Automatically disconnect non-connected pins ASoC: e750: Automatically disconnect non-connected pins ASoC: hx4700: Automatically disconnect non-connected pins ASoC: magician: Automatically disconnect non-connected pins ASoC: palm27x: Fix microphone route ASoC: palm27x: Automatically disconnect non-connected pins ASoC: spitz: Automatically disconnect non-connected pins ASoC: ad193x: Don't power down DAC in CODEC probe ASoC: mc13783: Update set_tdm_slot() semantics ASoC: fsl: Update set_tdm_slot() semantics ASoC: fsl: Remove fsl_asoc_xlate_tdm_slot_mask() ASoC: Update snd_soc_dai_set_tdm_slot() documentation ASoC: 88pm860x: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: ak4671: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: da732x: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: sgtl5000: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: sn95031: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: arizona: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm2000: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm5102: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm5102: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm5110: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8350: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8400: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8731: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8770: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8900: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8903: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8904: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8955: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8961: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8961: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8962: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8988: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8990: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8991: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8993: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8995: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8995: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8996: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm8997: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm9081: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm9090: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm9713: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm_adsp: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: wm_hubs: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: mioa701_wm9713: Fix speaker event ASoC: pcm3008: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320aic31xx: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320aic3x: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320dac33: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: twl4030: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: twl6040: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: alc5623: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: alc5632: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt286: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5631: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5640: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5645: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5651: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5670: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5677: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: Remove codec field from snd_soc_dapm_widget ASoC: jive: Automatically disconnect non-connected pins ASoC: smdk_wm8580: Pass card DAPM context to snd_soc_dapm_disable_pin() ASoC: samsung: Replace depends on REGMAP_I2C with depends on I2C ASoC: Add support for allocating AC'97 device before registering it ASoC: wm97xx: Reset AC'97 device before registering it Manuel Lauss (1): ASoC: wm8731: init mutex in i2c init path Mark Brown (27): ASoC: dapm: Don't use async I/O Merge branch 'topic/rt5670' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge remote-tracking branch 'asoc/fix/fsl-ssi' into asoc-fsl Merge branch 'topic/rt5677' of git://git.kernel.org/.../broonie/sound into asoc-w-codec Merge branch 'topic/dai' of git://git.kernel.org/.../broonie/sound into asoc-samsung Merge branch 'fix/atmel' of git://git.kernel.org/.../broonie/sound into asoc-atmel Merge branch 'topic/rt286' of git://git.kernel.org/.../broonie/sound into asoc-w-codec Merge branch 'fix/rt5677' of git://git.kernel.org/.../broonie/sound into asoc-w-codec Merge branch 'topic/arizona' of git://git.kernel.org/.../broonie/sound into asoc-w-codec Merge remote-tracking branches 'asoc/fix/ac97', 'asoc/fix/atmel', 'asoc/fix/intel', 'asoc/fix/rt286', 'asoc/fix/rt5640', 'asoc/fix/sgtl5000', 'asoc/fix/sta32x', 'asoc/fix/tlv320aic3x' and 'asoc/fix/wm8731' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dmaengine' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next Merge remote-tracking branch 'asoc/topic/w-codec' into asoc-next Merge remote-tracking branches 'asoc/topic/ad193x', 'asoc/topic/alc5632', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next Merge remote-tracking branches 'asoc/topic/bt-sco', 'asoc/topic/cs35l32', 'asoc/topic/cs42l52' and 'asoc/topic/cs42l56' into asoc-next Merge remote-tracking branches 'asoc/topic/cs42l73', 'asoc/topic/dai' and 'asoc/topic/davinci' into asoc-next Merge remote-tracking branches 'asoc/topic/doc', 'asoc/topic/dwc', 'asoc/topic/fsi', 'asoc/topic/fsl' and 'asoc/topic/fsl-asrc' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/imx', 'asoc/topic/intel', 'asoc/topic/mxs-saif' and 'asoc/topic/nuc900' into asoc-next Merge remote-tracking branches 'asoc/topic/omap', 'asoc/topic/pxa' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/rockchip', 'asoc/topic/rt5645' and 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branches 'asoc/topic/rx51', 'asoc/topic/samsung', 'asoc/topic/sh', 'asoc/topic/simple' and 'asoc/topic/sta32x' into asoc-next Merge remote-tracking branches 'asoc/topic/tdm-slot', 'asoc/topic/tegra', 'asoc/topic/tlv320aic3x', 'asoc/topic/ts3a227e' and 'asoc/topic/ts3a277e' into asoc-next Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/wm8750', 'asoc/topic/wm8804' and 'asoc/topic/wm8904' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8960', 'asoc/topic/wm8988' and 'asoc/topic/xtfpga' into asoc-next Markus Elfring (3): ASoC: Intel: Delete an unnecessary check before the function call "sst_dma_free" ASoC: fsi: Deletion of unnecessary checks before the function call "clk_enable" ASoC: Intel: Delete an unnecessary check before the function call "release_firmware" Masanari Iida (2): ASoC: Fix typo in bt-sco.c ASoC: Fix warning with make xmldocs caused by soc-devres.c Max Filippov (1): ASoC: add xtensa xtfpga I2S interface and platform Mengdong Lin (2): ASoC: rt5670: add API to select ASRC clock source ASoC: Intel: Select RT5672 ASRC clock source on Cherrytrail and Braswell Oder Chiou (6): ASoC: rt5677: Revise the filter powers ASoC: rt5677: Add the ASRC support ASoC: rt5677: Adjust the routing of "PLL1" ASoC: rt5677: Add the MICBIAS VDD setting in the platform data ASoC: rt5677: Use the regmap functions instead of the snd_soc functions ASoC: rt5677: Add the slot_width "25" support in the TDM mode Peter Rosin (10): ASoC: pcm512x: Also support PCM514x devices ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS ASoC: pcm512x: Fixup warning splat ASoC: pcm512x: Use the correct range constraints for S24_LE ASoC: atmel_ssc_dai: Support SND_SOC_DAIFMT_CBM_CFS on I2S Peter Ujfalusi (7): ASoC: davinci-mcasp: Support for combined tx/rx interrupt line ASoC: rx51: Correct DAPM route for capture ASoC: tlv320aic3x: Fix data delay configuration ASoC: davinci-evm: Do not include edma headers ASoC: davinci-evm: Do not include davinci-pcm.h and davinci-i2s.h ASoC: davicni-mcasp: Mark the common irq line as shared ASoC: davinci-mcasp: Fix DIT only McASP instance support Sylwester Nawrocki (13): ASoC: samsung: i2s: Remove unused gpios field from struct i2s ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clk provider DT binding documentation ASoC: samsung: i2s: Add clock provider for the I2S internal clocks Takashi Iwai (2): ASoC: intel: Remove superfluous backslash in Kconfig ASoC: Add sysfs entries via static attribute groups Thomas Niederprüm (9): ASoC: sta32x: correct bit shift value for IDE register ASoC: sta32x: Convert to direct regmap API usage. ASoC: sta32x: make sta32x a gpio consumer for the reset GPIO ASoC: sta32x: use DECLARE_TLV_DB_RANGE macro. ASoC: sta32x: move code to calculate mclk divider and extrapolation ratio to sta32x_hw_params() ASoC: sta32x: add device tree binding. ASoC: sta32x: use dev_dbg() for debug output. ASoC: sta32x: minor Kconfig update. ASoC: sta32x: change dai name to be in line with the sta350 driver. Vaishali Thakkar (1): ASoC: wm5102: Use put_unaligned_be16 Wei Yongjun (1): ASoC: Intel: fix sparse non static symbol warnings Wolfram Sang (4): ASoC: intel: sst: drop owner assignment from platform_drivers ASoC: omap: drop owner assignment from platform_drivers ASoC: pxa: drop owner assignment from platform_drivers ASoC: samsung: drop owner assignment from platform_drivers Zidan Wang (1): ASoC: wm8960: Let wm8960 codec driver manage its own MCLK kbuild test robot (2): ASoC: Intel: fix platform_no_drv_owner.cocci warnings ASoC: Intel: fix platform_no_drv_owner.cocci warnings .../devicetree/bindings/sound/cdns,xtfpga-i2s.txt | 18 + .../devicetree/bindings/sound/designware-i2s.txt | 31 + .../bindings/sound/nvidia,tegra-audio-rt5677.txt | 67 ++ .../devicetree/bindings/sound/pcm512x.txt | 28 +- .../devicetree/bindings/sound/samsung-i2s.txt | 22 + .../devicetree/bindings/sound/simple-card.txt | 5 + .../devicetree/bindings/sound/st,sta32x.txt | 92 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 10 +- .../devicetree/bindings/sound/ts3a227e.txt | 5 + Documentation/devicetree/bindings/sound/wm8904.txt | 2 +- MAINTAINERS | 11 + include/dt-bindings/sound/samsung-i2s.h | 8 + include/sound/pcm.h | 12 + include/sound/rcar_snd.h | 1 + include/sound/rt5677.h | 3 + include/sound/simple_card.h | 1 + include/sound/soc-dapm.h | 4 +- include/sound/soc.h | 4 + include/sound/sta32x.h | 18 +- sound/core/pcm_lib.c | 85 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel-pcm-dma.c | 12 +- sound/soc/atmel/atmel_ssc_dai.c | 132 +-- sound/soc/atmel/sam9g20_wm8731.c | 31 +- sound/soc/au1x/db1200.c | 19 +- sound/soc/au1x/dbdma2.c | 6 - sound/soc/au1x/dma.c | 6 - sound/soc/codecs/88pm860x-codec.c | 4 +- sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/ad193x.c | 4 +- sound/soc/codecs/ak4671.c | 2 +- sound/soc/codecs/alc5623.c | 8 +- sound/soc/codecs/alc5632.c | 12 +- sound/soc/codecs/arizona.c | 78 +- sound/soc/codecs/arizona.h | 5 + sound/soc/codecs/bt-sco.c | 2 +- sound/soc/codecs/cs35l32.c | 4 +- sound/soc/codecs/cs42l52.c | 4 +- sound/soc/codecs/cs42l56.c | 4 +- sound/soc/codecs/cs42l73.c | 4 +- sound/soc/codecs/da732x.c | 4 +- sound/soc/codecs/mc13783.c | 10 +- sound/soc/codecs/pcm3008.c | 4 +- sound/soc/codecs/pcm512x-i2c.c | 4 + sound/soc/codecs/pcm512x-spi.c | 4 + sound/soc/codecs/pcm512x.c | 932 ++++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- sound/soc/codecs/rt286.c | 51 +- sound/soc/codecs/rt5631.c | 28 +- sound/soc/codecs/rt5640.c | 12 +- sound/soc/codecs/rt5645.c | 174 +++- sound/soc/codecs/rt5645.h | 15 + sound/soc/codecs/rt5651.c | 18 +- sound/soc/codecs/rt5670.c | 137 ++- sound/soc/codecs/rt5670.h | 80 +- sound/soc/codecs/rt5677.c | 243 +++++- sound/soc/codecs/sgtl5000.c | 27 +- sound/soc/codecs/sn95031.c | 33 +- sound/soc/codecs/sta32x.c | 534 +++++++----- sound/soc/codecs/sta32x.h | 2 +- sound/soc/codecs/tlv320aic31xx.c | 9 +- sound/soc/codecs/tlv320aic3x.c | 349 +++++--- sound/soc/codecs/tlv320dac33.c | 9 +- sound/soc/codecs/ts3a227e.c | 35 +- sound/soc/codecs/twl4030.c | 55 +- sound/soc/codecs/twl6040.c | 4 +- sound/soc/codecs/wm2000.c | 2 +- sound/soc/codecs/wm5100.c | 5 +- sound/soc/codecs/wm5102.c | 23 +- sound/soc/codecs/wm5110.c | 20 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8731.c | 5 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8770.c | 8 +- sound/soc/codecs/wm8804.c | 4 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 35 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8958-dsp2.c | 2 +- sound/soc/codecs/wm8960.c | 51 +- sound/soc/codecs/wm8961.c | 4 +- sound/soc/codecs/wm8962.c | 6 +- sound/soc/codecs/wm8988.c | 6 +- sound/soc/codecs/wm8990.c | 9 +- sound/soc/codecs/wm8991.c | 9 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 23 +- sound/soc/codecs/wm8995.c | 20 +- sound/soc/codecs/wm8996.c | 8 +- sound/soc/codecs/wm8997.c | 11 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9705.c | 16 +- sound/soc/codecs/wm9712.c | 12 +- sound/soc/codecs/wm9713.c | 14 +- sound/soc/codecs/wm_adsp.c | 6 +- sound/soc/codecs/wm_hubs.c | 10 +- sound/soc/davinci/Kconfig | 3 +- sound/soc/davinci/davinci-evm.c | 6 - sound/soc/davinci/davinci-mcasp.c | 103 ++- sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/designware_i2s.c | 311 +++++-- sound/soc/fsl/eukrea-tlv320.c | 23 +- sound/soc/fsl/fsl_asrc.c | 5 +- sound/soc/fsl/fsl_asrc.h | 3 - sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 17 +- sound/soc/fsl/fsl_ssi.c | 10 +- sound/soc/fsl/fsl_utils.c | 27 - sound/soc/fsl/fsl_utils.h | 3 - sound/soc/fsl/imx-mc13783.c | 5 +- sound/soc/fsl/imx-spdif.c | 1 + sound/soc/fsl/imx-ssi.c | 5 +- sound/soc/fsl/mx27vis-aic32x4.c | 12 +- sound/soc/fsl/wm1133-ev1.c | 16 +- sound/soc/generic/simple-card.c | 34 + sound/soc/intel/Kconfig | 4 +- sound/soc/intel/broadwell.c | 10 - sound/soc/intel/byt-rt5640.c | 12 +- sound/soc/intel/bytcr_dpcm_rt5640.c | 1 - sound/soc/intel/cht_bsw_rt5672.c | 15 +- sound/soc/intel/sst-baytrail-pcm.c | 6 - sound/soc/intel/sst-dsp.c | 3 +- sound/soc/intel/sst-firmware.c | 3 + sound/soc/intel/sst-haswell-dsp.c | 17 +- sound/soc/intel/sst-haswell-ipc.c | 9 +- sound/soc/intel/sst-haswell-pcm.c | 105 ++- sound/soc/intel/sst-mfld-platform-pcm.c | 7 - sound/soc/intel/sst/sst_acpi.c | 7 +- sound/soc/intel/sst/sst_loader.c | 3 +- sound/soc/mxs/mxs-saif.c | 10 +- sound/soc/mxs/mxs-saif.h | 1 - sound/soc/mxs/mxs-sgtl5000.c | 27 +- sound/soc/nuc900/nuc900-pcm.c | 6 - sound/soc/omap/ams-delta.c | 18 +- sound/soc/omap/omap-hdmi-audio.c | 1 - sound/soc/omap/omap-twl4030.c | 20 +- sound/soc/omap/rx51.c | 8 +- sound/soc/pxa/Kconfig | 2 +- sound/soc/pxa/corgi.c | 16 +- sound/soc/pxa/e740_wm9705.c | 20 +- sound/soc/pxa/e750_wm9705.c | 20 +- sound/soc/pxa/hx4700.c | 8 +- sound/soc/pxa/magician.c | 21 +- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/pxa/palm27x.c | 15 +- sound/soc/pxa/raumfeld.c | 35 +- sound/soc/pxa/spitz.c | 23 +- sound/soc/pxa/ttc-dkb.c | 4 - sound/soc/pxa/zylonite.c | 12 +- sound/soc/rockchip/rockchip_i2s.c | 4 + sound/soc/samsung/Kconfig | 4 +- sound/soc/samsung/arndale_rt5631.c | 1 - sound/soc/samsung/goni_wm8994.c | 23 +- sound/soc/samsung/h1940_uda1380.c | 15 +- sound/soc/samsung/i2s.c | 362 ++++---- sound/soc/samsung/jive_wm8750.c | 34 +- sound/soc/samsung/neo1973_wm8753.c | 25 +- sound/soc/samsung/odroidx2_max98090.c | 6 +- sound/soc/samsung/rx1950_uda1380.c | 15 +- sound/soc/samsung/s3c24xx_simtec.c | 20 +- sound/soc/samsung/s3c24xx_uda134x.c | 12 +- sound/soc/samsung/smartq_wm8987.c | 16 +- sound/soc/samsung/smdk_wm8580.c | 26 +- sound/soc/samsung/smdk_wm8580pcm.c | 19 +- sound/soc/samsung/smdk_wm8994pcm.c | 16 +- sound/soc/sh/dma-sh7760.c | 6 - sound/soc/sh/fsi.c | 15 +- sound/soc/sh/migor.c | 12 +- sound/soc/sh/rcar/adg.c | 14 +- sound/soc/sh/rcar/core.c | 111 ++- sound/soc/sh/rcar/dvc.c | 63 +- sound/soc/sh/rcar/gen.c | 15 + sound/soc/sh/rcar/rsnd.h | 81 +- sound/soc/sh/rcar/src.c | 269 ++++-- sound/soc/sh/rcar/ssi.c | 100 +-- sound/soc/sh/siu_pcm.c | 1 - sound/soc/soc-ac97.c | 36 +- sound/soc/soc-core.c | 207 +++-- sound/soc/soc-dapm.c | 125 +-- sound/soc/soc-devres.c | 2 +- sound/soc/soc-generic-dmaengine-pcm.c | 15 +- sound/soc/soc-pcm.c | 7 +- sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_rt5677.c | 347 ++++++++ sound/soc/txx9/txx9aclc.c | 6 - sound/soc/ux500/mop500_ab8500.c | 16 +- sound/soc/xtensa/Kconfig | 7 + sound/soc/xtensa/Makefile | 3 + sound/soc/xtensa/xtfpga-i2s.c | 675 +++++++++++++++ 196 files changed, 5566 insertions(+), 2227 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cdns,xtfpga-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-rt5677.txt create mode 100644 Documentation/devicetree/bindings/sound/st,sta32x.txt create mode 100644 include/dt-bindings/sound/samsung-i2s.h create mode 100644 sound/soc/tegra/tegra_rt5677.c create mode 100644 sound/soc/xtensa/Kconfig create mode 100644 sound/soc/xtensa/Makefile create mode 100644 sound/soc/xtensa/xtfpga-i2s.c

2 4

[alsa-devel] [PATCH_V4 0/3] sound:soc: jz4740: Add DT, dynamic sampling
by Zubair Lutfullah Kakakhel 05 Feb '15

05 Feb '15

Hi, Here are a few simple patches for the jz4740. First adds dynamic sampling support to jz4740-i2s. Then two to add a simple binding and DT support. Then a patch to set the clock rate for i2s. These are in preparation for jz4780 and ci20 later on. Patches are based on 3.19-rc7. Quite disjoint and stay within jz4740 so should apply easily on other trees. If you would like to have them rebased to a different tree, please tell. Thank-you V4 changes: Added Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Rebased on 3.19-rc7 Removed patch 4 which was redundant V3 changes: Added dma binding to example but forgot to put it in required properties. V2 changes: Removed a redundant #define Added #ifdef CONFIG_OF Added dma in binding document Removed a redundant call to clk_prepare_enable Zubair Lutfullah Kakakhel (3): sound: soc: jz4740: Add dynamic sampling rate support to jz4740-i2s dt: sound: jz4740: Add binding documentation for jz4740-i2s sound: soc: jz4740: Add DT support to jz4740-i2s driver .../bindings/sound/ingenic,jz4740-i2s.txt | 23 ++++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.c | 21 +++++++++++++++++++- 2 files changed, 43 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: Intel: add a status for runtime suspend/resume
by Jie Yang 05 Feb '15

05 Feb '15

For runtime suspend/resume, it is some different with suspend/resume, e.g. codec power supply won't be switch off, codec jack detection still working(to wake up system from Jack event), won't call call snd_soc_suspend/resume, etc. So here, we add a platform PM status, HSW_PM_STATE_RTD3, to make the status clearer, when in idle, it will enter this status, to transfer from HSW_PM_STATE_RTD3 to HSW_PM_STATE_D3, we will do those extra jobs, and vice versa for resuming. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-pcm.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index ad7f4a5..06b149e 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -119,8 +119,9 @@ struct hsw_pcm_data { }; enum hsw_pm_state { - HSW_PM_STATE_D3 = 0, - HSW_PM_STATE_D0 = 1, + HSW_PM_STATE_D0 = 0, + HSW_PM_STATE_RTD3 = 1, + HSW_PM_STATE_D3 = 2, }; /* private data for the driver */ @@ -1035,12 +1036,12 @@ static int hsw_pcm_runtime_suspend(struct device *dev) struct hsw_priv_data *pdata = dev_get_drvdata(dev); struct sst_hsw *hsw = pdata->hsw; - if (pdata->pm_state == HSW_PM_STATE_D3) + if (pdata->pm_state >= HSW_PM_STATE_RTD3) return 0; sst_hsw_dsp_runtime_suspend(hsw); sst_hsw_dsp_runtime_sleep(hsw); - pdata->pm_state = HSW_PM_STATE_D3; + pdata->pm_state = HSW_PM_STATE_RTD3; return 0; } @@ -1051,7 +1052,7 @@ static int hsw_pcm_runtime_resume(struct device *dev) struct sst_hsw *hsw = pdata->hsw; int ret; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_RTD3) return 0; ret = sst_hsw_dsp_load(hsw); @@ -1091,7 +1092,7 @@ static void hsw_pcm_complete(struct device *dev) struct hsw_pcm_data *pcm_data; int i, err; - if (pdata->pm_state == HSW_PM_STATE_D0) + if (pdata->pm_state != HSW_PM_STATE_D3) return; err = sst_hsw_dsp_load(hsw); @@ -1139,6 +1140,9 @@ static int hsw_pcm_prepare(struct device *dev) if (pdata->pm_state == HSW_PM_STATE_D3) return 0; + else if (pdata->pm_state == HSW_PM_STATE_RTD3) + goto suspend; + /* suspend all active streams */ for (i = 0; i < ARRAY_SIZE(mod_map); i++) { pcm_data = &pdata->pcm[mod_map[i].dai_id][mod_map[i].stream]; @@ -1152,9 +1156,6 @@ static int hsw_pcm_prepare(struct device *dev) msleep(2); } - snd_soc_suspend(pdata->soc_card->dev); - snd_soc_poweroff(pdata->soc_card->dev); - /* enter D3 state and stall */ sst_hsw_dsp_runtime_suspend(hsw); @@ -1174,6 +1175,11 @@ static int hsw_pcm_prepare(struct device *dev) /* put the DSP to sleep */ sst_hsw_dsp_runtime_sleep(hsw); + +suspend: + snd_soc_suspend(pdata->soc_card->dev); + snd_soc_poweroff(pdata->soc_card->dev); + pdata->pm_state = HSW_PM_STATE_D3; return 0; -- 1.9.1

3 2

[alsa-devel] [PATCH] sound: soc: codecs/ts3a227e: fix sparse warning
by Lad Prabhakar 04 Feb '15

04 Feb '15

From: "Lad, Prabhakar" <prabhakar.csengg(a)gmail.com> this patch fixes following sparse warning: ts3a227e.c:222:5: warning: symbol 'ts3a227e_enable_jack_detect' was not declared. Should it be static? Signed-off-by: Lad, Prabhakar <prabhakar.csengg(a)gmail.com> --- Found this issue on linux-next (gcc version 4.8.2, sparse version 0.4.5-rc1)and applies on top linux-next. sound/soc/codecs/ts3a227e.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 5ecfa31..9fd80ac 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -20,6 +20,8 @@ #include <sound/jack.h> #include <sound/soc.h> +#include "ts3a227e.h" + struct ts3a227e { struct regmap *regmap; struct snd_soc_jack *jack; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add support for tlv320aic3104
by Jyri Sarha 04 Feb '15

04 Feb '15

Disables GPIO support and LINE2 input and renames Mic3 input to Mic2, if tlv320aic3104 mode is seleced. Devicetree binding document is updated accordingly. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- .../devicetree/bindings/sound/tlv320aic3x.txt | 10 +- sound/soc/codecs/tlv320aic3x.c | 344 ++++++++++++++------ 2 files changed, 252 insertions(+), 102 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 5e6040c..47a213c 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -9,6 +9,7 @@ Required properties: "ti,tlv320aic33" - TLV320AIC33 "ti,tlv320aic3007" - TLV320AIC3007 "ti,tlv320aic3106" - TLV320AIC3106 + "ti,tlv320aic3104" - TLV320AIC3104 - reg - <int> - I2C slave address @@ -18,6 +19,7 @@ Optional properties: - gpio-reset - gpio pin number used for codec reset - ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + - Not supported on tlv320aic3104 - ai3x-micbias-vg - MicBias Voltage required. 1 - MICBIAS output is powered to 2.0V, 2 - MICBIAS output is powered to 2.5V, @@ -36,7 +38,13 @@ CODEC output pins: * HPLCOM * HPRCOM -CODEC input pins: +CODEC input pins for TLV320AIC3104: + * MIC2L + * MIC2R + * LINE1L + * LINE1R + +CODEC input pins for other compatible codecs: * MIC3L * MIC3R * LINE1L diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b7ebce0..49ba3c7 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -87,6 +87,7 @@ struct aic3x_priv { #define AIC3X_MODEL_3X 0 #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 +#define AIC3X_MODEL_3104 3 u16 model; /* Selects the micbias voltage */ @@ -316,52 +317,37 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * only for swapped L-to-R and R-to-L routes. See below stereo controls * for direct L-to-L and R-to-R routes. */ - SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume", - LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left Line Mixer PGAR Bypass Volume", PGAR_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left Line Mixer DACR1 Playback Volume", DACR1_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume", - LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right Line Mixer PGAL Bypass Volume", PGAL_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right Line Mixer DACL1 Playback Volume", DACL1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume", - LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HP Mixer PGAR Bypass Volume", PGAR_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HP Mixer DACR1 Playback Volume", DACR1_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume", - LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HP Mixer PGAL Bypass Volume", PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HP Mixer DACL1 Playback Volume", DACL1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume", - LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HPCOM Mixer PGAR Bypass Volume", PGAR_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Left HPCOM Mixer DACR1 Playback Volume", DACR1_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), - SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume", - LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HPCOM Mixer PGAL Bypass Volume", PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), SOC_SINGLE_TLV("Right HPCOM Mixer DACL1 Playback Volume", DACL1_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), /* Stereo output controls for direct L-to-L and R-to-R routes */ - SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume", - LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("Line PGA Bypass Volume", PGAL_2_LLOPM_VOL, PGAR_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), @@ -369,9 +355,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", - LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("HP PGA Bypass Volume", PGAL_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), @@ -379,9 +362,6 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), - SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume", - LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, - 0, 118, 1, output_stage_tlv), SOC_DOUBLE_R_TLV("HPCOM PGA Bypass Volume", PGAL_2_HPLCOM_VOL, PGAR_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), @@ -424,6 +404,45 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), }; +/* For other than tlv320aic3104 */ +static const struct snd_kcontrol_new aic3x_extra_snd_controls[] = { + /* + * Output controls that map to output mixer switches. Note these are + * only for swapped L-to-R and R-to-L routes. See below stereo controls + * for direct L-to-L and R-to-R routes. + */ + SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume", + LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume", + LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume", + LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume", + LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume", + LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv), + + SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume", + LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), + + /* Stereo output controls for direct L-to-L and R-to-R routes */ + SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume", + LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume", + LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, + 0, 118, 1, output_stage_tlv), + + SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume", + LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, + 0, 118, 1, output_stage_tlv), +}; + static const struct snd_kcontrol_new aic3x_mono_controls[] = { SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume", LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL, @@ -464,22 +483,24 @@ SOC_DAPM_ENUM("Route", aic3x_right_hpcom_enum); /* Left Line Mixer */ static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_LLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_LLOPM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), }; /* Right Line Mixer */ static const struct snd_kcontrol_new aic3x_right_line_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_RLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), }; /* Mono Mixer */ @@ -494,42 +515,46 @@ static const struct snd_kcontrol_new aic3x_mono_mixer_controls[] = { /* Left HP Mixer */ static const struct snd_kcontrol_new aic3x_left_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLOUT_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0), }; /* Right HP Mixer */ static const struct snd_kcontrol_new aic3x_right_hp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPROUT_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), }; /* Left HPCOM Mixer */ static const struct snd_kcontrol_new aic3x_left_hpcom_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLCOM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0), }; /* Right HPCOM Mixer */ static const struct snd_kcontrol_new aic3x_right_hpcom_mixer_controls[] = { - SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPRCOM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0), + /* Not on tlv320aic3104 */ + SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), }; /* Left PGA Mixer */ @@ -550,6 +575,22 @@ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), }; +/* Left PGA Mixer for tlv320aic3104 */ +static const struct snd_kcontrol_new aic3104_left_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1), +}; + +/* Right PGA Mixer for tlv320aic3104 */ +static const struct snd_kcontrol_new aic3104_right_pga_mixer_controls[] = { + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic2R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), +}; + /* Left Line1 Mux */ static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls = SOC_DAPM_ENUM("Route", aic3x_line1l_2_l_enum); @@ -593,26 +634,56 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), - SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_left_pga_mixer_controls[0], - ARRAY_SIZE(aic3x_left_pga_mixer_controls)), SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1l_mux_controls), SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1r_mux_controls), - SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, - &aic3x_left_line2_mux_controls), /* Inputs to Right ADC */ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", LINE1R_2_RADC_CTRL, 2, 0), - SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, - &aic3x_right_pga_mixer_controls[0], - ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1l_mux_controls), SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1r_mux_controls), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("LLOUT"), + SND_SOC_DAPM_OUTPUT("RLOUT"), + SND_SOC_DAPM_OUTPUT("HPLOUT"), + SND_SOC_DAPM_OUTPUT("HPROUT"), + SND_SOC_DAPM_OUTPUT("HPLCOM"), + SND_SOC_DAPM_OUTPUT("HPRCOM"), + + SND_SOC_DAPM_INPUT("LINE1L"), + SND_SOC_DAPM_INPUT("LINE1R"), + + /* + * Virtual output pin to detection block inside codec. This can be + * used to keep codec bias on if gpio or detection features are needed. + * Force pin on or construct a path with an input jack and mic bias + * widgets. + */ + SND_SOC_DAPM_OUTPUT("Detection"), +}; + +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_widget aic3x_extra_dapm_widgets[] = { + /* Inputs to Left ADC */ + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_left_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line2_mux_controls), + + /* Inputs to Right ADC */ + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3x_right_pga_mixer_controls)), SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line2_mux_controls), @@ -637,11 +708,6 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), - /* Mic Bias */ - SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, - mic_bias_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - /* Output mixers */ SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_line_mixer_controls[0], @@ -662,27 +728,46 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { &aic3x_right_hpcom_mixer_controls[0], ARRAY_SIZE(aic3x_right_hpcom_mixer_controls)), - SND_SOC_DAPM_OUTPUT("LLOUT"), - SND_SOC_DAPM_OUTPUT("RLOUT"), - SND_SOC_DAPM_OUTPUT("HPLOUT"), - SND_SOC_DAPM_OUTPUT("HPROUT"), - SND_SOC_DAPM_OUTPUT("HPLCOM"), - SND_SOC_DAPM_OUTPUT("HPRCOM"), - SND_SOC_DAPM_INPUT("MIC3L"), SND_SOC_DAPM_INPUT("MIC3R"), - SND_SOC_DAPM_INPUT("LINE1L"), - SND_SOC_DAPM_INPUT("LINE1R"), SND_SOC_DAPM_INPUT("LINE2L"), SND_SOC_DAPM_INPUT("LINE2R"), +}; - /* - * Virtual output pin to detection block inside codec. This can be - * used to keep codec bias on if gpio or detection features are needed. - * Force pin on or construct a path with an input jack and mic bias - * widgets. - */ - SND_SOC_DAPM_OUTPUT("Detection"), +/* For tlv320aic3104 */ +static const struct snd_soc_dapm_widget aic3104_extra_dapm_widgets[] = { + /* Inputs to Left ADC */ + SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3104_left_pga_mixer_controls[0], + ARRAY_SIZE(aic3104_left_pga_mixer_controls)), + + /* Inputs to Right ADC */ + SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, + &aic3104_right_pga_mixer_controls[0], + ARRAY_SIZE(aic3104_right_pga_mixer_controls)), + + /* Output mixers */ + SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_line_mixer_controls[0], + ARRAY_SIZE(aic3x_left_line_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_line_mixer_controls[0], + ARRAY_SIZE(aic3x_right_line_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hp_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right HP Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hp_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hp_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Left HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_left_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_left_hpcom_mixer_controls) - 2), + SND_SOC_DAPM_MIXER("Right HPCOM Mixer", SND_SOC_NOPM, 0, 0, + &aic3x_right_hpcom_mixer_controls[0], + ARRAY_SIZE(aic3x_right_hpcom_mixer_controls) - 2), + + SND_SOC_DAPM_INPUT("MIC2L"), + SND_SOC_DAPM_INPUT("MIC2R"), }; static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = { @@ -712,17 +797,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Line1R Mux", "single-ended", "LINE1R"}, {"Left Line1R Mux", "differential", "LINE1R"}, - {"Left Line2L Mux", "single-ended", "LINE2L"}, - {"Left Line2L Mux", "differential", "LINE2L"}, - {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, - {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, - {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, - {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Left ADC", NULL, "Left PGA Mixer"}, - {"Left ADC", NULL, "GPIO1 dmic modclk"}, /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, @@ -730,25 +808,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line1L Mux", "single-ended", "LINE1L"}, {"Right Line1L Mux", "differential", "LINE1L"}, - {"Right Line2R Mux", "single-ended", "LINE2R"}, - {"Right Line2R Mux", "differential", "LINE2R"}, - {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"}, {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"}, - {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, - {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, - {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, - {"Right ADC", NULL, "GPIO1 dmic modclk"}, - - /* - * Logical path between digital mic enable and GPIO1 modulator clock - * output function - */ - {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, - {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, - {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, /* Left DAC Output */ {"Left DAC Mux", "DAC_L1", "Left DAC"}, @@ -761,10 +824,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right DAC Mux", "DAC_R3", "Right DAC"}, /* Left Line Output */ - {"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left Line Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left Line Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -773,10 +834,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"LLOUT", NULL, "Left Line Out"}, /* Right Line Output */ - {"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right Line Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right Line Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -785,10 +844,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"RLOUT", NULL, "Right Line Out"}, /* Left HP Output */ - {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left HP Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left HP Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -797,10 +854,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPLOUT", NULL, "Left HP Out"}, /* Right HP Output */ - {"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right HP Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right HP Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -809,10 +864,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPROUT", NULL, "Right HP Out"}, /* Left HPCOM Output */ - {"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Left HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Left HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Left HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Left HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -823,10 +876,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPLCOM", NULL, "Left HP Com"}, /* Right HPCOM Output */ - {"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, {"Right HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"}, {"Right HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"}, - {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, {"Right HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"}, {"Right HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"}, @@ -839,6 +890,72 @@ static const struct snd_soc_dapm_route intercon[] = { {"HPRCOM", NULL, "Right HP Com"}, }; +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_route intercon_extra[] = { + /* Left Input */ + {"Left Line2L Mux", "single-ended", "LINE2L"}, + {"Left Line2L Mux", "differential", "LINE2L"}, + + {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, + {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Left ADC", NULL, "GPIO1 dmic modclk"}, + + /* Right Input */ + {"Right Line2R Mux", "single-ended", "LINE2R"}, + {"Right Line2R Mux", "differential", "LINE2R"}, + + {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, + {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, + + {"Right ADC", NULL, "GPIO1 dmic modclk"}, + + /* + * Logical path between digital mic enable and GPIO1 modulator clock + * output function + */ + {"GPIO1 dmic modclk", NULL, "DMic Rate 128"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 64"}, + {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, + + /* Left Line Output */ + {"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right Line Output */ + {"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Left HP Output */ + {"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right HP Output */ + {"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Left HPCOM Output */ + {"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, + + /* Right HPCOM Output */ + {"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, + {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, +}; + +/* For other than tlv320aic3104 */ +static const struct snd_soc_dapm_route intercon_extra_3104[] = { + /* Left Input */ + {"Left PGA Mixer", "Mic2L Switch", "MIC2L"}, + {"Left PGA Mixer", "Mic2R Switch", "MIC2R"}, + + /* Right Input */ + {"Right PGA Mixer", "Mic2L Switch", "MIC2L"}, + {"Right PGA Mixer", "Mic2R Switch", "MIC2R"}, +}; + static const struct snd_soc_dapm_route intercon_mono[] = { /* Mono Output */ {"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"}, @@ -867,17 +984,31 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets, + ARRAY_SIZE(aic3x_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra, + ARRAY_SIZE(intercon_extra)); snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets, ARRAY_SIZE(aic3x_dapm_mono_widgets)); snd_soc_dapm_add_routes(dapm, intercon_mono, ARRAY_SIZE(intercon_mono)); break; case AIC3X_MODEL_3007: + snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets, + ARRAY_SIZE(aic3x_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra, + ARRAY_SIZE(intercon_extra)); snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); snd_soc_dapm_add_routes(dapm, intercon_3007, ARRAY_SIZE(intercon_3007)); break; + case AIC3X_MODEL_3104: + snd_soc_dapm_new_controls(dapm, aic3104_extra_dapm_widgets, + ARRAY_SIZE(aic3104_extra_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, intercon_extra_3104, + ARRAY_SIZE(intercon_extra_3104)); + break; } return 0; @@ -1438,23 +1569,33 @@ static int aic3x_probe(struct snd_soc_codec *codec) aic3x_init(codec); if (aic3x->setup) { - /* setup GPIO functions */ - snd_soc_write(codec, AIC3X_GPIO1_REG, - (aic3x->setup->gpio_func[0] & 0xf) << 4); - snd_soc_write(codec, AIC3X_GPIO2_REG, - (aic3x->setup->gpio_func[1] & 0xf) << 4); + if (aic3x->model != AIC3X_MODEL_3104) { + /* setup GPIO functions */ + snd_soc_write(codec, AIC3X_GPIO1_REG, + (aic3x->setup->gpio_func[0] & 0xf) << 4); + snd_soc_write(codec, AIC3X_GPIO2_REG, + (aic3x->setup->gpio_func[1] & 0xf) << 4); + } else { + dev_warn(codec->dev, "GPIO functionality is not supported on tlv320aic3104\n"); + } } switch (aic3x->model) { case AIC3X_MODEL_3X: case AIC3X_MODEL_33: + snd_soc_add_codec_controls(codec, aic3x_extra_snd_controls, + ARRAY_SIZE(aic3x_extra_snd_controls)); snd_soc_add_codec_controls(codec, aic3x_mono_controls, ARRAY_SIZE(aic3x_mono_controls)); break; case AIC3X_MODEL_3007: + snd_soc_add_codec_controls(codec, aic3x_extra_snd_controls, + ARRAY_SIZE(aic3x_extra_snd_controls)); snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); break; + case AIC3X_MODEL_3104: + break; } /* set mic bias voltage */ @@ -1522,6 +1663,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, { "tlv320aic3106", AIC3X_MODEL_3X }, + { "tlv320aic3104", AIC3X_MODEL_3104 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); -- 1.7.9.5

4 10

[alsa-devel] [PATCH] ASoC: rt286: add irq enable/disable APIs and export
by Jie Yang 04 Feb '15

04 Feb '15

Some platforms, e.g. WSB, don't need irq handler and jack detection when system is in Suspend, for power save reason. Here add irq enable/disable APIs and export them, also disable LDO1, which is used for jack detection when headphone is plugged in. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Reviewed-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 24 ++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 2 ++ 2 files changed, 26 insertions(+) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8104d22..66afb9b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -400,6 +400,30 @@ int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(rt286_mic_detect); +int rt286_enable_irq(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + if (rt286->jack->status | SND_JACK_HEADPHONE) + snd_soc_dapm_force_enable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + enable_irq(rt286->i2c->irq); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_enable_irq); + +int rt286_disable_irq(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + disable_irq(rt286->i2c->irq); + + snd_soc_dapm_disable_pin(&rt286->codec->dapm, "LDO1"); + snd_soc_dapm_sync(&rt286->codec->dapm); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_disable_irq); + static int is_mclk_mode(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h index b539b73..fb18331 100644 --- a/sound/soc/codecs/rt286.h +++ b/sound/soc/codecs/rt286.h @@ -193,6 +193,8 @@ enum { }; int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +int rt286_enable_irq(struct snd_soc_codec *codec); +int rt286_disable_irq(struct snd_soc_codec *codec); #endif /* __RT286_H__ */ -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: Intel: initial scalar variable ba
by Jie Yang 04 Feb '15

04 Feb '15

Reported by Coverity: CID 1267985 CID 1267986 Fix these two Defects: Uninitialized scalar variable. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-firmware.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index a2ae2c5..5f71ef6 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -791,6 +791,7 @@ int sst_module_alloc_blocks(struct sst_module *module) struct sst_block_allocator ba; int ret; + memset(&ba, 0, sizeof(ba)); ba.size = module->size; ba.type = module->type; ba.offset = module->offset; @@ -864,6 +865,7 @@ int sst_module_runtime_alloc_blocks(struct sst_module_runtime *runtime, if (module->persistent_size == 0) return 0; + memset(&ba, 0, sizeof(ba)); ba.size = module->persistent_size; ba.type = SST_MEM_DRAM; -- 1.9.1

2 1

[alsa-devel] [RFC PATCH] ASoC: wm8731: let codec to manage clock by itself
by Bo Shen 04 Feb '15

04 Feb '15

Let the wm8731 codec to manage clock by itself. As all at91 related boards have been switch to CCF framework. So, on the at91 related boards which use wm8731 will let it manage the clock, or else the board use wm8731 is broken. However, at the same time the wm8731 codec is used on other boards, I am sure this change will broken some boards. For example: poodle and corgi based on PXA SoC (in default configuration file, no one use it). DB1200 board which is a mips based board. So I have no idea how to fix them. So, my suggestion is to add CCF check based on the following patch? Any idea or suggestions? Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8731.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b9211b4..83f75d66 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -13,6 +13,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -45,6 +46,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { /* codec private data */ struct wm8731_priv { struct regmap *regmap; + struct clk *mclk; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; @@ -389,6 +391,8 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (clk_id) { case WM8731_SYSCLK_XTAL: case WM8731_SYSCLK_MCLK: + if (clk_set_rate(wm8731->mclk, freq)) + return -EINVAL; wm8731->sysclk_type = clk_id; break; default: @@ -490,6 +494,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + clk_prepare_enable(wm8731->mclk); break; case SND_SOC_BIAS_PREPARE: break; @@ -508,6 +513,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: + clk_disable_unprepare(wm8731->mclk); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -666,6 +672,13 @@ static int wm8731_spi_probe(struct spi_device *spi) if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&spi->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + dev_err(&spi->dev, "Failed to get MCLK\n"); + return ret; + } + mutex_init(&wm8731->lock); wm8731->regmap = devm_regmap_init_spi(spi, &wm8731_regmap); @@ -717,6 +730,13 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + wm8731->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(wm8731->mclk)) { + ret = PTR_ERR(wm8731->mclk); + dev_err(&i2c->dev, "Failed to get MCLK\n"); + return ret; + } + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); -- 2.3.0.rc0

5 10

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix bad comment before intercon_extra_3104 definition
by Jyri Sarha 04 Feb '15

04 Feb '15

The intercon_extra_3104 is obviously for tlv320aic3104. Reported-by: Benoît Thébaudeau <benoit.thebaudeau.dev(a)gmail.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cb92cdb..ed35e8f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -945,7 +945,7 @@ static const struct snd_soc_dapm_route intercon_extra[] = { {"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"}, }; -/* For other than tlv320aic3104 */ +/* For tlv320aic3104 */ static const struct snd_soc_dapm_route intercon_extra_3104[] = { /* Left Input */ {"Left PGA Mixer", "Mic2L Switch", "MIC2L"}, -- 1.9.1

2 1

[alsa-devel] Single vs. multiple seq client ports
by Tim E. Real 04 Feb '15

04 Feb '15

Hi. Our app currently uses a single alsa midi seq client port to address all external midi ports. I have an opportunity now to change this to allow as many client ports as desired, for example one client port for each external port found. If I proceed, it is possible the user may have many client ports if desired. What are the advantages and disadvantages if any, in terms of speed and resources, of having a single client port versus multiple client ports? Thanks. Tim.

2 1

[alsa-devel] [PATCH V3 00/15] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 03 Feb '15

03 Feb '15

This series is an attempt to resolve the CDCLK clock gating issue on Odroid X2/U3 as reported by Daniel Drake [1], by exposing the CDCLK gate clock (and the two other clocks) through clk API. The upside is we can switch Odroid X2/U3 to the simple-card, once the CDCLK clock is taken care of by the clk core and DT. Changes since v2: - skipped the first, already merged patch, - modified description of the patch moving clk_prepare_enable() from DAI to the platform device probe(), - the last patch marked for stable. The patch series has been created on top of branch: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/samsung Changes since the first version: - removed check for the i2s_opclk1 mux input clock while creating the mux and div clocks, - the DT binding documentation changes reworked (addressing review comments), - added include/dt-bindings/sound/samsung-i2s.h header file defining the clk indices, it's been put into a separate patch together with the I2S DT binding documentation updates to make merging of the ASoC and the dts patches separately easier, - a patch fixing compatible strings of I2S1, I2S2 in exynos4.dtsi is included in this series. This whole series may need more testing on other SoCs, so far I only tested it on Odroid Exynos4412 X2, with the I2S working in slave mode. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081753.… Sylwester Nawrocki (15): ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clk provider DT binding documentation ASoC: samsung: i2s: Add clock provider for the I2S internal clocks ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update ARM: dts: Switch Odroid X2/U2 to simple-audio-card ARM: dts: Fix I2S1, I2S2 compatible for exynos4 SoCs .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++ arch/arm/boot/dts/exynos4.dtsi | 13 +- arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 27 +- arch/arm/boot/dts/exynos4412-odroidu3.dts | 8 +- arch/arm/boot/dts/exynos4412-odroidx2.dts | 8 +- include/dt-bindings/sound/samsung-i2s.h | 8 + sound/soc/samsung/i2s.c | 361 ++++++++++++-------- sound/soc/samsung/odroidx2_max98090.c | 6 +- 8 files changed, 295 insertions(+), 158 deletions(-) create mode 100644 include/dt-bindings/sound/samsung-i2s.h -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: sgtl5000: add delay before first I2C access
by Eric Nelson 03 Feb '15

03 Feb '15

To quote from section 1.3.1 of the data sheet: The SGTL5000 has an internal reset that is deasserted 8 SYS_MCLK cycles after all power rails have been brought up. After this time, communication can start ... 1.0μs represents 8 SYS_MCLK cycles at the minimum 8.0 MHz SYS_MCLK. Signed-off-by: Eric Nelson <eric.nelson(a)boundarydevices.com> --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 29cf7ce..7d65a5b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1462,6 +1462,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (ret) return ret; + /* Need 8 clocks before I2C accesses */ + usleep_range(1, 2); + /* read chip information */ ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); if (ret) -- 1.9.1

4 7

[alsa-devel] [PATCH] add Steinberg UR44 USB Audio Interface support to quirks-table
by Cyrus Harmon 03 Feb '15

03 Feb '15

Signed-off-by: Cyrus Harmon <cyrus(a)bobobeach.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..987ca01 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -436,6 +436,36 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR44", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- 1.8.3.2

5 32

[alsa-devel] [PATCH 2/2] ALSA: line6: fixup of line6_start_timer argument type
by Nicholas Mc Guire 03 Feb '15

03 Feb '15

line6_start_timer passes an unsigned int as argument to be used in mod_timer which is then used by mod_timer as unsigned long, this just fixes up the argument type. This change helps make static code checkers happy. Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- The change of argument type was checked against all call-sites - in all cases it is passing in a constant (either VARIAX_STARTUP_DELAY1 or POD_STARTUP_DELAY) which should be fine with the argument type change. Patch was compile tested for x86_64_defconfig + CONFIG_SOUND=y, CONFIG_SND_USB_POD=m (implies CONFIG_SND_USB_LINE6=y) Patch is against 3.0.19-rc6 (localversion = -next-20150202) sound/usb/line6/driver.c | 2 +- sound/usb/line6/driver.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 3cf12cc..e2a2603 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -162,7 +162,7 @@ static int line6_send_raw_message_async_part(struct message *msg, /* Setup and start timer. */ -void line6_start_timer(struct timer_list *timer, unsigned int msecs, +void line6_start_timer(struct timer_list *timer, unsigned long msecs, void (*function)(unsigned long), unsigned long data) { setup_timer(timer, function, data); diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index fa877a3..2276b78 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -159,7 +159,7 @@ extern int line6_send_sysex_message(struct usb_line6 *line6, const char *buffer, int size); extern ssize_t line6_set_raw(struct device *dev, struct device_attribute *attr, const char *buf, size_t count); -extern void line6_start_timer(struct timer_list *timer, unsigned int msecs, +extern void line6_start_timer(struct timer_list *timer, unsigned long msecs, void (*function)(unsigned long), unsigned long data); extern int line6_version_request_async(struct usb_line6 *line6); -- 1.7.10.4

2 1

[alsa-devel] [PATCH 1/2] ALSA: line6: use msecs_to_jiffies for conversion
by Nicholas Mc Guire 03 Feb '15

03 Feb '15

This is only an API consolidation and should make things more readable it replaces var * HZ / 1000 by msecs_to_jiffies(var). Signed-off-by: Nicholas Mc Guire <hofrat(a)osadl.org> --- Converting milliseconds to jiffies by val * HZ / 1000 is technically not wrong but msecs_to_jiffies(val) is the cleaner solution and handles all corner cases correctly. This is a minor API cleanup only. Patch was only compile tested for x86_64_defconfig + CONFIG_SOUND=y, CONFIG_SND_USB_POD=m (implies CONFIG_SND_USB_LINE6=y) Patch is against 3.0.19-rc6 (localversion = -next-20150202) sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 2328ec9..3cf12cc 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -166,7 +166,7 @@ void line6_start_timer(struct timer_list *timer, unsigned int msecs, void (*function)(unsigned long), unsigned long data) { setup_timer(timer, function, data); - mod_timer(timer, jiffies + msecs * HZ / 1000); + mod_timer(timer, jiffies + msecs_to_jiffies(msecs)); } EXPORT_SYMBOL_GPL(line6_start_timer); -- 1.7.10.4

2 1

[alsa-devel] [PATCH v4 0/3] Sound support for Exynos4412 Trats2 board
by Inha Song 03 Feb '15

03 Feb '15

This patch-set adds basic sound support for the Trats2 boards. It just support primary I2s and external speaker playback. Changes in v4: - Move to clock out parent settings in exynos4412-trats2.dts - Remove unnecessary lines in Trats2 machine driver. - Change name of dai stream to more cleary. - Add MFD_WM8994 to select in machine driver kconfig. Changes in v3: - Set clock out parent to XUSBXTI in pmu_system_controller DT node. Changes in v2: - Remove unnecessary lines in Trats2 machine driver. - move clock enable/disable to .startup / .shutdown. - Fix code style issues. - Remove mclk DT property in Trats2 sound and add MCLK1 property in WM1811 node. (MCLK1 is optional property in WM8994 node - refer: Document/devicetree/bindings/sound/wm8994.txt) - Use mclk that defined in WM1811 DT node. - Add "samsung,audio-codec" property to required properties. Inha Song (3): ASoC: samsung: Add machine driver for Trats2 ASoC: samsung: Document Trats2 audio subsystem bindings ARM: dts: Add sound nodes for exynos4412-trats2 .../bindings/sound/samsung,trats2-wm1811.txt | 25 +++ arch/arm/boot/dts/exynos4412-trats2.dts | 42 ++++ sound/soc/samsung/Kconfig | 9 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/trats2_wm1811.c | 211 +++++++++++++++++++++ 5 files changed, 289 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,trats2-wm1811.txt create mode 100644 sound/soc/samsung/trats2_wm1811.c -- 2.0.0.390.gcb682f8

4 11

[alsa-devel] [PATCH v2] ASoC: Add sysfs entries via static attribute groups
by Takashi Iwai 02 Feb '15

02 Feb '15

Instead of calling device_create_file() manually, assign the static attribute group entries at the device registration. This simplifies the error handling and avoids the possible races. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Enable attributes conditionally via is_visible callback include/sound/soc-dapm.h | 2 +- sound/soc/soc-core.c | 58 ++++++++++++++++++++++++++++-------------------- sound/soc/soc-dapm.c | 14 ++++-------- 3 files changed, 39 insertions(+), 35 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 89823cfe6f04..e8e0220404d4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -405,7 +405,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_update *update); /* dapm sys fs - used by the core */ -int snd_soc_dapm_sys_add(struct device *dev); +extern struct attribute *soc_dapm_dev_attrs[]; void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2c62620abca6..4bd4d77d4ba8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -191,6 +191,39 @@ static ssize_t pmdown_time_set(struct device *dev, static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); +static struct attribute *soc_dev_attrs[] = { + &dev_attr_codec_reg.attr, + &dev_attr_pmdown_time.attr, + NULL +}; + +static umode_t soc_dev_attr_is_visible(struct kobject *kobj, + struct attribute *attr, int idx) +{ + struct device *dev = kobj_to_dev(kobj); + struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); + + if (attr == &dev_attr_pmdown_time.attr) + return attr->mode; /* always visible */ + return rtd->codec ? attr->mode : 0; /* enabled only with codec */ +} + +static const struct attribute_group soc_dapm_dev_group = { + .attrs = soc_dapm_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group soc_dev_roup = { + .attrs = soc_dev_attrs, + .is_visible = soc_dev_attr_is_visible, +}; + +static const struct attribute_group *soc_dev_attr_groups[] = { + &soc_dapm_dev_group, + &soc_dev_roup, + NULL +}; + #ifdef CONFIG_DEBUG_FS static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) @@ -949,8 +982,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(rtd->dev, &dev_attr_pmdown_time); - device_remove_file(rtd->dev, &dev_attr_codec_reg); device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1120,6 +1151,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; + rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); @@ -1136,23 +1168,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return ret; } rtd->dev_registered = 1; - - if (rtd->codec) { - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", - ret); - - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", - ret); - } - return 0; } @@ -1308,11 +1323,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } #endif - ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); - if (ret < 0) - dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", - ret); - if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ ret = soc_new_compress(rtd, num); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..6c5cf7d1ff54 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2127,15 +2127,10 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -int snd_soc_dapm_sys_add(struct device *dev) -{ - return device_create_file(dev, &dev_attr_dapm_widget); -} - -static void snd_soc_dapm_sys_remove(struct device *dev) -{ - device_remove_file(dev, &dev_attr_dapm_widget); -} +struct attribute *soc_dapm_dev_attrs[] = { + &dev_attr_dapm_widget.attr, + NULL +}; static void dapm_free_path(struct snd_soc_dapm_path *path) { @@ -3903,7 +3898,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) */ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_sys_remove(dapm->dev); dapm_debugfs_cleanup(dapm); dapm_free_widgets(dapm); list_del(&dapm->list); -- 2.2.2

2 1

[alsa-devel] [PATCH] ASoC: rx51: Correct DAPM route for capture
by Peter Ujfalusi 02 Feb '15

02 Feb '15

"Mic Bias" is DAPM_SUPPLY so it has to be connected in the route accordingly. Fixes audio capture on the board. Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Tested-by: Pavel Machek <pavel(a)ucw.cz> --- sound/soc/omap/rx51.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 04896d6252a2..7f299357c2d2 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -250,14 +250,14 @@ static const struct snd_soc_dapm_route audio_map[] = { {"FM Transmitter", NULL, "LLOUT"}, {"FM Transmitter", NULL, "RLOUT"}, - {"DMic Rate 64", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "DMic"}, + {"DMic Rate 64", NULL, "DMic"}, + {"DMic", NULL, "Mic Bias"}, {"b LINE2R", NULL, "MONO_LOUT"}, {"Earphone", NULL, "b HPLOUT"}, - {"LINE1L", NULL, "b Mic Bias"}, - {"b Mic Bias", NULL, "HS Mic"} + {"LINE1L", NULL, "HS Mic"}, + {"HS Mic", NULL, "b Mic Bias"}, }; static const char * const spk_function[] = {"Off", "On"}; -- 2.2.2

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Re: [alsa-devel] Digital Input on ALC3661 (from laptop HDMI IN) not working under Linux
by Oliver Freyermuth 02 Feb '15

02 Feb '15

Am 02.02.2015 um 08:05 schrieb Raymond Yau: > >> >> >> I have an Alienware 17 laptop (sometimes called M17RX5) which has a >> > Realtek ALC3661. >> >>The problem I describe here should also be valid for the Alienware 18 >> > (all the late 2013 models, which are still the most current). At least >> > some scattered forum reports suggest so. >> > >> >>The card works almost perfect in Linux, but one functionality is >> > missing: The laptop features an HDMI input, and the Realtek card is >> > supposed to take care of the incoming digital audio. >> > This works fine in Windows, but in Linux, I only get the video and the >> > audio stays missing. >> > >> >> Here you can find my alsa-info: >> >>http://www.alsa-project.org/db/?f=0a0502e49212a0436a30fde9d4ee56163bbc2ebc > <http://www.alsa-project.org/db/?f=0a0502e49212a0436a30fde9d4ee56163bbc2ebc> >> > >> > The only digial input is node 0x0a >> > >> > Try hda-jack-retask to change node 0x1f with pin default [Jack] > digital In >> >> Thanks a lot for your suggestion! >> I tried, 0x0a stays as "not present" but I can select it in alsa-mixer >> as expected. Sadly, I can only record silence. >> Also setting Jack Detection to "Present" in the advanced overrides of >> hdajackretask did not help. > > http://git.alsa-project.org/?p=alsa-tools.git;a=blob;f=hdajackretask/README… > > You have to select "show unconnected pin" and "advanced override" > > Change the pin default of node 0x1f to 0x3c61130 [Jack] SPDIF In > > Set as boot default and reboot > > Digital capture device 1 will be created after you fix the pin default > of node 0x1f > > Turn on iec958 capture switch and try > > arecord -D IEC958:CARD=PCH -f dat -v test.wav Thanks a lot for your continued help! Here's what I did - the sad news, I still record only silence: 1. use hdajackretask as you explained, apply as boot default. 2. I got the following files: /etc/modprobe.d/hda-jack-retask.conf (below the automatic comment): options snd-hda-intel patch=hda-jack-retask.fw,hda-jack-retask.fw,hda-jack-retask.fw,hda-jack-retask.fw /lib64/firmware/hda-jack-retask.fw: [codec] 0x10ec0668 0x102805aa 0 [pincfg] 0x12 0x99a30140 0x14 0x90170110 0x15 0x0321101f 0x16 0x03211020 0x18 0x40030008 0x19 0x03a11030 0x1a 0x411111f0 0x1b 0x411111f0 0x1d 0x41000001 0x1e 0x411111f0 0x1f 0x03c61130 It contains exactly the line 0x1f 0x03c61130, so I hope I clicked correctly in the GUI. 3. After reboot, I see in dmesg (I assume this means it worked): [ 6.021374] snd_hda_intel 0000:00:1b.0: Applying patch firmware 'hda-jack-retask.fw' [ 6.021468] snd_hda_intel 0000:01:00.1: Disabling MSI [ 6.021471] snd_hda_intel 0000:01:00.1: Applying patch firmware 'hda-jack-retask.fw' [ 6.022749] snd_hda_intel 0000:00:1b.0: irq 29 for MSI/MSI-X 4. I activate S/PDIF capture in alsamixer (IEC958 is not shown there, but S/PDIF appeared new after the change). 5. arecord -L tells me at the end: iec958:CARD=PCH,DEV=0 HDA Intel PCH, ALC3661 Digital IEC958 (S/PDIF) Digital Audio Output 6. I run, as suggested: arecord -D iec958:CARD=PCH -f dat -v test.wav Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo Hooks PCM Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 Slave: Hardware PCM card 1 'HDA Intel PCH' device 1 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 1 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 appl_ptr : 0 hw_ptr : 0 But I get only silence, I tried to first attach the HDMI + switch to the video and start recording when everything is connected (i.e. when I would hear audio in Windows). I also tried to just record with the HDMI-cable connected and audio coming in (but not switched to video - in this case, I would hear nothing in Windows, but the audio should still arrive). Both cases, I record just silence. Any further ideas? Thanks a lot for your assistance! > >> > >> > Node 0x1f [Pin Complex] wcaps 0x400681: Stereo Digital >> > Pincap 0x00000020: IN >> > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear >> > Conn = 1/8, Color = Black >> > DefAssociation = 0xf, Sequence = 0x0 >> > Misc = NO_PRESENCE >> > Pin-ctls: 0x20: IN >> > Unsolicited: tag=00, enabled=0 >> > Power states: D0 D1 D2 D3 EPSS >> > >> >>In Windows, I observe the HDMI-IN audio routing to the speakers is only >> > done if the Realtek Audio Service is running. In Linux, I observe >> > keypress-events when I either switch manually to the HDMI-in signal > and also >> > if a new signal is detected (something (the UEFI?) injects a keypress >> > then). >> > I assume the service in Windows catches these events and changes >> > "something" in the Realtek Audio Settings. I see no special recording >> > device in Windows, nor any explicit HDMI-input setting, nor any related >> > setting in the driver. Dell links a standard Realtek driver in their >> > support section. >> >

