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

[alsa-devel] [PATCH - hwmixvolume 9/9] hwmixvolume: mention the new dependencies in the README
by Emmanuel Gil Peyrot 14 Aug '18

14 Aug '18

Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/README b/hwmixvolume/README index 9884ec2..db1b874 100644 --- a/hwmixvolume/README +++ b/hwmixvolume/README @@ -8,7 +8,8 @@ that use hardware mixing, i.e., those based on the following chips: * Yamaha DS-1 (YMF-724/740/744/754) (driver: snd-ymfpci) -This tool requires Python, pygtk, and alsa-pyton 1.0.22 or later. +This tool requires Python, PyGObject, GTK+ 3.0 or later, and alsa-pyton 1.0.22 +or later. It is recommended to use at least Linux kernel 2.6.32 or alsa-driver 1.0.22; otherwise, the name of the program that is using a stream cannot be shown. -- 2.18.0

2 1

[alsa-devel] [PATCH] uapi: fix sound/skl-tplg-interface.h userspace compilation errors
by Dmitry V. Levin 13 Aug '18

13 Aug '18

Include <linux/types.h> and consistently use types it provides to fix the following sound/skl-tplg-interface.h userspace compilation errors: /usr/include/sound/skl-tplg-interface.h:146:2: error: unknown type name 'u32' u32 set_params:2; /usr/include/sound/skl-tplg-interface.h:147:2: error: unknown type name 'u32' u32 rsvd:30; /usr/include/sound/skl-tplg-interface.h:148:2: error: unknown type name 'u32' u32 param_id; /usr/include/sound/skl-tplg-interface.h:149:2: error: unknown type name 'u32' u32 max; /usr/include/sound/skl-tplg-interface.h:166:2: error: unknown type name 'u16' u16 module_id; /usr/include/sound/skl-tplg-interface.h:167:2: error: unknown type name 'u16' u16 instance_id; /usr/include/sound/skl-tplg-interface.h:171:2: error: unknown type name 'u32' u32 channels; /usr/include/sound/skl-tplg-interface.h:172:2: error: unknown type name 'u32' u32 freq; /usr/include/sound/skl-tplg-interface.h:173:2: error: unknown type name 'u32' u32 bit_depth; /usr/include/sound/skl-tplg-interface.h:174:2: error: unknown type name 'u32' u32 valid_bit_depth; /usr/include/sound/skl-tplg-interface.h:175:2: error: unknown type name 'u32' u32 ch_cfg; /usr/include/sound/skl-tplg-interface.h:176:2: error: unknown type name 'u32' u32 interleaving_style; /usr/include/sound/skl-tplg-interface.h:177:2: error: unknown type name 'u32' u32 sample_type; /usr/include/sound/skl-tplg-interface.h:178:2: error: unknown type name 'u32' u32 ch_map; /usr/include/sound/skl-tplg-interface.h:182:2: error: unknown type name 'u32' u32 set_params:2; /usr/include/sound/skl-tplg-interface.h:183:2: error: unknown type name 'u32' u32 rsvd:30; /usr/include/sound/skl-tplg-interface.h:184:2: error: unknown type name 'u32' u32 param_id; /usr/include/sound/skl-tplg-interface.h:185:2: error: unknown type name 'u32' u32 caps_size; /usr/include/sound/skl-tplg-interface.h:186:2: error: unknown type name 'u32' u32 caps[HDA_SST_CFG_MAX]; /usr/include/sound/skl-tplg-interface.h:190:2: error: unknown type name 'u8' u8 pipe_id; /usr/include/sound/skl-tplg-interface.h:191:2: error: unknown type name 'u8' u8 pipe_priority; /usr/include/sound/skl-tplg-interface.h:192:2: error: unknown type name 'u16' u16 conn_type:4; /usr/include/sound/skl-tplg-interface.h:193:2: error: unknown type name 'u16' u16 rsvd:4; /usr/include/sound/skl-tplg-interface.h:194:2: error: unknown type name 'u16' u16 memory_pages:8; /usr/include/sound/skl-tplg-interface.h:200:2: error: unknown type name 'u16' u16 module_id; /usr/include/sound/skl-tplg-interface.h:201:2: error: unknown type name 'u16' u16 instance_id; /usr/include/sound/skl-tplg-interface.h:202:2: error: unknown type name 'u32' u32 max_mcps; /usr/include/sound/skl-tplg-interface.h:203:2: error: unknown type name 'u32' u32 mem_pages; /usr/include/sound/skl-tplg-interface.h:204:2: error: unknown type name 'u32' u32 obs; /usr/include/sound/skl-tplg-interface.h:205:2: error: unknown type name 'u32' u32 ibs; /usr/include/sound/skl-tplg-interface.h:206:2: error: unknown type name 'u32' u32 vbus_id; /usr/include/sound/skl-tplg-interface.h:208:2: error: unknown type name 'u32' u32 max_in_queue:8; /usr/include/sound/skl-tplg-interface.h:209:2: error: unknown type name 'u32' u32 max_out_queue:8; /usr/include/sound/skl-tplg-interface.h:210:2: error: unknown type name 'u32' u32 time_slot:8; /usr/include/sound/skl-tplg-interface.h:211:2: error: unknown type name 'u32' u32 core_id:4; /usr/include/sound/skl-tplg-interface.h:212:2: error: unknown type name 'u32' u32 rsvd1:4; /usr/include/sound/skl-tplg-interface.h:214:2: error: unknown type name 'u32' u32 module_type:8; /usr/include/sound/skl-tplg-interface.h:215:2: error: unknown type name 'u32' u32 conn_type:4; /usr/include/sound/skl-tplg-interface.h:216:2: error: unknown type name 'u32' u32 dev_type:4; /usr/include/sound/skl-tplg-interface.h:217:2: error: unknown type name 'u32' u32 hw_conn_type:4; /usr/include/sound/skl-tplg-interface.h:218:2: error: unknown type name 'u32' u32 rsvd2:12; /usr/include/sound/skl-tplg-interface.h:220:2: error: unknown type name 'u32' u32 params_fixup:8; /usr/include/sound/skl-tplg-interface.h:221:2: error: unknown type name 'u32' u32 converter:8; /usr/include/sound/skl-tplg-interface.h:222:2: error: unknown type name 'u32' u32 input_pin_type:1; /usr/include/sound/skl-tplg-interface.h:223:2: error: unknown type name 'u32' u32 output_pin_type:1; /usr/include/sound/skl-tplg-interface.h:224:2: error: unknown type name 'u32' u32 is_dynamic_in_pin:1; /usr/include/sound/skl-tplg-interface.h:225:2: error: unknown type name 'u32' u32 is_dynamic_out_pin:1; /usr/include/sound/skl-tplg-interface.h:226:2: error: unknown type name 'u32' u32 is_loadable:1; /usr/include/sound/skl-tplg-interface.h:227:2: error: unknown type name 'u32' u32 rsvd3:11; Fixes: 0c24fdc00244 ("ASoC: topology: Move skl-tplg-interface.h to uapi") Cc: <stable(a)vger.kernel.org> # v4.18 Signed-off-by: Dmitry V. Levin <ldv(a)altlinux.org> --- include/uapi/sound/skl-tplg-interface.h | 106 ++++++++++++------------ 1 file changed, 54 insertions(+), 52 deletions(-) diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index f58cafa42f18..f39352cef382 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -10,6 +10,8 @@ #ifndef __HDA_TPLG_INTERFACE_H__ #define __HDA_TPLG_INTERFACE_H__ +#include <linux/types.h> + /* * Default types range from 0~12. type can range from 0 to 0xff * SST types start at higher to avoid any overlapping in future @@ -143,10 +145,10 @@ enum skl_module_param_type { }; struct skl_dfw_algo_data { - u32 set_params:2; - u32 rsvd:30; - u32 param_id; - u32 max; + __u32 set_params:2; + __u32 rsvd:30; + __u32 param_id; + __u32 max; char params[0]; } __packed; @@ -163,68 +165,68 @@ enum skl_tuple_type { /* v4 configuration data */ struct skl_dfw_v4_module_pin { - u16 module_id; - u16 instance_id; + __u16 module_id; + __u16 instance_id; } __packed; struct skl_dfw_v4_module_fmt { - u32 channels; - u32 freq; - u32 bit_depth; - u32 valid_bit_depth; - u32 ch_cfg; - u32 interleaving_style; - u32 sample_type; - u32 ch_map; + __u32 channels; + __u32 freq; + __u32 bit_depth; + __u32 valid_bit_depth; + __u32 ch_cfg; + __u32 interleaving_style; + __u32 sample_type; + __u32 ch_map; } __packed; struct skl_dfw_v4_module_caps { - u32 set_params:2; - u32 rsvd:30; - u32 param_id; - u32 caps_size; - u32 caps[HDA_SST_CFG_MAX]; + __u32 set_params:2; + __u32 rsvd:30; + __u32 param_id; + __u32 caps_size; + __u32 caps[HDA_SST_CFG_MAX]; } __packed; struct skl_dfw_v4_pipe { - u8 pipe_id; - u8 pipe_priority; - u16 conn_type:4; - u16 rsvd:4; - u16 memory_pages:8; + __u8 pipe_id; + __u8 pipe_priority; + __u16 conn_type:4; + __u16 rsvd:4; + __u16 memory_pages:8; } __packed; struct skl_dfw_v4_module { char uuid[SKL_UUID_STR_SZ]; - u16 module_id; - u16 instance_id; - u32 max_mcps; - u32 mem_pages; - u32 obs; - u32 ibs; - u32 vbus_id; - - u32 max_in_queue:8; - u32 max_out_queue:8; - u32 time_slot:8; - u32 core_id:4; - u32 rsvd1:4; - - u32 module_type:8; - u32 conn_type:4; - u32 dev_type:4; - u32 hw_conn_type:4; - u32 rsvd2:12; - - u32 params_fixup:8; - u32 converter:8; - u32 input_pin_type:1; - u32 output_pin_type:1; - u32 is_dynamic_in_pin:1; - u32 is_dynamic_out_pin:1; - u32 is_loadable:1; - u32 rsvd3:11; + __u16 module_id; + __u16 instance_id; + __u32 max_mcps; + __u32 mem_pages; + __u32 obs; + __u32 ibs; + __u32 vbus_id; + + __u32 max_in_queue:8; + __u32 max_out_queue:8; + __u32 time_slot:8; + __u32 core_id:4; + __u32 rsvd1:4; + + __u32 module_type:8; + __u32 conn_type:4; + __u32 dev_type:4; + __u32 hw_conn_type:4; + __u32 rsvd2:12; + + __u32 params_fixup:8; + __u32 converter:8; + __u32 input_pin_type:1; + __u32 output_pin_type:1; + __u32 is_dynamic_in_pin:1; + __u32 is_dynamic_out_pin:1; + __u32 is_loadable:1; + __u32 rsvd3:11; struct skl_dfw_v4_pipe pipe; struct skl_dfw_v4_module_fmt in_fmt[MAX_IN_QUEUE]; -- ldv