1 0

[alsa-devel] [PATCH 0/2] ASoC: davinci-mcasp: Fixes for 3.20
by Peter Ujfalusi 02 Feb '15

02 Feb '15

Hi, While doing the conversion of davinci DAI drivers to use edma-pcm I have noticed couple of issues with the driver regarding to old daVinci device support which resulted in the following two patch. Some SoC uses shared IRQ line for all McASP instances There are SoCs where a given McASP instance only supports DIT mode. Currently the arch code feeds dummy resource in place of RX (which is not available). With the second patch this is not going to be needed anymore. If it is not too late for 3.20.. regards, Peter --- Peter Ujfalusi (2): ASoC: davicni-mcasp: Mark the common irq line as shared ASoC: davinci-mcasp: Fix DIT only McASP instance support sound/soc/davinci/davinci-mcasp.c | 75 +++++++++++++++++++++------------------ 1 file changed, 41 insertions(+), 34 deletions(-) -- 2.2.2

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[alsa-devel] [PATCH 0/2] ASoC: davinci-evm: Remove not needed header files
by Peter Ujfalusi 02 Feb '15

02 Feb '15

Hi, Trivial changes to remove headers should not be included in the machine driver. The first patch was sent as part of the edma header cleanup series in November: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084734.h… http://mailman.alsa-project.org/pipermail/alsa-devel/2014-November/084735.h… The series missed 3.19 and it seams to not going to make it to 3.20 either.. With these header files gone from the davinci-evm I will be able to send my davinci audio conversion from davinci-pcm to edma-pcm for 3.21. Mark: is it possible to take these for 3.20 still? Regards, Peter --- Peter Ujfalusi (2): ASoC: davinci-evm: Do not include edma headers ASoC: davinci-evm: Do not include davinci-pcm.h and davinci-i2s.h sound/soc/davinci/davinci-evm.c | 6 ------ 1 file changed, 6 deletions(-) -- 2.2.2

2 4

[alsa-devel] [PATCH v4 00/10] audio timestamping evolutions
by Pierre-Louis Bossart 02 Feb '15

02 Feb '15

This series of patches was inspired by recent threads on the alsa mailing list, as well issues detected with existing and upcoming hardware: 1. there was a request from the PulseAudio community to get more information from drivers to make rewinds safer. While the conclusion was that it's nearly impossible for a driver developer to provide this information, there are however ways to assess the granularity of the hw_ptr updates using timestamping capabilities, and indirectly understand how safe rewinds might be. 2. There was also a request to add a start_at capability based either on system hr timers or a wall clock, which requires a means to expose both types of information to applications. Rather than adding new sets of timestamps, it is suggested the new start_at functionality relies on the new definition provides by these patches 3. For new hardware, there is a neeed to let low-level drivers handle timestamping instead of having the ALSA core do it. Similarly there is a need to let the low-level driver update the initial estimate for the trigger timestamp if there are delays to start a stream (eg. with USB) These patches try to provide an answer to these multiple needs by building on the work done two years ago to expose wall clock information to applications. The evolution is to let application select which audio timestamp they are interested in, track the delay and drifts between recurring measurements and get, when possible, an idea of the accuracy of the underlying hardware. A backwards compatible mode is provided in case userspace doesn't provide any timestamp selection (results based on HDAudio wallclock for playback, hw_ptr in all other cases). The first 4 patches are corrections for misses in the way the system and trigger timestamps are handled, and the last 6 provide the additional audio timestamping selection. A second batch is planned to enable hardware capabilities in a low-level drivers. A corresponding set of patches is available for alsa-lib. V2 changes: trigger_tstamp: trigger_tstamp_latched, pending redefined as bool trigger_tstamp_latched reset in snd_pcm_pre_start() audio_ts_config, report: keep separate structure but use different bitfields for in and out. use u32 instead of __u32, add comments that these structures are internal COMPAT backwards compatible mode, uses WALL_CLOCK/LINK for HDAudio playback and DEFAULT (hw_ptr) everywhere else INFO bits: reclaimed 32-bits from hw_params, renamed as info_ext moved all timestamp info to info_ext snd_pcm_status: read only 32-bit audio_tstamp_data, ignore all other fields V3/4: Addressed feedback from Jaroslav: no change to STATUS ioctl, new functionality introduced in STATUS_EXT ioctl with r/w params. bumped PCM protocol for detection on STATUS_EXT in userspace used 32-bit word for accuracy report, no mantissa/exponent packing rolled back info_ext changes, all INFO fields remain in same word Merged comments from Liam (code simplifications) fixed packing Added driver_tstamp field in case there is a delay in passing the tstamp and audio_tstamp over IPC. Pierre-Louis Bossart (10): ALSA: core: don't override timestamp unconditionally ALSA: core: allow for trigger_tstamp snapshot in .trigger ALSA: hda: read trigger_timestamp immediately after starting DMA ALSA: usb: update trigger timestamp on first non-zero URB submitted ALSA: core: selection of audio_tstamp type and accuracy reports ALSA: core: pass audio tstamp config from userspace ALSA: core: pass audio tstamp config from userspace in compat mode ALSA: core: replace .wall_clock by .get_time_info ALSA: hda: replace .wallclock by .get_time_info ALSA: bump PCM protocol to 2.0.13 Documentation/sound/alsa/timestamping.txt | 200 ++++++++++++++++++++++++++++++ include/sound/pcm.h | 67 +++++++++- include/uapi/sound/asound.h | 36 +++++- sound/core/pcm_compat.c | 36 +++++- sound/core/pcm_lib.c | 88 ++++++++----- sound/core/pcm_native.c | 58 ++++++++- sound/pci/hda/hda_controller.c | 43 +++++-- sound/usb/pcm.c | 9 ++ 8 files changed, 478 insertions(+), 59 deletions(-) create mode 100644 Documentation/sound/alsa/timestamping.txt -- 1.9.1

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[alsa-devel] [PATCH 00/14] Refactoring ALSA core device management
by Takashi Iwai 02 Feb '15

02 Feb '15

Hi, this is a series of relatively small patches for refactoring the ALSA core device management. I experimented a similar thing some time ago but didn't merge it in the end due to too intrusive changes. At this time, the approach is rather minimalistic: only the device registration part is replaced by embedding the struct device into each ALSA device object. To the outside of ALSA core, there shouldn't be many visible changes. The only possible breakage would be an access to some renamed / dropped fields by a downstream driver. But it must be easy to correct. Some patches in the series are merely cleanup patches that can be applied individually, but included just for completeness. The whole patches are found in test/snd-device branch of sound git tree, too. Takashi === Takashi Iwai (14): ALSA: Allow to pass the device object to snd_register_device*() ALSA: control: Provide a helper to look for the preferred subdevice ALSA: Add a helper to initialize device ALSA: control: Embed struct device ALSA: hwdep: Embed struct device ALSA: pcm: Embed struct device ALSA: rawmidi: Embed struct device ALSA: rawmidi: Use rawmidi device file for kernel messages ALSA: timer: Propagate the error at initialization ALSA: timer: Handle the device directly ALSA: seq: Handle the device directly ALSA: compress: Embed struct device ALSA: Simplify snd_device_register() variants ALSA: Drop snd_get_device() helper include/sound/compress_driver.h | 4 +- include/sound/control.h | 11 +++- include/sound/core.h | 41 ++---------- include/sound/hwdep.h | 3 +- include/sound/pcm.h | 2 +- include/sound/rawmidi.h | 4 +- sound/aoa/soundbus/i2sbus/pcm.c | 5 +- sound/core/compress_offload.c | 26 ++++++-- sound/core/control.c | 66 +++++++++++-------- sound/core/hwdep.c | 88 ++++++++++--------------- sound/core/init.c | 19 ++++++ sound/core/pcm.c | 70 ++++++-------------- sound/core/rawmidi.c | 47 +++++++------- sound/core/seq/seq_clientmgr.c | 14 +++- sound/core/sound.c | 112 ++++++++++---------------------- sound/core/timer.c | 42 ++++++++---- sound/pci/hda/hda_controller.c | 3 +- sound/pci/hda/hda_hwdep.c | 7 +- sound/soc/intel/sst-mfld-platform-pcm.c | 1 - 19 files changed, 265 insertions(+), 300 deletions(-) -- 2.2.2

3 20

[alsa-devel] [PATCH v2] ASoC: tlv320aic3x: Fix data delay configuration
by Peter Ujfalusi 02 Feb '15

02 Feb '15

The CTRLC register were not receiving the correct delay configuration, which will corrupt DSP_A audio mode. Fixes: 368494093354 ("ASoC: tlv320aic3x: Add TDM support") Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Tested-by: Pavel Machek <pavel(a)ucw.cz> Tested-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Changes since v1: Fixes tag and Tested-by from Pavel and Jarkko added to commit message. Regards, Peter sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a78ad361f494..07603d142923 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, delay += aic3x->tdm_delay; /* Configure data delay */ - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } -- 2.2.2

2 2

[alsa-devel] [PATCH] ASoC: atmel-pcm-dma: won't check direction when configure dma
by Bo Shen 02 Feb '15

02 Feb '15

As DMA framework request DMA using direction only in prep_slave function, (The At91 xdma driver has adapted to this request). So won't check direction when do DMA configuration. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/atmel/atmel-pcm-dma.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 33fb3bb..b8e7bad 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -105,13 +105,11 @@ static int atmel_pcm_configure_dma(struct snd_pcm_substream *substream, return ret; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - slave_config->dst_addr = ssc->phybase + SSC_THR; - slave_config->dst_maxburst = 1; - } else { - slave_config->src_addr = ssc->phybase + SSC_RHR; - slave_config->src_maxburst = 1; - } + slave_config->dst_addr = ssc->phybase + SSC_THR; + slave_config->dst_maxburst = 1; + + slave_config->src_addr = ssc->phybase + SSC_RHR; + slave_config->src_maxburst = 1; prtd->dma_intr_handler = atmel_pcm_dma_irq; -- 2.3.0.rc0

2 1

[alsa-devel] [PATCH] MAINTAINERS: ASoC: add maintainer for Intel BDW/HSW ASoC driver
by Jie Yang 02 Feb '15

02 Feb '15

Adding myself as the Intel BDW/HSW ASoC driver maintainer. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Acked-by: Liam Girdwood <lgirdwood(a)gmail.com> --- MAINTAINERS | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index dc1e9a8..78a0913 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4943,6 +4943,16 @@ F: Documentation/input/multi-touch-protocol.txt F: drivers/input/input-mt.c K: \b(ABS|SYN)_MT_ +INTEL ASoC BDW/HSW DRIVERS +M: Jie Yang <yang.jie(a)linux.intel.com> +L: alsa-devel(a)alsa-project.org +S: Supported +F: sound/soc/intel/sst-haswell* +F: sound/soc/intel/sst-dsp* +F: sound/soc/intel/sst-firmware.c +F: sound/soc/intel/broadwell.c +F: sound/soc/intel/haswell.c + INTEL C600 SERIES SAS CONTROLLER DRIVER M: Intel SCU Linux support <intel-linux-scu(a)intel.com> M: Artur Paszkiewicz <artur.paszkiewicz(a)intel.com> -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] Remove unused functions
by Chris Rorvick 01 Feb '15

01 Feb '15

Found a couple more things to rip out. Regards, Chris Chris Rorvick (2): ALSA: line6: Remove unused line6_midibuf_status() ALSA: line6: Remove unused line6_midibuf_skip_message() sound/usb/line6/midibuf.c | 18 ------------------ 1 file changed, 18 deletions(-) -- 2.1.0

2 4

[alsa-devel] [PATCH 0/3] snd-bcd2000: add audio support and control interface
by Mario Kicherer 01 Feb '15

01 Feb '15

The current version of the driver only provides a MIDI interface to read controller status and switch LEDs. These patches add audio playback and capture support as well as a control interface to switch between Phono A and Mic input.

2 7

[alsa-devel] [PATCH alsa-lib] conf: add support for USB AudioSport Quattro (tm)
by Alexander E. Patrakov 01 Feb '15

01 Feb '15

This patch resulted from discussion with redlion_ on #alsa IRC channel on Freenode. 4-channel playback now works. 4-channel capture works too, but not simultaneously with playback (hardware limitation). Alsa-info before the fix: http://www.alsa-project.org/db/?f=a3673622074b88a1abf4ccc6e7f37d0b5b72f34a Signed-off-by: Alexander E. Patrakov <patrakov(a)gmail.com> --- src/conf/cards/USB-Audio.conf | 1 + 1 file changed, 1 insertion(+) diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 05e5d09..4cbf85e 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -30,6 +30,7 @@ USB-Audio.pcm.surround40_type { "OmniStudio" two_stereo_devices "Quattro" two_stereo_devices "SB Audigy 2 NX" six_channels + "USB AudioSport Quattro (tm)" two_stereo_devices } # If a device does not use the first PCM device for digital data, the device -- 2.2.1

2 1

[alsa-devel] [PATCH v2 12/12] dt-bindings: Add documentation for Rockchip dw-hdmi-audio
by Yakir Yang 31 Jan '15

31 Jan '15

Required properties: - compatible: platform specific - cpu-of-node: the device node of cpu_dai Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Changes in v2: - remove codec-name and codec-dai-name - rename rockchip,rockchip-hdmi-audio.txt to rockchip,rockchip-dw-hdmi-audio.txt .../bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt diff --git a/Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt b/Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt new file mode 100644 index 0000000..5b86eed --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rockchip-dw-hdmi-audio.txt @@ -0,0 +1,12 @@ +Rockchip hdmi audio bindings + +Required properties: +- compatible: platform specific +- cpu-of-node: the device node of cpu_dai + +Example: + +sound { + compatible = "rockchip,rk3288-hdmi-audio"; + cpu-of-node = <&i2s>; +}; -- 2.1.2

3 2

[alsa-devel] [PATCH v2 10/12] ASoC: dw-hdmi-audio: add codec driver for dw hdmi audio
by Yakir Yang 31 Jan '15

31 Jan '15

codec driver creat an standard alsa device, than config audio and report jack status through some callback interfaces that dw_hdmi driver support. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Changes in v2: - Update dw_hdmi audio control interfaces, and adjust jack report process sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/dw-hdmi-audio.c | 314 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/dw-hdmi-audio.h | 17 +++ 4 files changed, 337 insertions(+) create mode 100644 sound/soc/codecs/dw-hdmi-audio.c create mode 100644 sound/soc/codecs/dw-hdmi-audio.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8349f98..b34dd12 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_DW_HDMI_AUDIO select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 @@ -459,6 +460,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_DW_HDMI_AUDIO + tristate + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bbdfd1e..0ebb664 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -68,6 +68,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-dw-hdmi-audio-objs := dw-hdmi-audio.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o @@ -249,6 +250,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_DW_HDMI_AUDIO) += snd-soc-dw-hdmi-audio.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o diff --git a/sound/soc/codecs/dw-hdmi-audio.c b/sound/soc/codecs/dw-hdmi-audio.c new file mode 100644 index 0000000..ed046ce --- /dev/null +++ b/sound/soc/codecs/dw-hdmi-audio.c @@ -0,0 +1,314 @@ +/* + * dw-hdmi-codec.c + * + * DesignerWare ALSA SoC DAI driver for DW HDMI audio. + * Copyright (c) 2014, CORPORATION. All rights reserved. + * Authors: Yakir Yang <ykk(a)rock-chips.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>.* + * + */ +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/moduleparam.h> + +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> + +#include <drm/bridge/dw_hdmi.h> +#include "dw-hdmi-audio.h" + +struct snd_dw_hdmi { + struct device *dev; + struct dw_hdmi_audio_data data; + + u8 jack_status; + struct snd_soc_jack *jack; + struct delayed_work jack_work; +}; + +/** + * dw_hdmi_jack_detect - Enable hdmi detection via the HDMI IRQ + * + * @codec: HDMI codec + * @jack: jack to report detection events on + * + * Enable HDMI detection via IRQ on the HDMI. + * + * If no jack is supplied detection will be disabled. + */ +int dw_hdmi_jack_detect(struct snd_soc_codec *codec_dai, + struct snd_soc_jack *jack) +{ + struct snd_dw_hdmi *hdmi = snd_soc_codec_get_drvdata(codec_dai); + + hdmi->jack = jack; + hdmi->jack_status = 0; + + snd_soc_jack_report(hdmi->jack, 0, SND_JACK_LINEOUT); + + queue_delayed_work(system_power_efficient_wq, &hdmi->jack_work, + msecs_to_jiffies(50)); + + return 0; +} +EXPORT_SYMBOL_GPL(dw_hdmi_jack_detect); + +static void dw_hdmi_jack_work(struct work_struct *work) +{ + struct snd_dw_hdmi *hdmi = container_of(work, struct snd_dw_hdmi, + jack_work.work); + u8 jack_status, status; + + status = hdmi->data.get_connect_status(hdmi->data.dw); + jack_status = status ? SND_JACK_LINEOUT : 0; + if (jack_status != hdmi->jack_status) { + snd_soc_jack_report(hdmi->jack, jack_status, + SND_JACK_LINEOUT); + hdmi->jack_status = jack_status; + + dev_dbg(hdmi->dev, "jack report [%d]\n", hdmi->jack_status); + } +} + +static irqreturn_t snd_dw_hdmi_irq(int irq, void *dev_id) +{ + struct snd_dw_hdmi *hdmi = dev_id; + + queue_delayed_work(system_power_efficient_wq, &hdmi->jack_work, + msecs_to_jiffies(50)); + + return IRQ_HANDLED; +} + +static int dw_hdmi_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_dw_hdmi *hdmi = snd_soc_dai_get_drvdata(codec_dai); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hdmi_audio_fmt hdmi_fmt; + unsigned int fmt, rate, chan, width; + + fmt = rtd->dai_link->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK; + switch (fmt) { + case SND_SOC_DAIFMT_I2S: + hdmi_fmt.dai_fmt = AUDIO_DAIFMT_IIS; + break; + case SND_SOC_DAIFMT_LEFT_J: + hdmi_fmt.dai_fmt = AUDIO_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + hdmi_fmt.dai_fmt = AUDIO_DAIFMT_RIGHT_J; + break; + default: + dev_err(codec_dai->dev, "DAI format unsupported"); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: dai_fmt = %d.\n", fmt); + + width = params_width(params); + switch (width) { + case 16: + case 24: + hdmi_fmt.word_length = width; + break; + default: + dev_err(codec_dai->dev, "width[%d] not support!\n", width); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: word_length = %d.\n", width); + + chan = params_channels(params); + switch (chan) { + case 2: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_2; + break; + case 4: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_4; + break; + case 6: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_6; + break; + case 8: + hdmi_fmt.chan_num = AUDIO_CHANNELNUM_8; + break; + default: + dev_err(codec_dai->dev, "channel[%d] not support!\n", chan); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: chan_num = %d.\n", chan); + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + case 176400: + case 192000: + hdmi_fmt.sample_rate = rate; + break; + default: + dev_err(codec_dai->dev, "rate[%d] not support!\n", rate); + return -EINVAL; + } + dev_dbg(codec_dai->dev, "[codec_dai]: sample_rate = %d.\n", rate); + + hdmi->data.config_audio(hdmi->data.dw, &hdmi_fmt); + + return 0; +} + +static int dw_hdmi_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_dw_hdmi *hdmi = snd_soc_dai_get_drvdata(codec_dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + hdmi->data.enable_clk(hdmi->data.dw); + dev_dbg(codec_dai->dev, "[codec_dai]: trigger enable.\n"); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + hdmi->data.disable_clk(hdmi->data.dw); + dev_dbg(codec_dai->dev, "[codec_dai]: trigger disable.\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget dw_hdmi_widgets[] = { + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route dw_hdmi_routes[] = { + { "TX", NULL, "Playback" }, +}; + +static const struct snd_soc_dai_ops dw_hdmi_dai_ops = { + .hw_params = dw_hdmi_dai_hw_params, + .trigger = dw_hdmi_dai_trigger, +}; + +static struct snd_soc_dai_driver dw_hdmi_audio_dai = { + .name = "dw-hdmi-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &dw_hdmi_dai_ops, +}; + +static const struct snd_soc_codec_driver dw_hdmi_audio = { + .dapm_widgets = dw_hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(dw_hdmi_widgets), + .dapm_routes = dw_hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(dw_hdmi_routes), +}; + +static int dw_hdmi_audio_probe(struct platform_device *pdev) +{ + struct dw_hdmi_audio_data *data = pdev->dev.platform_data; + struct snd_dw_hdmi *hdmi; + int ret; + + hdmi = devm_kzalloc(&pdev->dev, sizeof(*hdmi), GFP_KERNEL); + if (!hdmi) + return -ENOMEM; + + hdmi->data = *data; + hdmi->dev = &pdev->dev; + platform_set_drvdata(pdev, hdmi); + + INIT_DELAYED_WORK(&hdmi->jack_work, dw_hdmi_jack_work); + + ret = devm_request_irq(&pdev->dev, hdmi->data.irq, snd_dw_hdmi_irq, + IRQF_SHARED, "dw-hdmi-audio", hdmi); + if (ret) { + dev_err(&pdev->dev, "request irq failed (%d)\n", ret); + goto free_hdmi_data; + } + + ret = snd_soc_register_codec(&pdev->dev, &dw_hdmi_audio, + &dw_hdmi_audio_dai, 1); + if (ret) { + dev_err(&pdev->dev, "register codec failed (%d)\n", ret); + goto free_irq; + } + + dev_info(&pdev->dev, "hdmi audio init success.\n"); + + return 0; + +free_irq: + devm_free_irq(&pdev->dev, hdmi->data.irq, hdmi); +free_hdmi_data: + devm_kfree(&pdev->dev, hdmi); + + return ret; +} + +static int dw_hdmi_audio_remove(struct platform_device *pdev) +{ + struct snd_dw_hdmi *hdmi = platform_get_drvdata(pdev); + + snd_soc_unregister_codec(&pdev->dev); + devm_free_irq(&pdev->dev, hdmi->data.irq, hdmi); + devm_kfree(&pdev->dev, hdmi); + + return 0; +} + +static const struct of_device_id dw_hdmi_audio_ids[] = { + { .compatible = "dw-hdmi-audio", }, + { } +}; + +static struct platform_driver dw_hdmi_audio_driver = { + .driver = { + .name = "dw-hdmi-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(dw_hdmi_audio_ids), + }, + .probe = dw_hdmi_audio_probe, + .remove = dw_hdmi_audio_remove, +}; +module_platform_driver(dw_hdmi_audio_driver); + +MODULE_AUTHOR("Yakir Yang <ykk(a)rock-chips.com>"); +MODULE_DESCRIPTION("DW HDMI Audio ASoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" "dw-hdmi-audio"); +MODULE_DEVICE_TABLE(of, dw_hdmi_audio_ids); diff --git a/sound/soc/codecs/dw-hdmi-audio.h b/sound/soc/codecs/dw-hdmi-audio.h new file mode 100644 index 0000000..2a7b5b4 --- /dev/null +++ b/sound/soc/codecs/dw-hdmi-audio.h @@ -0,0 +1,17 @@ +/* + * dw-hdmi-audio.h -- DW HDMI ALSA SoC Audio driver + * + * Copyright 2011-2012 DesignerWare Products + * + * 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 _DW_HDMI_AUDIO_H +#define _DW_HDMI_AUDIO_H + +int dw_hdmi_jack_detect(struct snd_soc_codec *codec_dai, + struct snd_soc_jack *jack); + +#endif -- 2.1.2

2 1

[alsa-devel] [PATCH 1/5] sync asound.h with kernel version
by Pierre-Louis Bossart 31 Jan '15

31 Jan '15

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/asound.h | 37 ++++++++++++++++++++++++++++++++----- 1 file changed, 32 insertions(+), 5 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 1f23cd6..3d46e9a 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -140,7 +140,7 @@ struct snd_hwdep_dsp_image { * * *****************************************************************************/ -#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 12) +#define SNDRV_PCM_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 13) typedef unsigned long snd_pcm_uframes_t; typedef signed long snd_pcm_sframes_t; @@ -267,9 +267,17 @@ typedef int __bitwise snd_pcm_subformat_t; #define SNDRV_PCM_INFO_JOINT_DUPLEX 0x00200000 /* playback and capture stream are somewhat correlated */ #define SNDRV_PCM_INFO_SYNC_START 0x00400000 /* pcm support some kind of sync go */ #define SNDRV_PCM_INFO_NO_PERIOD_WAKEUP 0x00800000 /* period wakeup can be disabled */ -#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_WALL_CLOCK 0x01000000 /* (Deprecated)has audio wall clock for audio/system time sync */ +#define SNDRV_PCM_INFO_HAS_LINK_ATIME 0x01000000 /* report hardware link audio time, reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ABSOLUTE_ATIME 0x02000000 /* report absolute hardware link audio time, not reset on startup */ +#define SNDRV_PCM_INFO_HAS_LINK_ESTIMATED_ATIME 0x04000000 /* report estimated link audio time */ +#define SNDRV_PCM_INFO_HAS_LINK_SYNCHRONIZED_ATIME 0x08000000 /* report synchronized audio/system time */ + +#define SNDRV_PCM_INFO_DRAIN_TRIGGER 0x40000000 /* internal kernel flag - trigger in drain */ #define SNDRV_PCM_INFO_FIFO_IN_FRAMES 0x80000000 /* internal kernel flag - FIFO size is in frames */ + + typedef int __bitwise snd_pcm_state_t; #define SNDRV_PCM_STATE_OPEN ((__force snd_pcm_state_t) 0) /* stream is open */ #define SNDRV_PCM_STATE_SETUP ((__force snd_pcm_state_t) 1) /* stream has a setup */ @@ -407,6 +415,22 @@ struct snd_pcm_channel_info { unsigned int step; /* samples distance in bits */ }; +enum { + /* + * first definition for backwards compatibility only, + * maps to wallclock/link time for HDAudio playback and DEFAULT/DMA time for everything else + */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_COMPAT = 0, + + /* timestamp definitions */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 1, /* DMA time, reported as per hw_ptr */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK = 2, /* link time reported by sample or wallclock counter, reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /* link time reported by sample or wallclock counter, not reset on startup */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /* link time estimated indirectly */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /* link time synchronized with system time */ + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED +}; + struct snd_pcm_status { snd_pcm_state_t state; /* stream state */ struct timespec trigger_tstamp; /* time when stream was started/stopped/paused */ @@ -418,9 +442,11 @@ struct snd_pcm_status { snd_pcm_uframes_t avail_max; /* max frames available on hw since last status */ snd_pcm_uframes_t overrange; /* count of ADC (capture) overrange detections from last status */ snd_pcm_state_t suspended_state; /* suspended stream state */ - __u32 reserved_alignment; /* must be filled with zero */ - struct timespec audio_tstamp; /* from sample counter or wall clock */ - unsigned char reserved[56-sizeof(struct timespec)]; /* must be filled with zero */ + __u32 audio_tstamp_data; /* needed for 64-bit alignment, used for configs/report to/from userspace */ + struct timespec audio_tstamp; /* sample counter, wall clock, PHC or on-demand sync'ed */ + struct timespec driver_tstamp; /* useful in case reference system tstamp is reported with delay */ + __u32 audio_tstamp_accuracy; /* in ns units, only valid if indicated in audio_tstamp_data */ + unsigned char reserved[52-2*sizeof(struct timespec)]; /* must be filled with zero */ }; struct snd_pcm_mmap_status { @@ -533,6 +559,7 @@ enum { #define SNDRV_PCM_IOCTL_DELAY _IOR('A', 0x21, snd_pcm_sframes_t) #define SNDRV_PCM_IOCTL_HWSYNC _IO('A', 0x22) #define SNDRV_PCM_IOCTL_SYNC_PTR _IOWR('A', 0x23, struct snd_pcm_sync_ptr) +#define SNDRV_PCM_IOCTL_STATUS_EXT _IOWR('A', 0x24, struct snd_pcm_status) #define SNDRV_PCM_IOCTL_CHANNEL_INFO _IOR('A', 0x32, struct snd_pcm_channel_info) #define SNDRV_PCM_IOCTL_PREPARE _IO('A', 0x40) #define SNDRV_PCM_IOCTL_RESET _IO('A', 0x41) -- 1.9.1

1 4

[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix data delay configuration
by Peter Ujfalusi 30 Jan '15

30 Jan '15

Fix the issue introduced by: 368494093354 ASoC: tlv320aic3x: Add TDM support The CTRLC register were not receiving the correct delay configuration, which will corrupt DSP_A audio mode. Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, most of my boards are using DSP_B or I2S mode... This is why this is slipped. Peter sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index a78ad361f494..07603d142923 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1046,7 +1046,7 @@ static int aic3x_prepare(struct snd_pcm_substream *substream, delay += aic3x->tdm_delay; /* Configure data delay */ - snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, aic3x->tdm_delay); + snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } -- 2.2.2

4 3

[alsa-devel] [PATCH] ASoC: Add sysfs entries via static attribute groups
by Takashi Iwai 30 Jan '15

30 Jan '15

Instead of calling device_create_file() manually, assign the static attribute group entries at the device registration. This simplifies the code and avoids the possible races. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/soc-dapm.h | 2 +- sound/soc/soc-core.c | 40 ++++++++++++++++------------------------ sound/soc/soc-dapm.c | 16 +++++++--------- 3 files changed, 24 insertions(+), 34 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 89823cfe6f04..9fecd6752fcb 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -405,7 +405,7 @@ int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_update *update); /* dapm sys fs - used by the core */ -int snd_soc_dapm_sys_add(struct device *dev); +extern const struct attribute_group soc_dapm_dev_attr_group; void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2c62620abca6..386365f4a628 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -949,8 +949,6 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) /* unregister the rtd device */ if (rtd->dev_registered) { - device_remove_file(rtd->dev, &dev_attr_pmdown_time); - device_remove_file(rtd->dev, &dev_attr_codec_reg); device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -1103,6 +1101,21 @@ err_probe: return ret; } +static struct attribute *soc_dev_attrs[] = { + &dev_attr_codec_reg.attr, + &dev_attr_pmdown_time.attr, + NULL +}; + +static const struct attribute_group soc_dev_attr_group = { + .attrs = soc_dev_attrs, +}; + +static const struct attribute_group *soc_dev_attr_groups[] = { + &soc_dev_attr_group, + &soc_dapm_dev_attr_group, +}; + static void rtd_release(struct device *dev) { kfree(dev); @@ -1120,6 +1133,7 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, device_initialize(rtd->dev); rtd->dev->parent = rtd->card->dev; rtd->dev->release = rtd_release; + rtd->dev->groups = soc_dev_attr_groups; dev_set_name(rtd->dev, "%s", name); dev_set_drvdata(rtd->dev, rtd); mutex_init(&rtd->pcm_mutex); @@ -1136,23 +1150,6 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return ret; } rtd->dev_registered = 1; - - if (rtd->codec) { - /* add DAPM sysfs entries for this codec */ - ret = snd_soc_dapm_sys_add(rtd->dev); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec dapm sysfs entries: %d\n", - ret); - - /* add codec sysfs entries */ - ret = device_create_file(rtd->dev, &dev_attr_codec_reg); - if (ret < 0) - dev_err(rtd->dev, - "ASoC: failed to add codec sysfs files: %d\n", - ret); - } - return 0; } @@ -1308,11 +1305,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } #endif - ret = device_create_file(rtd->dev, &dev_attr_pmdown_time); - if (ret < 0) - dev_warn(rtd->dev, "ASoC: failed to add pmdown_time sysfs: %d\n", - ret); - if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ ret = soc_new_compress(rtd, num); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c5136bb1f982..a997bb8cae93 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2127,15 +2127,14 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -int snd_soc_dapm_sys_add(struct device *dev) -{ - return device_create_file(dev, &dev_attr_dapm_widget); -} +static struct attribute *soc_dapm_dev_attrs[] = { + &dev_attr_dapm_widget.attr, + NULL +}; -static void snd_soc_dapm_sys_remove(struct device *dev) -{ - device_remove_file(dev, &dev_attr_dapm_widget); -} +const struct attribute_group soc_dapm_dev_attr_group = { + .attrs = soc_dapm_dev_attrs, +}; static void dapm_free_path(struct snd_soc_dapm_path *path) { @@ -3903,7 +3902,6 @@ void snd_soc_dapm_auto_nc_pins(struct snd_soc_card *card) */ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_sys_remove(dapm->dev); dapm_debugfs_cleanup(dapm); dapm_free_widgets(dapm); list_del(&dapm->list); -- 2.2.2

2 2

[alsa-devel] [PATCH] ASoC: rt286: Add rt288 codec support
by Bard Liao 30 Jan '15

30 Jan '15

This patch adds support for rt288 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 81bdd276..2eafbcf 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -34,6 +34,7 @@ #include "rt286.h" #define RT286_VENDOR_ID 0x10ec0286 +#define RT288_VENDOR_ID 0x10ec0288 struct rt286_priv { struct regmap *regmap; @@ -1210,7 +1211,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_read(rt286->regmap, RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); - if (ret != RT286_VENDOR_ID) { + if (ret != RT286_VENDOR_ID && ret != RT288_VENDOR_ID) { dev_err(&i2c->dev, "Device with ID register %x is not rt286\n", ret); return -ENODEV; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt286: set idle_bias_off to false
by Bard Liao 30 Jan '15

30 Jan '15

We do the power down process at bias level STANDBY. And do nothing at bias level OFF. As a result, the register setting is the same between bias level STANDBY and OFF. So we don't need to set idle_bias_off true. Signed-off-by: Jie Yang <yang.jie(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 847cc4b..81bdd276 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1144,7 +1144,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt286 = { .suspend = rt286_suspend, .resume = rt286_resume, .set_bias_level = rt286_set_bias_level, - .idle_bias_off = true, + .idle_bias_off = false, .controls = rt286_snd_controls, .num_controls = ARRAY_SIZE(rt286_snd_controls), .dapm_widgets = rt286_dapm_widgets, -- 1.8.1.1.439.g50a6b54

3 7

[alsa-devel] [PATCH] ASoC: sgtl5000: Use shift mask when setting codec mode
by Filip Brozovic 30 Jan '15

30 Jan '15

Shift the I2S mode value by the necessary amount before writing the registers. This makes Right Justified and PCM mode work in addition to the default Left Justified mode. Signed-off-by: Filip Brozovic <fbrozovic(a)gmail.com> --- sound/soc/codecs/sgtl5000.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 29cf7ce..7665016 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -483,21 +483,21 @@ static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* setting i2s data format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_DSP_B: - i2sctl |= SGTL5000_I2S_MODE_PCM; + i2sctl |= SGTL5000_I2S_MODE_PCM << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; case SND_SOC_DAIFMT_I2S: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; break; case SND_SOC_DAIFMT_RIGHT_J: - i2sctl |= SGTL5000_I2S_MODE_RJ; + i2sctl |= SGTL5000_I2S_MODE_RJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRPOL; break; case SND_SOC_DAIFMT_LEFT_J: - i2sctl |= SGTL5000_I2S_MODE_I2S_LJ; + i2sctl |= SGTL5000_I2S_MODE_I2S_LJ << SGTL5000_I2S_MODE_SHIFT; i2sctl |= SGTL5000_I2S_LRALIGN; break; default: -- 2.1.4

1 0

[alsa-devel] [PATCH v2 11/12] ASoC: rockchip-hdmi-audio: add sound driver for hdmi audio
by Yakir Yang 30 Jan '15

30 Jan '15

Add a sound driver that combines rockchip-i2s cpu_dai and dw-hdmi-codec as codec_dai to provide hdmi audio output on rk3288 platforms. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Changes in v2: - give "codec-name" & "codec-dai-name" an const name sound/soc/rockchip/Kconfig | 9 ++ sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_hdmi_audio.c | 196 +++++++++++++++++++++++++++++++ 3 files changed, 207 insertions(+) create mode 100644 sound/soc/rockchip/rockchip_hdmi_audio.c diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index e181826..ed2b7f0 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -14,3 +14,12 @@ config SND_SOC_ROCKCHIP_I2S Say Y or M if you want to add support for I2S driver for Rockchip I2S device. The device supports upto maximum of 8 channels each for play and record. + +config SND_SOC_ROCKCHIP_HDMI_AUDIO + tristate "ASoC support for Rockchip HDMI audio" + depends on SND_SOC_ROCKCHIP + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_DW_HDMI_AUDIO + help + Say Y or M here if you want to add support for SoC audio on Rockchip + HDMI, such as rk3288 hdmi. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index b921909..b9185b3 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -1,4 +1,6 @@ # ROCKCHIP Platform Support snd-soc-i2s-objs := rockchip_i2s.o +snd-soc-rockchip-hdmi-audio-objs := rockchip_hdmi_audio.o obj-$(CONFIG_SND_SOC_ROCKCHIP_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SOC_ROCKCHIP_HDMI_AUDIO) += snd-soc-rockchip-hdmi-audio.o diff --git a/sound/soc/rockchip/rockchip_hdmi_audio.c b/sound/soc/rockchip/rockchip_hdmi_audio.c new file mode 100644 index 0000000..0c9f497 --- /dev/null +++ b/sound/soc/rockchip/rockchip_hdmi_audio.c @@ -0,0 +1,196 @@ +/* + * rockchip-hdmi-card.c + * + * ROCKCHIP ALSA SoC DAI driver for HDMI audio on rockchip processors. + * Copyright (c) 2014, ROCKCHIP CORPORATION. All rights reserved. + * Authors: Yakir Yang <ykk(a)rock-chips.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>.* + * + */ +#include <linux/module.h> +#include <linux/device.h> +#include <linux/platform_device.h> + +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/jack.h> +#include <sound/core.h> +#include <sound/pcm_params.h> + +#include "rockchip_i2s.h" +#include "../codecs/dw-hdmi-audio.h" + +#define DRV_NAME "rockchip-hdmi-audio" + +struct hdmi_audio_private { + struct snd_soc_jack hdmi_jack; +}; + +static int rockchip_hdmi_audio_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned int dai_fmt = rtd->dai_link->dai_fmt; + int mclk, ret; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, dai_fmt); + if (ret < 0) { + dev_err(cpu_dai->dev, "failed to set cpu_dai fmt.\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(cpu_dai->dev, "failed to set cpu_dai sysclk.\n"); + return ret; + } + + return 0; +} + +static int hdmi_audio_dai_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_card *card = runtime->card; + struct hdmi_audio_private *drv = snd_soc_card_get_drvdata(card); + + /* Enable headphone jack detection */ + snd_soc_jack_new(codec, "HDMI Jack", SND_JACK_LINEOUT, + &drv->hdmi_jack); + + return dw_hdmi_jack_detect(codec, &drv->hdmi_jack); +} + +static struct snd_soc_ops hdmi_audio_dai_ops = { + .hw_params = rockchip_hdmi_audio_hw_params, +}; + +static struct snd_soc_dai_link hdmi_audio_dai = { + .name = "RockchipHDMI", + .stream_name = "RockchipHDMI", + .codec_name = "dw-hdmi-audio", + .codec_dai_name = "dw-hdmi-hifi", + .init = hdmi_audio_dai_init, + .ops = &hdmi_audio_dai_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, +}; + +static struct snd_soc_card rockchip_hdmi_audio_card = { + .name = "RockchipHDMI", + .owner = THIS_MODULE, + .dai_link = &hdmi_audio_dai, + .num_links = 1, +}; + +static int rockchip_hdmi_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_hdmi_audio_card; + struct device_node *np = pdev->dev.of_node; + struct hdmi_audio_private *drv; + int ret; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (!drv) + return -ENOMEM; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, drv); + + hdmi_audio_dai.cpu_of_node = of_parse_phandle(np, "i2s-controller", 0); + if (!hdmi_audio_dai.cpu_of_node) { + dev_err(&pdev->dev, "Property 'i2s-controller' missing !\n"); + goto free_priv_data; + } + + hdmi_audio_dai.platform_of_node = hdmi_audio_dai.cpu_of_node; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "register card failed (%d)\n", ret); + card->dev = NULL; + goto free_cpu_of_node; + } + + dev_info(&pdev->dev, "hdmi audio init success.\n"); + + return 0; + +free_cpu_of_node: + hdmi_audio_dai.cpu_of_node = NULL; + hdmi_audio_dai.platform_of_node = NULL; +free_priv_data: + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + devm_kfree(&pdev->dev, drv); + card->dev = NULL; + + return ret; +} + +static int rockchip_hdmi_audio_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + snd_soc_card_set_drvdata(card, NULL); + platform_set_drvdata(pdev, NULL); + card->dev = NULL; + + return 0; +} + +static const struct of_device_id rockchip_hdmi_audio_of_match[] = { + { .compatible = "rockchip,rk3288-hdmi-audio", }, + {}, +}; + +static struct platform_driver rockchip_hdmi_audio_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .of_match_table = rockchip_hdmi_audio_of_match, + }, + .probe = rockchip_hdmi_audio_probe, + .remove = rockchip_hdmi_audio_remove, +}; +module_platform_driver(rockchip_hdmi_audio_driver); + +MODULE_AUTHOR("Yakir Yang <ykk(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip HDMI Audio ASoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_hdmi_audio_of_match); -- 2.1.2

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Re: [alsa-devel] [PATCH 0/2 v5] dmaengine: rcar-audmapp: independent from SH_DMAE_BASE
by Kuninori Morimoto 30 Jan '15

30 Jan '15

Hi Laurent, Vinod, and Arnd # I added Linux ALSA ML and Mark > > >> This time I added SoC/platform side setting patches too (= 3) - 6)). > > >> SoC/platform side setting needs many entries for this rcar-audmapp, > > >> because it has many combinations. > > >> But, I believe this is very normal DMAEngine style, not special. > > > > > > Vinod commented on 22/12/2014 (Message-ID: > > > <20141222151447.GL16827(a)intel.com>) > > > > > > "I think this makes sense. Going thru the driver, it was clear that we > > > were not really gaining anything for using dmaengine API here. So agree > > > that lets use dmanegine for 1st dmac thru dmaengine library and then > > > configure this in your sound driver.." > > > > > > My understanding is that a solution specific to the sound driver was > > > preferred, instead of a generic DMA engine driver. Have I missed something > > > ? > > > > Grr... my understanding was that > > "1st DMAC will use dmaengine library (= sound framework has sound-dma-xxx > > functions) 2nd DMAC will use normal dmaengine API" > > > > But, I need to fixup sound driver ? > > - 1st DMAC = normal DMAEngine API > > - 2nd DMAC = part of sound driver > > > > Sorry, for discuss it again here, but I want flexible switching > > for 1st/2nd DMAC (because of 1st/2nd DMAC path combination). > > So, using same DMAEngine interface from sound driver is easy to > > implement/understanding. > > - 1st DMAC = normal DMAEngine API > > - 2nd DMAC = normal DMAEngine API > > The first DMA engine (the one handling transfer from/to memory) is a general- > purpose DMA engine. It should be handled by a driver that implement the DMA > engine API, no doubt about that. > > The second "DMA engine" is dedicated to the sound IP cores and is far from > being general-purpose, given that it only supports peripheral-to-peripheral > transfers, without even interrupts to report transfer completion. I'm not even > sure we can call it a DMA engine as there's no Direct Memory Access involved > here :-) The hardware looks more like a crossbar switch with programmable > routing of audio channels. That's why Vinod and I were wondering whether it > really makes sense to implement it using the DMA engine API, given the > resulting complexity of the DT bindings (the sound DT nodes look a bit scary), > or if it could be simpler to implement it as part of the Renesas sound > drivers. If you are caring about naming (= DMA), it is "Audio *DMAC* peri peri". I wonder dma_transfer_direction has DMA_DEV_TO_DEV (this driver is not using it though...) it is for peripheral-to-peripheral ? And API point of view, 2nd DMAC doesn't need new DMAEngine API. From DRY (= Don't Repeat Yourself) point of view, I don't want to re-create "similar but different" implementation for naming issue. From DT bindings complexity point of view, which is complex ? DMAC driver side ? DT node side ? Indeed sound driver needs many node, but is is regular arrangement, not complex, and, it needs many node for 1st DMAC too. I don't understand why 1st is OK, 2nd is not OK ? From DMAC driver side complexity point of view, 1st DMAC has same complexity (= it accepts many node from many drivers) ? If I need to move 2nd DMAC from DMAEngine to sound driver side, please explain it to Mark Brown (= ALSA SoC maintainer) Best regards --- Kuninori Morimoto Best regards --- Kuninori Morimoto

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[alsa-devel] [RFC PATCH] ASoC: atmel_ssc_dai: Allow more rates
by Peter Rosin 29 Jan '15

29 Jan '15

From: Peter Rosin <peda(a)axentia.se> When the SSC acts as BCK master, use a ratnum rule to limit the rate instead of only doing the standard rates. When the SSC acts as BCK slave, allow any BCK frequency up to within 500ppm of the SSC master clock, possibly divided by 2, 3 or 6. Put a cap at 384kHz. Who's /ever/ going to need more than that? Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 98 +++++++++++++++++++++++++++++++++++++-- sound/soc/atmel/atmel_ssc_dai.h | 1 + 2 files changed, 95 insertions(+), 4 deletions(-) This patch depends on the patch I sent earlier that enables the _CBS_CFS mode on the atmel SSC dai (1). I'm unsure about the rate rules for reception on the TK pin when outputting LRCLK and the rate rules for transmitting on the RK pin when inputting LRCLK. The datasheet for "my" chip (SAMA5D31) doesn't say anything about what rates are allowed when using the RK pin for transmitting or the TK pin for receiving. Another question I have is if these rate rules are the same across all chips supported by this driver, or if there is need for some amount of special casing? And what special casing if that is the case? The reason I'm digging around in this area is that I need support for inputting a BCK of 16MHz on the TK pin during playback, i.e. with the codec mastering BCK. And the only way I can get that is to use _CBM_CFS. My SSC master clock is 66MHz, which means that the max supported rate for _CBM_CFM is 11MHz, i.e. not enough. This patch unlocks my use case, but I would like to correctly stop rates that the SSC does not support in other use cases as well. I simply do not have the hardware to fully test this myself. An lastly, is it reasonable to deduct 500ppm to allow for some clock skew? FYI, the 500ppm was pulled out of my hat... Cheers, Peter (1) ASoC: atmel_ssc_dai: Support SND_SOC_DAIFMT_CBM_CFS on I2S https://lkml.org/lkml/2015/1/29/373 diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f55f3aab8bdd..11eab65c86c8 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -187,6 +187,76 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) return IRQ_HANDLED; } +static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct atmel_ssc_info *ssc_p = rule->private; + struct ssc_device *ssc = ssc_p->ssc; + struct snd_interval *i = hw_param_interval(params, rule->var); + struct snd_interval t; + struct snd_ratnum r = { + .den_min = 1, + .den_max = 4095, + .den_step = 1, + }; + unsigned int num = 0, den = 0; + int frame_size; + int mck_div = 2; + int ret; + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) + return frame_size; + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFS: + if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) + && ssc->clk_from_rk_pin) + /* Receiver Frame Synchro is output on RK pin. + * What rules are there when outputting on the TK pin? + */ + mck_div = 3; + break; + + case SND_SOC_DAIFMT_CBM_CFM: + if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) + && !ssc->clk_from_rk_pin) + /* Transmit Frame Synchro is input on TK pin. + * What rules are there when inputting from the RK pin? + */ + mck_div = 6; + break; + } + + switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + r.num = ssc_p->mck_rate / mck_div / frame_size; + + ret = snd_interval_ratnum(i, 1, &r, &num, &den); + if (ret >= 0 && den && rule->var == SNDRV_PCM_HW_PARAM_RATE) { + params->rate_num = num; + params->rate_den = den; + } + break; + + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBM_CFM: + t.min = 8000; + t.max = ssc_p->mck_rate / mck_div / frame_size; + /* Take away 500ppm, just to be on the safe side. */ + t.max -= t.max / 2000; + t.openmin = t.openmax = 0; + t.integer = 0; + ret = snd_interval_refine(i, &t); + break; + + default: + ret = -EINVAL; + break; + } + + return ret; +} /*-------------------------------------------------------------------------*\ * DAI functions @@ -200,10 +270,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; + int ret; pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", ssc_readl(ssc_p->ssc->regs, SR)); + if (!ssc_p->initialized) + clk_enable(ssc_p->ssc->clk); + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk) * 2; + if (!ssc_p->initialized) + clk_disable(ssc_p->ssc->clk); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dir = 0; dir_mask = SSC_DIR_MASK_PLAYBACK; @@ -212,6 +289,17 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, dir_mask = SSC_DIR_MASK_CAPTURE; } + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + atmel_ssc_hw_rule_rate, + ssc_p, + SNDRV_PCM_HW_PARAM_FRAME_BITS, + SNDRV_PCM_HW_PARAM_CHANNELS, -1); + if (ret < 0) { + dev_err(dai->dev, "Failed to specify rate rule: %d\n", ret); + return ret; + } + dma_params = &ssc_dma_params[dai->id][dir]; dma_params->ssc = ssc_p->ssc; dma_params->substream = substream; @@ -787,8 +875,6 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) # define atmel_ssc_resume NULL #endif /* CONFIG_PM */ -#define ATMEL_SSC_RATES (SNDRV_PCM_RATE_8000_96000) - #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -808,12 +894,16 @@ static struct snd_soc_dai_driver atmel_ssc_dai = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .capture = { .channels_min = 1, .channels_max = 2, - .rates = ATMEL_SSC_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = ATMEL_SSC_FORMATS,}, .ops = &atmel_ssc_dai_ops, }; diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index b1f08d511495..80b153857a88 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -115,6 +115,7 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + unsigned long mck_rate; }; int atmel_ssc_set_audio(int ssc_id); -- 1.7.10.4

1 0

[alsa-devel] simple-audio-card vs. platform question
by Ricard Wanderlof 29 Jan '15

29 Jan '15

I'm trying to set up a simple device tree framework so I can do some development on an I2S driver. I'm trying to use simple-audio-card, but I keep running into the problem that once soc_bind_dai_link() gets called, dai_link->platform is set to NULL (because it is not initialized by simple-card.c when there is a DT node), while dai_link->platform_of_node has been set to dai_link->cpu_of_node in asoc_simple_card_parse_of(). This results in the following code in soc-core:soc_bind_dai_link() to set platform to NULL, /* if there's no platform we match on the empty platform */ platform_name = dai_link->platform_name; if (!platform_name && !dai_link->platform_of_node) platform_name = "snd-soc-dummy"; /* find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { if (dai_link->platform_of_node) { if (platform->dev->of_node != dai_link->platform_of_node) continue; } else { if (strcmp(platform->component.name, platform_name)) continue; } rtd->platform = platform; } if (!rtd->platform) { dev_err(card->dev, "ASoC: platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } with -EPROBE_DEFER as a result. I can't figure out either how to set up a platform (=DMA) driver in this environment. Perhaps the case of having a CPU DAI, DMA driver and codec DAI is not supported by simple-audio-card? On other hand, the way it looks there is no way the resulting setup can get through soc_bind_dai_link() unless there is a platform and associated driver. This indicates to me that I'm doing something seriously wrong in my setup, but having looked at various DT configurations in arch/arm/boot/dts in this case, as well as the documentation in Documentation/devicetree/bindings/sound/simple-card.txt I can't figure how to set it up any other way. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH] ASoC: pxa: make TTC DKB tristate
by Arnd Bergmann 29 Jan '15

29 Jan '15

In a rare combination of Kconfig settings, the 88pm860x-codec module may be selected as a loadable module, while it's also being used by the ttb-dkb code that is built-in, resulting in a link error: sound/built-in.o: In function `ttc_pm860x_init': :(.text+0x3e888): undefined reference to `pm860x_hs_jack_detect' :(.text+0x3e898): undefined reference to `pm860x_mic_jack_detect' Changing ttb-tkb to a tristate option tells Kconfig that 88pm86x actually needs to be built-in if ttc-dkb is also built-in. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 2434b6d61675..39cea80846c3 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -140,7 +140,7 @@ config SND_PXA910_SOC Marvell PXA910 reference platform. config SND_SOC_TTC_DKB - bool "SoC Audio support for TTC DKB" + tristate "SoC Audio support for TTC DKB" depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y select PXA_SSP select SND_PXA_SOC_SSP

1 0

[alsa-devel] [PATCH] ASoC: Samsung: add missing I2C/SPI dependencies
by Arnd Bergmann 29 Jan '15

29 Jan '15

The SND_SOC_ARNDALE_RT5631_ALC5631 selects the rt5631 codec that requires I2C to work, so we get a build error if I2C is disabled: codecs/rt5631.c:1737:1: warning: data definition has no type or storage class 31_i2c_driver); codecs/rt5631.c:1737:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int] codecs/rt5631.c:1737:1: warning: parameter names (without types) in function declaration codecs/rt5631.c:1726:26: warning: 'rt5631_i2c_driver' defined but not used [-Wunused-variable] The same thing can happen on odroid-x2, littlemill, bells and speyside, so this patch adds all the dependencies. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index e817a2f43ea8..dd781d9a1c66 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -185,7 +185,7 @@ config SND_SOC_SMDK_WM8994_PCM config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM8996 select SND_SOC_WM9081 @@ -200,7 +200,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && MFD_ARIZONA && I2C && SPI_MASTER select SND_SAMSUNG_I2S select SND_SOC_WM5102 select SND_SOC_WM5110 @@ -217,7 +217,7 @@ config SND_SOC_LOWLAND config SND_SOC_LITTLEMILL tristate "Audio support for Wolfson Littlemill" - depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 && I2C select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 @@ -234,7 +234,7 @@ config SND_SOC_SNOW config SND_SOC_ODROIDX2 tristate "Audio support for Odroid-X2 and Odroid-U3" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SOC_MAX98090 select SND_SAMSUNG_I2S help @@ -242,6 +242,6 @@ config SND_SOC_ODROIDX2 config SND_SOC_ARNDALE_RT5631_ALC5631 tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on SND_SOC_SAMSUNG + depends on SND_SOC_SAMSUNG && I2C select SND_SAMSUNG_I2S select SND_SOC_RT5631

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[alsa-devel] [PATCH 0/2] ASoC: pcm512x: Fixups for the Clock master modes series
by Peter Rosin 29 Jan '15

29 Jan '15

From: Peter Rosin <peda(a)axentia.se> These fixups can either be squashed in with 5/7 from the Clock master mode series, or they can be added on top of the current topic/pcm512x. Sorry for the trouble. I can squash them, and resend the Clock master modes series if that helps. Let me know how you'd like it. Cheers, Peter Peter Rosin (2): ASoC: pcm512x: Fixup warning splat ASoC: pcm512x: Use the correct range constraints for S24_LE sound/soc/codecs/pcm512x.c | 41 +++++++++++++++++++++-------------------- 1 file changed, 21 insertions(+), 20 deletions(-) -- 1.7.10.4

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[alsa-devel] [RFC] ASoC: rt5677: fix SPI dependency
by Arnd Bergmann 29 Jan '15

29 Jan '15

The rt5677 codec has gained code that requires SPI to work correctly, but there is no provision in Kconfig to prevent the driver from being used when SPI is disabled or a loadable module, resulting in this build error: sound/built-in.o: In function `rt5677_spi_write': :(.text+0xa7ba0): undefined reference to `spi_sync' sound/built-in.o: In function `rt5677_spi_driver_init': :(.init.text+0x253c): undefined reference to `spi_register_driver' ERROR: "spi_sync" [sound/soc/codecs/snd-soc-rt5677-spi.ko] undefined! ERROR: "spi_register_driver" [sound/soc/codecs/snd-soc-rt5677-spi.ko] undefined! This makes the SPI portion of the driver depend on the SPI subsystem, and disables the function that uses SPI for firmware download if SPI is disabled. The latter may not be the correct solution, but I could not come up with a better one. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: af48f1d08a54741 ("ASoC: rt5677: Support DSP function for VAD application") diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8349f982a586..ef2c70e77d91 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -525,7 +525,7 @@ config SND_SOC_RT5677 config SND_SOC_RT5677_SPI tristate - default SND_SOC_RT5677 + default SND_SOC_RT5677 && SPI #Freescale sgtl5000 codec config SND_SOC_SGTL5000 diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 26fc538f03b1..5d0bb8748dd1 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -702,6 +702,9 @@ static int rt5677_set_dsp_vad(struct snd_soc_codec *codec, bool on) static bool activity; int ret; + if (!IS_ENABLED(CONFIG_SND_SOC_RT5677_SPI)) + return -ENXIO; + if (on && !activity) { activity = true;

2 1

[alsa-devel] [PATCH] ASoC: davinci: fix DM365_EVM codec selection
by Arnd Bergmann 29 Jan '15

29 Jan '15

An earlier bug fix of mine made the SND_DM365_VOICE_CODEC symbol tristate to avoid creating an undefined reference from the davinci-vcif.c driver to the davinci_soc_platform_register function that may be in a module. However, this may now lead to a different error on randconfig kernels: "warning: SND_DM365_VOICE_CODEC creates inconsistent choice state" This happens because we now have a choice statement with one bool and one tristate option, and the latter might not support being set to 'y' because of dependencies. This new change turns the other option into 'tristate' as well, which avoids the problem. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 19926c6de0c3 ("ASoC: davinci: vcif must be a module if SND_DAVINCI_SOC is") diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 8e948c63f3d9..2b81ca418d2a 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -58,13 +58,12 @@ choice depends on MACH_DAVINCI_DM365_EVM config SND_DM365_AIC3X_CODEC - bool "Audio Codec - AIC3101" + tristate "Audio Codec - AIC3101" help Say Y if you want to add support for AIC3101 audio codec config SND_DM365_VOICE_CODEC tristate "Voice Codec - CQ93VC" - depends on SND_DAVINCI_SOC select MFD_DAVINCI_VOICECODEC select SND_DAVINCI_SOC_VCIF select SND_SOC_CQ0093VC

2 1

[alsa-devel] ESI Juli@ crash with external clock switch - patch
by Pavel Hofman 28 Jan '15

28 Jan '15

Hi, I have investigated a crash/kernel thread lockup when Juli@ is switched to external SPDIF clock and the incoming SPDIF stream changes samplerate. The problem appears to occur in the timed thread in charge of reading incoming samplerate and acting upon its change. The problem disappears with the following patch: diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index c7f5633..68bb326 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -154,7 +154,7 @@ void snd_ak4114_reinit(struct ak4114 *chip) { chip->init = 1; mb(); - flush_delayed_work(&chip->work); + //flush_delayed_work(&chip->work); ak4114_init_regs(chip); /* bring up statistics / event queing */ chip->init = 0; I am afraid I do not know enough about kernel workqueues to determine whether this "fix" is OK. Interestingly, the almost identical driver ak4113.c for a very similar card Infrasonic Quartet (ice1724/quartet.c) does not suffer from this problem (tested now). Thanks a lot for your opinion. Best regards, Pavel Hofman.

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[alsa-devel] [PATCH v2 0/7] ASoC: pcm512x: Clock master modes
by Peter Rosin 28 Jan '15

28 Jan '15

From: Peter Rosin <peda(a)axentia.se> Hi! [ Note that the dt change is in patch 5/7, for those only interested in that particular bit. ] This series implements BCLK master modes for the pcm512x driver. It has only been tested with the pcm5142 chip, but they are from the same family and should be compatible. I have mainly used the spec for the newer pcm5242 chip (also from the same family) as it fills in a lot of blanks in the pcm512x/pcm514x specs. The code has also seen most of its testing in a 3.10 environment, so there might be some forward-porting warts. But it is able to play sound in 3.18 as well, and most of the changes have little to do with anything but the clocking in the chip itself. Changes since v1: - Use snd_soc_params_to_frame_size and snd_soc_params_to_bclk instead of implementing home-grown versions based on the physical width. The I2S-communication will no longer have padding bits for S24_LE. Cheers, Peter Peter Rosin (7): ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Change register default to match actual content after reset ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS .../devicetree/bindings/sound/pcm512x.txt | 25 +- include/sound/pcm.h | 12 + sound/core/pcm_lib.c | 85 ++ sound/soc/codecs/pcm512x.c | 933 +++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- 5 files changed, 1139 insertions(+), 25 deletions(-) -- 1.7.10.4

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[alsa-devel] [PATCH v3 1/2] ASoC: codecs: wm8904: add dt ids table
by Bo Shen 28 Jan '15

28 Jan '15

From: Alexander Morozov <linux(a)meltdown.ru> The WM8904 and WM8918 has the same data type, while the WM8912 has different data type. So, use the data in dt ids table to distinguish them. Signed-off-by: Alexander Morozov <linux(a)meltdown.ru> [voice.shen(a)atmel.com: add code to distinguish device type] Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v3: - Get the device type from match data in probe function. Changes in v2: - Add driver data for distinguish the device capability. sound/soc/codecs/wm8904.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..a80bc52 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2098,6 +2098,24 @@ static const struct regmap_config wm8904_regmap = { .num_reg_defaults = ARRAY_SIZE(wm8904_reg_defaults), }; +#ifdef CONFIG_OF +static enum wm8904_type wm8904_data = WM8904; +static enum wm8904_type wm8912_data = WM8912; + +static const struct of_device_id wm8904_of_match[] = { + { + .compatible = "wlf,wm8904", + .data = &wm8904_data, + }, { + .compatible = "wlf,wm8912", + .data = &wm8912_data, + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, wm8904_of_match); +#endif + static int wm8904_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -2125,7 +2143,17 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, return ret; } - wm8904->devtype = id->driver_data; + if (i2c->dev.of_node) { + const struct of_device_id *match; + + match = of_match_node(wm8904_of_match, i2c->dev.of_node); + if (match == NULL) + return -EINVAL; + wm8904->devtype = *((enum wm8904_type *)match->data); + } else { + wm8904->devtype = id->driver_data; + } + i2c_set_clientdata(i2c, wm8904); wm8904->pdata = i2c->dev.platform_data; @@ -2259,6 +2287,7 @@ static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(wm8904_of_match), }, .probe = wm8904_i2c_probe, .remove = wm8904_i2c_remove, -- 2.3.0.rc0

2 2

[alsa-devel] [PATCH v3 00/10] Improvements to Tegra-based Chromebook support
by Tomeu Vizoso 28 Jan '15

28 Jan '15

v3: * Added bindings for the LTN140AT29 panel * Removed the delay in pwrseq, as what was actually needed was to add a dependency on the power supplies of the host * Uses the pinmux for the Blaze as generated by tegra-pinmux-scripts * Uses the pinmux for the Big as in the last patch from Simon Glass Hello, this series adds support for the Tegra-based HP Chromebook 14 (aka nyan blaze), which is very similar to the Acer Chromebook 13 (aka nyan big). Because they both include tegra124-nyan.dtsi, some improvements to Blaze support have also benefitted the Big. I have tested that USB2, the panels, HDMI, the trackpad, Wifi and sound work on both. The DT for the Big includes the pinmux configuration as generated by tegra-pinmux-scripts with Simon's patch at: https://patchwork.ozlabs.org/patch/417779/ These patches are based on top of linux-next 20150128. http://cgit.collabora.com/git/user/tomeu/linux.git/log/?h=nyan-v3 Regards, Tomeu Stéphane Marchesin (1): drm/panel: add support for Samsung LTN140AT29 panel Tomeu Vizoso (9): ARM: tegra: Set the sound card model that alsaucm expects ARM: tegra: Move out nyan-generic parts out from the nyan-big DT ARM: tegra: Add DTS for the nyan-blaze board ARM: tegra: Add node for trackpad in Nyan boards ASoC: tegra: Add a control for the headphone switch ASoC: tegra: add sink for the internal mic to tegra_max98090 ARM: tegra: Use pwrseq-simple for the wifi in Nyan ARM: tegra: Use the generated pinmux data ARM: tegra: Set spi-max-frequency property to flash node .../bindings/panel/samsung,ltn140at29-301.txt | 7 + .../bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + arch/arm/boot/dts/Makefile | 1 + arch/arm/boot/dts/tegra124-nyan-big.dts | 2112 +++++++++++--------- arch/arm/boot/dts/tegra124-nyan-blaze.dts | 1325 ++++++++++++ arch/arm/boot/dts/tegra124-nyan.dtsi | 692 +++++++ drivers/gpu/drm/panel/panel-simple.c | 26 + sound/soc/tegra/tegra_max98090.c | 3 + 8 files changed, 3206 insertions(+), 961 deletions(-) create mode 100644 Documentation/devicetree/bindings/panel/samsung,ltn140at29-301.txt create mode 100644 arch/arm/boot/dts/tegra124-nyan-blaze.dts create mode 100644 arch/arm/boot/dts/tegra124-nyan.dtsi -- 1.9.3

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[alsa-devel] [PATCH PING] ASoC: dwc: Remove unnecessary conditional compilation
by Andrew Jackson 28 Jan '15

28 Jan '15

From: Andrew Jackson <Andrew.Jackson(a)arm.com> of_match_ptr is already conditionally compiled based on CONFIG_OF so further conditional compilation is not required. Remove conditional compilation surrounding of_match_ptr. Signed-off-by: Andrew Jackson <Andrew.Jackson(a)arm.com> --- sound/soc/dwc/designware_i2s.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 99cf64b..a3e97b4 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -624,9 +624,7 @@ static struct platform_driver dw_i2s_driver = { .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", -#ifdef CONFIG_OF .of_match_table = of_match_ptr(dw_i2s_of_match), -#endif }, }; -- 1.7.1

2 1

[alsa-devel] [RFC] XEN support in aserver
by Vitaly V. Ch 28 Jan '15

28 Jan '15

Hi all! What do you think about introducing XEN support into aserver? It will allow to share sound between domains with few little changes. \\wbr Vitaly Ch

3 3

[alsa-devel] [PATCH 0/7] Yet another line6 cleanups
by Takashi Iwai 28 Jan '15

28 Jan '15

OK, I did enough for this month. The only real fix is the first patch for big-endianess. The rest are rather cleanups and beautifying the code. As usual, the latest code is found in test/line6 branch. thanks, Takashi === Takashi Iwai (7): ALSA: line6: Fix volume calculation for big-endian ALSA: line6: Tidy up and typo fixes in comments ALSA: line6: Remove revision.h ALSA: line6: Move the contents of usbdefs.h into driver.h ALSA: line6: Remove struct usb_line6_podhd ALSA: line6: Remove invalid capability bits for PODxt Live Variax ALSA: line6: Remove snd_line6_ prefix of pcm property fields sound/usb/line6/capture.c | 5 ++- sound/usb/line6/driver.c | 4 +-- sound/usb/line6/driver.h | 89 ++++++++++++++++++++-------------------------- sound/usb/line6/midi.c | 1 - sound/usb/line6/midi.h | 32 +++++------------ sound/usb/line6/pcm.h | 57 +++++++++-------------------- sound/usb/line6/playback.c | 24 +++++++------ sound/usb/line6/pod.c | 39 ++++++-------------- sound/usb/line6/podhd.c | 16 +++------ sound/usb/line6/revision.h | 4 --- sound/usb/line6/toneport.c | 31 +++++----------- sound/usb/line6/usbdefs.h | 27 -------------- sound/usb/line6/variax.c | 25 ++++--------- 13 files changed, 109 insertions(+), 245 deletions(-) delete mode 100644 sound/usb/line6/revision.h delete mode 100644 sound/usb/line6/usbdefs.h -- 2.2.2

1 8

[alsa-devel] [PATCH 0/4] Misc cleanup for 3.20 kernel
by Takashi Iwai 28 Jan '15

28 Jan '15

Hi, here is a few trivial cleanup patches, targeted for 3.20. All these shouldn't result in any functional change. Takashi === Takashi Iwai (4): ALSA: Include linux/io.h instead of asm/io.h ALSA: Include linux/uaccess.h and linux/bitopts.h instead of asm/* ALSA: au88x0: Remove superfluous ifdef __KERNEL__ ALSA: usx2y: Move UAPI definition into include/uapi/sound/usb_stream.h include/sound/emu10k1.h | 2 +- include/sound/gus.h | 2 +- include/sound/sb.h | 2 +- include/uapi/sound/usb_stream.h | 76 ++++++++++++++++++++++++++++++++++ sound/aoa/soundbus/i2sbus/control.c | 2 +- sound/aoa/soundbus/i2sbus/pcm.c | 2 +- sound/core/memory.c | 4 +- sound/core/pcm_memory.c | 2 +- sound/core/pcm_native.c | 2 +- sound/drivers/mpu401/mpu401_uart.c | 2 +- sound/drivers/opl3/opl3_lib.c | 2 +- sound/drivers/opl4/opl4_lib.c | 2 +- sound/drivers/opl4/opl4_synth.c | 2 +- sound/drivers/pcsp/pcsp.c | 2 +- sound/drivers/pcsp/pcsp_input.c | 2 +- sound/drivers/pcsp/pcsp_lib.c | 2 +- sound/drivers/serial-u16550.c | 3 +- sound/drivers/vx/vx_core.c | 2 +- sound/i2c/other/ak4xxx-adda.c | 2 +- sound/isa/ad1816a/ad1816a_lib.c | 2 +- sound/isa/azt2320.c | 2 +- sound/isa/cs423x/cs4236_lib.c | 2 +- sound/isa/es1688/es1688_lib.c | 2 +- sound/isa/es18xx.c | 2 +- sound/isa/opl3sa2.c | 3 +- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- sound/isa/sb/emu8000.c | 4 +- sound/isa/sb/emu8000_patch.c | 2 +- sound/isa/sb/sb16_main.c | 2 +- sound/isa/sb/sb8_main.c | 2 +- sound/isa/sb/sb8_midi.c | 2 +- sound/isa/sb/sb_common.c | 2 +- sound/isa/sb/sb_mixer.c | 2 +- sound/isa/wavefront/wavefront_fx.c | 2 +- sound/isa/wavefront/wavefront_midi.c | 2 +- sound/isa/wavefront/wavefront_synth.c | 2 +- sound/isa/wss/wss_lib.c | 2 +- sound/parisc/harmony.c | 2 +- sound/pci/ad1889.c | 3 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/als300.c | 3 +- sound/pci/als4000.c | 2 +- sound/pci/asihpi/hpioctl.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au88x0.h | 4 +- sound/pci/aw2/aw2-saa7146.c | 2 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 2 +- sound/pci/ca0106/ca0106_mixer.c | 2 +- sound/pci/ca0106/ca0106_proc.c | 2 +- sound/pci/cmipci.c | 2 +- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx_lib.c | 3 +- sound/pci/cs46xx/dsp_spos.c | 2 +- sound/pci/cs46xx/dsp_spos_scb_lib.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/es1938.c | 3 +- sound/pci/es1968.c | 2 +- sound/pci/ice1712/ak4xxx.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 3 +- sound/pci/maestro3.c | 2 +- sound/pci/mixart/mixart_core.c | 2 +- sound/pci/mixart/mixart_hwdep.c | 2 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/oxygen_io.c | 2 +- sound/pci/pcxhr/pcxhr_core.c | 2 +- sound/pci/pcxhr/pcxhr_hwdep.c | 2 +- sound/pci/riptide/riptide.c | 2 +- sound/pci/rme32.c | 3 +- sound/pci/rme96.c | 3 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 2 +- sound/pci/sonicvibes.c | 3 +- sound/pci/trident/trident_main.c | 3 +- sound/pci/trident/trident_memory.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/vx222/vx222_ops.c | 2 +- sound/pci/ymfpci/ymfpci_main.c | 2 +- sound/ppc/awacs.c | 2 +- sound/ppc/beep.c | 2 +- sound/ppc/burgundy.c | 2 +- sound/ppc/pmac.c | 2 +- sound/ppc/tumbler.c | 2 +- sound/sh/aica.c | 2 +- sound/sparc/amd7930.c | 2 +- sound/synth/emux/emux_hwdep.c | 2 +- sound/synth/emux/emux_oss.c | 2 +- sound/synth/emux/soundfont.c | 2 +- sound/usb/usx2y/usb_stream.h | 78 ++--------------------------------- 108 files changed, 188 insertions(+), 195 deletions(-) create mode 100644 include/uapi/sound/usb_stream.h -- 2.2.2

1 4

[alsa-devel] sigmadsp: adau1701: Version mismatch while loading firmware in 3.18.3
by Pascal Huerst 28 Jan '15

28 Jan '15

Hello Lars-Peter, i followed the instructions on: http://wiki.analog.com/resources/tools-software/linux-software/sigmadsp_gen… and built the firmware from my xml-file. But when the driver loads, I always get: adau1701 2-0034: Failed to load firmware: Invalid version 2. Supported firmware versions: 1 adau1701 2-0034: Failed to load firmware adau1701 2-0034: ASoC: can't set adau1701 hw params: -22 I have been using that .xml/.bin file in 3.14.13 for some time, but now in 3.18.3 I always get the version mismatch from above. What am I doing wrong? regards pascal

2 2

[alsa-devel] [PATCH] sound: sscape: add missing include of asm/io.h
by Arnd Bergmann 28 Jan '15

28 Jan '15

The soundscape driver uses the ISA inb/outb functions declared in asm/io.h, so it needs to include this header to avoid a build error: sscape.c: In function 'sscape_write_unsafe': sscape.c:203:2: error: implicit declaration of function 'outb' [-Werror=implicit-function-declaration] Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 018ab140c2be..7b248cdf06e2 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -23,6 +23,7 @@ #include <linux/init.h> #include <linux/err.h> +#include <linux/io.h> #include <linux/isa.h> #include <linux/delay.h> #include <linux/firmware.h>

2 3

[alsa-devel] [PATCH] ASoC: rx51: do not fail if could not get jack detection gpio
by Pali Rohár 28 Jan '15

28 Jan '15

qemu does not emulate it Signed-off-by: Pali Rohár <pali.rohar(a)gmail.com> --- sound/soc/omap/rx51.c | 45 +++++++++++++++++++++++++-------------------- 1 file changed, 25 insertions(+), 20 deletions(-) diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 04896d6..a689021 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -310,25 +310,27 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) return err; } - /* AV jack detection */ - err = snd_soc_jack_new(codec, "AV Jack", - SND_JACK_HEADSET | SND_JACK_VIDEOOUT, - &rx51_av_jack); - if (err) { - dev_err(card->dev, "Failed to add AV Jack\n"); - return err; - } + if (!IS_ERR(pdata->jack_detection_gpio)) { + /* AV jack detection */ + err = snd_soc_jack_new(codec, "AV Jack", + SND_JACK_HEADSET | SND_JACK_VIDEOOUT, + &rx51_av_jack); + if (err) { + dev_err(card->dev, "Failed to add AV Jack\n"); + return err; + } - /* prepare gpio for snd_soc_jack_add_gpios */ - rx51_av_jack_gpios[0].gpio = desc_to_gpio(pdata->jack_detection_gpio); - devm_gpiod_put(card->dev, pdata->jack_detection_gpio); + /* prepare gpio for snd_soc_jack_add_gpios */ + rx51_av_jack_gpios[0].gpio = desc_to_gpio(pdata->jack_detection_gpio); + devm_gpiod_put(card->dev, pdata->jack_detection_gpio); - err = snd_soc_jack_add_gpios(&rx51_av_jack, - ARRAY_SIZE(rx51_av_jack_gpios), - rx51_av_jack_gpios); - if (err) { - dev_err(card->dev, "Failed to add GPIOs\n"); - return err; + err = snd_soc_jack_add_gpios(&rx51_av_jack, + ARRAY_SIZE(rx51_av_jack_gpios), + rx51_av_jack_gpios); + if (err) { + dev_err(card->dev, "Failed to add GPIOs\n"); + return err; + } } return err; @@ -336,8 +338,11 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd) static int rx51_card_remove(struct snd_soc_card *card) { - snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), - rx51_av_jack_gpios); + struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card); + + if (!IS_ERR(pdata->jack_detection_gpio)) + snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), + rx51_av_jack_gpios); return 0; } @@ -470,8 +475,8 @@ static int rx51_soc_probe(struct platform_device *pdev) pdata->jack_detection_gpio = devm_gpiod_get(card->dev, "jack-detection"); if (IS_ERR(pdata->jack_detection_gpio)) { + /* Do not fail, qemu does not emulate jack detection gpio */ dev_err(card->dev, "could not get jack detection gpio\n"); - return PTR_ERR(pdata->jack_detection_gpio); } pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch"); -- 1.7.9.5

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[alsa-devel] [PATCH 00/16] Yet more cleanups of line6 drivers
by Takashi Iwai 28 Jan '15

28 Jan '15

Hi, here is a series of cleanup patches for line6 stuff. They are applied on top of my previous patches. The latest patchset is found in test/line6 branch of sound.git tree. Takashi Takashi Iwai (16): ALSA: line6: Minor refactoring ALSA: line6: Fix memory leak at probe error path ALSA: line6: Remove unused line6_nop_read() ALSA: line6: Reduce superfluous spinlock in midi.c ALSA: line6: Fix missing error handling in line6_pcm_acquire() ALSA: line6: Use logical OR ALSA: line6: Fix the error recovery in line6_pcm_acquire() ALSA: line6: Drop superfluous spinlock for trigger ALSA: line6: Use incremental loop ALSA: line6: Drop voodoo workarounds ALSA: line6: Rearrange PCM structure ALSA: line6: Consolidate URB unlink and sync helpers ALSA: line6: Use dev_err() ALSA: line6: Consolidate PCM stream buffer allocation and free ALSA: line6: Do clipping in volume / monitor manipulations ALSA: line6: Skip volume manipulation during silence copying sound/usb/line6/capture.c | 122 ++++++-------------------- sound/usb/line6/capture.h | 5 -- sound/usb/line6/driver.c | 157 ++++++++++++++++----------------- sound/usb/line6/driver.h | 2 - sound/usb/line6/midi.c | 18 ++-- sound/usb/line6/midi.h | 7 +- sound/usb/line6/pcm.c | 210 +++++++++++++++++++++++++++------------------ sound/usb/line6/pcm.h | 163 ++++++++++------------------------- sound/usb/line6/playback.c | 159 ++++++++++------------------------ sound/usb/line6/playback.h | 5 -- 10 files changed, 329 insertions(+), 519 deletions(-) -- 2.2.2

2 26

[alsa-devel] [patch] ASoC: Intel: remove an unused struct member
by Dan Carpenter 28 Jan '15

28 Jan '15

We never set the ->scratch pointer, so let's delete it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index eeef9a5..78daa46 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -277,7 +277,6 @@ struct sst_hsw { /* FW config */ struct sst_hsw_ipc_fw_ready fw_ready; struct sst_hsw_ipc_fw_version version; - struct sst_module *scratch; bool fw_done; struct sst_fw *sst_fw; @@ -2135,7 +2134,6 @@ void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) dma_free_coherent(hsw->dsp->dma_dev, SST_HSW_DX_CONTEXT_SIZE, hsw->dx_context, hsw->dx_context_paddr); sst_dsp_free(hsw->dsp); - kfree(hsw->scratch); kthread_stop(hsw->tx_thread); kfree(hsw->msg); }

2 1

[alsa-devel] [PATCH 0/5] Reduce races in line6 drivers
by Takashi Iwai 28 Jan '15

28 Jan '15

Hi, another day, another patchset. This time it mostly covers the races in PCM stream handling. This should work for most cases, but there are slightly uncovered conditions that is fairly difficult to fix. Maybe moving to the non-atomic PCM ops is one way, but it's too far ahead, ther are much more things to clean up before that. Takashi === Takashi Iwai (5): ALSA: line6: Fix racy loopback handling ALSA: line6: Clear prev_fbuf and prev_fsize properly ALSA: line6: Reorganize PCM stream handling ALSA: line6: Make common PCM pointer callback ALSA: line6: Handle error from line6_pcm_acquire() sound/usb/line6/capture.c | 113 +++------------ sound/usb/line6/capture.h | 1 - sound/usb/line6/pcm.c | 351 ++++++++++++++++++++++++++------------------- sound/usb/line6/pcm.h | 145 ++++++------------- sound/usb/line6/playback.c | 139 ++++-------------- sound/usb/line6/playback.h | 1 - sound/usb/line6/toneport.c | 17 ++- 7 files changed, 302 insertions(+), 465 deletions(-) -- 2.2.2

2 7

[alsa-devel] [PATCH 0/7 RESEND] ASoC: pcm512x: Clock master modes
by Peter Rosin 27 Jan '15

27 Jan '15

From: Peter Rosin <peda(a)axentia.se> Hi! [ Note that the dt change is in patch 5/7, for those only interested in that particular bit. ] This series implements BCLK master modes for the pcm512x driver. It has only been tested with the pcm5142 chip, but they are from the same family and should be compatible. I have mainly used the spec for the newer pcm5242 chip (also from the same family) as it fills in a lot of blanks in the pcm512x/pcm514x specs. The code has also seen most of its testing in a 3.10 environment, so there might be some forward-porting warts. But it is able to play sound in 3.18 as well, and most of the changes have little to do with anything but the clocking in the chip itself. This is technically not a clean resend, since I'm now sending the whole series to all parties, hopefully using the correct email addesses this time. I have also augmented the commit message of patch 1/7 according to the review comments by Lars-Peter Clausen. Cheers, Peter Peter Rosin (7): ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Change register default to match actual content after reset ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS .../devicetree/bindings/sound/pcm512x.txt | 25 +- include/sound/pcm.h | 12 + sound/core/pcm_lib.c | 85 ++ sound/soc/codecs/pcm512x.c | 955 +++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- 5 files changed, 1161 insertions(+), 25 deletions(-) -- 1.7.10.4

1 7

[alsa-devel] [PATCH] ASoC: rt5645: Add rt5650 codec support
by Bard Liao 27 Jan '15

27 Jan '15

This patch adds support for rt5650 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 158 +++++++++++++++++++++++++++++++++++++++++----- sound/soc/codecs/rt5645.h | 15 +++++ 2 files changed, 156 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 6e9cd8e..306db11 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -31,6 +31,7 @@ #include "rt5645.h" #define RT5645_DEVICE_ID 0x6308 +#define RT5650_DEVICE_ID 0x6419 #define RT5645_PR_RANGE_BASE (0xff + 1) #define RT5645_PR_SPACING 0x100 @@ -59,6 +60,10 @@ static const struct reg_default init_list[] = { }; #define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) +static const struct reg_default rt5650_init_list[] = { + {0xf6, 0x0100}, +}; + static const struct reg_default rt5645_reg[] = { { 0x00, 0x0000 }, { 0x01, 0xc8c8 }, @@ -86,6 +91,7 @@ static const struct reg_default rt5645_reg[] = { { 0x2a, 0x5656 }, { 0x2b, 0x5454 }, { 0x2c, 0xaaa0 }, + { 0x2d, 0x0000 }, { 0x2f, 0x1002 }, { 0x31, 0x5000 }, { 0x32, 0x0000 }, @@ -192,6 +198,8 @@ static const struct reg_default rt5645_reg[] = { { 0xdb, 0x0003 }, { 0xdc, 0x0049 }, { 0xdd, 0x001b }, + { 0xdf, 0x0008 }, + { 0xe0, 0x4000 }, { 0xe6, 0x8000 }, { 0xe7, 0x0200 }, { 0xec, 0xb300 }, @@ -241,6 +249,7 @@ static bool rt5645_volatile_register(struct device *dev, unsigned int reg) case RT5645_IRQ_CTRL3: case RT5645_INT_IRQ_ST: case RT5645_IL_CMD: + case RT5650_4BTN_IL_CMD1: case RT5645_VENDOR_ID: case RT5645_VENDOR_ID1: case RT5645_VENDOR_ID2: @@ -286,6 +295,7 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_STO_DAC_MIXER: case RT5645_MONO_DAC_MIXER: case RT5645_DIG_MIXER: + case RT5650_A_DAC_SOUR: case RT5645_DIG_INF1_DATA: case RT5645_PDM_OUT_CTRL: case RT5645_REC_L1_MIXER: @@ -376,6 +386,8 @@ static bool rt5645_readable_register(struct device *dev, unsigned int reg) case RT5645_IL_CMD: case RT5645_IL_CMD2: case RT5645_IL_CMD3: + case RT5650_4BTN_IL_CMD1: + case RT5650_4BTN_IL_CMD2: case RT5645_DRC1_HL_CTRL1: case RT5645_DRC2_HL_CTRL1: case RT5645_ADC_MONO_HP_CTRL1: @@ -1005,6 +1017,44 @@ static SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5645_if1_adc_in_mux = SOC_DAPM_ENUM("IF1 ADC IN source", rt5645_if1_adc_in_enum); +/* MX-2d [3] [2] */ +static const char * const rt5650_a_dac1_src[] = { + "DAC1", "Stereo DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_L_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_l_mux = + SOC_DAPM_ENUM("A DAC1 L source", rt5650_a_dac1_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac1_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC1_R_IN_SFT, rt5650_a_dac1_src); + +static const struct snd_kcontrol_new rt5650_a_dac1_r_mux = + SOC_DAPM_ENUM("A DAC1 R source", rt5650_a_dac1_r_enum); + +/* MX-2d [1] [0] */ +static const char * const rt5650_a_dac2_src[] = { + "Stereo DAC Mixer", "Mono DAC Mixer" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_l_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_L_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_l_mux = + SOC_DAPM_ENUM("A DAC2 L source", rt5650_a_dac2_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5650_a_dac2_r_enum, RT5650_A_DAC_SOUR, + RT5650_A_DAC2_R_IN_SFT, rt5650_a_dac2_src); + +static const struct snd_kcontrol_new rt5650_a_dac2_r_mux = + SOC_DAPM_ENUM("A DAC2 R source", rt5650_a_dac2_r_enum); + /* MX-2F [13:12] */ static const char * const rt5645_if2_adc_in_src[] = { "IF_ADC1", "IF_ADC2", "VAD_ADC" @@ -1149,11 +1199,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: hp_amp_power(codec, 1); /* headphone unmute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, RT5645_CP_FQ1_MASK | - RT5645_CP_FQ2_MASK | RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1173,12 +1228,16 @@ static int rt5645_hp_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_PRE_PMD: /* headphone mute sequence */ - snd_soc_update_bits(codec, RT5645_DEPOP_M3, - RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | - RT5645_CP_FQ3_MASK, - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | - (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | - (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + snd_soc_write(codec, RT5645_DEPOP_M3, 0x0737); + } else { + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | + RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + } regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, 0xfc00); snd_soc_update_bits(codec, RT5645_DEPOP_M1, @@ -1572,6 +1631,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPOR"), }; +static const struct snd_soc_dapm_widget rt5650_specific_dapm_widgets[] = { + SND_SOC_DAPM_MUX("A DAC1 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_l_mux), + SND_SOC_DAPM_MUX("A DAC1 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac1_r_mux), + SND_SOC_DAPM_MUX("A DAC2 L Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_l_mux), + SND_SOC_DAPM_MUX("A DAC2 R Mux", SND_SOC_NOPM, + 0, 0, &rt5650_a_dac2_r_mux), +}; + static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, @@ -1777,13 +1847,9 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, - { "DAC L1", NULL, "Stereo DAC MIXL" }, { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R1", NULL, "Stereo DAC MIXR" }, { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC L2", NULL, "Mono DAC MIXL" }, { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, - { "DAC R2", NULL, "Mono DAC MIXR" }, { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, { "SPK MIXL", "BST1 Switch", "BST1" }, @@ -1872,6 +1938,30 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "SPOR", NULL, "SPK amp" }, }; +static const struct snd_soc_dapm_route rt5650_specific_dapm_routes[] = { + { "A DAC1 L Mux", "DAC1", "DAC1 MIXL"}, + { "A DAC1 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC1 R Mux", "DAC1", "DAC1 MIXR"}, + { "A DAC1 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + + { "A DAC2 L Mux", "Stereo DAC Mixer", "Stereo DAC MIXL"}, + { "A DAC2 L Mux", "Mono DAC Mixer", "Mono DAC MIXL"}, + { "A DAC2 R Mux", "Stereo DAC Mixer", "Stereo DAC MIXR"}, + { "A DAC2 R Mux", "Mono DAC Mixer", "Mono DAC MIXR"}, + + { "DAC L1", NULL, "A DAC1 L Mux" }, + { "DAC R1", NULL, "A DAC1 R Mux" }, + { "DAC L2", NULL, "A DAC2 L Mux" }, + { "DAC R2", NULL, "A DAC2 R Mux" }, +}; + +static const struct snd_soc_dapm_route rt5645_specific_dapm_routes[] = { + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC R2", NULL, "Mono DAC MIXR" }, +}; + static int rt5645_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2287,6 +2377,22 @@ static int rt5645_probe(struct snd_soc_codec *codec) rt5645->codec = codec; + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5645: + snd_soc_dapm_add_routes(&codec->dapm, + rt5645_specific_dapm_routes, + ARRAY_SIZE(rt5645_specific_dapm_routes)); + break; + case CODEC_TYPE_RT5650: + snd_soc_dapm_new_controls(&codec->dapm, + rt5650_specific_dapm_widgets, + ARRAY_SIZE(rt5650_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5650_specific_dapm_routes, + ARRAY_SIZE(rt5650_specific_dapm_routes)); + break; + } + rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); @@ -2418,6 +2524,7 @@ static const struct regmap_config rt5645_regmap = { static const struct i2c_device_id rt5645_i2c_id[] = { { "rt5645", 0 }, + { "rt5650", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); @@ -2450,9 +2557,18 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, } regmap_read(rt5645->regmap, RT5645_VENDOR_ID2, &val); - if (val != RT5645_DEVICE_ID) { + + switch (val) { + case RT5645_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5645; + break; + case RT5650_DEVICE_ID: + rt5645->codec_type = CODEC_TYPE_RT5650; + break; + default: dev_err(&i2c->dev, - "Device with ID register %x is not rt5645\n", val); + "Device with ID register %x is not rt5645 or rt5650\n", + val); return -ENODEV; } @@ -2463,6 +2579,14 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + if (rt5645->codec_type == CODEC_TYPE_RT5650) { + ret = regmap_register_patch(rt5645->regmap, rt5650_init_list, + ARRAY_SIZE(rt5650_init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Apply rt5650 patch failed: %d\n", + ret); + } + if (rt5645->pdata.in2_diff) regmap_update_bits(rt5645->regmap, RT5645_IN2_CTRL, RT5645_IN_DF2, RT5645_IN_DF2); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index a815e36..7454231 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -47,6 +47,7 @@ #define RT5645_STO_DAC_MIXER 0x2a #define RT5645_MONO_DAC_MIXER 0x2b #define RT5645_DIG_MIXER 0x2c +#define RT5650_A_DAC_SOUR 0x2d #define RT5645_DIG_INF1_DATA 0x2f /* Mixer - PDM */ #define RT5645_PDM_OUT_CTRL 0x31 @@ -150,6 +151,8 @@ #define RT5645_IL_CMD 0xdb #define RT5645_IL_CMD2 0xdc #define RT5645_IL_CMD3 0xdd +#define RT5650_4BTN_IL_CMD1 0xdf +#define RT5650_4BTN_IL_CMD2 0xe0 #define RT5645_DRC1_HL_CTRL1 0xe7 #define RT5645_DRC2_HL_CTRL1 0xe9 #define RT5645_MUTI_DRC_CTRL1 0xea @@ -472,6 +475,12 @@ #define RT5645_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) #define RT5645_DAC_L2_DAC_R_VOL_SFT 4 +/* Analog DAC1/2 Input Source Control (0x2d) */ +#define RT5650_A_DAC1_L_IN_SFT 3 +#define RT5650_A_DAC1_R_IN_SFT 2 +#define RT5650_A_DAC2_L_IN_SFT 1 +#define RT5650_A_DAC2_R_IN_SFT 0 + /* Digital Interface Data Control (0x2f) */ #define RT5645_IF1_ADC2_IN_SEL (0x1 << 15) #define RT5645_IF1_ADC2_IN_SFT 15 @@ -2175,6 +2184,11 @@ enum { RT5645_DMIC_DATA_GPIO11, }; +enum { + CODEC_TYPE_RT5645, + CODEC_TYPE_RT5650, +}; + struct rt5645_priv { struct snd_soc_codec *codec; struct rt5645_platform_data pdata; @@ -2184,6 +2198,7 @@ struct rt5645_priv { struct snd_soc_jack *mic_jack; struct delayed_work jack_detect_work; + int codec_type; int sysclk; int sysclk_src; int lrck[RT5645_AIFS]; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: Intel: Used lock version to update shim registers
by Jie Yang 27 Jan '15

27 Jan '15

We need hold lock each time updating shirm registers, otherwise, we may set unexpected values to them when they are set in different thread at different time sequence. The notification work will be scheduled in global work queue, which won't hold this sst->spinlock itself, so here we need change to use the lock version to update shim registers. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index eeef9a5..e18cb07 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -653,11 +653,11 @@ static void hsw_notification_work(struct work_struct *work) } /* tell DSP that notification has been handled */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD, + sst_dsp_shim_update_bits(hsw->dsp, SST_IPCD, SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); /* unmask busy interrupt */ - sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); + sst_dsp_shim_update_bits(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); } static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: ts3a227e: Add dts property that allows to specify micbias voltage
by Anatol Pomozov 27 Jan '15

27 Jan '15

The voltage controls key press threshold. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/ts3a227e.txt | 5 ++++ sound/soc/codecs/ts3a227e.c | 27 ++++++++++++++++++++++ 2 files changed, 32 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ts3a227e.txt b/Documentation/devicetree/bindings/sound/ts3a227e.txt index e8bf23e..a836881 100644 --- a/Documentation/devicetree/bindings/sound/ts3a227e.txt +++ b/Documentation/devicetree/bindings/sound/ts3a227e.txt @@ -13,6 +13,11 @@ Required properties: - interrupt-parent: The parent interrupt controller - interrupts: Interrupt number for /INT pin from the 227e +Optional properies: + - ti,micbias: Intended MICBIAS voltage (datasheet section 9.6.7). + Select 0/1/2/3/4/5/6/7 to specify MACBIAS voltage + 2.1V/2.2V/2.3V/2.4V/2.5V/2.6V/2.7V/2.8V + Default value is "1" (2.2V). Examples: diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 54d431c..5ecfa31 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -79,6 +79,10 @@ static const int ts3a227e_buttons[] = { /* TS3A227E_REG_SETTING_2 0x05 */ #define KP_ENABLE 0x04 +/* TS3A227E_REG_SETTING_3 0x06 */ +#define MICBIAS_SETTING_SFT (3) +#define MICBIAS_SETTING_MASK (0x7 << MICBIAS_SETTING_SFT) + /* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ #define TYPE_3_POLE 0x01 #define TYPE_4_POLE_OMTP 0x02 @@ -248,6 +252,21 @@ static const struct regmap_config ts3a227e_regmap_config = { .num_reg_defaults = ARRAY_SIZE(ts3a227e_reg_defaults), }; +static int ts3a227e_parse_dt(struct ts3a227e *ts3a227e, struct device_node *np) +{ + u32 micbias; + int err; + + err = of_property_read_u32(np, "ti,micbias", &micbias); + if (!err) { + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_3, + MICBIAS_SETTING_MASK, + (micbias & 0x07) << MICBIAS_SETTING_SFT); + } + + return 0; +} + static int ts3a227e_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -266,6 +285,14 @@ static int ts3a227e_i2c_probe(struct i2c_client *i2c, if (IS_ERR(ts3a227e->regmap)) return PTR_ERR(ts3a227e->regmap); + if (dev->of_node) { + ret = ts3a227e_parse_dt(ts3a227e, dev->of_node); + if (ret) { + dev_err(dev, "Failed to parse device tree: %d\n", ret); + return ret; + } + } + ret = devm_request_threaded_irq(dev, i2c->irq, NULL, ts3a227e_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "TS3A227E", ts3a227e); -- 2.2.0.rc0.207.ga3a616c

2 1

[alsa-devel] [PATCH] ASoC: ts3a227e: Remap keys to match Android headset specification
by Anatol Pomozov 27 Jan '15

27 Jan '15

ts3a227e datasheet says typical key resistance is key1 50 Om key2 135 Om key3 240 Om key4 470 Om The android headset specification expect buttons impedance: A (MEDIA) 0-70 Om D (VOICECOMMAND) 110-180 Om B (VOLUMEUP) 210-290 Om C (VOLUMEDOWN) 360-680 Om Thus key mapping should be key1 - MEDIA key2 - VOICECOMMAND key3 - VOLUMEUP key3 - VOLUMEDOWN Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/ts3a227e.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index 9f2dced..54d431c 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -221,9 +221,9 @@ int ts3a227e_enable_jack_detect(struct snd_soc_component *component, struct ts3a227e *ts3a227e = snd_soc_component_get_drvdata(component); snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); ts3a227e->jack = jack; ts3a227e_jack_report(ts3a227e); -- 2.2.0.rc0.207.ga3a616c

3 3

[alsa-devel] [PATCH] ASoC: wm8731: init mutex in i2c init path
by Manuel Lauss 27 Jan '15

27 Jan '15

The I2C init path forgot to init the mutex, leading to an oops when controls are accessed. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Fixes an oops on bootup when alsactl tries to restore codec parameters on my MIPS DB1300 board. sound/soc/codecs/wm8731.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index b9211b4..b115ed8 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -717,6 +717,8 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, if (wm8731 == NULL) return -ENOMEM; + mutex_init(&wm8731->lock); + wm8731->regmap = devm_regmap_init_i2c(i2c, &wm8731_regmap); if (IS_ERR(wm8731->regmap)) { ret = PTR_ERR(wm8731->regmap); -- 2.2.1 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: atmel_ssc_dai: fix start event for I2S mode
by Bo Shen 27 Jan '15

27 Jan '15

According to the I2S specification information as following: - WS = 0, channel 1 (left) - WS = 1, channel 2 (right) So, the start event should be TF/RF falling edge. Reported-by: Songjun Wu <songjun.wu(a)atmel.com> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: None sound/soc/atmel/atmel_ssc_dai.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 99ff35e..e691aab 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -348,7 +348,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; - int start_event; int ret; int fslen, fslen_ext; @@ -457,19 +456,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * The SSC transmit clock is obtained from the BCLK signal on * on the TK line, and the SSC receive clock is * generated from the transmit clock. - * - * For single channel data, one sample is transferred - * on the falling edge of the LRC clock. - * For two channel data, one sample is - * transferred on both edges of the LRC clock. */ - start_event = ((channels == 1) - ? SSC_START_FALLING_RF - : SSC_START_EDGE_RF); - rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) - | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? @@ -478,14 +468,14 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(RFMR_FSLEN, 0) - | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) | SSC_BF(RFMR_DATLEN, (bits - 1)); tcmr = SSC_BF(TCMR_PERIOD, 0) | SSC_BF(TCMR_STTDLY, START_DELAY) - | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? @@ -495,7 +485,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(TFMR_FSLEN, 0) - | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) | SSC_BF(TFMR_DATLEN, (bits - 1)); -- 2.3.0.rc0

2 3

[alsa-devel] add Steinberg UR44 USB Audio Interface support to quirks-table
by Vlastimil Wagner 27 Jan '15

27 Jan '15

hi In what state is the driver for Steinberg UR44 thanks Bukowski

2 1

[alsa-devel] [PATCH 00/10] ASoC: sta32x: Add device tree binding
by Thomas Niederprüm 27 Jan '15

27 Jan '15

This patchset is the result of comparing the sta32x driver with the sta350 driver with the goal to make the sta32x driver device tree capable. In the merging process the coding style of the sta350 driver was adopted since it seemed more up to date. I did my best to break up the changes in logical chunks. Additionally some small fixes were added. The comparison with the sta350 lead to the following changes: - Switch to direct usage of the regmap API - Use the kernel gpio framework to control the reset pin of the sta32x - adopt calculation of mlck divider and extrapolation ratio from sta350 - add device tree bindings - Change the Codecs DAI name to be coherent with the sta350 driver Moreover some minor fixes were added: - Add the status register - Fix bit shift value for the IDE bit in the CONFF register - Add description for the driver and I2C dependency in Kconfig Thomas Niederprüm (10): ASoC: sta32x: Convert to direct regmap API usage. ASoC: sta32x: make sta32x a gpio consumer for the reset GPIO ASoC: sta32x: use DECLARE_TLV_DB_RANGE macro. ASoC: sta32x: add status register. ASoC: sta32x: move code to calculate mclk divider and extrapolation ratio to sta32x_hw_params() ASoC: sta32x: add device tree binding. ASoC: sta32x: use dev_dbg() for debug output. ASoC: sta32x: minor Kconfig update. ASoC: sta32x: correct bit shift value for IDE register. ASoC: sta32x: change dai name to be in line with the sta350 driver. .../devicetree/bindings/sound/st,sta32x.txt | 92 ++++ include/sound/sta32x.h | 18 +- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/sta32x.c | 537 +++++++++++++-------- sound/soc/codecs/sta32x.h | 4 +- 5 files changed, 450 insertions(+), 205 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/st,sta32x.txt -- 2.1.1

4 17

[alsa-devel] [PATCH_V3 0/4] sound:soc: jz4740: DT, dynamic sampling, enable clocks
by Zubair Lutfullah Kakakhel 27 Jan '15

27 Jan '15

Hi, Here are a few simple patches for the jz4740. First adds dynamic sampling support to jz4740-i2s. Then two to add a simple binding and DT support. Then a patch to set the clock rate for i2s. These are in preparation for jz4780 and ci20 later on. Patches are based on 3.19-rc6. Quite disjoint and stay within jz4740 so should apply easily on other trees. If you would like to have them rebased to a different tree, please tell. Thank-you V3 changes: Added dma binding to example but forgot to put it in required properties. V2 changes: Removed a redundant #define Added #ifdef CONFIG_OF Added dma in binding document Removed a redundant call to clk_prepare_enable Zubair Lutfullah Kakakhel (4): sound: soc: jz4740: Add dynamic sampling rate support to jz4740-i2s dt: sound: jz4740: Add binding documentation for jz4740-i2s sound: soc: jz4740: Add DT support to jz4740-i2s driver sound: jz4740: set i2s clk rate to 12MHz .../bindings/sound/ingenic,jz4740-i2s.txt | 23 ++++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.c | 22 ++++++++++++++++++++- 2 files changed, 44 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt -- 1.9.1

2 6

Re: [alsa-devel] [PATCH v2 00/16] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 27 Jan '15

27 Jan '15

Hello Tobias, On 24/01/15 02:53, Tobias Jakobi wrote: > Hello, > > I've tested the series on my X2 and so far I haven't encountered any > obvious issues with it. > > I have a small question though. With the move to simple-audio-card the > old driver (selected by SND_SOC_ODROIDX2) is probably going away after > some time. > Currently SND_SOC_ODROIDX2 also selects SND_SOC_MAX98090 and > SND_SAMSUNG_I2S, which I believe (correct me if I'm wrong) are necessary > to use the sound subsystem even with simple-audio-card (since the > max98090 codec has to be built at least). I've found no way of selecting > these two manually, so at the moment I need both CONFIG_SND_SIMPLE_CARD > and SND_SOC_ODROIDX2 enabled to use sound. Is this intended behaviour? Thanks for the feedback. It seems I finally forgot to update Kconfig, presumably we should make SND_SOC_ODROIDX2 selecting SND_SIMPLE_CARD, i.e. diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 58c797a..7fb728c 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -247,6 +247,7 @@ config SND_SOC_ODROIDX2 depends on SND_SOC_SAMSUNG select SND_SOC_MAX98090 select SND_SAMSUNG_I2S + select SND_SIMPLE_CARD help Say Y here to enable audio support for the Odroidx2. Making SND_SOC_SAMSUNG selecting SND_SIMPLE_CARD is probably not a good idea and still we would be missing the codec part. I'm going to post a patch as above. > And while I'm at it. I'm booting the board with an upstream u-boot > version which sets the MPLL to 800MHz, which AFAIK is the clocking rate > that makes less 'trouble' (but at the expense of performance). > I was wondering if anyone can comment on whether the recent rework has > any influence on the behaviour when booting with the vendor u-boot > (which sets the MPLL to 880MHz). IIRC then clock rounding issues arose > there. > > It would be interesting to know if this series gets us closer to drive > the MPLL with the recommended (?) higher clocking rate. No, (unfortunately) this series has nothing to do with MPLL. The whole clock tree for sound is configured by the kernel, including the EPLL, which provides reference clock for the audio subsystem. MPLL and EPLL are normally used in different clock domains. So there should be no any clock rounding issues related to bootloader as far as audio is concerned. Please let me know if you find any related problems. -- Regards, Sylwester

1 0

[alsa-devel] [PATCH v2 1/3] ASoC: codecs: wm8904: add dt ids table
by Bo Shen 27 Jan '15

27 Jan '15

From: Alexander Morozov <linux(a)meltdown.ru> Signed-off-by: Alexander Morozov <linux(a)meltdown.ru> [Add driver data to distinguish device type] Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: - Add driver data for distinguish the device capability. sound/soc/codecs/wm8904.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..6e3f175 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2255,10 +2255,32 @@ static const struct i2c_device_id wm8904_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id); +#ifdef CONFIG_OF +static enum wm8904_type wm8904_data = WM8904; +static enum wm8904_type wm8912_data = WM8912; + +static const struct of_device_id wm8904_of_match[] = { + { + .compatible = "wlf,wm8904", + .data = &wm8904_data, + }, { + .compatible = "wlf,wm8912", + .data = &wm8912_data, + }, { + .compatible = "wlf,wm8918", + .data = &wm8904_data, + }, { + /* sentinel */ + } +}; +MODULE_DEVICE_TABLE(of, wm8904_of_match); +#endif + static struct i2c_driver wm8904_i2c_driver = { .driver = { .name = "wm8904", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(wm8904_of_match), }, .probe = wm8904_i2c_probe, .remove = wm8904_i2c_remove, -- 2.1.0.24.g4109c28

4 9

[alsa-devel] Regression in kernel 3.19-rc5 for usb2 scarlett 18i20
by Guido Aulisi 27 Jan '15

27 Jan '15

Hi, I noticed a regression in 3.19-rc5 kernel (as shipped by fedora rawhide) with usb2 focusrite scarlett 18i20. I get a lot of xruns with jack, even at 1024 frame buffer and 2 periods. In 3.17.8 it works quite well without xruns at the same settings. I use a fedora 20 (64bit) laptop with 8GB ram, I only installed the rawhide kernel to try the new included scarlett mixer, no other software components changed. I can try to do some other test, but compiling the kernel on this laptop is time consuming. Thanks Guido

3 5

[alsa-devel] [PATCH 0/7] ASoC: pcm512x: Clock master modes
by Peter Rosin 27 Jan '15

27 Jan '15

From: Peter Rosin <peda(a)axentia.se> Hi! This series implements BCLK master modes for the pcm512x driver. It has only been tested with the pcm5142 chip, but they are from the same family and should be compatible. I have mainly used the spec for the newer pcm5242 chip (also from the same family) as it fills in a lot of blanks in the pcm512x/pcm514x specs. The code has also seen most of its testing in a 3.10 environment, so there might be some forward-porting warts. But it is able to play sound in 3.18 as well, and most of the changes have little to do with anything but the clocking in the chip itself. Cheers, Peter Peter Rosin (7): ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges() ASoC: pcm512x: Fix spelling of register field names. ASoC: pcm512x: Change register default to match actual content after reset ASoC: pcm512x: Support mastering BCLK/LRCLK without using the PLL ASoC: pcm512x: Support mastering BCLK/LRCLK using the PLL ASoC: pcm512x: Avoid the PLL for the DAC clock, if possible ASoC: pcm512x: Support SND_SOC_DAIFMT_CBM_CFS .../devicetree/bindings/sound/pcm512x.txt | 25 +- include/sound/pcm.h | 12 + sound/core/pcm_lib.c | 85 ++ sound/soc/codecs/pcm512x.c | 955 +++++++++++++++++++- sound/soc/codecs/pcm512x.h | 109 ++- 5 files changed, 1161 insertions(+), 25 deletions(-) -- 1.7.10.4

2 8

[alsa-devel] [PATCH] ASoC: rt5640: Add RT5642 ACPI ID for Intel Baytrail
by Jarkko Nikula 26 Jan '15

26 Jan '15

Asus T100TAF uses ACPI ID "10EC5642" for its audio codec. I suppose it is updated ACPI ID for the RT5642 codec since some earlier platforms are using "10EC5640" with the RT5642 too. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: Bard Liao <bardliao(a)realtek.com> Cc: Oder Chiou <oder_chiou(a)realtek.com> --- ACPI ID found from the DSDT table of Asus T100TAF. This makes the rt5640 codec driver probing but the audio itself is not working due a bit different ACPI data and perhaps HW differences too compared to for instance Asus T100TA. --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b386958de042..178e55d4d481 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2125,6 +2125,7 @@ MODULE_DEVICE_TABLE(of, rt5640_of_match); static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { "10EC5640", 0 }, + { "10EC5642", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: rt286: Fix capture volume setting issue
by Bard Liao 26 Jan '15

26 Jan '15

The purpose of rt286_adc_event is to mute/numnte the ADC mixer. However, it will also set the capture volume to default value. As a result, "ADC0 Capture Volume" is not working if it is set before capture start. This patch remove rt286_adc_event and add "ADC0 Capture Switch" to mute/unmute ADC mixer. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 38 ++++++-------------------------------- 1 file changed, 6 insertions(+), 32 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 1d1c7f8..847cc4b 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -417,6 +417,8 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); static const struct snd_kcontrol_new rt286_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R("ADC0 Capture Switch", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 7, 1, 1), SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, @@ -538,32 +540,6 @@ static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, return 0; } -static int rt286_adc_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - unsigned int nid; - - nid = (w->reg >> 20) & 0xff; - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7000); - break; - case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(codec, - VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), - 0x7080, 0x7080); - break; - default: - return 0; - } - - return 0; -} - static int rt286_vref_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -667,12 +643,10 @@ static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), /* ADC Mux */ - SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, - &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), - SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, - &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | - SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux), + SND_SOC_DAPM_MUX("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux), /* Audio Interface */ SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 0/2] ASoC: Fix wm97xx touchscreen driver
by Lars-Peter Clausen 26 Jan '15

26 Jan '15

The wm97xx touchscreen driver will bind itself to the snd_ac97 device registered by CODEC drivers and expects that the device has already been reset by the CODEC driver. Previous to commit 6794f709b712 ("ASoC: ac97: Drop delayed device registration") the snd_ac97 device was only registered after the CODEC driver probe function had finished running, but starting with the commit it is registered within snd_soc_new_ac97_codec(). This breaks the touchscreen driver as the reset is no longer performed before the touchscreen driver probe function runs. This patch series introduces a new function snd_soc_alloc_ac97_codec() which allocates the snd_ac97 device, but does not yet register it yet. This allows the CODEC drivers to perform the reset before the device is registered. The series is meant to perform a minimum amount of changes while fixing the issue to prevent introducing other regressions. I have a few more patches that consolidate the reset handling in the drivers and puts it into the core in snd_soc_new_ac97_codec() which makes the split in the drivers unnecessary, but that's something for next and not for stable. Manuel can you give this another round of testing? Thanks, - Lars Lars-Peter Clausen (2): ASoC: Add support for allocating AC'97 device before registering it ASoC: wm97xx: Reset AC'97 device before registering it include/sound/soc.h | 1 + sound/soc/codecs/wm9705.c | 16 ++++++++++------ sound/soc/codecs/wm9712.c | 12 ++++++++---- sound/soc/codecs/wm9713.c | 12 ++++++++---- sound/soc/soc-ac97.c | 34 +++++++++++++++++++++++++++++----- 5 files changed, 56 insertions(+), 19 deletions(-) -- 1.8.0

4 6

[alsa-devel] broadwell-audio is unable to initialize on certain laptops
by Tom Li 26 Jan '15

26 Jan '15

Hello everyone. I'm here for reporting an issue. I bought a new ThinkPad Helix 2, but I found that the built-in sound output device is not recognize, only external HDMI sound output is possible. # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 When I check dmesg output, I found some error message that indicated that broadwell-audio is unable to initialize. # dmesg | egrep "sound|audio" [ 1.827797] input: HDA Intel HDMI HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:03.0/sound/card0/input8 [ 1.827935] input: HDA Intel HDMI HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:03.0/sound/card0/input9 [ 1.828071] input: HDA Intel HDMI HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:03.0/sound/card0/input10 [ 1.952913] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 1.952924] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.062812] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.062837] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.350951] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.350987] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.375744] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.375759] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.386019] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.386037] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 2.706751] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 2.706777] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.042190] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.042202] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.048200] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.048212] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.048382] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.048391] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.199788] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.199812] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.208241] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.208259] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.216577] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.216600] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.270629] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.270659] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.324400] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.324419] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.604723] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.604755] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.604800] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.604817] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.673159] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.673190] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.727312] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.727352] platform broadwell-audio: Driver broadwell-audio requests probe deferral [ 3.727387] broadwell-audio broadwell-audio: ASoC: CPU DAI System Pin not registered [ 3.727401] platform broadwell-audio: Driver broadwell-audio requests probe deferral I tried both 3.18.3 and 3.19.0-rc6, the results were same. It seems that we need to add some quirks for this hardware. I also found a similar bug on Launchpad [0]. I filed a kernel bug [1] and I hope someone could fix it. [0]: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1413446 [1]: https://bugzilla.kernel.org/show_bug.cgi?id=92061 Thanks.