2 1

[alsa-devel] [GIT PULL] ASoC updates for v4.19
by Mark Brown 13 Aug '18

13 Aug '18

The following changes since commit 1ffaddd029c867d134a1dde39f540dcc8c52e274: Linux 4.18-rc8 (2018-08-05 12:37:41 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.19 for you to fetch changes up to 4aa5db22d35588e1a5d2ee88472348ea73d9fb23: Merge branch 'asoc-4.19' into asoc-next (2018-08-09 14:47:05 +0100) ---------------------------------------------------------------- ASoC: Updates for v4.19 A fairly big update, including quite a bit of core activity this time around (which is good to see) along with a fairly large set of new drivers. - A new snd_pcm_stop_xrun() helper which is now used in several drivers. - Support for providing name prefixes to generic component nodes. - Quite a few fixes for DPCM as it gains a bit wider use and more robust testing. - Generalization of the DIO2125 support to a simple amplifier driver. - Accessory detection support for the audio graph card. - DT support for PXA AC'97 devices. - Quirks for a number of new x86 systems. - Support for AM Logic Meson, Everest ES7154, Intel systems with RT5682, Qualcomm QDSP6 and WCD9335, Realtek RT5682 and TI TAS5707. ---------------------------------------------------------------- Adam Thomson (1): ASoC: da7219: Add delays to capture path to remove DC offset noise Agrawal, Akshu (4): ASoC: AMD: Add NULL pointer check ASoC: AMD: Change codec to channel link as per hardware redesign ASoC: AMD: Send correct channel for configuring DMA descriptors ASoC: AMD: For capture have interrupts on I2S->ACP channel Ajit Pandey (1): ASoC: wm_adsp: Correct DSP pointer for preloader control Akshu Agrawal (6): ASoC: AMD: Configure channel 1 or channel 0 for capture ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002 ASoC: AMD: Fix build warning ASoC: soc-pcm: Use delay set in component pointer function ASoC: AMD: Make ACP->SYSMEM DMA non circular ASoC: AMD: Set delay value for the capture case Alexey Khoroshilov (3): ASoC: rockchip: put device_node on remove ASoC: tegra: fix device_node refcounting ASoC: tegra_alc5632: fix device_node refcounting Andrew Gabbasov (1): ASoC: rsnd: cmd: Add missing newline to debug message Arnaud Pouliquen (2): MAINTAINERS: add entry for STI audio drivers MAINTAINERS: add entry for STM32 audio drivers Arnd Bergmann (2): ASoC: pxa: make SND_PXA_SOC_SSP depend on PLAT_PXA ASoC: meson: axg-spdifout: select SND_PCM_IEC958 Bard Liao (3): ASoC: rt5682: add rt5682 codec driver ASoC: rt5682: add button detection mode control ASoC: rt5631: add Volume to the name of volume control Charles Keepax (5): ASoC: pcm: Tidy up open/hw_params handling ASoC: wm_adsp: Simplify handling of alg offset and length ASoC: arizona: Set compressed IRQ to a wake source ASoC: wm_adsp: Correct algorithm list allocation size ASoC: wm_adsp: Take prefix into account in control name length Codrin Ciubotariu (2): ASoC: atmel-i2s: Remove unnecessary audio PLL clock (aclk) ASoC: atmel-i2s: dt-bindings: Remove unnecessary phandle to aclk Colin Ian King (5): ASoC: twl6040: make pointer dmic_codec_dev static ASoC: ak5558: make two structures static ASoC: ak4458: make structure soc_codec_dev_ak4458 static const ASoC: nau8540: remove redundant variable osrate ASoC: stm32: remove redundant pointers 'priv' and 'rtd' Dan Carpenter (3): ASoC: qdsp6: qdafe: fix some off by one bugs ASoC: qdsp6: q6afe-dai: fix a range check in of_q6afe_parse_dai_data() ASoC: qdsp6: q6routing: off by one in routing_hw_params() Daniel Kurtz (6): ASoC: AMD: Always stop ch2 first ASoC: AMD: Always subtract bytescount ASoC: AMD: Fix Capture DMA channel names ASoC: AMD: Do not generate interrups for every captured sample ASoC: AMD: Reset bytescount when starting transaction ASoC: AMD: Simplify trigger handler Daniel Mack (11): ASoC: fold pxa2xx-pcm into its only user, pxa2xx-ac97 ASoC: pxa: clean up function names in pxa2xx-lib ASoC: pxa: move some functions to pxa2xx-lib ASoC: pxa2xx-pcm-lib: fix indenting ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components ASoC: pxa: remove bindings from pxa2xx-pcm ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable ASoC: pxa: select SND_PXA2XX_LIB for drivers that depend on it ASoC: pxa-ssp: remove .set_pll() and .set_clkdiv() callbacks ASoC: pxa-ssp: add support for an external clock in devicetree ASoC: pxa: remove clock divider and pll setup from zylonite and magician Edward Cragg (1): ASoC: tegra: i2s: Fix typo/broken macro Fabio Estevam (5): ASoC: soc-utils: Fix unregistration order ASoC: fsl-asoc-card: Switch to SPDX identifier ASoC: fsl_asrc: Switch to SPDX identifier ASoC: fsl_utils: Switch to SPDX identifier ASoC: imx-sgtl5000: Switch to SPDX identifier Geert Uytterhoeven (1): ASoC: rt1305: Use ULL suffixes for 64-bit constants Gustavo A. R. Silva (37): ASoC: pxa-ssp: mark expected switch fall-through ASoC: nau8825: use 64-bit arithmetic instead of 32-bit ASoC: fsl_spdif: Use 64-bit arithmetic instead of 32-bit ASoC: nau8824: use 64-bit arithmetic instead of 32-bit ASoC: sirf: Fix potential NULL pointer dereference ASoC: wm8961: Mark expected switch fall-through ASoC: rt5640: Mark expected switch fall-through ASoC: rt5677: Mark expected switch fall-through ASoC: wm8955: Mark expected switch fall-through ASoC: wm8960: Mark expected switch fall-through ASoC: wm8904: Mark expected switch fall-through ASoC: wm8996: Mark expected switch fall-through ASoC: wm8962: Mark expected switch fall-through ASoC: wm8995: Mark expected switch fall-through ASoC: wm9081: Mark expected switch fall-through ASoC: wm8994: Mark expected switch fall-through ASoC: smd845: fix memory leak ASoC: davinci-i2s: mark expected switch fall-through ASoC: omap-mcpdm: Mark expected switch fall-throughs ASoC: core: mark expected switch fall-through ASoC: fsl_esai: Mark expected switch fall-through ASoC: omap-dmic: Mark expected switch fall-throughs ASoC: samsung: i2s: Mark expected switch fall-through ASoC: max9850: use true and false for boolean values ASoC: rt5631: use true and false for boolean values ASoC: tda7419: use true and false for boolean values ASoC: wm8990: use true and false for boolean values ASoC: cs4270: use true and false for boolean values ASoC: wm8996: use true and false for boolean values ASoC: da7219: use true and false for boolean values ASoC: twl6040: use true and false for boolean values ASoC: da7213: use true and false for boolean values ASoC: wm5100-tables: use true and false for boolean values ASoC: da9055: use true and false for boolean values ASoC: wm8903: use true and false for boolean values ASoC: wm8994: Fix missing break in switch ASoC: adav80x: mark expected switch fall-through Hans de Goede (28): ASoC: Intel: bytcr_rt5640: Add quirk for the Chuwi Vi10 tablet ASoC: Intel: bytcr_rt5640: Add quirk for the Nuvison/TMax TM800W560 tablet ASoC: Intel: bytcr_rt5640: Fix Acer Iconia 8 over-current detect threshold ASoC: Intel: bytcr_rt5651: Add BYT_RT5651_DEFAULT_QUIRKS define ASoC: Intel: bytcr_rt5651: Change default input map from in2 to in1 ASoC: Intel: bytcr_rt5651: Fix IN1_IN2_MAP quirk not being logged ASoC: Intel: bytcr_rt5651: Remove IN2 input mappings ASoC: Intel: bytcr_rt5651: Fix IN1 map headsetmic mapping ASoC: Intel: bytcr_rt5651: Fix DMIC map headsetmic mapping ASoC: Intel: bytcr_rt5651: Simplify card long-name ASoC: Intel: bytcr_rt5651: Add BYT_RT5651_HP_LR_SWAPPED quirk ASoC: Intel: bytcr_rt5651: Sort DMI table entries alphabetically ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix2 8 tablet ASoC: Intel: bytcr_rt5651: Remove is_valleyview helper ASoC: Intel: bytcr_rt5651: Move getting of codec_dev into probe() ASoC: Intel: bytcr_rt5651: Add support for externar amplifier enable GPIO ASoC: rt5651: Fix workqueue cancel vs irq free race on remove ASoC: rt5651: Allow disabling jack-detect by calling set_jack(NULL) ASoC: rt5651: Add button press support ASoC: Intel: bytcr_rt5651: Disable jack-detect over suspend/resume ASoC: Intel: bytcr_rt5651: Reporting button presses ASoC: rt5651: Add IN3 Boost volume control ASoC: Intel: bytcr_rt5651: Fix using the wrong GPIO for the ext-amp on some boards ASoC: Intel: bytcr_rt5651: Set OVCD limit for VIOS LTH17 to 2000uA ASoC: Intel: bytcr_rt5651: Add IN2 input mapping ASoC: Intel: bytcr_rt5651: Add mono speaker quirk ASoC: Intel: bytcr_rt5651: Add quirk table entries for various devices ASoC: Intel: bytcr_rt5640: Add quirk for the "Connect Tablet 9" tablet Hiroyuki Yokoyama (1): ASoC: rsnd: Document R-Car M3-N support Janusz Krzysztofik (1): ASoC: cx20442: Don't ignore regulator_get() errors. Jerome Brunet (39): clk: add duty cycle support ASoC: simple-amplifier: remame dio2125 documentation ASoC: simple-amplifier: rename dio2125 to simple-amplifer ASoC: simple-amplifer: add simple-amplifier compatible ASoC: tas571x: add tas5707 compatible ASoC: tas517x: add tas5707 support ASoC: dpcm: improve runtime update predictability ASoC: dpcm: extend channel merging to the backend cpu dai ASoC: dpcm: don't merge format from invalid codec dai ASoC: es7134: remove 64kHz rate from the supported rates ASoC: es7134: check if mclk rate is valid ASoC: es7134: Add VDD and AVDD power supplies ASoC: es7241: add dt-bindings documentation for the es7241 adc ASoC: es7241: add es7241 codec support ASoC: es7134: update DT binding with new compatible and supplies ASoC: es7134: correct required power supplies ASoC: es7134: add support for the es7154 ASoC: dpcm: rework runtime stream merge ASoC: dpcm: add rate merge to the BE stream merge ASoC: add DT documentation for the sound-name-prefix property ASoC: allow soc-core to pick up name prefixes from component nodes ASoC: meson: add axg fifos DT binding documentation ASoC: meson: add axg fifo base driver ASoC: meson: add axg frddr driver ASoC: meson: add axg toddr driver ASoC: meson: add axg spdif output DT bindings documentation ASoC: meson: add axg spdif output ASoC: meson: add axg tdm formatters DT bindings documentation ASoC: meson: add axg tdm interface DT bindings documentation ASoC: meson: add tdm formatter base driver ASoC: meson: add tdm interface driver ASoC: meson: add tdm output driver ASoC: meson: add tdm input driver ASoC: export snd_soc_of_get_slot_mask ASoC: meson: add axg sound card DT bindings documentation ASoC: meson: add axg sound card support ASoC: meson: use IRQ_RETVAL in the fifo irq handler ASoC: meson: update axg sound card bindings ASoC: meson: align axg card driver with DT bindings documentation Jiada Wang (1): ASoC: soc-pcm: DPCM cares BE channel constraint Jianqun Xu (1): ASoC: rockchip: add config for rockchip dmaengine pcm register KaiChieh Chuang (2): ASoC: mediatek: sub dai use list_head ASoC: mt6797: sub dai use list_head Katsuhiro Suzuki (7): ASoC: simple-card-utils: move hp and mic detect gpios from simple-card ASoC: simple-card: move hp and mic detection to soc_card probe ASoC: audio-graph-card: add hp and mic detect gpios same as simple-card ASoC: uniphier: remove redundant check of PLL ID ASoC: add hp-det-gpio and mic-det-gpio to audio graph card binding ASoC: uniphier: add support for multichannel output ASoC: uniphier: change functions to static Kuninori Morimoto (37): ASoC: fsi: convert to SPDX identifiers ASoC: hac: convert to SPDX identifiers ASoC: ssi: convert to SPDX identifiers ASoC: siu: convert to SPDX identifiers ASoC: rsnd: convert to SPDX identifiers ASoC: migor: convert to SPDX identifiers ASoC: dma-sh7760: convert to SPDX identifiers ASoC: sh7760-ac97: convert to SPDX identifiers ASoC: rsnd: has .symmetric_rates if SSIs are sharing WS pin ASoC: rsnd: SSI parent cares SWSP bit ASoC: rsnd: add rsnd_daidrv_get() ASoC: simple-card: convert to SPDX identifiers ASoC: simple-card-utils: convert to SPDX identifiers ASoC: simple-scu-card.c: convert to SPDX identifiers ASoC: audio-graph-card.c: convert to SPDX identifiers ASoC: audio-graph-scu-card.c: convert to SPDX identifiers ASoC: soc-io.c: convert to SPDX identifiers ASoC: soc-dai.h: convert to SPDX identifiers ASoC: soc-ops.c: convert to SPDX identifiers ASoC: soc-pcm.c: convert to SPDX identifiers ASoC: soc-dpcm.h: convert to SPDX identifiers ASoC: soc-jack.c: convert to SPDX identifiers ASoC: soc-utils.c: convert to SPDX identifiers ASoC: soc-devres.c: convert to SPDX identifiers ASoC: soc-acpi: convert to SPDX identifiers ASoC: soc-core: convert to SPDX identifiers ASoC: soc-dapm: convert to SPDX identifiers ASoC: soc-topology: convert to SPDX identifiers ASoC: soc-compress: convert to SPDX identifiers ASoC: soc-generic-dmaengine-pcm: convert to SPDX identifiers ASoC: ac97: convert to SPDX identifiers ASoC: ak4642: convert to SPDX identifiers ASoC: ak4554: convert to SPDX identifiers ASoC: da7210: convert to SPDX identifiers ASoC: ak4613: convert to SPDX identifiers ASoC: sh: Kconfig: convert to SPDX identifiers ASoC: fsi: convert to SPDX identifiers Ladislav Michl (1): ASoC: wm8988: fix typo in rate constraints Lars-Peter Clausen (1): ASoC: adau171x1: Connect playback DAI to the DSP Liam Girdwood (4): ASoC: dapm: Fix potential DAI widget pointer deref when linking DAIs ASoC: topology: Give more data to clients via callbacks ASoC: topology: Add callback for DAPM route load/unload ASoC: core: Allow topology to override machine driver FE DAI link config. Liang Chen (1): ASoC: rockchip-i2s: add description for px30 Mac Chiang (1): ASoC: Intel: kbl_da7219_max98357a: add fe_ops for kbl Audio Capture Port Marcel Ziswiler (2): ASoC: sgtl5000: fix spelling in devicetree binding document ASoC: tegra: improve goto error label Mark Brown (12): Merge branch 'asoc-4.17' into asoc-linus Merge branch 'asoc-4.18' into asoc-next Merge branch 'asoc-4.18' into asoc-4.19 for amd dep Merge tag 'pxa-for-4.19-dma_slave_map' of https://github.com/rjarzmik/linux into asoc-4.19 for ac'97 deps Merge tag 'alsa-add-snd-sgbuf-aligned-pages' of https://git.kernel.org/.../broonie/sound into asoc-4.19 Merge branch 'topic/drm_audio_component' of https://git.kernel.org/.../tiwai/sound into asoc-4.19 Merge tag 'clk-core-duty-cycle-for-mark' of git://git.kernel.org/.../clk/linux into asoc-4.19 ASoC: wcd9335: Fix build due to CLASS-H Controller support ASoC: wcd9335: Fix build Merge branch 'asoc-4.18' into asoc-4.19 wmadsp dep Merge branch 'asoc-4.18' into asoc-linus Merge branch 'asoc-4.19' into asoc-next Matthias Kaehlcke (1): ASoC: rt5677: Fix initialization of rt5677_of_match.data Mukunda, Vijendar (1): ASoC: AMD: Modified DMA pointer for capture Naveen Manohar (3): ASoC: Intel: broxton: reduce machine name for bxt_da7219_max98357a ASoC: Intel: common: Add Geminilake Dialog+Maxim machine driver entry ASoC: Intel: Boards: Add GLK Realtek Maxim I2S machine driver Oder Chiou (1): ASoC: rt5514: Fix the issue of the delay volume applied Pierre-Louis Bossart (13): ASoC: Intel: Skylake: cleanup before moving ACPI tables ASoC: Intel: move SKL+ codec ACPI tables to common directory ASoC: Intel: common: add firmware/topology information for SOF ASoC: Intel: common: add entries for SOF-based machine drivers ASoC: Intel: common: fix copy/paste issue with SOF/broadwell topology file ASoC: Intel: common: rename 'reef' to 'sof' in ACPI matching table ASoC: Intel: common: fix missing rename from 'reef' to 'sof' ASoC: Intel: Haswell: fix endianness handling ASoC: Intel: Skylake: BDL definitions should be __le32 ASoC: Intel: common: make sst_dma functions static ASoC: Intel: Atom: simplify iomem address and casts ASoC: Intel: Atom: fix inversion between __iowrite32 and __ioread32 ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally Rakesh Ughreja (8): ALSA: hdac: Remove usage of struct hdac_ext_device and use hdac_device instead ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead ALSA: hdac: ext: add wait for codec to respond after link reset ALSA: hda: split snd_hda_codec_new function ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init ALSA: hdac: add extended ops in the hdac_bus ASoC: Intel: Skylake: fix widget handling Richard Fitzgerald (4): ASoC: wm_adsp: Allow up to 8 channels for voice control ASoC: wm_adsp: Parse HOST_BUFFER controls ASoC: wm_adsp: Declare firmware controls from codec driver ASoC: wm_adsp: Make DSP name configurable by codec driver Robert Jarzmik (13): ASoC: pxa: add binding for pxa2xx-ac97 audio complex ASoC: pxa: add devicetree support dmaengine: pxa: use a dma slave map ARM: pxa: add dma slave map dmaengine: pxa: add a default requestor policy mmc: pxamci: remove the dmaengine compat need media: pxa_camera: remove the dmaengine compat need mtd: rawnand: marvell: remove the dmaengine compat need ata: pata_pxa: remove the dmaengine compat need dmaengine: pxa: document pxad_param ARM: pxa: remove the DMA IO resources ARM: pxa: change SSP DMA channels allocation ASoC: pxa: remove the dmaengine compat need Rohit Kumar (1): ASoC: qcom: apq8096: set card as device drvdata Rohit kumar (5): ASoC: qcom: dt-bindings: Add sdm845 machine bindings ASoC: dt-bindings: Update dt binding name for apq8096 ASoC: qcom: Add support to parse common audio device nodes ASoC: qcom: add sdm845 sound card support ASoC: qcom: Fix unmet dependency warning for SND_SOC_SDM845 Russell King (1): ASoC: hdmi-codec: fix routing Ryan Lee (1): ASoC: max98373: Added software reset register to readable registers Srinivas Kandagatla (36): ASoC: q6adm: dt-bindings: add compatible string to routing ASoC: q6asm: dt-bindings: add compatible string to dais ASoC: q6afe: dt-bindings: add compatible string to dais ASoC: qdsp6: q6adm: use of_platform_populate/depopulate() ASoC: qdsp6: q6asm: use of_platform_populate/depopulate() ASoC: qdsp6: q6afe: use of_platform_populate/depopulate() ASoC: qdsp6: q6afe-dai: support dt based module loading ASoC: qdsp6: q6asm-dai: support dt based module loading ASoC: qdsp6: q6routing: support dt based module loading ASoC: qcom: apq8096: remove redundant owner assignment ASoC: qdsp6: q6routing: add proper error check ASoC: qdsp6: q6asm: remove unused struct q6asm member ASoC: qdsp6: q6afe-dai: do not close port if its not opened ASoC: qdsp6: q6asm-dai: do not close port if its not opened ASoC: qdsp6: q6afe: Add missing slimbus capture ports ASoC: qdsp6: q6afe-dai: add support to slim tx dais ASoC: qdsp6: q6routing: add slim rx routings ASoC: qdsp6: q6afe-dai: Do not overwrite slim dai num_channels ASoC: core: add support to card re-bind using component framework ASoC: qdsp6: q6afe-dai: remove component fw related code ASoC: qdsp6: q6asm-dai: remove component framework related code ASoC: qdsp6: q6routing: remove component framework related code ASoC: qcom: apq8096: remove component framework related code ASoC: core: add support to snd_soc_dai_get_channel_map() ASoC: dt-bindings: add dt bindings for wcd9335 audio codec ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: apq8096: remove auto rebinding ASoC: smd845: remove auto rebinding ASoC: core: remove support for card rebind using component framework ASoC: qcom: make common.c as proper module ASoC: apq8096: remove unused header files ASoC: sdm845: remove unused header files ASoC: qcom: remove unused header files from common.h ASoC: qdsp6: q6afe-dai: add SLIM tx AIF_IN dapm ASoC: compress: make BE and FE order inline with dpcm Sriram Periyasamy (1): ASoC: hdac_hdmi: Add documentation for power management Stefan Agner (1): ASoC: codecs: PCM1789: unconditionally flush work Steven Eckhoff (3): ASoC: TSCS42xx: Add mic bias boost control ASoC: TSCS42xx: Remove Playback/Capture in names ASoC: TSCS42xx: Add headphone auto switching Stuart Henderson (1): ASoC: wm_adsp: Ensure DSP boot work complete before preloader_put return Takashi Iwai (18): ASoC: davinci: Use snd_pcm_stop_xrun() helper ASoC: qcom: Use snd_pcm_stop_xrun() helper ASoC: sti: Use snd_pcm_stop_xrun() helper ASoC: stm32: Use snd_pcm_stop_xrun() helper Merge branch 'topic/hda-core-intel' into topic/hda-acomp drm/i915: Split audio component to a generic type ALSA: hda/i915: Associate audio component with devres ALSA: hda: Make audio component support more generic ASoC: zte: Fix incorrect PCM format bit usages ASoC: dmaengine: Fix missing __user prefix in copy_user callback ASoC: dapm: Use int for format bit position ALSA: pcm: Add snd_mask_set_format() helper for standard usages ASoC: doc: Replace open code with params_set_format() ASoC: intel: Fix snd_pcm_format_t handling ASoC: fsl: Use snd_mask_set_format() ASoC: pcm186x: Declare PCM format with snd_pcm_format_t ASoC: dmaengine: Use standard pcm_format_to_bits() macro ALSA: memalloc: Fix missing PAGE_SIZE definition Wei Yongjun (2): ASoC: rt5682: use devm_snd_soc_register_component() ASoC: rt1305: use devm_snd_soc_register_component() Yong Zhi (1): ASoC: Intel: bxt: Use refcap device for mono recording benjamin.gaignard(a)linaro.org (1): ASoC: stm32: replace "%p" with "%pK" olivier moysan (1): ASoC: stm32: sai: add iec958 controls support .../devicetree/bindings/sound/amlogic,axg-fifo.txt | 23 + .../bindings/sound/amlogic,axg-sound-card.txt | 124 + .../bindings/sound/amlogic,axg-spdifout.txt | 20 + .../bindings/sound/amlogic,axg-tdm-formatters.txt | 28 + .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 + .../devicetree/bindings/sound/atmel-i2s.txt | 11 +- .../devicetree/bindings/sound/audio-graph-card.txt | 2 + .../devicetree/bindings/sound/dioo,dio2125.txt | 12 - .../devicetree/bindings/sound/everest,es7134.txt | 7 +- .../devicetree/bindings/sound/everest,es7241.txt | 28 + .../bindings/sound/marvell,pxa2xx-ac97.txt | 27 + .../devicetree/bindings/sound/mrvl,pxa-ssp.txt | 8 + .../devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt | 15 - .../devicetree/bindings/sound/name-prefix.txt | 24 + .../devicetree/bindings/sound/qcom,apq8096.txt | 15 +- .../devicetree/bindings/sound/qcom,q6adm.txt | 6 + .../devicetree/bindings/sound/qcom,q6afe.txt | 6 + .../devicetree/bindings/sound/qcom,q6asm.txt | 6 + .../devicetree/bindings/sound/qcom,sdm845.txt | 80 + .../devicetree/bindings/sound/qcom,wcd9335.txt | 123 + .../devicetree/bindings/sound/renesas,rsnd.txt | 1 + .../devicetree/bindings/sound/rockchip-i2s.txt | 1 + Documentation/devicetree/bindings/sound/rt5682.txt | 50 + .../devicetree/bindings/sound/sgtl5000.txt | 2 +- .../devicetree/bindings/sound/simple-amplifier.txt | 12 + .../devicetree/bindings/sound/tas571x.txt | 1 + Documentation/sound/soc/dpcm.rst | 4 +- MAINTAINERS | 15 + arch/arm/mach-pxa/devices.c | 148 +- arch/arm/mach-pxa/devices.h | 6 +- arch/arm/mach-pxa/pxa25x.c | 38 +- arch/arm/mach-pxa/pxa27x.c | 39 +- arch/arm/mach-pxa/pxa3xx.c | 41 +- arch/arm/plat-pxa/ssp.c | 47 - drivers/ata/pata_pxa.c | 10 +- drivers/clk/clk.c | 199 +- drivers/dma/pxa_dma.c | 15 +- drivers/gpu/drm/i915/Kconfig | 1 + drivers/gpu/drm/i915/intel_audio.c | 22 +- drivers/media/platform/pxa_camera.c | 22 +- drivers/mmc/host/pxamci.c | 29 +- drivers/mtd/nand/raw/marvell_nand.c | 17 +- include/drm/drm_audio_component.h | 118 + include/drm/i915_component.h | 85 +- include/linux/clk-provider.h | 26 + include/linux/clk.h | 33 + include/linux/dma/pxa-dma.h | 9 + include/linux/platform_data/mmp_dma.h | 4 + include/linux/pxa2xx_ssp.h | 10 +- include/sound/ac97/codec.h | 8 +- include/sound/ac97/compat.h | 9 +- include/sound/ac97/controller.h | 8 +- include/sound/ac97/regs.h | 20 +- include/sound/ac97_codec.h | 25 +- include/sound/compress_driver.h | 21 +- include/sound/dmaengine_pcm.h | 14 +- include/sound/hda_component.h | 61 + include/sound/hda_i915.h | 37 +- include/sound/hdaudio.h | 35 +- include/sound/hdaudio_ext.h | 123 +- include/sound/memalloc.h | 18 +- include/sound/pcm_params.h | 10 +- include/sound/pxa2xx-lib.h | 13 +- include/sound/rt5682.h | 40 + include/sound/sh_fsi.h | 13 +- include/sound/simple_card.h | 7 +- include/sound/simple_card_utils.h | 23 +- include/sound/soc-acpi-intel-match.h | 19 +- include/sound/soc-acpi.h | 13 +- include/sound/soc-dai.h | 15 +- include/sound/soc-dapm.h | 11 +- include/sound/soc-dpcm.h | 7 +- include/sound/soc-topology.h | 37 +- include/sound/soc.h | 31 +- include/trace/events/clk.h | 36 + sound/arm/Kconfig | 5 - sound/arm/Makefile | 3 - sound/arm/pxa2xx-ac97-lib.c | 12 + sound/arm/pxa2xx-ac97.c | 124 +- sound/arm/pxa2xx-pcm-lib.c | 75 +- sound/arm/pxa2xx-pcm.c | 129 - sound/arm/pxa2xx-pcm.h | 27 - sound/hda/Kconfig | 7 +- sound/hda/Makefile | 1 + sound/hda/ext/hdac_ext_bus.c | 80 +- sound/hda/ext/hdac_ext_controller.c | 64 +- sound/hda/ext/hdac_ext_stream.c | 104 +- sound/hda/hdac_component.c | 335 +++ sound/hda/hdac_i915.c | 335 +-- sound/pci/hda/hda_codec.c | 68 +- sound/pci/hda/hda_codec.h | 2 + sound/pci/hda/patch_hdmi.c | 57 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 109 +- sound/soc/amd/acp-pcm-dma.c | 214 +- sound/soc/amd/acp.h | 13 +- sound/soc/atmel/atmel-i2s.c | 46 +- sound/soc/codecs/Kconfig | 25 +- sound/soc/codecs/Makefile | 8 +- sound/soc/codecs/adau17x1.c | 1 + sound/soc/codecs/adav80x.c | 1 + sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/ak4554.c | 17 +- sound/soc/codecs/ak4613.c | 26 +- sound/soc/codecs/ak4642.c | 26 +- sound/soc/codecs/ak5558.c | 4 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs47l24.c | 11 + sound/soc/codecs/cx20442.c | 23 +- sound/soc/codecs/da7210.c | 27 +- sound/soc/codecs/da7213.c | 4 +- sound/soc/codecs/da7219-aad.c | 5 + sound/soc/codecs/da7219.c | 48 +- sound/soc/codecs/da7219.h | 8 +- sound/soc/codecs/da9055.c | 4 +- sound/soc/codecs/es7134.c | 227 +- sound/soc/codecs/es7241.c | 322 +++ sound/soc/codecs/hdac_hdmi.c | 495 ++-- sound/soc/codecs/hdmi-codec.c | 21 +- sound/soc/codecs/max98373.c | 1 + sound/soc/codecs/max9850.c | 4 +- sound/soc/codecs/nau8540.c | 3 - sound/soc/codecs/nau8824.c | 2 +- sound/soc/codecs/nau8825.c | 2 +- sound/soc/codecs/pcm1789.c | 3 +- sound/soc/codecs/pcm186x.c | 2 +- sound/soc/codecs/rt1305.c | 15 +- sound/soc/codecs/rt5514.c | 8 +- sound/soc/codecs/rt5631.c | 12 +- sound/soc/codecs/rt5640.c | 2 + sound/soc/codecs/rt5651.c | 235 +- sound/soc/codecs/rt5651.h | 8 + sound/soc/codecs/rt5677.c | 3 +- sound/soc/codecs/rt5682.c | 2681 ++++++++++++++++++++ sound/soc/codecs/rt5682.h | 1324 ++++++++++ sound/soc/codecs/{dio2125.c => simple-amplifier.c} | 42 +- sound/soc/codecs/tas571x.c | 110 + sound/soc/codecs/tas571x.h | 16 + sound/soc/codecs/tda7419.c | 4 +- sound/soc/codecs/tscs42xx.c | 37 +- sound/soc/codecs/tscs42xx.h | 8 + sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/wm2200.c | 10 +- sound/soc/codecs/wm5100-tables.c | 12 +- sound/soc/codecs/wm5102.c | 10 + sound/soc/codecs/wm5110.c | 13 + sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8904.c | 1 + sound/soc/codecs/wm8955.c | 1 + sound/soc/codecs/wm8960.c | 1 + sound/soc/codecs/wm8961.c | 1 + sound/soc/codecs/wm8962.c | 1 + sound/soc/codecs/wm8988.c | 4 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/codecs/wm8994.c | 1 + sound/soc/codecs/wm8995.c | 1 + sound/soc/codecs/wm8996.c | 9 +- sound/soc/codecs/wm9081.c | 1 + sound/soc/codecs/wm_adsp.c | 216 +- sound/soc/codecs/wm_adsp.h | 12 +- sound/soc/codecs/wmfw.h | 1 + sound/soc/davinci/davinci-i2s.c | 1 + sound/soc/davinci/davinci-mcasp.c | 16 +- sound/soc/fsl/fsl-asoc-card.c | 20 +- sound/soc/fsl/fsl_asrc.c | 18 +- sound/soc/fsl/fsl_asrc.h | 5 +- sound/soc/fsl/fsl_asrc_dma.c | 18 +- sound/soc/fsl/fsl_esai.c | 1 + sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_utils.c | 18 +- sound/soc/fsl/fsl_utils.h | 7 +- sound/soc/fsl/imx-sgtl5000.c | 15 +- sound/soc/generic/audio-graph-card.c | 41 +- sound/soc/generic/audio-graph-scu-card.c | 25 +- sound/soc/generic/simple-card-utils.c | 74 +- sound/soc/generic/simple-card.c | 106 +- sound/soc/generic/simple-scu-card.c | 21 +- sound/soc/intel/atom/sst/sst_drv_interface.c | 29 +- sound/soc/intel/atom/sst/sst_loader.c | 6 +- sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5677.c | 4 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 20 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 55 +- sound/soc/intel/boards/bytcr_rt5651.c | 364 ++- sound/soc/intel/boards/glk_rt5682_max98357a.c | 643 +++++ sound/soc/intel/boards/kbl_da7219_max98357a.c | 3 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/common/Makefile | 6 +- sound/soc/intel/common/soc-acpi-intel-bxt-match.c | 59 + sound/soc/intel/common/soc-acpi-intel-byt-match.c | 40 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 56 +- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 32 + sound/soc/intel/common/soc-acpi-intel-glk-match.c | 41 + .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 16 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 91 + sound/soc/intel/common/soc-acpi-intel-skl-match.c | 47 + sound/soc/intel/common/sst-firmware.c | 6 +- sound/soc/intel/haswell/sst-haswell-dsp.c | 53 +- sound/soc/intel/skylake/skl-messages.c | 50 +- sound/soc/intel/skylake/skl-nhlt.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 119 +- sound/soc/intel/skylake/skl-sst-cldma.c | 8 +- sound/soc/intel/skylake/skl-sst-cldma.h | 2 +- sound/soc/intel/skylake/skl-topology.c | 28 +- sound/soc/intel/skylake/skl-topology.h | 11 +- sound/soc/intel/skylake/skl.c | 360 +-- sound/soc/intel/skylake/skl.h | 7 +- .../soc/mediatek/common/mtk-afe-platform-driver.c | 64 +- sound/soc/mediatek/common/mtk-base-afe.h | 6 +- sound/soc/mediatek/mt6797/mt6797-afe-common.h | 1 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 65 +- sound/soc/mediatek/mt6797/mt6797-dai-adda.c | 20 +- sound/soc/mediatek/mt6797/mt6797-dai-hostless.c | 16 +- sound/soc/mediatek/mt6797/mt6797-dai-pcm.c | 19 +- sound/soc/meson/Kconfig | 65 + sound/soc/meson/Makefile | 21 + sound/soc/meson/axg-card.c | 671 +++++ sound/soc/meson/axg-fifo.c | 341 +++ sound/soc/meson/axg-fifo.h | 80 + sound/soc/meson/axg-frddr.c | 141 + sound/soc/meson/axg-spdifout.c | 456 ++++ sound/soc/meson/axg-tdm-formatter.c | 381 +++ sound/soc/meson/axg-tdm-formatter.h | 39 + sound/soc/meson/axg-tdm-interface.c | 542 ++++ sound/soc/meson/axg-tdm.h | 78 + sound/soc/meson/axg-tdmin.c | 229 ++ sound/soc/meson/axg-tdmout.c | 259 ++ sound/soc/meson/axg-toddr.c | 199 ++ sound/soc/omap/omap-abe-twl6040.c | 2 +- sound/soc/omap/omap-dmic.c | 2 + sound/soc/omap/omap-mcpdm.c | 4 + sound/soc/pxa/Kconfig | 6 +- sound/soc/pxa/magician.c | 106 +- sound/soc/pxa/pxa-ssp.c | 181 +- sound/soc/pxa/pxa2xx-ac97.c | 47 +- sound/soc/pxa/pxa2xx-i2s.c | 9 +- sound/soc/pxa/pxa2xx-pcm.c | 73 - sound/soc/pxa/zylonite.c | 9 - sound/soc/qcom/Kconfig | 14 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/apq8096.c | 188 +- sound/soc/qcom/common.c | 112 + sound/soc/qcom/common.h | 11 + sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6adm.c | 16 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 225 +- sound/soc/qcom/qdsp6/q6afe.c | 43 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 42 +- sound/soc/qcom/qdsp6/q6asm.c | 17 +- sound/soc/qcom/qdsp6/q6routing.c | 71 +- sound/soc/qcom/sdm845.c | 285 +++ sound/soc/rockchip/Makefile | 3 +- sound/soc/rockchip/rockchip_i2s.c | 3 +- sound/soc/rockchip/rockchip_pcm.c | 45 + sound/soc/rockchip/rockchip_pcm.h | 14 + sound/soc/rockchip/rockchip_rt5645.c | 27 +- sound/soc/samsung/i2s.c | 1 + sound/soc/sh/Kconfig | 1 + sound/soc/sh/dma-sh7760.c | 26 +- sound/soc/sh/fsi.c | 22 +- sound/soc/sh/hac.c | 20 +- sound/soc/sh/migor.c | 14 +- sound/soc/sh/rcar/Makefile | 1 + sound/soc/sh/rcar/adg.c | 15 +- sound/soc/sh/rcar/cmd.c | 19 +- sound/soc/sh/rcar/core.c | 41 +- sound/soc/sh/rcar/ctu.c | 15 +- sound/soc/sh/rcar/dma.c | 17 +- sound/soc/sh/rcar/dvc.c | 16 +- sound/soc/sh/rcar/gen.c | 16 +- sound/soc/sh/rcar/mix.c | 14 +- sound/soc/sh/rcar/rsnd.h | 17 +- sound/soc/sh/rcar/src.c | 16 +- sound/soc/sh/rcar/ssi.c | 59 +- sound/soc/sh/rcar/ssiu.c | 15 +- sound/soc/sh/sh7760-ac97.c | 14 +- sound/soc/sh/siu.h | 26 +- sound/soc/sh/siu_dai.c | 26 +- sound/soc/sh/siu_pcm.c | 27 +- sound/soc/sh/ssi.c | 21 +- sound/soc/sirf/sirf-usp.c | 7 +- sound/soc/soc-ac97.c | 29 +- sound/soc/soc-acpi.c | 20 +- sound/soc/soc-compress.c | 120 +- sound/soc/soc-core.c | 200 +- sound/soc/soc-dapm.c | 51 +- sound/soc/soc-devres.c | 15 +- sound/soc/soc-generic-dmaengine-pcm.c | 30 +- sound/soc/soc-io.c | 19 +- sound/soc/soc-jack.c | 19 +- sound/soc/soc-ops.c | 29 +- sound/soc/soc-pcm.c | 469 ++-- sound/soc/soc-topology.c | 91 +- sound/soc/soc-utils.c | 24 +- sound/soc/sti/uniperif_player.c | 6 +- sound/soc/sti/uniperif_reader.c | 2 +- sound/soc/stm/Kconfig | 1 + sound/soc/stm/stm32_adfsdm.c | 10 +- sound/soc/stm/stm32_sai_sub.c | 146 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra30_i2s.h | 2 +- sound/soc/tegra/tegra_alc5632.c | 17 +- sound/soc/tegra/tegra_rt5677.c | 17 +- sound/soc/uniphier/aio-core.c | 84 +- sound/soc/uniphier/aio-cpu.c | 5 - sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-reg.h | 1 + sound/soc/uniphier/aio.h | 6 +- sound/soc/zte/zx-tdm.c | 4 +- 317 files changed, 15598 insertions(+), 4320 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt delete mode 100644 Documentation/devicetree/bindings/sound/dioo,dio2125.txt create mode 100644 Documentation/devicetree/bindings/sound/everest,es7241.txt create mode 100644 Documentation/devicetree/bindings/sound/marvell,pxa2xx-ac97.txt delete mode 100644 Documentation/devicetree/bindings/sound/mrvl,pxa2xx-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/name-prefix.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd9335.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5682.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-amplifier.txt create mode 100644 include/drm/drm_audio_component.h create mode 100644 include/sound/hda_component.h create mode 100644 include/sound/rt5682.h delete mode 100644 sound/arm/pxa2xx-pcm.c delete mode 100644 sound/arm/pxa2xx-pcm.h create mode 100644 sound/hda/hdac_component.c create mode 100644 sound/soc/codecs/es7241.c create mode 100644 sound/soc/codecs/rt5682.c create mode 100644 sound/soc/codecs/rt5682.h rename sound/soc/codecs/{dio2125.c => simple-amplifier.c} (68%) create mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-bxt-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-cnl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-glk-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-kbl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-skl-match.c create mode 100644 sound/soc/meson/Kconfig create mode 100644 sound/soc/meson/Makefile create mode 100644 sound/soc/meson/axg-card.c create mode 100644 sound/soc/meson/axg-fifo.c create mode 100644 sound/soc/meson/axg-fifo.h create mode 100644 sound/soc/meson/axg-frddr.c create mode 100644 sound/soc/meson/axg-spdifout.c create mode 100644 sound/soc/meson/axg-tdm-formatter.c create mode 100644 sound/soc/meson/axg-tdm-formatter.h create mode 100644 sound/soc/meson/axg-tdm-interface.c create mode 100644 sound/soc/meson/axg-tdm.h create mode 100644 sound/soc/meson/axg-tdmin.c create mode 100644 sound/soc/meson/axg-tdmout.c create mode 100644 sound/soc/meson/axg-toddr.c create mode 100644 sound/soc/qcom/common.c create mode 100644 sound/soc/qcom/common.h create mode 100644 sound/soc/qcom/sdm845.c create mode 100644 sound/soc/rockchip/rockchip_pcm.c create mode 100644 sound/soc/rockchip/rockchip_pcm.h

2 1

Re: [alsa-devel] A bug about cache inconsistency report
by Hans Hu(SH-RD) 13 Aug '18

13 Aug '18

> > > > >Then the next step would be to fake sg-buffer from this > > > > >straight buffer. Revert the above, and modify sgbuf.c to the following: > > > > >- Allocate a large continuous buffer > > > > >- Assign each page in this large buffer > > > > > > > > >If this still works, it's not about vmap, but it just means > > > > >that the physically ordered pages do matter -- implicitly > > > > >showing that the snooping behavior isn't properly turned on / off on the controller. > > > > > > > > To fake SG-buffer, I did this test: restore all the codes to the original, then added some codes in snd_malloc_sgbuf_pages() like below, the result is badly niose. > > > > > > >Not really what I meant. Rather something like below (totally untested). > > > > > > [Hans :] I know what you mean now and complete code like below, but the result is still noise. > > > > >OK, so indeed the vmapped address does seem matter. Interesting. > > >What kind of user access does produce the noise? Does it via aplay mmap mode, too? > > > > >In anyway, if the vmap is a problem, it might be worked around a patch like below (again totally untested and not sure whether it's correct). > > > > [Hans :] As I know, In the hardware layer HDAC’s stream have two data transport path : non-snoop & snoop; In the software layer ALSA-Driver have two data transport path : mmap & not mmap(test shows, it is dependent on wav's format or mmap_flag in aplay.c). When hardware at non-snoop mode, without hardware module's help, software must mark the mem to uncacheable type: when mmap used, the mark action happened at pcm_mmap_prepare(), the not-mmap mode's mark action happened at __mark_pages_wc(). And when at not-mmap mode, the vmapped address directly used in snd_pcm_lib_write_transfer() -> copy_from_user(). > > > >Well, that's usually no problem regarding that cache coherency. > >At least it hasn't been any issue with Intel and AMD CPUs. > >Does the problem happen with Intel CPU instead of VIA? > > [Hans :] I don't have enough Intel/AMD machine here, up to now, only find one Intel's : HW: mother board is Dell 042P49, HDA controller is 8086:1c20, codec is cx20641. >And the same issue is seen on that machine? [Hans :] Yes Thanks, Hans > > Yes, of course, this path combinations(non-snoop + not-mmap) unlikely be used by most music players. > > > > [Hans :] I complete the patch in your last mail like below, not used the vunmap & re-vmap way(it has a little problem). Result is OK now(no noise anymore). > > >What problem did you get with vunmap / vmap? > > [Hans :] if patch like below, even the new vmapped address new_area be assigned to dmab->area, it will not use this time, but next. This time used dmab->area still the last one, but the old one had been vunmap, so Oops reported. >That's bad. But the problem isn't as you guessed, I guess. >The allocator re-uses the pre-allocated buffer if available, and this makes things inconsistent. I'll cook later. 保密声明：本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 CONFIDENTIAL NOTE: This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

2 2

[alsa-devel] [PATCH] Suppress udev detection of sound card for some units on IEEE 1394 bus
by Takashi Sakamoto 11 Aug '18

11 Aug '18

A bug was filed to bugzilla.kernel.org for a quirk of some models which ALSA BeBoB driver supports. Bug 199365 - repeating bus resets on Firewire bus with Focusrite Saffaire 26/io https://bugzilla.kernel.org/show_bug.cgi?id=199365 Some models (two models as long as I know) have a quirk to disappear from IEEE 1394 bus at disconnections of packet streaming. Corresponding character devices are removed according to 'remove' callbacks of relevant drivers from Linux dd core. Then the models re-appear on the bus by generating bus resets and corresponding character devices are added according to 'probe' callbacks from Linux dd core. In a view of ALSA applications, this looks that plug-out/plug-in occur in a sequential order for the models when they stop playback/capture substream. For most applications, this doesn't cause large issue. However, this quirk is not good for combination of below modules in PulseAudio. PulseAudio enters endless loop to detect the models and start/stop PCM substream. - module-udev-detect - module-alsa-card - module-suspend-on-idle In detail, please read my comment no.6: https://bugzilla.kernel.org/show_bug.cgi?id=199365#c6 This commit suppressed udev detection of sound card for the issued models. For the models, 'PULSE_IGNORE' flag is added to udev rules, then module-udev-detect don't handle the models and PulseAudio never uses the models automatically. In a scenario for users to load module-alsa-card/module-alsa-sink/module-alsa-source by hand, although these modules can still stop PCM substreams with module-suspend-on-idle, PulseAudio never enters the endless loop because udev detection doesn't work for the models. In this case, as long as special files for ALSA character devices for these models are the same, corresponding sinks and sources are available even if the voluntary plug-out/plug-in occur. (Focusrite Saffire Pro 10 i/o with systemd 237) $ udevadm info -q all -p /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 P: /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 E: DEVPATH=/devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 E: ID_BUS=firewire E: ID_FOR_SEAT=sound-pci-0000_00_07_0 E: ID_ID=firewire-0x00130e01000606e0 E: ID_MODEL=Pro10IO E: ID_MODEL_FROM_DATABASE=XIO2213A/B/XIO2221 IEEE-1394b OHCI Controller [Cheetah Express] E: ID_MODEL_ID=0x000006 E: ID_PATH=pci-0000:00:07.0 E: ID_PATH_TAG=pci-0000_00_07_0 E: ID_PCI_CLASS_FROM_DATABASE=Serial bus controller E: ID_PCI_INTERFACE_FROM_DATABASE=OHCI E: ID_PCI_SUBCLASS_FROM_DATABASE=FireWire (IEEE 1394) E: ID_SERIAL=0x00130e01000606e0 E: ID_SERIAL_SHORT=0x00130e01000606e0 E: ID_VENDOR=Focusrite E: ID_VENDOR_FROM_DATABASE=Texas Instruments E: ID_VENDOR_ID=0x00130e E: SOUND_INITIALIZED=1 E: SUBSYSTEM=sound E: SYSTEMD_WANTS=sound.target E: TAGS=:systemd:seat: E: USEC_INITIALIZED=957089064 (Focusrite Saffire Pro 26 i/o with systemd 237) $ udevadm info -q all -p /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 P: /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 E: DEVPATH=/devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1 E: ID_BUS=firewire E: ID_FOR_SEAT=sound-pci-0000_00_07_0 E: ID_ID=firewire-0x00130e0100030cdd E: ID_MODEL=Pro26IO E: ID_MODEL_FROM_DATABASE=XIO2213A/B/XIO2221 IEEE-1394b OHCI Controller [Cheetah Express] E: ID_MODEL_ID=0x000003 E: ID_PATH=pci-0000:00:07.0 E: ID_PATH_TAG=pci-0000_00_07_0 E: ID_PCI_CLASS_FROM_DATABASE=Serial bus controller E: ID_PCI_INTERFACE_FROM_DATABASE=OHCI E: ID_PCI_SUBCLASS_FROM_DATABASE=FireWire (IEEE 1394) E: ID_SERIAL=0x00130e0100030cdd E: ID_SERIAL_SHORT=0x00130e0100030cdd E: ID_VENDOR=Focusrite E: ID_VENDOR_FROM_DATABASE=Texas Instruments E: ID_VENDOR_ID=0x00130e E: SOUND_INITIALIZED=1 E: SUBSYSTEM=sound E: SYSTEMD_WANTS=sound.target E: TAGS=:systemd:seat: E: USEC_INITIALIZED=1071026684 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- .../alsa/mixer/profile-sets/90-pulseaudio.rules | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules b/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules index ceb61d9e5..19b826b52 100644 --- a/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules +++ b/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules @@ -19,6 +19,7 @@ SUBSYSTEM!="sound", GOTO="pulseaudio_end" ACTION!="change", GOTO="pulseaudio_end" KERNEL!="card*", GOTO="pulseaudio_end" SUBSYSTEMS=="usb", GOTO="pulseaudio_check_usb" +SUBSYSTEMS=="firewire", GOTO="pulseaudio_firewire_quirk" SUBSYSTEMS=="platform", DRIVERS=="thinkpad_acpi", ENV{PULSE_IGNORE}="1" @@ -109,4 +110,13 @@ ATTRS{idVendor}=="1038", ATTRS{idProduct}=="1250", ENV{PULSE_PROFILE_SET}="steel ATTRS{idVendor}=="1038", ATTRS{idProduct}=="1260", ENV{PULSE_PROFILE_SET}="steelseries-arctis-7-usb-audio.conf" ATTRS{idVendor}=="147a", ATTRS{idProduct}=="e055", ENV{PULSE_PROFILE_SET}="cmedia-high-speed-true-hdaudio.conf" +GOTO="pulseaudio_end" + +LABEL="pulseaudio_firewire_quirk" + +# Focusrite Saffire Pro 10/26 i/o has a quirk to disappear from IEEE 1394 bus when losing connections. +# https://bugzilla.kernel.org/show_bug.cgi?id=199365 +ENV{ID_VENDOR_ID}=="0x00130e", ENV{ID_MODEL_ID}=="0x000003", ENV{PULSE_IGNORE}="1" +ENV{ID_VENDOR_ID}=="0x00130e", ENV{ID_MODEL_ID}=="0x000006", ENV{PULSE_IGNORE}="1" + LABEL="pulseaudio_end" -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: adau17x1: Implemented safeload support
by Robert Rosengren 10 Aug '18