1 0

[alsa-devel] [alsa] Crackling after 6.45 hours
by a.renault＠kaptalia.com 26 Jan '15

26 Jan '15

Hi everyone, I currently facing the following issue : My sound crackling after 6.45 hours of music without reinitialization. I am using sdl-mixer, sdl and alsa. I deployed a bug on libsdl at the address : https://bugzilla.libsdl.org/show_bug.cgi?id=2794 And after some works we discover that the snd_pcm_writei answer "-1073741824". The error code "-1073741824" is not one of -EINTR, -EPIPE, -ESTRPIPE, -EAGAIN. So "snd_pcm_recover" does not handle anything. If you are brave enought please take a look at the link, Any idea would be greatly appreciated. Best regards, -- Adrien

4 6

[alsa-devel] [PATCH_V2 0/4] sound:soc: jz4740: DT, dynamic sampling, enable clocks
by Zubair Lutfullah Kakakhel 26 Jan '15

26 Jan '15

Hi, Here are a few simple patches for the jz4740. First adds dynamic sampling support to jz4740-i2s. Then two to add a simple binding and DT support. Then a patch to set the clock rate for i2s. These are in preparation for jz4780 and ci20 later on. Patches are based on 3.19-rc6. Quite disjoint and stay within jz4740 so should apply easily on other trees. If you would like to have them rebased to a different tree, please tell. Thank-you V2 changes: Removed a redundant #define Added #ifdef CONFIG_OF Added dma in binding document Removed a redundant call to clk_prepare_enable Zubair Lutfullah Kakakhel (4): sound: soc: jz4740: Add dynamic sampling rate support to jz4740-i2s dt: sound: jz4740: Add binding documentation for jz4740-i2s sound: soc: jz4740: Add DT support to jz4740-i2s driver sound: jz4740: set i2s clk rate to 12MHz .../bindings/sound/ingenic,jz4740-i2s.txt | 21 +++++++++++++++++++++ sound/soc/jz4740/jz4740-i2s.c | 22 +++++++++++++++++++++- 2 files changed, 42 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt -- 1.9.1

2 5

[alsa-devel] [PATCH 0/6] sequencer fixes
by Clemens Ladisch 26 Jan '15

26 Jan '15

One bug fix, and various unimportant cleanups for -next. include/sound/seq_kernel.h | 9 +-------- sound/core/seq/seq_clientmgr.c | 3 +-- sound/core/seq/seq_dummy.c | 31 ------------------------------- sound/core/seq/seq_ports.c | 9 +++------ sound/core/seq/seq_ports.h | 1 - 5 files changed, 5 insertions(+), 48 deletions(-)

2 8

[alsa-devel] [PATCH 0/3] Another round of line6 cleanup
by Takashi Iwai 26 Jan '15

26 Jan '15

Hi, here is yet another series of cleanup patches for line6 drivers. This time, the extra allocation for struct usb_line6 was removed by using snd_card_new()'s allocation feature. In addition to that, a few trivial cleanups. The test/line6 branch of sound git tree was updated with these patches, too. Takashi === Takashi Iwai (3): ALSA: line6: Drop interface argument from private_init and disconnect callbacks ALSA: line6: Let snd_card_new() allocate private data ALSA: line6: Minor tidy up in line6_probe() sound/usb/line6/driver.c | 52 ++++++++++++++++++++-------------------------- sound/usb/line6/driver.h | 8 ++++--- sound/usb/line6/pod.c | 21 +++++++------------ sound/usb/line6/podhd.c | 13 ++++-------- sound/usb/line6/toneport.c | 21 +++++++------------ sound/usb/line6/variax.c | 19 ++++++----------- 6 files changed, 54 insertions(+), 80 deletions(-) -- 2.2.2

1 3

[alsa-devel] ASoC fixes for v3.19
by Mark Brown 26 Jan '15

26 Jan '15

The following changes since commit 26bc420b59a38e4e6685a73345a0def461136dce: Linux 3.19-rc6 (2015-01-25 20:04:41 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.19-rc6 for you to fetch changes up to 88343ee38d7ae55e39cda74c5d137cb2c454e66c: Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/simple', 'asoc/fix/ts3a227e', 'asoc/fix/wm8904' and 'asoc/fix/wm8960' into asoc-linus (2015-01-26 11:29:58 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.19 There's a lot more fixes here than I'd like since I've been lax in sending things throughout the release cycle but there's only two in generic code and they've had quite some time to cook in -next so hopefully on balance are OK. The two generic fixes are a fix for crashes on capture DAIs in the compress code and a fix for error handling on probe failures which would have been harmless in the past but now oopses with the new dynamic probe code. ---------------------------------------------------------------- Andrew Jackson (1): ASoC: adi: Add missing return statement. Aurelien BOUIN (1): ASoC: fsl_esai: Fix incorrect xDC field width of xCCR registers Bard Liao (1): ASoC: rt286: set the same format for dac and adc Bo Shen (1): ASoC: wm8904: fix runtime warning Cheng-Yi Chiang (1): ASoC: ts3a227e: Check and report jack status at probe Fabio Estevam (2): ASoC: fsl_ssi: Fix irq error check ASoC: fsl: imx-wm8962: Set the card owner field Geert Uytterhoeven (1): ASoC: simple-card: Fix crash in asoc_simple_card_unref() Jianqun Xu (1): ASoC: rockchip: i2s: applys rate symmetry for CPU DAI Jie Yang (2): ASoC: Intel: Don't change offset of block allocator during fixed allocate ASoC: Intel: Add NULL checks for the stream pointer Mark Brown (4): Merge remote-tracking branch 'asoc/fix/fsl-esai' into asoc-linus Merge remote-tracking branch 'asoc/fix/pcm512x' into asoc-linus Merge remote-tracking branches 'asoc/fix/adi', 'asoc/fix/compress', 'asoc/fix/fsl-ssi', 'asoc/fix/imx', 'asoc/fix/intel', 'asoc/fix/omap', 'asoc/fix/rockchip' and 'asoc/fix/rt286' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5677', 'asoc/fix/simple', 'asoc/fix/ts3a227e', 'asoc/fix/wm8904' and 'asoc/fix/wm8960' into asoc-linus Oder Chiou (1): ASoC: rt5677: Modify the behavior that updates the PLL parameter. Peter Rosin (1): ASoC: pcm512x: Fix DSP program selection Peter Ujfalusi (1): ASoC: omap-mcbsp: Correct CBM_CFS dai format configuration Qais Yousef (1): ASoC: soc-compress.c: fix NULL dereference Zidan Wang (1): ASoC: wm8960: Fix capture sample rate from 11250 to 11025 sound/soc/adi/axi-i2s.c | 2 ++ sound/soc/codecs/pcm512x.c | 2 +- sound/soc/codecs/rt286.c | 6 ++---- sound/soc/codecs/rt5677.c | 18 ++++++++++++++---- sound/soc/codecs/ts3a227e.c | 6 ++++++ sound/soc/codecs/wm8904.c | 23 +++++++++++++++-------- sound/soc/codecs/wm8960.c | 2 +- sound/soc/fsl/fsl_esai.h | 2 +- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/imx-wm8962.c | 1 + sound/soc/generic/simple-card.c | 7 +++---- sound/soc/intel/sst-firmware.c | 13 +++++++------ sound/soc/intel/sst-haswell-ipc.c | 30 ++++++++++++++++++++++++++++++ sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 1 + sound/soc/soc-compress.c | 9 ++++++--- 16 files changed, 93 insertions(+), 35 deletions(-)

2 1

[alsa-devel] [PATCH 0/4] sound:soc: jz4740: DT, dynamic sampling, enable clocks
by Zubair Lutfullah Kakakhel 26 Jan '15

26 Jan '15

Hi, Here are a few simple patches for the jz4740. First adds dynamic sampling support to jz4740-i2s. Then two to add a simple binding and DT support. Then a patch to enable the codec clock. These are in preparation for jz4780 and ci20 later on. These are based on 3.19-rc6. If you would like to have them rebased to a different tree, please tell. Thank-you Zubair Lutfullah Kakakhel (4): sound: soc: jz4740: Add dynamic sampling rate support to jz4740-i2s dt: sound: jz4740: Add binding documentation for jz4740-i2s sound: soc: jz4740: Add DT support to jz4740-i2s driver sound: jz4740: Enable codec clock during dai_probe .../bindings/sound/ingenic,jz4740-i2s.txt | 18 ++++++++++++++++ sound/soc/jz4740/jz4740-i2s.c | 24 +++++++++++++++++++++- 2 files changed, 41 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/ingenic,jz4740-i2s.txt -- 1.9.1

3 12

[alsa-devel] [PATCH - UCM 1/1] ucm: add binary configure file parse
by han.lu＠intel.com 26 Jan '15

26 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> with cset command, UCM set kcontrol parameters directly: cset "name='<KCONTROL_NAME>' 1<,2,3,...>" This patch enables UCM to set kcontrol with parameters from configure file: cset-bin-file "name='<KCONTROL_NAME>' <path/to/file>" where "cset-bin-file" is a newly added keyword alongside of "cset", to indicate cset with binary data in file. The binary data in file is parameter for audio DSPs, and it's just passed by UCM/ALSA as raw data. The data type of parameter elements must be byte, and the count must matches driver definition. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 37ae4c8..182f174 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -34,6 +34,7 @@ #include <ctype.h> #include <stdarg.h> #include <pthread.h> +#include <sys/stat.h> /* * misc @@ -160,11 +161,65 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int binary_file_parse(snd_ctl_elem_value_t *dst, + snd_ctl_elem_info_t *info, + const char *filepath) +{ + int err = 0; + int fd; + struct stat st; + size_t sz; + ssize_t sz_read; + char *res; + snd_ctl_elem_type_t type; + unsigned int idx, count; + + type = snd_ctl_elem_info_get_type(info); + if (type != SND_CTL_ELEM_TYPE_BYTES) { + uc_error("only support byte type!"); + err = -EINVAL; + return err; + } + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + count = snd_ctl_elem_info_get_count(info); + if (sz != count || sz > sizeof(dst->value.bytes)) { + uc_error("invalid parameter size %d!", sz); + err = -EINVAL; + goto __fail; + } + res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + sz_read = read(fd, res, sz); + if (sz_read < 0 || (size_t)sz_read != sz) { + err = -errno; + goto __fail_read; + } + for (idx = 0; idx < sz; idx++) + snd_ctl_elem_value_set_byte(dst, idx, *(res + idx)); + __fail_read: + free(res); + __fail: + close(fd); + return err; +} + extern int __snd_ctl_ascii_elem_id_parse(snd_ctl_elem_id_t *dst, const char *str, const char **ret_ptr); -static int execute_cset(snd_ctl_t *ctl, const char *cset) +static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) { const char *pos; int err; @@ -194,7 +249,10 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset) err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); if (err < 0) goto __fail; err = snd_ctl_elem_write(ctl, value); @@ -239,6 +297,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail_nomem; break; case SEQUENCE_ELEMENT_TYPE_CSET: + case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: if (cdev == NULL) { const char *cdev1 = NULL, *cdev2 = NULL; err = get_value3(&cdev1, "PlaybackCTL", @@ -274,7 +333,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail; } } - err = execute_cset(ctl, s->data.cset); + err = execute_cset(ctl, s->data.cset, s->type); if (err < 0) { uc_error("unable to execute cset '%s'\n", s->data.cset); goto __fail; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index d7517f6..9e1cb41 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -306,6 +306,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-bin-file") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-bin-file requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 87f14a2..c1655c7 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -47,6 +47,7 @@ #define SEQUENCE_ELEMENT_TYPE_CSET 2 #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 +#define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 struct ucm_value { struct list_head list; -- 2.1.0

2 1

[alsa-devel] [PATCH 0/3] ARM: edma: Correct header file usage
by Peter Ujfalusi 26 Jan '15

26 Jan '15

Hi, The linux/platform_data/edma.h file was used for API definition as well, which is not correct since the header should only contain platform data related structures, defines, etc. Mark: I think this series can go via arm since it is just changing include stuff under ASoC. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-evm: Do not include edma headers ARM: edma: Rename header file for dmaengine filter function definition ARM: edma: Split up header file to platform_data and API file arch/arm/common/edma.c | 3 +- arch/arm/mach-davinci/devices.c | 1 + arch/arm/mach-davinci/include/mach/da8xx.h | 1 + drivers/dma/edma.c | 2 +- drivers/mmc/host/davinci_mmc.c | 3 +- drivers/spi/spi-davinci.c | 2 +- include/linux/edma-dmaengine.h | 29 ++++++ include/linux/edma.h | 152 ++++++++++++++++++++++++++--- include/linux/platform_data/edma.h | 148 ++-------------------------- sound/soc/davinci/davinci-evm.c | 3 - sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/davinci/edma-pcm.c | 2 +- 12 files changed, 182 insertions(+), 165 deletions(-) create mode 100644 include/linux/edma-dmaengine.h -- 2.1.3

7 19

[alsa-devel] [PATCH 0/3] Add master mode, tmd and right-j mode support
by Zidan Wang 23 Jan '15

23 Jan '15

Add sai master mode support. Add tdm slots support. Add Right-J mode support. Zidan Wang (3): SoC: fsl_sai: add sai master mode support ASoC: fsl_sai: Add support for tdm slots operation ASoC: fsl_sai: Add support for Right-J mode sound/soc/fsl/fsl_sai.c | 164 ++++++++++++++++++++++++++++++++++++++++++++---- sound/soc/fsl/fsl_sai.h | 13 +++- 2 files changed, 164 insertions(+), 13 deletions(-) -- 1.9.1

4 18

[alsa-devel] [PATCH v3 0/4] Sound support for Exynos4412 Trats2 board
by Inha Song 23 Jan '15

23 Jan '15

This patch-set adds basic sound support for the Trats2 boards. It just support primary I2s and external speaker playback. Changes in v3: - Set clock out parent to XUSBXTI in pmu_system_controller DT node. Changes in v2: - Remove unnecessary lines in Trats2 machine driver - move clock enable/disable to .startup / .shutdown - Fix code style issues - Remove mclk DT property in Trats2 sound and add MCLK1 property in WM1811 node. (MCLK1 is optional property in WM8994 node - refer: Document/devicetree/bindings/sound/wm8994.txt) - Use mclk that defined in WM1811 DT node. - Add "samsung,audio-codec" property to required properties. Inha Song (4): ASoC: samsung: Add machine driver for Trats2 ASoC: samsung: Document Trats2 audio subsystem bindings ARM: dts: set clock out parent to XUSBXTI ARM: dts: Add sound nodes for exynos4412-trats2 .../bindings/sound/samsung,trats2-wm1811.txt | 25 +++ arch/arm/boot/dts/exynos4.dtsi | 2 + arch/arm/boot/dts/exynos4412-trats2.dts | 37 ++++ sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/trats2_wm1811.c | 218 +++++++++++++++++++++ 6 files changed, 292 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,trats2-wm1811.txt create mode 100644 sound/soc/samsung/trats2_wm1811.c -- 2.0.0.390.gcb682f8

3 10

[alsa-devel] alsa-lib support for compress offload
by Qais Yousef 23 Jan '15

23 Jan '15

Hi, As far as I understand I need to use something similar to tinycompress to use alsa devices that supports compress offload. When I asked on gstreamer list [1] about alsasink support for compress offload they said yes. But I can see that gstalsasink.c [2] only uses the standard alsa-lib api which AFAICT doesn't support compress offload. Did I misread the code and alsa-lib actually works with compress offload? Gstreamer refers to the feature "passthrough" and associate it with SPDIF[3], are they taking advantage of some other alsa feature that looks like compress offload? Thanks, Qais [1] http://gstreamer-devel.966125.n4.nabble.com/ALSA-Compress-Offload-support-t… [2] http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/alsa/gstals… [3] http://cgit.freedesktop.org/gstreamer/gst-plugins-base/tree/ext/alsa/gstals…

6 16

[alsa-devel] [PATCH 1/1] ASoC: tas5086: Reinit register values in probe function
by pascal.huerst＠gmail.com 22 Jan '15

22 Jan '15

From: Pascal Huerst <pascal.huerst(a)gmail.com> If the machine driver has been un/reloaded, the register values of the tas5086 codec have to be reinitialized, in order to work properly. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 249ef5c..cd19171 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -851,10 +851,16 @@ static int tas5086_probe(struct snd_soc_codec *codec) } tas5086_reset(priv); + regcache_mark_dirty(priv->regmap); + ret = tas5086_init(codec->dev, priv); if (ret < 0) goto exit_disable_regulators; + ret = regcache_sync(priv->regmap); + if (ret < 0) + goto exit_disable_regulators; + /* set master volume to 0 dB */ ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); if (ret < 0) -- 1.9.3

1 0

[alsa-devel] [PATCH - UCM 1/1] ucm: add binary configure file parse
by han.lu＠intel.com 22 Jan '15

22 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> with cset command, UCM set kcontrol parameters directly: cset "name='<KCONTROL_NAME>' 1<,2,3,...>" This patch enables UCM to set kcontrol with parameters from configure file: cset-bin-file "name='<KCONTROL_NAME>' <path/to/file>" where "cset-bin-file" is a newly added keyword alongside of "cset", to indicate cset with binary data in file. The binary data in file is parameter for audio DSPs, and it's just passed by UCM/ALSA as raw data. The data type of parameter elements must be byte, and the count must matches driver definition. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 37ae4c8..4c1e4b2 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -160,11 +160,62 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int binary_file_parse(snd_ctl_elem_value_t *dst, + snd_ctl_elem_info_t *info, + const char *filepath) +{ + int err, fd; + struct stat st; + size_t sz; + char *res; + snd_ctl_elem_type_t type; + unsigned int idx, count; + + type = snd_ctl_elem_info_get_type(info); + if (type != SND_CTL_ELEM_TYPE_BYTES) { + uc_error("only support byte type!"); + err = -EINVAL; + return err; + } + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + count = snd_ctl_elem_info_get_count(info); + if (sz != count || sz > sizeof(dst->value.bytes)) { + uc_error("invalid parameter size %d!", sz); + err = -EINVAL; + goto __fail; + } + res = malloc(sz); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + if (read(fd, res, sz) != sz) { + err = -errno; + goto __fail_read; + } + for (idx = 0; idx < sz; idx++) + snd_ctl_elem_value_set_byte(dst, idx, *(res + idx)); + __fail_read: + free(res); + __fail: + close(fd); + return err; +} + extern int __snd_ctl_ascii_elem_id_parse(snd_ctl_elem_id_t *dst, const char *str, const char **ret_ptr); -static int execute_cset(snd_ctl_t *ctl, const char *cset) +static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) { const char *pos; int err; @@ -194,7 +245,10 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset) err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); if (err < 0) goto __fail; err = snd_ctl_elem_write(ctl, value); @@ -239,6 +293,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail_nomem; break; case SEQUENCE_ELEMENT_TYPE_CSET: + case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: if (cdev == NULL) { const char *cdev1 = NULL, *cdev2 = NULL; err = get_value3(&cdev1, "PlaybackCTL", @@ -274,7 +329,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail; } } - err = execute_cset(ctl, s->data.cset); + err = execute_cset(ctl, s->data.cset, s->type); if (err < 0) { uc_error("unable to execute cset '%s'\n", s->data.cset); goto __fail; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index d7517f6..9e1cb41 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -306,6 +306,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-bin-file") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-bin-file requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 87f14a2..c1655c7 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -47,6 +47,7 @@ #define SEQUENCE_ELEMENT_TYPE_CSET 2 #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 +#define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 struct ucm_value { struct list_head list; -- 2.1.0

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[alsa-devel] [PATCH v2 0/3] Sound support for Exynos4412 Trats2 board
by Inha Song 22 Jan '15

22 Jan '15

This patch-set adds basic sound support for the Trats2 boards. It just support primary I2s and external speaker playback. Changes in v2: - Remove unnecessary lines in Trats2 machine driver - move clock enable/disable to .startup / .shutdown - Fix code style issues - Remove mclk DT property in Trats2 sound and add MCLK1 property in WM1811 node. (MCLK1 is optional property in WM8994 node - refer: Document/devicetree/bindings/sound/wm8994.txt) - Use mclk that defined in WM1811 DT node. - Add "samsung,audio-codec" property to required properties. Inha Song (3): ASoC: samsung: Add machine driver for Trats2 ASoC: samsung: Document Trats2 audio subsystem bindings ARM: dts: Add sound nodes for exynos4412-trats2 .../bindings/sound/samsung,trats2-wm1811.txt | 25 +++ arch/arm/boot/dts/exynos4412-trats2.dts | 39 ++++ sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/trats2_wm1811.c | 218 +++++++++++++++++++++ 5 files changed, 292 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,trats2-wm1811.txt create mode 100644 sound/soc/samsung/trats2_wm1811.c -- 2.0.0.390.gcb682f8

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[alsa-devel] Question about codec which is supporting TDM / 5.1ch
by Kuninori Morimoto 22 Jan '15

22 Jan '15

Hi ALSA / ALSA SoC Now, we want to create new board, and then, we want to use below codec if exist. Can you show it to us if it was exist ? (I researched it on upstream Linux source tree, but couldn't find....) - upstream Linux supports its driver - it supports TDM format - it supports 5.1ch sound - it supports DT bindings (if possible) Best regards --- Kuninori Morimoto

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[alsa-devel] Hauppauge USB-Live2 recording fails on USB3 port
by Alistair Grant 22 Jan '15

22 Jan '15

Hi, I've been attempting to capture video through a Hauppauge USB-Live2 device using: mythffmpeg -y -loglevel debug -f alsa -i plughw:1,0 -f v4l2 -s pal -pix_fmt yuyv422 -i /dev/video1 -b:v 2048k a.nut This works fine from Ubuntu 14.04 on an older laptop that only has USB2, but fails on my Dell XPS13, which only has USB3 ports (Ubuntu 14.10, Kernel 3.16, 3.18, 3.19). I have included a larger extract from /var/log/syslog below, however three of the lines are: Jan 18 16:49:23 alistair-XPS13 pulseaudio[3296]: [alsa-sink-ALC3661 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Jan 18 16:49:23 alistair-XPS13 pulseaudio[3296]: [alsa-sink-ALC3661 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jan 18 16:49:23 alistair-XPS13 pulseaudio[3296]: [alsa-sink-ALC3661 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. I have already posted this on the linux-usb mailing list and received a confirmation from the author of the USB-Live2 USB driver that there are problems with USB3. So I expect that this is a USB3 xHCI problem and not an ALSA problem, but if there is a workaround I would be very interested. If you'd like more information or me to do more testing, please let me know. Thanks very much, Alistair Original report to Ubuntu Bugs: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1412121 linux-usb report: http://thread.gmane.org/gmane.linux.usb.general/121094 Additional discussion on linux-usb: http://www.spinics.net/lists/linux-usb/msg120276.html /var/log/syslog extract: Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.790754] usb 1-1: new high-speed USB device number 5 using xhci_hcd Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.921414] usb 1-1: New USB device found, idVendor=2040, idProduct=c200 Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.921422] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.921426] usb 1-1: Product: Hauppauge Device Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.921430] usb 1-1: Manufacturer: Hauppauge Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.921433] usb 1-1: SerialNumber: 0011446325 Jan 18 16:48:16 alistair-XPS13 mtp-probe: checking bus 1, device 5: "/sys/devices/pci0000:00/0000:00:14.0/usb1/1-1" Jan 18 16:48:16 alistair-XPS13 mtp-probe: bus: 1, device: 5 was not an MTP device Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.985669] cx231xx 1-1:1.1: New device Hauppauge Hauppauge Device @ 480 Mbps (2040:c200) with 6 interfaces Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.985716] cx231xx 1-1:1.1: can't change interface 3 alt no. to 3: Max. Pkt size = 0 Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.987108] cx231xx 1-1:1.1: can't change interface 4 alt no. to 1: Max. Pkt size = 0 Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.988279] cx231xx 1-1:1.1: Identified as Hauppauge USB Live 2 (card=9) Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.988572] i2c i2c-10: Added multiplexed i2c bus 12 Jan 18 16:48:16 alistair-XPS13 kernel: [ 83.988633] i2c i2c-10: Added multiplexed i2c bus 13 Jan 18 16:48:16 alistair-XPS13 kernel: [ 84.088325] cx25840 9-0044: cx23102 A/V decoder found @ 0x88 (cx231xx #0-0) Jan 18 16:48:18 alistair-XPS13 kernel: [ 86.036671] cx25840 9-0044: loaded v4l-cx231xx-avcore-01.fw firmware (16382 bytes) Jan 18 16:48:18 alistair-XPS13 kernel: [ 86.067096] cx231xx 1-1:1.1: v4l2 driver version 0.0.3 Jan 18 16:48:19 alistair-XPS13 kernel: [ 86.153723] cx231xx 1-1:1.1: Registered video device video1 [v4l2] Jan 18 16:48:19 alistair-XPS13 kernel: [ 86.153864] cx231xx 1-1:1.1: Registered VBI device vbi0 Jan 18 16:48:19 alistair-XPS13 kernel: [ 86.153869] cx231xx 1-1:1.1: video EndPoint Addr 0x84, Alternate settings: 5 Jan 18 16:48:19 alistair-XPS13 kernel: [ 86.153872] cx231xx 1-1:1.1: VBI EndPoint Addr 0x85, Alternate settings: 2 Jan 18 16:48:19 alistair-XPS13 kernel: [ 86.153875] cx231xx 1-1:1.1: sliced CC EndPoint Addr 0x86, Alternate settings: 2 Jan 18 16:48:19 alistair-XPS13 kernel: [ 86.154013] usbcore: registered new interface driver cx231xx Jan 18 16:48:19 alistair-XPS13 kernel: [ 86.176396] cx231xx 1-1:1.1: audio EndPoint Addr 0x83, Alternate settings: 3 Jan 18 16:48:19 alistair-XPS13 kernel: [ 86.176402] cx231xx 1-1:1.1: Cx231xx Audio Extension initialized Jan 18 16:48:19 alistair-XPS13 pulseaudio[3296]: [pulseaudio] source.c: Default and alternate sample rates are the same. Jan 18 16:48:19 alistair-XPS13 rtkit-daemon[2205]: Successfully made thread 3819 of process 3296 (n/a) owned by '1000' RT at priority 5. Jan 18 16:48:19 alistair-XPS13 rtkit-daemon[2205]: Supervising 5 threads of 1 processes of 1 users. Jan 18 16:49:23 alistair-XPS13 pulseaudio[3296]: [alsa-sink-ALC3661 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Jan 18 16:49:23 alistair-XPS13 pulseaudio[3296]: [alsa-sink-ALC3661 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jan 18 16:49:23 alistair-XPS13 pulseaudio[3296]: [alsa-sink-ALC3661 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.336417] xhci_hcd 0000:00:14.0: xHCI host not responding to stop endpoint command. Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.336428] xhci_hcd 0000:00:14.0: Assuming host is dying, halting host. Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.336542] usb 1-1: Not enough bandwidth for altsetting 0 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.336553] cx231xx 1-1:1.1: can't change interface 2 alt no. to 0 (err=-19) Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.336554] xhci_hcd 0000:00:14.0: HC died; cleaning up Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.336563] cx231xx 1-1:1.1: failed to set alternate setting ! Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.336669] cx231xx 1-1:1.1: cx231xx_send_usb_command: failed with status --19 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.336679] cx231xx 1-1:1.1: cx231xx_send_usb_command: failed with status --19 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.337094] usb 1-1: USB disconnect, device number 5 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.337451] cx231xx 1-1:1.1: device video1 is open! Deregistration and memory deallocation are deferred on close. Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.372776] usb 1-3: USB disconnect, device number 2 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373439] usb 1-1:1.1: V4L2 device vbi0 deregistered Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373449] ------------[ cut here ]------------ Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373462] WARNING: CPU: 3 PID: 3992 at /home/apw/COD/linux/fs/sysfs/group.c:219 sysfs_remove_group+0xa3/0xb0() Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373466] sysfs group ffffffff81cb60a0 not found for kobject 'vbi0' Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373469] Modules linked in: cx231xx_alsa cx25840 cx231xx videobuf_vmalloc tveeprom cx2341x rc_core videobuf_core i2c_mux pci_stub vboxpci(OE) vboxnetadp(OE) vboxnetflt(OE) vboxdrv(OE) xt_CHECKSUM iptable_mangle ipt_MASQUERADE nf_nat_masquerade_ipv4 iptable_nat nf_conntrack_ipv4 nf_defrag_ipv4 nf_nat_ipv4 nf_nat nf_conntrack xt_tcpudp bridge stp llc iptable_filter ip_tables x_tables ctr ccm dm_crypt rpcsec_gss_krb5 nfsv4 joydev arc4 snd_soc_rt5640 hid_rmi btusb dell_wmi snd_soc_rl6231 sparse_keymap intel_rapl iosf_mbi snd_soc_core uvcvideo videobuf2_vmalloc x86_pkg_temp_thermal videobuf2_memops snd_compress videobuf2_core v4l2_common snd_pcm_dmaengine hid_multitouch intel_powerclamp videodev snd_seq_midi snd_seq_midi_event media snd_rawmidi kvm_intel kvm snd_seq iwlmvm crct10dif_pclmul crc32_pclmul mac80211 dell_laptop dcdbas ghash_clmulni_intel aesni_intel aes_x86_64 snd_hda_codec_hdmi lrw snd_hda_codec_realtek snd_hda_codec_generic gf128mul rfcomm glue_helper iwlwifi bnep ablk_helper cryptd snd_hda_intel snd_seq_device snd_hda_controller bluetooth snd_hda_codec serio_raw snd_hwdep cfg80211 snd_pcm mei_me mei snd_timer lpc_ich shpchp snd dw_dmac snd_soc_sst_acpi nfsd soundcore i2c_hid auth_rpcgss nfs_acl i2c_designware_platform 8250_dw dw_dmac_core i2c_designware_core spi_pxa2xx_platform nfs intel_rst intel_smartconnect lockd grace sunrpc fscache mac_hid parport_pc ppdev coretemp lp parport usbhid hid i915 i2c_algo_bit drm_kms_helper psmouse ahci libahci drm wmi sdhci_acpi video sdhci Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373655] CPU: 3 PID: 3992 Comm: mythffmpeg Tainted: G OE 3.19.0-031900rc5-generic #201501180935 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373659] Hardware name: Dell Inc. XPS13 9333/099TN4, BIOS A01 11/11/2013 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373662] 00000000000000db ffff8800ae8a3c68 ffffffff817c5098 0000000000000007 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373669] ffff8800ae8a3cb8 ffff8800ae8a3ca8 ffffffff81076dd7 ffff8800ae8a3c98 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373675] ffffffff81cb60a0 ffff8800be448898 0000000000000000 ffff8801f0f8b398 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373682] Call Trace: Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373694] [<ffffffff817c5098>] dump_stack+0x45/0x57 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373703] [<ffffffff81076dd7>] warn_slowpath_common+0x97/0xe0 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373710] [<ffffffff81076ed6>] warn_slowpath_fmt+0x46/0x50 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373718] [<ffffffff8126ab93>] ? kernfs_find_and_get_ns+0x53/0x70 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373725] [<ffffffff8126ef43>] sysfs_remove_group+0xa3/0xb0 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373731] [<ffffffff814fe409>] dpm_sysfs_remove+0x59/0x70 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373740] [<ffffffff814f2166>] device_del+0x56/0x230 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373747] [<ffffffff814f2362>] device_unregister+0x22/0x60 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373765] [<ffffffffc0a2825d>] video_unregister_device+0x4d/0x60 [videodev] Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373777] [<ffffffffc0d53f55>] cx231xx_release_analog_resources+0x125/0x170 [cx231xx] Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373787] [<ffffffffc0d56666>] cx231xx_release_resources+0x16/0x60 [cx231xx] Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373797] [<ffffffffc0d52b86>] cx231xx_close.isra.11+0x1d6/0x2a0 [cx231xx] Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373805] [<ffffffffc0d52c8f>] cx231xx_v4l2_close+0x3f/0x60 [cx231xx] Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373818] [<ffffffffc0a27491>] v4l2_release+0x41/0x90 [videodev] Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373825] [<ffffffff811f7e30>] __fput+0xc0/0x250 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373832] [<ffffffff811f800e>] ____fput+0xe/0x10 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373838] [<ffffffff81093f3f>] task_work_run+0xaf/0xf0 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373847] [<ffffffff81015077>] do_notify_resume+0xc7/0xd0 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373853] [<ffffffff817d264f>] int_signal+0x12/0x17 Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.373858] ---[ end trace cfaf8b5d940102a2 ]--- Jan 18 16:50:14 alistair-XPS13 kernel: [ 201.374207] usb 1-1:1.1: V4L2 device video1 deregistered More errors follow.

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[alsa-devel] [PATCH - UCM 1/1] ucm: add binary configure file parse
by han.lu＠intel.com 21 Jan '15

21 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> with cset command, UCM set kcontrol parameters directly: cset "name='<KCONTROL_NAME>' 1<,2,3,...>" This patch enables UCM to set kcontrol with parameters from configure file: cset-bin-file "name='<KCONTROL_NAME>' <path/to/file>" where "cset-bin-file" is a newly added keyword alongside of "cset", to indicate cset with binary data in file. The binary data in file is parameter for audio DSPs, and it's just passed by UCM/ALSA as raw data. The data type of the parameter element must be byte, and the count is limited by the size of struct snd_ctl_elem_value and driver definition. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 37ae4c8..818465a 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -160,11 +160,62 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int binary_file_parse(snd_ctl_elem_value_t *dst, + snd_ctl_elem_info_t *info, + const char *filepath) +{ + int err, fd; + struct stat st; + size_t sz; + char *res; + snd_ctl_elem_type_t type; + unsigned int idx, count; + + type = snd_ctl_elem_info_get_type(info); + if (type != SND_CTL_ELEM_TYPE_BYTES) { + uc_error("only support byte type!"); + err = -EINVAL; + return err; + } + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + count = snd_ctl_elem_info_get_count(info); + if (sz == 0 || sz > count || sz > sizeof(dst->value.bytes)) { + uc_error("invalid parameter size!"); + err = -EINVAL; + goto __fail; + } + res = calloc(sz, 1); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + if (read(fd, res, sz) != sz) { + err = -errno; + goto __fail_read; + } + for (idx = 0; idx < sz; idx++) + snd_ctl_elem_value_set_byte(dst, idx, *(res + idx)); + __fail_read: + free(res); + __fail: + close(fd); + return err; +} + extern int __snd_ctl_ascii_elem_id_parse(snd_ctl_elem_id_t *dst, const char *str, const char **ret_ptr); -static int execute_cset(snd_ctl_t *ctl, const char *cset) +static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) { const char *pos; int err; @@ -194,7 +245,10 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset) err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); if (err < 0) goto __fail; err = snd_ctl_elem_write(ctl, value); @@ -239,6 +293,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail_nomem; break; case SEQUENCE_ELEMENT_TYPE_CSET: + case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: if (cdev == NULL) { const char *cdev1 = NULL, *cdev2 = NULL; err = get_value3(&cdev1, "PlaybackCTL", @@ -274,7 +329,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail; } } - err = execute_cset(ctl, s->data.cset); + err = execute_cset(ctl, s->data.cset, s->type); if (err < 0) { uc_error("unable to execute cset '%s'\n", s->data.cset); goto __fail; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index d7517f6..9e1cb41 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -306,6 +306,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-bin-file") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-bin-file requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 87f14a2..c1655c7 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -47,6 +47,7 @@ #define SEQUENCE_ELEMENT_TYPE_CSET 2 #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 +#define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 struct ucm_value { struct list_head list; -- 2.1.0

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[alsa-devel] [PATCH] ALSA: fireworks/bebob: fix refering to released resources at unplugging
by Takashi Sakamoto 21 Jan '15

21 Jan '15

Unplugging devices can cause disorder by releasing streaming resources before stopping playbacking/capturing substreams. This commit forces the state of ALSA character devices into disconnect before releasing the resources. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 3 ++- sound/firewire/fireworks/fireworks.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index fc19c99..41eb1ce 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -306,10 +306,11 @@ static void bebob_remove(struct fw_unit *unit) if (bebob == NULL) return; + snd_card_disconnect(bebob->card); + kfree(bebob->maudio_special_quirk); snd_bebob_stream_destroy_duplex(bebob); - snd_card_disconnect(bebob->card); snd_card_free_when_closed(bebob->card); } diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 3e2ed8e..21fb50f 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -289,10 +289,11 @@ static void efw_remove(struct fw_unit *unit) { struct snd_efw *efw = dev_get_drvdata(&unit->device); + snd_card_disconnect(efw->card); + snd_efw_stream_destroy_duplex(efw); snd_efw_transaction_remove_instance(efw); - snd_card_disconnect(efw->card); snd_card_free_when_closed(efw->card); } -- 2.1.0

2 2

[alsa-devel] arecord: strange --duration behaviour with some sample rates
by Daniel Drake 20 Jan '15

20 Jan '15

Hi, With alsa-utils-1.0.28, we're seeing some strange behaviour when running: arecord -t wav -r 44100 -d 1 out.wav After recording 1 second of data to out.wav, a load of extra files are created like out-01.wav, out-02.wav, out-03.wav, etc. These extra files are 44 bytes in length. arecord seems to continue like this forever rather than exiting after the 1 second recording. The problem is somehow linked to the sample-rate. At 48000 or 8000, things work fine, just 44100 is bad. Tracing some values in the source: count is 44100, and rest is initialized to the same. In each iteration of the /* capture */ loop, f has value 5512, i.e. 5512 bytes are read. So rest decreases through values 38588, 33076, 27564, 22052, 16540, 11028, 5516. Now we read another 5512 bytes, rest is decreased to 4. So we now call pcm_read(audiobuf, f) where f has value 4. pcm_read adjusts that to 5512 and returns 5512. Now we hit: if (pcm_read(audiobuf, f) != f) break; We break out of that inner loop, count still has value 4 so we reach the end of the outer loop, and loop again, which causes a new file to be opened, etc. Sorry that I'm not quite up-to-scratch with pcm basics to suggest a quick fix, but hopefully this diagnosis is useful at least. Thanks, Daniel -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 1

[alsa-devel] [PATCH] ALSA: hda - Enable docking station for an HP machine
by David Henningsson 20 Jan '15

20 Jan '15

From: TienFu Chen <tienfu.chen(a)canonical.com> On this machine we need to inform the driver where the docking station pins are, because it has not been set up by BIOS. Tested-by: TienFu Chen <tienfu.chen(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1412800 Signed-off-by: TienFu Chen <tienfu.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1720f8a..0403061 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4417,6 +4417,7 @@ enum { ALC280_FIXUP_HP_GPIO4, ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, + ALC280_FIXUP_HP_DOCK_PINS, }; static const struct hda_fixup alc269_fixups[] = { @@ -4894,6 +4895,17 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_gpio2_mic_hotkey, }, + [ALC280_FIXUP_HP_DOCK_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x21011020 }, /* line-out */ + { 0x1a, 0x01a1903c }, /* headset mic */ + { 0x18, 0x2181103f }, /* line-in */ + { }, + }, + .chained = true, + .chain_id = ALC280_FIXUP_HP_GPIO4 + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4937,6 +4949,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS), SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.9.1

2 1

[alsa-devel] [PATCH] fm801: move to pcim_* and devm_* functions
by Andy Shevchenko 20 Jan '15

20 Jan '15

The managed functions allow to get ->probe() and ->remove() simplier. This patch converts code to use managed functions. While here remove the dead code and fix the value printed in error message. Signed-off-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> --- sound/pci/fm801.c | 29 +++++++---------------------- 1 file changed, 7 insertions(+), 22 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index dcda3c1..2708500 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1178,12 +1178,8 @@ static int snd_fm801_free(struct fm801 *chip) v4l2_device_unregister(&chip->v4l2_dev); } #endif - if (chip->irq >= 0) - free_irq(chip->irq, chip); pci_release_regions(chip->pci); - pci_disable_device(chip->pci); - kfree(chip); return 0; } @@ -1206,28 +1202,23 @@ static int snd_fm801_create(struct snd_card *card, }; *rchip = NULL; - if ((err = pci_enable_device(pci)) < 0) + if ((err = pcim_enable_device(pci)) < 0) return err; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) { - pci_disable_device(pci); + chip = devm_kzalloc(&pci->dev, sizeof(*chip), GFP_KERNEL); + if (chip == NULL) return -ENOMEM; - } spin_lock_init(&chip->reg_lock); chip->card = card; chip->pci = pci; chip->irq = -1; chip->tea575x_tuner = tea575x_tuner; - if ((err = pci_request_regions(pci, "FM801")) < 0) { - kfree(chip); - pci_disable_device(pci); + if ((err = pci_request_regions(pci, "FM801")) < 0) return err; - } chip->port = pci_resource_start(pci, 0); if ((tea575x_tuner & TUNER_ONLY) == 0) { - if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED, - KBUILD_MODNAME, chip)) { - dev_err(card->dev, "unable to grab IRQ %d\n", chip->irq); + if (devm_request_irq(&pci->dev, pci->irq, snd_fm801_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); snd_fm801_free(chip); return -EBUSY; } @@ -1242,12 +1233,6 @@ static int snd_fm801_create(struct snd_card *card, /* init might set tuner access method */ tea575x_tuner = chip->tea575x_tuner; - if (chip->irq >= 0 && (tea575x_tuner & TUNER_ONLY)) { - pci_clear_master(pci); - free_irq(chip->irq, chip); - chip->irq = -1; - } - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_fm801_free(chip); return err; -- 1.8.3.101.g727a46b

2 17

[alsa-devel] [RESEND PATCH] mfd: wm8994: set mfd id-base for regulator devs creation to avoid conflicts
by Inha Song 20 Jan '15

20 Jan '15

After commit: 6e3f62f0793e ("mfd: core: Fix platform-device id generation") We must set the id base when register a duplicate name of mfd_cell. If not, duplicate filename error was reported. - sysfs: cannot create duplicate filename '/devices/.../wm8994-ldo' Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index e6fab94..47cd3cc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -36,12 +36,12 @@ static const struct mfd_cell wm8994_regulator_devs[] = { { .name = "wm8994-ldo", - .id = 1, + .id = 0, .pm_runtime_no_callbacks = true, }, { .name = "wm8994-ldo", - .id = 2, + .id = 1, .pm_runtime_no_callbacks = true, }, }; @@ -344,7 +344,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) dev_set_drvdata(wm8994->dev, wm8994); /* Add the on-chip regulators first for bootstrapping */ - ret = mfd_add_devices(wm8994->dev, -1, + ret = mfd_add_devices(wm8994->dev, 0, wm8994_regulator_devs, ARRAY_SIZE(wm8994_regulator_devs), NULL, 0, NULL); -- 2.0.0.390.gcb682f8

3 2

[alsa-devel] [PATCH 0/7] More line6 fixes / rewrites
by Takashi Iwai 20 Jan '15

20 Jan '15

Hi, this is yet another series of patches for line6 drivers. Most of patches are cleanups and trivial fixes while the last one is to reimplement the LED control via the standard LED class. As usual, patches are found in test/line6 branch for now. Takashi

1 7

[alsa-devel] [PATCH 0/3] Minor line6 cleanup
by Chris Rorvick 20 Jan '15

20 Jan '15

These apply on top of test/line6. Chris Rorvick (3): ALSA: line6: Remove superfluous NULL checks ALSA: line6: Refer to manufacturer as "Line 6" ALSA: line6: Remove driver version from header comment sound/usb/line6/Kconfig | 4 ++-- sound/usb/line6/capture.c | 2 +- sound/usb/line6/capture.h | 2 +- sound/usb/line6/driver.c | 36 +++++++++++++----------------------- sound/usb/line6/driver.h | 14 +++++++------- sound/usb/line6/midi.c | 6 +++--- sound/usb/line6/midi.h | 4 ++-- sound/usb/line6/midibuf.c | 4 ++-- sound/usb/line6/midibuf.h | 2 +- sound/usb/line6/pcm.c | 2 +- sound/usb/line6/pcm.h | 10 +++++----- sound/usb/line6/playback.c | 2 +- sound/usb/line6/playback.h | 2 +- sound/usb/line6/pod.c | 6 +++--- sound/usb/line6/podhd.c | 6 +++--- sound/usb/line6/toneport.c | 4 ++-- sound/usb/line6/usbdefs.h | 2 +- sound/usb/line6/variax.c | 4 ++-- 18 files changed, 51 insertions(+), 61 deletions(-) -- 2.1.0

2 4

[alsa-devel] [PATCH v2] ASoC: OMAP: mcbsp: ensure that CLKX and CLKR are not used as ouput pins when they are used as input clock for the SRG.
by Thomas Niederprüm 20 Jan '15

20 Jan '15

This patch fixes faulty behaviour in a setup where the input clock for the SRG is fed through the CLKR/CLKX pin but the McBSP is configured to be master (SND_SOC_DAIFMT_CBS_CFS). In that case of course CLKR/CLKX must not be configured as output pin. Otherwise the input clock is messed up horribly. This patch makes it possible to use the CLKR/CLKX pin rather than CLKS to inject a reference clock in setups where McBSP is master and not both rx and tx are used. However for this to work it has to be ensured that set_dai_sysclk() is called after set_dai_fmt(). This was tested on a beagleboard-xm using McBSP1 to drive a i2s DAC through the tx lines (CLKX,FSX,DX). Using this patch the CLKR pin is used to inject an external reference clock. changes since v1: - added comments explaining the bit masking to disable output on CLKR/CLKX Signed-off-by: Thomas Niederprüm <niederp(a)physik.uni-kl.de> --- sound/soc/omap/omap-mcbsp.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index bd3ef2a..c37f606 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_SYSCLK_CLKX_EXT: regs->srgr2 |= CLKSM; + regs->pcr0 |= SCLKME; + /* + * If McBSP is master but yet the CLKX/CLKR pin drives the SRG, + * disable output on those pins. This enables to inject the + * reference clock through CLKX/CLKR. For this to work + * set_dai_sysclk() _needs_ to be called after set_dai_fmt(). + */ + regs->pcr0 &= ~CLKXM; + break; case OMAP_MCBSP_SYSCLK_CLKR_EXT: regs->pcr0 |= SCLKME; + /* Disable ouput on CLKR pin in master mode */ + regs->pcr0 &= ~CLKRM; break; default: err = -ENODEV; -- 2.1.1

2 1

[alsa-devel] [PATCH 00/17] More line6 driver code reorg and cleanups
by Takashi Iwai 20 Jan '15

20 Jan '15

Hi, here is a series of patches I hacked quickly after merging line6 stuff into sound git tree. There are two significant changes: - Split to each driver and renames: now there are 4 card modules, snd-usb-pod, snd-usb-podhd, snd-usb-toneport and snd-usb-variax, while snd-usb-line6 serves as a common helper module. - Many codes restrucruring, especially cleanup of the resource management; the destructor is solely called from card's private_free now The current merged code is found in topic/line6 branch of sound.git tree while this development code is found in test/line6 branch. The devel branch hasn't been merged to linux-next yet, I'm waiting for any good/bad test results. So, give me any feedback. Thanks! (BTW, the next step would be to straighten the init code in pod.c, etc instead of (ab)using the system timer. Instead, it should be a flat function with wait_event() of such.) Takashi

2 24

[alsa-devel] [PATCH 1/2] ASoC: atmel_ssc_dai: fix start event for I2S mode
by Bo Shen 20 Jan '15

20 Jan '15

According to the I2S specification information as following: - WS = 0, channel 1 (left) - WS = 1, channel 2 (right) So, the start event should be TF/RF falling edge. Reported-by: Songjun Wu <songjun.wu(a)atmel.com> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 99ff35e..e691aab 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -348,7 +348,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; - int start_event; int ret; int fslen, fslen_ext; @@ -457,19 +456,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * The SSC transmit clock is obtained from the BCLK signal on * on the TK line, and the SSC receive clock is * generated from the transmit clock. - * - * For single channel data, one sample is transferred - * on the falling edge of the LRC clock. - * For two channel data, one sample is - * transferred on both edges of the LRC clock. */ - start_event = ((channels == 1) - ? SSC_START_FALLING_RF - : SSC_START_EDGE_RF); - rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) - | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? @@ -478,14 +468,14 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(RFMR_FSLEN, 0) - | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) | SSC_BF(RFMR_DATLEN, (bits - 1)); tcmr = SSC_BF(TCMR_PERIOD, 0) | SSC_BF(TCMR_STTDLY, START_DELAY) - | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? @@ -495,7 +485,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(TFMR_FSLEN, 0) - | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) | SSC_BF(TFMR_DATLEN, (bits - 1)); -- 2.3.0.rc0

2 3

[alsa-devel] [PATCH 1/3] ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format
by Lars-Peter Clausen 20 Jan '15

20 Jan '15

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> --- 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] wm97xx touch does not work after ac97 delayed dev reg removal
by Manuel Lauss 19 Jan '15

19 Jan '15

Hi Lars, Since commit 6794f709b7124ff1e574c4f4c9494418ab56c4b4 (ASoC: ac97: Drop delayed device registration) the wm9712 ts on my MIPS DB1300 board doesn't work any more: wm97xx-ts 0-0:wm9712-codec.1: Device with vendor 0000 is not a wm97xx db1200-ac97 db1300-ac97.0: wm9712-hifi <-> au1xpsc_ac97.1 mapping ok wm9712-codec wm9712-codec.1: ASoC: mux Differential Source has no paths wm9712-codec wm9712-codec.1: ASoC: mux Capture Phone Mux has no paths db1200-ac97 db1300-i2s.0: wm8731-hifi <-> au1xpsc_i2s.2 mapping ok whereas before this patch: [ 1.820000] db1200-ac97 db1300-ac97.0: wm9712-hifi <-> au1xpsc_ac97.1 mapping ok [ 1.820000] wm9712-codec wm9712-codec.1: ASoC: mux Differential Source has no paths [ 1.830000] wm9712-codec wm9712-codec.1: ASoC: mux Capture Phone Mux has no paths [ 1.840000] wm97xx-ts 0-0:wm9712-codec.1: detected a wm9712 codec [ 1.850000] input: wm97xx touchscreen as /devices/platform/db1300-ac97.0/0-0:wm9712-codec.1/input/input0 [ 1.860000] db1200-ac97 db1300-i2s.0: wm8731-hifi <-> au1xpsc_i2s.2 mapping ok Do you have any ideas how to solve this? Thanks! Manuel

3 4

Re: [alsa-devel] [PATCH] ASoC: arizona: Add delay for output disable
by Mark Brown 19 Jan '15

19 Jan '15

On Mon, Jan 19, 2015 at 03:49:46PM +0000, Charles Keepax wrote: > + case SND_SOC_DAPM_POST_PMD: > + switch (w->shift) { > + case ARIZONA_OUT1L_ENA_SHIFT: > + case ARIZONA_OUT1R_ENA_SHIFT: > + case ARIZONA_OUT2L_ENA_SHIFT: > + case ARIZONA_OUT2R_ENA_SHIFT: > + case ARIZONA_OUT3L_ENA_SHIFT: > + case ARIZONA_OUT3R_ENA_SHIFT: > + udelay(750); > + break; That's a really quite long udelay() especially given that as the code is written it's going to be cumalative over all the outputs being torn down so typically will be 1.5ms for headphones (not sure if that's the intention or not). msleep() would be more friendly than udelay() and it'd also be good to arrange to coalesce the delays. Perhaps set a flag in _PRE_PMD and check if a delay is needed in the clock teardown? Or better yet record a timestamp and then figure out if a delay is needed when tearing down the clock though that is probably overengineering.

2 3

[alsa-devel] [PATCH v2 1/1] alsa-lib: Add snd_pcm_start_at.
by Tim Cussins 19 Jan '15

19 Jan '15

Add notion of TSTAMP_CLASS (SYSTEM/AUDIO) as per Pierre's suggestion. Add snd_pcm_start_at() Signed-off-by: Tim Cussins <timcussins(a)eml.cc> diff --git a/include/pcm.h b/include/pcm.h index 0655e7f..c57edca 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -323,13 +323,25 @@ typedef enum _snd_pcm_tstamp { SND_PCM_TSTAMP_LAST = SND_PCM_TSTAMP_ENABLE } snd_pcm_tstamp_t; +typedef enum _snd_pcm_tstamp_class { + SND_PCM_TSTAMP_CLASS_SYSTEM = 0, + SND_PCM_TSTAMP_CLASS_AUDIO, + SND_PCM_TSTAMP_CLASS_LAST = SND_PCM_TSTAMP_CLASS_AUDIO, +} snd_pcm_tstamp_class_t; + typedef enum _snd_pcm_tstamp_type { SND_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /** gettimeofday equivalent */ - SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ + SND_PCM_TSTAMP_TYPE_MONOTONIC, /** posix_clock_monotonic equivalent */ SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /** monotonic_raw (no NTP) */ SND_PCM_TSTAMP_TYPE_LAST = SND_PCM_TSTAMP_TYPE_MONOTONIC_RAW, } snd_pcm_tstamp_type_t; +typedef enum _snd_pcm_audio_tstamp_type { + SND_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 0, + SND_PCM_AUDIO_TSTAMP_TYPE_LINK, + SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SND_PCM_AUDIO_TSTAMP_TYPE_LINK, +} snd_pcm_audio_tstamp_t; + /** Unsigned frames quantity */ typedef unsigned long snd_pcm_uframes_t; /** Signed frames quantity */ @@ -478,6 +490,7 @@ int snd_pcm_prepare(snd_pcm_t *pcm); int snd_pcm_reset(snd_pcm_t *pcm); int snd_pcm_status(snd_pcm_t *pcm, snd_pcm_status_t *status); int snd_pcm_start(snd_pcm_t *pcm); +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t* start_time); int snd_pcm_drop(snd_pcm_t *pcm); int snd_pcm_drain(snd_pcm_t *pcm); int snd_pcm_pause(snd_pcm_t *pcm, int enable); diff --git a/include/sound/asound.h b/include/sound/asound.h index 1f23cd6..1592160 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -466,12 +466,30 @@ struct snd_xfern { }; enum { + SNDRV_PCM_TSTAMP_CLASS_SYSTEM = 0, + SNDRV_PCM_TSTAMP_CLASS_AUDIO, + SNDRV_PCM_TSTAMP_CLASS_LAST = SNDRV_PCM_TSTAMP_CLASS_AUDIO, +}; + +enum { SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY = 0, /* gettimeofday equivalent */ SNDRV_PCM_TSTAMP_TYPE_MONOTONIC, /* posix_clock_monotonic equivalent */ - SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ + SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, /* monotonic_raw (no NTP) */ SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW, }; +enum { + SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT = 0, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, + SNDRV_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK, +}; + +struct snd_start_at { + int tstamp_class; + int tstamp_type; + struct timespec start_time; +}; + /* channel positions */ enum { SNDRV_CHMAP_UNKNOWN = 0, @@ -550,6 +568,8 @@ enum { #define SNDRV_PCM_IOCTL_READN_FRAMES _IOR('A', 0x53, struct snd_xfern) #define SNDRV_PCM_IOCTL_LINK _IOW('A', 0x60, int) #define SNDRV_PCM_IOCTL_UNLINK _IO('A', 0x61) +#define SNDRV_PCM_IOCTL_START_AT _IOW('A', 0x62, struct snd_start_at) + /***************************************************************************** * * diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index baa47c7..40a4689 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1085,6 +1085,37 @@ int snd_pcm_start(snd_pcm_t *pcm) } /** + * \brief Start a PCM at a specified point in the future + * \param pcm PCM handle + * \param tstamp_class specifies the class of tstamp_type + * \param tstamp_type specifies the clock with which to interpret \p start_time + * \param start_time Absolute time at which to start the stream + * \return 0 on success otherwise a negative error code + * \retval -ENOSYS operation not supported for the current timestamp type + * \retval -EINVAL timespec, tstamp_class or tstamp_type is invalid + * \retval -ETIME requested start_time cannot be satisfied + * + * This method is non-blocking: It establishes an appropriate timer in the kernel + * that will start the stream on expiry. + * + * The timer is unconditionally cancelled upon any \a attempt to change the stream + * state e.g. drop, drain, start, start_at. + */ +int snd_pcm_start_at(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t *start_time) +{ + assert(pcm); + assert(start_time); + if (CHECK_SANITY(! pcm->setup)) { + SNDMSG("PCM not set up"); + return -EIO; + } + if (pcm->fast_ops->start_at) { + return pcm->fast_ops->start_at(pcm->fast_op_arg, tstamp_class, tstamp_type, start_time); + } + return -EINVAL; +} + +/** * \brief Stop a PCM dropping pending frames * \param pcm PCM handle * \return 0 on success otherwise a negative error code diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index c34b766..0190787 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -620,6 +620,26 @@ static int snd_pcm_hw_start(snd_pcm_t *pcm) return 0; } +static int snd_pcm_hw_start_at(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t *start_time) +{ + snd_pcm_hw_t *hw = pcm->private_data; + int err; + + struct snd_start_at start_at = { + .tstamp_class = tstamp_class, + .tstamp_type = tstamp_type, + .start_time = *start_time + }; + + sync_ptr(hw, 0); + if (ioctl(hw->fd, SNDRV_PCM_IOCTL_START_AT, &start_at) < 0) { + err = -errno; + SYSMSG("SNDRV_PCM_IOCTL_START_AT failed (%i)", err); + return err; + } + return 0; +} + static int snd_pcm_hw_drop(snd_pcm_t *pcm) { snd_pcm_hw_t *hw = pcm->private_data; @@ -1336,6 +1356,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops = { .prepare = snd_pcm_hw_prepare, .reset = snd_pcm_hw_reset, .start = snd_pcm_hw_start, + .start_at = snd_pcm_hw_start_at, .drop = snd_pcm_hw_drop, .drain = snd_pcm_hw_drain, .pause = snd_pcm_hw_pause, diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index 394505f..5cdfd3a 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -154,6 +154,7 @@ typedef struct { int (*prepare)(snd_pcm_t *pcm); int (*reset)(snd_pcm_t *pcm); int (*start)(snd_pcm_t *pcm); + int (*start_at)(snd_pcm_t *pcm, snd_pcm_tstamp_class_t tstamp_class, int tstamp_type, const snd_htimestamp_t *start_time); int (*drop)(snd_pcm_t *pcm); int (*drain)(snd_pcm_t *pcm); int (*pause)(snd_pcm_t *pcm, int enable); -- 1.7.10.4

3 10

[alsa-devel] [PATCH 00/18] Use setup_timer() and mod_timer()
by Takashi Iwai 19 Jan '15

19 Jan '15

Hi, this is a series of simple cleanup patches to replace the open codes with the standard helpers, setup_timer() and mod_timer(). Takashi

1 18

Re: [alsa-devel] [PATCH v2 2/2] sound: dummy, avoid race with timer
by Takashi Iwai 19 Jan '15

19 Jan '15

At Mon, 19 Jan 2015 10:56:41 +0100, Jiri Slaby wrote: > > There is a race between timer and process contexts. Process context > does not disable irqs, so when a timer ticks inside process' critical > section, the system can deadlock. Fix this by a traditional _irqsave > variant of spin_lock. > > [v2] removed ->pointer change as it is called with BH off Sorry, my previous comment was incorrect. The start function here is also called from PCM trigger callback that is already irq-disabled. So, this is also superfluous. thanks, Takashi > > Signed-off-by: Jiri Slaby <jslaby(a)suse.cz> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org > --- > sound/drivers/dummy.c | 7 +++++-- > 1 file changed, 5 insertions(+), 2 deletions(-) > > diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c > index d11baaf0f0b4..0791210cb4e7 100644 > --- a/sound/drivers/dummy.c > +++ b/sound/drivers/dummy.c > @@ -271,10 +271,13 @@ static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) > static int dummy_systimer_start(struct snd_pcm_substream *substream) > { > struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; > - spin_lock(&dpcm->lock); > + unsigned long flags; > + > + spin_lock_irqsave(&dpcm->lock, flags); > dpcm->base_time = jiffies; > dummy_systimer_rearm(dpcm); > - spin_unlock(&dpcm->lock); > + spin_unlock_irqrestore(&dpcm->lock, flags); > + > return 0; > } > > -- > 2.2.1 >

1 0

Re: [alsa-devel] [PATCH 2/3] sound: dummy, use del_timer_sync
by Takashi Iwai 19 Jan '15

19 Jan '15

At Mon, 19 Jan 2015 10:42:55 +0100, Jiri Slaby wrote: > > Wait for the timer to really quit, not only by ensuring it is not in > the critical section. > > BTW the spin_lock in here does not disable BH, so that it could > deadlock. The stop function is called from the PCM trigger callback which is irq-disabled. So, you can't use del_timer_sync() here. thanks, Takashi > > Signed-off-by: Jiri Slaby <jslaby(a)suse.cz> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org > --- > sound/drivers/dummy.c | 6 +++--- > 1 file changed, 3 insertions(+), 3 deletions(-) > > diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c > index d11baaf0f0b4..6d6bf4093583 100644 > --- a/sound/drivers/dummy.c > +++ b/sound/drivers/dummy.c > @@ -281,9 +281,9 @@ static int dummy_systimer_start(struct snd_pcm_substream *substream) > static int dummy_systimer_stop(struct snd_pcm_substream *substream) > { > struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; > - spin_lock(&dpcm->lock); > - del_timer(&dpcm->timer); > - spin_unlock(&dpcm->lock); > + > + del_timer_sync(&dpcm->timer); > + > return 0; > } > > -- > 2.2.1 >

2 1

Re: [alsa-devel] [PATCH 3/3] sound: dummy, avoid races with timer
by Takashi Iwai 19 Jan '15

19 Jan '15

At Mon, 19 Jan 2015 10:42:56 +0100, Jiri Slaby wrote: > > There is a race between timer and process contexts. Process context > does not disable irqs, so when a timer ticks inside process' critical > section, the system can deadlock. Fix this by a traditional _irqsave > variant of spin_lock. > > Signed-off-by: Jiri Slaby <jslaby(a)suse.cz> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org > --- > sound/drivers/dummy.c | 13 +++++++++---- > 1 file changed, 9 insertions(+), 4 deletions(-) > > diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c > index 6d6bf4093583..18a7cfc99078 100644 > --- a/sound/drivers/dummy.c > +++ b/sound/drivers/dummy.c > @@ -271,10 +271,13 @@ static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) > static int dummy_systimer_start(struct snd_pcm_substream *substream) > { > struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; > - spin_lock(&dpcm->lock); > + unsigned long flags; > + > + spin_lock_irqsave(&dpcm->lock, flags); > dpcm->base_time = jiffies; > dummy_systimer_rearm(dpcm); > - spin_unlock(&dpcm->lock); > + spin_unlock_irqrestore(&dpcm->lock, flags); > + This looks correct, but... > return 0; > } > > @@ -322,12 +325,14 @@ static snd_pcm_uframes_t > dummy_systimer_pointer(struct snd_pcm_substream *substream) > { > struct dummy_systimer_pcm *dpcm = substream->runtime->private_data; > + unsigned long flags; > snd_pcm_uframes_t pos; > > - spin_lock(&dpcm->lock); > + spin_lock_irqsave(&dpcm->lock, flags); > dummy_systimer_update(dpcm); > pos = dpcm->frac_pos / HZ; > - spin_unlock(&dpcm->lock); > + spin_unlock_irqrestore(&dpcm->lock, flags); > + This chunk is superfluous. The pointer callback is guaranteed to be called in the irq-disabled context. thanks, Takashi

2 1

Re: [alsa-devel] [PATCH 1/3] sound: dummy, use setup_timer and mod_timer
by Takashi Iwai 19 Jan '15

19 Jan '15

At Mon, 19 Jan 2015 10:42:54 +0100, Jiri Slaby wrote: > > From: Roman Kollar <rkollar(a)mail.muni.cz> > > Use setup_timer and mod_timer instead of structure assignments as it > is the preferred way to setup and set the timer. > > Signed-off-by: Roman Kollar <rkollar(a)mail.muni.cz> > Signed-off-by: Jiri Slaby <jslaby(a)suse.cz> > Cc: Jaroslav Kysela <perex(a)perex.cz> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org Applied this one, thanks. Takashi > --- > sound/drivers/dummy.c | 10 ++++------ > 1 file changed, 4 insertions(+), 6 deletions(-) > > diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c > index 5d0dfb787cec..d11baaf0f0b4 100644 > --- a/sound/drivers/dummy.c > +++ b/sound/drivers/dummy.c > @@ -245,9 +245,8 @@ struct dummy_systimer_pcm { > > static void dummy_systimer_rearm(struct dummy_systimer_pcm *dpcm) > { > - dpcm->timer.expires = jiffies + > - (dpcm->frac_period_rest + dpcm->rate - 1) / dpcm->rate; > - add_timer(&dpcm->timer); > + mod_timer(&dpcm->timer, jiffies + > + (dpcm->frac_period_rest + dpcm->rate - 1) / dpcm->rate); > } > > static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) > @@ -340,9 +339,8 @@ static int dummy_systimer_create(struct snd_pcm_substream *substream) > if (!dpcm) > return -ENOMEM; > substream->runtime->private_data = dpcm; > - init_timer(&dpcm->timer); > - dpcm->timer.data = (unsigned long) dpcm; > - dpcm->timer.function = dummy_systimer_callback; > + setup_timer(&dpcm->timer, dummy_systimer_callback, > + (unsigned long) dpcm); > spin_lock_init(&dpcm->lock); > dpcm->substream = substream; > return 0; > -- > 2.2.1 >

1 0

[alsa-devel] [PATCH v2] conf/ucm: GoogleNyan: Add configuration
by Tomeu Vizoso 19 Jan '15

19 Jan '15

Taken from the ChromeOS sources, this configuration should apply to all Nyan boards from Google, so far HP Chromebook 14 (nyan-blaze) and Acer Chromebook 13 (nyan-big). Signed-off-by: Tomeu Vizoso <tomeu.vizoso(a)collabora.com> --- The board name matches what is proposed in this patch: http://article.gmane.org/gmane.linux.ports.arm.kernel/386036 v2: * Fix whitespace * Remove ChromeOS-specific stuff * Use Headphones control instead of HP Left/Right Out --- configure.ac | 1 + src/conf/ucm/GoogleNyan/GoogleNyan.conf | 5 +++ src/conf/ucm/GoogleNyan/HiFi.conf | 77 +++++++++++++++++++++++++++++++++ src/conf/ucm/GoogleNyan/Makefile.am | 4 ++ src/conf/ucm/Makefile.am | 2 +- 5 files changed, 88 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/GoogleNyan/GoogleNyan.conf create mode 100644 src/conf/ucm/GoogleNyan/HiFi.conf create mode 100644 src/conf/ucm/GoogleNyan/Makefile.am diff --git a/configure.ac b/configure.ac index 734163c..916280b 100644 --- a/configure.ac +++ b/configure.ac @@ -654,6 +654,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/tegraalc5632/Makefile \ src/conf/ucm/PAZ00/Makefile \ + src/conf/ucm/GoogleNyan/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ alsalisp/Makefile aserver/Makefile \ test/Makefile test/lsb/Makefile \ diff --git a/src/conf/ucm/GoogleNyan/GoogleNyan.conf b/src/conf/ucm/GoogleNyan/GoogleNyan.conf new file mode 100644 index 0000000..efe6ad8 --- /dev/null +++ b/src/conf/ucm/GoogleNyan/GoogleNyan.conf @@ -0,0 +1,5 @@ +Comment "Nyan internal card" +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} diff --git a/src/conf/ucm/GoogleNyan/HiFi.conf b/src/conf/ucm/GoogleNyan/HiFi.conf new file mode 100644 index 0000000..b28fe91 --- /dev/null +++ b/src/conf/ucm/GoogleNyan/HiFi.conf @@ -0,0 +1,77 @@ +SectionVerb { + EnableSequence [ + cdev "hw:GoogleNyan" + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Headphone Left Switch' on" + cset "name='Headphone Right Switch' on" + cset "name='Digital EQ 3 Band Switch' off" + cset "name='Digital EQ 5 Band Switch' off" + cset "name='Digital EQ 7 Band Switch' off" + cset "name='Biquad Switch' off" + cset "name='Filter Mode' Music" + cset "name='ADC Oversampling Rate' 0" + cset "name='DMIC Mux' DMIC" + cset "name='MIC2 Mux' IN34" + cset "name='Right ADC Mixer MIC2 Switch' on" + cset "name='Left ADC Mixer MIC2 Switch' on" + cset "name='MIC2 Volume' 10" + cset "name='MIC2 Boost Volume' 0" + cset "name='Mic Jack Switch' off" + cset "name='Int Mic Switch' on" + cset "name='ADCR Boost Volume' 4" + cset "name='ADCL Boost Volume' 4" + cset "name='ADCR Volume' 11" + cset "name='ADCL Volume' 11" + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Speaker Left Mixer Volume' 2" + cset "name='Speaker Right Mixer Volume' 2" + cset "name='Record Path DC Blocking' on" + cset "name='Playback Path DC Blocking' on" + cset "name='Headphone Left Switch' on" + cset "name='Headphone Right Switch' on" + cset "name='Headphones Switch' off" + cset "name='Speaker Left Switch' on" + cset "name='Speaker Right Switch' on" + cset "name='Speakers Switch' on" + ] + DisableSequence [ + ] +} + +SectionDevice."Headphone".0 { + Value { + OutputDspName "" + } + EnableSequence [ + cdev "hw:GoogleNyan" + cset "name='Speakers Switch' off" + cset "name='Headphones Switch' on" + ] + DisableSequence [ + cdev "hw:GoogleNyan" + cset "name='Headphones Switch' off" + cset "name='Speakers Switch' on" + ] +} + +SectionDevice."Mic".0 { + Value { + CaptureControl "MIC2" + } + EnableSequence [ + cdev "hw:GoogleNyan" + cset "name='Int Mic Switch' off" + cset "name='DMIC Mux' ADC" + cset "name='Mic Jack Switch' on" + cset "name='Record Path DC Blocking' on" + ] + DisableSequence [ + cdev "hw:GoogleNyan" + cset "name='Mic Jack Switch' off" + cset "name='DMIC Mux' DMIC" + cset "name='Int Mic Switch' on" + cset "name='Record Path DC Blocking' off" + ] +} diff --git a/src/conf/ucm/GoogleNyan/Makefile.am b/src/conf/ucm/GoogleNyan/Makefile.am new file mode 100644 index 0000000..47c7fc9 --- /dev/null +++ b/src/conf/ucm/GoogleNyan/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/GoogleNyan +ucm_DATA = GoogleNyan.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index bde89df..14fc7ae 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan -- 1.9.3

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[alsa-devel] [PATCH - UCM 1/1] ucm: add binary configure file parse
by han.lu＠intel.com 19 Jan '15

19 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> with cset command, UCM set kcontrol parameters directly: cset "name='<KCONTROL_NAME>' 1<,2,3,...>" This patch enables UCM to set kcontrol with parameters from configure file: cset-bin-file "name='<KCONTROL_NAME>' <path/to/file>" where "cset-bin-file" is a newly added keyword alongside of "cset", to indicate cset with binary data in file. The binary data in file is parameter for audio DSPs, and it's just passed by UCM/ALSA as raw data. The data type of the parameter element must be byte, and the count is limited by the size of struct snd_ctl_elem_value and driver definition. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 37ae4c8..a1b1fb6 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -160,11 +160,57 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int binary_file_parse(snd_ctl_elem_value_t *dst, + snd_ctl_elem_info_t *info, + const char *filepath) +{ + int err, fd; + struct stat st; + size_t sz; + char *res; + snd_ctl_elem_type_t type; + unsigned int idx, count; + + type = snd_ctl_elem_info_get_type(info); + if (type != SND_CTL_ELEM_TYPE_BYTES) { + uc_error("only support byte type!"); + err = -EINVAL; + return err; + } + fd = open(filepath, O_RDONLY); + if (fd < 0) { + err = -errno; + return err; + } + if (stat(filepath, &st) == -1) { + err = -errno; + goto __fail; + } + sz = st.st_size; + if (sz > sizeof(dst->value.bytes)) + sz = sizeof(dst->value.bytes); + count = snd_ctl_elem_info_get_count(info); + if (sz > count) + sz = count; + res = calloc(sz, 1); + if (res == NULL) { + err = -ENOMEM; + goto __fail; + } + read(fd, res, sz); + for (idx = 0; idx < sz; idx++) + snd_ctl_elem_value_set_byte(dst, idx, *(res + idx)); + free(res); + __fail: + close(fd); + return err; +} + extern int __snd_ctl_ascii_elem_id_parse(snd_ctl_elem_id_t *dst, const char *str, const char **ret_ptr); -static int execute_cset(snd_ctl_t *ctl, const char *cset) +static int execute_cset(snd_ctl_t *ctl, const char *cset, unsigned int type) { const char *pos; int err; @@ -194,7 +240,10 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset) err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (type == SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE) + err = binary_file_parse(value, info, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); if (err < 0) goto __fail; err = snd_ctl_elem_write(ctl, value); @@ -239,6 +288,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail_nomem; break; case SEQUENCE_ELEMENT_TYPE_CSET: + case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: if (cdev == NULL) { const char *cdev1 = NULL, *cdev2 = NULL; err = get_value3(&cdev1, "PlaybackCTL", @@ -274,7 +324,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail; } } - err = execute_cset(ctl, s->data.cset); + err = execute_cset(ctl, s->data.cset, s->type); if (err < 0) { uc_error("unable to execute cset '%s'\n", s->data.cset); goto __fail; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index d7517f6..9e1cb41 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -306,6 +306,16 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-bin-file") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-bin-file requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 87f14a2..c1655c7 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -47,6 +47,7 @@ #define SEQUENCE_ELEMENT_TYPE_CSET 2 #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 +#define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 struct ucm_value { struct list_head list; -- 2.1.0

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[alsa-devel] Strange bug in alsa-lib
by plcmn 19 Jan '15

19 Jan '15

Greetings everyone! I'm sorry in advance for my poor English. Unfortunately your bugtracker is down so I've had to send this bug report (or description, huh) here. Alsa information about my system is in the attachments. I've came across this bug a few days ago, it appeared since I've installed alsa. Everytime when I try to quickly change the volume on Master channel the sound brakes: alsamixer hangs up and some music players (for example, Music on Console) play music by fits and starts (it's similar to watching hd video on a slow connection). Some music players, for example DeadBeeF, or browsers, such as Chromium work quite normal. I've searched forums, but I didn't come across anyone with a similar problem. So, I've decided to do a little research to find out what caused the problems. I've started with a simple script: plcmn@equestria:~$ cat test.sh export i=0 while [ $i -lt 80 ] do amixer set Master $i% let i=$(($i+1)) done Here are the results of the execution before the bug appears: plcmn@equestria:~$ time ./test.sh real 0m0.743s user 0m0.350s sys 0m0.203s And this how it looks like when the bug comes out (I've just launched this script two times in a row): plcmn@equestria~$ time test.sh real 4m41.763s user 0m0.607s sys 0m12.037s That looked really weird, so I've tried to figure out what caused the problem. Checking the source with cppcheck didn't gave any results, except some memory leaks (like [src/rawmidi/rawmidi_hw.c:304]: (error) Memory leak: hw), but none of them referred to my problem. So I've digged into the code and found out the line that caused the error: --- From src/control/control_hw.c: 193 if (ioctl(hw->fd, SNDRV_CTL_IOCTL_ELEM_WRITE, control) < 0) { --- I'm a dummy in system programming, so I can't exactly tell what's the problem with this line (I know ioctl's purpose, but I don't what to do next -- a mess of macros and structures were hard for understanding, so I stopped at this point). The delay measured by clock() from time.h was about 0.18 ms here. Here is gdb's output for arguments passed to the problematic function: Breakpoint 2, snd_ctl_hw_elem_write (handle=<optimized out>, control=0x7fffffffd9e0) at control_hw.c:193 (gdb) n 194 snd_ctl_hw_t *hw = handle->private_data; (gdb) p handle $2 = <optimized out> (gdb) p control $3 = (snd_ctl_elem_value_t *) 0x7fffffffd9e0 (gdb) p control->id $4 = {numid = 16, iface = 2, device = 0, subdevice = 0, name = "Master Playback Volume", '\000' <repeats 21 times>, index = 0} (gdb) p control->indirect $5 = 0 (gdb) p control->reserved $6 = '\000' <repeats 111 times> (gdb) p control->tstamp $7 = {tv_sec = 0, tv_nsec = 0} (gdb) p control->value $8 = {integer = {value = {70, 0 <repeats 127 times>}, value_ptr = 0x46}, integer64 = {value = {70, 0 <repeats 63 times>}, value_ptr = 0x46}, enumerated = {item = {70, 0 <repeats 127 times>}, item_ptr = 0x46}, bytes = {data = "F", '\000' <repeats 510 times>, data_ptr = 0x46 <error: Cannot access memory at address 0x46>}, iec958 = {status = "F", '\000' <repeats 22 times>, subcode = '\000' <repeats 146 times>, pad = 0 '\000', dig_subframe = "\000\000\000"}} For some reason this function is called twice, the second call doesn't cause such a delay. I can provide any information you'll need. I hope that my report will be useful. Sorry again for my poor English and sending a bug report to dev mail lists.

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[alsa-devel] [PATCH] ASoC: OMAP: mcbsp: ensure that CLKX and CLKR are not used as ouput pins when they are used as input clock for the SRG.
by Thomas Niederprüm 19 Jan '15

19 Jan '15

This patch fixes faulty behaviour in a setup where the input clock for the SRG is fed through the CLKR pin but the McBSP is configured to be master (SND_SOC_DAIFMT_CBS_CFS). In that case of course CLKR must not be configured as output pin. Otherwise the input clock is messed up horribly. The same reasoning applies if CLKX is configured as input for the SRG. Signed-off-by: Thomas Niederprüm <niederp(a)physik.uni-kl.de> --- sound/soc/omap/omap-mcbsp.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index bd3ef2a..c89f562 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -530,8 +530,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_SYSCLK_CLKX_EXT: regs->srgr2 |= CLKSM; + regs->pcr0 |= SCLKME; + regs->pcr0 &= ~CLKXM; + break; case OMAP_MCBSP_SYSCLK_CLKR_EXT: regs->pcr0 |= SCLKME; + regs->pcr0 &= ~CLKRM; break; default: err = -ENODEV; -- 2.1.1

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[alsa-devel] [PATCH] Add Logitech Webcam C210 to list of Logitech webcams that require mic volume fix
by Jason Lee Cragg 18 Jan '15

18 Jan '15

From: Jason Lee Cragg <jcragg(a)gmail.com> Signed-off-by: Jason Lee Cragg <jcragg(a)gmail.com> --- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 41650d5..3e2ef61 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -913,6 +913,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x0807): /* Logitech Webcam C500 */ case USB_ID(0x046d, 0x0808): case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x0819): /* Logitech Webcam C210 */ case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: asihpi: asihpi: Remove unused function
by Rickard Strandqvist 17 Jan '15

17 Jan '15

Remove the function hpi_stream_group_get_map() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/asihpi/asihpi.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e9273fb..920b15e 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -271,15 +271,6 @@ static inline u16 hpi_stream_group_reset(u32 h_stream) return hpi_instream_group_reset(h_stream); } -static inline u16 hpi_stream_group_get_map( - u32 h_stream, u32 *mo, u32 *mi) -{ - if (hpi_handle_object(h_stream) == HPI_OBJ_OSTREAM) - return hpi_outstream_group_get_map(h_stream, mo, mi); - else - return hpi_instream_group_get_map(h_stream, mo, mi); -} - static u16 handle_error(u16 err, int line, char *filename) { if (err) -- 1.7.10.4

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Re: [alsa-devel] Digital Input on ALC3661 (from laptop HDMI IN) not working under Linux
by Oliver Freyermuth 17 Jan '15

17 Jan '15

>> I have an Alienware 17 laptop (sometimes called M17RX5) which has a > Realtek ALC3661. >>The problem I describe here should also be valid for the Alienware 18 > (all the late 2013 models, which are still the most current). At least > some scattered forum reports suggest so. > >>The card works almost perfect in Linux, but one functionality is > missing: The laptop features an HDMI input, and the Realtek card is > supposed to take care of the incoming digital audio. > This works fine in Windows, but in Linux, I only get the video and the > audio stays missing. > >> Here you can find my alsa-info: > http://www.alsa-project.org/db/?f=0a0502e49212a0436a30fde9d4ee56163bbc2ebc > > The only digial input is node 0x0a > > Try hda-jack-retask to change node 0x1f with pin default [Jack] digital In Thanks a lot for your suggestion! I tried, 0x0a stays as "not present" but I can select it in alsa-mixer as expected. Sadly, I can only record silence. Also setting Jack Detection to "Present" in the advanced overrides of hdajackretask did not help. Do you have further ideas / is there more information I can provide / collect? Cheers, Oliver > > Node 0x0a [Audio Input] wcaps 0x100791: Stereo Digital > Converter: stream=0, channel=0 > SDI-Select: 0 > Digital: > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x570]: 32000 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Connection: 1 > 0x1f > > Node 0x1f [Pin Complex] wcaps 0x400681: Stereo Digital > Pincap 0x00000020: IN > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > >>In Windows, I observe the HDMI-IN audio routing to the speakers is only > done if the Realtek Audio Service is running. In Linux, I observe > keypress-events when I either switch manually to the HDMI-in signal and also > if a new signal is detected (something (the UEFI?) injects a keypress > then). > I assume the service in Windows catches these events and changes > "something" in the Realtek Audio Settings. I see no special recording > device in Windows, nor any explicit HDMI-input setting, nor any related > setting in the driver. Dell links a standard Realtek driver in their > support section. > >>Note the laptop has a single HDMI connector which can be used both for > input and output (so the nvidia-card > also shows one HDMI-out). The second NVIDIA audio out is very likely > from the mini-display-port of the > laptop. > -- Oliver Freyermuth Physikalisches Institut der Universität Bonn Nußallee 12 53115 Bonn -- Raum W 0.018 (W152), Tel.: 0228 73 3697 --

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[alsa-devel] [RESEND PATCH 0/6] Add support for the WM8280 and WM8281 codecs
by Charles Keepax 17 Jan '15

17 Jan '15

This set of patches adds support for the WM8280 and WM8281 codecs to the Wolfson Arizona drivers. I made one small correction to the first patch to removed a potentially uninitialised variable, but otherwise just a resend of Richard's work. I would suggest we put the whole series through the MFD tree I believe we have all the required Acks and I don't believe there arn't any other patches making their way through at the moment that this should conflict with. Thanks, Charles Richard Fitzgerald (6): mfd: arizona: Add support for WM8280/WM8281 Documentation: devicetree: arizona: Add bindings for WM8280 regulator: arizona-micsupp: Add support for WM8280/WM8281 gpio: arizona: Add support for WM8280/WM8281 extcon: arizona: Add support for WM8280/WM8281 ASoC: arizona: Add support for WM8280/WM8281 Documentation/devicetree/bindings/mfd/arizona.txt | 15 +++++++++++---- drivers/extcon/extcon-arizona.c | 1 + drivers/gpio/gpio-arizona.c | 1 + drivers/mfd/Kconfig | 5 +++-- drivers/mfd/arizona-core.c | 15 +++++++++++++-- drivers/mfd/arizona-i2c.c | 2 ++ drivers/mfd/arizona-irq.c | 1 + drivers/mfd/arizona-spi.c | 2 ++ drivers/regulator/arizona-micsupp.c | 1 + include/linux/mfd/arizona/core.h | 1 + sound/soc/codecs/arizona.c | 2 ++ 11 files changed, 38 insertions(+), 8 deletions(-) -- 1.7.2.5

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[alsa-devel] ags-0_4_2-34
by Joël Krähemann 17 Jan '15

17 Jan '15

Hi all Definitively a success story by itselves http://ags.sf.net Thank you very much for your help. best regards Joël

1 0

[alsa-devel] [PATCH 0/4] add Studio Evolution SE6X support
by Clemens Ladisch 17 Jan '15

17 Jan '15

This is a driver for an unimportant sound card, and related changes in the support code. Please apply to -next. sound/pci/Kconfig | 9 + sound/pci/oxygen/Makefile | 2 sound/pci/oxygen/oxygen.h | 2 sound/pci/oxygen/oxygen_lib.c | 26 ++++- sound/pci/oxygen/oxygen_mixer.c | 36 +++++++ sound/pci/oxygen/oxygen_pcm.c | 45 ++++++++- sound/pci/oxygen/se6x.c | 160 ++++++++++++++++++++++++++++++++ 7 files changed, 272 insertions(+), 8 deletions(-)

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[alsa-devel] [PATCH] [ALSA] ASoC: Add max98925 codec driver
by Anish Kumar 17 Jan '15

17 Jan '15

From: Anish Kumar <anish.kumar(a)maximintegrated.com> This patch adds the max98925 codec driver. Signed-off-by: Anish Kumar <anish.kumar(a)maximintegrated.com> --- sound/soc/codecs/max98925.c | 954 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max98925.h | 842 ++++++++++++++++++++++++++++++++++++++ 2 files changed, 1796 insertions(+) create mode 100644 sound/soc/codecs/max98925.c create mode 100644 sound/soc/codecs/max98925.h diff --git a/sound/soc/codecs/max98925.c b/sound/soc/codecs/max98925.c new file mode 100644 index 0000000..ab78064 --- /dev/null +++ b/sound/soc/codecs/max98925.c @@ -0,0 +1,954 @@ +/* + * max98925.c -- ALSA SoC Stereo MAX98925 driver + * + * Copyright 2013-15 Maxim Integrated Products + * + * 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. + */ + +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/cdev.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/max98925.h> +#include "max98925.h" + +static struct reg_default max98925_reg[] = { + { 0x00, 0x00 }, /* Battery Voltage Data */ + { 0x01, 0x00 }, /* Boost Voltage Data */ + { 0x02, 0x00 }, /* Live Status0 */ + { 0x03, 0x00 }, /* Live Status1 */ + { 0x04, 0x00 }, /* Live Status2 */ + { 0x05, 0x00 }, /* State0 */ + { 0x06, 0x00 }, /* State1 */ + { 0x07, 0x00 }, /* State2 */ + { 0x08, 0x00 }, /* Flag0 */ + { 0x09, 0x00 }, /* Flag1 */ + { 0x0A, 0x00 }, /* Flag2 */ + { 0x0B, 0x00 }, /* IRQ Enable0 */ + { 0x0C, 0x00 }, /* IRQ Enable1 */ + { 0x0D, 0x00 }, /* IRQ Enable2 */ + { 0x0E, 0x00 }, /* IRQ Clear0 */ + { 0x0F, 0x00 }, /* IRQ Clear1 */ + { 0x10, 0x00 }, /* IRQ Clear2 */ + { 0x11, 0xC0 }, /* Map0 */ + { 0x12, 0x00 }, /* Map1 */ + { 0x13, 0x00 }, /* Map2 */ + { 0x14, 0xF0 }, /* Map3 */ + { 0x15, 0x00 }, /* Map4 */ + { 0x16, 0xAB }, /* Map5 */ + { 0x17, 0x89 }, /* Map6 */ + { 0x18, 0x00 }, /* Map7 */ + { 0x19, 0x00 }, /* Map8 */ + { 0x1A, 0x06 }, /* DAI Clock Mode 1 */ + { 0x1B, 0xC0 }, /* DAI Clock Mode 2 */ + { 0x1C, 0x00 }, /* DAI Clock Divider Denominator MSBs */ + { 0x1D, 0x00 }, /* DAI Clock Divider Denominator LSBs */ + { 0x1E, 0xF0 }, /* DAI Clock Divider Numerator MSBs */ + { 0x1F, 0x00 }, /* DAI Clock Divider Numerator LSBs */ + { 0x20, 0x50 }, /* Format */ + { 0x21, 0x00 }, /* TDM Slot Select */ + { 0x22, 0x00 }, /* DOUT Configuration VMON */ + { 0x23, 0x00 }, /* DOUT Configuration IMON */ + { 0x24, 0x00 }, /* DOUT Configuration VBAT */ + { 0x25, 0x00 }, /* DOUT Configuration VBST */ + { 0x26, 0x00 }, /* DOUT Configuration FLAG */ + { 0x27, 0xFF }, /* DOUT HiZ Configuration 1 */ + { 0x28, 0xFF }, /* DOUT HiZ Configuration 2 */ + { 0x29, 0xFF }, /* DOUT HiZ Configuration 3 */ + { 0x2A, 0xFF }, /* DOUT HiZ Configuration 4 */ + { 0x2B, 0x02 }, /* DOUT Drive Strength */ + { 0x2C, 0x90 }, /* Filters */ + { 0x2D, 0x00 }, /* Gain */ + { 0x2E, 0x02 }, /* Gain Ramping */ + { 0x2F, 0x00 }, /* Speaker Amplifier */ + { 0x30, 0x0A }, /* Threshold */ + { 0x31, 0x00 }, /* ALC Attack */ + { 0x32, 0x80 }, /* ALC Atten and Release */ + { 0x33, 0x00 }, /* ALC Infinite Hold Release */ + { 0x34, 0x92 }, /* ALC Configuration */ + { 0x35, 0x01 }, /* Boost Converter */ + { 0x36, 0x00 }, /* Block Enable */ + { 0x37, 0x00 }, /* Configuration */ + { 0x38, 0x00 }, /* Global Enable */ + { 0x3A, 0x00 }, /* Boost Limiter */ + { 0xFF, 0x50 }, /* Revision ID */ +}; + +static bool max98925_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_R000_VBAT_DATA: + case MAX98925_R001_VBST_DATA: + case MAX98925_R002_LIVE_STATUS0: + case MAX98925_R003_LIVE_STATUS1: + case MAX98925_R004_LIVE_STATUS2: + case MAX98925_R005_STATE0: + case MAX98925_R006_STATE1: + case MAX98925_R007_STATE2: + case MAX98925_R008_FLAG0: + case MAX98925_R009_FLAG1: + case MAX98925_R00A_FLAG2: + case MAX98925_R0FF_VERSION: + return true; + default: + return false; + } +} + +static bool max98925_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98925_R00E_IRQ_CLEAR0: + case MAX98925_R00F_IRQ_CLEAR1: + case MAX98925_R010_IRQ_CLEAR2: + case MAX98925_R033_ALC_HOLD_RLS: + return false; + default: + return true; + } +}; + +static const unsigned int max98925_spk_tlv[] = { + TLV_DB_RANGE_HEAD(1), + 1, 31, TLV_DB_SCALE_ITEM(-600, 100, 0), +}; + +static int max98925_spk_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = max98925->spk_gain; + pr_debug("%s: spk_gain setting returned %d\n", __func__, + (int) ucontrol->value.integer.value[0]); + return 0; +} + +static int max98925_spk_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + if (sel < ((1 << M98925_SPK_GAIN_WIDTH) - 1)) { + regmap_update_bits(max98925->regmapL, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, sel << M98925_SPK_GAIN_SHIFT); + regmap_update_bits(max98925->regmapR, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, sel << M98925_SPK_GAIN_SHIFT); + + max98925->spk_gain = sel; + + pr_debug("%s: spk_gain set to %d\n", __func__, sel); + } else { + pr_debug("%s: valid speaker gain settings: %d to %d\n", + __func__, 0, + ((1 << M98925_SPK_GAIN_WIDTH) - 1)); + } + return 0; +} + +static int max98925_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + int data; + + regmap_read(max98925->regmapL, reg, &data); + ucontrol->value.integer.value[0] = + (data & mask) >> shift; + return 0; +} + +static int max98925_reg_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol, unsigned int reg, + unsigned int mask, unsigned int shift) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + unsigned int sel = ucontrol->value.integer.value[0]; + + regmap_update_bits(max98925->regmapL, reg, mask, sel << shift); + regmap_update_bits(max98925->regmapR, reg, mask, sel << shift); + + pr_debug("%s: register 0x%02X, value 0x%02X\n", __func__, reg, sel); + return 0; +} + +static int max98925_spk_ramp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_RMP_EN_MASK, M98925_SPK_RMP_EN_SHIFT); +} + +static int max98925_spk_ramp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_RMP_EN_MASK, M98925_SPK_RMP_EN_SHIFT); +} + +static int max98925_spk_zcd_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_ZCD_EN_MASK, M98925_SPK_ZCD_EN_SHIFT); +} + +static int max98925_spk_zcd_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R02E_GAIN_RAMPING, + M98925_SPK_ZCD_EN_MASK, M98925_SPK_ZCD_EN_SHIFT); +} + +static int max98925_alc_en_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_EN_MASK, M98925_ALC_EN_SHIFT); +} + +static int max98925_alc_en_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_EN_MASK, M98925_ALC_EN_SHIFT); +} + +static int max98925_alc_threshold_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_TH_MASK, M98925_ALC_TH_SHIFT); +} + +static int max98925_alc_threshold_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R030_THRESHOLD, + M98925_ALC_TH_MASK, M98925_ALC_TH_SHIFT); +} + +static const char * const max98925_boost_voltage_text[] = {"8.5V", "8.25V", + "8.0V", "7.75V", "7.5V", "7.25V", "7.0V", "6.75V", + "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", "6.5V", + "6.5V", "6.5V"}; + +static const struct soc_enum max98925_boost_voltage_enum = + SOC_ENUM_SINGLE(MAX98925_R037_CONFIGURATION, M98925_BST_VOUT_SHIFT, 15, + max98925_boost_voltage_text); + +static int max98925_boost_voltage_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_get(kcontrol, ucontrol, MAX98925_R037_CONFIGURATION, + M98925_BST_VOUT_MASK, M98925_BST_VOUT_SHIFT); +} + +static int max98925_boost_voltage_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return max98925_reg_put(kcontrol, ucontrol, MAX98925_R037_CONFIGURATION, + M98925_BST_VOUT_MASK, M98925_BST_VOUT_SHIFT); +} + +static const struct snd_kcontrol_new max98925_snd_controls[] = { + + SOC_SINGLE_EXT_TLV("Speaker Gain", MAX98925_R02D_GAIN, + M98925_SPK_GAIN_SHIFT, (1<<M98925_SPK_GAIN_WIDTH)-1, 0, + max98925_spk_gain_get, max98925_spk_gain_put, max98925_spk_tlv), + + SOC_SINGLE_EXT("Speaker Ramp", 0, 0, 1, 0, + max98925_spk_ramp_get, max98925_spk_ramp_put), + + SOC_SINGLE_EXT("Speaker ZCD", 0, 0, 1, 0, + max98925_spk_zcd_get, max98925_spk_zcd_put), + + SOC_SINGLE_EXT("ALC Enable", 0, 0, 1, 0, + max98925_alc_en_get, max98925_alc_en_put), + + SOC_SINGLE_EXT("ALC Threshold", 0, 0, (1<<M98925_ALC_TH_WIDTH)-1, 0, + max98925_alc_threshold_get, max98925_alc_threshold_put), + + SOC_ENUM_EXT("Boost Output Voltage", max98925_boost_voltage_enum, + max98925_boost_voltage_get, max98925_boost_voltage_put), +}; + +static int max98925_add_widgets(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_add_codec_controls(codec, max98925_snd_controls, + ARRAY_SIZE(max98925_snd_controls)); + return 0; +} + +/* codec sample rate and n/m dividers parameter table */ +static const struct { + int rate; + int sr; + int divisors[3][2]; +} rate_table[] = { + { + .rate = 8000, + .sr = 0, + .divisors = { {1, 375}, {5, 1764}, {1, 384} } + }, + { + .rate = 11025, + .sr = 1, + .divisors = { {147, 40000}, {1, 256}, {147, 40960} } + }, + { + .rate = 12000, + .sr = 2, + .divisors = { {1, 250}, {5, 1176}, {1, 256} } + }, + { + .rate = 16000, + .sr = 3, + .divisors = { {2, 375}, {5, 882}, {1, 192} } + }, + { + .rate = 22050, + .sr = 4, + .divisors = { {147, 20000}, {1, 128}, {147, 20480} } + }, + { + .rate = 24000, + .sr = 5, + .divisors = { {1, 125}, {5, 588}, {1, 128} } + }, + { + .rate = 32000, + .sr = 6, + .divisors = { {4, 375}, {5, 441}, {1, 96} } + }, + { + .rate = 44100, + .sr = 7, + .divisors = { {147, 10000}, {1, 64}, {147, 10240} } + }, + { + .rate = 48000, + .sr = 8, + .divisors = { {2, 125}, {5, 294}, {1, 64} } + }, +}; + +static inline int max98925_rate_value(int rate, + int clock, int *value, int *n, int *m) +{ + int ret = -EINVAL; + int i; + + for (i = 0; i < ARRAY_SIZE(rate_table); i++) { + if (rate_table[i].rate >= rate) { + *value = rate_table[i].sr; + *n = rate_table[i].divisors[clock][0]; + *m = rate_table[i].divisors[clock][1]; + ret = 0; + break; + } + } + pr_debug("%s: sample rate is %d, returning %d\n", + __func__, rate_table[i].rate, *value); + return ret; +} + +static int max98925_set_tdm_slot(struct snd_soc_dai *codec_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + pr_info("%s: tx_mask 0x%X, rx_mask 0x%X, slots %d, slot width %d\n", + __func__, tx_mask, rx_mask, slots, slot_width); + return 0; +} + +static void max98925_set_sense_data(struct max98925_priv *max98925) +{ + /* + * 1. set VMON slots + */ + regmap_update_bits(max98925->regmapL, MAX98925_R022_DOUT_CFG_VMON, + M98925_DAI_VMON_EN_MASK, M98925_DAI_VMON_EN_MASK); + regmap_update_bits(max98925->regmapL, MAX98925_R022_DOUT_CFG_VMON, + M98925_DAI_VMON_SLOT_MASK, M98925_DAI_VMON_SLOT_00_01); + /* + * 2. set IMON slots + */ + regmap_update_bits(max98925->regmapL, MAX98925_R023_DOUT_CFG_IMON, + M98925_DAI_IMON_EN_MASK, M98925_DAI_IMON_EN_MASK); + regmap_update_bits(max98925->regmapL, MAX98925_R023_DOUT_CFG_IMON, + M98925_DAI_IMON_SLOT_MASK, M98925_DAI_IMON_SLOT_02_03); +} + +static void max98925_set_slave(struct max98925_priv *max98925) +{ + /* + * 1. use BCLK instead of MCLK + */ + regmap_update_bits(max98925->regmapL, MAX98925_R01A_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, M98925_DAI_CLK_SOURCE_MASK); + regmap_update_bits(max98925->regmapR, MAX98925_R01A_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, M98925_DAI_CLK_SOURCE_MASK); + /* + * 2. set DAI to slave mode + */ + regmap_update_bits(max98925->regmapL, MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, 0); + regmap_update_bits(max98925->regmapR, MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, 0); + /* + * 3. set BLCKs to LRCLKs to 64 + */ + regmap_update_bits(max98925->regmapL, MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_64); + regmap_update_bits(max98925->regmapR, MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_BSEL_MASK, M98925_DAI_BSEL_64); + + max98925_set_sense_data(max98925); +} + +static void max98925_set_master(struct max98925_priv *max98925) +{ + /* + * 1. use MCLK for Left channel, right channel always BCLK + */ + regmap_update_bits(max98925->regmapL, MAX98925_R01A_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, 0); + regmap_update_bits(max98925->regmapR, MAX98925_R01A_DAI_CLK_MODE1, + M98925_DAI_CLK_SOURCE_MASK, M98925_DAI_CLK_SOURCE_MASK); + /* + * 2. set left channel DAI to master mode, right channel always slave + */ + regmap_update_bits(max98925->regmapL, MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, M98925_DAI_MAS_MASK); + regmap_update_bits(max98925->regmapR, MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_MAS_MASK, 0); +} + +static int max98925_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct max98925_cdata *cdata; + unsigned int invert = 0; + + pr_debug("%s: fmt 0x%08X\n", __func__, fmt); + cdata = &max98925->dai[0]; + cdata->fmt = fmt; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + max98925_set_slave(max98925); + break; + case SND_SOC_DAIFMT_CBM_CFM: + max98925_set_master(max98925); + break; + case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + break; + case SND_SOC_DAIFMT_DSP_A: + default: + dev_err(codec->dev, "DAI format unsupported, fmt:0x%x", fmt); + return -EINVAL; + } + pr_debug("DAI format supported, fmt:0x%x", fmt); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + invert = M98925_DAI_WCI_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + invert = M98925_DAI_BCI_MASK; + break; + case SND_SOC_DAIFMT_IB_IF: + invert = M98925_DAI_BCI_MASK | M98925_DAI_WCI_MASK; + break; + default: + dev_err(codec->dev, "DAI invert mode unsupported"); + return -EINVAL; + } + + regmap_update_bits(max98925->regmapL, MAX98925_R020_FORMAT, + M98925_DAI_BCI_MASK | M98925_DAI_BCI_MASK, invert); + regmap_update_bits(max98925->regmapR, MAX98925_R020_FORMAT, + M98925_DAI_BCI_MASK | M98925_DAI_BCI_MASK, invert); + return 0; +} + +static int max98925_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + codec->dapm.bias_level = level; + return 0; +} + +static int max98925_set_clock(struct max98925_priv *max98925, unsigned int rate) +{ + unsigned int dai_sr = 0; + unsigned int clock; + unsigned int mdll; + unsigned int n; + unsigned int m; + + switch (max98925->sysclk) { + case 6000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx16; + break; + case 11289600: + clock = 1; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12000000: + clock = 0; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + case 12288000: + clock = 2; + mdll = M98925_MDLL_MULT_MCLKx8; + break; + default: + dev_info(max98925->codec->dev, "unsupported sysclk %d\n", + max98925->sysclk); + return -EINVAL; + } + + if (max98925_rate_value(rate, clock, &dai_sr, &n, &m)) + return -EINVAL; + + /* + * 1. set DAI_SR to correct LRCLK frequency + */ + regmap_update_bits(max98925->regmapL, MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_SR_MASK, dai_sr << M98925_DAI_SR_SHIFT); + regmap_update_bits(max98925->regmapR, MAX98925_R01B_DAI_CLK_MODE2, + M98925_DAI_SR_MASK, dai_sr << M98925_DAI_SR_SHIFT); + /* + * 2. set DAI m divider + */ + regmap_write(max98925->regmapL, MAX98925_R01C_DAI_CLK_DIV_M_MSBS, + m >> 8); + regmap_write(max98925->regmapL, MAX98925_R01D_DAI_CLK_DIV_M_LSBS, + m & 0xFF); + /* + * 3. set DAI n divider + */ + regmap_write(max98925->regmapL, MAX98925_R01E_DAI_CLK_DIV_N_MSBS, + n >> 8); + regmap_write(max98925->regmapL, MAX98925_R01F_DAI_CLK_DIV_N_LSBS, + n & 0xFF); + /* + * 4. set MDLL + */ + regmap_update_bits(max98925->regmapL, MAX98925_R01A_DAI_CLK_MODE1, + M98925_MDLL_MULT_MASK, mdll << M98925_MDLL_MULT_SHIFT); + regmap_update_bits(max98925->regmapR, MAX98925_R01A_DAI_CLK_MODE1, + M98925_MDLL_MULT_MASK, mdll << M98925_MDLL_MULT_SHIFT); + + return 0; +} + +static int max98925_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct max98925_cdata *cdata; + unsigned int rate; + + cdata = &max98925->dai[0]; + rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98925->regmapL, MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_16); + regmap_update_bits(max98925->regmapR, MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98925->regmapL, MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + regmap_update_bits(max98925->regmapR, MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + pr_debug("%s: (really set to 32 bits)\n", __func__); + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98925->regmapL, MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + regmap_update_bits(max98925->regmapR, MAX98925_R020_FORMAT, + M98925_DAI_CHANSZ_MASK, M98925_DAI_CHANSZ_32); + break; + default: + pr_err("%s: format unsupported %d", + __func__, params_format(params)); + return -EINVAL; + } + pr_debug("%s: format supported %d", __func__, params_format(params)); + return max98925_set_clock(max98925, rate); +} + +static int max98925_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + + pr_debug("%s: clk_id %d, freq %d, dir %d\n", + __func__, clk_id, freq, dir); + max98925->sysclk = freq; + return 0; +} + +static int max98925_dai_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct max98925_priv *max98925 = + snd_soc_codec_get_drvdata(codec_dai->codec); + + pr_debug("%s: mute %d\n", __func__, mute); + if (mute) { + regmap_update_bits(max98925->regmapL, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, 0x00); + regmap_update_bits(max98925->regmapR, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, 0x00); + + regmap_update_bits(max98925->regmapL, + MAX98925_R038_GLOBAL_ENABLE, M98925_EN_MASK, 0x0); + regmap_update_bits(max98925->regmapR, + MAX98925_R038_GLOBAL_ENABLE, M98925_EN_MASK, 0x0); + } else { + regmap_update_bits(max98925->regmapL, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, max98925->spk_gain); + regmap_update_bits(max98925->regmapR, MAX98925_R02D_GAIN, + M98925_SPK_GAIN_MASK, max98925->spk_gain); + + regmap_update_bits(max98925->regmapL, + MAX98925_R036_BLOCK_ENABLE, + M98925_BST_EN_MASK | M98925_SPK_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, + M98925_BST_EN_MASK | M98925_SPK_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK); + regmap_update_bits(max98925->regmapR, + MAX98925_R036_BLOCK_ENABLE, + M98925_BST_EN_MASK | M98925_SPK_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK, + M98925_BST_EN_MASK | M98925_SPK_EN_MASK | + M98925_ADC_IMON_EN_MASK | M98925_ADC_VMON_EN_MASK); + + regmap_write(max98925->regmapL, MAX98925_R038_GLOBAL_ENABLE, + M98925_EN_MASK); + regmap_write(max98925->regmapR, MAX98925_R038_GLOBAL_ENABLE, + M98925_EN_MASK); + } + return 0; +} + +#define MAX98925_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops max98925_dai_ops = { + .set_sysclk = max98925_dai_set_sysclk, + .set_fmt = max98925_dai_set_fmt, + .set_tdm_slot = max98925_set_tdm_slot, + .hw_params = max98925_dai_hw_params, + .digital_mute = max98925_dai_digital_mute, +}; + +static struct snd_soc_dai_driver max98925_dai[] = { + { + .name = "max98925-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98925_FORMATS, + }, + .ops = &max98925_dai_ops, + } +}; + +static void max98925_handle_pdata(struct snd_soc_codec *codec) +{ + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct max98925_pdata *pdata = max98925->pdata; + + if (!pdata) { + dev_dbg(codec->dev, "No platform data\n"); + return; + } +} + +static int max98925_probe(struct snd_soc_codec *codec) +{ + struct max98925_priv *max98925 = snd_soc_codec_get_drvdata(codec); + struct max98925_cdata *cdata; + int ret = 0; + int reg = 0; + + dev_info(codec->dev, "build number %s\n", MAX98925_REVISION); + + max98925->codec = codec; + codec->control_data = max98925->regmapL; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + max98925->sysclk = (unsigned)-1; + max98925->spk_gain = 0x14; + + cdata = &max98925->dai[0]; + cdata->rate = (unsigned)-1; + cdata->fmt = (unsigned)-1; + + reg = 0; + ret = regmap_read(max98925->regmapL, MAX98925_R0FF_VERSION, &reg); + if ((ret < 0) || + ((reg != MAX98925_VERSION) && (reg != MAX98925_VERSION1))) { + dev_err(codec->dev, + "L device initialization error (%d 0x%02X)\n", + ret, + reg); + goto err_access; + } + dev_info(codec->dev, "L device version 0x%02X\n", reg); + + if (max98925->regmapR != NULL) { + reg = 0; + ret = regmap_read(max98925->regmapR, + MAX98925_R0FF_VERSION, &reg); + if ((ret < 0) || ((reg != MAX98925_VERSION) + && (reg != MAX98925_VERSION1))) { + dev_err(codec->dev, + "R device initialization error (%d 0x%02X)\n", + ret, reg); + goto err_access; + } + dev_info(codec->dev, "R device version 0x%02X\n", reg); + } + regmap_write(max98925->regmapL, MAX98925_R038_GLOBAL_ENABLE, 0x00); + regmap_write(max98925->regmapR, MAX98925_R038_GLOBAL_ENABLE, 0x00); + + /* It's not the default but we need to set DAI_DLY */ + regmap_write(max98925->regmapL, + MAX98925_R020_FORMAT, M98925_DAI_DLY_MASK); + regmap_write(max98925->regmapR, + MAX98925_R020_FORMAT, M98925_DAI_DLY_MASK); + + regmap_write(max98925->regmapL, + MAX98925_R021_TDM_SLOT_SELECT, 0xC8); + regmap_write(max98925->regmapR, + MAX98925_R021_TDM_SLOT_SELECT, 0xC8); + + regmap_write(max98925->regmapL, + MAX98925_R027_DOUT_HIZ_CFG1, 0xFF); + regmap_write(max98925->regmapL, + MAX98925_R028_DOUT_HIZ_CFG2, 0xFF); + regmap_write(max98925->regmapL, + MAX98925_R029_DOUT_HIZ_CFG3, 0xFF); + regmap_write(max98925->regmapL, + MAX98925_R02A_DOUT_HIZ_CFG4, 0xF0); + regmap_write(max98925->regmapR, + MAX98925_R027_DOUT_HIZ_CFG1, 0xFF); + regmap_write(max98925->regmapR, + MAX98925_R028_DOUT_HIZ_CFG2, 0xFF); + regmap_write(max98925->regmapR, + MAX98925_R029_DOUT_HIZ_CFG3, 0xFF); + regmap_write(max98925->regmapR, + MAX98925_R02A_DOUT_HIZ_CFG4, 0x0F); + + regmap_write(max98925->regmapL, + MAX98925_R02C_FILTERS, 0xD8); + regmap_write(max98925->regmapR, + MAX98925_R02C_FILTERS, 0xD8); + + regmap_write(max98925->regmapL, + MAX98925_R034_ALC_CONFIGURATION, 0xF8); + regmap_write(max98925->regmapR, + MAX98925_R034_ALC_CONFIGURATION, 0xF8); + + regmap_write(max98925->regmapL, + MAX98925_R037_CONFIGURATION, 0xF0); + regmap_write(max98925->regmapR, + MAX98925_R037_CONFIGURATION, 0xF0); + + /* Disable ALC muting */ + regmap_write(max98925->regmapL, + MAX98925_R03A_BOOST_LIMITER, 0xF8); + regmap_write(max98925->regmapR, + MAX98925_R03A_BOOST_LIMITER, 0xF8); + + regmap_update_bits(max98925->regmapL, MAX98925_R02D_GAIN, + M98925_DAC_IN_SEL_MASK, M98925_DAC_IN_SEL_LEFT_DAI); + regmap_update_bits(max98925->regmapR, MAX98925_R02D_GAIN, + M98925_DAC_IN_SEL_MASK, M98925_DAC_IN_SEL_RIGHT_DAI); + + max98925_handle_pdata(codec); + max98925_add_widgets(codec); +err_access: + return ret; +} + +static struct snd_soc_codec_driver soc_codec_dev_max98925 = { + .probe = max98925_probe, + .set_bias_level = max98925_set_bias_level, +}; + +static const struct regmap_config max98925_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98925_R0FF_VERSION, + .reg_defaults = max98925_reg, + .num_reg_defaults = ARRAY_SIZE(max98925_reg), + .volatile_reg = max98925_volatile_register, + .readable_reg = max98925_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +/* There should be a second MAX98925 on the board */ +static struct i2c_board_info max98925_i2c_second[] = { + { + I2C_BOARD_INFO("max98925R", 0x32), + } +}; + +struct i2c_client *add_second_device(int busnum) +{ + struct i2c_client *i2c = NULL; + struct i2c_adapter *adapter; + + adapter = i2c_get_adapter(busnum); + if (adapter != NULL) + i2c = i2c_new_device(adapter, max98925_i2c_second); + + return i2c; +} + +static int max98925_i2c_probe(struct i2c_client *i2c_l, + const struct i2c_device_id *id) +{ + struct max98925_priv *max98925; + struct i2c_client *i2c_r; + int ret; + + pr_info("%s: enter, device '%s'\n", __func__, id->name); + if (strcmp(id->name, "max98925R") == 0) + return 0; + + max98925 = kzalloc(sizeof(struct max98925_priv), GFP_KERNEL); + if (max98925 == NULL) + return -ENOMEM; + + max98925->devtype = id->driver_data; + i2c_set_clientdata(i2c_l, max98925); + max98925->control_data = i2c_l; + max98925->pdata = i2c_l->dev.platform_data; + + max98925->regmapL = regmap_init_i2c(i2c_l, &max98925_regmap); + if (IS_ERR(max98925->regmapL)) { + ret = PTR_ERR(max98925->regmapL); + dev_err(&i2c_l->dev, "Failed to allocate regmapL: %d\n", ret); + goto err_out; + } + + /* Check for second MAX98925 */ + i2c_r = add_second_device(2); + + if (IS_ERR(i2c_r)) { + dev_err(&i2c_l->dev, "second MAX98925 not found\n"); + ret = -ENODEV; + goto err_out; + } else { + max98925->regmapR = regmap_init_i2c(i2c_r, &max98925_regmap); + if (IS_ERR(max98925->regmapR)) { + ret = PTR_ERR(max98925->regmapR); + dev_err(&i2c_r->dev, + "Failed to allocate regmapR: %d\n", ret); + goto err_out; + } + } + ret = snd_soc_register_codec(&i2c_l->dev, &soc_codec_dev_max98925, + max98925_dai, ARRAY_SIZE(max98925_dai)); +err_out: + if (ret < 0) { + if (max98925->regmapL) + regmap_exit(max98925->regmapL); + if (max98925->regmapR) + regmap_exit(max98925->regmapR); + kfree(max98925); + } + return ret; +} + +static int max98925_i2c_remove(struct i2c_client *client) +{ + struct max98925_priv *max98925 = dev_get_drvdata(&client->dev); + + snd_soc_unregister_codec(&client->dev); + if (max98925->regmapL) + regmap_exit(max98925->regmapL); + if (max98925->regmapR) + regmap_exit(max98925->regmapL); + kfree(i2c_get_clientdata(client)); + return 0; +} + +static const struct i2c_device_id max98925_i2c_id[] = { + { "max98925", MAX98925 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max98925_i2c_id); + +static struct i2c_driver max98925_i2c_driver = { + .driver = { + .name = "max98925", + .owner = THIS_MODULE, + .pm = NULL, + }, + .probe = max98925_i2c_probe, + .remove = max98925_i2c_remove, + .id_table = max98925_i2c_id, +}; + +module_i2c_driver(max98925_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC MAX98925 Stereo driver"); +MODULE_AUTHOR("Ralph Birt <rdbirt(a)gmail.com>, Anish kumar <anish.kumar(a)maximintegrated.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98925.h b/sound/soc/codecs/max98925.h new file mode 100644 index 0000000..9d89d34 --- /dev/null +++ b/sound/soc/codecs/max98925.h @@ -0,0 +1,842 @@ +/* + * max98925.h -- MAX98925 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * 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 _MAX98925_H +#define _MAX98925_H + +/* Maximum number of MAX98925 devices in the system this driver can support */ +#define MAX_NUM_MAX98925 2 +#define MAX98925_VERSION 0x51 +#define MAX98925_VERSION1 0x80 + +#define MAX98925_REVISION "0.00.0332" + +#define MAX98925_R000_VBAT_DATA 0x00 +#define MAX98925_R001_VBST_DATA 0x01 +#define MAX98925_R002_LIVE_STATUS0 0x02 +#define MAX98925_R003_LIVE_STATUS1 0x03 +#define MAX98925_R004_LIVE_STATUS2 0x04 +#define MAX98925_R005_STATE0 0x05 +#define MAX98925_R006_STATE1 0x06 +#define MAX98925_R007_STATE2 0x07 +#define MAX98925_R008_FLAG0 0x08 +#define MAX98925_R009_FLAG1 0x09 +#define MAX98925_R00A_FLAG2 0x0A +#define MAX98925_R00B_IRQ_ENABLE0 0x0B +#define MAX98925_R00C_IRQ_ENABLE1 0x0C +#define MAX98925_R00D_IRQ_ENABLE2 0x0D +#define MAX98925_R00E_IRQ_CLEAR0 0x0E +#define MAX98925_R00F_IRQ_CLEAR1 0x0F +#define MAX98925_R010_IRQ_CLEAR2 0x10 +#define MAX98925_R011_MAP0 0x11 +#define MAX98925_R012_MAP1 0x12 +#define MAX98925_R013_MAP2 0x13 +#define MAX98925_R014_MAP3 0x14 +#define MAX98925_R015_MAP4 0x15 +#define MAX98925_R016_MAP5 0x16 +#define MAX98925_R017_MAP6 0x17 +#define MAX98925_R018_MAP7 0x18 +#define MAX98925_R019_MAP8 0x19 +#define MAX98925_R01A_DAI_CLK_MODE1 0x1A +#define MAX98925_R01B_DAI_CLK_MODE2 0x1B +#define MAX98925_R01C_DAI_CLK_DIV_M_MSBS 0x1C +#define MAX98925_R01D_DAI_CLK_DIV_M_LSBS 0x1D +#define MAX98925_R01E_DAI_CLK_DIV_N_MSBS 0x1E +#define MAX98925_R01F_DAI_CLK_DIV_N_LSBS 0x1F +#define MAX98925_R020_FORMAT 0x20 +#define MAX98925_R021_TDM_SLOT_SELECT 0x21 +#define MAX98925_R022_DOUT_CFG_VMON 0x22 +#define MAX98925_R023_DOUT_CFG_IMON 0x23 +#define MAX98925_R024_DOUT_CFG_VBAT 0x24 +#define MAX98925_R025_DOUT_CFG_VBST 0x25 +#define MAX98925_R026_DOUT_CFG_FLAG 0x26 +#define MAX98925_R027_DOUT_HIZ_CFG1 0x27 +#define MAX98925_R028_DOUT_HIZ_CFG2 0x28 +#define MAX98925_R029_DOUT_HIZ_CFG3 0x29 +#define MAX98925_R02A_DOUT_HIZ_CFG4 0x2A +#define MAX98925_R02B_DOUT_DRV_STRENGTH 0x2B +#define MAX98925_R02C_FILTERS 0x2C +#define MAX98925_R02D_GAIN 0x2D +#define MAX98925_R02E_GAIN_RAMPING 0x2E +#define MAX98925_R02F_SPK_AMP 0x2F +#define MAX98925_R030_THRESHOLD 0x30 +#define MAX98925_R031_ALC_ATTACK 0x31 +#define MAX98925_R032_ALC_ATTEN_RLS 0x32 +#define MAX98925_R033_ALC_HOLD_RLS 0x33 +#define MAX98925_R034_ALC_CONFIGURATION 0x34 +#define MAX98925_R035_BOOST_CONVERTER 0x35 +#define MAX98925_R036_BLOCK_ENABLE 0x36 +#define MAX98925_R037_CONFIGURATION 0x37 +#define MAX98925_R038_GLOBAL_ENABLE 0x38 +#define MAX98925_R03A_BOOST_LIMITER 0x3A +#define MAX98925_R0FF_VERSION 0xFF + +#define MAX98925_REG_CNT (MAX98925_R03A_BOOST_LIMITER+1) + +/* MAX98925 Register Bit Fields */ + +/* MAX98925_R002_LIVE_STATUS0 */ +#define M98925_THERMWARN_STATUS_MASK (1<<3) +#define M98925_THERMWARN_STATUS_SHIFT 3 +#define M98925_THERMWARN_STATUS_WIDTH 1 +#define M98925_THERMSHDN_STATUS_MASK (1<<1) +#define M98925_THERMSHDN_STATUS_SHIFT 1 +#define M98925_THERMSHDN_STATUS_WIDTH 1 + +/* MAX98925_R003_LIVE_STATUS1 */ +#define M98925_SPKCURNT_STATUS_MASK (1<<5) +#define M98925_SPKCURNT_STATUS_SHIFT 5 +#define M98925_SPKCURNT_STATUS_WIDTH 1 +#define M98925_WATCHFAIL_STATUS_MASK (1<<4) +#define M98925_WATCHFAIL_STATUS_SHIFT 4 +#define M98925_WATCHFAIL_STATUS_WIDTH 1 +#define M98925_ALCINFH_STATUS_MASK (1<<3) +#define M98925_ALCINFH_STATUS_SHIFT 3 +#define M98925_ALCINFH_STATUS_WIDTH 1 +#define M98925_ALCACT_STATUS_MASK (1<<2) +#define M98925_ALCACT_STATUS_SHIFT 2 +#define M98925_ALCACT_STATUS_WIDTH 1 +#define M98925_ALCMUT_STATUS_MASK (1<<1) +#define M98925_ALCMUT_STATUS_SHIFT 1 +#define M98925_ALCMUT_STATUS_WIDTH 1 +#define M98925_ACLP_STATUS_MASK (1<<0) +#define M98925_ACLP_STATUS_SHIFT 0 +#define M98925_ACLP_STATUS_WIDTH 1 + +/* MAX98925_R004_LIVE_STATUS2 */ +#define M98925_SLOTOVRN_STATUS_MASK (1<<6) +#define M98925_SLOTOVRN_STATUS_SHIFT 6 +#define M98925_SLOTOVRN_STATUS_WIDTH 1 +#define M98925_INVALSLOT_STATUS_MASK (1<<5) +#define M98925_INVALSLOT_STATUS_SHIFT 5 +#define M98925_INVALSLOT_STATUS_WIDTH 1 +#define M98925_SLOTCNFLT_STATUS_MASK (1<<4) +#define M98925_SLOTCNFLT_STATUS_SHIFT 4 +#define M98925_SLOTCNFLT_STATUS_WIDTH 1 +#define M98925_VBSTOVFL_STATUS_MASK (1<<3) +#define M98925_VBSTOVFL_STATUS_SHIFT 3 +#define M98925_VBSTOVFL_STATUS_WIDTH 1 +#define M98925_VBATOVFL_STATUS_MASK (1<<2) +#define M98925_VBATOVFL_STATUS_SHIFT 2 +#define M98925_VBATOVFL_STATUS_WIDTH 1 +#define M98925_IMONOVFL_STATUS_MASK (1<<1) +#define M98925_IMONOVFL_STATUS_SHIFT 1 +#define M98925_IMONOVFL_STATUS_WIDTH 1 +#define M98925_VMONOVFL_STATUS_MASK (1<<0) +#define M98925_VMONOVFL_STATUS_SHIFT 0 +#define M98925_VMONOVFL_STATUS_WIDTH 1 + +/* MAX98925_R005_STATE0 */ +#define M98925_THERMWARN_END_STATE_MASK (1<<3) +#define M98925_THERMWARN_END_STATE_SHIFT 3 +#define M98925_THERMWARN_END_STATE_WIDTH 1 +#define M98925_THERMWARN_BGN_STATE_MASK (1<<2) +#define M98925_THERMWARN_BGN_STATE_SHIFT 1 +#define M98925_THERMWARN_BGN_STATE_WIDTH 1 +#define M98925_THERMSHDN_END_STATE_MASK (1<<1) +#define M98925_THERMSHDN_END_STATE_SHIFT 1 +#define M98925_THERMSHDN_END_STATE_WIDTH 1 +#define M98925_THERMSHDN_BGN_STATE_MASK (1<<0) +#define M98925_THERMSHDN_BGN_STATE_SHIFT 0 +#define M98925_THERMSHDN_BGN_STATE_WIDTH 1 + +/* MAX98925_R006_STATE1 */ +#define M98925_SPRCURNT_STATE_MASK (1<<5) +#define M98925_SPRCURNT_STATE_SHIFT 5 +#define M98925_SPRCURNT_STATE_WIDTH 1 +#define M98925_WATCHFAIL_STATE_MASK (1<<4) +#define M98925_WATCHFAIL_STATE_SHIFT 4 +#define M98925_WATCHFAIL_STATE_WIDTH 1 +#define M98925_ALCINFH_STATE_MASK (1<<3) +#define M98925_ALCINFH_STATE_SHIFT 3 +#define M98925_ALCINFH_STATE_WIDTH 1 +#define M98925_ALCACT_STATE_MASK (1<<2) +#define M98925_ALCACT_STATE_SHIFT 2 +#define M98925_ALCACT_STATE_WIDTH 1 +#define M98925_ALCMUT_STATE_MASK (1<<1) +#define M98925_ALCMUT_STATE_SHIFT 1 +#define M98925_ALCMUT_STATE_WIDTH 1 +#define M98925_ALCP_STATE_MASK (1<<0) +#define M98925_ALCP_STATE_SHIFT 0 +#define M98925_ALCP_STATE_WIDTH 1 + +/* MAX98925_R007_STATE2 */ +#define M98925_SLOTOVRN_STATE_MASK (1<<6) +#define M98925_SLOTOVRN_STATE_SHIFT 6 +#define M98925_SLOTOVRN_STATE_WIDTH 1 +#define M98925_INVALSLOT_STATE_MASK (1<<5) +#define M98925_INVALSLOT_STATE_SHIFT 5 +#define M98925_INVALSLOT_STATE_WIDTH 1 +#define M98925_SLOTCNFLT_STATE_MASK (1<<4) +#define M98925_SLOTCNFLT_STATE_SHIFT 4 +#define M98925_SLOTCNFLT_STATE_WIDTH 1 +#define M98925_VBSTOVFL_STATE_MASK (1<<3) +#define M98925_VBSTOVFL_STATE_SHIFT 3 +#define M98925_VBSTOVFL_STATE_WIDTH 1 +#define M98925_VBATOVFL_STATE_MASK (1<<2) +#define M98925_VBATOVFL_STATE_SHIFT 2 +#define M98925_VBATOVFL_STATE_WIDTH 1 +#define M98925_IMONOVFL_STATE_MASK (1<<1) +#define M98925_IMONOVFL_STATE_SHIFT 1 +#define M98925_IMONOVFL_STATE_WIDTH 1 +#define M98925_VMONOVFL_STATE_MASK (1<<0) +#define M98925_VMONOVFL_STATE_SHIFT 0 +#define M98925_VMONOVFL_STATE_WIDTH 1 + +/* MAX98925_R008_FLAG0 */ +#define M98925_THERMWARN_END_FLAG_MASK (1<<3) +#define M98925_THERMWARN_END_FLAG_SHIFT 3 +#define M98925_THERMWARN_END_FLAG_WIDTH 1 +#define M98925_THERMWARN_BGN_FLAG_MASK (1<<2) +#define M98925_THERMWARN_BGN_FLAG_SHIFT 2 +#define M98925_THERMWARN_BGN_FLAG_WIDTH 1 +#define M98925_THERMSHDN_END_FLAG_MASK (1<<1) +#define M98925_THERMSHDN_END_FLAG_SHIFT 1 +#define M98925_THERMSHDN_END_FLAG_WIDTH 1 +#define M98925_THERMSHDN_BGN_FLAG_MASK (1<<0) +#define M98925_THERMSHDN_BGN_FLAG_SHIFT 0 +#define M98925_THERMSHDN_BGN_FLAG_WIDTH 1 + +/* MAX98925_R009_FLAG1 */ +#define M98925_SPKCURNT_FLAG_MASK (1<<5) +#define M98925_SPKCURNT_FLAG_SHIFT 5 +#define M98925_SPKCURNT_FLAG_WIDTH 1 +#define M98925_WATCHFAIL_FLAG_MASK (1<<4) +#define M98925_WATCHFAIL_FLAG_SHIFT 4 +#define M98925_WATCHFAIL_FLAG_WIDTH 1 +#define M98925_ALCINFH_FLAG_MASK (1<<3) +#define M98925_ALCINFH_FLAG_SHIFT 3 +#define M98925_ALCINFH_FLAG_WIDTH 1 +#define M98925_ALCACT_FLAG_MASK (1<<2) +#define M98925_ALCACT_FLAG_SHIFT 2 +#define M98925_ALCACT_FLAG_WIDTH 1 +#define M98925_ALCMUT_FLAG_MASK (1<<1) +#define M98925_ALCMUT_FLAG_SHIFT 1 +#define M98925_ALCMUT_FLAG_WIDTH 1 +#define M98925_ALCP_FLAG_MASK (1<<0) +#define M98925_ALCP_FLAG_SHIFT 0 +#define M98925_ALCP_FLAG_WIDTH 1 + +/* MAX98925_R00A_FLAG2 */ +#define M98925_SLOTOVRN_FLAG_MASK (1<<6) +#define M98925_SLOTOVRN_FLAG_SHIFT 6 +#define M98925_SLOTOVRN_FLAG_WIDTH 1 +#define M98925_INVALSLOT_FLAG_MASK (1<<5) +#define M98925_INVALSLOT_FLAG_SHIFT 5 +#define M98925_INVALSLOT_FLAG_WIDTH 1 +#define M98925_SLOTCNFLT_FLAG_MASK (1<<4) +#define M98925_SLOTCNFLT_FLAG_SHIFT 4 +#define M98925_SLOTCNFLT_FLAG_WIDTH 1 +#define M98925_VBSTOVFL_FLAG_MASK (1<<3) +#define M98925_VBSTOVFL_FLAG_SHIFT 3 +#define M98925_VBSTOVFL_FLAG_WIDTH 1 +#define M98925_VBATOVFL_FLAG_MASK (1<<2) +#define M98925_VBATOVFL_FLAG_SHIFT 2 +#define M98925_VBATOVFL_FLAG_WIDTH 1 +#define M98925_IMONOVFL_FLAG_MASK (1<<1) +#define M98925_IMONOVFL_FLAG_SHIFT 1 +#define M98925_IMONOVFL_FLAG_WIDTH 1 +#define M98925_VMONOVFL_FLAG_MASK (1<<0) +#define M98925_VMONOVFL_FLAG_SHIFT 0 +#define M98925_VMONOVFL_FLAG_WIDTH 1 + +/* MAX98925_R00B_IRQ_ENABLE0 */ +#define M98925_THERMWARN_END_EN_MASK (1<<3) +#define M98925_THERMWARN_END_EN_SHIFT 3 +#define M98925_THERMWARN_END_EN_WIDTH 1 +#define M98925_THERMWARN_BGN_EN_MASK (1<<2) +#define M98925_THERMWARN_BGN_EN_SHIFT 2 +#define M98925_THERMWARN_BGN_EN_WIDTH 1 +#define M98925_THERMSHDN_END_EN_MASK (1<<1) +#define M98925_THERMSHDN_END_EN_SHIFT 1 +#define M98925_THERMSHDN_END_EN_WIDTH 1 +#define M98925_THERMSHDN_BGN_EN_MASK (1<<0) +#define M98925_THERMSHDN_BGN_EN_SHIFT 0 +#define M98925_THERMSHDN_BGN_EN_WIDTH 1 + +/* MAX98925_R00C_IRQ_ENABLE1 */ +#define M98925_SPKCURNT_EN_MASK (1<<5) +#define M98925_SPKCURNT_EN_SHIFT 5 +#define M98925_SPKCURNT_EN_WIDTH 1 +#define M98925_WATCHFAIL_EN_MASK (1<<4) +#define M98925_WATCHFAIL_EN_SHIFT 4 +#define M98925_WATCHFAIL_EN_WIDTH 1 +#define M98925_ALCINFH_EN_MASK (1<<3) +#define M98925_ALCINFH_EN_SHIFT 3 +#define M98925_ALCINFH_EN_WIDTH 1 +#define M98925_ALCACT_EN_MASK (1<<2) +#define M98925_ALCACT_EN_SHIFT 2 +#define M98925_ALCACT_EN_WIDTH 1 +#define M98925_ALCMUT_EN_MASK (1<<1) +#define M98925_ALCMUT_EN_SHIFT 1 +#define M98925_ALCMUT_EN_WIDTH 1 +#define M98925_ALCP_EN_MASK (1<<0) +#define M98925_ALCP_EN_SHIFT 0 +#define M98925_ALCP_EN_WIDTH 1 + +/* MAX98925_R00D_IRQ_ENABLE2 */ +#define M98925_SLOTOVRN_EN_MASK (1<<6) +#define M98925_SLOTOVRN_EN_SHIFT 6 +#define M98925_SLOTOVRN_EN_WIDTH 1 +#define M98925_INVALSLOT_EN_MASK (1<<5) +#define M98925_INVALSLOT_EN_SHIFT 5 +#define M98925_INVALSLOT_EN_WIDTH 1 +#define M98925_SLOTCNFLT_EN_MASK (1<<4) +#define M98925_SLOTCNFLT_EN_SHIFT 4 +#define M98925_SLOTCNFLT_EN_WIDTH 1 +#define M98925_VBSTOVFL_EN_MASK (1<<3) +#define M98925_VBSTOVFL_EN_SHIFT 3 +#define M98925_VBSTOVFL_EN_WIDTH 1 +#define M98925_VBATOVFL_EN_MASK (1<<2) +#define M98925_VBATOVFL_EN_SHIFT 2 +#define M98925_VBATOVFL_EN_WIDTH 1 +#define M98925_IMONOVFL_EN_MASK (1<<1) +#define M98925_IMONOVFL_EN_SHIFT 1 +#define M98925_IMONOVFL_EN_WIDTH 1 +#define M98925_VMONOVFL_EN_MASK (1<<0) +#define M98925_VMONOVFL_EN_SHIFT 0 +#define M98925_VMONOVFL_EN_WIDTH 1 + +/* MAX98925_R00E_IRQ_CLEAR0 */ +#define M98925_THERMWARN_END_CLR_MASK (1<<3) +#define M98925_THERMWARN_END_CLR_SHIFT 3 +#define M98925_THERMWARN_END_CLR_WIDTH 1 +#define M98925_THERMWARN_BGN_CLR_MASK (1<<2) +#define M98925_THERMWARN_BGN_CLR_SHIFT 2 +#define M98925_THERMWARN_BGN_CLR_WIDTH 1 +#define M98925_THERMSHDN_END_CLR_MASK (1<<1) +#define M98925_THERMSHDN_END_CLR_SHIFT 1 +#define M98925_THERMSHDN_END_CLR_WIDTH 1 +#define M98925_THERMSHDN_BGN_CLR_MASK (1<<0) +#define M98925_THERMSHDN_BGN_CLR_SHIFT 0 +#define M98925_THERMSHDN_BGN_CLR_WIDTH 1 + +/* MAX98925_R00F_IRQ_CLEAR1 */ +#define M98925_SPKCURNT_CLR_MASK (1<<5) +#define M98925_SPKCURNT_CLR_SHIFT 5 +#define M98925_SPKCURNT_CLR_WIDTH 1 +#define M98925_WATCHFAIL_CLR_MASK (1<<4) +#define M98925_WATCHFAIL_CLR_SHIFT 4 +#define M98925_WATCHFAIL_CLR_WIDTH 1 +#define M98925_ALCINFH_CLR_MASK (1<<3) +#define M98925_ALCINFH_CLR_SHIFT 3 +#define M98925_ALCINFH_CLR_WIDTH 1 +#define M98925_ALCACT_CLR_MASK (1<<2) +#define M98925_ALCACT_CLR_SHIFT 2 +#define M98925_ALCACT_CLR_WIDTH 1 +#define M98925_ALCMUT_CLR_MASK (1<<1) +#define M98925_ALCMUT_CLR_SHIFT 1 +#define M98925_ALCMUT_CLR_WIDTH 1 +#define M98925_ALCP_CLR_MASK (1<<0) +#define M98925_ALCP_CLR_SHIFT 0 +#define M98925_ALCP_CLR_WIDTH 1 + +/* MAX98925_R010_IRQ_CLEAR2 */ +#define M98925_SLOTOVRN_CLR_MASK (1<<6) +#define M98925_SLOTOVRN_CLR_SHIFT 6 +#define M98925_SLOTOVRN_CLR_WIDTH 1 +#define M98925_INVALSLOT_CLR_MASK (1<<5) +#define M98925_INVALSLOT_CLR_SHIFT 5 +#define M98925_INVALSLOT_CLR_WIDTH 1 +#define M98925_SLOTCNFLT_CLR_MASK (1<<4) +#define M98925_SLOTCNFLT_CLR_SHIFT 4 +#define M98925_SLOTCNFLT_CLR_WIDTH 1 +#define M98925_VBSTOVFL_CLR_MASK (1<<3) +#define M98925_VBSTOVFL_CLR_SHIFT 3 +#define M98925_VBSTOVFL_CLR_WIDTH 1 +#define M98925_VBATOVFL_CLR_MASK (1<<2) +#define M98925_VBATOVFL_CLR_SHIFT 2 +#define M98925_VBATOVFL_CLR_WIDTH 1 +#define M98925_IMONOVFL_CLR_MASK (1<<1) +#define M98925_IMONOVFL_CLR_SHIFT 1 +#define M98925_IMONOVFL_CLR_WIDTH 1 +#define M98925_VMONOVFL_CLR_MASK (1<<0) +#define M98925_VMONOVFL_CLR_SHIFT 0 +#define M98925_VMONOVFL_CLR_WIDTH 1 + +/* MAX98925_R011_MAP0 */ +#define M98925_ER_THERMWARN_EN_MASK (1<<7) +#define M98925_ER_THERMWARN_EN_SHIFT 7 +#define M98925_ER_THERMWARN_EN_WIDTH 1 +#define M98925_ER_THERMWARN_MAP_MASK (0x07<<4) +#define M98925_ER_THERMWARN_MAP_SHIFT 4 +#define M98925_ER_THERMWARN_MAP_WIDTH 3 + +/* MAX98925_R012_MAP1 */ +#define M98925_ER_ALCMUT_EN_MASK (1<<7) +#define M98925_ER_ALCMUT_EN_SHIFT 7 +#define M98925_ER_ALCMUT_EN_WIDTH 1 +#define M98925_ER_ALCMUT_MAP_MASK (0x07<<4) +#define M98925_ER_ALCMUT_MAP_SHIFT 4 +#define M98925_ER_ALCMUT_MAP_WIDTH 3 +#define M98925_ER_ALCP_EN_MASK (1<<3) +#define M98925_ER_ALCP_EN_SHIFT 3 +#define M98925_ER_ALCP_EN_WIDTH 1 +#define M98925_ER_ALCP_MAP_MASK (0x07<<0) +#define M98925_ER_ALCP_MAP_SHIFT 0 +#define M98925_ER_ALCP_MAP_WIDTH 3 + +/* MAX98925_R013_MAP2 */ +#define M98925_ER_ALCINFH_EN_MASK (1<<7) +#define M98925_ER_ALCINFH_EN_SHIFT 7 +#define M98925_ER_ALCINFH_EN_WIDTH 1 +#define M98925_ER_ALCINFH_MAP_MASK (0x07<<4) +#define M98925_ER_ALCINFH_MAP_SHIFT 4 +#define M98925_ER_ALCINFH_MAP_WIDTH 3 +#define M98925_ER_ALCACT_EN_MASK (1<<3) +#define M98925_ER_ALCACT_EN_SHIFT 3 +#define M98925_ER_ALCACT_EN_WIDTH 1 +#define M98925_ER_ALCACT_MAP_MASK (0x07<<0) +#define M98925_ER_ALCACT_MAP_SHIFT 0 +#define M98925_ER_ALCACT_MAP_WIDTH 3 + +/* MAX98925_R014_MAP3 */ +#define M98925_ER_SPKCURNT_EN_MASK (1<<7) +#define M98925_ER_SPKCURNT_EN_SHIFT 7 +#define M98925_ER_SPKCURNT_EN_WIDTH 1 +#define M98925_ER_SPKCURNT_MAP_MASK (0x07<<4) +#define M98925_ER_SPKCURNT_MAP_SHIFT 4 +#define M98925_ER_SPKCURNT_MAP_WIDTH 3 + +/* MAX98925_R015_MAP4 */ +/* RESERVED */ + +/* MAX98925_R016_MAP5 */ +#define M98925_ER_IMONOVFL_EN_MASK (1<<7) +#define M98925_ER_IMONOVFL_EN_SHIFT 7 +#define M98925_ER_IMONOVFL_EN_WIDTH 1 +#define M98925_ER_IMONOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_IMONOVFL_MAP_SHIFT 4 +#define M98925_ER_IMONOVFL_MAP_WIDTH 3 +#define M98925_ER_VMONOVFL_EN_MASK (1<<3) +#define M98925_ER_VMONOVFL_EN_SHIFT 3 +#define M98925_ER_VMONOVFL_EN_WIDTH 1 +#define M98925_ER_VMONOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VMONOVFL_MAP_SHIFT 0 +#define M98925_ER_VMONOVFL_MAP_WIDTH 3 + +/* MAX98925_R017_MAP6 */ +#define M98925_ER_VBSTOVFL_EN_MASK (1<<7) +#define M98925_ER_VBSTOVFL_EN_SHIFT 7 +#define M98925_ER_VBSTOVFL_EN_WIDTH 1 +#define M98925_ER_VBSTOVFL_MAP_MASK (0x07<<4) +#define M98925_ER_VBSTOVFL_MAP_SHIFT 4 +#define M98925_ER_VBSTOVFL_MAP_WIDTH 3 +#define M98925_ER_VBATOVFL_EN_MASK (1<<3) +#define M98925_ER_VBATOVFL_EN_SHIFT 3 +#define M98925_ER_VBATOVFL_EN_WIDTH 1 +#define M98925_ER_VBATOVFL_MAP_MASK (0x07<<0) +#define M98925_ER_VBATOVFL_MAP_SHIFT 0 +#define M98925_ER_VBATOVFL_MAP_WIDTH 3 + +/* MAX98925_R018_MAP7 */ +#define M98925_ER_INVALSLOT_EN_MASK (1<<7) +#define M98925_ER_INVALSLOT_EN_SHIFT 7 +#define M98925_ER_INVALSLOT_EN_WIDTH 1 +#define M98925_ER_INVALSLOT_MAP_MASK (0x07<<4) +#define M98925_ER_INVALSLOT_MAP_SHIFT 4 +#define M98925_ER_INVALSLOT_MAP_WIDTH 3 +#define M98925_ER_SLOTCNFLT_EN_MASK (1<<3) +#define M98925_ER_SLOTCNFLT_EN_SHIFT 3 +#define M98925_ER_SLOTCNFLT_EN_WIDTH 1 +#define M98925_ER_SLOTCNFLT_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTCNFLT_MAP_SHIFT 0 +#define M98925_ER_SLOTCNFLT_MAP_WIDTH 3 + +/* MAX98925_R019_MAP8 */ +#define M98925_ER_SLOTOVRN_EN_MASK (1<<3) +#define M98925_ER_SLOTOVRN_EN_SHIFT 3 +#define M98925_ER_SLOTOVRN_EN_WIDTH 1 +#define M98925_ER_SLOTOVRN_MAP_MASK (0x07<<0) +#define M98925_ER_SLOTOVRN_MAP_SHIFT 0 +#define M98925_ER_SLOTOVRN_MAP_WIDTH 3 + +/* MAX98925_R01A_DAI_CLK_MODE1 */ +#define M98925_DAI_CLK_SOURCE_MASK (1<<6) +#define M98925_DAI_CLK_SOURCE_SHIFT 6 +#define M98925_DAI_CLK_SOURCE_WIDTH 1 +#define M98925_MDLL_MULT_MASK (0x0F<<0) +#define M98925_MDLL_MULT_SHIFT 0 +#define M98925_MDLL_MULT_WIDTH 4 + +#define M98925_MDLL_MULT_MCLKx8 6 +#define M98925_MDLL_MULT_MCLKx16 8 + +/* MAX98925_R01B_DAI_CLK_MODE2 */ +#define M98925_DAI_SR_MASK (0x0F<<4) +#define M98925_DAI_SR_SHIFT 4 +#define M98925_DAI_SR_WIDTH 4 +#define M98925_DAI_MAS_MASK (1<<3) +#define M98925_DAI_MAS_SHIFT 3 +#define M98925_DAI_MAS_WIDTH 1 +#define M98925_DAI_BSEL_MASK (0x07<<0) +#define M98925_DAI_BSEL_SHIFT 0 +#define M98925_DAI_BSEL_WIDTH 3 + +#define M98925_DAI_BSEL_32 (0 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_48 (1 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_64 (2 << M98925_DAI_BSEL_SHIFT) +#define M98925_DAI_BSEL_256 (6 << M98925_DAI_BSEL_SHIFT) + +/* MAX98925_R01C_DAI_CLK_DIV_M_MSBS */ +#define M98925_DAI_M_MSBS_MASK (0xFF<<0) +#define M98925_DAI_M_MSBS_SHIFT 0 +#define M98925_DAI_M_MSBS_WIDTH 8 + +/* MAX98925_R01D_DAI_CLK_DIV_M_LSBS */ +#define M98925_DAI_M_LSBS_MASK (0xFF<<0) +#define M98925_DAI_M_LSBS_SHIFT 0 +#define M98925_DAI_M_LSBS_WIDTH 8 + +/* MAX98925_R01E_DAI_CLK_DIV_N_MSBS */ +#define M98925_DAI_N_MSBS_MASK (0x7F<<0) +#define M98925_DAI_N_MSBS_SHIFT 0 +#define M98925_DAI_N_MSBS_WIDTH 7 + +/* MAX98925_R01F_DAI_CLK_DIV_N_LSBS */ +#define M98925_DAI_N_LSBS_MASK (0xFF<<0) +#define M98925_DAI_N_LSBS_SHIFT 0 +#define M98925_DAI_N_LSBS_WIDTH 8 + +/* MAX98925_R020_FORMAT */ +#define M98925_DAI_CHANSZ_MASK (0x03<<6) +#define M98925_DAI_CHANSZ_SHIFT 6 +#define M98925_DAI_CHANSZ_WIDTH 2 +#define M98925_DAI_EXTBCLK_HIZ_MASK (1<<4) +#define M98925_DAI_EXTBCLK_HIZ_SHIFT 4 +#define M98925_DAI_EXTBCLK_HIZ_WIDTH 1 +#define M98925_DAI_WCI_MASK (1<<3) +#define M98925_DAI_WCI_SHIFT 3 +#define M98925_DAI_WCI_WIDTH 1 +#define M98925_DAI_BCI_MASK (1<<2) +#define M98925_DAI_BCI_SHIFT 2 +#define M98925_DAI_BCI_WIDTH 1 +#define M98925_DAI_DLY_MASK (1<<1) +#define M98925_DAI_DLY_SHIFT 1 +#define M98925_DAI_DLY_WIDTH 1 +#define M98925_DAI_TDM_MASK (1<<0) +#define M98925_DAI_TDM_SHIFT 0 +#define M98925_DAI_TDM_WIDTH 1 + +#define M98925_DAI_CHANSZ_16 (1 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_24 (2 << M98925_DAI_CHANSZ_SHIFT) +#define M98925_DAI_CHANSZ_32 (3 << M98925_DAI_CHANSZ_SHIFT) + +/* MAX98925_R021_TDM_SLOT_SELECT */ +#define M98925_DAI_DO_EN_MASK (1<<7) +#define M98925_DAI_DO_EN_SHIFT 7 +#define M98925_DAI_DO_EN_WIDTH 1 +#define M98925_DAI_DIN_EN_MASK (1<<6) +#define M98925_DAI_DIN_EN_SHIFT 6 +#define M98925_DAI_DIN_EN_WIDTH 1 +#define M98925_DAI_INR_SOURCE_MASK (0x07<<3) +#define M98925_DAI_INR_SOURCE_SHIFT 3 +#define M98925_DAI_INR_SOURCE_WIDTH 3 +#define M98925_DAI_INL_SOURCE_MASK (0x07<<0) +#define M98925_DAI_INL_SOURCE_SHIFT 0 +#define M98925_DAI_INL_SOURCE_WIDTH 3 + +/* MAX98925_R022_DOUT_CFG_VMON */ +#define M98925_DAI_VMON_EN_MASK (1<<5) +#define M98925_DAI_VMON_EN_SHIFT 5 +#define M98925_DAI_VMON_EN_WIDTH 1 +#define M98925_DAI_VMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VMON_SLOT_SHIFT 0 +#define M98925_DAI_VMON_SLOT_WIDTH 5 + +#define M98925_DAI_VMON_SLOT_00_01 (0 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_01_02 (1 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_02_03 (2 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_03_04 (3 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_04_05 (4 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_05_06 (5 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_06_07 (6 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_07_08 (7 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_08_09 (8 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_09_0A (9 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0A_0B (10 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0B_0C (11 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0C_0D (12 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0D_0E (13 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0E_0F (14 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_0F_10 (15 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_10_11 (16 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_11_12 (17 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_12_13 (18 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_13_14 (19 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_14_15 (20 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_15_16 (21 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_16_17 (22 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_17_18 (23 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_18_19 (24 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_19_1A (25 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1A_1B (26 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1B_1C (27 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1C_1D (28 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1D_1E (29 << M98925_DAI_VMON_SLOT_SHIFT) +#define M98925_DAI_VMON_SLOT_1E_1F (30 << M98925_DAI_VMON_SLOT_SHIFT) + +/* MAX98925_R023_DOUT_CFG_IMON */ +#define M98925_DAI_IMON_EN_MASK (1<<5) +#define M98925_DAI_IMON_EN_SHIFT 5 +#define M98925_DAI_IMON_EN_WIDTH 1 +#define M98925_DAI_IMON_SLOT_MASK (0x1F<<0) +#define M98925_DAI_IMON_SLOT_SHIFT 0 +#define M98925_DAI_IMON_SLOT_WIDTH 5 + +#define M98925_DAI_IMON_SLOT_00_01 (0 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_01_02 (1 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_02_03 (2 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_03_04 (3 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_04_05 (4 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_05_06 (5 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_06_07 (6 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_07_08 (7 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_08_09 (8 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_09_0A (9 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0A_0B (10 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0B_0C (11 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0C_0D (12 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0D_0E (13 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0E_0F (14 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_0F_10 (15 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_10_11 (16 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_11_12 (17 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_12_13 (18 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_13_14 (19 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_14_15 (20 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_15_16 (21 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_16_17 (22 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_17_18 (23 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_18_19 (24 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_19_1A (25 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1A_1B (26 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1B_1C (27 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1C_1D (28 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1D_1E (29 << M98925_DAI_IMON_SLOT_SHIFT) +#define M98925_DAI_IMON_SLOT_1E_1F (30 << M98925_DAI_IMON_SLOT_SHIFT) + +/* MAX98925_R024_DOUT_CFG_VBAT */ +#define M98925_DAI_VBAT_EN_MASK (1<<5) +#define M98925_DAI_VBAT_EN_SHIFT 5 +#define M98925_DAI_VBAT_EN_WIDTH 1 +#define M98925_DAI_VBAT_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBAT_SLOT_SHIFT 0 +#define M98925_DAI_VBAT_SLOT_WIDTH 5 + +/* MAX98925_R025_DOUT_CFG_VBST */ +#define M98925_DAI_VBST_EN_MASK (1<<5) +#define M98925_DAI_VBST_EN_SHIFT 5 +#define M98925_DAI_VBST_EN_WIDTH 1 +#define M98925_DAI_VBST_SLOT_MASK (0x1F<<0) +#define M98925_DAI_VBST_SLOT_SHIFT 0 +#define M98925_DAI_VBST_SLOT_WIDTH 5 + +/* MAX98925_R026_DOUT_CFG_FLAG */ +#define M98925_DAI_FLAG_EN_MASK (1<<5) +#define M98925_DAI_FLAG_EN_SHIFT 5 +#define M98925_DAI_FLAG_EN_WIDTH 1 +#define M98925_DAI_FLAG_SLOT_MASK (0x1F<<0) +#define M98925_DAI_FLAG_SLOT_SHIFT 0 +#define M98925_DAI_FLAG_SLOT_WIDTH 5 + +/* MAX98925_R027_DOUT_HIZ_CFG1 */ +#define M98925_DAI_SLOT_HIZ_CFG1_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG1_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG1_WIDTH 8 + +/* MAX98925_R028_DOUT_HIZ_CFG2 */ +#define M98925_DAI_SLOT_HIZ_CFG2_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG2_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG2_WIDTH 8 + +/* MAX98925_R029_DOUT_HIZ_CFG3 */ +#define M98925_DAI_SLOT_HIZ_CFG3_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG3_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG3_WIDTH 8 + +/* MAX98925_R02A_DOUT_HIZ_CFG4 */ +#define M98925_DAI_SLOT_HIZ_CFG4_MASK (0xFF<<0) +#define M98925_DAI_SLOT_HIZ_CFG4_SHIFT 0 +#define M98925_DAI_SLOT_HIZ_CFG4_WIDTH 8 + +/* MAX98925_R02B_DOUT_DRV_STRENGTH */ +#define M98925_DAI_OUT_DRIVE_MASK (0x03<<0) +#define M98925_DAI_OUT_DRIVE_SHIFT 0 +#define M98925_DAI_OUT_DRIVE_WIDTH 2 + +/* MAX98925_R02C_FILTERS */ +#define M98925_ADC_DITHER_EN_MASK (1<<7) +#define M98925_ADC_DITHER_EN_SHIFT 7 +#define M98925_ADC_DITHER_EN_WIDTH 1 +#define M98925_IV_DCB_EN_MASK (1<<6) +#define M98925_IV_DCB_EN_SHIFT 6 +#define M98925_IV_DCB_EN_WIDTH 1 +#define M98925_DAC_DITHER_EN_MASK (1<<4) +#define M98925_DAC_DITHER_EN_SHIFT 4 +#define M98925_DAC_DITHER_EN_WIDTH 1 +#define M98925_DAC_FILTER_MODE_MASK (1<<3) +#define M98925_DAC_FILTER_MODE_SHIFT 3 +#define M98925_DAC_FILTER_MODE_WIDTH 1 +#define M98925_DAC_HPF_MASK (0x07<<0) +#define M98925_DAC_HPF_SHIFT 0 +#define M98925_DAC_HPF_WIDTH 3 + +/* MAX98925_R02D_GAIN */ +#define M98925_DAC_IN_SEL_MASK (0x03<<5) +#define M98925_DAC_IN_SEL_SHIFT 5 +#define M98925_DAC_IN_SEL_WIDTH 2 +#define M98925_SPK_GAIN_MASK (0x1F<<0) +#define M98925_SPK_GAIN_SHIFT 0 +#define M98925_SPK_GAIN_WIDTH 5 + +#define M98925_DAC_IN_SEL_LEFT_DAI (0 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_RIGHT_DAI (1 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_SUMMED_DAI (2 << M98925_DAC_IN_SEL_SHIFT) +#define M98925_DAC_IN_SEL_DIV2_SUMMED_DAI (3 << M98925_DAC_IN_SEL_SHIFT) + +/* MAX98925_R02E_GAIN_RAMPING */ +#define M98925_SPK_RMP_EN_MASK (1<<1) +#define M98925_SPK_RMP_EN_SHIFT 1 +#define M98925_SPK_RMP_EN_WIDTH 1 +#define M98925_SPK_ZCD_EN_MASK (1<<0) +#define M98925_SPK_ZCD_EN_SHIFT 0 +#define M98925_SPK_ZCD_EN_WIDTH 1 + +/* MAX98925_R02F_SPK_AMP */ +#define M98925_SPK_MODE_MASK (1<<0) +#define M98925_SPK_MODE_SHIFT 0 +#define M98925_SPK_MODE_WIDTH 1 + +/* MAX98925_R030_THRESHOLD */ +#define M98925_ALC_EN_MASK (1<<5) +#define M98925_ALC_EN_SHIFT 5 +#define M98925_ALC_EN_WIDTH 1 +#define M98925_ALC_TH_MASK (0x1F<<0) +#define M98925_ALC_TH_SHIFT 0 +#define M98925_ALC_TH_WIDTH 5 + +/* MAX98925_R031_ALC_ATTACK */ +#define M98925_ALC_ATK_STEP_MASK (0x0F<<4) +#define M98925_ALC_ATK_STEP_SHIFT 4 +#define M98925_ALC_ATK_STEP_WIDTH 4 +#define M98925_ALC_ATK_RATE_MASK (0x7<<0) +#define M98925_ALC_ATK_RATE_SHIFT 0 +#define M98925_ALC_ATK_RATE_WIDTH 3 + +/* MAX98925_R032_ALC_ATTEN_RLS */ +#define M98925_ALC_MAX_ATTEN_MASK (0x0F<<4) +#define M98925_ALC_MAX_ATTEN_SHIFT 4 +#define M98925_ALC_MAX_ATTEN_WIDTH 4 +#define M98925_ALC_RLS_RATE_MASK (0x7<<0) +#define M98925_ALC_RLS_RATE_SHIFT 0 +#define M98925_ALC_RLS_RATE_WIDTH 3 + +/* MAX98925_R033_ALC_HOLD_RLS */ +#define M98925_ALC_RLS_TGR_MASK (1<<0) +#define M98925_ALC_RLS_TGR_SHIFT 0 +#define M98925_ALC_RLS_TGR_WIDTH 1 + +/* MAX98925_R034_ALC_CONFIGURATION */ +#define M98925_ALC_MUTE_EN_MASK (1<<7) +#define M98925_ALC_MUTE_EN_SHIFT 7 +#define M98925_ALC_MUTE_EN_WIDTH 1 +#define M98925_ALC_MUTE_DLY_MASK (0x07<<4) +#define M98925_ALC_MUTE_DLY_SHIFT 4 +#define M98925_ALC_MUTE_DLY_WIDTH 3 +#define M98925_ALC_RLS_DBT_MASK (0x07<<0) +#define M98925_ALC_RLS_DBT_SHIFT 0 +#define M98925_ALC_RLS_DBT_WIDTH 3 + +/* MAX98925_R035_BOOST_CONVERTER */ +#define M98925_BST_SYNC_MASK (1<<7) +#define M98925_BST_SYNC_SHIFT 7 +#define M98925_BST_SYNC_WIDTH 1 +#define M98925_BST_PHASE_MASK (0x03<<4) +#define M98925_BST_PHASE_SHIFT 4 +#define M98925_BST_PHASE_WIDTH 2 +#define M98925_BST_SKIP_MODE_MASK (0x03<<0) +#define M98925_BST_SKIP_MODE_SHIFT 0 +#define M98925_BST_SKIP_MODE_WIDTH 2 + +/* MAX98925_R036_BLOCK_ENABLE */ +#define M98925_BST_EN_MASK (1<<7) +#define M98925_BST_EN_SHIFT 7 +#define M98925_BST_EN_WIDTH 1 +#define M98925_WATCH_EN_MASK (1<<6) +#define M98925_WATCH_EN_SHIFT 6 +#define M98925_WATCH_EN_WIDTH 1 +#define M98925_CLKMON_EN_MASK (1<<5) +#define M98925_CLKMON_EN_SHIFT 5 +#define M98925_CLKMON_EN_WIDTH 1 +#define M98925_SPK_EN_MASK (1<<4) +#define M98925_SPK_EN_SHIFT 4 +#define M98925_SPK_EN_WIDTH 1 +#define M98925_ADC_VBST_EN_MASK (1<<3) +#define M98925_ADC_VBST_EN_SHIFT 3 +#define M98925_ADC_VBST_EN_WIDTH 1 +#define M98925_ADC_VBAT_EN_MASK (1<<2) +#define M98925_ADC_VBAT_EN_SHIFT 2 +#define M98925_ADC_VBAT_EN_WIDTH 1 +#define M98925_ADC_IMON_EN_MASK (1<<1) +#define M98925_ADC_IMON_EN_SHIFT 1 +#define M98925_ADC_IMON_EN_WIDTH 1 +#define M98925_ADC_VMON_EN_MASK (1<<0) +#define M98925_ADC_VMON_EN_SHIFT 0 +#define M98925_ADC_VMON_EN_WIDTH 1 + +/* MAX98925_R037_CONFIGURATION */ +#define M98925_BST_VOUT_MASK (0x0F<<4) +#define M98925_BST_VOUT_SHIFT 4 +#define M98925_BST_VOUT_WIDTH 4 +#define M98925_THERMWARN_LEVEL_MASK (0x03<<2) +#define M98925_THERMWARN_LEVEL_SHIFT 2 +#define M98925_THERMWARN_LEVEL_WIDTH 2 +#define M98925_WATCH_TIME_MASK (0x03<<0) +#define M98925_WATCH_TIME_SHIFT 0 +#define M98925_WATCH_TIME_WIDTH 2 + +/* MAX98925_R038_GLOBAL_ENABLE */ +#define M98925_EN_MASK (1<<7) +#define M98925_EN_SHIFT 7 +#define M98925_EN_WIDTH 1 + +/* MAX98925_R03A_BOOST_LIMITER */ +#define M98925_BST_ILIM_MASK (0x1F<<3) +#define M98925_BST_ILIM_SHIFT 3 +#define M98925_BST_ILIM_WIDTH 5 + +/* MAX98925_R0FF_VERSION */ +#define M98925_REV_ID_MASK (0xFF<<0) +#define M98925_REV_ID_SHIFT 0 +#define M98925_REV_ID_WIDTH 8 + +enum max98925_type { + MAX98925, +}; + +struct max98925_cdata { + unsigned int rate; + unsigned int fmt; +}; + +struct max98925_priv { + struct regmap *regmapL; + struct regmap *regmapR; + struct snd_soc_codec *codec; + enum max98925_type devtype; + void *control_data; + struct max98925_pdata *pdata; + unsigned int spk_gain; + unsigned int sysclk; + unsigned int iv_status; + struct max98925_cdata dai[1]; +}; + +#endif -- 1.9.3