10 Aug '18

From: Danny Smith <dannys(a)axis.com> Safeload support has been implemented which is used when updating for instance filter parameters using alsa controls. Without safeload support audio can become distorted during update. Signed-off-by: Danny Smith <dannys(a)axis.com> Signed-off-by: Robert Rosengren <robertr(a)axis.com> --- sound/soc/codecs/adau17x1.c | 69 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/adau17x1.h | 1 + 2 files changed, 68 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 57169b8ff14e..342ed0e04b1d 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -26,6 +26,10 @@ #include "adau17x1.h" #include "adau-utils.h" +#define ADAU17X1_SAFELOAD_TARGET_ADDRESS 0x0006 +#define ADAU17X1_SAFELOAD_TRIGGER 0x0007 +#define ADAU17X1_SAFELOAD_DATA(i) (0x0001 + (i)) + static const char * const adau17x1_capture_mixer_boost_text[] = { "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", }; @@ -326,6 +330,18 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +bool adau17x1_has_safeload(struct adau *adau) +{ + switch (adau->type) { + case ADAU1761: + case ADAU1781: + return true; + default: + return false; + } +} +EXPORT_SYMBOL_GPL(adau17x1_has_safeload); + static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { @@ -957,6 +973,50 @@ int adau17x1_resume(struct snd_soc_component *component) } EXPORT_SYMBOL_GPL(adau17x1_resume); +static int adau17x1_safeload(struct sigmadsp *sigmadsp, unsigned int addr, + const uint8_t bytes[], size_t len) +{ + uint8_t buf[4]; + unsigned int i; + int ret; + + /* target address is offset by 1 */ + unsigned int addr_offset = addr - 1; + + /* write safeload addresses */ + for (i = 0; i < len / 4; i++) { + memcpy(buf, bytes + i * 4, 4); + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_DATA(i), buf, 4); + if (ret < 0) + return ret; + } + + /* write target address */ + buf[0] = (addr_offset >> 24) & 0xff; + buf[1] = (addr_offset >> 16) & 0xff; + buf[2] = (addr_offset >> 8) & 0xff; + buf[3] = (addr_offset) & 0xff; + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TARGET_ADDRESS, buf, 4); + if (ret < 0) + return ret; + + /* write nbr of words to trigger address */ + buf[0] = 0x00; + buf[1] = 0x00; + buf[2] = 0x00; + buf[3] = i & 0xff; + ret = regmap_raw_write(sigmadsp->control_data, + ADAU17X1_SAFELOAD_TRIGGER, buf, 4); + + return ret; +} + +static const struct sigmadsp_ops adau17x1_sigmadsp_ops = { + .safeload = adau17x1_safeload, +}; + int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name) @@ -1002,8 +1062,13 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, dev_set_drvdata(dev, adau); if (firmware_name) { - adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, NULL, - firmware_name); + if (adau17x1_has_safeload(adau)) { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + &adau17x1_sigmadsp_ops, firmware_name); + } else { + adau->sigmadsp = devm_sigmadsp_init_regmap(dev, regmap, + NULL, firmware_name); + } if (IS_ERR(adau->sigmadsp)) { dev_warn(dev, "Could not find firmware file: %ld\n", PTR_ERR(adau->sigmadsp)); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index e6fe87beec07..8efa2df81c49 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -71,6 +71,7 @@ extern const struct snd_soc_dai_ops adau17x1_dai_ops; int adau17x1_setup_firmware(struct snd_soc_component *component, unsigned int rate); bool adau17x1_has_dsp(struct adau *adau); +bool adau17x1_has_safeload(struct adau *adau); #define ADAU17X1_CLOCK_CONTROL 0x4000 #define ADAU17X1_PLL_CONTROL 0x4002 -- 2.11.0

3 3

Re: [alsa-devel] : Kernal+debug installed: Help with TASCAM US-122
by Takashi Iwai 10 Aug '18

10 Aug '18

On Wed, 08 Aug 2018 18:46:13 +0200, Joseph Spencer wrote: > > I installed the kernal-debug package along with debug info (Linux > localhost.localdomain 4.17.11-200.fc28.x86_64+debug #1 SMP Mon Jul 30 > 14:53:06 UTC 2018 x86_64 x86_64 x86_64 GNU/Linux). > > I looked in /boot/config-4.17.11-200.fc28.x86_64+debug and I see that the > debug options are now enabled (attaching this file in case there are more > options I can enable). > > I repeated another failure. The output in journalctl -f appears to me to > be about the same (see attached) > > Any pointers on what I could enable or do to debug this further? If so, then try to disable i915 KMS. e.g. boot with nomodeset option, and check whether the crash still happens. Takashi

2 6

[alsa-devel] alsa bug - Martel Electronics USB soundcard recording sampling rate not automatically detected
by Susan Cragin 09 Aug '18

09 Aug '18

I run Dragon NaturallySpeaking through wine. The program works great with my old Logitech USB soundcard. But I just bought a new state-of-the-art dragon-optimized USB soundcard and stenomask, and the correct sampling rate cannot be automatically selected. Under proc/asound the device is described as EarMicrophone /tmp/alsa-info.txt.RjFRV2vThr

1 0

[alsa-devel] Fwd: ALSA queries
by Chakravarthi Pradeep 09 Aug '18

09 Aug '18

---------- Forwarded message --------- From: Chakravarthi Pradeep <doubleq7(a)gmail.com> Date: Sun 5 Aug, 2018, 14:19 Subject: Re: [alsa-devel] ALSA queries To: <tiwai(a)suse.de> Hello Takashi, Thanks for your reply. periods_min = the minimal number of periods period_bytes_min = the minimal size of bytes for one period period_bytes_max = the maximal size of bytes for one period periods_min = the minimal number of periods : what is meaning of periods , is it minimal number of interrupts ? period_bytes_min = the minimal size of bytes for one period : it means, minimal size of bytes per interrupt in case of device, Is it correct ? what about period_max ? It's most likely the ALSA PCM core's safety stop; your driver seem to have missed snd_pcm_period_elapsed() calls, so the hwptr isn't updated, resulting in XRUN. ALSA PCM core checks such XRUN condition with the own timer. I'm attaching my driver thread along with this mail. Can you please let me know if I have missed something in audio thread. ?. How to make sure in driver that, trigger stop should be called only when stop command is sent from application. I'm getting cut cut cut ... noise along with audio in VLC application. Initially for 2 or 4 seconds, cut cut cut noise is not heard in VLC application, However after almost after 5 sec ,I can hear cut cut cut noise in VLC application. With my analysis, hwptr is getting updated properly however I have doubt in app_ptr. I'm attaching the excl sheet with hw_ptr,app_ptr and buf_pos values. How to remove the cut cut cut ... noise in audio ? Regards, Chakravarthi On Fri, Aug 3, 2018 at 1:05 PM Takashi Iwai <tiwai(a)suse.de> wrote: > > On Thu, 02 Aug 2018 16:31:13 +0200, > Chakravarthi Pradeep wrote: > > > > I'm working on ALSA driver for PCIe card. My ALSA driver and it's > > initializing struct snd_pcm_hardware with below parameter. > > > > /************************ code start > > ************************************************/ > > static struct snd_pcm_hardware audio_pcm_hardware = { > > .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | > > SNDRV_PCM_INFO_MMAP_VALID | > > SNDRV_PCM_INFO_BLOCK_TRANSFER | > > SNDRV_PCM_INFO_RESUME ), > > .formats = (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S24_LE), > > .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | > > SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), > > .rate_min = 44100, > > .rate_max = 192000, > > .channels_min = 2, > > .channels_max = 8, > > .buffer_bytes_max = 76800, /*75 kbytes */ > > .period_bytes_min = 512,//4410, /* (channel (2) * sample_rate (44100) > > * bit width (24)) / (60 * 8) */ > > .period_bytes_max = 16*1024, > > .periods_min = 10, > > .periods_max = 255, > > > > }; > > /************************ code end > > ************************************************/ > > > > 1) I did not understand what is significance of periods_min , > > period_bytes_min , period_bytes_max and periods_max. Can you please > > tell me what is importance of these parameter and what value should be > > mentioned according into ALSA. > > These three defines the values your hardware may accept: > periods_min = the minimal number of periods > period_bytes_min = the minimal size of bytes for one period > period_bytes_max = the maximal size of bytes for one period > > > 2) snd_pcm_ops trigger callback is getting called in the driver when > > application sends "start" command. But ALSA driver is stopping by > > itself after one frame is copied to ALSA framework, without waiting > > for "stop" command. > > > > For instance: > > In trigger callback , I'm getting these logs after one frame is copied. > > Trigger:Start (When Play button is selected/clicked in application, > > Start command is sent to ALSA driver) > > Dma transfer is completed. > > Trigger:Stop. (When Stop button is selected/clicked in application, > > Stop command is sent to ALSA driver. But stop button is not clicked in > > this case) > > It's most likely the ALSA PCM core's safety stop; your driver seem to > have missed snd_pcm_period_elapsed() calls, so the hwptr isn't > updated, resulting in XRUN. ALSA PCM core checks such XRUN condition > with the own timer. > > > Takashi -- Thanks and Regards Chakravarthi Pradeep.K Ph: 91 9980434900

2 19

[alsa-devel] [PATCH] ASoC: adav80x: mark expected switch fall-through
by Gustavo A. R. Silva 09 Aug '18

09 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1056531 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/adav80x.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index db21ecb..8b9ca7e 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -648,6 +648,7 @@ static int adav80x_set_pll(struct snd_soc_component *component, int pll_id, pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; break; } + /* fall through */ default: return -EINVAL; } -- 2.7.4

3 4

[alsa-devel] [PATCH] ASoC: adau1761: Mark expected switch fall-though
by Gustavo A. R. Silva 09 Aug '18

09 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 114879 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/adau1761.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index be136e9..bef3e9e 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -518,7 +518,8 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_component *component) ARRAY_SIZE(adau1761_jack_detect_controls)); if (ret) return ret; - case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */ + /* fall through */ + case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes, ARRAY_SIZE(adau1761_no_dmic_routes)); if (ret) -- 2.7.4

3 2

[alsa-devel] [PATCH] ASoC: da7219: Add delays to capture path to remove DC offset noise
by Adam Thomson 09 Aug '18

09 Aug '18

On some platforms it has been noted that a pop noise can be witnessed when capturing audio, mainly for first time after a headset jack has been inserted. This is due to a DC offset in the Mic PGA and so to avoid this delays are required when powering up the capture path. This commit rectifies the problem by adding delays post Mic PGA and post Mixin PGA. The post Mic PGA delay is determined based on Mic Bias voltage, and is only applied the first time after a headset jack is inserted. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219-aad.c | 5 +++++ sound/soc/codecs/da7219.c | 44 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/da7219.h | 8 ++++++-- 3 files changed, 49 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index a49ab75..2c7d508 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -59,6 +59,7 @@ static void da7219_aad_btn_det_work(struct work_struct *work) container_of(work, struct da7219_aad_priv, btn_det_work); struct snd_soc_component *component = da7219_aad->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 statusa, micbias_ctrl; bool micbias_up = false; int retries = 0; @@ -86,6 +87,8 @@ static void da7219_aad_btn_det_work(struct work_struct *work) if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES) dev_warn(component->dev, "Mic bias status check timed out"); + da7219->micbias_on_event = true; + /* * Mic bias pulse required to enable mic, must be done before enabling * button detection to prevent erroneous button readings. @@ -439,6 +442,8 @@ static irqreturn_t da7219_aad_irq_thread(int irq, void *data) snd_soc_component_update_bits(component, DA7219_ACCDET_CONFIG_1, DA7219_BUTTON_CONFIG_MASK, 0); + da7219->micbias_on_event = false; + /* Disable mic bias */ snd_soc_dapm_disable_pin(dapm, "Mic Bias"); snd_soc_dapm_sync(dapm); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index c0144f2..e46e9f4 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -768,6 +768,30 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, * DAPM Events */ +static int da7219_mic_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (da7219->micbias_on_event) { + /* + * Delay only for first capture after bias enabled to + * avoid possible DC offset related noise. + */ + da7219->micbias_on_event = false; + msleep(da7219->mic_pga_delay); + } + break; + default: + break; + } + + return 0; +} + static int da7219_dai_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -937,12 +961,12 @@ static int da7219_gain_ramp_event(struct snd_soc_dapm_widget *w, SND_SOC_DAPM_INPUT("MIC"), /* Input PGAs */ - SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL, - DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), - SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL, - DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, - NULL, 0), + SND_SOC_DAPM_PGA_E("Mic PGA", DA7219_MIC_1_CTRL, + DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_mic_pga_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("Mixin PGA", DA7219_MIXIN_L_CTRL, + DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT, + NULL, 0, da7219_settling_event, SND_SOC_DAPM_POST_PMU), /* Input Filters */ SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT, @@ -1847,6 +1871,14 @@ static void da7219_handle_pdata(struct snd_soc_component *component) snd_soc_component_write(component, DA7219_MICBIAS_CTRL, micbias_lvl); + /* + * Calculate delay required to compensate for DC offset in + * Mic PGA, based on Mic Bias voltage. + */ + da7219->mic_pga_delay = DA7219_MIC_PGA_BASE_DELAY + + (pdata->micbias_lvl * + DA7219_MIC_PGA_OFFSET_DELAY); + /* Mic */ switch (pdata->mic_amp_in_sel) { case DA7219_MIC_AMP_IN_SEL_DIFF: diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 1b00023..3a00686 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -781,8 +781,10 @@ #define DA7219_SYS_STAT_CHECK_DELAY 50 /* Power up/down Delays */ -#define DA7219_SETTLING_DELAY 40 -#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_SETTLING_DELAY 40 +#define DA7219_MIN_GAIN_DELAY 30 +#define DA7219_MIC_PGA_BASE_DELAY 100 +#define DA7219_MIC_PGA_OFFSET_DELAY 40 enum da7219_clk_src { DA7219_CLKSRC_MCLK = 0, @@ -828,6 +830,8 @@ struct da7219_priv { bool master; bool alc_en; + bool micbias_on_event; + unsigned int mic_pga_delay; u8 gain_ramp_ctrl; }; -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Mark expected switch fall-through
by Gustavo A. R. Silva 09 Aug '18

09 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1357413 ("Missing break in switch") Addresses-Coverity-ID: 114917 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/usb/mixer.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 73e811f..c63c84b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -856,12 +856,14 @@ static int check_input_term(struct mixer_build *state, int id, term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ else /* UAC_VERSION_2 */ term->type = UAC3_EFFECT_UNIT << 16; /* virtual type */ + /* fall through */ case UAC1_EXTENSION_UNIT: /* UAC2_PROCESSING_UNIT_V2 */ if (protocol == UAC_VERSION_1 && !term->type) term->type = UAC3_EXTENSION_UNIT << 16; /* virtual type */ else if (protocol == UAC_VERSION_2 && !term->type) term->type = UAC3_PROCESSING_UNIT << 16; /* virtual type */ + /* fall through */ case UAC2_EXTENSION_UNIT_V2: { struct uac_processing_unit_descriptor *d = p1; -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: mixart_core: Mark expected switch fall-through
by Gustavo A. R. Silva 09 Aug '18

09 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Notice that in this particular case, I replaced the code comment with a proper "fall through" annotation, which is what GCC is expecting to find. Addresses-Coverity-ID: 114889 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/mixart/mixart_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 46c292b..71776bf 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -540,7 +540,7 @@ irqreturn_t snd_mixart_threaded_irq(int irq, void *dev_id) dev_err(&mgr->pci->dev, "canceled notification %x !\n", msg); } - /* no break, continue ! */ + /* fall through */ case MSG_TYPE_ANSWER: /* answer or notification to a message we are waiting for*/ mutex_lock(&mgr->msg_lock); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: opl3: Mark expected switch fall-through
by Gustavo A. R. Silva 08 Aug '18

08 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 114878 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/drivers/opl3/opl3_midi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 471916ca..a33cb74 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -368,6 +368,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) instr_4op = 1; break; } + /* fall through */ default: spin_unlock_irqrestore(&opl3->voice_lock, flags); return; -- 2.7.4

2 1

[alsa-devel] [RESEND PATCH] ASoC: Intel: Skylake: Mark expected switch fall-through
by Gustavo A. R. Silva 08 Aug '18

08 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1357418 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/intel/skylake/skl-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 823e391..62d6f2f 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -494,6 +494,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } + /* fall through */ case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: -- 2.7.4

1 0

[alsa-devel] [PATCH 00/11] Add Recon3D support + Driver cleanup
by Connor McAdams 08 Aug '18

08 Aug '18

This patch series adds support for the Recon3D PCI-E card. I have personally tested it on a Recon3D that was sent to me, and included that PCI ID in the quirk table. Much of this patch series renames Recon3Di functions to just r3d because both cards behave similarly. Aside from that, I do have a question- If the PCI ID's are included in the Window's driver .inf file, would I be good to add them in as quirks? Or should I wait for someone with the card to test it, and then add it once I know it works on that specific card? Connor McAdams (11): ALSA: hda/ca0132 - Create mmio gpio function to make code clearer ALSA: hda/ca0132 - Clean up ca0132_init function. ALSA: hda/ca0132 - Add alt_functions unsolicited response ALSA: hda/ca0132 - Add quirk ID and enum for Recon3D ALSA: hda/ca0132 - Add Recon3D pincfg ALSA: hda/ca0132 - Add bool variable to enable/disable pci region2 mmio ALSA: hda/ca0132 - Add Recon3D startup functions and setup ALSA: hda/ca0132 - Add DSP setup defaults for Recon3D ALSA: hda/ca0132 - Add Recon3D input and output select commands ALSA: hda/ca0132 - Change mixer controls for Recon3D ALSA: hda/ca0132 - Add exit commands for Recon3D sound/pci/hda/patch_ca0132.c | 276 ++++++++++++++++++++++++++++--------------- 1 file changed, 180 insertions(+), 96 deletions(-) -- 2.7.4

2 12

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Declare firmware controls from codec driver
by Richard Fitzgerald 08 Aug '18

08 Aug '18

To allow for more flexibility in naming of DSP-type cores move the creation of the firmware controls to the codec drivers instead of having a hardcoded list in wm_adsp. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs47l24.c | 3 +++ sound/soc/codecs/wm2200.c | 10 ++++------ sound/soc/codecs/wm5102.c | 2 ++ sound/soc/codecs/wm5110.c | 5 +++++ sound/soc/codecs/wm_adsp.c | 35 +++++++++-------------------------- sound/soc/codecs/wm_adsp.h | 10 +++++++++- 6 files changed, 32 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 0da52ead91e0..45e50fe3bf25 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -235,6 +235,9 @@ ARIZONA_MIXER_CONTROLS("AIF2TX6", ARIZONA_AIF2TX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 3663b9fd4d65..deff65161504 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1180,6 +1180,9 @@ SOC_DOUBLE_R_TLV("OUT2 Digital Volume", WM2200_DAC_DIGITAL_VOLUME_2L, SOC_DOUBLE("OUT2 Switch", WM2200_PDM_1, WM2200_SPK1L_MUTE_SHIFT, WM2200_SPK1R_MUTE_SHIFT, 1, 1), SOC_ENUM("RxANC Src", wm2200_rxanc_input_sel), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), }; WM2200_MIXER_ENUMS(OUT1L, WM2200_OUT1LMIX_INPUT_1_SOURCE); @@ -1553,15 +1556,10 @@ static const struct snd_soc_dapm_route wm2200_dapm_routes[] = { static int wm2200_probe(struct snd_soc_component *component) { struct wm2200_priv *wm2200 = snd_soc_component_get_drvdata(component); - int ret; wm2200->component = component; - ret = snd_soc_add_component_controls(component, wm_adsp_fw_controls, 2); - if (ret != 0) - return ret; - - return ret; + return 0; } static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a01a0c0e01eb..7e817e1877c2 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -985,6 +985,8 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 00c735c585d9..b0789a03d699 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -927,6 +927,11 @@ ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), + +WM_ADSP_FW_CONTROL("DSP1", 0), +WM_ADSP_FW_CONTROL("DSP2", 1), +WM_ADSP_FW_CONTROL("DSP3", 2), +WM_ADSP_FW_CONTROL("DSP4", 3), }; ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ae131376cea1..1c12c78dbcce 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -684,8 +684,8 @@ static inline void wm_adsp_debugfs_clear(struct wm_adsp *dsp) } #endif -static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -695,9 +695,10 @@ static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, return 0; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_get); -static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -721,8 +722,9 @@ static int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, return ret; } +EXPORT_SYMBOL_GPL(wm_adsp_fw_put); -static const struct soc_enum wm_adsp_fw_enum[] = { +const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 1, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 2, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), @@ -731,24 +733,7 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 5, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), SOC_ENUM_SINGLE(0, 6, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; - -const struct snd_kcontrol_new wm_adsp_fw_controls[] = { - SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP5 Firmware", wm_adsp_fw_enum[4], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP6 Firmware", wm_adsp_fw_enum[5], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM_EXT("DSP7 Firmware", wm_adsp_fw_enum[6], - wm_adsp_fw_get, wm_adsp_fw_put), -}; -EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); +EXPORT_SYMBOL_GPL(wm_adsp_fw_enum); static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -2890,9 +2875,7 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp dsp->component = component; - return snd_soc_add_component_controls(component, - &wm_adsp_fw_controls[dsp->num - 1], - 1); + return 0; } EXPORT_SYMBOL_GPL(wm_adsp2_component_probe); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index bc6d359f0533..8d58cb9d9bb9 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -121,7 +121,11 @@ struct wm_adsp { .reg = SND_SOC_NOPM, .shift = num, .event = wm_adsp2_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } -extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; +#define WM_ADSP_FW_CONTROL(dspname, num) \ + SOC_ENUM_EXT(dspname " Firmware", wm_adsp_fw_enum[num], \ + wm_adsp_fw_get, wm_adsp_fw_put) + +extern const struct soc_enum wm_adsp_fw_enum[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); @@ -144,6 +148,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int wm_adsp_fw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream); int wm_adsp_compr_free(struct snd_compr_stream *stream); -- 2.11.0

2 3

[alsa-devel] [PATCH - hwmixvolume 8/9] hwmixvolume: add my copyright
by Emmanuel Gil Peyrot 08 Aug '18

08 Aug '18

Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 7b90613..c34904e 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -2,6 +2,7 @@ # hwmixvolume - ALSA hardware mixer volume control applet # Copyright (c) 2009-2010 Clemens Ladisch +# Copyright (c) 2018 Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> # # Permission to use, copy, modify, and/or distribute this software for any # purpose with or without fee is hereby granted, provided that the above -- 2.18.0

1 0

[alsa-devel] [PATCH - hwmixvolume 7/9] hwmixvolume: switch the shebang to python, as it is now compatible with both 2 and 3
by Emmanuel Gil Peyrot 08 Aug '18

08 Aug '18

Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index ec2d05d..7b90613 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -1,4 +1,4 @@ -#!/usr/bin/python2 +#!/usr/bin/env python # hwmixvolume - ALSA hardware mixer volume control applet # Copyright (c) 2009-2010 Clemens Ladisch -- 2.18.0

1 0

[alsa-devel] [PATCH - hwmixvolume 6/9] hwmixvolume: fix deprecation warnings given by G_ENABLE_DIAGNOSTIC=1
by Emmanuel Gil Peyrot 08 Aug '18

08 Aug '18

Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index af0c1e0..ec2d05d 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -60,7 +60,7 @@ class Stream: value = alsahcontrol.Value(self.element) value.read() values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = Gtk.Label(self.get_label(info)) + self.label = Gtk.Label.new(self.get_label(info)) self.label.set_single_line_mode(True) self.parent.scales_vbox.add(self.label) for i in range(info.count): @@ -201,15 +201,15 @@ class MixerWindow(Gtk.Window): self.scales_vbox = Gtk.Grid() self.scales_vbox.set_orientation(Gtk.Orientation.VERTICAL) - scrollwin.add_with_viewport(self.scales_vbox) + scrollwin.add(self.scales_vbox) label = Gtk.Label() label.set_single_line_mode(True) - line_height = label.size_request().height + line_height = max(label.get_size_request().height, 0) label.destroy() scale = Gtk.Scale(orientation=Gtk.Orientation.HORIZONTAL) scale.set_draw_value(False) - line_height += scale.size_request().height + line_height += max(scale.get_size_request().height, 0) scale.destroy() # always have space for at least four sliders scrollwin.set_size_request(width=-1, height=line_height*4+4) @@ -267,7 +267,7 @@ class MixerWindow(Gtk.Window): else: msg = "This card does not have stream controls." if not has_msg: - self.msg_label = Gtk.Label(msg) + self.msg_label = Gtk.Label.new(msg) self.msg_label.set_vexpand(True) self.scales_vbox.add(self.msg_label) self.scales_vbox.show_all() -- 2.18.0

1 0

[alsa-devel] [PATCH - hwmixvolume 5/9] hwmixvolume: replace deprecated Gtk.HScale with Gtk.Scale
by Emmanuel Gil Peyrot 08 Aug '18

08 Aug '18

Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index e82f32d..af0c1e0 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -69,7 +69,7 @@ class Stream: step_incr=1, page_incr=(info.max-info.min+1)/8) adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = Gtk.HScale(adj) + scale = Gtk.Scale(orientation=Gtk.Orientation.HORIZONTAL, adjustment=adj) scale.set_draw_value(False) self.parent.scales_vbox.add(scale) self.scales.append(scale) @@ -207,7 +207,7 @@ class MixerWindow(Gtk.Window): label.set_single_line_mode(True) line_height = label.size_request().height label.destroy() - scale = Gtk.HScale() + scale = Gtk.Scale(orientation=Gtk.Orientation.HORIZONTAL) scale.set_draw_value(False) line_height += scale.size_request().height scale.destroy() -- 2.18.0

1 0

[alsa-devel] [PATCH - hwmixvolume 3/9] hwmixvolume: use four spaces instead of one tab for indent
by Emmanuel Gil Peyrot 08 Aug '18