6 8

[alsa-devel] [PATCH 1/2] ALSA: firewire-lib: remove rx_blocks_for_midi quirk
by Clemens Ladisch 16 Jan '15

16 Jan '15

There are several devices that expect to receive MIDI data only in the first eight data blocks of a packet. If the driver restricts the data rate to the allowed rate (as mandated by the specification, but not yet implemented by this driver), this happens naturally. Therefore, there is no reason to ever try to use more data packets with any device. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/firewire/amdtp.c | 10 +++++++--- sound/firewire/amdtp.h | 3 --- sound/firewire/bebob/bebob_stream.c | 7 ------- sound/firewire/fireworks/fireworks_stream.c | 5 ----- 4 files changed, 7 insertions(+), 18 deletions(-) diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index 3badc70..17548d8 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -21,6 +21,12 @@ #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) +/* + * Several devices look only at the first eight data blocks. + * In any case, this is more than enough for the MIDI data rate. + */ +#define MAX_MIDI_RX_BLOCKS 8 + #define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 Âµs */ /* isochronous header parameters */ @@ -78,8 +84,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->callbacked = false; s->sync_slave = NULL; - s->rx_blocks_for_midi = UINT_MAX; - return 0; } EXPORT_SYMBOL(amdtp_stream_init); @@ -474,7 +478,7 @@ static void amdtp_fill_midi(struct amdtp_stream *s, b = (u8 *)&buffer[s->midi_position]; port = (s->data_block_counter + f) % 8; - if ((f >= s->rx_blocks_for_midi) || + if ((f >= MAX_MIDI_RX_BLOCKS) || (s->midi[port] == NULL) || (snd_rawmidi_transmit(s->midi[port], b + 1, 1) <= 0)) b[0] = 0x80; diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp.h index e6e8926..cd4c4df 100644 --- a/sound/firewire/amdtp.h +++ b/sound/firewire/amdtp.h @@ -152,9 +152,6 @@ struct amdtp_stream { /* quirk: fixed interval of dbc between previos/current packets. */ unsigned int tx_dbc_interval; - /* quirk: the first count of data blocks in an rx packet for MIDI */ - unsigned int rx_blocks_for_midi; - bool callbacked; wait_queue_head_t callback_wait; struct amdtp_stream *sync_slave; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 1aab0a32..0ebcabf 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -484,13 +484,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) amdtp_stream_destroy(&bebob->rx_stream); destroy_both_connections(bebob); } - /* - * The firmware for these devices ignore MIDI messages in more than - * first 8 data blocks of an received AMDTP packet. - */ - if (bebob->spec == &maudio_fw410_spec || - bebob->spec == &maudio_special_spec) - bebob->rx_stream.rx_blocks_for_midi = 8; end: return err; } diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index b985fc5..4f440e1 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -179,11 +179,6 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) destroy_stream(efw, &efw->tx_stream); goto end; } - /* - * Fireworks ignores MIDI messages in more than first 8 data - * blocks of an received AMDTP packet. - */ - efw->rx_stream.rx_blocks_for_midi = 8; /* set IEC61883 compliant mode (actually not fully compliant...) */ err = snd_efw_command_set_tx_mode(efw, SND_EFW_TRANSPORT_MODE_IEC61883);

3 9

[alsa-devel] [PATCH 0/4] ak411x fix patches for ice1724 Juli@
by Takashi Iwai 16 Jan '15

16 Jan '15

Hi, here is a bit more cooked patch series for ak411x and juli@ device. [PATCH 1/4] ALSA: ak411x: Fix stall in work callback [PATCH 2/4] ALSA: ak4114: Move EXPORT_SYMBOL() after each function [PATCH 3/4] ALSA: ak411x: Add PM helper functions [PATCH 4/4] ALSA: ak411x: Fix race of reinit() calls Takashi

2 6

[alsa-devel] [PATCH] ASoC: samsung: Replace depends on REGMAP_I2C with depends on I2C
by Lars-Peter Clausen 16 Jan '15

16 Jan '15

regmap is a library function that gets selected by drivers that need it. No driver modules should depend on it, since this can create dependency loops. E.g. if machine driver depends on REGMAP_I2C and selects the CODEC driver and the CODEC driver selects REGMAP_I2C, then the machine driver selects by extension one of its dependencies. The proper way to handle this is for machine drivers to depend on the same dependencies as the CODEC driver they select. In this case it is I2C. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fc67f97..e817a2f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -54,7 +54,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) - depends on REGMAP_I2C + depends on I2C select SND_SOC_WM8580 select SND_SAMSUNG_I2S help @@ -167,7 +167,7 @@ config SND_SOC_SAMSUNG_SMDK_SPDIF config SND_SOC_SMDK_WM8580_PCM tristate "SoC PCM Audio support for WM8580 on SMDK" depends on SND_SOC_SAMSUNG && (MACH_SMDKV210 || MACH_SMDKC110) - depends on REGMAP_I2C + depends on I2C select SND_SOC_WM8580 select SND_SAMSUNG_PCM help -- 1.8.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: jive: Automatically disconnect non-connected pins
by Lars-Peter Clausen 16 Jan '15

16 Jan '15

All DAPM input and output pins of the wm8750 are either used in the card's DAPM routing table or are marked as not connected. Set the fully_routed flag of the card instead of manually marking the unused inputs and outputs as not connected. This makes the code a bit shorter and cleaner. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/jive_wm8750.c | 18 +----------------- 1 file changed, 1 insertion(+), 17 deletions(-) diff --git a/sound/soc/samsung/jive_wm8750.c b/sound/soc/samsung/jive_wm8750.c index 6c3b359..7fcb51f 100644 --- a/sound/soc/samsung/jive_wm8750.c +++ b/sound/soc/samsung/jive_wm8750.c @@ -83,22 +83,6 @@ static struct snd_soc_ops jive_ops = { .hw_params = jive_hw_params, }; -static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* These endpoints are not being used. */ - snd_soc_dapm_nc_pin(dapm, "LINPUT2"); - snd_soc_dapm_nc_pin(dapm, "RINPUT2"); - snd_soc_dapm_nc_pin(dapm, "LINPUT3"); - snd_soc_dapm_nc_pin(dapm, "RINPUT3"); - snd_soc_dapm_nc_pin(dapm, "OUT3"); - snd_soc_dapm_nc_pin(dapm, "MONO"); - - return 0; -} - static struct snd_soc_dai_link jive_dai = { .name = "wm8750", .stream_name = "WM8750", @@ -106,7 +90,6 @@ static struct snd_soc_dai_link jive_dai = { .codec_dai_name = "wm8750-hifi", .platform_name = "s3c2412-i2s", .codec_name = "wm8750.0-001a", - .init = jive_wm8750_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ops = &jive_ops, @@ -123,6 +106,7 @@ static struct snd_soc_card snd_soc_machine_jive = { .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), + .fully_routed = true, }; static struct platform_device *jive_snd_device; -- 1.8.0

2 3

[alsa-devel] [PATCH v8] ASoC: dapm: add code to configure dai link parameters
by Nikesh Oswal 16 Jan '15

16 Jan '15

dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 3 + include/sound/soc.h | 1 + sound/soc/soc-core.c | 6 +- sound/soc/soc-dapm.c | 155 ++++++++++++++++++++++++++++++++++++++++++++-- 4 files changed, 157 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5c0a798..a2098e4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -378,6 +378,7 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card); void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card); int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -531,6 +532,8 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ const struct snd_soc_pcm_stream *params; /* params for dai links */ + unsigned int num_params; /* number of params for dai links */ + unsigned int params_select; /* currently selected param for dai link */ /* dapm control */ int reg; /* negative reg = no direct dapm */ diff --git a/include/sound/soc.h b/include/sound/soc.h index edd4a0a..7556885 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -940,6 +940,7 @@ struct snd_soc_dai_link { int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; + unsigned int num_params; unsigned int dai_fmt; /* format to set on init */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ededb97..893a6e5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1230,7 +1230,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = cpu_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); @@ -1242,7 +1243,8 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, capture_w = codec_dai->capture_widget; if (play_w && capture_w) { ret = snd_soc_dapm_new_pcm(card, dai_link->params, - capture_w, play_w); + dai_link->num_params, capture_w, + play_w); if (ret != 0) { dev_err(card->dev, "ASoC: Can't link %s to %s: %d\n", play_w->name, capture_w->name, ret); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bc2d7a0..286413b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -853,6 +853,36 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } +/* create new dapm dai link control */ +static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) +{ + int i, ret; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_card *card = dapm->card->snd_card; + + /* create control for links with > 1 config */ + if (w->num_params <= 1) + return 0; + + /* add kcontrol */ + for (i = 0; i < w->num_kcontrols; i++) { + kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, + w->name, NULL); + ret = snd_ctl_add(card, kcontrol); + if (ret < 0) { + dev_err(dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[i].name, ret); + return ret; + } + kcontrol->private_data = w; + w->kcontrols[i] = kcontrol; + } + + return 0; +} + /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. @@ -2724,6 +2754,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_out_drv: dapm_new_pga(w); break; + case snd_soc_dapm_dai_link: + dapm_new_dai_link(w); + break; default: break; } @@ -3198,7 +3231,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, { struct snd_soc_dapm_path *source_p, *sink_p; struct snd_soc_dai *source, *sink; - const struct snd_soc_pcm_stream *config = w->params; + const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; u64 fmt; @@ -3290,8 +3323,39 @@ out: return ret; } +static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = w->params_select; + + return 0; +} + +static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); + + /* Can't change the config when widget is already powered */ + if (w->power) + return -EBUSY; + + if (ucontrol->value.integer.value[0] == w->params_select) + return 0; + + if (ucontrol->value.integer.value[0] >= w->num_params) + return -EINVAL; + + w->params_select = ucontrol->value.integer.value[0]; + + return 0; +} + int snd_soc_dapm_new_pcm(struct snd_soc_card *card, const struct snd_soc_pcm_stream *params, + unsigned int num_params, struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { @@ -3299,14 +3363,52 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *w; size_t len; char *link_name; - int ret; + int ret, count; + unsigned long private_value; + const char **w_param_text; + struct soc_enum w_param_enum[] = { + SOC_ENUM_SINGLE(0, 0, 0, NULL), + }; + struct snd_kcontrol_new kcontrol_dai_link[] = { + SOC_ENUM_EXT(NULL, w_param_enum[0], + snd_soc_dapm_dai_link_get, + snd_soc_dapm_dai_link_put), + }; + const struct snd_soc_pcm_stream *config = params; + + w_param_text = devm_kcalloc(card->dev, num_params, + sizeof(char *), GFP_KERNEL); + if (!w_param_text) + return -ENOMEM; len = strlen(source->name) + strlen(sink->name) + 2; link_name = devm_kzalloc(card->dev, len, GFP_KERNEL); - if (!link_name) - return -ENOMEM; + if (!link_name) { + ret = -ENOMEM; + goto outfree_w_param; + } snprintf(link_name, len, "%s-%s", source->name, sink->name); + for (count = 0 ; count < num_params; count++) { + if (!config->stream_name) { + dev_warn(card->dapm.dev, + "ASoC: anonymous config %d for dai link %s\n", + count, link_name); + } else { + w_param_text[count] = devm_kmemdup(card->dev, + config->stream_name, + strlen(config->stream_name) + 1, + GFP_KERNEL); + if (!w_param_text[count]) { + ret = -ENOMEM; + goto outfree_link_name; + } + } + config++; + } + w_param_enum[0].items = num_params; + w_param_enum[0].texts = w_param_text; + memset(&template, 0, sizeof(template)); template.reg = SND_SOC_NOPM; template.id = snd_soc_dapm_dai_link; @@ -3314,6 +3416,30 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, template.event = snd_soc_dai_link_event; template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD; + template.num_kcontrols = 1; + /* duplicate w_param_enum on heap so that memory persists */ + private_value = + (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_dai_link[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_link_name; + } + kcontrol_dai_link[0].private_value = private_value; + /* duplicate kcontrol_dai_link on heap so that memory persists */ + template.kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!template.kcontrol_news) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + link_name); + ret = -ENOMEM; + goto outfree_private_value; + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); @@ -3321,15 +3447,32 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!w) { dev_err(card->dev, "ASoC: Failed to create %s widget\n", link_name); - return -ENOMEM; + ret = -ENOMEM; + goto outfree_kcontrol_news; } w->params = params; + w->num_params = num_params; ret = snd_soc_dapm_add_path(&card->dapm, source, w, NULL, NULL); if (ret) - return ret; + goto outfree_w; return snd_soc_dapm_add_path(&card->dapm, w, sink, NULL, NULL); + +outfree_w: + devm_kfree(card->dev, w); +outfree_kcontrol_news: + devm_kfree(card->dev, (void *)template.kcontrol_news); +outfree_private_value: + devm_kfree(card->dev, (void *)private_value); +outfree_link_name: + devm_kfree(card->dev, link_name); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); + + return ret; } int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, -- 1.7.9.5

2 1

[alsa-devel] [PATCH] soc: intel: sst-haswell-ipc: Remove some unused functions
by Rickard Strandqvist 16 Jan '15

16 Jan '15

Removes some functions that are not used anywhere: sst_hsw_stream_unmute() sst_hsw_stream_mute() msg_set_stage_type() sst_hsw_dx_get_state() sst_hsw_dx_set_state() sst_hsw_stream_set_write_position() sst_hsw_stream_get_vol_reg() sst_hsw_stream_get_peak_reg() sst_hsw_stream_get_pointer_reg() sst_hsw_stream_get_read_reg() sst_hsw_stream_get_mixer_id() sst_hsw_stream_get_hw_id() sst_hsw_mixer_set_volume_curve() sst_hsw_mixer_unmute() sst_hsw_mixer_mute() sst_hsw_stream_set_volume_curve() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/intel/sst-haswell-ipc.c | 226 ------------------------------------- sound/soc/intel/sst-haswell-ipc.h | 34 ------ 2 files changed, 260 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index b629151..26f856c 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -333,12 +333,6 @@ static inline u32 msg_get_stage_type(u32 msg) return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; } -static inline u32 msg_set_stage_type(u32 msg, u32 type) -{ - return (msg & ~IPC_STG_TYPE_MASK) + - (type << IPC_STG_TYPE_SHIFT); -} - static inline u32 msg_get_stream_id(u32 msg) { return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; @@ -965,45 +959,6 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, } /* Mixer Controls */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, - &stream->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - stream->mute[channel] = 1; - return 0; -} - -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel) - -{ - int ret; - - stream->mute[channel] = 0; - ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, - stream->mute_volume[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", - stream->reply.stream_hw_id, channel); - return ret; - } - - return 0; -} - int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume) { @@ -1017,17 +972,6 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream return 0; } -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - stream->vol_req.curve_duration = curve_duration; - stream->vol_req.curve_type = curve; - - return 0; -} - /* stream volume */ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) @@ -1065,42 +1009,6 @@ int sst_hsw_stream_set_volume(struct sst_hsw *hsw, return 0; } -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, - &hsw->mute_volume[channel]); - if (ret < 0) - return ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 1; - return 0; -} - -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) -{ - int ret; - - ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, - hsw->mixer_info.volume_register_address[channel]); - if (ret < 0) { - dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", - channel); - return ret; - } - - hsw->mute[channel] = 0; - return 0; -} - int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume) { @@ -1114,16 +1022,6 @@ int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, return 0; } -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve) -{ - /* curve duration in steps of 100ns */ - hsw->curve_duration = curve_duration; - hsw->curve_type = curve; - - return 0; -} - /* global mixer volume */ int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume) @@ -1425,50 +1323,6 @@ int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) return 0; } -/* Stream Information - these calls could be inline but we want the IPC - ABI to be opaque to client PCM drivers to cope with any future ABI changes */ -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.stream_hw_id; -} - -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.mixer_hw_id; -} - -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.read_position_register_address; -} - -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream) -{ - return stream->reply.presentation_position_register_address; -} - -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.peak_meter_register_address[channel]; -} - -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel) -{ - if (channel >= 2) - return 0; - - return stream->reply.volume_register_address[channel]; -} - int sst_hsw_mixer_get_info(struct sst_hsw *hsw) { struct sst_hsw_ipc_stream_info_reply *reply; @@ -1591,30 +1445,6 @@ u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, return ppos; } -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position) -{ - u32 header; - int ret; - - trace_stream_write_position(stream->reply.stream_hw_id, position); - - header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | - IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); - header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); - header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); - header |= (stage_id << IPC_STG_ID_SHIFT); - stream->wpos.position = position; - - ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, - sizeof(stream->wpos)); - if (ret < 0) - dev_err(hsw->dev, "error: stream %d set position %d failed\n", - stream->reply.stream_hw_id, position); - - return ret; -} - /* physical BE config */ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, @@ -1644,63 +1474,7 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, } EXPORT_SYMBOL_GPL(sst_hsw_device_set_config); -/* DX Config */ -int sst_hsw_dx_set_state(struct sst_hsw *hsw, - enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx) -{ - u32 header, state_; - int ret, item; - - header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE); - state_ = state; - - trace_ipc_request("PM enter Dx state", state); - - ret = ipc_tx_message_wait(hsw, header, &state_, sizeof(state_), - dx, sizeof(*dx)); - if (ret < 0) { - dev_err(hsw->dev, "ipc: error set dx state %d failed\n", state); - return ret; - } - - for (item = 0; item < dx->entries_no; item++) { - dev_dbg(hsw->dev, - "Item[%d] offset[%x] - size[%x] - source[%x]\n", - item, dx->mem_info[item].offset, - dx->mem_info[item].size, - dx->mem_info[item].source); - } - dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", - dx->entries_no, state); - - memcpy(&hsw->dx, dx, sizeof(*dx)); - return 0; -} - /* Used to save state into hsw->dx_reply */ -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source) -{ - struct sst_hsw_ipc_dx_memory_item *dx_mem; - struct sst_hsw_ipc_dx_reply *dx_reply; - int entry_no; - - dx_reply = &hsw->dx; - entry_no = dx_reply->entries_no; - - trace_ipc_request("PM get Dx state", entry_no); - - if (item >= entry_no) - return -EINVAL; - - dx_mem = &dx_reply->mem_info[item]; - *offset = dx_mem->offset; - *size = dx_mem->size; - *source = dx_mem->source; - - return 0; -} - static int msg_empty_list_init(struct sst_hsw *hsw) { int i; diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h index 2ac194a..b6297cb 100644 --- a/sound/soc/intel/sst-haswell-ipc.h +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -371,32 +371,18 @@ int sst_hsw_fw_get_version(struct sst_hsw *hsw, u32 create_channel_map(enum sst_hsw_channel_config config); /* Stream Mixer Controls - */ -int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); -int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, - u32 stage_id, u32 channel); int sst_hsw_stream_set_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u64 curve_duration, - enum sst_hsw_volume_curve curve); - /* Global Mixer Controls - */ -int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); -int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); - int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 volume); int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, u32 *volume); -int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, - u64 curve_duration, enum sst_hsw_volume_curve curve); - /* Stream API */ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), @@ -436,18 +422,6 @@ int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 offset, u32 size); -int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream); -u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); -u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 channel); int sst_hsw_mixer_get_info(struct sst_hsw *hsw); /* Stream ALSA trigger operations */ @@ -462,8 +436,6 @@ int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, struct sst_hsw_stream *stream, u32 *position); -int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, - struct sst_hsw_stream *stream, u32 stage_id, u32 position); u32 sst_hsw_get_dsp_position(struct sst_hsw *hsw, struct sst_hsw_stream *stream); u64 sst_hsw_get_dsp_presentation_position(struct sst_hsw *hsw, @@ -474,12 +446,6 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, enum sst_hsw_device_mode mode, u32 clock_divider); -/* DX Config */ -int sst_hsw_dx_set_state(struct sst_hsw *hsw, - enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); -int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, - u32 *offset, u32 *size, u32 *source); - /* init */ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata); -- 1.7.10.4

3 2

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Correct CBM_CFS dai format configuration
by Peter Ujfalusi 16 Jan '15

16 Jan '15

We should select FSR also to be driven by McBSP, not only FSX. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Clearly the CBM_CFS is not widely used with McBSP, this is why it has not been noticed I believe. Peter sound/soc/omap/omap-mcbsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 8b79cafab1e2..c7eb9dd67f60 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -434,7 +434,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_CBM_CFS: /* McBSP slave. FS clock as output */ regs->srgr2 |= FSGM; - regs->pcr0 |= FSXM; + regs->pcr0 |= FSXM | FSRM; break; case SND_SOC_DAIFMT_CBM_CFM: /* McBSP slave */ -- 2.2.1

3 2

[alsa-devel] Playback in infinitely loop - bug in alsa-jack-plugins?
by Valentin Corfu 16 Jan '15

16 Jan '15

Hello guys, Few days ago I have started to look over a new challenging issue with alsa.. The problem is that the sound entry in a loop infinitely and it seems that the last samples in the buffer are just repeated over and over again. This behaviour it happens when I have too less and only sporadically data ready to be playbacked on my embedded system configured with alsa jack pcm plugin. To solve this issue, (I don't want to use snd_pcm_drain()), I see 2 ways: 1) to clear(or overwrite with silence) the "just played" frames from the ring buffer 2) when I am in this situation, to gracefully stop/close the alsa opened port In both situations I have tried without success, _set_silence_threshold(), set_stop_threshold and other googled solutions.. More over, I also tested with "default" device and the result is as expected... the sound can be hear just once, then it's going into underrun state. I also created a small test(based on pcm_min.c example), where I intentionally put a ~1s(can be more) delay after _writei() function, to see the issue reproduced. (http://pastebin.com/4KyUG0BA) log with pcm jack plugin: http://pastebin.com/MtKQ4ASz log with "default": http://pastebin.com/G0zq6fmJ Now I'm suspecting one issue with the underrun, over alsa jack plugin... (In case I'm not wrong with the test case or my assumption..) - Isn't it implemented underrun detection over jack plugin or it is a new bug here? Best Regards, Valentin Corfu

1 0

[alsa-devel] [PATCH v2 10/11] ASoC: tegra: Add a control for the headphone switch
by Tomeu Vizoso 15 Jan '15

15 Jan '15

To be used by userspace when the headphones jack is plugged in. Signed-off-by: Tomeu Vizoso <tomeu.vizoso(a)collabora.com> --- sound/soc/tegra/tegra_max98090.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index af3fb99..8df71a4 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -136,6 +136,7 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), }; -- 1.9.3

2 3

[alsa-devel] [PATCH v2 11/11] ASoC: tegra: add sink for the internal mic to tegra_max98090
by Tomeu Vizoso 15 Jan '15

15 Jan '15

This is used in the Tegra-based Chromebooks. Signed-off-by: Tomeu Vizoso <tomeu.vizoso(a)collabora.com> --- Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + sound/soc/tegra/tegra_max98090.c | 2 ++ 2 files changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt index c949abc..c3495be 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -18,6 +18,7 @@ Required properties: * Headphones * Speakers * Mic Jack + * Int Mic - nvidia,i2s-controller : The phandle of the Tegra I2S controller that's connected to the CODEC. diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index 8df71a4..29ea87c 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -133,11 +133,13 @@ static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), }; static const struct snd_kcontrol_new tegra_max98090_controls[] = { SOC_DAPM_PIN_SWITCH("Headphones"), SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Int Mic"), }; static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) -- 1.9.3

1 0

[alsa-devel] [PATCH v4] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 15 Jan '15

15 Jan '15

In soc_new_compress() when rtd->dai_link->dynamic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by using rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture as playback_count and capture_count to snd_pcm_new_internal(). Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- v4->v3: - the typo fix went wrong, apologies for the noise v3->v2: - fix a typo s/dynmic/dynamic/ in commit message - use if else if because either dpcm_playback or dpcm_capture should be on for compressed devices v2->v1: - use better way to fix it than just removing the line that caused the oops sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..025c38fbe3c0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + else if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

2 1

[alsa-devel] [PATCH 00/17] ASoC: Remove codec field from snd_soc_dapm_widget
by Lars-Peter Clausen 15 Jan '15

15 Jan '15

This series is the last part of the codec field removal from the snd_soc_dapm_widget. One small fix for an issue that slipped through earlier and the conversion of the TI and Realtek CODEC drivers and then finally the removal itself. The reason why it is removed is because it doesn't fit to well anymore in the componentized ASoC hierarchy. And the alternative of snd_soc_dapm_to_codec(w->dapm) typically generates the same amount of code, so there is really no reason to keep it. Depends on topic/w-codec and topic/rt5677. - Lars Lars-Peter Clausen (17): ASoC: mioa701_wm9713: Fix speaker event ASoC: pcm3008: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320aic31xx: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320aic3x: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: tlv320dac33: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: twl4030: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: twl6040: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: alc5623: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: alc5632: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt286: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5631: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5640: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5645: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5651: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5670: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: rt5677: Replace w->codec snd_soc_dapm_to_codec(w->dapm) ASoC: Remove codec field from snd_soc_dapm_widget include/sound/soc-dapm.h | 1 - sound/soc/codecs/alc5623.c | 8 +++--- sound/soc/codecs/alc5632.c | 8 +++--- sound/soc/codecs/pcm3008.c | 4 +-- sound/soc/codecs/rt286.c | 15 ++++++----- sound/soc/codecs/rt5631.c | 28 ++++++++++++-------- sound/soc/codecs/rt5640.c | 11 ++++---- sound/soc/codecs/rt5645.c | 16 +++++++----- sound/soc/codecs/rt5651.c | 18 +++++++------ sound/soc/codecs/rt5670.c | 16 +++++++----- sound/soc/codecs/rt5677.c | 23 +++++++++-------- sound/soc/codecs/tlv320aic31xx.c | 9 ++++--- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320dac33.c | 9 ++++--- sound/soc/codecs/twl4030.c | 55 +++++++++++++++++++++++++--------------- sound/soc/codecs/twl6040.c | 4 +-- sound/soc/pxa/mioa701_wm9713.c | 2 +- sound/soc/soc-dapm.c | 2 -- 18 files changed, 132 insertions(+), 99 deletions(-) -- 1.8.0

2 18

[alsa-devel] [PATCH RFC] ASoC: simple-card: Enable and disable DAI clocks as needed
by Jyri Sarha 15 Jan '15

15 Jan '15

Call clk_prepare_enable() and clk_disable_unprepare() for cpu dai clock and codec dai clock in dai statup and shutdown callbacks. This to make sure the related clock are enabled when the audio device is used. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- This patch is needed for Beaglebone-back HDMI audio. There is no direct dependency between this patch and the other (dts and DRM) changes needed for the HDMI audio so these changes can be merged independently. There are also some other TI EVMs that require this functionality if they are to be used with simple-card. .../devicetree/bindings/sound/simple-card.txt | 5 +++ include/sound/simple_card.h | 1 + sound/soc/generic/simple-card.c | 34 ++++++++++++++++++++ 3 files changed, 40 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c3cba60..73bf314 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -75,6 +75,11 @@ Optional CPU/CODEC subnodes properties: it can be specified via "clocks" if system has clock node (= common clock), or "system-clock-frequency" (if system doens't support common clock) + If a clock is specified, it is + enabled with clk_prepare_enable() + in dai startup() and disabled with + clk_disable_unprepare() in dai + shutdown(). Example 1 - single DAI link: diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h index 9b0ac77..1255ddb 100644 --- a/include/sound/simple_card.h +++ b/include/sound/simple_card.h @@ -20,6 +20,7 @@ struct asoc_simple_dai { unsigned int sysclk; int slots; int slot_width; + struct clk *clk; }; struct asoc_simple_card_info { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fb9240f..cb3998d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -39,6 +39,37 @@ struct simple_card_data { #define simple_priv_to_link(priv, i) ((priv)->snd_card.dai_link + i) #define simple_priv_to_props(priv, i) ((priv)->dai_props + i) +static int asoc_simple_card_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = + &priv->dai_props[rtd - rtd->card->rtd]; + int ret; + + ret = clk_prepare_enable(dai_props->cpu_dai.clk); + if (ret) + return ret; + + ret = clk_prepare_enable(dai_props->codec_dai.clk); + if (ret) + clk_disable_unprepare(dai_props->cpu_dai.clk); + + return ret; +} + +static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); + struct simple_dai_props *dai_props = + &priv->dai_props[rtd - rtd->card->rtd]; + + clk_disable_unprepare(dai_props->cpu_dai.clk); + + clk_disable_unprepare(dai_props->codec_dai.clk); +} + static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -58,6 +89,8 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, } static struct snd_soc_ops asoc_simple_card_ops = { + .startup = asoc_simple_card_startup, + .shutdown = asoc_simple_card_shutdown, .hw_params = asoc_simple_card_hw_params, }; @@ -219,6 +252,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, } dai->sysclk = clk_get_rate(clk); + dai->clk = clk; } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) { dai->sysclk = val; } else { -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/16] ASoC: rsnd: cleanup patches
by Kuninori Morimoto 15 Jan '15

15 Jan '15

Hi Mark These are cleanup patches for Renesas R-Car sound driver. I noticed current rsnd driver has un-used function, no-meaning parameter, dirty code, etc... These are cleanup patches for it. Kuninori Morimoto (16): ASoC: rsnd: remove unsed .rate ASoC: rsnd: tidyup rsnd_io_to_xxx() macro declaration position ASoC: rsnd: rsnd_mod controls clock ASoC: rsnd: rename rsnd_dai_get() to rsnd_rdai_get() ASoC: rsnd: rename rsnd_dai_is_clk_master() to rsnd_rdai_is_clk_master() ASoC: rsnd: remove unused rsnd_dai_id() ASoC: rsnd: remove rsnd_dai_get_platform_info() macro ASoC: rsnd: add rsnd_io_to_rdai() ASoC: rsnd: replace rsnd_dai_is_play() to rsnd_io_is_play() ASoC: rsnd: remove unsed rdai parameter from each function ASoC: rsnd: replace rdai to priv in each function ASoC: rsnd: save priv in struct rsnd_dai ASoC: rsnd: error meesage indicates its port ASoC: rsnd: ignore DIRQ when DMA transfer ASoC: rsnd: tidyup DMA start/stop timing on SSI ASoC: rsnd: tidyup DVC/SRC/SSI order sound/soc/sh/rcar/adg.c | 14 +++---- sound/soc/sh/rcar/core.c | 91 +++++++++++++++++------------------------ sound/soc/sh/rcar/dvc.c | 48 ++++++++++------------ sound/soc/sh/rcar/rsnd.h | 68 ++++++++++++++----------------- sound/soc/sh/rcar/src.c | 94 +++++++++++++++++++------------------------ sound/soc/sh/rcar/ssi.c | 100 +++++++++++++++++++++------------------------- 6 files changed, 178 insertions(+), 237 deletions(-) Best regards --- Kuninori Morimoto