08 Aug '18

Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 039ca17..267228c 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -29,289 +29,289 @@ EVENT_INFO = alsahcontrol.event_mask['INFO'] EVENT_REMOVE = alsahcontrol.event_mask_remove class Stream: - def __init__(self, element, parent): - self.element = element - self.element.set_callback(self) - self.parent = parent - self.label = None - self.scales = [] - self.adjustments = [] - self.callback(self.element, EVENT_INFO) - - def destroy(self): - self.deactivate() - - def callback(self, e, mask): - if mask == EVENT_REMOVE: - self.deactivate() - elif (mask & EVENT_INFO) != 0: - info = alsahcontrol.Info(self.element) - if info.is_inactive: - self.deactivate() - else: - self.activate() - elif (mask & EVENT_VALUE) != 0: - self.update_scales_from_ctl() - - def activate(self): - if self.label: - return - info = alsahcontrol.Info(self.element) - value = alsahcontrol.Value(self.element) - value.read() - values = value.get_tuple(TYPE_INTEGER, info.count) - self.label = Gtk.Label(self.get_label(info)) - self.label.set_single_line_mode(True) - self.parent.scales_vbox.add(self.label) - for i in range(info.count): - adj = Gtk.Adjustment(value=values[i], - lower=info.min, upper=info.max, - step_incr=1, - page_incr=(info.max-info.min+1)/8) - adj.connect('value-changed', self.update_ctl_from_scale, i) - scale = Gtk.HScale(adj) - scale.set_draw_value(False) - self.parent.scales_vbox.add(scale) - self.scales.append(scale) - self.adjustments.append(adj) - self.parent.scales_vbox.show_all() - self.parent.update_msg_label() - - def deactivate(self): - if not self.label: - return - self.label.destroy() - for s in self.scales: - s.destroy() - self.label = None - self.scales = [] - self.adjustments = [] - self.parent.update_msg_label() - - def update_scales_from_ctl(self): - if not self.label: - return - count = len(self.adjustments) - value = alsahcontrol.Value(self.element) - value.read() - values = value.get_tuple(TYPE_INTEGER, count) - for i in range(count): - self.adjustments[i].set_value(values[i]) - - def update_ctl_from_scale(self, adj, index): - scale_value = adj.get_value() - value_to_set = int(round(adj.get_value())) - count = len(self.adjustments) - value = alsahcontrol.Value(self.element) - if self.parent.lock_check.get_active(): - values = [value_to_set for i in range(count)] - else: - value.read() - values = value.get_array(TYPE_INTEGER, count) - values[index] = value_to_set - value.set_array(TYPE_INTEGER, values) - value.write() - if value_to_set != scale_value: - adj.set_value(value_to_set) - - def get_label(self, info): - pid = self.get_pid(info) - if pid: - cmdline = self.get_pid_cmdline(pid) - if cmdline: - return cmdline - else: - return "PID %d" % pid - else: - name = info.name - if name[-7:] == " Volume": - name = name[:-7] - if name[-9:] == " Playback": - name = name[:-9] - return name - - def get_pid(self, info): - card = self.parent.current_card - device = info.device - subdevice = info.subdevice - if subdevice == 0: - subdevice = info.index - filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) - try: - f = open(filename, "r") - except IOError: - return None - try: - for line in f.readlines(): - if line[:9] == "owner_pid": - return int(line.split(':')[1].strip()) - finally: - f.close() - return None - - def get_pid_cmdline(self, pid): - try: - f = open("/proc/%d/cmdline" % pid, "r") - except IOError: - return None - try: - cmdline = f.read() - finally: - f.close() - return cmdline.replace('\x00', ' ').strip() + def __init__(self, element, parent): + self.element = element + self.element.set_callback(self) + self.parent = parent + self.label = None + self.scales = [] + self.adjustments = [] + self.callback(self.element, EVENT_INFO) + + def destroy(self): + self.deactivate() + + def callback(self, e, mask): + if mask == EVENT_REMOVE: + self.deactivate() + elif (mask & EVENT_INFO) != 0: + info = alsahcontrol.Info(self.element) + if info.is_inactive: + self.deactivate() + else: + self.activate() + elif (mask & EVENT_VALUE) != 0: + self.update_scales_from_ctl() + + def activate(self): + if self.label: + return + info = alsahcontrol.Info(self.element) + value = alsahcontrol.Value(self.element) + value.read() + values = value.get_tuple(TYPE_INTEGER, info.count) + self.label = Gtk.Label(self.get_label(info)) + self.label.set_single_line_mode(True) + self.parent.scales_vbox.add(self.label) + for i in range(info.count): + adj = Gtk.Adjustment(value=values[i], + lower=info.min, upper=info.max, + step_incr=1, + page_incr=(info.max-info.min+1)/8) + adj.connect('value-changed', self.update_ctl_from_scale, i) + scale = Gtk.HScale(adj) + scale.set_draw_value(False) + self.parent.scales_vbox.add(scale) + self.scales.append(scale) + self.adjustments.append(adj) + self.parent.scales_vbox.show_all() + self.parent.update_msg_label() + + def deactivate(self): + if not self.label: + return + self.label.destroy() + for s in self.scales: + s.destroy() + self.label = None + self.scales = [] + self.adjustments = [] + self.parent.update_msg_label() + + def update_scales_from_ctl(self): + if not self.label: + return + count = len(self.adjustments) + value = alsahcontrol.Value(self.element) + value.read() + values = value.get_tuple(TYPE_INTEGER, count) + for i in range(count): + self.adjustments[i].set_value(values[i]) + + def update_ctl_from_scale(self, adj, index): + scale_value = adj.get_value() + value_to_set = int(round(adj.get_value())) + count = len(self.adjustments) + value = alsahcontrol.Value(self.element) + if self.parent.lock_check.get_active(): + values = [value_to_set for i in range(count)] + else: + value.read() + values = value.get_array(TYPE_INTEGER, count) + values[index] = value_to_set + value.set_array(TYPE_INTEGER, values) + value.write() + if value_to_set != scale_value: + adj.set_value(value_to_set) + + def get_label(self, info): + pid = self.get_pid(info) + if pid: + cmdline = self.get_pid_cmdline(pid) + if cmdline: + return cmdline + else: + return "PID %d" % pid + else: + name = info.name + if name[-7:] == " Volume": + name = name[:-7] + if name[-9:] == " Playback": + name = name[:-9] + return name + + def get_pid(self, info): + card = self.parent.current_card + device = info.device + subdevice = info.subdevice + if subdevice == 0: + subdevice = info.index + filename = "/proc/asound/card%d/pcm%dp/sub%d/status" % (card, device, subdevice) + try: + f = open(filename, "r") + except IOError: + return None + try: + for line in f.readlines(): + if line[:9] == "owner_pid": + return int(line.split(':')[1].strip()) + finally: + f.close() + return None + + def get_pid_cmdline(self, pid): + try: + f = open("/proc/%d/cmdline" % pid, "r") + except IOError: + return None + try: + cmdline = f.read() + finally: + f.close() + return cmdline.replace('\x00', ' ').strip() class MixerWindow(Gtk.Window): - card_numbers = alsacard.card_list() - current_card = -1 - hcontrol = None - scales_vbox = None - msg_label = None - streams = [] - hctl_sources = [] - - def __init__(self): - Gtk.Window.__init__(self) - self.connect('destroy', lambda w: Gtk.main_quit()) - self.set_title("Hardware Mixer Volumes") - - vbox = Gtk.Grid() - vbox.set_orientation(Gtk.Orientation.VERTICAL) - self.add(vbox) - - hbox = Gtk.Grid() - vbox.add(hbox) - - label = Gtk.Label.new_with_mnemonic("_Sound Card: ") - hbox.add(label) - - combo = Gtk.ComboBoxText() - combo.set_hexpand(True) - for i in self.card_numbers: - str = "%d: %s" % (i, alsacard.card_get_name(i)) - combo.append_text(str) - if len(self.card_numbers) > 0: - combo.set_active(0) - combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) - hbox.add(combo) - label.set_mnemonic_widget(combo) - - self.lock_check = Gtk.CheckButton.new_with_mnemonic(label="_Lock Channels") - self.lock_check.set_active(True) - vbox.add(self.lock_check) - - scrollwin = Gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=Gtk.PolicyType.NEVER, vscrollbar_policy=Gtk.PolicyType.AUTOMATIC) - scrollwin.set_shadow_type(Gtk.ShadowType.NONE) - scrollwin.set_vexpand(True) - vbox.add(scrollwin) - - self.scales_vbox = Gtk.Grid() - self.scales_vbox.set_orientation(Gtk.Orientation.VERTICAL) - scrollwin.add_with_viewport(self.scales_vbox) - - label = Gtk.Label() - label.set_single_line_mode(True) - line_height = label.size_request().height - label.destroy() - scale = Gtk.HScale() - scale.set_draw_value(False) - line_height += scale.size_request().height - scale.destroy() - # always have space for at least four sliders - scrollwin.set_size_request(width=-1, height=line_height*4+4) - - # TODO: select the default card or the first card with stream controls - if len(self.card_numbers) > 0: - self.change_card(self.card_numbers[0]) - self.update_msg_label() - - self.show_all() - - def change_card(self, cardnum): - for s in self.hctl_sources: - GLib.source_remove(s) - self.hctl_sources = [] - - self.hcontrol = self.open_hcontrol_for_card(cardnum) - - for s in self.streams: - s.destroy() - self.streams = [] - - self.current_card = cardnum - - if not self.hcontrol: - self.update_msg_label() - return - - for id in self.hcontrol.list(): - if not self.is_stream_elem(id): - continue - elem = alsahcontrol.Element(self.hcontrol, id[0]) - info = alsahcontrol.Info(elem) - if not self.is_stream_info(info): - continue - stream = Stream(elem, self) - self.streams.append(stream) - - for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(GLib.io_add_watch(fd, 0, GLib.IOCondition(condition), self.hctl_io_callback)) - - self.update_msg_label() - - self.scales_vbox.show_all() - - def update_msg_label(self): - needs_msg = len(self.scales_vbox.get_children()) < 2 - has_msg = self.msg_label - if has_msg and not needs_msg: - self.msg_label.destroy() - self.msg_label = None - elif needs_msg: - if len(self.streams) > 0: - msg = "There are no open streams." - else: - msg = "This card does not have stream controls." - if not has_msg: - self.msg_label = Gtk.Label(msg) - self.msg_label.set_vexpand(True) - self.scales_vbox.add(self.msg_label) - self.scales_vbox.show_all() - elif self.msg_label.get_text() != msg: - self.msg_label.set_text(msg) - - def open_hcontrol_for_card(self, cardnum): - devname = "hw:CARD=" + str(cardnum) - try: - hc = alsahcontrol.HControl(name=devname, - mode=alsahcontrol.open_mode['NONBLOCK']) - except: - # TODO: alsa error msg - dlg = Gtk.MessageDialog(self, - Gtk.DialogFlags.MODAL | Gtk.DialogFlags.DESTROY_WITH_PARENT, - Gtk.MessageType.ERROR, Gtk.ButtonsType.OK, - "Cannot open sound card control device.") - dlg.run() - dlg.destroy() - return None - return hc - - def is_stream_elem(self, id): - return ((id[1] == INTF_PCM and - id[4] in ("PCM Playback Volume", "EMU10K1 PCM Volume")) or - (id[1] == INTF_MIXER and - id[4] == "VIA DXS Playback Volume")) - - def is_stream_info(self, info): - return info.is_readable and info.is_writable and info.type == TYPE_INTEGER - - def hctl_io_callback(self, source, condition): - self.hcontrol.handle_events() - return True + card_numbers = alsacard.card_list() + current_card = -1 + hcontrol = None + scales_vbox = None + msg_label = None + streams = [] + hctl_sources = [] + + def __init__(self): + Gtk.Window.__init__(self) + self.connect('destroy', lambda w: Gtk.main_quit()) + self.set_title("Hardware Mixer Volumes") + + vbox = Gtk.Grid() + vbox.set_orientation(Gtk.Orientation.VERTICAL) + self.add(vbox) + + hbox = Gtk.Grid() + vbox.add(hbox) + + label = Gtk.Label.new_with_mnemonic("_Sound Card: ") + hbox.add(label) + + combo = Gtk.ComboBoxText() + combo.set_hexpand(True) + for i in self.card_numbers: + str = "%d: %s" % (i, alsacard.card_get_name(i)) + combo.append_text(str) + if len(self.card_numbers) > 0: + combo.set_active(0) + combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) + hbox.add(combo) + label.set_mnemonic_widget(combo) + + self.lock_check = Gtk.CheckButton.new_with_mnemonic(label="_Lock Channels") + self.lock_check.set_active(True) + vbox.add(self.lock_check) + + scrollwin = Gtk.ScrolledWindow() + scrollwin.set_policy(hscrollbar_policy=Gtk.PolicyType.NEVER, vscrollbar_policy=Gtk.PolicyType.AUTOMATIC) + scrollwin.set_shadow_type(Gtk.ShadowType.NONE) + scrollwin.set_vexpand(True) + vbox.add(scrollwin) + + self.scales_vbox = Gtk.Grid() + self.scales_vbox.set_orientation(Gtk.Orientation.VERTICAL) + scrollwin.add_with_viewport(self.scales_vbox) + + label = Gtk.Label() + label.set_single_line_mode(True) + line_height = label.size_request().height + label.destroy() + scale = Gtk.HScale() + scale.set_draw_value(False) + line_height += scale.size_request().height + scale.destroy() + # always have space for at least four sliders + scrollwin.set_size_request(width=-1, height=line_height*4+4) + + # TODO: select the default card or the first card with stream controls + if len(self.card_numbers) > 0: + self.change_card(self.card_numbers[0]) + self.update_msg_label() + + self.show_all() + + def change_card(self, cardnum): + for s in self.hctl_sources: + GLib.source_remove(s) + self.hctl_sources = [] + + self.hcontrol = self.open_hcontrol_for_card(cardnum) + + for s in self.streams: + s.destroy() + self.streams = [] + + self.current_card = cardnum + + if not self.hcontrol: + self.update_msg_label() + return + + for id in self.hcontrol.list(): + if not self.is_stream_elem(id): + continue + elem = alsahcontrol.Element(self.hcontrol, id[0]) + info = alsahcontrol.Info(elem) + if not self.is_stream_info(info): + continue + stream = Stream(elem, self) + self.streams.append(stream) + + for fd,condition in self.hcontrol.poll_fds: + self.hctl_sources.append(GLib.io_add_watch(fd, 0, GLib.IOCondition(condition), self.hctl_io_callback)) + + self.update_msg_label() + + self.scales_vbox.show_all() + + def update_msg_label(self): + needs_msg = len(self.scales_vbox.get_children()) < 2 + has_msg = self.msg_label + if has_msg and not needs_msg: + self.msg_label.destroy() + self.msg_label = None + elif needs_msg: + if len(self.streams) > 0: + msg = "There are no open streams." + else: + msg = "This card does not have stream controls." + if not has_msg: + self.msg_label = Gtk.Label(msg) + self.msg_label.set_vexpand(True) + self.scales_vbox.add(self.msg_label) + self.scales_vbox.show_all() + elif self.msg_label.get_text() != msg: + self.msg_label.set_text(msg) + + def open_hcontrol_for_card(self, cardnum): + devname = "hw:CARD=" + str(cardnum) + try: + hc = alsahcontrol.HControl(name=devname, + mode=alsahcontrol.open_mode['NONBLOCK']) + except: + # TODO: alsa error msg + dlg = Gtk.MessageDialog(self, + Gtk.DialogFlags.MODAL | Gtk.DialogFlags.DESTROY_WITH_PARENT, + Gtk.MessageType.ERROR, Gtk.ButtonsType.OK, + "Cannot open sound card control device.") + dlg.run() + dlg.destroy() + return None + return hc + + def is_stream_elem(self, id): + return ((id[1] == INTF_PCM and + id[4] in ("PCM Playback Volume", "EMU10K1 PCM Volume")) or + (id[1] == INTF_MIXER and + id[4] == "VIA DXS Playback Volume")) + + def is_stream_info(self, info): + return info.is_readable and info.is_writable and info.type == TYPE_INTEGER + + def hctl_io_callback(self, source, condition): + self.hcontrol.handle_events() + return True def main(): - MixerWindow() - Gtk.main() + MixerWindow() + Gtk.main() main() -- 2.18.0

1 0

[alsa-devel] [PATCH - hwmixvolume 2/9] hwmixvolume: switch to GTK+ 3.0
by Emmanuel Gil Peyrot 08 Aug '18

08 Aug '18

Signed-off-by: Emmanuel Gil Peyrot <linkmauve(a)linkmauve.fr> diff --git a/hwmixvolume/hwmixvolume b/hwmixvolume/hwmixvolume index 28ce776..039ca17 100755 --- a/hwmixvolume/hwmixvolume +++ b/hwmixvolume/hwmixvolume @@ -62,7 +62,7 @@ class Stream: values = value.get_tuple(TYPE_INTEGER, info.count) self.label = Gtk.Label(self.get_label(info)) self.label.set_single_line_mode(True) - self.parent.scales_vbox.pack_start(self.label, expand=False) + self.parent.scales_vbox.add(self.label) for i in range(info.count): adj = Gtk.Adjustment(value=values[i], lower=info.min, upper=info.max, @@ -71,7 +71,7 @@ class Stream: adj.connect('value-changed', self.update_ctl_from_scale, i) scale = Gtk.HScale(adj) scale.set_draw_value(False) - self.parent.scales_vbox.pack_start(scale, expand=False) + self.parent.scales_vbox.add(scale) self.scales.append(scale) self.adjustments.append(adj) self.parent.scales_vbox.show_all() @@ -174,45 +174,48 @@ class MixerWindow(Gtk.Window): self.connect('destroy', lambda w: Gtk.main_quit()) self.set_title("Hardware Mixer Volumes") - vbox = Gtk.VBox() + vbox = Gtk.Grid() + vbox.set_orientation(Gtk.Orientation.VERTICAL) self.add(vbox) - hbox = Gtk.HBox() - vbox.pack_start(hbox, expand=False) + hbox = Gtk.Grid() + vbox.add(hbox) - label = Gtk.Label("_Sound Card: ") - label.set_use_underline(True) - hbox.pack_start(label, expand=False) + label = Gtk.Label.new_with_mnemonic("_Sound Card: ") + hbox.add(label) - combo = Gtk.combo_box_new_text() + combo = Gtk.ComboBoxText() + combo.set_hexpand(True) for i in self.card_numbers: str = "%d: %s" % (i, alsacard.card_get_name(i)) combo.append_text(str) if len(self.card_numbers) > 0: combo.set_active(0) combo.connect('changed', lambda c: self.change_card(self.card_numbers[combo.get_active()])) - hbox.pack_start(combo) + hbox.add(combo) label.set_mnemonic_widget(combo) - self.lock_check = Gtk.CheckButton(label="_Lock Channels") + self.lock_check = Gtk.CheckButton.new_with_mnemonic(label="_Lock Channels") self.lock_check.set_active(True) - vbox.pack_start(self.lock_check, expand=False) + vbox.add(self.lock_check) scrollwin = Gtk.ScrolledWindow() - scrollwin.set_policy(hscrollbar_policy=Gtk.POLICY_NEVER, vscrollbar_policy=Gtk.POLICY_AUTOMATIC) - scrollwin.set_shadow_type(Gtk.SHADOW_NONE) - vbox.pack_start(scrollwin) + scrollwin.set_policy(hscrollbar_policy=Gtk.PolicyType.NEVER, vscrollbar_policy=Gtk.PolicyType.AUTOMATIC) + scrollwin.set_shadow_type(Gtk.ShadowType.NONE) + scrollwin.set_vexpand(True) + vbox.add(scrollwin) - self.scales_vbox = Gtk.VBox() + self.scales_vbox = Gtk.Grid() + self.scales_vbox.set_orientation(Gtk.Orientation.VERTICAL) scrollwin.add_with_viewport(self.scales_vbox) label = Gtk.Label() label.set_single_line_mode(True) - line_height = label.size_request()[1] + line_height = label.size_request().height label.destroy() scale = Gtk.HScale() scale.set_draw_value(False) - line_height += scale.size_request()[1] + line_height += scale.size_request().height scale.destroy() # always have space for at least four sliders scrollwin.set_size_request(width=-1, height=line_height*4+4) @@ -252,7 +255,7 @@ class MixerWindow(Gtk.Window): self.streams.append(stream) for fd,condition in self.hcontrol.poll_fds: - self.hctl_sources.append(GLib.io_add_watch(fd, condition, self.hctl_io_callback)) + self.hctl_sources.append(GLib.io_add_watch(fd, 0, GLib.IOCondition(condition), self.hctl_io_callback)) self.update_msg_label() @@ -271,7 +274,8 @@ class MixerWindow(Gtk.Window): msg = "This card does not have stream controls." if not has_msg: self.msg_label = Gtk.Label(msg) - self.scales_vbox.pack_start(self.msg_label) + self.msg_label.set_vexpand(True) + self.scales_vbox.add(self.msg_label) self.scales_vbox.show_all() elif self.msg_label.get_text() != msg: self.msg_label.set_text(msg) @@ -284,8 +288,8 @@ class MixerWindow(Gtk.Window): except: # TODO: alsa error msg dlg = Gtk.MessageDialog(self, - Gtk.DIALOG_MODAL | Gtk.DIALOG_DESTROY_WITH_PARENT, - Gtk.MESSAGE_ERROR, Gtk.BUTTONS_OK, + Gtk.DialogFlags.MODAL | Gtk.DialogFlags.DESTROY_WITH_PARENT, + Gtk.MessageType.ERROR, Gtk.ButtonsType.OK, "Cannot open sound card control device.") dlg.run() dlg.destroy() -- 2.18.0

1 0

[alsa-devel] [PATCH v2] ASoC: rt5677: Fix initialization of rt5677_of_match.data
by Matthias Kaehlcke 08 Aug '18

08 Aug '18

The driver expects to find the device id in rt5677_of_match.data, however it is currently assigned to rt5677_of_match.type. Fix this. The problem was found with the help of clang: sound/soc/codecs/rt5677.c:5010:36: warning: expression which evaluates to zero treated as a null pointer constant of type 'const void *' [-Wnon-literal-null-conversion] { .compatible = "realtek,rt5677", RT5677 }, ^~~~~~ Fixes: ddc9e69b9dc2 ("ASoC: rt5677: Hide platform data in the module sources") Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> -- Changes in v2: - assign device id to data instead of 'type' - updated subject and commit message - added: 'Fixes' tag --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8a0181a2db08..47feef30dadb 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5007,7 +5007,7 @@ static const struct regmap_config rt5677_regmap = { }; static const struct of_device_id rt5677_of_match[] = { - { .compatible = "realtek,rt5677", RT5677 }, + { .compatible = "realtek,rt5677", .data = (const void *)RT5677 }, { } }; MODULE_DEVICE_TABLE(of, rt5677_of_match); -- 2.18.0.597.ga71716f1ad-goog

4 4

[alsa-devel] [PATCH] ASoC: wm_adsp: Correct DSP pointer for preloader control
by Charles Keepax 08 Aug '18

08 Aug '18

From: Ajit Pandey <ajit.pandey(a)cirrus.com> The offset of the DSP core needs to be taken into account for the DSP preloader control get and put. Currently the dsp->preloaded variable will only ever be read/updated on the first DSP, whilst this doesn't affect the operation of the control the readback will be incorrect. Signed-off-by: Ajit Pandey <ajit.pandey(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3bc1bb5504a1..ae131376cea1 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2665,7 +2665,10 @@ int wm_adsp2_preloader_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; ucontrol->value.integer.value[0] = dsp->preloaded; @@ -2677,10 +2680,11 @@ int wm_adsp2_preloader_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); - struct wm_adsp *dsp = snd_soc_component_get_drvdata(component); + struct wm_adsp *dsps = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + struct wm_adsp *dsp = &dsps[mc->shift - 1]; char preload[32]; snprintf(preload, ARRAY_SIZE(preload), "DSP%u Preload", mc->shift); -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: max98373: Added software reset register to readable registers
by Ryan Lee 08 Aug '18

08 Aug '18

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changes : Added 'MAX98373_R2000_SW_RESET' register to 'max98373_readable_register' Software reset didn't work properly after suspend/resume. sound/soc/codecs/max98373.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index a925861..92b7125 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -488,6 +488,7 @@ static const DECLARE_TLV_DB_RANGE(max98373_bde_gain_tlv, static bool max98373_readable_register(struct device *dev, unsigned int reg) { switch (reg) { + case MAX98373_R2000_SW_RESET: case MAX98373_R2001_INT_RAW1 ... MAX98373_R200C_INT_EN3: case MAX98373_R2010_IRQ_CTRL: case MAX98373_R2014_THERM_WARN_THRESH -- 2.7.4

2 1

Re: [alsa-devel] A bug about cache inconsistency report
by Hans Hu(SH-RD) 08 Aug '18

08 Aug '18

> > > 1. noise relate test: > > > When I annotate the return after set_pages_array_wc, then > > > set_memory_wc works, noise disappear static void > > > __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, > > > bool > > > on) { ... > > > set_pages_array_wb(sgbuf->page_table, sgbuf->pages); //return; } > > > #endif > > > > > set_memory_uc() performs __pa() for the given address, and in this case, it's vmapped address. Something bogus in your test, I'm afraid. > > > > [Hans:] I know __pa() handle the vmapped address is wrong, that why I don't know how to fix the problem, I try this because I see kernel older than 3.7.6. [bogus? pretending to be real or genuine?] no no no, but the noise problem machine(8086:1c20) in another city, I will make sure it again. > > >I meant about the reliability of the test result. Or whether we're heading to a wrong fish. > >In anyway, which kernel version are you testing? And exactly what is your problem? Which machine, which hardware setup (codec, etc) and which output, and how is the sound backend (PulseAudio or direct ALSA write)? > > [Hans:] OS: Ubuntu 17.10 + 4.13.0-21-generic (or any one after 3.7.6). > HW: mother board is Dell 042P49, HDA controller is 8086:1c20, codec is cx20641. > Output and sound backend: Except the command below, anything other is default setting. > > Problem(what I have done and found): When system setup, I add "modprobe snd-had-intel.snoop=0" in grub, after setup, execute command "aplay -D plughw:0,0 44.1k_16b_2c.wav", then I hear music with noise. The reason cause this is the buffer address should been mark as WC type and the __mark_pages_wc was patched for this, but if define CONFIG_SND_DMA_SGBUF, the actual marked address is the address before vmapped(0xffff88*...*), however the address used in snd_pcm_lib_write_transfer() is the address after vmapped(0xffffc9*...*), so the mark didn't work and cache inconsistency occur(noise). >Well, basically the vmapped address isn't used *at all* on the HD-audio controller hardware. The SG buffer addresses are passed on BDL of the chip, but the vmapped address is used only in PCM core for some data manipulation. And this is purely software stuff, hence it shouldn't matter there whether it's in snoop mode or not. The snoop mode difference appears mainly when accessed via mmap from user-space directly to kernel-space. >That said, I guess your suspect heading to a wrong direction. [Hans:] So, am I misunderstand? Or snoop has two different meanings? As I know, In the hardware layer HDAC’s stream have two data transport path : non-snoop & snoop; In the software layer ALSA-Driver have two data transport path : mmap & not mmap(test shows, it is dependent on wav's format or mmap_flag in aplay.c). When hardware at non-snoop mode, without hardware module's help, software must mark the mem to WC type: when mmap used, the mark action happened at pcm_mmap_prepare(), the not-mmap mode's mark action happened at __mark_pages_wc(). And when at not-mmap mode, the vmapped address directly used in snd_pcm_lib_write_transfer() -> copy_from_user(). Yes, of course, this path combinations(non-snoop + not-mmap) unlikely be used in most music players. > > > And moreover, all Intel chipsets do work with snooping well, AFAIK. > > [Hans:] yes, all Intel chipsets do work with snooping well here too. And the problem case here is "non-snoop"+"aplay -D plughw:*.*"+"44.1k 16bit"+"8086:1c20", I will see if I can find more machine do the test. > >Ah, so you explicitly test with non-snoop mode for this chipset? > >For what purpose...? > > [Hans:] Yes, I've been testing a HDA controller recently(predecessor is VIA), it has the same problem. I want to figure out if it is hardware's problem or not. > > [Hans:] I did the same test on another platform(8086:A170), it do work with non-snoop well. In addition, I did such a change, oneliner, the purpose is to mark nothing, and it still do work with non-snoop well, this supposed to be right? > > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > #ifdef CONFIG_X86 > static void __mark_pages_wc(struct azx *chip, struct snd_dma_buffer > *dmab, bool on) { int pages; > +return; >This is essentially other way round; it skips the non-cache setup. > > > 2. change *_SG relate test: > > > Just followed your proposal > > > -snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, > > > +snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > > > > >OK, you need one more oneliner: > > > > >diff --git a/sound/pci/hda/hda_controller.c > > >b/sound/pci/hda/hda_controller.c index a12e594d4e3b..2eaed4956e7f > > >100644 > > >--- a/sound/pci/hda/hda_controller.c > > >+++ b/sound/pci/hda/hda_controller.c > > >@@ -707,7 +707,7 @@ static const struct snd_pcm_ops azx_pcm_ops = { > > >.pointer = azx_pcm_pointer, > > >.get_time_info = azx_get_time_info, >.mmap = azx_pcm_mmap, > > >-.page = snd_pcm_sgbuf_ops_page, > > >+// .page = snd_pcm_sgbuf_ops_page, > > > }; > > > > > static void azx_pcm_free(struct snd_pcm *pcm) > > [Hans :] I did this, and the Oops report as the same. > > >My bad, there are other SG-buffer codes in sound/hda/*, too. > > >So scratch these two oneliners. The only thing you'd need is the big-hammer change (again oneliner :) > > >--- a/sound/core/Kconfig > >+++ b/sound/core/Kconfig > >@@ -181,7 +181,7 @@ config SND_VMASTER > >bool > > > > config SND_DMA_SGBUF > >-def_bool y > >+def_bool n > > depends on X86 > > > source "sound/core/seq/Kconfig" > > [Hans :] If this, All the platform are all right. >Then the next step would be to fake sg-buffer from this straight >buffer. Revert the above, and modify sgbuf.c to the following: >- Allocate a large continuous buffer >- Assign each page in this large buffer >If this still works, it's not about vmap, but it just means that the >physically ordered pages do matter -- implicitly showing that the >snooping behavior isn't properly turned on / off on the controller. To fake SG-buffer, I did this test: restore all the codes to the original, then added some codes in snd_malloc_sgbuf_pages() like below, the result is badly niose. ---a/sound/core/sgbuf.c +++b/sound/core/sgbuf.c void *snd_malloc_sgbuf_pages(struct device *device, size_t size, struct snd_dma_buffer *dmab, size_t *res_size) { struct snd_sg_buf *sgbuf; unsigned int i, pages, chunk, maxpages; struct snd_dma_buffer tmpb; struct snd_sg_page *table; struct page **pgtable; +size *= 2; /* expand the buffer two times */ dmab->area = NULL; dmab->addr = 0; ... ... if (chunk < maxpages) maxpages = chunk; } +for(i = 0; i < sgbuf->pages / 2; i ++) /* restore size to the original it wanted */ +sgbuf->page_table[i] = sgbuf->page_table[i * 2]; /* take a page every other page, just for test, ignore page release */ +size /= 2; +sgbuf->pages /= 2; sgbuf->size = size; dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); Thanks, Hans 保密声明：本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 CONFIDENTIAL NOTE: This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

2 7

[alsa-devel] [PATCH 00/14] conf/ucm: bytcr-rt5651: Various improvements + long-name profiles addition
by Hans de Goede 08 Aug '18

08 Aug '18

Hi All, Here is a set of patches to the bytcr-rt5651 UCM profiles. This series also adds a bunch of long-name profiles so that users automatically only see the inputs / outputs which their device actually has. The long-names are based on the long-names set by the kernel after the rt5651 patch series which is currently queued in -next: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/log/?h=fo… With these UCM profile patches and the kernel changes in place all inputs and outputs as well as jack-detect will work 100% OOTB on supported bytcr-rt5651 devices, without the user needing to do any manual config or mixer settings. Regards, Hans

3 22

[alsa-devel] Spectre x360 fixing the top speakers (again)
by Alex G. 08 Aug '18

08 Aug '18

It's only been since 4.18-rc2 that the speaker issue on the x360 [1] was resolved. I didn't even know I had four speakers until seeing that bug report. My HP Spectre x360 (13-ae0xx) has the same issue as the original x360 with Bang speakers. The subsystem PCI ID is different [103c:83b9]. Of the two fixes developed in [1], only the mute led fix works when the PCI id is added to patch-realtek. If I try to add the full ALC295_FIXUP_HP_X360 fix, there's no audio coming out at all. One thing I tried was to boot W10 virtual machine and redirect the audio PCI device to the VM: 0:1f.3 Audio device: Intel Corporation Sunrise Point-LP HD Audio (rev 21) It all worked in the VM, but as soon as I shut it down and tried to play audio from linux, the top speakers still didn't work. The difference being that PAVUcontrol could see a "4.0 output" configuration. The same issue: audio sent to the bottom speakers was heard, but audio sent to the top speakers was not. Since I'm likely going to have to change _everything_ anyway, here's what I'm trying to achieve: 1) Make the top speakers great again. Whether they produce the same sound as the bottom speakers, or have their own audio channel, I don't care that much. 2) Be able to use the speakers when something is plugged in the headphone jack. Right now it's stupid. I try to plug in a microphone (with the proper TRRS jack), and the speakers die. I haven't seen this before. I was playing with HDA-Analyzer and was able to get speaker sounds with a dummy plugged in the jack, so I know it's possible. I don't remember how I did it, and it only worked once. 3) (optional) see if there's a way to use the jack as a line in or stereo mic in. I'm not sure where to start. Alex [1] https://bugzilla.kernel.org/show_bug.cgi?id=189331

3 3

[alsa-devel] [PATCH] ASoC: Intel: hdac_hdmi: Add sampling frequencies constraint
by Yong Zhi 07 Aug '18

07 Aug '18

Playback of 44.1Khz contents with HDMI plugged returns "Invalid pipe config". This patch adds rate constraint to allow user space SRC to do the converting. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7b8533abf637..b222a2e91463 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -288,6 +288,38 @@ static unsigned int sad_sample_bits_lpcm(const u8 *sad) return (sad[2] & 7); } +/** + * sad_sample_rates_lpcm - Find supported sampling frequencies + * + * @sad: pointer to CEA_SADs entry byte 2 which details sampling frequencies + * supported according to CEA-861 EDID V3. In little endian byte order. + * + * bit 7: Reserved (0) + * bit 6: 192kHz + * bit 5: 176kHz + * bit 4: 96kHz + * bit 3: 88kHz + * bit 2: 48kHz + * bit 1: 44kHz + * bit 0: 32kHz + * + * Return: bitmask of supported sampling frequencies. + */ +static u8 sad_sample_rates_lpcm(const u8 *sad) +{ + return (sad[1] & 0x7F); +} + +static const unsigned int cea_sampling_freqs[7] = { + SNDRV_PCM_RATE_32000, /* 0: 32000Hz */ + SNDRV_PCM_RATE_44100, /* 1: 44100Hz */ + SNDRV_PCM_RATE_48000, /* 2: 48000Hz */ + SNDRV_PCM_RATE_88200, /* 3: 88200Hz */ + SNDRV_PCM_RATE_96000, /* 4: 96000Hz */ + SNDRV_PCM_RATE_176400, /* 5: 176400Hz */ + SNDRV_PCM_RATE_192000, /* 6: 192000Hz */ +}; + static int hdac_hdmi_eld_limit_formats(struct snd_pcm_runtime *runtime, void *eld) { @@ -301,6 +333,8 @@ static int hdac_hdmi_eld_limit_formats(struct snd_pcm_runtime *runtime, for (i = drm_eld_sad_count(eld_buf); i > 0; i--, sad += 3) { if (sad_format(sad) == 1) { /* AUDIO_CODING_TYPE_LPCM */ + unsigned int rates = 0; + u8 sad_rates, j; /* * the controller support 20 and 24 bits in 32 bit @@ -308,6 +342,16 @@ static int hdac_hdmi_eld_limit_formats(struct snd_pcm_runtime *runtime, */ if (sad_sample_bits_lpcm(sad) & 0x6) formats |= SNDRV_PCM_FMTBIT_S32; + + sad_rates = sad_sample_rates_lpcm(sad); + /* Filter out 44.1, 88.2 and 176.4Khz */ + for (j = 0; j < 7; j += 2) + if (sad_rates & BIT(j)) + rates |= cea_sampling_freqs[j]; + + snd_pcm_hw_constraint_mask64(runtime, + SNDRV_PCM_HW_PARAM_RATE, + rates); } } -- 2.7.4

4 6

[alsa-devel] [PATCH] ASoC: rsnd: Document R-Car M3-N support
by Yoshihiro Kaneko 07 Aug '18

07 Aug '18

From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Document support for the sound modules in the Renesas M3-N (r8a77965) SoC. No driver update is needed. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Yoshihiro Kaneko <ykaneko0929(a)gmail.com> --- This patch is based on the devel branch of Simon Horman's renesas tree. Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index b86d790..9e764270 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -352,6 +352,7 @@ Required properties: - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) + - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 1.9.1

4 3

[alsa-devel] [PATCH] ASoC: rt5677: Remove bogus device id from rt5677_of_match
by Matthias Kaehlcke 07 Aug '18

07 Aug '18

The initializer of rt5677_of_match contains a bogus device id (enum type) for the 'type' field (char []). Remove it. Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/codecs/rt5677.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8a0181a2db08..7dab53b35cbf 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5007,7 +5007,7 @@ static const struct regmap_config rt5677_regmap = { }; static const struct of_device_id rt5677_of_match[] = { - { .compatible = "realtek,rt5677", RT5677 }, + { .compatible = "realtek,rt5677", }, { } }; MODULE_DEVICE_TABLE(of, rt5677_of_match); -- 2.18.0.597.ga71716f1ad-goog

2 2

[alsa-devel] [RFC PATCH] ASoC: compress: make BE and FE order inline with dpcm
by Srinivas Kandagatla 07 Aug '18

07 Aug '18

For some reason order of startup/hw_params/prepare are reversed in dynamic compress usecase when compared to dpcm usecase. This is a issue with platforms like QCOM where it expects the BE to be initialized before FE. Interestingly the compress trigger callback order is inline with dpcm. Am not 100% sure why the compress audio case has been reversed. This patch is making the order inline with dpcm. If the reverse ordering is just co-incendental then this change makes sense and will avoid inventing some new mechanism to cope with ordering. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Hi All, This patch is making the order of startup/hw_params/prepare in dynamic compress usecase inline with dpcm case. As I could not figure out the reason why this was done in reverse order. So am sending this patch to get some insights of this. It would be great If someone from Intel can test this patch to verify if the new ordering is okay. Without this patch, we have to come up with a new mechanism to enforce ordering on QCom platform to get compress offload working. Thanks, Srini sound/soc/soc-compress.c | 96 +++++++++++++++++++++++------------------------- 1 file changed, 46 insertions(+), 50 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b9e1673fea51..409d082e80d1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -140,6 +140,30 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) stream = SNDRV_PCM_STREAM_CAPTURE; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + fe->dpcm[stream].runtime = fe_substream->runtime; + + ret = dpcm_path_get(fe, stream, &list); + if (ret < 0) + goto be_err; + else if (ret == 0) + dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", + fe->dai_link->name, stream ? "capture" : "playback"); + /* calculate valid and active FE <-> BE dpcms */ + dpcm_process_paths(fe, stream, &list, 1); + fe->dpcm[stream].runtime = fe_substream->runtime; + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_startup(fe, stream); + if (ret < 0) { + /* clean up all links */ + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) + dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; + + dpcm_be_disconnect(fe, stream); + fe->dpcm[stream].runtime = NULL; + goto out; + } if (cpu_dai->driver->cops && cpu_dai->driver->cops->startup) { ret = cpu_dai->driver->cops->startup(cstream, cpu_dai); @@ -153,7 +177,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) ret = soc_compr_components_open(cstream, &component); if (ret < 0) - goto machine_err; + goto open_err; if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->startup) { ret = fe->dai_link->compr_ops->startup(cstream); @@ -164,31 +188,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) } } - fe->dpcm[stream].runtime = fe_substream->runtime; - - ret = dpcm_path_get(fe, stream, &list); - if (ret < 0) - goto fe_err; - else if (ret == 0) - dev_dbg(fe->dev, "Compress ASoC: %s no valid %s route\n", - fe->dai_link->name, stream ? "capture" : "playback"); - - /* calculate valid and active FE <-> BE dpcms */ - dpcm_process_paths(fe, stream, &list, 1); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_startup(fe, stream); - if (ret < 0) { - /* clean up all links */ - list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) - dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - - dpcm_be_disconnect(fe, stream); - fe->dpcm[stream].runtime = NULL; - goto path_err; - } - dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); @@ -201,17 +200,14 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) return 0; -path_err: - dpcm_path_put(&list); -fe_err: - if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) - fe->dai_link->compr_ops->shutdown(cstream); machine_err: soc_compr_components_free(cstream, component); - +open_err: if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown) cpu_dai->driver->cops->shutdown(cstream, cpu_dai); out: + dpcm_path_put(&list); +be_err: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; mutex_unlock(&fe->card->mutex); return ret; @@ -551,6 +547,24 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); + /* + * Create an empty hw_params for the BE as the machine driver must + * fix this up to match DSP decoder and ASRC configuration. + * I.e. machine driver fixup for compressed BE is mandatory. + */ + memset(&fe->dpcm[fe_substream->stream].hw_params, 0, + sizeof(struct snd_pcm_hw_params)); + + fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; + + ret = dpcm_be_dai_hw_params(fe, stream); + if (ret < 0) + goto out; + + ret = dpcm_be_dai_prepare(fe, stream); + if (ret < 0) + goto out; + if (cpu_dai->driver->cops && cpu_dai->driver->cops->set_params) { ret = cpu_dai->driver->cops->set_params(cstream, params, cpu_dai); if (ret < 0) @@ -577,24 +591,6 @@ static int soc_compr_set_params_fe(struct snd_compr_stream *cstream, goto out; } - /* - * Create an empty hw_params for the BE as the machine driver must - * fix this up to match DSP decoder and ASRC configuration. - * I.e. machine driver fixup for compressed BE is mandatory. - */ - memset(&fe->dpcm[fe_substream->stream].hw_params, 0, - sizeof(struct snd_pcm_hw_params)); - - fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; - - ret = dpcm_be_dai_hw_params(fe, stream); - if (ret < 0) - goto out; - - ret = dpcm_be_dai_prepare(fe, stream); - if (ret < 0) - goto out; - dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_START); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_PREPARE; -- 2.16.2

2 1

[alsa-devel] [PATCH] soundwire: intel: Remove duplicate assignment
by Vinod Koul 07 Aug '18

07 Aug '18

From: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/intel.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index da7f91f1a588..c5ee97ee7886 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -845,9 +845,6 @@ static int intel_probe(struct platform_device *pdev) sdw_intel_ops.read_prop = intel_prop_read; sdw->cdns.bus.ops = &sdw_intel_ops; - sdw_intel_ops.read_prop = intel_prop_read; - sdw->cdns.bus.ops = &sdw_intel_ops; - platform_set_drvdata(pdev, sdw); ret = sdw_add_bus_master(&sdw->cdns.bus); -- 2.14.4

2 1

[alsa-devel] [PATCH] soundwire: intel: Fix uninitialized adev deref
by Vinod Koul 07 Aug '18

07 Aug '18

In case of error, we can dereference uninitialized 'adev' drivers/soundwire/intel_init.c:154 sdw_intel_acpi_cb() error: uninitialized symbol 'adev'. Fix that by not using adev for warn print and make it pr_err. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/intel_init.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/intel_init.c b/drivers/soundwire/intel_init.c index d1ea6b4d0ad3..5c8a20d99878 100644 --- a/drivers/soundwire/intel_init.c +++ b/drivers/soundwire/intel_init.c @@ -151,7 +151,7 @@ static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, struct acpi_device *adev; if (acpi_bus_get_device(handle, &adev)) { - dev_err(&adev->dev, "Couldn't find ACPI handle\n"); + pr_err("%s: Couldn't find ACPI handle\n", __func__); return AE_NOT_FOUND; } -- 2.14.4

2 1

[alsa-devel] [bug report] slimbus: ngd: Add qcom SLIMBus NGD driver
by Dan Carpenter 07 Aug '18

07 Aug '18

Hello Srinivas Kandagatla, This is a semi-automatic email about new static checker warnings. The patch 917809e2280b: "slimbus: ngd: Add qcom SLIMBus NGD driver" from Jun 19, 2018, leads to the following Smatch complaint: drivers/slimbus/qcom-ngd-ctrl.c:867 qcom_slim_ngd_xfer_msg() warn: variable dereferenced before check 'txn->msg' (see line 791) drivers/slimbus/qcom-ngd-ctrl.c 790 791 if (txn->msg->num_bytes > SLIM_MSGQ_BUF_LEN || ^^^^^^^^^ Dereference 792 txn->rl > SLIM_MSGQ_BUF_LEN) { 793 dev_err(ctrl->dev, "msg exeeds HW limit\n"); 794 return -EINVAL; 795 } 796 797 pbuf = qcom_slim_ngd_tx_msg_get(ctrl, txn->rl, &tx_sent); 798 if (!pbuf) { 799 dev_err(ctrl->dev, "Message buffer unavailable\n"); 800 return -ENOMEM; 801 } 802 803 if (txn->mt == SLIM_MSG_MT_CORE && 804 (txn->mc == SLIM_MSG_MC_CONNECT_SOURCE || 805 txn->mc == SLIM_MSG_MC_CONNECT_SINK || 806 txn->mc == SLIM_MSG_MC_DISCONNECT_PORT)) { 807 txn->mt = SLIM_MSG_MT_DEST_REFERRED_USER; 808 switch (txn->mc) { 809 case SLIM_MSG_MC_CONNECT_SOURCE: 810 txn->mc = SLIM_USR_MC_CONNECT_SRC; 811 break; 812 case SLIM_MSG_MC_CONNECT_SINK: 813 txn->mc = SLIM_USR_MC_CONNECT_SINK; 814 break; 815 case SLIM_MSG_MC_DISCONNECT_PORT: 816 txn->mc = SLIM_USR_MC_DISCONNECT_PORT; 817 break; 818 default: 819 return -EINVAL; 820 } 821 822 usr_msg = true; 823 i = 0; 824 wbuf[i++] = txn->la; 825 la = SLIM_LA_MGR; 826 wbuf[i++] = txn->msg->wbuf[0]; 827 if (txn->mc != SLIM_USR_MC_DISCONNECT_PORT) 828 wbuf[i++] = txn->msg->wbuf[1]; 829 830 txn->comp = &done; 831 ret = slim_alloc_txn_tid(sctrl, txn); 832 if (ret) { 833 dev_err(ctrl->dev, "Unable to allocate TID\n"); 834 return ret; 835 } 836 837 wbuf[i++] = txn->tid; 838 839 txn->msg->num_bytes = i; 840 txn->msg->wbuf = wbuf; 841 txn->msg->rbuf = rbuf; 842 txn->rl = txn->msg->num_bytes + 4; 843 } 844 845 /* HW expects length field to be excluded */ 846 txn->rl--; 847 puc = (u8 *)pbuf; 848 *pbuf = 0; 849 if (txn->dt == SLIM_MSG_DEST_LOGICALADDR) { 850 *pbuf = SLIM_MSG_ASM_FIRST_WORD(txn->rl, txn->mt, txn->mc, 0, 851 la); 852 puc += 3; 853 } else { 854 *pbuf = SLIM_MSG_ASM_FIRST_WORD(txn->rl, txn->mt, txn->mc, 1, 855 la); 856 puc += 2; 857 } 858 859 if (slim_tid_txn(txn->mt, txn->mc)) 860 *(puc++) = txn->tid; 861 862 if (slim_ec_txn(txn->mt, txn->mc)) { 863 *(puc++) = (txn->ec & 0xFF); 864 *(puc++) = (txn->ec >> 8) & 0xFF; 865 } 866 867 if (txn->msg && txn->msg->wbuf) ^^^^^^^^ This check is too late. 868 memcpy(puc, txn->msg->wbuf, txn->msg->num_bytes); 869 regards, dan carpenter

2 1

[alsa-devel] [PATCH v4 23/26] devicetree: fix a series of wrong file references
by Mauro Carvalho Chehab 07 Aug '18

07 Aug '18

As files got renamed, their references broke. Manually fix a series of broken refs at the DT bindings. Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> --- .../devicetree/bindings/input/rmi4/rmi_2d_sensor.txt | 2 +- Documentation/devicetree/bindings/mfd/sun6i-prcm.txt | 2 +- .../devicetree/bindings/pci/hisilicon-pcie.txt | 2 +- Documentation/devicetree/bindings/pci/kirin-pcie.txt | 2 +- .../devicetree/bindings/pci/pci-keystone.txt | 4 ++-- .../devicetree/bindings/sound/st,stm32-i2s.txt | 2 +- .../devicetree/bindings/sound/st,stm32-sai.txt | 2 +- MAINTAINERS | 12 ++++++------ 8 files changed, 14 insertions(+), 14 deletions(-) diff --git a/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt b/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt index f2c30c8b725d..9afffbdf6e28 100644 --- a/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt +++ b/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt @@ -12,7 +12,7 @@ Additional documentation for F11 can be found at: http://www.synaptics.com/sites/default/files/511-000136-01-Rev-E-RMI4-Inter… Optional Touch Properties: -Description in Documentation/devicetree/bindings/input/touch +Description in Documentation/devicetree/bindings/input/touchscreen - touchscreen-inverted-x - touchscreen-inverted-y - touchscreen-swapped-x-y diff --git a/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt b/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt index 4d21ffdb0fc1..daa091c2e67b 100644 --- a/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt +++ b/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt @@ -8,7 +8,7 @@ Required properties: - reg: The PRCM registers range The prcm node may contain several subdevices definitions: - - see Documentation/devicetree/clk/sunxi.txt for clock devices + - see Documentation/devicetree/bindings/clock/sunxi.txt for clock devices - see Documentation/devicetree/bindings/reset/allwinner,sunxi-clock-reset.txt for reset controller devices diff --git a/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt b/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt index 7bf9df047a1e..0dcb87d6554f 100644 --- a/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt +++ b/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt @@ -3,7 +3,7 @@ HiSilicon Hip05 and Hip06 PCIe host bridge DT description HiSilicon PCIe host controller is based on the Synopsys DesignWare PCI core. It shares common functions with the PCIe DesignWare core driver and inherits common properties defined in -Documentation/devicetree/bindings/pci/designware-pci.txt. +Documentation/devicetree/bindings/pci/designware-pcie.txt. Additional properties are described here: diff --git a/Documentation/devicetree/bindings/pci/kirin-pcie.txt b/Documentation/devicetree/bindings/pci/kirin-pcie.txt index 6e217c63123d..6bbe43818ad5 100644 --- a/Documentation/devicetree/bindings/pci/kirin-pcie.txt +++ b/Documentation/devicetree/bindings/pci/kirin-pcie.txt @@ -3,7 +3,7 @@ HiSilicon Kirin SoCs PCIe host DT description Kirin PCIe host controller is based on the Synopsys DesignWare PCI core. It shares common functions with the PCIe DesignWare core driver and inherits common properties defined in -Documentation/devicetree/bindings/pci/designware-pci.txt. +Documentation/devicetree/bindings/pci/designware-pcie.txt. Additional properties are described here: diff --git a/Documentation/devicetree/bindings/pci/pci-keystone.txt b/Documentation/devicetree/bindings/pci/pci-keystone.txt index 7e05487544ed..3d4a209b0fd0 100644 --- a/Documentation/devicetree/bindings/pci/pci-keystone.txt +++ b/Documentation/devicetree/bindings/pci/pci-keystone.txt @@ -3,9 +3,9 @@ TI Keystone PCIe interface Keystone PCI host Controller is based on the Synopsys DesignWare PCI hardware version 3.65. It shares common functions with the PCIe DesignWare core driver and inherits common properties defined in -Documentation/devicetree/bindings/pci/designware-pci.txt +Documentation/devicetree/bindings/pci/designware-pcie.txt -Please refer to Documentation/devicetree/bindings/pci/designware-pci.txt +Please refer to Documentation/devicetree/bindings/pci/designware-pcie.txt for the details of DesignWare DT bindings. Additional properties are described here as well as properties that are not applicable. diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt index 4bda52042402..58c341300552 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt @@ -18,7 +18,7 @@ Required properties: See Documentation/devicetree/bindings/dma/stm32-dma.txt. - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt Optional properties: - resets: Reference to a reset controller asserting the reset controller diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt index f301cdf0b7e6..3a3fc506e43a 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt @@ -37,7 +37,7 @@ SAI subnodes required properties: "tx": if sai sub-block is configured as playback DAI "rx": if sai sub-block is configured as capture DAI - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt SAI subnodes Optional properties: - st,sync: specify synchronization mode. diff --git a/MAINTAINERS b/MAINTAINERS index 783ce44b3e7a..fd3fc63f2759 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6966,7 +6966,7 @@ IIO MULTIPLEXER M: Peter Rosin <peda(a)axentia.se> L: linux-iio(a)vger.kernel.org S: Maintained -F: Documentation/devicetree/bindings/iio/multiplexer/iio-mux.txt +F: Documentation/devicetree/bindings/iio/multiplexer/io-channel-mux.txt F: drivers/iio/multiplexer/iio-mux.c IIO SUBSYSTEM AND DRIVERS @@ -9696,7 +9696,7 @@ MXSFB DRM DRIVER M: Marek Vasut <marex(a)denx.de> S: Supported F: drivers/gpu/drm/mxsfb/ -F: Documentation/devicetree/bindings/display/mxsfb-drm.txt +F: Documentation/devicetree/bindings/display/mxsfb.txt MYRICOM MYRI-10G 10GbE DRIVER (MYRI10GE) M: Chris Lee <christopher.lee(a)cspi.com> @@ -10885,7 +10885,7 @@ M: Will Deacon <will.deacon(a)arm.com> L: linux-pci(a)vger.kernel.org L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Maintained -F: Documentation/devicetree/bindings/pci/controller-generic-pci.txt +F: Documentation/devicetree/bindings/pci/host-generic-pci.txt F: drivers/pci/controller/pci-host-common.c F: drivers/pci/controller/pci-host-generic.c @@ -11066,7 +11066,7 @@ M: Xiaowei Song <songxiaowei(a)hisilicon.com> M: Binghui Wang <wangbinghui(a)hisilicon.com> L: linux-pci(a)vger.kernel.org S: Maintained -F: Documentation/devicetree/bindings/pci/pcie-kirin.txt +F: Documentation/devicetree/bindings/pci/kirin-pcie.txt F: drivers/pci/controller/dwc/pcie-kirin.c PCIE DRIVER FOR HISILICON STB @@ -12457,7 +12457,7 @@ L: linux-crypto(a)vger.kernel.org L: linux-samsung-soc(a)vger.kernel.org S: Maintained F: drivers/crypto/exynos-rng.c -F: Documentation/devicetree/bindings/crypto/samsung,exynos-rng4.txt +F: Documentation/devicetree/bindings/rng/samsung,exynos4-rng.txt SAMSUNG EXYNOS TRUE RANDOM NUMBER GENERATOR (TRNG) DRIVER M: Łukasz Stelmach <l.stelmach(a)samsung.com> @@ -13571,7 +13571,7 @@ F: drivers/*/stm32-*timer* F: drivers/pwm/pwm-stm32* F: include/linux/*/stm32-*tim* F: Documentation/ABI/testing/*timer-stm32 -F: Documentation/devicetree/bindings/*/stm32-*timer +F: Documentation/devicetree/bindings/*/stm32-*timer* F: Documentation/devicetree/bindings/pwm/pwm-stm32* STMMAC ETHERNET DRIVER -- 2.17.1

4 3

[alsa-devel] [PATCH v8 00/10] soundwire: Add multi link support
by Shreyas NC 07 Aug '18

07 Aug '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Fix to avoid duplicate stream state assignment - Avoid incorrect stream release after configuring Master - Fix to avoid bus lock acquired twice - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch Additionally, as suggested by Vinod, the generic fixes are arranged at the top of the series followed by the Documentation patch for multi link support and then the multi link patches. changes in v8: - Check both m_rt_count and bus->multi_link flag for clean up after single instance bankswitch as suggested by Pierre. changes in v7: - Fix the git bisect issue pointed by Pierre. Now, the initialization of the m_rt_count and its modification/usage is split into 2 patches. While the declaration and initialization is in one patch, modification of the refcount and usage is in the multilink bankswitch patch. - Re-arrange the patches as suggested by Vinod Sanyog Kale (3): soundwire: Fix acquiring bus lock twice during master release Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances Shreyas NC (6): soundwire: Fix duplicate stream state assignment soundwire: Fix incorrect exit after configuring stream soundwire: Initialize completion for defer messages soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 511 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 489 insertions(+), 137 deletions(-) -- 2.7.4

3 12

[alsa-devel] [PATCH 0/9] ALSA: remove redundant variables
by Colin King 06 Aug '18

06 Aug '18

From: Colin Ian King <colin.king(a)canonical.com> Clean up code by removing redundant variables that were detected by clang Colin Ian King (9): ALSA: asihpi: remove redundant variable max_streams ALSA: cs5535audio: remove redundant pointer 'dma' ALSA: emu10k1: remove redundant variable attn ALSA: ens137x: remove redundant array pcm_devs ALSA: sonicvibes: remove redundant pointer 'dir' ASoC: nau8540: remove redundant variable osrate ASoC: stm32: remove redundant pointers 'priv' and 'rtd' ALSA: usb-audio: remove redundant pointer 'urb' ALSA: intel_hdmi: remove redundant variable cfg_val sound/pci/asihpi/hpi6205.c | 5 ----- sound/pci/cs5535audio/cs5535audio.c | 7 +------ sound/pci/emu10k1/emupcm.c | 3 +-- sound/pci/ens1370.c | 3 +-- sound/pci/sonicvibes.c | 2 -- sound/soc/codecs/nau8540.c | 3 --- sound/soc/stm/stm32_adfsdm.c | 8 +------- sound/usb/endpoint.c | 2 -- sound/x86/intel_hdmi_audio.c | 2 -- 9 files changed, 4 insertions(+), 31 deletions(-) -- 2.17.1

4 22

[alsa-devel] Applied "ASoC: AMD: Make ACP->SYSMEM DMA non circular" to the asoc tree
by Mark Brown 06 Aug '18

06 Aug '18

The patch ASoC: AMD: Make ACP->SYSMEM DMA non circular has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bbdb7012b0736cda0b9b00a2949e9207cf2f892f Mon Sep 17 00:00:00 2001 From: Akshu Agrawal <akshu.agrawal(a)amd.com> Date: Mon, 6 Aug 2018 12:57:14 +0530 Subject: [PATCH] ASoC: AMD: Make ACP->SYSMEM DMA non circular In capture case we don't want ACP to SYSMEM dma to be circular. This is because if an in place DSP filter is applied to captured output then circular DMA can overwrite the filter value with stale data. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-pcm-dma.c | 36 +++++++++++++++++++++++++++++++----- 1 file changed, 31 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 94bcf69008df..816abd65a6ed 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -400,7 +400,7 @@ static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, } /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular) { u32 dma_ctrl; @@ -429,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) break; } - /* circular for both DMA channel */ - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* enable for ACP to SRAM DMA channel */ + if (is_circular == true) + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + else + dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -674,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { + u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -705,6 +709,15 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) == + CAPTURE_START_DMA_DESCR_CH15) + dscr_idx = CAPTURE_END_DMA_DESCR_CH14; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH14; + config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, + 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -712,6 +725,17 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, @@ -1053,9 +1077,11 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_cap_channel_enable(rtd->acp_mmio, CAP_CHANNEL1); } + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + } else { + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); } - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: -- 2.18.0

1 0

[alsa-devel] [PATCH 1/3] ASoC: AMD: Make ACP->SYSMEM DMA non circular
by Akshu Agrawal 06 Aug '18

06 Aug '18

In capture case we don't want ACP to SYSMEM dma to be circular. This is because if an in place DSP filter is applied to captured output then circular DMA can overwrite the filter value with stale data. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 36 +++++++++++++++++++++++++++++++----- 1 file changed, 31 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 94bcf69..816abd6 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -400,7 +400,7 @@ static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, } /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular) { u32 dma_ctrl; @@ -429,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) break; } - /* circular for both DMA channel */ - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* enable for ACP to SRAM DMA channel */ + if (is_circular == true) + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + else + dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -674,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { + u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -705,6 +709,15 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) == + CAPTURE_START_DMA_DESCR_CH15) + dscr_idx = CAPTURE_END_DMA_DESCR_CH14; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH14; + config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, + 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -712,6 +725,17 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, @@ -1053,9 +1077,11 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_cap_channel_enable(rtd->acp_mmio, CAP_CHANNEL1); } + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + } else { + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); } - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: -- 1.9.1

2 4

[alsa-devel] [bug report] soundwire: intel: Add Intel init module
by Dan Carpenter 06 Aug '18

06 Aug '18

Hello Vinod Koul, The patch d62a7d41f38e: "soundwire: intel: Add Intel init module" from Dec 14, 2017, leads to the following static checker warning: drivers/soundwire/intel_init.c:154 sdw_intel_acpi_cb() error: uninitialized symbol 'adev'. drivers/soundwire/intel_init.c 147 static acpi_status sdw_intel_acpi_cb(acpi_handle handle, u32 level, 148 void *cdata, void **return_value) 149 { 150 struct sdw_intel_res *res = cdata; 151 struct acpi_device *adev; 152 153 if (acpi_bus_get_device(handle, &adev)) { 154 dev_err(&adev->dev, "Couldn't find ACPI handle\n"); 155 return AE_NOT_FOUND; We can't dereference "adev" if the acpi_bus_get_device() fails. 156 } 157 158 res->handle = handle; 159 return AE_OK; 160 } Related: drivers/acpi/scan.c:2008 acpi_walk_dep_device_list() error: uninitialized symbol 'adev'. drivers/usb/core/usb-acpi.c:204 usb_acpi_find_companion() error: uninitialized symbol 'adev'. regards, dan carpenter

3 2

[alsa-devel] [PATCH] ASoC: wm8994: Fix missing break in switch
by Gustavo A. R. Silva 06 Aug '18

06 Aug '18

Add missing break statement in order to prevent the code from falling through to the default case. Cc: stable(a)vger.kernel.org Addresses-Coverity-ID: 115050 ("Missing break in switch") Reported-by: Valdis Kletnieks <valdis.kletnieks(a)vt.edu> Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 62f8c5b..14f1b0c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2432,7 +2432,7 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, snd_soc_component_update_bits(component, WM8994_POWER_MANAGEMENT_2, WM8994_OPCLK_ENA, 0); } - /* fall through */ + break; default: return -EINVAL; -- 2.7.4

3 2

[alsa-devel] [PATCH 00/11] ASoC: codecs: Mark expected switch fall-throughs
by Gustavo A. R. Silva 06 Aug '18

06 Aug '18

Hi all, In preparation to enabling -Wimplicit-fallthrough, this patchset aims to add some annotations in order to mark switch cases where we are expecting to fall through. Thanks Gustavo A. R. Silva (11): ASoC: wm8961: Mark expected switch fall-through ASoC: rt5640: Mark expected switch fall-through ASoC: rt5677: Mark expected switch fall-through ASoC: wm8955: Mark expected switch fall-through ASoC: wm8960: Mark expected switch fall-through ASoC: wm8904: Mark expected switch fall-through ASoC: wm8996: Mark expected switch fall-through ASoC: wm8962: Mark expected switch fall-through ASoC: wm8995: Mark expected switch fall-through ASoC: wm9081: Mark expected switch fall-through ASoC: wm8994: Mark expected switch fall-through sound/soc/codecs/rt5640.c | 2 ++ sound/soc/codecs/rt5677.c | 1 + sound/soc/codecs/wm8904.c | 1 + sound/soc/codecs/wm8955.c | 1 + sound/soc/codecs/wm8960.c | 1 + sound/soc/codecs/wm8961.c | 1 + sound/soc/codecs/wm8962.c | 1 + sound/soc/codecs/wm8994.c | 1 + sound/soc/codecs/wm8995.c | 1 + sound/soc/codecs/wm8996.c | 1 + sound/soc/codecs/wm9081.c | 1 + 11 files changed, 12 insertions(+) -- 2.7.4

3 32

[alsa-devel] [PATCH 0/5] ASoC: qcom: few trival fixes
by Srinivas Kandagatla 06 Aug '18

06 Aug '18

This patchset have few trivial fixes, 3 of them to do with cleaning up the unused header files, 1 patch is to do with fixing warning on common.c and other is to add missing AIF_IN dapm for slim tx ports. Thanks, srini Srinivas Kandagatla (5): ASoC: qcom: make common.c as proper module ASoC: apq8096: remove unused header files ASoC: sdm845: remove unused header files ASoC: qcom: remove unused header files from common.h ASoC: qdsp6: q6afe-dai: add SLIM tx AIF_IN dapm sound/soc/qcom/Kconfig | 5 +++++ sound/soc/qcom/Makefile | 6 ++++-- sound/soc/qcom/apq8096.c | 2 -- sound/soc/qcom/common.h | 1 - sound/soc/qcom/qdsp6/q6afe-dai.c | 7 +++++++ sound/soc/qcom/sdm845.c | 2 -- 6 files changed, 16 insertions(+), 7 deletions(-) -- 2.16.2

2 10

[alsa-devel] [PATCH 00/12] ASoC: codecs: use true and false for boolean values
by Gustavo A. R. Silva 06 Aug '18

06 Aug '18

Hi all, This patchset aims to replace the use of integer values with a proper true or false in return statements for functions returning bool. Thanks Gustavo A. R. Silva (12): ASoC: max9850: use true and false for boolean values ASoC: rt5631: use true and false for boolean values ASoC: tda7419: use true and false for boolean values ASoC: wm8990: use true and false for boolean values ASoC: cs4270: use true and false for boolean values ASoC: wm8996: use true and false for boolean values ASoC: da7219: use true and false for boolean values ASoC: twl6040: use true and false for boolean values ASoC: da7213: use true and false for boolean values ASoC: wm5100-tables: use true and false for boolean values ASoC: da9055: use true and false for boolean values ASoC: wm8903: use true and false for boolean values sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/da7213.c | 4 ++-- sound/soc/codecs/da7219.c | 4 ++-- sound/soc/codecs/da9055.c | 4 ++-- sound/soc/codecs/max9850.c | 4 ++-- sound/soc/codecs/rt5631.c | 8 ++++---- sound/soc/codecs/tda7419.c | 4 ++-- sound/soc/codecs/twl6040.c | 2 +- sound/soc/codecs/wm5100-tables.c | 12 ++++++------ sound/soc/codecs/wm8903.c | 4 ++-- sound/soc/codecs/wm8990.c | 4 ++-- sound/soc/codecs/wm8996.c | 8 ++++---- 12 files changed, 30 insertions(+), 30 deletions(-) -- 2.7.4

3 26

[alsa-devel] Applied "ASoC: tda7419: use true and false for boolean values" to the asoc tree
by Mark Brown 06 Aug '18

06 Aug '18

The patch ASoC: tda7419: use true and false for boolean values has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 10754bfc051256ab922b818de3ddd11391862a5f Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sat, 4 Aug 2018 16:50:27 -0500 Subject: [PATCH] ASoC: tda7419: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tda7419.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tda7419.c b/sound/soc/codecs/tda7419.c index 225c210ac38f..7f3b79c5a563 100644 --- a/sound/soc/codecs/tda7419.c +++ b/sound/soc/codecs/tda7419.c @@ -142,9 +142,9 @@ struct tda7419_vol_control { static inline bool tda7419_vol_is_stereo(struct tda7419_vol_control *tvc) { if (tvc->reg == tvc->rreg) - return 0; + return false; - return 1; + return true; } static int tda7419_vol_info(struct snd_kcontrol *kcontrol, -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: wm5100-tables: use true and false for boolean values" to the asoc tree
by Mark Brown 06 Aug '18