2 17

[alsa-devel] [PATCH] ASoC: rt286: set the same format for dac and adc
by Bard Liao 15 Jan '15

15 Jan '15

There is only one I2S I/F, AD/DA path must operate to the same format. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 2cd4fe4..1d1c7f8 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -861,10 +861,8 @@ static int rt286_hw_params(struct snd_pcm_substream *substream, RT286_I2S_CTRL1, 0x0018, d_len_code << 3); dev_dbg(codec->dev, "format val = 0x%x\n", val); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); - else - snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); return 0; } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] unload Audio drivers while playback stream is active case kernel crash
by Wang, Jiada (ESD) 15 Jan '15

15 Jan '15

Hi Community I am using i.MX6Q sabreSD board, which have imx_wm892 machine driver, wm8962 codec and SSI CPU DAI, I got Kernel crash when unloading audio drivers (playback stream is active) modprobe -r snd_soc_imx_wm8962 modprobe -r snd_soc_fsl_ssi modprobe -r snd_soc_wm8962 [ 208.666868] Unable to handle kernel paging request at virtc [ 208.674110] pgd = 80004000 [ 208.676867] [7f06541c] *pgd=4c334811, *pte=00000000, *ppte=00000000 [ 208.683211] Internal error: Oops: 80000007 [#1] SMP ARM [ 208.688445] Modules linked in: snd_soc_wm8962 snd_soc_fsl_ssi snd_soc_imx_audmux imx_pcm_fiq evbug] [ 208.700662] CPU: 2 PID: 99 Comm: kworker/2:2 Not tainted 3.19.0-rc3-00069-g11c8f01-dirty #2 [ 208.709019] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 208.715575] Workqueue: events_power_efficient close_delayed_work [ 208.721605] task: be20ba80 ti: bdb40000 task.ti: bdb40000 [ 208.727012] PC is at 0x7f06541c [ 208.730164] LR is at snd_soc_dapm_set_bias_level+0x3c/0xc0 [ 208.735660] pc : [<7f06541c>] lr : [<805293e8>] psr: 200f0113 [ 208.735660] sp : bdb41da8 ip : bdb41dc8 fp : bdb41dc4 [ 208.747144] r10: bdb41e0c r9 : bc2ca9c4 r8 : bc2ca9ac [ 208.752375] r7 : bdb41e14 r6 : 00000001 r5 : bc2ca9cc r4 : bc2ca86c [ 208.758908] r3 : 7f06541c r2 : 00000001 r1 : bc2ca9cc r0 : bc2ca86c [ 208.765444] Flags: nzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment kernel [ 208.772759] Control: 10c5387d Table: 4c05c04a DAC: 00000015 [ 208.778513] Process kworker/2:2 (pid: 99, stack limit = 0xbdb40238) [ 208.784787] Stack: (0xbdb41da8 to 0xbdb42000) [ 208.789154] 1da0: be20ba80 bc2ca9cc bc2ca9cc bc2ca86c bdb41ddc bdb41dc8 [ 208.797342] 1dc0: 8052ae88 805293b8 00000000 bc2ca9bc bdb41e4c bdb41de0 8052d194 8052ae4c [ 208.805529] 1de0: 80062798 8006259c bdb41e4c 00000002 bc2ca86c bc2ca9bc bdb41e0c bdb41e04 [ 208.813716] 1e00: bdb41e34 bdb41e04 bdb41e04 bdb41e0c bdb41e0c bdb41e14 bdb41e14 bdacd240 [ 208.821903] 1e20: 00000004 00000001 bb46b010 00000002 00000000 bc2ca8cc 00000000 00000001 [ 208.830089] 1e40: bdb41e74 bdb41e50 8052e5ac 8052ce20 00000000 bb46b5d4 bb46b01c bb46b010 [ 208.838276] 1e60: bdb41eb0 be7b6c00 bdb41e94 bdb41e78 8052fff8 8052e534 8052ffac be212680 [ 208.846462] 1e80: bb46b5d4 be7b3380 bdb41eec bdb41e98 8003ef74 8052ffb8 00000001 00000000 [ 208.854648] 1ea0: 8003ef08 8003f294 00000000 00000000 811cb670 80b2e650 00000000 808b0484 [ 208.862834] 1ec0: 806babe4 be212680 be7b33b0 809c1882 be212698 00000008 be7b3380 be7b3380 [ 208.871021] 1ee0: bdb41f24 bdb41ef0 8003f388 8003edec be214780 be212680 8003f228 00000000 [ 208.879207] 1f00: be214780 be212680 8003f228 00000000 00000000 00000000 bdb41fac bdb41f28 [ 208.887393] 1f20: 800446cc 8003f234 bdb41f44 00000000 80062798 be212680 00000000 00000000 [ 208.895579] 1f40: dead4ead ffffffff ffffffff 809c3f34 00000000 00000000 80832660 bdb41f5c [ 208.903764] 1f60: bdb41f5c 00000000 00000000 dead4ead ffffffff ffffffff 809c3f34 00000000 [ 208.911950] 1f80: 00000000 80832660 bdb41f88 bdb41f88 be214780 800445f0 00000000 00000000 [ 208.920137] 1fa0: 00000000 bdb41fb0 8000ed68 800445fc 00000000 00000000 00000000 00000000 [ 208.928323] 1fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 [ 208.936509] 1fe0: 00000000 00000000 00000000 00000000 00000013 00000000 e89ffefb b6dafaea [ 208.944689] Backtrace: [ 208.947173] [<805293ac>] (snd_soc_dapm_set_bias_level) from [<8052ae88>] (dapm_pre_sequence_async+) [ 208.957004] r6:bc2ca86c r5:bc2ca9cc r4:bc2ca9cc r3:be20ba80 [ 208.962747] [<8052ae40>] (dapm_pre_sequence_async) from [<8052d194>] (dapm_power_widgets+0x380/0x7) [ 208.971970] r4:bc2ca9bc r3:00000000 [ 208.975593] [<8052ce14>] (dapm_power_widgets) from [<8052e5ac>] (snd_soc_dapm_stream_event+0x84/0x) [ 208.984816] r10:00000001 r9:00000000 r8:bc2ca8cc r7:00000000 r6:00000002 r5:bb46b010 [ 208.992736] r4:00000001 [ 208.995298] [<8052e528>] (snd_soc_dapm_stream_event) from [<8052fff8>] (close_delayed_work+0x4c/0x) [ 209.004522] r8:be7b6c00 r7:bdb41eb0 r6:bb46b010 r5:bb46b01c r4:bb46b5d4 r3:00000000 [ 209.012371] [<8052ffac>] (close_delayed_work) from [<8003ef74>] (process_one_work+0x194/0x40c) [ 209.020988] r6:be7b3380 r5:bb46b5d4 r4:be212680 r3:8052ffac [ 209.026725] [<8003ede0>] (process_one_work) from [<8003f388>] (worker_thread+0x160/0x48c) [ 209.034906] r10:be7b3380 r9:be7b3380 r8:00000008 r7:be212698 r6:809c1882 r5:be7b33b0 [ 209.042823] r4:be212680 [ 209.045384] [<8003f228>] (worker_thread) from [<800446cc>] (kthread+0xdc/0xf8) [ 209.052611] r10:00000000 r9:00000000 r8:00000000 r7:8003f228 r6:be212680 r5:be214780 [ 209.060530] r4:00000000 [ 209.063096] [<800445f0>] (kthread) from [<8000ed68>] (ret_from_fork+0x14/0x2c) [ 209.070322] r7:00000000 r6:00000000 r5:800445f0 r4:be214780 [ 209.076057] Code: bad PC value [ 209.079194] ---[ end trace dd15f2865dbe27f9 ]--- Seems it's because when playback stream is active soc_cleanup_card_resources() -> snd_card_free() Will trigger soc_pcm_close() which will queue close_delayed_work(), and then when close_delayed_work() is scheduled the necessary resource has already been released. I tried to set rtd->pmdown_time to 0 in soc_cleanup_card_resources() to avoid any new delayed_work be queued. BUT when unload audio drivers, I got another kernel crash ######################### [ 28.234740] Unable to handle kernel paging request at virtual address 6faaaf90 [ 28.242075] pgd = bbde0000 [ 28.244818] [6faaaf90] *pgd=00000000 [ 28.248511] Internal error: Oops: 5 [#1] SMP ARM [ 28.253155] Modules linked in: snd_soc_imx_wm8962(-) snd_soc_imx_audmux snd_soc_wm8962 snd_soc_fsl] [ 28.267055] CPU: 1 PID: 188 Comm: pulseaudio Not tainted 3.19.0-rc3-00069-g11c8f01-dirty #7 [ 28.275435] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 28.281991] task: bc3e1d40 ti: bbc1e000 task.ti: bbc1e000 [ 28.287431] PC is at dapm_seq_insert+0x34/0xf4 [ 28.291902] LR is at dapm_power_widgets+0x234/0x798 [ 28.296809] pc : [<8052ad74>] lr : [<8052d090>] psr: 600f0113 [ 28.296809] sp : bbc1fdd0 ip : 809bdd90 fp : bbc1fdec [ 28.308308] r10: bbc3bb90 r9 : bc0405c4 r8 : bc0405ac [ 28.313554] r7 : bbc1fe24 r6 : 809bdd90 r5 : bbc3b480 r4 : bbc3b134 [ 28.320093] r3 : bbc3b0c0 r2 : 00000000 r1 : bbc1fe1c r0 : bbc3bb40 [ 28.326628] Flags: nZCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user [ 28.333771] Control: 10c5387d Table: 4bde004a DAC: 00000015 [ 28.339524] Process pulseaudio (pid: 188, stack limit = 0xbbc1e238) [ 28.345797] Stack: (0xbbc1fdd0 to 0xbbc20000) [ 28.350162] fdc0: 00000000 bbc3bb40 bbc3bba4 bbc1fe24 [ 28.358349] fde0: bbc1fe5c bbc1fdf0 8052d090 8052ad4c 80062798 8006259c bbc1fe5c 00000002 [ 28.366535] fe00: bc04046c bc0405bc bbc1fe1c bbc1fe14 bbc1fe44 bbc1fe14 bbc1fe14 bbc3b134 [ 28.374721] fe20: bbc3b134 bbc1fe24 bbc1fe24 bbc3bb40 00000004 00000001 bd916010 00000002 [ 28.382907] fe40: 00000000 bc0404cc 00000000 bd91601c bbc1fe84 bbc1fe60 8052e5f4 8052ce68 [ 28.391093] fe60: 00000000 bd916010 00000000 bc2c5a00 bbebb800 bdbe720c bbc1feac bbc1fe88 [ 28.399279] fe80: 805317cc 8052e57c bc2c5a00 bc2c5a00 bc3d9b80 bbdf7640 bc2c4ae8 00000008 [ 28.407465] fea0: bbc1fec4 bbc1feb0 8051b610 80531664 bc2c4a00 bc2c5a00 bbc1feec bbc1fec8 [ 28.415651] fec0: 8051b6a0 8051b5c0 8051b664 bbdf75c0 809bd508 bc3d9b80 bdba8f30 bdba8f30 [ 28.423837] fee0: bbc1ff0c bbc1fef0 8050eb2c 8051b670 bc3d9b80 bdba8f30 bd9b9850 bb8f75c0 [ 28.432023] ff00: bbc1ff4c bbc1ff10 800ef5bc 8050ea50 00000000 00000000 80042d68 bc3d9b88 [ 28.440208] ff20: bc3d9b80 bc3e20f0 00000000 809c9f14 bc3e1d40 00000000 bbc1e000 00000000 [ 28.448393] ff40: bbc1ff5c bbc1ff50 800ef760 800ef538 bbc1ff84 bbc1ff60 80042ec0 800ef75c [ 28.456579] ff60: bc3d9b80 8000ee64 bbc1ffb0 00000000 00000006 8000ee64 bbc1ffac bbc1ff88 [ 28.464765] ff80: 80011884 80042e14 8000ecac 00000004 00000000 0116aa50 0116ab58 0118e100 [ 28.472950] ffa0: 00000000 bbc1ffb0 8000ecf8 800117dc 00000000 76fcf084 76fcf4c0 725fcc94 [ 28.481136] ffc0: 0116aa50 0116ab58 0118e100 00000006 01118168 4ce839e4 00000000 00000000 [ 28.489322] ffe0: 00000000 7e957978 4e1e6d24 4e1e6d34 800f0010 00000016 4eff6821 4eff6c21 [ 28.497501] Backtrace: [ 28.499980] [<8052ad40>] (dapm_seq_insert) from [<8052d090>] (dapm_power_widgets+0x234/0x798) [ 28.508509] r7:bbc1fe24 r6:bbc3bba4 r5:bbc3bb40 r4:00000000 [ 28.514251] [<8052ce5c>] (dapm_power_widgets) from [<8052e5f4>] (snd_soc_dapm_stream_event+0x84/0x) [ 28.523473] r10:bd91601c r9:00000000 r8:bc0404cc r7:00000000 r6:00000002 r5:bd916010 [ 28.531393] r4:00000001 [ 28.533959] [<8052e570>] (snd_soc_dapm_stream_event) from [<805317cc>] (soc_pcm_close+0x174/0x28c) [ 28.542920] r8:bdbe720c r7:bbebb800 r6:bc2c5a00 r5:00000000 r4:bd916010 r3:00000000 [ 28.550764] [<80531658>] (soc_pcm_close) from [<8051b610>] (snd_pcm_release_substream+0x5c/0xb0) [ 28.559552] r10:00000008 r8:bc2c4ae8 r7:bbdf7640 r6:bc3d9b80 r5:bc2c5a00 r4:bc2c5a00 [ 28.567479] [<8051b5b4>] (snd_pcm_release_substream) from [<8051b6a0>] (snd_pcm_release+0x3c/0x88) [ 28.576440] r5:bc2c5a00 r4:bc2c4a00 [ 28.580062] [<8051b664>] (snd_pcm_release) from [<8050eb2c>] (snd_disconnect_release+0xe8/0xf8) [ 28.588764] r8:bdba8f30 r7:bdba8f30 r6:bc3d9b80 r5:809bd508 r4:bbdf75c0 r3:8051b664 [ 28.596604] [<8050ea44>] (snd_disconnect_release) from [<800ef5bc>] (__fput+0x90/0x1d4) [ 28.604611] r7:bb8f75c0 r6:bd9b9850 r5:bdba8f30 r4:bc3d9b80 [ 28.610345] [<800ef52c>] (__fput) from [<800ef760>] (____fput+0x10/0x14) [ 28.617050] r10:00000000 r9:bbc1e000 r8:00000000 r7:bc3e1d40 r6:809c9f14 r5:00000000 [ 28.624968] r4:bc3e20f0 [ 28.627535] [<800ef750>] (____fput) from [<80042ec0>] (task_work_run+0xb8/0xe8) [ 28.634865] [<80042e08>] (task_work_run) from [<80011884>] (do_work_pending+0xb4/0xd4) [ 28.642786] r8:8000ee64 r7:00000006 r6:00000000 r5:bbc1ffb0 r4:8000ee64 r3:bc3d9b80 [ 28.650624] [<800117d0>] (do_work_pending) from [<8000ecf8>] (work_pending+0xc/0x20) [ 28.658371] r6:0118e100 r5:0116ab58 r4:0116aa50 [ 28.663047] Code: e5905000 e3520000 e24c6070 01a0600c (e7965105) [ 28.669220] ---[ end trace 69b76cbdebd80b05 ]--- >From the backtrace, seems it's because when soc_pcm_close() is called, it will power widget due to stream event, but at that time, all widgets have already been freed by soc_remove_dai_links() Currently I am out of idea how to fix this issue, As I think this is a generic issue not i.MX platform specific, Someone in community may has already find this issue, Could anyone give some inputs/suggestion to fix this issue? Regards & Thanks, Jiada

6 19

[alsa-devel] ASoC Machine driver
by Ghadiya, Anil 15 Jan '15

15 Jan '15

Hi All, I am new to ALSA architecture and want to understand the flow of the driver. So I took one of the existing codec driver "adau1701" and build it for the my imx6q-sabrelite development board. I am able to build and insert it on the board. Now I want to write ASoC machine driver for the same. I am hoping that I do not need to write ASoC platform driver as my codec is not using any I2S, DMA, S/PDIF or any SOC specific driver. I have seen the machine driver for other platform. I can see that sound card is getting registered in the probe function of the machine driver, but . I am not able to understand the flow here. Could you please help me understand the flow of execution like........ when probe function of machine driver gets called? When will the probe function od codec driver gets executed? and so on.... Thanks in Advance, Anil Patel

1 0

[alsa-devel] [PATCH] ALSA: hda - Select INPUT for Realtek HD-audio codec
by Takashi Iwai 15 Jan '15

15 Jan '15

The commit commit [33f4acd3b214: ALSA: hda - Enable mic mute hotkey and LEDs for an HP machine] introduced a quirk for a HP machine involving with the input event handling. Although the relevant code is protected via IS_ENABLED(CONFIG_INPUT), this doesn't suffice when the audio driver is built in while the input is module. As an easy workaround, this patch forcibly selects CONFIG_INPUT in Kconfig. This shouldn't be a practical problem since CONFIG_INPUT is almost mandatory for all systems. Also, this allows to remove the ugly ifdefs in the code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 1 + sound/pci/hda/patch_realtek.c | 6 ------ 2 files changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ebf4c2fb99df..7f0f2c5a4e97 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -107,6 +107,7 @@ config SND_HDA_PATCH_LOADER config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC + select INPUT help Say Y or M here to include Realtek HD-audio codec support in snd-hda-intel driver, such as ALC880. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a50e15e166c6..1720f8a457d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -121,9 +121,7 @@ struct alc_spec { hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; unsigned int coef0; -#if IS_ENABLED(CONFIG_INPUT) struct input_dev *kb_dev; -#endif }; /* @@ -3476,7 +3474,6 @@ static void alc280_fixup_hp_gpio4(struct hda_codec *codec, } } -#if IS_ENABLED(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, struct hda_jack_callback *event) { @@ -3489,12 +3486,10 @@ static void gpio2_mic_hotkey_event(struct hda_codec *codec, input_report_key(spec->kb_dev, KEY_MICMUTE, 0); input_sync(spec->kb_dev); } -#endif static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -#if IS_ENABLED(CONFIG_INPUT) /* GPIO1 = set according to SKU external amp GPIO2 = mic mute hotkey GPIO3 = mute LED @@ -3550,7 +3545,6 @@ static void alc280_fixup_hp_gpio2_mic_hotkey(struct hda_codec *codec, input_unregister_device(spec->kb_dev); spec->kb_dev = NULL; } -#endif } static void alc269_fixup_hp_line1_mic1_led(struct hda_codec *codec, -- 2.2.1

2 1

[alsa-devel] [PATCH] ASoC: sirf: fix platform_no_drv_owner.cocci warnings
by kbuild test robot 15 Jan '15

15 Jan '15

sound/soc/sirf/sirf-atlas7-cs42xx8.c:192:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Yibo Cai <yibo.cai(a)csr.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> --- sirf-atlas7-cs42xx8.c | 1 - 1 file changed, 1 deletion(-) --- a/sound/soc/sirf/sirf-atlas7-cs42xx8.c +++ b/sound/soc/sirf/sirf-atlas7-cs42xx8.c @@ -189,7 +189,6 @@ MODULE_DEVICE_TABLE(of, sirf_hdmi_card_o static struct platform_driver sirf_hdmi_card_driver = { .driver = { .name = "sirf-hdmi-card", - .owner = THIS_MODULE, .pm = &snd_soc_pm_ops, .of_match_table = sirf_hdmi_card_of_match, },

2 1

[alsa-devel] [PATCH - UCM 1/1] ucm: add binary configure file parse
by han.lu＠intel.com 15 Jan '15

15 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> with cset command, UCM set kcontrol parameters directly: cset "name='<KCONTROL_NAME>' 1<,2,3,...>" This patch enables UCM to set kcontrol with parameters from configure file: cset-bin-file "name='<KCONTROL_NAME>' <path/to/file>" where "cset-bin-file" is a newly added keyword alongside of "cset", to indicate cset with binary data in file. The binary data is parameter only for audio DSPs, and it's just passed by UCM/ALSA as raw data. The data type of the parameter element must be byte, and the maxim number of parameter elements is 512 (the maxim value that struct snd_ctl_elem_value can hold). Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/ucm/main.c b/src/ucm/main.c index 37ae4c8..d2b1e15 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -160,11 +160,45 @@ static int open_ctl(snd_use_case_mgr_t *uc_mgr, return 0; } +static int binary_file_parse(snd_ctl_elem_value_t *dst, + const char *filepath) +{ + int err = 0; + FILE *in; + long len; + char *res; + unsigned int idx; + + in = fopen(filepath, "r"); + if (!in) { + err = -errno; + goto __fail; + } + fseek(in, 0L, SEEK_END); + len = ftell(in); + rewind(in); + if (len > 512) + len = 512; + res = calloc(1, (size_t)len); + if (res == NULL) { + err = -ENOMEM; + goto __fail_nomem; + } + fread(res, (size_t)len, 1, in); + for (idx = 0; idx < len; idx++) + snd_ctl_elem_value_set_byte(dst, idx, *(res + idx)); + free(res); + __fail_nomem: + fclose(in); + __fail: + return err; +} + extern int __snd_ctl_ascii_elem_id_parse(snd_ctl_elem_id_t *dst, const char *str, const char **ret_ptr); -static int execute_cset(snd_ctl_t *ctl, const char *cset) +static int execute_cset(snd_ctl_t *ctl, const char *cset, int isbin) { const char *pos; int err; @@ -194,7 +228,10 @@ static int execute_cset(snd_ctl_t *ctl, const char *cset) err = snd_ctl_elem_info(ctl, info); if (err < 0) goto __fail; - err = snd_ctl_ascii_value_parse(ctl, value, info, pos); + if (isbin) + err = binary_file_parse(value, pos); + else + err = snd_ctl_ascii_value_parse(ctl, value, info, pos); if (err < 0) goto __fail; err = snd_ctl_elem_write(ctl, value); @@ -239,6 +276,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail_nomem; break; case SEQUENCE_ELEMENT_TYPE_CSET: + case SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE: if (cdev == NULL) { const char *cdev1 = NULL, *cdev2 = NULL; err = get_value3(&cdev1, "PlaybackCTL", @@ -274,7 +312,7 @@ static int execute_sequence(snd_use_case_mgr_t *uc_mgr, goto __fail; } } - err = execute_cset(ctl, s->data.cset); + err = execute_cset(ctl, s->data.cset, s->isbin); if (err < 0) { uc_error("unable to execute cset '%s'\n", s->data.cset); goto __fail; diff --git a/src/ucm/parser.c b/src/ucm/parser.c index d7517f6..fb5a033 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -306,6 +306,17 @@ static int parse_sequence(snd_use_case_mgr_t *uc_mgr ATTRIBUTE_UNUSED, continue; } + if (strcmp(cmd, "cset-bin-file") == 0) { + curr->type = SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE; + curr->isbin = 1; + err = parse_string(n, &curr->data.cset); + if (err < 0) { + uc_error("error: cset-bin-file requires a string!"); + return err; + } + continue; + } + if (strcmp(cmd, "usleep") == 0) { curr->type = SEQUENCE_ELEMENT_TYPE_SLEEP; err = snd_config_get_integer(n, &curr->data.sleep); diff --git a/src/ucm/ucm_local.h b/src/ucm/ucm_local.h index 87f14a2..d0a609b 100644 --- a/src/ucm/ucm_local.h +++ b/src/ucm/ucm_local.h @@ -47,6 +47,7 @@ #define SEQUENCE_ELEMENT_TYPE_CSET 2 #define SEQUENCE_ELEMENT_TYPE_SLEEP 3 #define SEQUENCE_ELEMENT_TYPE_EXEC 4 +#define SEQUENCE_ELEMENT_TYPE_CSET_BIN_FILE 5 struct ucm_value { struct list_head list; @@ -63,6 +64,7 @@ struct sequence_element { char *cset; char *exec; } data; + int isbin; /* Indicate cset to handle binary or ascii array */ }; /* -- 1.9.1

2 1

[alsa-devel] Configuring slave output for ALC889
by Dheeraj CVR 15 Jan '15

15 Jan '15

Hi, I am using an Intel Desktop Board DX58SO. I am using S/PDIF for HDMI Audio and there is no sound output via HDMI. I have checked with alsamixer and the volumes for the S/PDIF are stuck at 0 and I can't increase them. By looking at the commit history, I have found this commit which enables 0x10 as a slave output for 0x6 on ALC889 for the exact same board that I am using. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/015901.html However, it appears that with the commit "ALSA: hda/realtek - Drop ALC882 desktop model quirks", the older workaround has been removed, which caused a regression. I have checked the output of alsainfo and it seems that 0x06 is still being used instead of 0x010. Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=1, device=0 Pincap 0x00000010: OUT Pin Default 0x99430130: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 I was planning to edit the "patch_alc882" function to configure the "spec->gen.slave_dig_outs" and set the values to "{0x10, 0, 0}". However, I was not sure if this was the right way of doing. Please advice. I am running Arch-Linux with 3.17 kernel. -- Regards, Dheeraj CVR.

2 1

[alsa-devel] [PATCH] ASoC: Intel: initial stream_hw_id to invalid value
by Jie Yang 14 Jan '15

14 Jan '15

The stream_hw_id for System stream is 0x0, if we use initial stream_hw_id value 0, it may return wrong(not committed) stream when calling function get_stream_by_id() with stream_id=0. Here initial stream_hw_id to invalid value to fix this issue. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 5bf1404..eeef9a5 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -94,6 +94,8 @@ /* Mailbox */ #define IPC_MAX_MAILBOX_BYTES 256 +#define INVALID_STREAM_HW_ID 0xffffffff + /* Global Message - Types and Replies */ enum ipc_glb_type { IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */ @@ -1208,6 +1210,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, return NULL; spin_lock_irqsave(&sst->spinlock, flags); + stream->reply.stream_hw_id = INVALID_STREAM_HW_ID; list_add(&stream->node, &hsw->stream_list); stream->notify_position = notify_position; stream->pdata = data; -- 1.9.1

2 1

[alsa-devel] [PATCH 01/29] ASoC: arizona: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 14 Jan '15

14 Jan '15

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/arizona.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 9550d74..65977bc 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -84,7 +84,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); bool manual_ena = false; @@ -692,7 +692,8 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); unsigned int reg; if (w->shift % 2) @@ -705,25 +706,25 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, priv->in_pending++; break; case SND_SOC_DAPM_POST_PMU: - snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0); + snd_soc_update_bits(codec, reg, ARIZONA_IN1L_MUTE, 0); /* If this is the last input pending then allow VU */ priv->in_pending--; if (priv->in_pending == 0) { msleep(1); - arizona_in_set_vu(w->codec, 1); + arizona_in_set_vu(codec, 1); } break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, reg, + snd_soc_update_bits(codec, reg, ARIZONA_IN1L_MUTE | ARIZONA_IN_VU, ARIZONA_IN1L_MUTE | ARIZONA_IN_VU); break; case SND_SOC_DAPM_POST_PMD: /* Disable volume updates if no inputs are enabled */ - reg = snd_soc_read(w->codec, ARIZONA_INPUT_ENABLES); + reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES); if (reg == 0) - arizona_in_set_vu(w->codec, 0); + arizona_in_set_vu(codec, 0); } return 0; @@ -760,7 +761,8 @@ int arizona_hp_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona *arizona = priv->arizona; unsigned int mask = 1 << w->shift; unsigned int val; -- 1.8.0

3 30

[alsa-devel] [PATCH 1/5] ASoC: 88pm860x: Replace w->codec snd_soc_dapm_to_codec(w->dapm)
by Lars-Peter Clausen 14 Jan '15

14 Jan '15

The codec field of the snd_soc_widget struct is eventually going to be removed, use snd_soc_dapm_to_codec(w->dapm) instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/88pm860x-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index a2bf27f..a0f2653 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -386,7 +386,7 @@ static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol, static int pm860x_rsync_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); /* * In order to avoid current on the load, mute power-on and power-off @@ -403,7 +403,7 @@ static int pm860x_rsync_event(struct snd_soc_dapm_widget *w, static int pm860x_dac_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); unsigned int dac = 0; int data; -- 1.8.0

4 7

[alsa-devel] [PATCH 0/4] ASoC: Make set_tdm_slot() semantics consistent
by Lars-Peter Clausen 14 Jan '15

14 Jan '15

Changes since v1: * Don't modify the fsl-esai driver as it already uses the non-inverted semantics Original cover letter below: This is something we shortly discussed during the ALSA mini conference in Duesseldorf last year. We want to do more useful stuff with the snd_soc_set_tdm_slot() API and build generic infrastructure on top of it. But there are currently two different incompatible semantics of how the tx_mask and rx_mask parameters of snd_soc_dai_set_tdm_slot() are used. If we want to be able to use the API in a generic way we need to fix this inconsistency. Both interpretations agree that the slot masks should be represented as bitmasks which bit 0 representing the first slot, bit 1 the second slot and so on. But where they differ is in the meaning of value of the bit. Some implementations assume that if a bit is set the channel is enabled and if the bit is cleared the channel is disabled, others have inverse semantics, they assume that if a bit is cleared the channel is enabled and if the bit is set the channel is disabled. The former is used by most drivers and later is only used by some the Freescale DAI drivers and the MC13783 CODEC driver. This appears to be a result of Freescale DAI drivers directly passing the masks through to the registers which expect the inverted semantics. And the MC13783 appears to have picked up the same meaning since it is used on a board which has a Freescale DAI on the CPU side of the DAI link. This series updates the MC13783 and Freescale DAI drivers to use the same semantics as other drivers and also updates the documentation of the snd_soc_dai_set_tdm_slot() function to be more specific on the intended semantics. - Lars Lars-Peter Clausen (4): ASoC: mc13783: Update set_tdm_slot() semantics ASoC: fsl: Update set_tdm_slot() semantics ASoC: fsl: Remove fsl_asoc_xlate_tdm_slot_mask() ASoC: Update snd_soc_dai_set_tdm_slot() documentation sound/soc/codecs/mc13783.c | 10 +++++----- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 4 ++-- sound/soc/fsl/fsl_utils.c | 27 --------------------------- sound/soc/fsl/fsl_utils.h | 3 --- sound/soc/fsl/imx-mc13783.c | 5 ++--- sound/soc/fsl/imx-ssi.c | 5 ++--- sound/soc/fsl/wm1133-ev1.c | 4 ++-- sound/soc/soc-core.c | 20 ++++++++++++++++---- 9 files changed, 30 insertions(+), 50 deletions(-) -- 1.8.0

3 6

[alsa-devel] [PATCH v3] ASoC: wm8904: fix runtime warning
by Bo Shen 14 Jan '15

14 Jan '15

Correct the path name for mux to get rid of the following warning: --->8--- wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTL wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTL wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTR wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTR wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACL wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACL wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACR wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACR ---8<--- Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v3: - Correct the DAC related route path Changes in v2: - Correct the mux path name. - Route the mux to the pins. sound/soc/codecs/wm8904.c | 23 +++++++++++++++-------- 1 file changed, 15 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..75b87c5 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL Mux", "Left", "ADCL" }, + { "AIFOUTL Mux", "Right", "ADCR" }, + { "AIFOUTR Mux", "Left", "ADCL" }, + { "AIFOUTR Mux", "Right", "ADCR" }, + + { "AIFOUTL", NULL, "AIFOUTL Mux" }, + { "AIFOUTR", NULL, "AIFOUTR Mux" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,12 +1092,16 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL Mux", "Left", "AIFINL" }, + { "DACL Mux", "Right", "AIFINR" }, + + { "DACR Mux", "Left", "AIFINL" }, + { "DACR Mux", "Right", "AIFINR" }, + + { "DACL", NULL, "DACL Mux" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "DACR Mux" }, { "DACR", NULL, "CLK_DSP" }, { "Charge pump", NULL, "SYSCLK" }, -- 2.1.0.24.g4109c28

3 2

[alsa-devel] [PATCH] Asoc: sam9x5_wm8731: depend on ARCH_AT91 instead of SOC_AT91SAM9X5
by Alexandre Belloni 14 Jan '15

14 Jan '15

The SOC_AT91SAM9X5 option is going to be removed, only depend on ARCH_AT91 like for the other drivers. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/soc/atmel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index fb3878312bf8..1579e994acf8 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -45,7 +45,7 @@ config SND_ATMEL_SOC_WM8904 config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" - depends on ATMEL_SSC && SND_ATMEL_SOC && SOC_AT91SAM9X5 + depends on ARCH_AT91 && ATMEL_SSC && SND_ATMEL_SOC select SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA select SND_SOC_WM8731 -- 2.1.0

4 3

[alsa-devel] [PATCH] ASoC: fsl: imx-spdif: Set the card owner field
by Fabio Estevam 14 Jan '15

14 Jan '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Set the card owner field to avoid getting a kernel crash when the imx-spdif is unloaded while the playback is active. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Build-tested only. sound/soc/fsl/imx-spdif.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index e94704f..33da26a 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -60,6 +60,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; data->card.dai_link = &data->dai; data->card.num_links = 1; + data->card.owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) -- 1.9.1

2 1

[alsa-devel] [PATCH] conf/ucm: GoogleNyan: Add configuration
by Tomeu Vizoso 14 Jan '15

14 Jan '15

Taken from the ChromeOS sources, this configuration should apply to all Nyan boards from Google, so far HP Chromebook 14 (nyan-blaze) and Acer Chromebook 13 (nyan-big). Signed-off-by: Tomeu Vizoso <tomeu.vizoso(a)collabora.com> --- configure.ac | 1 + src/conf/ucm/GoogleNyan/GoogleNyan.conf | 5 ++ src/conf/ucm/GoogleNyan/HiFi.conf | 83 +++++++++++++++++++++++++++++++++ src/conf/ucm/GoogleNyan/Makefile.am | 4 ++ src/conf/ucm/Makefile.am | 2 +- 5 files changed, 94 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/GoogleNyan/GoogleNyan.conf create mode 100644 src/conf/ucm/GoogleNyan/HiFi.conf create mode 100644 src/conf/ucm/GoogleNyan/Makefile.am diff --git a/configure.ac b/configure.ac index 734163c..916280b 100644 --- a/configure.ac +++ b/configure.ac @@ -654,6 +654,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/SDP4430/Makefile \ src/conf/ucm/tegraalc5632/Makefile \ src/conf/ucm/PAZ00/Makefile \ + src/conf/ucm/GoogleNyan/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ alsalisp/Makefile aserver/Makefile \ test/Makefile test/lsb/Makefile \ diff --git a/src/conf/ucm/GoogleNyan/GoogleNyan.conf b/src/conf/ucm/GoogleNyan/GoogleNyan.conf new file mode 100644 index 0000000..ebd0dd9 --- /dev/null +++ b/src/conf/ucm/GoogleNyan/GoogleNyan.conf @@ -0,0 +1,5 @@ + Comment "Nyan internal card" + SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" + } diff --git a/src/conf/ucm/GoogleNyan/HiFi.conf b/src/conf/ucm/GoogleNyan/HiFi.conf new file mode 100644 index 0000000..cb624e7 --- /dev/null +++ b/src/conf/ucm/GoogleNyan/HiFi.conf @@ -0,0 +1,83 @@ + SectionVerb { + Value { + OutputDspName "speaker_eq" + } + EnableSequence [ + cdev "hw:GoogleNyan" + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Headphone Left Switch' on" + cset "name='Headphone Right Switch' on" + cset "name='Digital EQ 3 Band Switch' off" + cset "name='Digital EQ 5 Band Switch' off" + cset "name='Digital EQ 7 Band Switch' off" + cset "name='Biquad Switch' off" + cset "name='Filter Mode' Music" + cset "name='ADC Oversampling Rate' 0" + cset "name='DMIC Mux' DMIC" + cset "name='MIC2 Mux' IN34" + cset "name='Right ADC Mixer MIC2 Switch' on" + cset "name='Left ADC Mixer MIC2 Switch' on" + cset "name='MIC2 Volume' 10" + cset "name='MIC2 Boost Volume' 0" + cset "name='Mic Jack Switch' off" + cset "name='Int Mic Switch' on" + cset "name='ADCR Boost Volume' 4" + cset "name='ADCL Boost Volume' 4" + cset "name='ADCR Volume' 11" + cset "name='ADCL Volume' 11" + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Speaker Left Mixer Volume' 2" + cset "name='Speaker Right Mixer Volume' 2" + cset "name='Record Path DC Blocking' on" + cset "name='Playback Path DC Blocking' on" + cset "name='Headphone Left Switch' on" + cset "name='Headphone Right Switch' on" + cset "name='HP Left Out Switch' off" + cset "name='HP Right Out Switch' off" + cset "name='Speaker Left Switch' on" + cset "name='Speaker Right Switch' on" + cset "name='Speakers Switch' on" + ] + DisableSequence [ + ] + } + SectionDevice."Headphone".0 { + Value { + JackName "Google Nyan Headphone Jack" + OutputDspName "" + } + EnableSequence [ + cdev "hw:GoogleNyan" + cset "name='Speakers Switch' off" + cset "name='HP Left Out Switch' on" + cset "name='HP Right Out Switch' on" + ] + DisableSequence [ + cdev "hw:GoogleNyan" + cset "name='HP Left Out Switch' off" + cset "name='HP Right Out Switch' off" + cset "name='Speakers Switch' on" + ] + } + SectionDevice."Mic".0 { + Value { + JackName "Google Nyan Mic Jack" + CaptureControl "MIC2" + } + EnableSequence [ + cdev "hw:GoogleNyan" + cset "name='Int Mic Switch' off" + cset "name='DMIC Mux' ADC" + cset "name='Mic Jack Switch' on" + cset "name='Record Path DC Blocking' on" + ] + DisableSequence [ + cdev "hw:GoogleNyan" + cset "name='Mic Jack Switch' off" + cset "name='DMIC Mux' DMIC" + cset "name='Int Mic Switch' on" + cset "name='Record Path DC Blocking' off" + ] + } diff --git a/src/conf/ucm/GoogleNyan/Makefile.am b/src/conf/ucm/GoogleNyan/Makefile.am new file mode 100644 index 0000000..47c7fc9 --- /dev/null +++ b/src/conf/ucm/GoogleNyan/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/GoogleNyan +ucm_DATA = GoogleNyan.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index bde89df..14fc7ae 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan -- 1.9.3

3 3

[alsa-devel] [PATCH 1/2] ASoC: rt5677: Use the regmap functions instead of the snd_soc functions
by Oder Chiou 14 Jan '15

14 Jan '15

The patch uses the regmap functions instead of the snd_soc functions in some cases. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 32c368c..24fc592 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -925,6 +925,8 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, static int is_using_asrc(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink) { + struct snd_soc_codec *codec = source->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); unsigned int reg, shift, val; if (source->reg == RT5677_ASRC_1) { @@ -991,7 +993,9 @@ static int is_using_asrc(struct snd_soc_dapm_widget *source, } } - val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + regmap_read(rt5677->regmap, reg, &val); + val = (val >> shift) & 0xf; + switch (val) { case 1 ... 6: return 1; @@ -4098,6 +4102,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; + struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); unsigned int val = 0; if (rx_mask || tx_mask) @@ -4135,10 +4140,12 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (dai->id) { case RT5677_AIF1: - snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val); + regmap_update_bits(rt5677->regmap, RT5677_TDM1_CTRL1, 0x1f00, + val); break; case RT5677_AIF2: - snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val); + regmap_update_bits(rt5677->regmap, RT5677_TDM2_CTRL1, 0x1f00, + val); break; default: break; -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ASoC: simple-card: Fix crash in asoc_simple_card_unref()
by Geert Uytterhoeven 14 Jan '15

14 Jan '15

If asoc_simple_card_probe() fails, asoc_simple_card_unref() may be called before dev_set_drvdata(), causing a NULL pointer dereference in asoc_simple_card_unref(): Unable to handle kernel NULL pointer dereference at virtual address 000000d4 ... PC is at asoc_simple_card_unref+0x14/0x48 LR is at asoc_simple_card_probe+0x3d4/0x40c This typically happens because asoc_simple_card_parse_of() returns -EPROBE_DEFER, but other failure modes are possible. devm_snd_soc_register_card()/snd_soc_register_card() may fail either before or after dev_set_drvdata(). Pass a snd_soc_card pointer instead of a platform_device pointer to asoc_simple_card_unref() to fix this. Note that if CONFIG_OF_DYNAMIC=n, of_node_put() is a dummy, and gcc may optimize away the loop over card->dai_link, never actually dereferencing card, and thus avoiding the crash... Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Fixes: e512e001dafa54e5 ("ASoC: simple-card: Fix the reference count of device nodes") --- Seen on sh73a0/kzm9g-multiplatform and r8a7740/armadillo-multiplatform, first with CONFIG_OF_DYNAMIC=y, later with extra debug code that caused the loop not to be optimized away. It took me a while to understand what was really happening... I think this should be applied to -stable. In v3.15 and v3.16, only 8xx, pseries, and wsp selected OF_DYNAMIC, and I believe they don't use aSoC? In v3.17 and later, OF_DYNAMIC is selected by OF_SELFTEST/OF_UNITTEST. --- sound/soc/generic/simple-card.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fb9240fdc9b70095..7fe3009b1c43c63c 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -452,9 +452,8 @@ static int asoc_simple_card_parse_of(struct device_node *node, } /* Decrease the reference count of the device nodes */ -static int asoc_simple_card_unref(struct platform_device *pdev) +static int asoc_simple_card_unref(struct snd_soc_card *card) { - struct snd_soc_card *card = platform_get_drvdata(pdev); struct snd_soc_dai_link *dai_link; int num_links; @@ -556,7 +555,7 @@ static int asoc_simple_card_probe(struct platform_device *pdev) return ret; err: - asoc_simple_card_unref(pdev); + asoc_simple_card_unref(&priv->snd_card); return ret; } @@ -572,7 +571,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) snd_soc_jack_free_gpios(&simple_card_mic_jack, 1, &simple_card_mic_jack_gpio); - return asoc_simple_card_unref(pdev); + return asoc_simple_card_unref(card); } static const struct of_device_id asoc_simple_of_match[] = { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Change irq type to 'int'
by Fabio Estevam 14 Jan '15

14 Jan '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Since commit 2ffa531078037a0 ("ASoC: fsl_ssi: Fix module unbound") the irq number is retrieved via platform_get_irq(), which may fail and return a negative number, so adapt its type to 'int'. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 65400be..46549de 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -160,7 +160,7 @@ struct fsl_ssi_soc_data { */ struct fsl_ssi_private { struct regmap *regs; - unsigned int irq; + int irq; struct snd_soc_dai_driver cpu_dai_drv; unsigned int dai_fmt; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl: imx-wm8962: Set the card owner field
by Fabio Estevam 14 Jan '15

14 Jan '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> The following crash happens when trying to unload the snd_soc_imx_wm8962 module while playback is active: [ 208.666868] Unable to handle kernel paging request at virtc [ 208.674110] pgd = 80004000 [ 208.676867] [7f06541c] *pgd=4c334811, *pte=00000000, *ppte=00000000 [ 208.683211] Internal error: Oops: 80000007 [#1] SMP ARM [ 208.688445] Modules linked in: snd_soc_wm8962 snd_soc_fsl_ssi snd_soc_imx_audmux imx_pcm_fiq evbug] ... In order to avoid such problem, fill the card owner field as suggested by Lars-Peter Clausen: "But looking at the source it seems that this is a core feature of ALSA and at least for the card module itself it will do the ref-counting when a stream is started/stopped. And we even support setting the owner of a card in ASoC. It's just that pretty much no ASoC card driver bothers to set the owner field in the snd_soc_card struct. So this particular problem can be fixed by updating the imx-wm8962 driver to set the owner field." By doing as suggested, we no longer see the crash when attempting to unload the snd_soc_imx_wm8962 module while playback is active: $ modprobe -r snd_soc_imx_wm8962 modprobe: can't unload module snd_soc_imx_wm8962: Resource temporarily unavailable Reported-by: Jiada Wang <jiada_wang(a)mentor.com> Suggested-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-wm8962.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 4caacb0..cd146d4 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -257,6 +257,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) if (ret) goto clk_fail; data->card.num_links = 1; + data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; data->card.dapm_widgets = imx_wm8962_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); -- 1.9.1

2 1

[alsa-devel] [PATCH - alsa-utils 1/1] configure: allow contitional compilation of alsaucm
by Matthieu Crapet 14 Jan '15

14 Jan '15

If alsa-lib have been compiled with --disable-ucm, alsaucm can't be built. Detection is dynamic, no configure command line option is available for now. Signed-off-by: Matthieu Crapet <mcrapet(a)gmail.com> diff --git a/Makefile.am b/Makefile.am index 248eb11..5bbe588 100644 --- a/Makefile.am +++ b/Makefile.am @@ -1,6 +1,6 @@ AM_CPPFLAGS=-I$(top_srcdir)/include -SUBDIRS = include alsactl alsaucm utils m4 po alsa-info +SUBDIRS = include alsactl utils m4 po alsa-info if ALSAMIXER SUBDIRS += alsamixer endif @@ -22,6 +22,9 @@ endif if HAVE_SEQ SUBDIRS += seq endif +if HAVE_UCM +SUBDIRS += alsaucm +endif EXTRA_DIST= TODO gitcompile AUTOMAKE_OPTIONS=foreign diff --git a/configure.ac b/configure.ac index 3c7032b..d953e5c 100644 --- a/configure.ac +++ b/configure.ac @@ -48,6 +48,8 @@ AC_CHECK_HEADERS([alsa/rawmidi.h], [have_rawmidi="yes"], [have_rawmidi="no"], [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/seq.h], [have_seq="yes"], [have_seq="no"], [#include <alsa/asoundlib.h>]) +AC_CHECK_HEADERS([alsa/use-case.h], [have_ucm="yes"], [have_ucm="no"], + [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"], [#include <samplerate.h>]) @@ -55,6 +57,7 @@ AM_CONDITIONAL(HAVE_PCM, test "$have_pcm" = "yes") AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") AM_CONDITIONAL(HAVE_SEQ, test "$have_seq" = "yes") +AM_CONDITIONAL(HAVE_UCM, test "$have_ucm" = "yes") AM_CONDITIONAL(HAVE_SAMPLERATE, test "$have_samplerate" = "yes") dnl Check for librt -- 1.8.5.4

2 2

[alsa-devel] [PATCH - alsalib 1/1] autotools: fix ucm partial build
by Matthieu Crapet 14 Jan '15

14 Jan '15

When --disable-ucm configure option is specified, don't install related include file. Signed-off-by: Matthieu Crapet <mcrapet(a)gmail.com> diff --git a/include/Makefile.am b/include/Makefile.am index 0127d5c..4baa03a 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -5,7 +5,7 @@ alsaincludedir = ${includedir}/alsa alsainclude_HEADERS = asoundlib.h asoundef.h \ version.h global.h input.h output.h error.h \ - conf.h control.h iatomic.h use-case.h + conf.h control.h iatomic.h if BUILD_CTL_PLUGIN_EXT alsainclude_HEADERS += control_external.h @@ -17,7 +17,7 @@ if BUILD_PCM_PLUGIN alsainclude_HEADERS += pcm_plugin.h endif if BUILD_PCM_PLUGIN_RATE -alsainclude_HEADERS += pcm_rate.h +alsainclude_HEADERS += pcm_rate.h endif if BUILD_PCM_PLUGIN_EXTPLUG alsainclude_HEADERS += pcm_external.h pcm_extplug.h @@ -46,6 +46,10 @@ if BUILD_SEQ alsainclude_HEADERS += seq_event.h seq.h seqmid.h seq_midi_event.h endif +if BUILD_UCM +alsainclude_HEADERS += use-case.h +endif + if BUILD_ALISP alsainclude_HEADERS += alisp.h endif -- 1.8.5.4

2 3

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Correct rx format unit configuration
by Peter Ujfalusi 14 Jan '15

14 Jan '15

In case of capture we should not use rotation. The reverse and mask is enough to get the data align correctly from the bus to MCU: Format data from bus after reverse (XRBUF) S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| With this patch all supported formats will work for playback and capture. Reported-by: Jyri Sarha <jsarha(a)ti.com> (broken S24_3LE capture) Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a6b2ff7d7d7..68347b55f6e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -467,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7; - u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; + /* + * For captured data we should not rotate, inversion and masking is + * enoguh to get the data to the right position: + * Format data from bus after reverse (XRBUF) + * S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| + * S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| + * S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB| + */ + u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() -- 2.1.0

3 6

[alsa-devel] [PATCH v9 0/4] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 14 Jan '15

14 Jan '15

Based on 3.19.0-rc3. v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (4): drm/i2c: tda998x: Add DT support for audio drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter drm/i2c: tda998x: set cts_n according to the sample width .../devicetree/bindings/drm/i2c/tda998x.txt | 18 ++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 213 +++++++++++++++++++-- include/sound/tda998x.h | 23 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 175 +++++++++++++++++ 7 files changed, 424 insertions(+), 12 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

6 60

[alsa-devel] [PATCH v2 00/16] ASoC: samsung: Add clk provider for I2S internal clocks
by Sylwester Nawrocki 14 Jan '15

14 Jan '15

This series is an attempt to resolve the CDCLK clock gating issue on Odroid X2/U3 as reported by Daniel Drake [1], by exposing the CDCLK gate clock (and the two other clocks) through clk API. Changes since previous version: - removed check for the i2s_opclk1 mux input clock while creating the mux and div clocks, - the DT binding documentation changes reworked (addressing review comments), - added include/dt-bindings/sound/samsung-i2s.h header file defining the clk indices, it's been put into a separate patch together with the I2S DT binding documentation updates to make merging of the ASoC and the dts patches separately easier, - a patch fixing compatible strings of I2S1, I2S2 in exynos4.dtsi is included in this series. This whole series may need more testing on other SoCs, so far I only tested it on Odroid Exynos4412 X2, with the I2S working in slave mode. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081753.… Sylwester Nawrocki (16): ASoC: samsung: i2s: Remove unused gpios field from struct i2s ASoC: samsung: i2s: samsung_i2s_get_driver_data() cleanup ASoC: samsung: i2s: Add return value checks in probe() ASoC: samsung: i2s: Request memory region in driver probe() ASoC: samsung: i2s: Move clk_get() to platform driver probe() ASoC: samsung: i2s: Move clk enable to the platform driver probe() ASoC: samsung: i2s: Add get_other_dai helper function ASoC: samsung: i2s: Remove an unneeded goto usage ASoC: samsung: i2s: Add spinlock in place of local_irq_* calls ASoC: samsung: i2s: Protect more registers with a spinlock ASoC: samsung: odroidx2: Handle I2S CDCLK clock conditionally ASoC: samsung: i2s: Add clk provider DT binding documentation ASoC: samsung: i2s: Add clock provider for the I2S internal clocks ARM: dts: Exynos4 and Odroid X2/U3 sound device nodes update ARM: dts: Switch Odroid X2/U2 to simple-audio-card ARM: dts: Fix I2S1, I2S2 compatible for exynos4 SoCs .../devicetree/bindings/sound/samsung-i2s.txt | 22 ++ arch/arm/boot/dts/exynos4.dtsi | 13 +- arch/arm/boot/dts/exynos4412-odroid-common.dtsi | 27 +- arch/arm/boot/dts/exynos4412-odroidu3.dts | 8 +- arch/arm/boot/dts/exynos4412-odroidx2.dts | 8 +- include/dt-bindings/sound/samsung-i2s.h | 8 + sound/soc/samsung/i2s.c | 362 ++++++++++++-------- sound/soc/samsung/odroidx2_max98090.c | 6 +- 8 files changed, 295 insertions(+), 159 deletions(-) create mode 100644 include/dt-bindings/sound/samsung-i2s.h -- 1.7.9.5

2 24

[alsa-devel] [PATCH] ALSA: snd-usb-usx2y: fix incorrect indentation on if statement
by Colin King 14 Jan '15

14 Jan '15

From: Colin Ian King <colin.king(a)canonical.com> Minor style issue, statement needs an extra tab indentation on the if statement. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/usx2y/usbusx2yaudio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 61d5dc2..dd40ca9 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -166,7 +166,7 @@ static int usX2Y_urb_play_prepare(struct snd_usX2Y_substream *subs, /* set the buffer pointer */ urb->transfer_buffer = runtime->dma_area + subs->hwptr * usX2Y->stride; if ((subs->hwptr += count) >= runtime->buffer_size) - subs->hwptr -= runtime->buffer_size; + subs->hwptr -= runtime->buffer_size; } else urb->transfer_buffer = subs->tmpbuf; -- 2.1.4

1 0

[alsa-devel] [PATCH - UCM 1/2] control: enable octal and hexadecimal parse
by han.lu＠intel.com 14 Jan '15

14 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 978977d..acaf734 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -59,10 +59,10 @@ static long get_integer(const char **ptr, long min, long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; - strtol(p, &p, 10); + strtol(p, &p, 0); } if (*p == '%') { val = (long)convert_prange1(strtod(s, NULL), min, max); @@ -87,10 +87,10 @@ static long long get_integer64(const char **ptr, long long min, long long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; - strtol(p, &p, 10); + strtol(p, &p, 0); } if (*p == '%') { val = (long long)convert_prange1(strtod(s, NULL), min, max); -- 2.1.0

4 9

[alsa-devel] [PATCH - UCM 1/2] control: enable octal and hexadecimal parse
by han.lu＠intel.com 14 Jan '15

14 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> Use zero-base for strtol(), so get_integer() and get_integer64() can parse decimal, octal and hexadecimal data from input string. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 978977d..8d6c385 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -59,7 +59,7 @@ static long get_integer(const char **ptr, long min, long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; strtol(p, &p, 10); @@ -87,7 +87,7 @@ static long long get_integer64(const char **ptr, long long min, long long max) goto out; s = p; - val = strtol(s, &p, 10); + val = strtol(s, &p, 0); if (*p == '.') { p++; strtol(p, &p, 10); -- 1.9.1

2 1

[alsa-devel] Regression with .set_bias_level in pcm512x?
by Peter Rosin 14 Jan '15

14 Jan '15

Hi! I'm working on adding support for DAIFMT_CBM_CFM and DAIFMT_CBM_CFS to pcm512x, and it works nicely in the 3.10 kernel + Atmel-patches were I have developed the code. Now I'm trying to forward-port it and test it on 3.18 + other-Atmel-patches in preparation for submission. However, there seems to be a regression in that the (pre-existing) pcm512x_set_bias_level callback is never called in the 3.18 version. I see that there has been some core work on set_bias_level, and suspect it has something to do with dapm and components, but I think someone else will find the problem much quicker than I am able to. I have feebly tried to revert 0bd2ac3dae74ee25c5ea171cb572731c7a89c248 and 68f831c2724ab72c0088471b2ed1dc99e81948ef, but that didn't help. That also seems a bit backwards. So, what needs to be done in the pcm512x driver to have set_bias_level called? Please help. Cheers, Peter

2 4

[alsa-devel] [PATCH v3] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 14 Jan '15

14 Jan '15

In soc_new_compress() when rtd->dai_link->daynamic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by using rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture as playback_count and capture_count to snd_pcm_new_internal(). Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- v3->v2: - fix a typo s/dynmic/dynamic/ in commit message - use if else if because either dpcm_playback or dpcm_capture should be on for compressed devices v2->v1: - use better way to fix it than just removing the line that caused the oops sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..025c38fbe3c0 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + else if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

1 0

[alsa-devel] [PATCH v2] ALSA: ASoC: soc-compress.c: fix NULL dereference
by Qais Yousef 14 Jan '15

14 Jan '15

In soc_new_compress() when rtd->dai_link->daynmic is set, we create the pcm substreams with this call: ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, 1, 0, &be_pcm); which passes 0 as capture_count leading to be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream being NULL, hence when trying to set rtd a few lines below we get an oops. Fix by using rtd->dai_link->dpcm_playback and rtd->dai_link->dpcm_capture as playback_count and capture_count to snd_pcm_new_internal(). Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- v2->v1: - use better way to fix it than just removing the line that caused the oops sound/soc/soc-compress.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 590a82f01d0b..27a668463ad7 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -659,7 +659,8 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->dai_link->stream_name); ret = snd_pcm_new_internal(rtd->card->snd_card, new_name, num, - 1, 0, &be_pcm); + rtd->dai_link->dpcm_playback, + rtd->dai_link->dpcm_capture, &be_pcm); if (ret < 0) { dev_err(rtd->card->dev, "ASoC: can't create compressed for %s\n", rtd->dai_link->name); @@ -668,8 +669,10 @@ int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) rtd->pcm = be_pcm; rtd->fe_compr = 1; - be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; - be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; + if (rtd->dai_link->dpcm_playback) + be_pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream->private_data = rtd; + if (rtd->dai_link->dpcm_capture) + be_pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream->private_data = rtd; memcpy(compr->ops, &soc_compr_dyn_ops, sizeof(soc_compr_dyn_ops)); } else memcpy(compr->ops, &soc_compr_ops, sizeof(soc_compr_ops)); -- 2.1.0

4 6

[alsa-devel] "ak" vs "asahi-kasei" (was: Re: [PATCH 3/3 v3] ARM: shmobile: lager: add sound support on DT)
by Geert Uytterhoeven 13 Jan '15

13 Jan '15

Hi Morimoto-san, CC devicetree, alsa-devel On Wed, Jun 11, 2014 at 9:40 AM, Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> wrote: >> On Wed, Jun 11, 2014 at 8:54 AM, Kuninori Morimoto >> <kuninori.morimoto.gx(a)renesas.com> wrote: >> > + compatible = "asahi-kasei,ak4643"; >> >> As Sergei already commented, >> Documentation/devicetree/bindings/vendor-prefixes.txt >> says >> >> ak Asahi Kasei Corp. > > I know, but, > > grep asahi-kasei sound/soc/codecs/* > sound/soc/codecs/ak4104.c: { .compatible = "asahi-kasei,ak4104", }, > sound/soc/codecs/ak4554.c: { .compatible = "asahi-kasei,ak4554" }, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4642", .data = &ak4642_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4643", .data = &ak4643_drvdata}, > sound/soc/codecs/ak4642.c: { .compatible = "asahi-kasei,ak4648", .data = &ak4648_drvdata}, > sound/soc/codecs/ak5386.c: { .compatible = "asahi-kasei,ak5386", }, > > ak4642 driver can't probe by "ak"; That's bad. And all but one of these match entries have been added _after_ the addition of "ak" to vendor-prefixes.txt... Outside sound, there's also "asahi-kasei,ak8975" vs. "ak,ak8975". > My code is based on "code", not "document" Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

14 44

Re: [alsa-devel] randconfig build error with next-20150113, in sound/pci/hda/patch_realtek.c
by Takashi Iwai 13 Jan '15

13 Jan '15

At Tue, 13 Jan 2015 11:39:12 -0700, Jim Davis wrote: > > Building with the attached random configuration file, > > sound/built-in.o: In function `alc280_fixup_hp_gpio2_mic_hotkey': > /home/jim/linux/sound/pci/hda/patch_realtek.c:3512: undefined > reference to `input_allocate_device' > /home/jim/linux/sound/pci/hda/patch_realtek.c:3520: undefined > reference to `input_register_device' > /home/jim/linux/sound/pci/hda/patch_realtek.c:3522: undefined > reference to `input_free_device' > /home/jim/linux/sound/pci/hda/patch_realtek.c:3550: undefined > reference to `input_unregister_device' > sound/built-in.o: In function `gpio2_mic_hotkey_event': > /home/jim/linux/include/linux/input.h:392: undefined reference to `input_event' > /home/jim/linux/include/linux/input.h:417: undefined reference to `input_event' > /home/jim/linux/include/linux/input.h:392: undefined reference to `input_event' > /home/jim/linux/include/linux/input.h:417: undefined reference to `input_event' > make: *** [vmlinux] Error 1 Thanks. So this is the case where the audio driver is built-in while the input driver is module. I guess the cleanest workaround is just to select INPUT from SND_HDA_CODEC_REALTEK in sound/pci/hda/Kconfig. Practically seen, CONFIG_INPUT is always set, so there must be no real drawback with it. (And we can reduce a few lines as a bonus.) David, care to prepare a patch? thanks, Takashi

1 0

[alsa-devel] [PATCH] pyalsa: alsaseq: fix memory leaks
by Hector Martin 13 Jan '15