06 Aug '18

The patch ASoC: wm5100-tables: use true and false for boolean values has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 380ae4ec420304aa59adb36776eaf78aef37192c Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Sat, 4 Aug 2018 16:54:54 -0500 Subject: [PATCH] ASoC: wm5100-tables: use true and false for boolean values Return statements in functions returning bool should use true or false instead of an integer value. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm5100-tables.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index e239f4bf2460..9e987cf07450 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -30,7 +30,7 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) case WM5100_OUTPUT_STATUS_2: case WM5100_INPUT_ENABLES_STATUS: case WM5100_MIC_DETECT_3: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -41,9 +41,9 @@ bool wm5100_volatile_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } @@ -798,7 +798,7 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) case WM5100_DSP3_CONTROL_28: case WM5100_DSP3_CONTROL_29: case WM5100_DSP3_CONTROL_30: - return 1; + return true; default: if ((reg >= WM5100_DSP1_PM_0 && reg <= WM5100_DSP1_PM_1535) || (reg >= WM5100_DSP1_ZM_0 && reg <= WM5100_DSP1_ZM_2047) || @@ -809,9 +809,9 @@ bool wm5100_readable_register(struct device *dev, unsigned int reg) (reg >= WM5100_DSP3_PM_0 && reg <= WM5100_DSP3_PM_1535) || (reg >= WM5100_DSP3_ZM_0 && reg <= WM5100_DSP3_ZM_2047) || (reg >= WM5100_DSP3_DM_0 && reg <= WM5100_DSP3_DM_511)) - return 1; + return true; else - return 0; + return false; } } -- 2.18.0

1 0

[alsa-devel] [PATCH] ALSA: dice: fix wrong copy to rx parameters for Alesis iO26
by Takashi Sakamoto 06 Aug '18

06 Aug '18

A commit 28b208f600a3 ('ALSA: dice: add parameters of stream formats for models produced by Alesis') adds wrong copy to rx parameters instead of tx parameters for Alesis iO26. This commit fixes the bug for v4.18-rc8. Fixes: 28b208f600a3 ('ALSA: dice: add parameters of stream formats for models produced by Alesis') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-alesis.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c index b2efb1c71a98..218292bdace6 100644 --- a/sound/firewire/dice/dice-alesis.c +++ b/sound/firewire/dice/dice-alesis.c @@ -37,7 +37,7 @@ int snd_dice_detect_alesis_formats(struct snd_dice *dice) MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); } else { - memcpy(dice->rx_pcm_chs, alesis_io26_tx_pcm_chs, + memcpy(dice->tx_pcm_chs, alesis_io26_tx_pcm_chs, MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); } -- 2.17.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: AMD: Make ACP->SYSMEM DMA non circular
by Akshu Agrawal 06 Aug '18

06 Aug '18

In capture case we don't want ACP to SYSMEM dma to be circular. This is because if an in place DSP filter is applied to captured output then circular DMA can overwrite the filter value with stale data. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 36 +++++++++++++++++++++++++++++++----- 1 file changed, 31 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 94bcf69..816abd6 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -400,7 +400,7 @@ static void acp_dma_cap_channel_disable(void __iomem *acp_mmio, } /* Start a given DMA channel transfer */ -static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) +static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num, bool is_circular) { u32 dma_ctrl; @@ -429,8 +429,11 @@ static void acp_dma_start(void __iomem *acp_mmio, u16 ch_num) break; } - /* circular for both DMA channel */ - dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + /* enable for ACP to SRAM DMA channel */ + if (is_circular == true) + dma_ctrl |= ACP_DMA_CNTL_0__Circular_DMA_En_MASK; + else + dma_ctrl &= ~ACP_DMA_CNTL_0__Circular_DMA_En_MASK; acp_reg_write(dma_ctrl, acp_mmio, mmACP_DMA_CNTL_0 + ch_num); } @@ -674,6 +677,7 @@ static int acp_deinit(void __iomem *acp_mmio) /* ACP DMA irq handler routine for playback, capture usecases */ static irqreturn_t dma_irq_handler(int irq, void *arg) { + u16 dscr_idx; u32 intr_flag, ext_intr_status; struct audio_drv_data *irq_data; void __iomem *acp_mmio; @@ -705,6 +709,15 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_14) == + CAPTURE_START_DMA_DESCR_CH15) + dscr_idx = CAPTURE_END_DMA_DESCR_CH14; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH14; + config_acp_dma_channel(acp_mmio, ACP_TO_SYSRAM_CH_NUM, dscr_idx, + 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_CH_NUM, false); + snd_pcm_period_elapsed(irq_data->capture_i2ssp_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_CH_NUM)) << 16, acp_mmio, mmACP_EXTERNAL_INTR_STAT); @@ -712,6 +725,17 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) if ((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) != 0) { valid_irq = true; + if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_10) == + CAPTURE_START_DMA_DESCR_CH11) + dscr_idx = CAPTURE_END_DMA_DESCR_CH10; + else + dscr_idx = CAPTURE_START_DMA_DESCR_CH10; + config_acp_dma_channel(acp_mmio, + ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + dscr_idx, 1, 0); + acp_dma_start(acp_mmio, ACP_TO_SYSRAM_BT_INSTANCE_CH_NUM, + false); + snd_pcm_period_elapsed(irq_data->capture_i2sbt_stream); acp_reg_write((intr_flag & BIT(I2S_TO_ACP_DMA_BT_INSTANCE_CH_NUM)) << 16, @@ -1053,9 +1077,11 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) acp_dma_cap_channel_enable(rtd->acp_mmio, CAP_CHANNEL1); } + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + } else { + acp_dma_start(rtd->acp_mmio, rtd->ch1, true); + acp_dma_start(rtd->acp_mmio, rtd->ch2, true); } - acp_dma_start(rtd->acp_mmio, rtd->ch1); - acp_dma_start(rtd->acp_mmio, rtd->ch2); ret = 0; break; case SNDRV_PCM_TRIGGER_STOP: -- 1.9.1

3 4

[alsa-devel] A question about period
by Yu-hsuan Hsu 06 Aug '18

06 Aug '18

Hi all, I have a question about the period we set in hw_params. I found different boards may have different explanations about it. I have two guesses about its meaning. 1. The period_size is the size of each hardware's consumption. If we set period size to N, the pcm will consume N frames each time. 2. The period_size is the size to control when hardware call interrupt. If we set period size to N, the pcm consume frames in its step. When the number of frames it consumes more than N, it will call interrupt. We can use snd_pcm_avail function to check the real available frames in the device. If guess 1 is correct, the size of consumption should be fixed. Else, setting period_size is nothing to do with hardware's consumption. I've checked some boards and found that each board has different behavior (Most of them meet guess 2). I'm confuse which one is correct. Thanks!

2 8

Re: [alsa-devel] 答复: 答复: A bug about cache inconsistency report
by Takashi Iwai 06 Aug '18

06 Aug '18

On Thu, 02 Aug 2018 15:59:31 +0200, Hans Hu(SH-RD) wrote: > > > 1. noise relate test: > > When I annotate the return after set_pages_array_wc, then > > set_memory_wc works, noise disappear static void > > __mark_pages_wc(struct azx *chip, struct snd_dma_buffer *dmab, bool > > on) { ... > > set_pages_array_wb(sgbuf->page_table, sgbuf->pages); //return; } > > #endif > > > set_memory_uc() performs __pa() for the given address, and in this case, it's vmapped address. Something bogus in your test, I'm afraid. > > [Hans:] I know __pa() handle the vmapped address is wrong, that why I don't know how to fix the problem, I try this because I see kernel older than 3.7.6. [bogus? pretending to be real or genuine?] no no no, but the noise problem machine(8086:1c20) in another city, I will make sure it again. I meant about the reliability of the test result. Or whether we're heading to a wrong fish. In anyway, which kernel version are you testing? And exactly what is your problem? Which machine, which hardware setup (codec, etc) and which output, and how is the sound backend (PulseAudio or direct ALSA write)? > > And moreover, all Intel chipsets do work with snooping well, AFAIK. > [Hans:] yes, all Intel chipsets do work with snooping well here too. And the problem case here is "non-snoop"+"aplay -D plughw:*.*"+"44.1k 16bit"+"8086:1c20", I will see if I can find more machine do the test. Ah, so you explicitly test with non-snoop mode for this chipset? For what purpose...? > > 2. change *_SG relate test: > > Just followed your proposal > > -snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, > > +snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > > >OK, you need one more oneliner: > > >diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index a12e594d4e3b..2eaed4956e7f 100644 > >--- a/sound/pci/hda/hda_controller.c > >+++ b/sound/pci/hda/hda_controller.c > >@@ -707,7 +707,7 @@ static const struct snd_pcm_ops azx_pcm_ops = { > >.pointer = azx_pcm_pointer, > >.get_time_info = azx_get_time_info, > >.mmap = azx_pcm_mmap, > >-.page = snd_pcm_sgbuf_ops_page, > >+// .page = snd_pcm_sgbuf_ops_page, > > }; > > > static void azx_pcm_free(struct snd_pcm *pcm) > [Hans :] I did this, and the Oops report as the same. My bad, there are other SG-buffer codes in sound/hda/*, too. So scratch these two oneliners. The only thing you'd need is the big-hammer change (again oneliner :) --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -181,7 +181,7 @@ config SND_VMASTER bool config SND_DMA_SGBUF - def_bool y + def_bool n depends on X86 source "sound/core/seq/Kconfig" > >And, again, please avoid top-posting. > [Hans :] sorry, I didn't understand at first. My mailbox is Outlook, am I right now? Yes, looks better now. thanks, Takashi

2 2

[alsa-devel] ca0132 firmware files
by Connor McAdams 06 Aug '18

06 Aug '18

Hi! I acquired permission from Creative Labs to distribute their firmware in the same fashion as the current Chromebook firmware a month or two ago, but the email had a disclaimer at the bottom saying I couldn't forward or copy the email to anyone else. Anyways, after requesting that the email be sent again without this, they have sent me the following email: >Dear Connor, > >Thank you for your reply. > >Based on your feedback, the updated linux driver is based on Chromebook project has been >accepted in the ALSA tree, which means that the code has been reviewed and approved by ALSA >maintainers. You may follow what has been done on Chromebook's project. The firmware must >remain in ALSA firmware tree and not subjected to GPL. > >If you require further assistance, please reply to this email and we will get back to you. > > >Regards, >Anthony Now, I don't really know what to do with that. I feel as though Creative probably has to initiate sending the firmware themselves, but I'm not sure. Any information on the next step I should take would be very helpful. Thanks, Connor.

1 0

Re: [alsa-devel] ALSA queries
by Chakravarthi Pradeep 05 Aug '18

05 Aug '18

On Sun 5 Aug, 2018, 14:19 Chakravarthi Pradeep, <doubleq7(a)gmail.com> wrote: > Hello Takashi, > > Thanks for your reply. > > periods_min = the minimal number of periods > period_bytes_min = the minimal size of bytes for one period > period_bytes_max = the maximal size of bytes for one period > > > periods_min = the minimal number of periods : what is meaning of > periods , is it minimal number of interrupts ? > period_bytes_min = the minimal size of bytes for one period : it > means, minimal size of bytes per interrupt in case of device, Is it > correct ? > what about period_max ? > > It's most likely the ALSA PCM core's safety stop; your driver seem to > have missed snd_pcm_period_elapsed() calls, so the hwptr isn't > updated, resulting in XRUN. ALSA PCM core checks such XRUN condition > with the own timer. > > I'm attaching my driver thread along with this mail. Can you please > let me know if I have missed something in audio thread. ?. How to make > sure in driver that, trigger stop should be called only when stop > command is sent from application. > > I'm getting cut cut cut ... noise along with audio in VLC application. > Initially for 2 or 4 seconds, cut cut cut noise is not heard in VLC > application, However after almost after 5 sec ,I can hear cut cut cut > noise in VLC application. With my analysis, hwptr is getting updated > properly however I have doubt in app_ptr. I'm attaching the excl sheet > with hw_ptr,app_ptr and buf_pos values. > > How to remove the cut cut cut ... noise in audio ? > > Regards, > Chakravarthi > > On Fri, Aug 3, 2018 at 1:05 PM Takashi Iwai <tiwai(a)suse.de> wrote: > > > > On Thu, 02 Aug 2018 16:31:13 +0200, > > Chakravarthi Pradeep wrote: > > > > > > I'm working on ALSA driver for PCIe card. My ALSA driver and it's > > > initializing struct snd_pcm_hardware with below parameter. > > > > > > /************************ code start > > > ************************************************/ > > > static struct snd_pcm_hardware audio_pcm_hardware = { > > > .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | > > > SNDRV_PCM_INFO_MMAP_VALID | > > > SNDRV_PCM_INFO_BLOCK_TRANSFER | > > > SNDRV_PCM_INFO_RESUME ), > > > .formats = (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S24_LE), > > > .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | > > > SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), > > > .rate_min = 44100, > > > .rate_max = 192000, > > > .channels_min = 2, > > > .channels_max = 8, > > > .buffer_bytes_max = 76800, /*75 kbytes */ > > > .period_bytes_min = 512,//4410, /* (channel (2) * sample_rate (44100) > > > * bit width (24)) / (60 * 8) */ > > > .period_bytes_max = 16*1024, > > > .periods_min = 10, > > > .periods_max = 255, > > > > > > }; > > > /************************ code end > > > ************************************************/ > > > > > > 1) I did not understand what is significance of periods_min , > > > period_bytes_min , period_bytes_max and periods_max. Can you please > > > tell me what is importance of these parameter and what value should be > > > mentioned according into ALSA. > > > > These three defines the values your hardware may accept: > > periods_min = the minimal number of periods > > period_bytes_min = the minimal size of bytes for one period > > period_bytes_max = the maximal size of bytes for one period > > > > > 2) snd_pcm_ops trigger callback is getting called in the driver when > > > application sends "start" command. But ALSA driver is stopping by > > > itself after one frame is copied to ALSA framework, without waiting > > > for "stop" command. > > > > > > For instance: > > > In trigger callback , I'm getting these logs after one frame is copied. > > > Trigger:Start (When Play button is selected/clicked in application, > > > Start command is sent to ALSA driver) > > > Dma transfer is completed. > > > Trigger:Stop. (When Stop button is selected/clicked in application, > > > Stop command is sent to ALSA driver. But stop button is not clicked in > > > this case) > > > > It's most likely the ALSA PCM core's safety stop; your driver seem to > > have missed snd_pcm_period_elapsed() calls, so the hwptr isn't > > updated, resulting in XRUN. ALSA PCM core checks such XRUN condition > > with the own timer. > > > > > > Takashi > > > > -- > Thanks and Regards > Chakravarthi Pradeep.K > Ph: 91 9980434900 >

1 0

[alsa-devel] Help with TASCAM US-122
by Joseph Spencer 05 Aug '18

05 Aug '18

Hello, I'm on Fedora 28 (Linux localhost.localdomain 4.17.11-200.fc28.x86_64 #1 SMP Mon Jul 30 15:21:09 UTC 2018 x86_64 x86_64 x86_64 GNU/Linux), and I'm trying to use the TASCAM US-122 Audio/MIDI Interface <https://alsa-project.org/main/index.php/Matrix:Vendor-Tascam>. It seems as though the alsa-firmware package is able to detect the USB device just fine; however, after a few moments my entire system freezes (mouse cursor doesn't move etc.). I'm left to forcefully power off. I'm wondering if anyone can point me in the right direction. I'm comfortable with C (haven't done much kernel development), good with shell scripts, not so familiar with alsa. I'd be more than happy to contribute to hopefully fix this for others and (of course) myself. I had journalctl -f up in a terminal when I plugged in the interface and was able to capture the system logs (in the post script of this email). Any help would be appreciated. Thanks, Joe Spencer 602.777.2307 P.S. Here's the relevant system logs of the issue: Aug 04 03:13:50 localhost.localdomain kernel: usb 3-1: new full-speed USB device number 2 using xhci_hcd Aug 04 03:13:50 localhost.localdomain kernel: usb 3-1: New USB device found, idVendor=1604, idProduct=8006, bcdDevice= 1.00 Aug 04 03:13:50 localhost.localdomain kernel: usb 3-1: New USB device strings: Mfr=0, Product=0, SerialNumber=0 Aug 04 03:13:50 localhost.localdomain /lib/udev/tascam_fw[2834]: load /usr/share/alsa/firmware/usx2yloader/us122fw.ihx for 1604/8006/100 to /dev/bus/usb/003/002 Aug 04 03:13:50 localhost.localdomain /lib/udev/tascam_fw[2837]: load /usr/share/alsa/firmware/usx2yloader/us122fw.ihx for 1604/8006/100 to Aug 04 03:13:50 localhost.localdomain kernel: usb 3-1: USB disconnect, device number 2 Aug 04 03:13:52 localhost.localdomain kernel: usb 3-1: new full-speed USB device number 3 using xhci_hcd Aug 04 03:13:52 localhost.localdomain kernel: usb 3-1: New USB device found, idVendor=1604, idProduct=8007, bcdDevice= 0.01 Aug 04 03:13:52 localhost.localdomain kernel: usb 3-1: New USB device strings: Mfr=0, Product=0, SerialNumber=0 Aug 04 03:13:52 localhost.localdomain /lib/udev/tascam_fpga[2859]: calling /usr/bin/usx2yloader for Aug 04 03:13:52 localhost.localdomain /lib/udev/tascam_fpga[2861]: leaving Aug 04 03:13:52 localhost.localdomain kernel: usbcore: registered new interface driver snd-usb-usx2y Aug 04 03:13:52 localhost.localdomain /lib/udev/tascam_fpga[2864]: calling /usr/bin/usx2yloader for *Aug 04 03:13:54 localhost.localdomain /lib/udev/tascam_fpga[2867]: leaving (this was the last log entry before my screen froze, the following logs were obtained after a reboot with my device disconnected)* Aug 04 03:13:58 localhost.localdomain kernel: BUG: unable to handle kernel paging request at 00000000178f90ca Aug 04 03:13:58 localhost.localdomain kernel: PGD 800000040aa8b067 P4D 800000040aa8b067 PUD 4050d8067 PMD 0 Aug 04 03:13:58 localhost.localdomain kernel: Oops: 0000 [#1] SMP PTI Aug 04 03:13:58 localhost.localdomain kernel: Modules linked in: snd_seq_midi snd_seq_midi_event snd_usb_usx2y snd_usbmidi_lib snd_rawmidi ccm rfcomm fuse vboxnetadp(OE) vboxnetflt(OE) vboxdrv> Aug 04 03:13:58 localhost.localdomain kernel: snd_hwdep snd_seq wmi snd_seq_device snd_pcm snd_timer parport_pc parport snd mei_me soundcore mei dell_rbtn rfkill lpc_ich shpchp dell_smo8800 p> Aug 04 03:13:58 localhost.localdomain kernel: CPU: 1 PID: 1125 Comm: Xorg Tainted: G OE 4.17.11-200.fc28.x86_64 #1 Aug 04 03:13:58 localhost.localdomain kernel: Hardware name: Dell Inc. Latitude E6530/, BIOS A17 08/19/2015 Aug 04 03:13:58 localhost.localdomain kernel: RIP: 0010:__kmalloc_track_caller+0xa2/0x230 Aug 04 03:13:58 localhost.localdomain kernel: RSP: 0018:ffffb4c7c22bfab8 EFLAGS: 00010206 Aug 04 03:13:58 localhost.localdomain kernel: RAX: 0000000000000000 RBX: 0000000000000000 RCX: 0000000000000000 Aug 04 03:13:58 localhost.localdomain kernel: RDX: 0000000000004956 RSI: 00000000014000c0 RDI: 0000000000026120 Aug 04 03:13:58 localhost.localdomain kernel: RBP: 00000000014000c0 R08: 00000000178f90ca R09: ffff930a8d803200 Aug 04 03:13:58 localhost.localdomain kernel: R10: ffff930a860b2c00 R11: 0000000000000000 R12: 00000000000000d0 Aug 04 03:13:58 localhost.localdomain kernel: R13: ffffffffc04a7e80 R14: ffff930a8d803200 R15: 00000000178f90ca Aug 04 03:13:58 localhost.localdomain kernel: FS: 00007f11c22baac0(0000) GS:ffff930a9e280000(0000) knlGS:0000000000000000 Aug 04 03:13:58 localhost.localdomain kernel: CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 Aug 04 03:13:58 localhost.localdomain kernel: CR2: 00000000178f90ca CR3: 0000000406ab4004 CR4: 00000000001606e0 Aug 04 03:13:58 localhost.localdomain kernel: Call Trace: Aug 04 03:13:58 localhost.localdomain kernel: kmemdup+0x17/0x40 Aug 04 03:13:58 localhost.localdomain kernel: intel_plane_duplicate_state+0x20/0x50 [i915] Aug 04 03:13:58 localhost.localdomain kernel: drm_atomic_get_plane_state+0x6a/0x110 [drm] Aug 04 03:13:58 localhost.localdomain kernel: drm_atomic_helper_disable_plane+0x36/0xc0 [drm_kms_helper] Aug 04 03:13:58 localhost.localdomain kernel: __setplane_internal+0x1bd/0x260 [drm] Aug 04 03:13:58 localhost.localdomain kernel: ? sock_wfree+0x3a/0x60 Aug 04 03:13:58 localhost.localdomain kernel: ? unix_destruct_scm+0x80/0xa0 Aug 04 03:13:58 localhost.localdomain kernel: drm_mode_cursor_universal+0x1b7/0x1f0 [drm] Aug 04 03:13:58 localhost.localdomain kernel: drm_mode_cursor_common+0x18b/0x1f0 [drm] Aug 04 03:13:58 localhost.localdomain kernel: ? drm_mode_setplane+0x230/0x230 [drm] Aug 04 03:13:58 localhost.localdomain kernel: drm_mode_cursor_ioctl+0x4d/0x70 [drm] Aug 04 03:13:58 localhost.localdomain kernel: drm_ioctl_kernel+0x5b/0xb0 [drm] Aug 04 03:13:58 localhost.localdomain kernel: drm_ioctl+0x1b3/0x370 [drm] Aug 04 03:13:58 localhost.localdomain kernel: ? drm_mode_setplane+0x230/0x230 [drm] Aug 04 03:13:58 localhost.localdomain kernel: ? timerqueue_add+0x52/0x80 Aug 04 03:13:58 localhost.localdomain kernel: ? selinux_file_ioctl+0x161/0x200 Aug 04 03:13:58 localhost.localdomain kernel: do_vfs_ioctl+0xa4/0x610 Aug 04 03:13:58 localhost.localdomain kernel: ksys_ioctl+0x60/0x90 Aug 04 03:13:58 localhost.localdomain kernel: __x64_sys_ioctl+0x16/0x20 Aug 04 03:13:58 localhost.localdomain kernel: do_syscall_64+0x5b/0x160 Aug 04 03:13:58 localhost.localdomain kernel: entry_SYSCALL_64_after_hwframe+0x44/0xa9 Aug 04 03:13:58 localhost.localdomain kernel: RIP: 0033:0x7f11bf53dec7 Aug 04 03:13:58 localhost.localdomain kernel: RSP: 002b:00007fffa0ee58d8 EFLAGS: 00003246 ORIG_RAX: 0000000000000010 Aug 04 03:13:58 localhost.localdomain kernel: RAX: ffffffffffffffda RBX: 00000000013cc100 RCX: 00007f11bf53dec7 Aug 04 03:13:58 localhost.localdomain kernel: RDX: 00007fffa0ee5910 RSI: 00000000c01c64a3 RDI: 0000000000000010 Aug 04 03:13:58 localhost.localdomain kernel: RBP: 00007fffa0ee5910 R08: 00007f11c2136000 R09: 0000000000000001 Aug 04 03:13:58 localhost.localdomain kernel: R10: 0000000000000780 R11: 0000000000003246 R12: 00000000c01c64a3 Aug 04 03:13:58 localhost.localdomain kernel: R13: 0000000000000010 R14: 000000000124c0d0 R15: 00000000000001cd Aug 04 03:13:58 localhost.localdomain kernel: Code: 50 08 65 4c 03 05 e7 26 da 5e 49 83 78 10 00 4d 8b 38 0f 84 22 01 00 00 4d 85 ff 0f 84 19 01 00 00 45 8b 41 20 49 8b 39 4d 01 f8 <4d> 8b 10 > Aug 04 03:13:58 localhost.localdomain kernel: RIP: __kmalloc_track_caller+0xa2/0x230 RSP: ffffb4c7c22bfab8 Aug 04 03:13:58 localhost.localdomain kernel: CR2: 00000000178f90ca Aug 04 03:13:58 localhost.localdomain kernel: ---[ end trace 59cb442553c1594d ]--- Aug 04 03:13:59 localhost.localdomain abrt-dump-journal-oops[1205]: abrt-dump-journal-oops: Found oopses: 1 Aug 04 03:13:59 localhost.localdomain abrt-dump-journal-oops[1205]: abrt-dump-journal-oops: Creating problem directories Aug 04 03:13:59 localhost.localdomain kernel: BUG: unable to handle kernel paging request at 00000000178f90ca Aug 04 03:13:59 localhost.localdomain kernel: PGD 0 P4D 0 Aug 04 03:13:59 localhost.localdomain kernel: Oops: 0000 [#2] SMP PTI Aug 04 03:13:59 localhost.localdomain kernel: Modules linked in: snd_seq_midi snd_seq_midi_event snd_usb_usx2y snd_usbmidi_lib snd_rawmidi ccm rfcomm fuse vboxnetadp(OE) vboxnetflt(OE) vboxdrv> Aug 04 03:13:59 localhost.localdomain kernel: snd_hwdep snd_seq wmi snd_seq_device snd_pcm snd_timer parport_pc parport snd mei_me soundcore mei dell_rbtn rfkill lpc_ich shpchp dell_smo8800 p> Aug 04 03:13:59 localhost.localdomain kernel: CPU: 1 PID: 1205 Comm: abrt-dump-journ Tainted: G D OE 4.17.11-200.fc28.x86_64 #1 Aug 04 03:13:59 localhost.localdomain kernel: Hardware name: Dell Inc. Latitude E6530/, BIOS A17 08/19/2015 Aug 04 03:13:59 localhost.localdomain kernel: RIP: 0010:kmem_cache_alloc_trace+0x8d/0x1d0 Aug 04 03:13:59 localhost.localdomain kernel: RSP: 0018:ffffb4c7c25dfc60 EFLAGS: 00010206 Aug 04 03:13:59 localhost.localdomain kernel: RAX: 0000000000000000 RBX: 00000000178f90ca RCX: 0000000000000002 Aug 04 03:13:59 localhost.localdomain kernel: RDX: 0000000000004956 RSI: 00000000014080c0 RDI: 0000000000026120 Aug 04 03:13:59 localhost.localdomain kernel: RBP: ffff930a8d803200 R08: ffff930a9e2a6120 R09: 0000000000000000 Aug 04 03:13:59 localhost.localdomain kernel: R10: 0000000000000000 R11: e7825477348aeba9 R12: 00000000014080c0 Aug 04 03:13:59 localhost.localdomain kernel: R13: 00000000000000f8 R14: ffff930a8d803200 R15: 00000000178f90ca Aug 04 03:13:59 localhost.localdomain kernel: FS: 00007f0fd1a99ac0(0000) GS:ffff930a9e280000(0000) knlGS:0000000000000000 Aug 04 03:13:59 localhost.localdomain kernel: CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 Aug 04 03:13:59 localhost.localdomain kernel: CR2: 00000000178f90ca CR3: 00000004082c2006 CR4: 00000000001606e0 Aug 04 03:13:59 localhost.localdomain kernel: Call Trace: Aug 04 03:13:59 localhost.localdomain kernel: ? full_proxy_open+0xc0/0x1d0 Aug 04 03:13:59 localhost.localdomain kernel: full_proxy_open+0xc0/0x1d0 Aug 04 03:13:59 localhost.localdomain kernel: ? debugfs_create_devm_seqfile+0x80/0x80 Aug 04 03:13:59 localhost.localdomain kernel: do_dentry_open+0x1af/0x2e0 Aug 04 03:13:59 localhost.localdomain kernel: path_openat+0x46e/0x15f0 Aug 04 03:13:59 localhost.localdomain kernel: ? 0xffffffffc02a8000 Aug 04 03:13:59 localhost.localdomain kernel: ? seq_puts+0x3a/0x50 Aug 04 03:13:59 localhost.localdomain kernel: ? __kmalloc_node+0x208/0x2d0 Aug 04 03:13:59 localhost.localdomain kernel: ? seq_read+0x2de/0x460 Aug 04 03:13:59 localhost.localdomain kernel: do_filp_open+0x93/0x100 Aug 04 03:13:59 localhost.localdomain kernel: ? __check_object_size+0x9c/0x171 Aug 04 03:13:59 localhost.localdomain kernel: do_sys_open+0x186/0x210 Aug 04 03:13:59 localhost.localdomain kernel: do_syscall_64+0x5b/0x160 Aug 04 03:13:59 localhost.localdomain kernel: entry_SYSCALL_64_after_hwframe+0x44/0xa9 Aug 04 03:13:59 localhost.localdomain kernel: RIP: 0033:0x7f0fd03b0c9e Aug 04 03:13:59 localhost.localdomain kernel: RSP: 002b:00007ffd7905f8b0 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 Aug 04 03:13:59 localhost.localdomain kernel: RAX: ffffffffffffffda RBX: 0000000000000001 RCX: 00007f0fd03b0c9e Aug 04 03:13:59 localhost.localdomain kernel: RDX: 0000000000000000 RSI: 000055cf0bd89780 RDI: 00000000ffffff9c Aug 04 03:13:59 localhost.localdomain kernel: RBP: 0000000000000000 R08: 00007f0fd067fc40 R09: 0000000000000007 Aug 04 03:13:59 localhost.localdomain kernel: R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000000 Aug 04 03:13:59 localhost.localdomain kernel: R13: 000055cf0d8c5700 R14: 0000000000000003 R15: 000055cf0d936d50 Aug 04 03:13:59 localhost.localdomain kernel: Code: 50 08 65 4c 03 05 7c 5f da 5e 49 83 78 10 00 4d 8b 38 0f 84 10 01 00 00 4d 85 ff 0f 84 07 01 00 00 41 8b 5e 20 49 8b 3e 4c 01 fb <48> 33 1b > Aug 04 03:13:59 localhost.localdomain kernel: RIP: kmem_cache_alloc_trace+0x8d/0x1d0 RSP: ffffb4c7c25dfc60 Aug 04 03:13:59 localhost.localdomain kernel: CR2: 00000000178f90ca Aug 04 03:13:59 localhost.localdomain kernel: ---[ end trace 59cb442553c1594e ]--- Aug 04 03:13:59 localhost.localdomain audit[1]: SERVICE_STOP pid=1 uid=0 auid=4294967295 ses=4294967295 subj=system_u:system_r:init_t:s0 msg='unit=abrt-oops comm="systemd" exe="/usr/lib/system> Aug 04 03:13:59 localhost.localdomain systemd[1]: abrt-oops.service: Main process exited, code=killed, status=9/KILL Aug 04 03:13:59 localhost.localdomain systemd[1]: abrt-oops.service: Failed with result 'signal'. Aug 04 03:15:24 localhost.localdomain chronyd[1036]: Selected source 208.75.89.4 Aug 04 03:15:56 localhost.localdomain systemd-logind[1024]: Power key pressed. Aug 04 03:15:56 localhost.localdomain systemd-logind[1024]: Suspending... Aug 04 03:15:56 localhost.localdomain NetworkManager[1054]: <info> [1533377756.3501] manager: sleep: sleep requested (sleeping: no enabled: yes) Aug 04 03:15:56 localhost.localdomain NetworkManager[1054]: <info> [1533377756.3506] manager: NetworkManager state is now ASLEEP Aug 04 03:15:56 localhost.localdomain NetworkManager[1054]: <info> [1533377756.3510] device (wlp2s0): state change: activated -> deactivating (reason 'sleeping', sys-iface-state: 'managed') Aug 04 03:15:56 localhost.localdomain NetworkManager[1054]: <info> [1533377756.3714] device (wlp2s0): state change: deactivating -> disconnected (reason 'sleeping', sys-iface-state: 'managed') Aug 04 03:15:56 localhost.localdomain dbus-daemon[1046]: [system] Activating via systemd: service name='org.freedesktop.nm_dispatcher' unit='dbus-org.freedesktop.nm-dispatcher.service' request> Aug 04 03:15:56 localhost.localdomain avahi-daemon[1030]: Withdrawing address record for fe80::f3ca:ecd8:3b4f:e72a on wlp2s0. Aug 04 03:15:56 localhost.localdomain avahi-daemon[1030]: Leaving mDNS multicast group on interface wlp2s0.IPv6 with address fe80::f3ca:ecd8:3b4f:e72a. Aug 04 03:15:56 localhost.localdomain avahi-daemon[1030]: Interface wlp2s0.IPv6 no longer relevant for mDNS. Aug 04 03:15:56 localhost.localdomain systemd[1]: Starting Network Manager Script Dispatcher Service... Aug 04 03:15:56 localhost.localdomain kernel: BUG: unable to handle kernel paging request at 00000000178f90ca Aug 04 03:15:56 localhost.localdomain kernel: PGD 0 P4D 0 Aug 04 03:15:56 localhost.localdomain kernel: Oops: 0000 [#3] SMP PTI Aug 04 03:15:56 localhost.localdomain kernel: Modules linked in: snd_seq_midi snd_seq_midi_event snd_usb_usx2y snd_usbmidi_lib snd_rawmidi ccm rfcomm fuse vboxnetadp(OE) vboxnetflt(OE) vboxdrv> Aug 04 03:15:56 localhost.localdomain kernel: snd_hwdep snd_seq wmi snd_seq_device snd_pcm snd_timer parport_pc parport snd mei_me soundcore mei dell_rbtn rfkill lpc_ich shpchp dell_smo8800 p> Aug 04 03:15:56 localhost.localdomain kernel: CPU: 1 PID: 2878 Comm: (spatcher) Tainted: G D OE 4.17.11-200.fc28.x86_64 #1 Aug 04 03:15:56 localhost.localdomain kernel: Hardware name: Dell Inc. Latitude E6530/, BIOS A17 08/19/2015 Aug 04 03:15:56 localhost.localdomain kernel: RIP: 0010:kmem_cache_alloc_trace+0x8d/0x1d0 Aug 04 03:15:56 localhost.localdomain kernel: RSP: 0018:ffffb4c7c261fe68 EFLAGS: 00010206 Aug 04 03:15:56 localhost.localdomain kernel: RAX: 0000000000000000 RBX: 00000000178f90ca RCX: 0000000000000000 Aug 04 03:15:56 localhost.localdomain kernel: RDX: 0000000000004956 RSI: 00000000014080c0 RDI: 0000000000026120 Aug 04 03:15:56 localhost.localdomain kernel: RBP: ffff930a8d803200 R08: ffff930a9e2a6120 R09: 0000555a9cca5d60 Aug 04 03:15:56 localhost.localdomain kernel: R10: fefefefefefefeff R11: 0000000000000fe0 R12: 00000000014080c0 Aug 04 03:15:56 localhost.localdomain kernel: R13: 0000000000000100 R14: ffff930a8d803200 R15: 00000000178f90ca Aug 04 03:15:56 localhost.localdomain kernel: FS: 00007f413dfaae40(0000) GS:ffff930a9e280000(0000) knlGS:0000000000000000 Aug 04 03:15:56 localhost.localdomain kernel: CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 Aug 04 03:15:56 localhost.localdomain kernel: CR2: 00000000178f90ca CR3: 00000003c9d3e001 CR4: 00000000001606e0 Aug 04 03:15:56 localhost.localdomain kernel: Call Trace: Aug 04 03:15:56 localhost.localdomain kernel: ? do_execveat_common.isra.37+0xeb/0x790 Aug 04 03:15:56 localhost.localdomain kernel: do_execveat_common.isra.37+0xeb/0x790 Aug 04 03:15:56 localhost.localdomain kernel: ? strncpy_from_user+0x4a/0x170 Aug 04 03:15:56 localhost.localdomain kernel: __x64_sys_execve+0x32/0x40 Aug 04 03:15:56 localhost.localdomain kernel: do_syscall_64+0x5b/0x160 Aug 04 03:15:56 localhost.localdomain kernel: entry_SYSCALL_64_after_hwframe+0x44/0xa9 Aug 04 03:15:56 localhost.localdomain kernel: RIP: 0033:0x7f413bc5dde7 Aug 04 03:15:56 localhost.localdomain kernel: RSP: 002b:00007ffc55bb2748 EFLAGS: 00000246 ORIG_RAX: 000000000000003b Aug 04 03:15:56 localhost.localdomain kernel: RAX: ffffffffffffffda RBX: 0000000000000000 RCX: 00007f413bc5dde7 Aug 04 03:15:56 localhost.localdomain kernel: RDX: 0000555a9cca5d60 RSI: 0000555a9cd8e4c0 RDI: 0000555a9cd92bc0 Aug 04 03:15:56 localhost.localdomain kernel: RBP: 00007ffc55bb29b0 R08: 000000000000fefe R09: 0000000000000000 Aug 04 03:15:56 localhost.localdomain kernel: R10: 0000555a9cd4a900 R11: 0000000000000246 R12: 0000555a9cd8e4c0 Aug 04 03:15:56 localhost.localdomain kernel: R13: 0000555a9cd933a0 R14: 0000555a9cd93840 R15: 0000000000000000 Aug 04 03:15:56 localhost.localdomain kernel: Code: 50 08 65 4c 03 05 7c 5f da 5e 49 83 78 10 00 4d 8b 38 0f 84 10 01 00 00 4d 85 ff 0f 84 07 01 00 00 41 8b 5e 20 49 8b 3e 4c 01 fb <48> 33 1b > Aug 04 03:15:56 localhost.localdomain kernel: RIP: kmem_cache_alloc_trace+0x8d/0x1d0 RSP: ffffb4c7c261fe68 Aug 04 03:15:56 localhost.localdomain kernel: CR2: 00000000178f90ca Aug 04 03:15:56 localhost.localdomain kernel: ---[ end trace 59cb442553c1594f ]--- Aug 04 03:15:56 localhost.localdomain kernel: wlp2s0: deauthenticating from 84:1b:5e:03:b3:bd by local choice (Reason: 3=DEAUTH_LEAVING) Aug 04 03:15:56 localhost.localdomain kernel: IPv6: ADDRCONF(NETDEV_UP): wlp2s0: link is not ready Aug 04 03:15:56 localhost.localdomain kernel: BUG: unable to handle kernel paging request at 00000000178f90ca -- Reboot -- Aug 04 03:16:21 localhost.localdomain kernel: microcode: microcode updated early to revision 0x1f, date = 2018-02-07 Aug 04 03:16:21 localhost.localdomain kernel: Linux version 4.17.11-200.fc28.x86_64 (mockbuild(a)bkernel01.phx2.fedoraproject.org) (gcc version 8.1.1 20180712 (Red Hat 8.1.1-5) (GCC)) #1 SMP Mon> Aug 04 03:16:21 localhost.localdomain kernel: Command line: BOOT_IMAGE=/vmlinuz-4.17.11-200.fc28.x86_64 root=/dev/mapper/fedora-root ro rd.luks.uuid=luks-9c1123dd-5aee-41e5-b3c4-b91249d5ed64 r> Aug 04 03:16:21 localhost.localdomain kernel: x86/fpu: Supporting XSAVE feature 0x001: 'x87 floating point registers' Aug 04 03:16:21 localhost.localdomain kernel: x86/fpu: Supporting XSAVE feature 0x002: 'SSE registers' Aug 04 03:16:21 localhost.localdomain kernel: x86/fpu: Supporting XSAVE feature 0x004: 'AVX registers' Aug 04 03:16:21 localhost.localdomain kernel: x86/fpu: xstate_offset[2]: 576, xstate_sizes[2]: 256 Aug 04 03:16:21 localhost.localdomain kernel: x86/fpu: Enabled xstate features 0x7, context size is 832 bytes, using 'standard' format. Aug 04 03:16:21 localhost.localdomain kernel: e820: BIOS-provided physical RAM map: Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x0000000000000000-0x000000000009d3ff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x000000000009d400-0x000000000009ffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000000e0000-0x00000000000fffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x0000000000100000-0x000000001fffffff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x0000000020000000-0x00000000201fffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x0000000020200000-0x0000000040003fff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x0000000040004000-0x0000000040004fff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x0000000040005000-0x00000000d7f09fff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000d7f0a000-0x00000000d7ffffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000d8000000-0x00000000d8752fff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000d8753000-0x00000000d87fffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000d8800000-0x00000000d8fb2fff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000d8fb3000-0x00000000d8ffffff] ACPI data Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000d9000000-0x00000000da6fbfff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000da6fc000-0x00000000da7fffff] ACPI NVS Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000da800000-0x00000000dbb1bfff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000dbb1c000-0x00000000dc7a6fff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000dc7a7000-0x00000000dc7e9fff] ACPI NVS Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000dc7ea000-0x00000000dcab2fff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000dcab3000-0x00000000dcfeffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000dcff0000-0x00000000dcffffff] usable Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000dd800000-0x00000000df9fffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000f8000000-0x00000000fbffffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000fec00000-0x00000000fec00fff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000fed00000-0x00000000fed03fff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000fed1c000-0x00000000fed1ffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000fee00000-0x00000000fee00fff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x00000000ff000000-0x00000000ffffffff] reserved Aug 04 03:16:21 localhost.localdomain kernel: BIOS-e820: [mem 0x0000000100000000-0x000000041e5fffff] usable Aug 04 03:16:21 localhost.localdomain kernel: NX (Execute Disable) protection: active Aug 04 03:16:21 localhost.localdomain kernel: random: fast init done Aug 04 03:16:21 localhost.localdomain kernel: SMBIOS 2.7 present. Aug 04 03:16:21 localhost.localdomain kernel: DMI: Dell Inc. Latitude E6530/, BIOS A17 08/19/2015 Aug 04 03:16:21 localhost.localdomain kernel: e820: update [mem 0x00000000-0x00000fff] usable ==> reserved Aug 04 03:16:21 localhost.localdomain kernel: e820: remove [mem 0x000a0000-0x000fffff] usable Aug 04 03:16:21 localhost.localdomain kernel: e820: last_pfn = 0x41e600 max_arch_pfn = 0x400000000 Aug 04 03:16:21 localhost.localdomain kernel: MTRR default type: uncachable Aug 04 03:16:21 localhost.localdomain kernel: MTRR fixed ranges enabled: Aug 04 03:16:21 localhost.localdomain kernel: 00000-9FFFF write-back

2 1

[alsa-devel] [PATCH] alsabat: Allow custom sample format for round trip latency test
by net147＠gmail.com 05 Aug '18

05 Aug '18

From: Jonathan Liu <net147(a)gmail.com> Setting the format to BAT_PCM_FORMAT_S16_LE in the round trip latency test initialization is redundant as it is already set by default to BAT_PCM_FORMAT_S16_LE unless a sample format is specified on the command line. Signed-off-by: Jonathan Liu <net147(a)gmail.com> --- bat/latencytest.c | 1 - 1 file changed, 1 deletion(-) diff --git a/bat/latencytest.c b/bat/latencytest.c index fae191c..ec3abe2 100644 --- a/bat/latencytest.c +++ b/bat/latencytest.c @@ -178,7 +178,6 @@ void roundtrip_latency_init(struct bat *bat) bat->latency.is_playing = false; bat->latency.error = 0; bat->latency.xrun_error = false; - bat->format = BAT_PCM_FORMAT_S16_LE; bat->frames = LATENCY_TEST_TIME_LIMIT * bat->rate; bat->periods_played = 0; } -- 2.18.0

2 1

[alsa-devel] [PATCH] ALSA: echoaudio: Mark expected switch fall-throughs
by Gustavo A. R. Silva 05 Aug '18

05 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115156 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/echoaudio/echoaudio.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 358ef7d..907cf1a 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -713,6 +713,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) break; case SNDRV_PCM_FORMAT_S32_BE: format.data_are_bigendian = 1; + /* fall through */ case SNDRV_PCM_FORMAT_S32_LE: format.bits_per_sample = 32; break; @@ -764,6 +765,7 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) pipe->last_counter = 0; pipe->position = 0; *pipe->dma_counter = 0; + /* fall through */ case PIPE_STATE_PAUSED: pipe->state = PIPE_STATE_STARTED; break; -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: emu10k1: Mark expected switch fall-throughs
by Gustavo A. R. Silva 05 Aug '18

05 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Notice that in this particular case, I replaced the code comment with a proper "fall through" annotation, which is what GCC is expecting to find. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/emu10k1/emupcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 26f6eda..9f2b609 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -778,7 +778,7 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); /* do we need this? */ snd_emu10k1_playback_invalidate_cache(emu, 0, epcm->voices[0]); - /* follow thru */ + /* fall through */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: if (cmd == SNDRV_PCM_TRIGGER_PAUSE_RELEASE) @@ -928,7 +928,7 @@ static int snd_emu10k1_efx_playback_trigger(struct snd_pcm_substream *substream, } snd_emu10k1_playback_invalidate_cache(emu, 1, epcm->extra); - /* follow thru */ + /* fall through */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: snd_emu10k1_playback_prepare_voice(emu, epcm->extra, 1, 1, NULL); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: mixart: Mark expected switch fall-through
by Gustavo A. R. Silva 05 Aug '18

05 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/pci/mixart/mixart.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index a74f1ad..9cd297a 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -182,6 +182,7 @@ static int mixart_set_clock(struct mixart_mgr *mgr, case PIPE_RUNNING: if(rate != 0) break; + /* fall through */ default: if(rate == 0) return 0; /* nothing to do */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Add mute LED quirk for HP Spectre x360
by Alexandru Gagniuc 05 Aug '18

05 Aug '18

This device has the same issues as the HP x360 wrt the MUTE LED and the front speakers not working. This patch fixes the MUTE LED issue, but doesn't touch the HDA verbs. The fix for the x360 does not work on the Spectre. Signed-off-by: Alexandru Gagniuc <mr.nuke.me(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f6af3e1c2b93..d14b05f68d6d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6530,6 +6530,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.17.1

2 1

[alsa-devel] ALSA driver rmmod
by Chakravarthi Pradeep 05 Aug '18

05 Aug '18

Dear ALSA Team, I have taken reference alsa driver from https://alsa-project.org/main/index.php/Minivosc and driver code path from https://github.com/stadaki/alsa-minivosc-src . If I, do a rmmod of aforesaid driver using below command "rmmod snd-minivosc" Output of the above command is "rmmod: ERROR: Module snd_minivosc is in use" Below is output from lsmod | grep snd-minivosc "snd_minivosc 16384 1 snd_pcm 98304 6 snd_hda_intel,snd_hda_codec,snd_usb_audio,snd_minivosc,snd_hda_core,snd_hda_codec_hdmi snd 81920 26 snd_hda_intel,snd_hwdep,snd_seq,snd_hda_codec,snd_usb_audio,snd_minivosc,snd_timer,snd_rawmidi,snd_hda_codec_hdmi,snd_hda_codec_generic,snd_usbmidi_lib,snd_seq_device,snd_hda_codec_realtek,snd_pcm" Is there is any way to remove alsa driver from kernel without restarting PC? Kindly let me know how to remove alsa driver safely from linux kernel and then reinsert it. Thanks and Regards Chakravarthi Pradeep.K Ph: 91 9980434900

1 0

Re: [alsa-devel] CS8409 Macbook Pro 2016 2017
by Takashi Iwai 04 Aug '18

04 Aug '18

On Sat, 04 Aug 2018 17:44:59 +0200, David Ulricht wrote: > > I used the following script running as root to verify the 8 GPIOs while > playing long youtube video, > I have running pulseaudio meanwhile and not restarting it, i think its okay > like that people on the ubuntu forums say that after using > hda-verb and setting a gpio active immediately sound appears, so i guess > there is no need to restart pulseaudio. > I have done the same test with headphones in. > No sound in both scenarios, no input no output detected (i'm watching > pavucontrol for input detection). This codec is fairly unique and doesn't provide the standard jack detection on each pin. Maybe it's via either GPIO or any other method. You can try to read GPIO pins (set as input) to see whether any of them corresponds to the jack detection, for example. Takashi > > #!/usr/bin/env python3 > import os,time > for x in range(0,256): > mhex=("0x%0.2X" % x) > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK "+mhex) > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION "+mhex) > os.system("hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA "+mhex) > time.sleep(3) > > Meanwhile script is running I'm watching /proc/asound/card0/codec#0 and > IO[0-7] changes its configuration, > in comparison Joel Krahemann in his scripts is changing the hex right after > /dev/snd/hwC0D0 (0x01) e.g. i dont think he is modifiying the GPIOs > correctly. > > I found the following post: > https://forum.manjaro.org/t/sound-not-working-on-a-2017-imac-27-5k-retina/4… > According to https://bugzilla.kernel.org/show_bug.cgi?id=195671 iMac27 has > a similar soundcard. > How can i setup with command line or any other utility: > """select “off” in the top menu that propose to setup the Digital > Surround""" > this is some setting in xfce4-mixer package which is not available for > ubuntu since 2 LTS versions behind. I tried current Manjaro and couldn't > find this setting in audio mixer of xfce4 either. > > On Sat, Aug 4, 2018 at 9:48 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 04 Aug 2018 04:07:43 +0200, > > David Ulricht wrote: > > > > > > Hello, > > > > > > Macbook pro models containing CS8409 for sure are: > > > MBP131 MBP141.I own MBP 14,1 e.g. 2017 model. > > > > > > I'm attaching working sound configuration from Windows 10 registry (note > > > that the cs420x might need to be ignored or might be important i don't > > > really know). I believe what is really used is in the PinConfigOverride > > > section. > > > > Comparing between BIOS default (init_pin_cfg in alsa-info.sh output) > > and your override (user_pin_cfg), there aren't so many changes. > > All changes are pretty minor, and I guess it won't influence on the > > driver behavior. > > > > > Interesting thing to note is that sound in MacOS is much louder than > > > Bootcamp Windows, would be nice to hear how loud is on linux. > > > > This is possibly with some vendor-specific GPIO or COEF verbs. > > Or it's some direct I2C control. The INI file mentions it. > > And that's above HD-audio driver's responsibility. > > > > > I did convert PinConfigOverride HEX strings to [pincfg] format in the > > > attached hda-jack-retask.fw thanks to Takashi's guidance. > > > > > > I'm attaching also alsamixer ASCII. Only PCM, no Master ? > > > > It's because the codec chip has no output amplifier at all. > > So there can be neither output volume nor mute control on this chip. > > > > > What I have tried is execute: > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x@@ > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION 0x@@ > > > hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x@@ > > > putting @@ from 0x00 to 0x50 > > > > There are eight GPIOs, so you'd need to test each bit, i.e. 0x01, > > 0x02, 0x04, 0x08, ... 0x80. > > > > And how is the current situation? You can't play *and* record > > anything from any inputs / outputs? > > > > > > Takashi > > > [2 <text/html; UTF-8 (quoted-printable)>] >

1 0

[alsa-devel] CS8409 Macbook Pro 2016 2017
by David Ulricht 04 Aug '18

04 Aug '18

Hello, Macbook pro models containing CS8409 for sure are: MBP131 MBP141.I own MBP 14,1 e.g. 2017 model. I'm attaching working sound configuration from Windows 10 registry (note that the cs420x might need to be ignored or might be important i don't really know). I believe what is really used is in the PinConfigOverride section. Interesting thing to note is that sound in MacOS is much louder than Bootcamp Windows, would be nice to hear how loud is on linux. I did convert PinConfigOverride HEX strings to [pincfg] format in the attached hda-jack-retask.fw thanks to Takashi's guidance. I'm attaching also alsamixer ASCII. Only PCM, no Master ? What I have tried is execute: hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_MASK 0x@@ hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DIRECTION 0x@@ hda-verb /dev/snd/hwC0D0 0x01 SET_GPIO_DATA 0x@@ putting @@ from 0x00 to 0x50 Please advise.

2 2

[alsa-devel] [PATCH] ALSA: seq_oss: Mark expected switch fall-through
by Gustavo A. R. Silva 04 Aug '18

04 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Warning level 2 was used: -Wimplicit-fallthrough=2 Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/seq/oss/seq_oss_timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index 4f24ea9..ba127c2 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -92,7 +92,7 @@ snd_seq_oss_process_timer_event(struct seq_oss_timer *rec, union evrec *ev) case TMR_WAIT_REL: parm += rec->cur_tick; rec->realtime = 0; - /* continue to next */ + /* fall through and continue to next */ case TMR_WAIT_ABS: if (parm == 0) { rec->realtime = 1; -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: seq: Mark expected switch fall-through
by Gustavo A. R. Silva 04 Aug '18

04 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Notice that in this particular case, I replaced the code comment with a proper "fall through" annotation, which is what GCC is expecting to find. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/seq/seq_midi_emul.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_midi_emul.c b/sound/core/seq/seq_midi_emul.c index f9f2133..c1975dd 100644 --- a/sound/core/seq/seq_midi_emul.c +++ b/sound/core/seq/seq_midi_emul.c @@ -318,7 +318,7 @@ do_control(struct snd_midi_op *ops, void *drv, struct snd_midi_channel_set *chse break; case MIDI_CTL_MSB_DATA_ENTRY: chan->control[MIDI_CTL_LSB_DATA_ENTRY] = 0; - /* go through here */ + /* fall through */ case MIDI_CTL_LSB_DATA_ENTRY: if (chan->param_type == SNDRV_MIDI_PARAM_TYPE_REGISTERED) rpn(ops, drv, chan, chset); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: smd845: fix memory leak
by Gustavo A. R. Silva 03 Aug '18

03 Aug '18

In case memory resources for *card* were allocated, release them before return. Addresses-Coverity-ID: 1472244 ("Resource leak") Fixes: 6b1687bf76ef ("ASoC: qcom: add sdm845 sound card support") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- NOTE: Notice that there are many places in which the return values of functions snd_soc_dai_set_sysclk() and snd_soc_dai_set_fmt() are not being checked. I wonder if this is intentional. Otherwise, I can send a patch for that. Please, confirm. Thanks! sound/soc/qcom/sdm845.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index be0cb11..c1adb77 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -222,8 +222,10 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) /* Allocate the private data */ data = kzalloc(sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; + if (!data) { + ret = -ENOMEM; + goto data_alloc_fail; + } card->dev = dev; dev_set_drvdata(dev, card); @@ -248,6 +250,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) kfree(card->dai_link); parse_dt_fail: kfree(data); +data_alloc_fail: kfree(card); return ret; } -- 2.7.4

2 1

[alsa-devel] [PATCH 0/7] ASoC: Mark expected switch fall-throughs
by Gustavo A. R. Silva 03 Aug '18

03 Aug '18

Hi all, In preparation to enabling -Wimplicit-fallthrough, this patchset aims to add some annotations in order to mark switch cases where we are expecting to fall through. Thanks Gustavo A. R. Silva (7): ASoC: davinci-i2s: mark expected switch fall-through ASoC: fsl_esai: Mark expected switch fall-through ASoC: Intel: skl-pcm: Mark expected switch fall-through ASoC: omap-dmic: Mark expected switch fall-throughs ASoC: omap-mcpdm: MArk expected switch fall-throughs ASoC: samsung: i2s: Mark expected switch fall-through ASoC: core: mark expected switch fall-through sound/soc/davinci/davinci-i2s.c | 1 + sound/soc/fsl/fsl_esai.c | 1 + sound/soc/intel/skylake/skl-pcm.c | 1 + sound/soc/omap/omap-dmic.c | 2 ++ sound/soc/omap/omap-mcpdm.c | 4 ++++ sound/soc/samsung/i2s.c | 1 + sound/soc/soc-core.c | 1 + 7 files changed, 11 insertions(+) -- 2.7.4

2 13

[alsa-devel] [PATCH 0/6] ALSA: Remove empty init and exit
by Takashi Iwai 03 Aug '18

03 Aug '18

Hi, here is a patchset of trivial cleanups I noticed while working on the recent sequencer bug. There are a bunch of driver codes that have empty module init and exit entries, and we can remove them gracefully. Takashi === Takashi Iwai (6): ALSA: compress: Remove empty init and exit ALSA: drivers: Remove empty init and exit ALSA: isa: Remove empty init and exit ALSA: i2c: Remove empty init and exit ALSA: pci: Remove empty init and exit ALSA: synth: Remove empty init and exit sound/core/compress_offload.c | 12 ------------ sound/drivers/mpu401/mpu401_uart.c | 16 ---------------- sound/drivers/opl3/opl3_lib.c | 16 ---------------- sound/drivers/opl4/opl4_lib.c | 12 ------------ sound/drivers/vx/vx_core.c | 15 --------------- sound/i2c/cs8427.c | 12 ------------ sound/i2c/i2c.c | 13 ------------- sound/i2c/other/ak4xxx-adda.c | 12 ------------ sound/i2c/tea6330t.c | 16 ---------------- sound/isa/es1688/es1688_lib.c | 16 ---------------- sound/isa/gus/gus_main.c | 16 ---------------- sound/isa/sb/sb16_csp.c | 16 ---------------- sound/isa/sb/sb16_main.c | 16 ---------------- sound/isa/sb/sb8_main.c | 16 ---------------- sound/isa/sb/sb_common.c | 16 ---------------- sound/isa/wss/wss_lib.c | 16 ---------------- sound/pci/ac97/ac97_codec.c | 16 ---------------- sound/pci/ice1712/ak4xxx.c | 12 ------------ sound/synth/emux/emux.c | 17 ----------------- sound/synth/util_mem.c | 16 ---------------- 20 files changed, 297 deletions(-) -- 2.18.0

1 6

[alsa-devel] [PATCH] [ALSA] sound/isa/gus: fix spelling mistake "acumulator" -> "accumulator"
by Colin King 03 Aug '18

03 Aug '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial spelling mistake fix in debug message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/isa/gus/gus_io.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c index ca79878d8d8c..2fd32ef22c30 100644 --- a/sound/isa/gus/gus_io.c +++ b/sound/isa/gus/gus_io.c @@ -461,7 +461,7 @@ void snd_gf1_print_voice_registers(struct snd_gus_card * gus) printk(KERN_INFO " -%i- GFA1 effect address = 0x%x\n", voice, snd_gf1_i_read_addr(gus, 0x11, ctrl & 4)); printk(KERN_INFO " -%i- GFA1 effect volume = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x16)); printk(KERN_INFO " -%i- GFA1 effect volume final = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x1d)); - printk(KERN_INFO " -%i- GFA1 effect acumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14)); + printk(KERN_INFO " -%i- GFA1 effect accumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14)); } if (mode & 0x20) { printk(KERN_INFO " -%i- GFA1 left offset = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x13), snd_gf1_i_read16(gus, 0x13) >> 4); -- 2.17.1

2 1

[alsa-devel] [PATCH 0/4] ALSA: Mark expected switch fall-throughs
by Gustavo A. R. Silva 03 Aug '18

03 Aug '18

Hi all, In preparation to enabling -Wimplicit-fallthrough, this patchset aims to add some annotations in order to mark switch cases where we are expecting to fall through. Thanks Gustavo A. R. Silva (4): ALSA: galaxy: Mark expected switch fall-through ALSA: opti92x: mark expected switch fall-through ALSA: opti9xx: mark expected switch fall-through ALSA: es18xx: mark expected switch fall-through sound/isa/es18xx.c | 1 + sound/isa/galaxy/galaxy.c | 3 +++ sound/isa/opti9xx/miro.c | 5 +++++ sound/isa/opti9xx/opti92x-ad1848.c | 3 +++ 4 files changed, 12 insertions(+) -- 2.7.4

2 5

[alsa-devel] [PATCH] ASoC: Intel: bxt: Use refcap device for mono recording
by Yong Zhi 03 Aug '18

03 Aug '18

The refcap capture device supports mono recording only, this patch adds the channel constraints, also change stream_name to "Wake on Voice". Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 16 +++++++++++++++- 1 file changed, 15 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index be6e4b4..097a089 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -319,6 +319,15 @@ static const unsigned int rates_16000[] = { 16000, }; +static const unsigned int ch_mono[] = { + 1, +}; + +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + static const struct snd_pcm_hw_constraint_list constraints_16000 = { .count = ARRAY_SIZE(rates_16000), .list = rates_16000, @@ -326,6 +335,11 @@ static const struct snd_pcm_hw_constraint_list constraints_16000 = { static int broxton_refcap_startup(struct snd_pcm_substream *substream) { + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_16000); @@ -372,7 +386,7 @@ static struct snd_soc_dai_link broxton_dais[] = { [BXT_DPCM_AUDIO_REF_CP] = { .name = "Bxt Audio Reference cap", - .stream_name = "Refcap", + .stream_name = "Wake on Voice", .cpu_dai_name = "Reference Pin", .codec_name = "snd-soc-dummy", .codec_dai_name = "snd-soc-dummy-dai", -- 2.7.4

4 5

[alsa-devel] ALSA queries
by Chakravarthi Pradeep 03 Aug '18

03 Aug '18

I'm working on ALSA driver for PCIe card. My ALSA driver and it's initializing struct snd_pcm_hardware with below parameter. /************************ code start ************************************************/ static struct snd_pcm_hardware audio_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME ), .formats = (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S24_LE), .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), .rate_min = 44100, .rate_max = 192000, .channels_min = 2, .channels_max = 8, .buffer_bytes_max = 76800, /*75 kbytes */ .period_bytes_min = 512,//4410, /* (channel (2) * sample_rate (44100) * bit width (24)) / (60 * 8) */ .period_bytes_max = 16*1024, .periods_min = 10, .periods_max = 255, }; /************************ code end ************************************************/ 1) I did not understand what is significance of periods_min , period_bytes_min , period_bytes_max and periods_max. Can you please tell me what is importance of these parameter and what value should be mentioned according into ALSA. 2) snd_pcm_ops trigger callback is getting called in the driver when application sends "start" command. But ALSA driver is stopping by itself after one frame is copied to ALSA framework, without waiting for "stop" command. For instance: In trigger callback , I'm getting these logs after one frame is copied. Trigger:Start (When Play button is selected/clicked in application, Start command is sent to ALSA driver) Dma transfer is completed. Trigger:Stop. (When Stop button is selected/clicked in application, Stop command is sent to ALSA driver. But stop button is not clicked in this case) Thanks in advance. -- Thanks and Regards Chakravarthi Pradeep.K Ph: 91 9980434900