13 Jan '15

Fix a bunch of dangling reference issues. Also switch to Py_BuildValue where appropriate, since that makes it easier to build complex dictionary/tuple structures without leaking integer/string references. Signed-off-by: Hector Martin <marcan(a)marcan.st> --- I haven't looked at the other parts of pyalsa; those could probably use some looking over too. I went through alsaseq since I'm writing a long-running MIDI client and I noticed it was leaking all event objects. pyalsa/alsaseq.c | 156 +++++++++++++++++++++---------------------------------- 1 file changed, 58 insertions(+), 98 deletions(-) diff --git a/pyalsa/alsaseq.c b/pyalsa/alsaseq.c index e34d3b4..ce332ed 100644 --- a/pyalsa/alsaseq.c +++ b/pyalsa/alsaseq.c @@ -124,7 +124,9 @@ if (PyModule_AddObject(module, name, tmp) < 0) { \ return; \ } \ - PyDict_SetItem(TDICT(subtype), PyInt_FromLong(value), tmp); \ + PyObject *key = PyInt_FromLong(value); \ + PyDict_SetItem(TDICT(subtype), key, tmp); \ + Py_DECREF(key); \ } #define TCONSTRETURN(subtype, value) { \ @@ -199,8 +201,11 @@ PyDict_SetItemString(dict, name, object) /* sets a integer into the dict */ -#define SETDICTINT(name, value) \ - PyDict_SetItemString(dict, name, PyInt_FromLong(value)) +#define SETDICTINT(name, value) { \ + PyObject *val = PyInt_FromLong(value); \ + PyDict_SetItemString(dict, name, val); \ + Py_DECREF(val); \ + } /* sets note info dict (used by SeqEvent_get_data) */ #define SETDICT_NOTE3 { \ @@ -288,6 +293,7 @@ PyList_SetItem(list, i, PyInt_FromLong(t[i])); \ } \ SETDICTOBJ("ext", list); \ + Py_DECREF(list); \ } /* gets integer from python param */ @@ -1767,7 +1773,7 @@ SeqEvent_repr(SeqEventObject *self) { return PyString_FromFormat("<alsaseq.SeqEvent type=%s(%d) flags=%d tag=%d " "queue=%d time=%s(%u.%u) from=%d:%d to=%d:%d " - "at 0x%p>", + "at %p>", typestr, self->event->type, self->event->flags, self->event->tag, self->event->queue, @@ -2367,34 +2373,23 @@ _query_connections_list(snd_seq_t *handle, PyObject *list = PyList_New(0); int index = 0; - long tmplong; snd_seq_query_subscribe_set_type(query, type); snd_seq_query_subscribe_set_index(query, index); while (snd_seq_query_port_subscribers(handle, query) >= 0) { const snd_seq_addr_t *addr = snd_seq_query_subscribe_get_addr(query); - PyObject *dict = PyDict_New(); - - tmplong = snd_seq_query_subscribe_get_queue(query); - PyDict_SetItemString(dict, "queue", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_query_subscribe_get_exclusive(query); - PyDict_SetItemString(dict, "exclusive", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_query_subscribe_get_time_update(query); - PyDict_SetItemString(dict, "time_update", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_query_subscribe_get_time_real(query); - PyDict_SetItemString(dict, "time_real", PyInt_FromLong(tmplong)); - - - PyObject *tuple = PyTuple_New(3); - PyTuple_SetItem(tuple, 0, PyInt_FromLong(addr->client)); - PyTuple_SetItem(tuple, 1, PyInt_FromLong(addr->port)); - PyTuple_SetItem(tuple, 2, dict); + PyObject *tuple = Py_BuildValue( + "(ii{sisisisi})", + (int)addr->client, + (int)addr->port, + "queue", (int)snd_seq_query_subscribe_get_queue(query), + "exclusive", (int)snd_seq_query_subscribe_get_exclusive(query), + "time_update", (int)snd_seq_query_subscribe_get_time_update(query), + "time_real", (int)snd_seq_query_subscribe_get_time_real(query)); PyList_Append(list, tuple); + Py_DECREF(tuple); snd_seq_query_subscribe_set_index(query, ++index); } return list; @@ -2466,11 +2461,13 @@ Sequencer_connection_list(SequencerObject *self, PyTuple_SetItem(porttuple, 2, conntuple); PyList_Append(portlist, porttuple); + Py_DECREF(porttuple); } PyTuple_SetItem(tuple, 2, portlist); /* append list of port tuples */ PyList_Append(list, tuple); + Py_DECREF(tuple); } return list; @@ -2506,9 +2503,6 @@ Sequencer_get_client_info(SequencerObject *self, snd_seq_client_info_t *cinfo; int client_id = -1; int ret; - PyObject *tmpobj; - long tmplong; - const char * tmpchar; char *kwlist[] = { "client_id", NULL}; if (!PyArg_ParseTupleAndKeywords(args, kwds, "|i", kwlist, @@ -2532,37 +2526,21 @@ Sequencer_get_client_info(SequencerObject *self, } } - PyObject *dict = PyDict_New(); - if (dict == NULL) { - return NULL; - } - - TCONSTASSIGN(ADDR_CLIENT, client_id, tmpobj); - PyDict_SetItemString(dict, "id", tmpobj); - - tmplong = snd_seq_client_info_get_type(cinfo); - TCONSTASSIGN(CLIENT_TYPE, tmplong, tmpobj); - PyDict_SetItemString(dict, "type", tmpobj); - - tmpchar = snd_seq_client_info_get_name(cinfo); - tmpchar = (tmpchar == NULL ? "" : tmpchar); - PyDict_SetItemString(dict, "name", PyString_FromString(tmpchar)); - - tmplong = snd_seq_client_info_get_broadcast_filter(cinfo); - PyDict_SetItemString(dict, "broadcast_filter", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_client_info_get_error_bounce(cinfo); - PyDict_SetItemString(dict, "error_bounce", PyInt_FromLong(tmplong)); - - tmpchar = (const char *)snd_seq_client_info_get_event_filter(cinfo); - tmpchar = (tmpchar == NULL ? "" : tmpchar); - PyDict_SetItemString(dict, "event_filter", PyString_FromString(tmpchar)); + PyObject *d_id, *d_type; + TCONSTASSIGN(ADDR_CLIENT, client_id, d_id); + TCONSTASSIGN(CLIENT_TYPE, snd_seq_client_info_get_type(cinfo), d_type); + const char *d_name = snd_seq_client_info_get_name(cinfo); - tmplong = snd_seq_client_info_get_num_ports(cinfo); - PyDict_SetItemString(dict, "num_ports", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_client_info_get_event_lost(cinfo); - PyDict_SetItemString(dict, "event_lost", PyInt_FromLong(tmplong)); + PyObject *dict = Py_BuildValue( + "{sNsNsssisiss#sisi}", + "id", d_id, + "type", d_type, + "name", d_name == NULL ? "" : d_name, + "broadcast_filter", (int)snd_seq_client_info_get_broadcast_filter(cinfo), + "error_bounce", (int)snd_seq_client_info_get_error_bounce(cinfo), + "event_filter", snd_seq_client_info_get_event_filter(cinfo), 32, + "num_ports", (int)snd_seq_client_info_get_num_ports(cinfo), + "event_lost", (int)snd_seq_client_info_get_event_lost(cinfo)); return dict; } @@ -2594,8 +2572,6 @@ Sequencer_get_port_info(SequencerObject *self, snd_seq_client_info_t *cinfo; int port_id; int client_id; - const char *tmpchar; - long tmplong; int ret; char *kwlist[] = { "port_id", "client_id", NULL }; @@ -2612,11 +2588,6 @@ Sequencer_get_port_info(SequencerObject *self, return NULL; } - PyObject *dict = PyDict_New(); - if (dict == NULL) { - return NULL; - } - snd_seq_port_info_alloca(&pinfo); ret = snd_seq_get_any_port_info(self->handle, client_id, port_id, pinfo); @@ -2625,17 +2596,11 @@ Sequencer_get_port_info(SequencerObject *self, return NULL; } - tmpchar = snd_seq_port_info_get_name(pinfo); - tmpchar = (tmpchar == NULL ? "" : tmpchar); - PyDict_SetItemString(dict, "name", PyString_FromString(tmpchar)); - - tmplong = snd_seq_port_info_get_capability(pinfo); - PyDict_SetItemString(dict, "capability", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_info_get_type(pinfo); - PyDict_SetItemString(dict, "type", PyInt_FromLong(tmplong)); - - return dict; + return Py_BuildValue( + "{sssIsI}", + "name", snd_seq_port_info_get_name(pinfo), + "capability", (unsigned int)snd_seq_port_info_get_capability(pinfo), + "type", (unsigned int)snd_seq_port_info_get_type(pinfo)); } /** alsaseq.Sequencer connect_ports() method: __doc__ */ @@ -2768,7 +2733,6 @@ Sequencer_get_connect_info(SequencerObject *self, snd_seq_addr_t sender, dest; snd_seq_port_subscribe_t *sinfo; int ret; - long tmplong; if (!PyArg_ParseTuple(args, "(BB)(BB)", &(sender.client), &(sender.port), &(dest.client), &(dest.port))) { @@ -2786,21 +2750,12 @@ Sequencer_get_connect_info(SequencerObject *self, return NULL; } - PyObject *dict = PyDict_New(); - - tmplong = snd_seq_port_subscribe_get_queue(sinfo); - PyDict_SetItemString(dict, "queue", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_subscribe_get_exclusive(sinfo); - PyDict_SetItemString(dict, "exclusive", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_subscribe_get_time_update(sinfo); - PyDict_SetItemString(dict, "time_update", PyInt_FromLong(tmplong)); - - tmplong = snd_seq_port_subscribe_get_time_real(sinfo); - PyDict_SetItemString(dict, "time_real", PyInt_FromLong(tmplong)); - - return dict; + return Py_BuildValue( + "{sisisisi}", + "queue", (int)snd_seq_port_subscribe_get_queue(sinfo), + "exclusive", (int)snd_seq_port_subscribe_get_exclusive(sinfo), + "time_update", (int)snd_seq_port_subscribe_get_time_update(sinfo), + "time_real", (int)snd_seq_port_subscribe_get_time_real(sinfo)); } /** alsaseq.Sequencer receive_events() method: __doc__ */ @@ -2878,6 +2833,7 @@ Sequencer_receive_events(SequencerObject *self, } PyList_Append(list, SeqEventObject); + Py_DECREF(SeqEventObject); maxevents --; @@ -3263,19 +3219,23 @@ Sequencer_registerpoll(SequencerObject *self, PyObject *args, PyObject *kwds) if (count <= 0) Py_RETURN_NONE; - reg = PyObject_GetAttr(pollObj, PyString_InternFromString("register")); + reg = PyObject_GetAttrString(pollObj, "register"); + if (!reg) + return NULL; for (i = 0; i < count; i++) { t = PyTuple_New(2); - if (t) { - PyTuple_SET_ITEM(t, 0, PyInt_FromLong(pfd[i].fd)); - PyTuple_SET_ITEM(t, 1, PyInt_FromLong(pfd[i].events)); - Py_XDECREF(PyObject_CallObject(reg, t)); - Py_DECREF(t); + if (!t) { + Py_DECREF(reg); + return NULL; } + PyTuple_SET_ITEM(t, 0, PyInt_FromLong(pfd[i].fd)); + PyTuple_SET_ITEM(t, 1, PyInt_FromLong(pfd[i].events)); + Py_XDECREF(PyObject_CallObject(reg, t)); + Py_DECREF(t); } - Py_XDECREF(reg); + Py_DECREF(reg); Py_RETURN_NONE; } -- 2.2.1

2 1

[alsa-devel] [PATCH] hdajackretask: Add dock hp/mic/line to simple options
by David Henningsson 13 Jan '15

13 Jan '15

This allows for simpler setting of dock headphone, dock mic, dock line out and dock line in. Also improve detection a bit so that an existing dock headphone shows up as such and not as regular headphone. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hdajackretask/sysfs-pin-configs.c | 32 ++++++++++++++++++++------------ 1 file changed, 20 insertions(+), 12 deletions(-) diff --git a/hdajackretask/sysfs-pin-configs.c b/hdajackretask/sysfs-pin-configs.c index 592ffe2..693842f 100644 --- a/hdajackretask/sysfs-pin-configs.c +++ b/hdajackretask/sysfs-pin-configs.c @@ -289,6 +289,8 @@ unsigned long actual_pin_config(pin_configs_t* pins) #define AC_DEFCFG_DEVICE_SHIFT 20 #define AC_DEFCFG_LOCATION (0x3f<<24) #define AC_DEFCFG_LOCATION_SHIFT 24 +#define AC_DEFCFG_GROSSLOC (0x3<<28) +#define AC_DEFCFG_GROSSLOC_SHIFT 28 #define AC_DEFCFG_PORT_CONN (0x3<<30) #define AC_DEFCFG_PORT_CONN_SHIFT 30 @@ -594,21 +596,27 @@ static gboolean disabled_caps(unsigned long pin_caps, unsigned long wid_caps) return TRUE; } +#define DEFAULT_MASK (AC_DEFCFG_PORT_CONN | AC_DEFCFG_GROSSLOC | AC_DEFCFG_DEVICE) + static typical_pins_t simple_typical_pins[] = { - {"Headphone", headphone_caps, 0x0321403f, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE, }, - {"Microphone", extmic_caps, 0x03a19020, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, - {"Line out (Front)", lineout_caps, 0x01014410, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line out (Center/LFE)", lineout_caps, 0x01014411, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line out (Back)", lineout_caps, 0x01014412, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line out (Side)", lineout_caps, 0x01014413, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE,}, - {"Line in", linein_caps, 0x0181344f, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, - {"Internal speaker", lineout_caps, 0x90170150, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE, }, - {"Internal speaker (LFE)", lineout_caps, 0x90170151, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE, }, - {"Internal speaker (Back)", lineout_caps, 0x90170152, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE | AC_DEFCFG_SEQUENCE, }, - {"Internal mic", intmic_caps, 0x90a60160, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, - {"HDMI", hdmi_caps, 0x18560070, AC_DEFCFG_PORT_CONN | AC_DEFCFG_LOCATION,}, + {"Headphone", headphone_caps, 0x0321403f, DEFAULT_MASK, }, + {"Microphone", extmic_caps, 0x03a19020, DEFAULT_MASK,}, + {"Line out (Front)", lineout_caps, 0x01014010, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line out (Center/LFE)", lineout_caps, 0x01014011, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line out (Back)", lineout_caps, 0x01014012, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line out (Side)", lineout_caps, 0x01014013, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Line in", linein_caps, 0x0181304f, DEFAULT_MASK,}, + {"Internal speaker", lineout_caps, 0x90170150, DEFAULT_MASK | AC_DEFCFG_SEQUENCE, }, + {"Internal speaker (LFE)", lineout_caps, 0x90170151, DEFAULT_MASK | AC_DEFCFG_SEQUENCE, }, + {"Internal speaker (Back)", lineout_caps, 0x90170152, DEFAULT_MASK | AC_DEFCFG_SEQUENCE, }, + {"Internal mic", intmic_caps, 0x90a60160, DEFAULT_MASK,}, + {"HDMI / DisplayPort", hdmi_caps, 0x18560070, AC_DEFCFG_PORT_CONN | AC_DEFCFG_LOCATION,}, {"SPDIF out", spdifout_caps, 0x014b1180, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, {"SPDIF in", spdifin_caps, 0x01cb6190, AC_DEFCFG_PORT_CONN | AC_DEFCFG_DEVICE,}, + {"Dock Headphone", headphone_caps, 0x222140af, DEFAULT_MASK, }, + {"Dock Microphone", extmic_caps, 0x22a190a0, DEFAULT_MASK,}, + {"Dock Line out", lineout_caps, 0x220140b0, DEFAULT_MASK | AC_DEFCFG_SEQUENCE,}, + {"Dock Line in", linein_caps, 0x228130bf, DEFAULT_MASK,}, {"Not connected", disabled_caps, 0x40f000f0, AC_DEFCFG_PORT_CONN,}, {} }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: compress_driver.h: include sound/core.h explicitly
by Qais Yousef 13 Jan '15

13 Jan '15

Fixes the following compilation error: include/sound/compress_driver.h: In function ‘snd_compr_drain_notify’: include/sound/compress_driver.h:177:2: error: implicit declaration of function ‘snd_BUG_ON’ [-Werror=implicit-function-declaration] if (snd_BUG_ON(!stream)) snd_BUG_ON() is defined in sound/core.h but the file is not included explicitly, so include it. Signed-off-by: Qais Yousef <qais.yousef(a)imgtec.com> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: linux-kernel(a)vger.kernel.org --- include/sound/compress_driver.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 396e8f73670a..1e2531058b7e 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -27,6 +27,7 @@ #include <linux/types.h> #include <linux/sched.h> +#include <sound/core.h> #include <sound/compress_offload.h> #include <sound/asound.h> #include <sound/pcm.h> -- 2.1.0

3 3

[alsa-devel] Configuring slave output for ALC889
by Dheeraj CVR 13 Jan '15

13 Jan '15

Hi, I am using an Intel Desktop Board DX58SO. I am using S/PDIF for HDMI Audio and there is no sound output via HDMI. I have checked with alsamixer and the volumes for the S/PDIF are stuck at 0 and I can't increase them. By looking at the commit history, I have found this commit which enables 0x10 as a slave output for 0x6 on ALC889 for the exact same board that I am using. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/015901.html However, it appears that with the commit "ALSA: hda/realtek - Drop ALC882 desktop model quirks", the older workaround has been removed, which caused a regression. I have checked the output of alsainfo and it seems that 0x06 is still being used instead of 0x010. Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="SPDIF Phantom Jack", index=1, device=0 Pincap 0x00000010: OUT Pin Default 0x99430130: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 I was planning to edit the "patch_alc882" function to configure the "spec->gen.slave_dig_outs" and set the values to "{0x10, 0, 0}". However, I was not sure if this was the right way of doing. Please advice. I am running Arch-Linux with 3.17 kernel. -- Regards, Dheeraj CVR.

1 0

[alsa-devel] [PATCH v2] ASoC: wm8904: fix runtime warning
by Bo Shen 13 Jan '15

13 Jan '15

Correct the path name for mux to get rid of the following warning: --->8--- wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTL wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTL wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTR wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTR wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACL wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACL wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACR wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACR ---8<--- Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- Changes in v2: - Correct the mux path name. - Route the mux to the pins. sound/soc/codecs/wm8904.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..110d0e6 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1076,10 +1076,13 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL Mux", "Left", "ADCL" }, + { "AIFOUTL Mux", "Right", "ADCR" }, + { "AIFOUTR Mux", "Left", "ADCL" }, + { "AIFOUTR Mux", "Right", "ADCR" }, + + { "AIFOUTL", NULL, "AIFOUTL Mux" }, + { "AIFOUTR", NULL, "AIFOUTR Mux" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,14 +1092,20 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL", NULL, "AIFINL" }, + { "DACL", NULL, "AIFINR" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "AIFINL" }, + { "DACR", NULL, "AIFINR" }, { "DACR", NULL, "CLK_DSP" }, + { "DACL Mux", "Left", "DACL" }, + { "DACL Mux", "Right", "DACR" }, + + { "DACR Mux", "Left", "DACL" }, + { "DACR Mux", "Right", "DACR" }, + { "Charge pump", NULL, "SYSCLK" }, { "Headphone Output", NULL, "HPL PGA" }, -- 2.1.0.24.g4109c28

2 1

[alsa-devel] [PATCH] ASoC: wm8904: fix runtime warning
by Bo Shen 13 Jan '15

13 Jan '15

As the commit 5fe5b76 (ASoC: dapm: Do not pretend to support controls for non mixer/mux widgets) remove the fake control. Then, the wm8904 throw out the following warning at runtime. --->8--- wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTL wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTL wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTL wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTL wm8904 1-001a: Control not supported for path ADCL -> [Left] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCL --> Left --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCL -> Left -> AIFOUTR wm8904 1-001a: Control not supported for path ADCR -> [Right] -> AIFOUTR wm8904 1-001a: ASoC: no dapm match for ADCR --> Right --> AIFOUTR wm8904 1-001a: ASoC: Failed to add route ADCR -> Right -> AIFOUTR wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACL wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACL wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACL wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACL wm8904 1-001a: Control not supported for path AIFINR -> [Right] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINR --> Right --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINR -> Right -> DACR wm8904 1-001a: Control not supported for path AIFINL -> [Left] -> DACR wm8904 1-001a: ASoC: no dapm match for AIFINL --> Left --> DACR wm8904 1-001a: ASoC: Failed to add route AIFINL -> Left -> DACR ---8<--- This patch is used to remove these warning. Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/codecs/wm8904.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4d2d2b1..38582d7 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1076,10 +1076,10 @@ static const struct snd_soc_dapm_route adc_intercon[] = { { "Right Capture PGA", NULL, "Right Capture Mux" }, { "Right Capture PGA", NULL, "Right Capture Inverting Mux" }, - { "AIFOUTL", "Left", "ADCL" }, - { "AIFOUTL", "Right", "ADCR" }, - { "AIFOUTR", "Left", "ADCL" }, - { "AIFOUTR", "Right", "ADCR" }, + { "AIFOUTL", NULL, "ADCL" }, + { "AIFOUTL", NULL, "ADCR" }, + { "AIFOUTR", NULL, "ADCL" }, + { "AIFOUTR", NULL, "ADCR" }, { "ADCL", NULL, "CLK_DSP" }, { "ADCL", NULL, "Left Capture PGA" }, @@ -1089,12 +1089,12 @@ static const struct snd_soc_dapm_route adc_intercon[] = { }; static const struct snd_soc_dapm_route dac_intercon[] = { - { "DACL", "Right", "AIFINR" }, - { "DACL", "Left", "AIFINL" }, + { "DACL", NULL, "AIFINR" }, + { "DACL", NULL, "AIFINL" }, { "DACL", NULL, "CLK_DSP" }, - { "DACR", "Right", "AIFINR" }, - { "DACR", "Left", "AIFINL" }, + { "DACR", NULL, "AIFINR" }, + { "DACR", NULL, "AIFINL" }, { "DACR", NULL, "CLK_DSP" }, { "Charge pump", NULL, "SYSCLK" }, -- 2.1.0.24.g4109c28

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[alsa-devel] [PULL] topic/i915-hda-componentized
by Daniel Vetter 13 Jan '15

13 Jan '15

Hi Takashi, So here's the stable tag with Imre's i915/hda componentized refactoring, based upon 3.19-rc4. I'll pull in the same tag into drm-intel-next. Cheers, Daniel The following changes since commit eaa27f34e91a14cdceed26ed6c6793ec1d186115: linux 3.19-rc4 (2015-01-11 12:44:53 -0800) are available in the git repository at: git://anongit.freedesktop.org/drm-intel tags/topic/i915-hda-componentized-2015-01-12 for you to fetch changes up to fcf3aac5fc307f0cae429f5844ddc25761662858: drm/i915: remove unused power_well/get_cdclk_freq api (2015-01-12 02:48:24 +0100) ---------------------------------------------------------------- Imre Deak (6): drm/i915: add dev_to_i915 helper drm/i915: add component support ALSA: hda: export struct hda_intel ALSA: hda: pass intel_hda to all i915 interface functions ALSA: hda: add component support drm/i915: remove unused power_well/get_cdclk_freq api drivers/gpu/drm/i915/i915_dma.c | 4 + drivers/gpu/drm/i915/i915_drv.c | 9 +- drivers/gpu/drm/i915/i915_drv.h | 8 ++ drivers/gpu/drm/i915/intel_audio.c | 110 +++++++++++++++++++++++ drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_runtime_pm.c | 56 ------------ include/drm/i915_component.h | 38 ++++++++ include/drm/i915_powerwell.h | 37 -------- sound/pci/hda/hda_i915.c | 154 ++++++++++++++++++++++---------- sound/pci/hda/hda_i915.h | 37 -------- sound/pci/hda/hda_intel.c | 60 ++++--------- sound/pci/hda/hda_intel.h | 71 +++++++++++++++ 12 files changed, 361 insertions(+), 225 deletions(-) create mode 100644 include/drm/i915_component.h delete mode 100644 include/drm/i915_powerwell.h delete mode 100644 sound/pci/hda/hda_i915.h create mode 100644 sound/pci/hda/hda_intel.h -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

2 1

[alsa-devel] [PATCH - UCM 1/2] control: add hexadecimal parse
by han.lu＠intel.com 13 Jan '15

13 Jan '15

From: "Lu, Han" <han.lu(a)intel.com> Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/src/control/ctlparse.c b/src/control/ctlparse.c index 978977d..92e2f95 100644 --- a/src/control/ctlparse.c +++ b/src/control/ctlparse.c @@ -59,7 +59,10 @@ static long get_integer(const char **ptr, long min, long max) goto out; s = p; - val = strtol(s, &p, 10); + if (*p == '0' && (*(p+1) == 'x' || *(p+1) == 'X')) + val = strtol(s, &p, 16); + else + val = strtol(s, &p, 10); if (*p == '.') { p++; strtol(p, &p, 10); @@ -87,7 +90,10 @@ static long long get_integer64(const char **ptr, long long min, long long max) goto out; s = p; - val = strtol(s, &p, 10); + if (*p == '0' && (*(p+1) == 'x' || *(p+1) == 'X')) + val = strtol(s, &p, 16); + else + val = strtol(s, &p, 10); if (*p == '.') { p++; strtol(p, &p, 10); -- 1.9.1

3 5

[alsa-devel] Question on HDA Audio - resume issue
by N, Harshapriya 13 Jan '15

13 Jan '15

Hi, On running a suspend/resume stress test, after a few hundred runs, I see that resume takes a longer time (~40 sec) because of HDA audio (00:03 is the audio device in the system) . HDA audio takes around 35 sec to resume in the following case. dmesg: >>>2015-01-11T05:43:39.993107+08:00 kernel: [46300.269994] call 0000:00:03.0+ returned 0 after 35524554 usecs >>>2015-01-11T05:43:39.993109+08:00 kernel: [46300.270033] PM: Wakeup pending, aborting suspend Can anyone help understand why would HDA audio take so long to resume sporadically? -Harsha

1 0

[alsa-devel] [PATCH 0/3] Sound support for Exynos4412 Trats2 board
by Inha Song 13 Jan '15

13 Jan '15

This patch-set adds basic sound support for the Trats2 boards. It just support primary I2s and external speaker playback. Inha Song (3): ASoC: samsung: Add machine driver for Trats2 ASoC: samsung: Document Trats2 audio subsystem bindings ARM: dts: Add sound nodes for exynos4412-trats2 .../bindings/sound/samsung,trats2-wm1811.txt | 31 +++ arch/arm/boot/dts/exynos4412-trats2.dts | 38 ++++ sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/trats2_wm1811.c | 216 +++++++++++++++++++++ 5 files changed, 295 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,trats2-wm1811.txt create mode 100644 sound/soc/samsung/trats2_wm1811.c -- 2.0.0.390.gcb682f8

3 7

[alsa-devel] [PATCH] sound: firewire: fireworks: fireworks_command: Remove unused function
by Rickard Strandqvist 13 Jan '15

13 Jan '15

Remove the function snd_efw_command_set_resp_addr() that is not used anywhere. This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/firewire/fireworks/fireworks.h | 2 -- sound/firewire/fireworks/fireworks_command.c | 21 --------------------- 2 files changed, 23 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 4f0201a..693b76b 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -189,8 +189,6 @@ enum snd_efw_transport_mode { SND_EFW_TRANSPORT_MODE_WINDOWS = 0, SND_EFW_TRANSPORT_MODE_IEC61883 = 1, }; -int snd_efw_command_set_resp_addr(struct snd_efw *efw, - u16 addr_high, u32 addr_low); int snd_efw_command_set_tx_mode(struct snd_efw *efw, enum snd_efw_transport_mode mode); int snd_efw_command_get_hwinfo(struct snd_efw *efw, diff --git a/sound/firewire/fireworks/fireworks_command.c b/sound/firewire/fireworks/fireworks_command.c index 166f805..e0cc467 100644 --- a/sound/firewire/fireworks/fireworks_command.c +++ b/sound/firewire/fireworks/fireworks_command.c @@ -175,27 +175,6 @@ end: } /* - * The address in host system for transaction response is changable when the - * device supports. struct hwinfo.flags includes its flag. The default is - * MEMORY_SPACE_EFW_RESPONSE. - */ -int snd_efw_command_set_resp_addr(struct snd_efw *efw, - u16 addr_high, u32 addr_low) -{ - __be32 addr[2]; - - addr[0] = cpu_to_be32(addr_high); - addr[1] = cpu_to_be32(addr_low); - - if (!efw->resp_addr_changable) - return -ENOSYS; - - return efw_transaction(efw, EFC_CAT_HWCTL, - EFC_CMD_HWINFO_SET_RESP_ADDR, - addr, sizeof(addr), NULL, 0); -} - -/* * This is for timestamp processing. In Windows mode, all 32bit fields of second * CIP header in AMDTP transmit packet is used for 'presentation timestamp'. In * 'no data' packet the value of this field is 0x90ffffff. -- 1.7.10.4

3 2

Re: [alsa-devel] Card name size?
by Takashi Iwai 13 Jan '15

13 Jan '15

At Sun, 11 Jan 2015 20:52:00 +0000, Alan Horstmann wrote: > > Hi, > > Just a quick question... > is there a (guaranteed) max size to the name string returned by > snd_ctl_card_info_get_name() > or is it just whatever the driver sets? It's max 32 letters (including NUL). longname and mixername are max 80, id and driver are max 16. Takashi

2 1

Re: [alsa-devel] [PATCH 00/25] line6usb cleanup
by Takashi Iwai 13 Jan '15

13 Jan '15

At Mon, 12 Jan 2015 17:35:01 +0100, Takashi Iwai wrote: > > At Sun, 11 Jan 2015 15:04:55 -0600, > Chris Rorvick wrote: > > > > > At Fri, 9 Jan 2015 23:35:46 -0600, > > > Chris Rorvick wrote: > > >> > > >> I have a TonePort UX2 that I've used for testing, meaning that some of > > >> this is really only compile-tested. > > > > > > If anyone is responsible for testing with real hardware, I'll happily > > > review. > > > > To be clear, the TonePort UX2 is real hardware. But this driver > > basically supports four classes of Line 6 devices and I'm only covering > > one of them. > > > > So this series is a first step in trying to address this. Having this > > as a single driver probably made sense when it was a separate project, > > but now that it is in-tree it seems like the POD, PODHD, TonePort, and > > Variax pieces should each be separate drivers that each depend on a core > > Line 6 driver. I think the cleanup in this series will make that > > easier. None of this is my area of expertise, though, so advice and > > feedback is very welcome. > > > > > are there any active developers for this driver? > > > > I intended to do further work. I know there is quite a bit of mundane > > checkpatch cleanup that would need to get done before this could get > > promoted, and I believe I read that it's using sysfs for stuff that > > would normally be done via an ALSA interface, and the sysfs interface > > has not been documented nor has it been justified. All stuff I thought > > I might look into. > > > > But I'm just doing this for fun so I can't promise anything. :-) > > OK, so the situation looks fairly good, we have a few active > developers and/or testers. And the current code doesn't look so > terrible despite of it being in staging directory. That said, I think > we can promote this stuff into sound/usb/line6 directory, then apply > Chris' cleanup patches, and work on it further. > > Does it sound OK for you guys? Greg? > > Once when I get approval, I'll start a new clean branch on sound.git > tree so that you guys can work on it further for 3.20 kernel. JFYI, I merged all patches Greg forwarded to me into topic/line6 branch of sound git tree. It's included to linux-next, so you can work on linux-next or my branch as well. Now the code has been moved from driver/staging/line6 to sound/usb/line6. I'm going to apply some driver rename and cleanup patches to follow to other ALSA driver standards, and think of splitting to each driver with a core line6 helper module after that. Feel free to send the further development patches to me and/or alsa-devel ML like other sound drivers. thanks, Takashi

1 0

Re: [alsa-devel] Possible regression in patch_realtek.c
by David Henningsson 12 Jan '15

12 Jan '15

On 2015-01-11 22:11, Thanos Kyritsis wrote: > Hello, > > I hope it's ok to e-mail you directly instead of going through an > official bug tracker, sorry in advance otherwise. Ok. I have added alsa-devel to CC. > I'd like your opinion on a patch of yours that dates a year back: > > https://git.kernel.org/cgit/linux/kernel/git/stable/linux-stable.git/commit… > https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1248116 > > This patch has a negative impact for my hardware. I've completely lost > the kernel's front panel jack auto-mute functionality. > > In detail: my motherboard is Gigabyte Z68AP-D3, utilizing ALC889 > chipset (http://www.gigabyte.com/products/product-page.aspx?pid=3897) > with PCI id: 0x1458, 0xa002. > > Without this patch (that is kernel up to 3.13.x), all is OK, when I > plug headphones via the front panel jack, the kernel auto-mutes the > speaker-phone jack. > With the patch (that is kernel 3.14.x and all later versions), > auto-mute does not work. Plugging headphones results on sound getting > through both jacks simultaneously. > I can, of course, control the volume/mute of these two jacks > independently via CLI alsamixer, but I think this is a regression. Do > you agree ? I'm noticing that the patch's purpose was to fix a > slightly different problem, though. > Is it possible to satisfy both requirements for both motherboards ? > > Can I do something to further help ? If you'd like me to comment on > bug #1248116, that will be no problem of course. > > Thank you in advance for your help and for your contribution towards a > better kernel and distros :-) Okay. So the real question is if there's a way we can tell the working and the faulty ones apart. Until we have found such a way, I'll be hesitant to revert the patch. Here's alsa-info for a faulty one: http://www.alsa-project.org/db/?f=435c404946687b445f6c60da5edc5c3f083bc1da Both Codec SSID and PCI SSID are 1458:a002 on that machine, codec vendor ID is 10ec:0899 - maybe we should only apply this quirk for certain codec vendor IDs, then? Anyhow, you can bring back jack detection for yourself (as a workaround) by reconfiguring the codec. This is easiest done through hda-jack-retask (available in the alsa-tools repository). In hda-jack-retask click "advanced override", then make sure pin 0x1b has jack detect available. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

3 2

[alsa-devel] [PATCH] ASoC: rt5677: Add the slot_width "25" support in the TDM mode
by Oder Chiou 12 Jan '15

12 Jan '15

Add the slot_width "25" support in the TDM mode for the Intel platform. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5677.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 32c368c..af34ffc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4123,6 +4123,7 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, val |= (1 << 8); break; case 24: + case 25: val |= (2 << 8); break; case 32: @@ -4136,9 +4137,14 @@ static int rt5677_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (dai->id) { case RT5677_AIF1: snd_soc_update_bits(codec, RT5677_TDM1_CTRL1, 0x1f00, val); + if (slot_width == 25) + snd_soc_update_bits(codec, RT5677_DIG_MISC, 0x8000, + 0x8000); break; case RT5677_AIF2: snd_soc_update_bits(codec, RT5677_TDM2_CTRL1, 0x1f00, val); + if (slot_width == 25) + snd_soc_update_bits(codec, RT5677_DIG_MISC, 0x80, 0x80); break; default: break; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 4/5] sound: dmasound_atari: Remove obsolete IRQ_TYPE_SLOW
by Geert Uytterhoeven 12 Jan '15

12 Jan '15

IRQ_TYPE_SLOW is no longer used by the Atari platform interrupt code since commit 734085651c9b80aa ("[PATCH] m68k: convert atari irq code") in v2.6.18-rc1, so drop it. Note that its value has been reused for a different purpose (IRQ_TYPE_NONE) since commit 6a6de9ef5850d063 ("[PATCH] genirq: core") in v2.6.18-rc1. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: alsa-devel(a)alsa-project.org --- sound/oss/dmasound/dmasound_atari.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 13c214466d3b9f0e..1c56bf58eff906e0 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -851,7 +851,7 @@ static int __init AtaIrqInit(void) st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ st_mfp.tim_ct_a = 8; /* Turn on event counting. */ /* Register interrupt handler. */ - if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", + if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, 0, "DMA sound", AtaInterrupt)) return 0; st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */ -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] alsactl/init: Change the "basic rules" description
by David Henningsson 12 Jan '15

12 Jan '15

We now enable more outputs than just the main speaker, so update the basic rules to reflect the actual situation. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- alsactl/init/default | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/alsactl/init/default b/alsactl/init/default index 5fb0113..64f933f 100644 --- a/alsactl/init/default +++ b/alsactl/init/default @@ -3,7 +3,7 @@ # # Basic rules are: # - keep volumes at minimal level, but sound should be hearable -# - enable just main speakers for playback and main microphone for recording +# - enable standard outputs for playback and main microphone for recording # # ************************************************************************** -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: rme*: Use snd_pcm_format_t
by Takashi Iwai 12 Jan '15

12 Jan '15

Fix sparse warnings below by using snd_pcm_format_t properly: sound/pci/rme32.c:682:60: sparse: incorrect type in argument 2 (different base types) sound/pci/rme96.c:1006:69: sparse: incorrect type in argument 2 (different base types) ..... Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 6 ++---- 2 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index 6c60dcd2e5a1..1a7affad7164 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -632,7 +632,7 @@ snd_rme32_setframelog(struct rme32 * rme32, int n_channels, int is_playback) } } -static int snd_rme32_setformat(struct rme32 * rme32, int format) +static int snd_rme32_setformat(struct rme32 *rme32, snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S16_LE: diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index e33e79eadf83..236ac1d48184 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -922,8 +922,7 @@ snd_rme96_setframelog(struct rme96 *rme96, } static int -snd_rme96_playback_setformat(struct rme96 *rme96, - int format) +snd_rme96_playback_setformat(struct rme96 *rme96, snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S16_LE: @@ -940,8 +939,7 @@ snd_rme96_playback_setformat(struct rme96 *rme96, } static int -snd_rme96_capture_setformat(struct rme96 *rme96, - int format) +snd_rme96_capture_setformat(struct rme96 *rme96, snd_pcm_format_t format) { switch (format) { case SNDRV_PCM_FORMAT_S16_LE: -- 2.2.1

1 0

Re: [alsa-devel] [FFADO-devel] [RFC] libhinawa: a light-weight I/O library for status/transactions to ALSA FireWire devices
by Jonathan Woithe 12 Jan '15

12 Jan '15

Hi Takashi S On Tue, Sep 23, 2014 at 08:39:34PM +0900, Takashi Sakamoto wrote: > For some reasons (I describe later), I start to develop a light-weight > I/O library for status/transactions to ALSA FireWire devices. > https://github.com/takaswie/libhinawa > > This library gives ways to use ALSA FireWire driver functionality such as > lock/unlock kernel streaming, EFW transaction. This is one of my > solutions to some issues between ALSA/FFADO. As I understand it, this library will be a thin wrapper around whatever kernel interface is used to communicate these details out to userspace (I think byte-type mixer controls are the preferred avenue at present). In principle I have no problem with this approach so long as it remains lightweight. The situation to avoid is the one we had with libraw1394; when the new stack came about the preferred way to interact with it switched back to the native kernel interface since (as I understand the situation) it was felt that libraw1394 didn't really provide a worthwhile API abstraction. Out of interest, is there any signficance in the library's name "hinawa"? > I hope to merge this library into alsa-tools package in future, and this > is a first RFC for it. It's under development and I require your > comments about its APIs and structure ... I will try to find some time to look at the current proposal in the next week or so. > Dice notification is an actual example. Dice based devices transfer > notification to a specific address on host controller. The address space > is exclusive resources on an system. For Dice notification, ALSA Dice > driver gives a way to utilize it for applications As well as Fireworks > situation, some userspace stuffs are needed to use this functionality. For sure. One thing to keep in mind is that while this is an issue for DICE devices it may not apply to others. The library API should be structured to keep in mind that not all features will be needed across the board. We don't want to burden all subsequent firewire-audio streaming drivers with baggage which is only required for a small subset. Regards jonathan

2 2

[alsa-devel] [PATCHv3 1/2] ALSA: fix emu8000 DRAM sizing for AWE64 Value
by David Flater 12 Jan '15

12 Jan '15

Applicable to any kernel since 2013: The special case added in commit 1338fc97d07a did not handle the possibility that the address space on an AWE64 Value would wrap around at 512 KiB. That is what it does, so the memory is still not detected on those cards. Fix that with a logic clean-up that eliminates the need for a special case. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-11 v3: In response to feedback from Takashi Iwai, Patch 1: Repeat code to avoid goto middle of loop. Patch 2: Use pr_info not printk. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-09 v2: In response to feedback from Takashi Iwai, 1. Optimize for diff size. 1a. Use goto to avoid indenting and repeating code. 1b. Jettison new debugging printouts. 2. Split printk into second patch. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). sound/isa/sb/emu8000.c | 31 +++++++++++++++---------------- 1 file changed, 15 insertions(+), 16 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 45fcdff..d25e384 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -378,13 +378,12 @@ init_arrays(struct snd_emu8000 *emu) static void size_dram(struct snd_emu8000 *emu) { - int i, size, detected_size; + int i, size; if (emu->dram_checked) return; size = 0; - detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -392,10 +391,19 @@ size_dram(struct snd_emu8000 *emu) EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET); EMU8000_SMLD_WRITE(emu, UNIQUE_ID1); snd_emu8000_init_fm(emu); /* This must really be here and not 2 lines back even */ + snd_emu8000_write_wait(emu); - while (size < EMU8000_MAX_DRAM) { + /* + * Detect first 512 KiB. If a write succeeds at the beginning of a + * 512 KiB page we assume that the whole page is there. + */ + EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); + EMU8000_SMLD_READ(emu); /* discard stale data */ + if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1) + goto skip_detect; /* No RAM */ + snd_emu8000_read_wait(emu); - size += 512 * 1024; /* increment 512kbytes */ + for (size = 512 * 1024; size < EMU8000_MAX_DRAM; size += 512 * 1024) { /* Write a unique data on the test address. * if the address is out of range, the data is written on @@ -431,18 +439,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_read_wait(emu); /* Otherwise, it's valid memory. */ - detected_size = size + 512 * 1024; - } - - /* Distinguish 512 KiB from 0. */ - if (detected_size == 0) { - snd_emu8000_read_wait(emu); - EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); - EMU8000_SMLD_READ(emu); /* discard stale data */ - if (EMU8000_SMLD_READ(emu) == UNIQUE_ID1) - detected_size = 512 * 1024; } +skip_detect: /* wait until FULL bit in SMAxW register is false */ for (i = 0; i < 10000; i++) { if ((EMU8000_SMALW_READ(emu) & 0x80000000) == 0) @@ -455,9 +454,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, detected_size/1024); + emu->port1, size/1024); - emu->mem_size = detected_size; + emu->mem_size = size; emu->dram_checked = 1; } -- 1.8.4

2 5

[alsa-devel] [PATCH 1/2] lib: devres: add a helper function for ioremap_wc
by Abhilash Kesavan 12 Jan '15

12 Jan '15

Implement a resource managed writecombine ioremap function. Signed-off-by: Abhilash Kesavan <a.kesavan(a)samsung.com> --- Documentation/driver-model/devres.txt | 1 + include/linux/io.h | 2 ++ lib/devres.c | 28 ++++++++++++++++++++++++++++ 3 files changed, 31 insertions(+) diff --git a/Documentation/driver-model/devres.txt b/Documentation/driver-model/devres.txt index b5ab416..0f80cee 100644 --- a/Documentation/driver-model/devres.txt +++ b/Documentation/driver-model/devres.txt @@ -274,6 +274,7 @@ IOMAP devm_ioport_unmap() devm_ioremap() devm_ioremap_nocache() + devm_ioremap_wc() devm_ioremap_resource() : checks resource, requests memory region, ioremaps devm_iounmap() pcim_iomap() diff --git a/include/linux/io.h b/include/linux/io.h index fa02e55..42b33f0 100644 --- a/include/linux/io.h +++ b/include/linux/io.h @@ -64,6 +64,8 @@ void __iomem *devm_ioremap(struct device *dev, resource_size_t offset, resource_size_t size); void __iomem *devm_ioremap_nocache(struct device *dev, resource_size_t offset, resource_size_t size); +void __iomem *devm_ioremap_wc(struct device *dev, resource_size_t offset, + resource_size_t size); void devm_iounmap(struct device *dev, void __iomem *addr); int check_signature(const volatile void __iomem *io_addr, const unsigned char *signature, int length); diff --git a/lib/devres.c b/lib/devres.c index 0f1dd2e..e8e1738 100644 --- a/lib/devres.c +++ b/lib/devres.c @@ -72,6 +72,34 @@ void __iomem *devm_ioremap_nocache(struct device *dev, resource_size_t offset, EXPORT_SYMBOL(devm_ioremap_nocache); /** + * devm_ioremap_wc - Managed ioremap_wc() + * @dev: Generic device to remap IO address for + * @offset: BUS offset to map + * @size: Size of map + * + * Managed ioremap_wc(). Map is automatically unmapped on driver detach. + */ +void __iomem *devm_ioremap_wc(struct device *dev, resource_size_t offset, + resource_size_t size) +{ + void __iomem **ptr, *addr; + + ptr = devres_alloc(devm_ioremap_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return NULL; + + addr = ioremap_wc(offset, size); + if (addr) { + *ptr = addr; + devres_add(dev, ptr); + } else + devres_free(ptr); + + return addr; +} +EXPORT_SYMBOL(devm_ioremap_wc); + +/** * devm_iounmap - Managed iounmap() * @dev: Generic device to unmap for * @addr: Address to unmap -- 1.7.9.5

9 14

[alsa-devel] [PATCH] ASoC: ad193x: Don't power down DAC in CODEC probe
by Lars-Peter Clausen 12 Jan '15

12 Jan '15

The DAC powerdown bit is managed by DAPM. Manually powering down the DAC in the CODEC probe function may cause unnecessary power state transitions which can lead to click and pop noises. So leave the DAC powerdown bit in its default poweron-reset state and let DAPM do all the management. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad193x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 387530b..17c9535 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -333,8 +333,8 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_DAC_CHNL_MUTE, 0x0); /* de-emphasis: 48kHz, powedown dac */ regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A); - /* powerdown dac, dac in tdm mode */ - regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x41); + /* dac in tdm mode */ + regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40); /* high-pass filter enable */ regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3); /* sata delay=1, adc aux mode */ -- 1.8.0

2 1

[alsa-devel] [PATCH v4 0/6] sanitize hda/i915 interface using the component fw
by Imre Deak 12 Jan '15

12 Jan '15

This is v4 of [1] addressing the review comments from Takashi and Jani. [1] http://lists.freedesktop.org/archives/intel-gfx/2014-December/056992.html Imre Deak (6): drm/i915: add dev_to_i915 helper drm/i915: add component support ALSA: hda: export struct hda_intel ALSA: hda: pass intel_hda to all i915 interface functions ALSA: hda: add component support drm/i915: remove unused power_well/get_cdclk_freq api drivers/gpu/drm/i915/i915_dma.c | 4 + drivers/gpu/drm/i915/i915_drv.c | 9 +- drivers/gpu/drm/i915/i915_drv.h | 8 ++ drivers/gpu/drm/i915/intel_audio.c | 110 +++++++++++++++++++++++ drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_runtime_pm.c | 56 ------------ include/drm/i915_component.h | 38 ++++++++ include/drm/i915_powerwell.h | 37 -------- sound/pci/hda/hda_i915.c | 152 ++++++++++++++++++++++---------- sound/pci/hda/hda_i915.h | 37 -------- sound/pci/hda/hda_intel.c | 60 ++++--------- sound/pci/hda/hda_intel.h | 71 +++++++++++++++ 12 files changed, 360 insertions(+), 224 deletions(-) create mode 100644 include/drm/i915_component.h delete mode 100644 include/drm/i915_powerwell.h delete mode 100644 sound/pci/hda/hda_i915.h create mode 100644 sound/pci/hda/hda_intel.h -- 2.1.0

3 9

Re: [alsa-devel] Regression for Lenovo X240 sound in 3.18
by Raymond Yau 12 Jan '15

12 Jan '15

> > > >control.14 { > >iface CARD > >name 'Mic Jack' > >value false > >comment { access read type BOOLEAN count 1 } } > > > >What is the usage of this jack detect control if conventional mic is not > >supported ? > > "jack detect" reads like "detect if something is present", and that > is overly general, which could also mean it is used to detect > when to turn off the built-in speaker. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem those dock stations also have combo jack but the patch did not create dock headset mic jack / dock headset mic phantom jack ThinkPad® Ultra Dock(40A20090XX), ThinkPad® Pro Dock (40A10065XX), ThinkPad® Dock (40A00065) http://support.lenovo.com/en/documents/pd028010 http://support.lenovo.com/en/documents/pd028011 1x Stereo/Mic Combo Audio Port Refer to the maunal of those dock station, the icon near the combo jack is also a headset

1 0

[alsa-devel] [PATCHv3 2/2] ALSA: log emu8000 DRAM size at level INFO
by David Flater 11 Jan '15

11 Jan '15

Applicable after PATCHv3 1/2: Detected sound font memory goes unreported unless the kernel was built with ALSA debugging enabled. Elevate that to a pr_info. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-11 v3: In response to feedback from Takashi Iwai, Patch 1: Repeat code to avoid goto middle of loop. Patch 2: Use pr_info not printk. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-09 v2: In response to feedback from Takashi Iwai, 1. Optimize for diff size. 1a. Use goto to avoid indenting and repeating code. 1b. Jettison new debugging printouts. 2. Split printk into second patch. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). sound/isa/sb/emu8000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index d25e384..96e9d94 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -453,7 +453,7 @@ skip_detect: snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_CLOSE); snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); - snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", + pr_info("EMU8000 [0x%lx]: %d KiB on-board DRAM detected\n", emu->port1, size/1024); emu->mem_size = size; -- 1.8.4

1 0

[alsa-devel] [PATCH] ALSA: lx6464es: lx_core: Remove some unused functions
by Rickard Strandqvist 11 Jan '15

11 Jan '15

Removes some functions that are not used anywhere: lx_buffer_cancel() lx_buffer_free() lx_stream_sample_position() lx_stream_state() lx_pipe_sample_count() This was partially found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/pci/lx6464es/lx_core.c | 107 ------------------------------------------ sound/pci/lx6464es/lx_core.h | 10 ---- 2 files changed, 117 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index f3d6202..5621cd7 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -582,33 +582,6 @@ int lx_pipe_pause(struct lx6464es *chip, u32 pipe, int is_capture) } -int lx_pipe_sample_count(struct lx6464es *chip, u32 pipe, int is_capture, - u64 *rsample_count) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_0A_GET_PIPE_SPL_COUNT); - - chip->rmh.cmd[0] |= pipe_cmd; - chip->rmh.stat_len = 2; /* need all words here! */ - - err = lx_message_send_atomic(chip, &chip->rmh); /* don't sleep! */ - - if (err != 0) - dev_err(chip->card->dev, - "could not query pipe's sample count\n"); - else { - *rsample_count = ((u64)(chip->rmh.stat[0] & MASK_SPL_COUNT_HI) - << 24) /* hi part */ - + chip->rmh.stat[1]; /* lo part */ - } - - mutex_unlock(&chip->msg_lock); - return err; -} - int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate) { int err; @@ -714,46 +687,6 @@ int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, return err; } -int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture, - int *rstate) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_0E_GET_STREAM_SPL_COUNT); - - chip->rmh.cmd[0] |= pipe_cmd; - - err = lx_message_send_atomic(chip, &chip->rmh); - - *rstate = (chip->rmh.stat[0] & SF_START) ? START_STATE : PAUSE_STATE; - - mutex_unlock(&chip->msg_lock); - return err; -} - -int lx_stream_sample_position(struct lx6464es *chip, u32 pipe, int is_capture, - u64 *r_bytepos) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_0E_GET_STREAM_SPL_COUNT); - - chip->rmh.cmd[0] |= pipe_cmd; - - err = lx_message_send_atomic(chip, &chip->rmh); - - *r_bytepos = ((u64) (chip->rmh.stat[0] & MASK_SPL_COUNT_HI) - << 32) /* hi part */ - + chip->rmh.stat[1]; /* lo part */ - - mutex_unlock(&chip->msg_lock); - return err; -} - /* low-level buffer handling */ int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture, u32 buffer_size, u32 buf_address_lo, u32 buf_address_hi, @@ -803,46 +736,6 @@ int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture, return err; } -int lx_buffer_free(struct lx6464es *chip, u32 pipe, int is_capture, - u32 *r_buffer_size) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_11_CANCEL_BUFFER); - - chip->rmh.cmd[0] |= pipe_cmd; - chip->rmh.cmd[0] |= MASK_BUFFER_ID; /* ask for the current buffer: the - * microblaze will seek for it */ - - err = lx_message_send_atomic(chip, &chip->rmh); - - if (err == 0) - *r_buffer_size = chip->rmh.stat[0] & MASK_DATA_SIZE; - - mutex_unlock(&chip->msg_lock); - return err; -} - -int lx_buffer_cancel(struct lx6464es *chip, u32 pipe, int is_capture, - u32 buffer_index) -{ - int err; - u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - - mutex_lock(&chip->msg_lock); - lx_message_init(&chip->rmh, CMD_11_CANCEL_BUFFER); - - chip->rmh.cmd[0] |= pipe_cmd; - chip->rmh.cmd[0] |= buffer_index; - - err = lx_message_send_atomic(chip, &chip->rmh); - - mutex_unlock(&chip->msg_lock); - return err; -} - /* low-level gain/peak handling * diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 0cc140c..f95ed1b 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -120,8 +120,6 @@ int lx_dsp_get_mac(struct lx6464es *chip); int lx_pipe_allocate(struct lx6464es *chip, u32 pipe, int is_capture, int channels); int lx_pipe_release(struct lx6464es *chip, u32 pipe, int is_capture); -int lx_pipe_sample_count(struct lx6464es *chip, u32 pipe, int is_capture, - u64 *rsample_count); int lx_pipe_state(struct lx6464es *chip, u32 pipe, int is_capture, u16 *rstate); int lx_pipe_stop(struct lx6464es *chip, u32 pipe, int is_capture); int lx_pipe_start(struct lx6464es *chip, u32 pipe, int is_capture); @@ -133,10 +131,6 @@ int lx_pipe_wait_for_idle(struct lx6464es *chip, u32 pipe, int is_capture); /* low-level stream handling */ int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, u32 pipe, int is_capture); -int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture, - int *rstate); -int lx_stream_sample_position(struct lx6464es *chip, u32 pipe, int is_capture, - u64 *r_bytepos); int lx_stream_set_state(struct lx6464es *chip, u32 pipe, int is_capture, enum stream_state_t state); @@ -168,10 +162,6 @@ int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, int lx_buffer_give(struct lx6464es *chip, u32 pipe, int is_capture, u32 buffer_size, u32 buf_address_lo, u32 buf_address_hi, u32 *r_buffer_index); -int lx_buffer_free(struct lx6464es *chip, u32 pipe, int is_capture, - u32 *r_buffer_size); -int lx_buffer_cancel(struct lx6464es *chip, u32 pipe, int is_capture, - u32 buffer_index); /* low-level gain/peak handling */ int lx_level_unmute(struct lx6464es *chip, int is_capture, int unmute); -- 1.7.10.4

3 3

[alsa-devel] No internal speakers on laptop, suspend-to-ram and resume makes them work
by Wolfy 11 Jan '15

11 Jan '15

I have a Panasonic Toughbook CF-52 with intel hda and AD1833. No sound from internal speakers, but sound from headphones jack works correctly. Based on a forum post I found regarding the same issue in Win7, I tried suspending to ram and resuming, and sound works afterwards, until the next reboot. debianuser (very very helpful!) on irc said I should post this here, as different settings after resume shouldn't be, suggesting a bug. Plus, sound not working for a pretty large laptop brand suggests a fix/quirk of some kind is needed, even if the real bug is in the laptop. Some notes / things tried: Headphones output seems to work correctly on boot. No alsamixer sliders/muting/switches result in sound from speakers. Googling shows this affects a pretty wide variety of panasonic laptops, not just mine or my model. Googling shows this happens in win7 (but not earlier windows versions) unless a "Hotkey Appendix Driver" from panasonic is some random combination (depending on post) of installed, uninstalled and reinstalled. The keyboard does have a few audio hotkeys, but they seem to be software only, and are simply grabbed by xfce's volumed, where they seem to do exactly what they should. But apparently the panasonic windows one does something else with audio from within the hotkey driver? Perhaps it mutes it by toggling some non-obvious bit? Or despite being named "hotkey", it also handles the headphones jack automute? Googling found some reports saying that booting into windows (with the proper extra driver installed) then rebooting into linux will make sound work "for a while", even over reboots. I do not have windows to test these posts. This suggests it's something being saved somewhere that survives poweroff. Googling didn't find anyone with this problem who managed to make sound work correctly without windows. (not good!) It always works (internal speakers come on) after a suspend-to-ram and resume. It always stops working after a restart, suspend-to-disk or any other kind of reboot. Reloading all snd* modules makes no change. If it's not working, it stays not working, if it's working, it stays working. Suspending and resuming fixes it even if all the snd* modules are unloaded before suspending and loaded after resuming. When working (after suspend and resume), the labels shown in alsamixer are wrong - for example, Headphones now controls the internal speakers, while the Speakers slider does exactly nothing. Loading the module with generic, nofixup, ref models did nothing. I installed hda-jack-retask to see if I could make it work that way, but I was unable to make hda-jack-retask do anything at all - even setting every pin to not connected results in absolutely no change in operation. One forum thread on this problem, with some hdajackretask screenshots, is: http://www.toughbooktalk.com/viewtopic.php?f=39&t=1909 I've attached three alsa-info.sh files. The "broken" one is with sound not working. I then suspended and resumed, then generated the "working" one. The third is a diff between them, to save people time trying to spot changes. Thanks, --Randy

1 0

[alsa-devel] [PATCHv2 1/2] ALSA: fix emu8000 DRAM sizing for AWE64 Value
by David Flater 11 Jan '15

11 Jan '15

Applicable to any kernel since 2013: The special case added in commit 1338fc97d07a did not handle the possibility that the address space on an AWE64 Value would wrap around at 512 KiB. That is what it does, so the memory is still not detected on those cards. Fix that with a logic clean-up that eliminates the need for a special case. Signed-off-by: David Flater <dave(a)flaterco.com> --- History: 2015-01-09 v2: In response to feedback from Takashi Iwai, 1. Optimize for diff size. 1a. Use goto to avoid indenting and repeating code. 1b. Jettison new debugging printouts. 2. Split printk into second patch. Retested on CT4390 (4 MiB) and CT4380 (512 KiB). 2015-01-08 v1 patch sent to LKML, Alsa Devel and maintainers. Tested on unexpanded CT4390 (4 MiB), CT4520 (512 KiB), and CT4380 (512 KiB). CT4380 is commonly said to come with 1 MiB of DRAM, but Creative's AWE Control app agreed that mine has only 512 KiB. It has the same memory chip as the CT4520. The affected function first appeared in alsa-driver-0.3.0 and was merged in linux-2.5.5. Its somewhat different ancestor was in sound/oss/awe_wave.c. sound/isa/sb/emu8000.c | 33 +++++++++++++++------------------ 1 file changed, 15 insertions(+), 18 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 45fcdff..3dcf80e 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -378,13 +378,12 @@ init_arrays(struct snd_emu8000 *emu) static void size_dram(struct snd_emu8000 *emu) { - int i, size, detected_size; + int i, size; if (emu->dram_checked) return; size = 0; - detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -392,11 +391,12 @@ size_dram(struct snd_emu8000 *emu) EMU8000_SMALW_WRITE(emu, EMU8000_DRAM_OFFSET); EMU8000_SMLD_WRITE(emu, UNIQUE_ID1); snd_emu8000_init_fm(emu); /* This must really be here and not 2 lines back even */ + snd_emu8000_write_wait(emu); + + goto size_dram_ID1_check; /* If that fails, we have no RAM. */ while (size < EMU8000_MAX_DRAM) { - size += 512 * 1024; /* increment 512kbytes */ - /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -424,23 +424,20 @@ size_dram(struct snd_emu8000 *emu) * wraps back to the beginning; so check to see if the * initial value has been overwritten. */ + + size_dram_ID1_check: EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); EMU8000_SMLD_READ(emu); /* discard stale data */ if (EMU8000_SMLD_READ(emu) != UNIQUE_ID1) - break; /* we must have wrapped around */ + break; snd_emu8000_read_wait(emu); - /* Otherwise, it's valid memory. */ - detected_size = size + 512 * 1024; - } - - /* Distinguish 512 KiB from 0. */ - if (detected_size == 0) { - snd_emu8000_read_wait(emu); - EMU8000_SMALR_WRITE(emu, EMU8000_DRAM_OFFSET); - EMU8000_SMLD_READ(emu); /* discard stale data */ - if (EMU8000_SMLD_READ(emu) == UNIQUE_ID1) - detected_size = 512 * 1024; + /* + * Otherwise, it's valid memory. If a write succeeds at + * the beginning of a 512 KiB page we assume that the whole + * page is there. + */ + size += 512 * 1024; /* increment 512 KiB */ } /* wait until FULL bit in SMAxW register is false */ @@ -455,9 +452,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, detected_size/1024); + emu->port1, size/1024); - emu->mem_size = detected_size; + emu->mem_size = size; emu->dram_checked = 1; } -- 1.8.4

2 2