2 2

[alsa-devel] bug report. Envy24 based card (terratec ews88d) won't slave to external clock via s/pdif.
by Tobias Eklund 03 Aug '18

03 Aug '18

Hi, when trying to clock my terratec ews88d from an apogee rosetta AD it just won't. The s/pdif input works but the card won't lock to the s/pdif clock but rather stay at whatever setting it was at before i change to s/pdif in envy24control or mudita24. Since I have another converter slaved via ADAT out I can tell what sample rate is being output via ADAT sync. Works perfectly for setting the sample rate in envy24control but not when i choose external clock and change the rate on the Apogee AD. Also, audio recorded through the s/pdif input is obviously at the wrong rate when changing the rate on the Apogee AD. I found this old discussion which seems to describe the same problem: http://alsa-devel.alsa-project.narkive.com/30mvygRD/m-audio-delta-1010lt-s-… Also, there is a problem with the alsa-info.sh script, it didn't give me url to share. Just the message Your ALSA information is located at Please inform the person helping you. If the alsa-info.sh output is required i'll post it in an e-mail. kind regards, -- Tobias Eklund Fatburs Brunnsgata 31 118 28 Stockholm 0046 707570196

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix wrong sizeof argument
by Gustavo A. R. Silva 02 Aug '18

02 Aug '18

The correct argument to be passed to kzalloc is sizeof(*pd) Addresses-Coverity-ID: 1472224 ("Wrong sizeof argument") Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/usb/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8fe3b0e..67cf849 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1037,7 +1037,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, fp->rate_max = UAC3_BADD_SAMPLING_RATE; fp->rates = SNDRV_PCM_RATE_CONTINUOUS; - pd = kzalloc(sizeof(pd), GFP_KERNEL); + pd = kzalloc(sizeof(*pd), GFP_KERNEL); if (!pd) { kfree(fp->rate_table); kfree(fp); -- 2.7.4

2 2

[alsa-devel] [PATCH 0/3] ASoC: core: remove support for card rebind using component framework
by Srinivas Kandagatla 02 Aug '18

02 Aug '18

DRM based audio components get registered inside the component framework bind callback. However component framework has a big mutex lock taken for every call to component_add, component_del and bind, unbind callbacks. This can lead to deadlock situation if we are trying to add new/remove component within a bind/unbind callbacks. Which is what was happening with bcm2837 rpi 3. Revert this change till we sort out the mutex issue. Thanks, srini Srinivas Kandagatla (3): ASoC: apq8096: remove auto rebinding ASoC: smd845: remove auto rebinding ASoC: core: remove support for card rebind using component framework include/sound/soc.h | 7 ------ sound/soc/qcom/apq8096.c | 2 -- sound/soc/qcom/sdm845.c | 2 -- sound/soc/soc-core.c | 62 ------------------------------------------------ 4 files changed, 73 deletions(-) -- 2.16.2

2 6

[alsa-devel] [PATCH v3 0/4] arm64: dts: rockchip: add basic dts file for PX30 SoCs
by cl＠rock-chips.com 02 Aug '18

02 Aug '18

From: Liang Chen <cl(a)rock-chips.com> Changes in v3: 1. backlight: remove properties <brightness-levels>,<num-interpolated-steps>, <default-brightness-level>, as the driver will compute a default table. Changes in V2: 1. change the subject to format "dt-bindings: module: ..." for the documents. 2. use new property <num-interpolated-steps> for backlight. 3. remove pinctrl of rk809, because the driver is not ready. Liang Chen (4): dt-bindings: usb: dwc2: add description for px30 dt-bindings: rockchip: grf: add grf and pmugrf description for px30 arm64: dts: rockchip: add core dtsi file for PX30 SoCs arm64: dts: rockchip: add PX30 eavluation board devicetree Documentation/devicetree/bindings/arm/rockchip.txt | 4 + .../devicetree/bindings/soc/rockchip/grf.txt | 2 + Documentation/devicetree/bindings/usb/dwc2.txt | 1 + arch/arm64/boot/dts/rockchip/Makefile | 1 + arch/arm64/boot/dts/rockchip/px30-evb.dts | 470 +++++ arch/arm64/boot/dts/rockchip/px30.dtsi | 2043 ++++++++++++++++++++ 6 files changed, 2521 insertions(+) create mode 100644 arch/arm64/boot/dts/rockchip/px30-evb.dts create mode 100644 arch/arm64/boot/dts/rockchip/px30.dtsi -- 1.9.1

2 8

[alsa-devel] [PATCH] ALSA: hda: Correct Asrock B85M-ITX power_save blacklist entry
by Hans de Goede 02 Aug '18

02 Aug '18

I added the subsys product-id for the HDMI HDA device rather then for the PCH one, this commit fixes this. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 Cc: stable(a)vger.kernel.org Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1ae1850b3bfd..647ae1a71e10 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2207,7 +2207,7 @@ static int azx_probe(struct pci_dev *pci, */ static struct snd_pci_quirk power_save_blacklist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ - SND_PCI_QUIRK(0x1849, 0x0c0c, "Asrock B85M-ITX", 0), + SND_PCI_QUIRK(0x1849, 0xc892, "Asrock B85M-ITX", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1849, 0x7662, "Asrock H81M-HDS", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ -- 2.18.0

2 1

Re: [alsa-devel] A bug about cache inconsistency report
by Hans Hu(SH-RD) 02 Aug '18

02 Aug '18

1. noise relate test: 8086:1C20 noise ; 8086:A170 ok; 8086:8D20 ok; Seems this issue related to the HDA Card version 2. change *_SG relate test: Both above platform report Oops Thanks Hans -----邮件原件----- 发件人: Takashi Iwai [mailto:tiwai@suse.de] 发送时间: 01 August 2018 12:45 收件人: Hans Hu(SH-RD) <HansHu(a)zhaoxin.com> 抄送: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org> 主题: Re: [alsa-devel] A bug about cache inconsistency report On Wed, 01 Aug 2018 04:05:18 +0200, Hans Hu(SH-RD) wrote: > > I did more test base on different Intel platform, and result is the same. Which platform did you test exactly? > Maybe I made some mistake? Can you do the test in yours platform? The negative test result implies that the analysis isn't right. Actually I haven't seen that problem on my test machines. BTW, please avoid top-posting. thanks, Takashi > > Thank you! > Hans > > -----邮件原件----- > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > 发送时间: 31 July 2018 21:55 > 收件人: Hans Hu(SH-RD) <HansHu(a)zhaoxin.com> > 抄送: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org> > 主题: Re: 答复: [alsa-devel] A bug about cache inconsistency report > > On Tue, 31 Jul 2018 15:51:06 +0200, > Hans Hu(SH-RD) wrote: > > > > I did the test and Oops report when install driver( both snoop and > > non-snoop mode) > > I don't understand. Did my one-liner change the behavior or not? > > > thanks, > > Takashi > > > > > -----邮件原件----- > > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > > 发送时间: 31 July 2018 19:59 > > 收件人: Hans Hu(SH-RD) <HansHu(a)zhaoxin.com> > > 抄送: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org> > > 主题: Re: [alsa-devel] A bug about cache inconsistency report > > > > On Tue, 31 Jul 2018 12:52:04 +0200, > > Hans Hu(SH-RD) wrote: > > > > > > Hi All, > > > > > > Try send an email. > > > > > > Appearance: > > > when use non-snoop + ioctl path combination, an issue occurs: > > > usage: aplay -D plughw:0,0 44100_S16_LE_2c.wav Ubuntu, it > > > performance intermittently noise; > > > > > > Root case: > > > Function set_pages_array_wc marked the PTE of the original address; Function set_memory_wc marked the PTE both original and after vmaped address. > > > [cid:image002.png@01D428FF.92621520] > > > However, cache or not in transmission process is based on the PTE of the later, and the set_pages_array_wc did't mark it, so a cache inconsistency issue occurs. > > > > > > Related bug: > > > SHA1 ID: 9ddf1aeb2134e72275c97a2c6ff2e3eb04f2f27a > > > > OK, so did you confirm that dropping the SG buffer support "fixes" the noise problem on your system? The easiest test would be like: > > > > --- a/sound/pci/hda/hda_controller.c > > +++ b/sound/pci/hda/hda_controller.c > > @@ -770,7 +770,7 @@ int snd_hda_attach_pcm_stream(struct hda_bus > > *_bus, struct hda_codec *codec, size = CONFIG_SND_HDA_PREALLOC_SIZE > > * 1024; if (size > MAX_PREALLOC_SIZE) size = MAX_PREALLOC_SIZE; > > -snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, > > +snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > > chip->card->dev, > > size, MAX_PREALLOC_SIZE); > > return 0; > > > > > > thanks, > > > > Takashi > > > > > > 保密声明： > > 本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 > > CONFIDENTIAL NOTE: > > This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited. > > > 保密声明： > 本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 > CONFIDENTIAL NOTE: > This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited. 保密声明：本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 CONFIDENTIAL NOTE: This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

2 3

[alsa-devel] [PATCH] ASoC: qcom: Fix unmet dependency warning for SND_SOC_SDM845
by Rohit kumar 02 Aug '18

02 Aug '18

Add DEPENDS_ON QCOM_APR for SND_SOC_SDM845 to fix the warning: unmet direct dependencies detected for SND_SOC_QDSP6. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> --- sound/soc/qcom/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 3507308..5f03312 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -93,6 +93,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" + depends on QCOM_APR select SND_SOC_QDSP6 help To add support for audio on Qualcomm Technologies Inc. -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

[alsa-devel] [PATCH] ASoC: fsi: convert to SPDX identifiers
by Kuninori Morimoto 02 Aug '18

02 Aug '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/sh_fsi.h | 13 +++++-------- 1 file changed, 5 insertions(+), 8 deletions(-) diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h index 7a9710b..89eafe2 100644 --- a/include/sound/sh_fsi.h +++ b/include/sound/sh_fsi.h @@ -1,16 +1,13 @@ -#ifndef __SOUND_FSI_H -#define __SOUND_FSI_H - -/* +/* SPDX-License-Identifier: GPL-2.0 + * * Fifo-attached Serial Interface (FSI) support for SH7724 * * Copyright (C) 2009 Renesas Solutions Corp. * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ +#ifndef __SOUND_FSI_H +#define __SOUND_FSI_H + #include <linux/clk.h> #include <sound/soc.h> -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: sb: Fix a typo
by Takashi Iwai 02 Aug '18

02 Aug '18

There was a typo of COPY_USER in the dead code (that is disabled as default). Fixes: 4b83eff81c81 ("ALSA: sb: Convert to the new PCM ops") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/sb/emu8000_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index bc5af71d3bdb..f46f6ec3ea0c 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -470,7 +470,7 @@ static int emu8k_pcm_copy(struct snd_pcm_substream *subs, /* convert to word unit */ pos = (pos << 1) + rec->loop_start[voice]; count <<= 1; - LOOP_WRITE(rec, pos, src, count, COPY_UESR); + LOOP_WRITE(rec, pos, src, count, COPY_USER); return 0; } -- 2.18.0

1 0

[alsa-devel] [PATCH -next] ALSA: usb-audio: Fix invalid use of sizeof in parse_uac_endpoint_attributes()
by Wei Yongjun 02 Aug '18

02 Aug '18

sizeof() when applied to a pointer typed expression gives the size of the pointer, not that of the pointed data. Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/usb/stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8fe3b0e..67cf849 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1037,7 +1037,7 @@ static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip, fp->rate_max = UAC3_BADD_SAMPLING_RATE; fp->rates = SNDRV_PCM_RATE_CONTINUOUS; - pd = kzalloc(sizeof(pd), GFP_KERNEL); + pd = kzalloc(sizeof(*pd), GFP_KERNEL); if (!pd) { kfree(fp->rate_table); kfree(fp);

2 1

[alsa-devel] [PATCH] ALSA: usb: Mark expected switch fall-through
by Gustavo A. R. Silva 01 Aug '18

01 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 115084 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/usb/pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bbc7116..3828471 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1694,6 +1694,7 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea switch (cmd) { case SNDRV_PCM_TRIGGER_START: subs->trigger_tstamp_pending_update = true; + /* fall through */ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: subs->data_endpoint->prepare_data_urb = prepare_playback_urb; subs->data_endpoint->retire_data_urb = retire_playback_urb; -- 2.7.4

2 1

[alsa-devel] [RESEND][PATCH] ASoC: Intel: Power down links before turning off display audio power
by Sriram Periyasamy 01 Aug '18

01 Aug '18

On certain platforms, Display HDMI HDA codec was not going to sleep state after the use when links are powered down after turning off the display power. As per the HW recommendation, links are powered down before turning off the display power to ensure that the codec goes to sleep state. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 +++++------- sound/soc/intel/skylake/skl.c | 11 ++++++----- 2 files changed, 11 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 0cdceca2b216..c98c4c2babc9 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -2235,12 +2235,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) */ snd_hdac_codec_read(hdev, hdev->afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); - err = snd_hdac_display_power(bus, false); - if (err < 0) { - dev_err(bus->dev, "Cannot turn on display power on i915\n"); - return err; - } - hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); if (!hlink) { dev_err(dev, "hdac link not found\n"); @@ -2249,7 +2243,11 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) snd_hdac_ext_bus_link_put(ebus, hlink); - return 0; + err = snd_hdac_display_power(bus, false); + if (err < 0) + dev_err(bus->dev, "Cannot turn off display power on i915\n"); + + return err; } static int hdac_hdmi_runtime_resume(struct device *dev) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 670ff9aaca55..c62e474717ff 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -762,6 +762,12 @@ static void skl_probe_work(struct work_struct *work) } } + /* + * we are done probing so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { err = snd_hdac_display_power(bus, false); if (err < 0) { @@ -771,11 +777,6 @@ static void skl_probe_work(struct work_struct *work) } } - /* - * we are done probing so decrement link counts - */ - list_for_each_entry(hlink, &ebus->hlink_list, list) - snd_hdac_ext_bus_link_put(ebus, hlink); /* configure PM */ pm_runtime_put_noidle(bus->dev); -- 2.7.4

2 1

[alsa-devel] [PATCH 4.14 211/246] ASoC: topology: Add missing clock gating parameter when parsing hw_configs
by Greg Kroah-Hartman 01 Aug '18

01 Aug '18

4.14-stable review patch. If anyone has any objections, please let me know. ------------------ From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit 933e1c4a667103c4d10ebdc9505a0a6abd8c3fbd ] Clock gating parameter is a part of `dai_fmt`. It is supported by `alsa-lib` when creating a topology binary file, but ignored by kernel when loading this topology file. After applying this commit, the clock gating parameter is not ignored any more. This solution is backwards compatible. The existing behaviour is not broken, because by default the parameter value is 0 and is ignored. snd_soc_tplg_hw_config.clock_gated = 0 => no effect snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. pm_gate_clocks "true" # clock can be gated } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" ] default_hw_conf_id "1" } ~~~~ Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pan Xiuli <xiuli.pan(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 7 ++++++- sound/soc/soc-topology.c | 7 +++++++ 2 files changed, 13 insertions(+), 1 deletion(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -139,6 +139,11 @@ #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) +/* DAI clock gating */ +#define SND_SOC_TPLG_DAI_CLK_GATE_UNDEFINED 0 +#define SND_SOC_TPLG_DAI_CLK_GATE_GATED 1 +#define SND_SOC_TPLG_DAI_CLK_GATE_CONT 2 + /* DAI physical PCM data formats. * Add new formats to the end of the list. */ @@ -324,7 +329,7 @@ struct snd_soc_tplg_hw_config { __le32 size; /* in bytes of this structure */ __le32 id; /* unique ID - - used to match */ __le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */ - __u8 clock_gated; /* 1 if clock can be gated to save power */ + __u8 clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1997,6 +1997,13 @@ static void set_link_hw_format(struct sn link->dai_fmt = hw_config->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + /* clock gating */ + if (hw_config->clock_gated == SND_SOC_TPLG_DAI_CLK_GATE_GATED) + link->dai_fmt |= SND_SOC_DAIFMT_GATED; + else if (hw_config->clock_gated == + SND_SOC_TPLG_DAI_CLK_GATE_CONT) + link->dai_fmt |= SND_SOC_DAIFMT_CONT; + /* clock signal polarity */ invert_bclk = hw_config->invert_bclk; invert_fsync = hw_config->invert_fsync;

1 0

[alsa-devel] [PATCH 4.14 210/246] ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()
by Greg Kroah-Hartman 01 Aug '18

01 Aug '18

4.14-stable review patch. If anyone has any objections, please let me know. ------------------ From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit a941e2fab3207cb0d57dc4ec47b1b12c8ea78b84 ] The values of bclk and fsync are inverted WRT the codec. But the existing solution already works for Broadwell, see the alsa-lib config: `alsa-lib/src/conf/topology/broadwell/broadwell.conf` This commit provides the backwards-compatible solution to fix this misuse. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 16 ++++++++++++++-- sound/soc/soc-topology.c | 12 +++++++----- 2 files changed, 21 insertions(+), 7 deletions(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -160,6 +160,18 @@ #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) +/* DAI topology BCLK parameter + * For the backwards capability, by default codec is bclk master + */ +#define SND_SOC_TPLG_BCLK_CM 0 /* codec is bclk master */ +#define SND_SOC_TPLG_BCLK_CS 1 /* codec is bclk slave */ + +/* DAI topology FSYNC parameter + * For the backwards capability, by default codec is fsync master + */ +#define SND_SOC_TPLG_FSYNC_CM 0 /* codec is fsync master */ +#define SND_SOC_TPLG_FSYNC_CS 1 /* codec is fsync slave */ + /* * Block Header. * This header precedes all object and object arrays below. @@ -315,8 +327,8 @@ struct snd_soc_tplg_hw_config { __u8 clock_gated; /* 1 if clock can be gated to save power */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ - __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ - __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */ + __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* 0 for input, 1 for output */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2010,13 +2010,15 @@ static void set_link_hw_format(struct sn link->dai_fmt |= SND_SOC_DAIFMT_IB_IF; /* clock masters */ - bclk_master = hw_config->bclk_master; - fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + bclk_master = (hw_config->bclk_master == + SND_SOC_TPLG_BCLK_CM); + fsync_master = (hw_config->fsync_master == + SND_SOC_TPLG_FSYNC_CM); + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;

1 0

[alsa-devel] [PATCH 4.17 290/336] ASoC: topology: Add missing clock gating parameter when parsing hw_configs
by Greg Kroah-Hartman 01 Aug '18

01 Aug '18

4.17-stable review patch. If anyone has any objections, please let me know. ------------------ From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit 933e1c4a667103c4d10ebdc9505a0a6abd8c3fbd ] Clock gating parameter is a part of `dai_fmt`. It is supported by `alsa-lib` when creating a topology binary file, but ignored by kernel when loading this topology file. After applying this commit, the clock gating parameter is not ignored any more. This solution is backwards compatible. The existing behaviour is not broken, because by default the parameter value is 0 and is ignored. snd_soc_tplg_hw_config.clock_gated = 0 => no effect snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. pm_gate_clocks "true" # clock can be gated } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" ] default_hw_conf_id "1" } ~~~~ Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pan Xiuli <xiuli.pan(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 7 ++++++- sound/soc/soc-topology.c | 7 +++++++ 2 files changed, 13 insertions(+), 1 deletion(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -139,6 +139,11 @@ #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) +/* DAI clock gating */ +#define SND_SOC_TPLG_DAI_CLK_GATE_UNDEFINED 0 +#define SND_SOC_TPLG_DAI_CLK_GATE_GATED 1 +#define SND_SOC_TPLG_DAI_CLK_GATE_CONT 2 + /* DAI physical PCM data formats. * Add new formats to the end of the list. */ @@ -324,7 +329,7 @@ struct snd_soc_tplg_hw_config { __le32 size; /* in bytes of this structure */ __le32 id; /* unique ID - - used to match */ __le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */ - __u8 clock_gated; /* 1 if clock can be gated to save power */ + __u8 clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2006,6 +2006,13 @@ static void set_link_hw_format(struct sn link->dai_fmt = hw_config->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + /* clock gating */ + if (hw_config->clock_gated == SND_SOC_TPLG_DAI_CLK_GATE_GATED) + link->dai_fmt |= SND_SOC_DAIFMT_GATED; + else if (hw_config->clock_gated == + SND_SOC_TPLG_DAI_CLK_GATE_CONT) + link->dai_fmt |= SND_SOC_DAIFMT_CONT; + /* clock signal polarity */ invert_bclk = hw_config->invert_bclk; invert_fsync = hw_config->invert_fsync;

1 0

[alsa-devel] [PATCH 4.17 289/336] ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()
by Greg Kroah-Hartman 01 Aug '18

01 Aug '18

4.17-stable review patch. If anyone has any objections, please let me know. ------------------ From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit a941e2fab3207cb0d57dc4ec47b1b12c8ea78b84 ] The values of bclk and fsync are inverted WRT the codec. But the existing solution already works for Broadwell, see the alsa-lib config: `alsa-lib/src/conf/topology/broadwell/broadwell.conf` This commit provides the backwards-compatible solution to fix this misuse. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 16 ++++++++++++++-- sound/soc/soc-topology.c | 12 +++++++----- 2 files changed, 21 insertions(+), 7 deletions(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -160,6 +160,18 @@ #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) +/* DAI topology BCLK parameter + * For the backwards capability, by default codec is bclk master + */ +#define SND_SOC_TPLG_BCLK_CM 0 /* codec is bclk master */ +#define SND_SOC_TPLG_BCLK_CS 1 /* codec is bclk slave */ + +/* DAI topology FSYNC parameter + * For the backwards capability, by default codec is fsync master + */ +#define SND_SOC_TPLG_FSYNC_CM 0 /* codec is fsync master */ +#define SND_SOC_TPLG_FSYNC_CS 1 /* codec is fsync slave */ + /* * Block Header. * This header precedes all object and object arrays below. @@ -315,8 +327,8 @@ struct snd_soc_tplg_hw_config { __u8 clock_gated; /* 1 if clock can be gated to save power */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ - __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ - __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */ + __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* 0 for input, 1 for output */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2019,13 +2019,15 @@ static void set_link_hw_format(struct sn link->dai_fmt |= SND_SOC_DAIFMT_IB_IF; /* clock masters */ - bclk_master = hw_config->bclk_master; - fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + bclk_master = (hw_config->bclk_master == + SND_SOC_TPLG_BCLK_CM); + fsync_master = (hw_config->fsync_master == + SND_SOC_TPLG_FSYNC_CM); + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS;

1 0

[alsa-devel] [PATCH] ALSA: pcm: Mark expected switch fall-through
by Gustavo A. R. Silva 01 Aug '18

01 Aug '18

In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Addresses-Coverity-ID: 1357375 ("Missing break in switch") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/oss/pcm_plugin.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 85bab92..0391cb1a 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -353,6 +353,7 @@ snd_pcm_format_t snd_pcm_plug_slave_format(snd_pcm_format_t format, if (snd_mask_test(format_mask, (__force int)format1)) return format1; } + /* fall through */ default: return (__force snd_pcm_format_t)-EINVAL; } -- 2.7.4

2 1

[alsa-devel] [PATCH 0/5] ALSA: seq: Some minor cleanups
by Takashi Iwai 01 Aug '18

01 Aug '18

Hi, while debugging the recent virmidi bug, I've found a few ugly appearance in the sequencer code, so spent a bit time for cleaning it up. Most of changes are cosmetic, non-functional changes. Takashi === Takashi Iwai (5): ALSA: seq: Minor cleanup of MIDI event parser helpers ALSA: seq: Remove dead codes ALSA: seq: Fix leftovers at probe error path ALSA: seq: Use no intrruptible mutex_lock ALSA: seq: Drop unused 64bit division macros include/sound/seq_midi_event.h | 6 +-- sound/core/seq/oss/seq_oss_midi.c | 2 +- sound/core/seq/seq.c | 33 ++++++------ sound/core/seq/seq_clientmgr.c | 28 +++------- sound/core/seq/seq_info.c | 10 +--- sound/core/seq/seq_info.h | 6 +-- sound/core/seq/seq_memory.c | 12 ----- sound/core/seq/seq_memory.h | 6 --- sound/core/seq/seq_midi.c | 24 ++++----- sound/core/seq/seq_midi_emul.c | 12 ----- sound/core/seq/seq_midi_event.c | 87 +++---------------------------- sound/core/seq/seq_queue.c | 12 +---- sound/core/seq/seq_queue.h | 27 ---------- sound/core/seq/seq_virmidi.c | 37 +------------ 14 files changed, 52 insertions(+), 250 deletions(-) -- 2.18.0

1 5

[alsa-devel] [PATCH v5 0/8] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 01 Aug '18

01 Aug '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v5 is based on Takashi's topic/drm_audio_component merged on top on Marks' for-next branch - the merge is not included here. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Both the HDaudio and HDMI outputs were tested. Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. TODO in future update: fix the HDMI jack detection which only works after the mixer values are set, which isn't practical for headless devices always connected (this is a problem in the hdac_hdmi codec that was present before this series was submitted) Changes v5 (comments from Vinod) - fix SPDIF style across patches - patch 1: update commit message to remove reference to "fix" - patch 3: remove platform name assignment and Xmas tree style - patch 7: simplify error handling, add missing _put() on errors, remove tests on ops, add switch statement, fix alignment - patch 8: fix indentation issues, use bool instead of tristate Changes v4: - rebase/update on Takashi's topic/drm_audio_component branch - changes in the HDaudio detection to avoid adding a fake ACPI ID - new Kconfigs to control HDaudio detection Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Pierre-Louis Bossart (2): ASoC: Intel: common: add table for HDA-based platforms ASoC: Intel: Skylake: add option to control HDAudio + DSP usage Rakesh Ughreja (6): ASoC: Intel: Skylake: extend widget handling ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers include/sound/soc-acpi-intel-match.h | 6 + sound/pci/hda/hda_bind.c | 12 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 487 ++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 ++ sound/soc/intel/Kconfig | 19 + sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 130 ++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 37 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 176 ++++++++ sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 40 ++ sound/soc/intel/skylake/skl-pcm.c | 70 +++- sound/soc/intel/skylake/skl-topology.c | 3 + sound/soc/intel/skylake/skl.c | 100 ++++- sound/soc/intel/skylake/skl.h | 12 +- 18 files changed, 1111 insertions(+), 26 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c -- 2.14.1

3 14

[alsa-devel] Applied "ASoC: wcd9335: Fix build due to CLASS-H Controller support" to the asoc tree
by Mark Brown 01 Aug '18

01 Aug '18

The patch ASoC: wcd9335: Fix build due to CLASS-H Controller support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b0a39d356ae1478a5876bb02ba08af155a3a5554 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Wed, 1 Aug 2018 14:18:50 +0100 Subject: [PATCH] ASoC: wcd9335: Fix build due to CLASS-H Controller support This reverts commit c8cb5f775c8dac (ASoC: vert "ASoC: wcd9335: add CLASS-H Controller support) due to missing dependencies. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/wcd-clsh.c | 605 ------------------------------------ sound/soc/codecs/wcd-clsh.h | 49 --- sound/soc/codecs/wcd9335.c | 10 - 4 files changed, 1 insertion(+), 665 deletions(-) delete mode 100644 sound/soc/codecs/wcd-clsh.c delete mode 100644 sound/soc/codecs/wcd-clsh.h diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e3a3d4694e15..01410b63daac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,7 +192,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o -snd-soc-wcd9335-objs := wcd-clsh.o wcd9335.o +snd-soc-wcd9335-objs := wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o diff --git a/sound/soc/codecs/wcd-clsh.c b/sound/soc/codecs/wcd-clsh.c deleted file mode 100644 index 2393456cbd97..000000000000 --- a/sound/soc/codecs/wcd-clsh.c +++ /dev/null @@ -1,605 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. -// Copyright (c) 2017-2018, Linaro Limited - -#include <linux/slab.h> -#include <sound/soc.h> -#include <linux/kernel.h> -#include <linux/delay.h> -#include <linux/mfd/wcd9335/wcd9335.h> -#include <linux/mfd/wcd9335/registers.h> -#include "wcd-clsh.h" - -struct wcd_clsh_ctrl { - int state; - int mode; - int flyback_users; - int buck_users; - int clsh_users; - int codec_version; - struct snd_soc_component *comp; -}; - -/* Class-H registers for codecs from and above WCD9335 */ -#define WCD9XXX_A_CDC_RX0_RX_PATH_CFG0 WCD9335_REG(0xB, 0x42) -#define WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK BIT(6) -#define WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE BIT(6) -#define WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE 0 -#define WCD9XXX_A_CDC_RX1_RX_PATH_CFG0 WCD9335_REG(0xB, 0x56) -#define WCD9XXX_A_CDC_RX2_RX_PATH_CFG0 WCD9335_REG(0xB, 0x6A) -#define WCD9XXX_A_CDC_CLSH_K1_MSB WCD9335_REG(0xC, 0x08) -#define WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK GENMASK(3, 0) -#define WCD9XXX_A_CDC_CLSH_K1_LSB WCD9335_REG(0xC, 0x09) -#define WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK GENMASK(7, 0) -#define WCD9XXX_A_ANA_RX_SUPPLIES WCD9335_REG(0x6, 0x08) -#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK BIT(1) -#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H 0 -#define WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB BIT(1) -#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK BIT(2) -#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA BIT(2) -#define WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT 0 -#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK BIT(3) -#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA BIT(3) -#define WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT 0 -#define WCD9XXX_A_ANA_RX_VNEG_EN_MASK BIT(6) -#define WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT 6 -#define WCD9XXX_A_ANA_RX_VNEG_ENABLE BIT(6) -#define WCD9XXX_A_ANA_RX_VNEG_DISABLE 0 -#define WCD9XXX_A_ANA_RX_VPOS_EN_MASK BIT(7) -#define WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT 7 -#define WCD9XXX_A_ANA_RX_VPOS_ENABLE BIT(7) -#define WCD9XXX_A_ANA_RX_VPOS_DISABLE 0 -#define WCD9XXX_A_ANA_HPH WCD9335_REG(0x6, 0x09) -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK GENMASK(3, 2) -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA 0x08 -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP 0x04 -#define WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL 0x0 -#define WCD9XXX_A_CDC_CLSH_CRC WCD9335_REG(0xC, 0x01) -#define WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK BIT(0) -#define WCD9XXX_A_CDC_CLSH_CRC_CLK_ENABLE BIT(0) -#define WCD9XXX_A_CDC_CLSH_CRC_CLK_DISABLE 0 -#define WCD9XXX_FLYBACK_EN WCD9335_REG(0x6, 0xA4) -#define WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK GENMASK(6, 5) -#define WCD9XXX_FLYBACK_EN_DELAY_26P25_US 0x40 -#define WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK BIT(4) -#define WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY BIT(4) -#define WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY 0 -#define WCD9XXX_RX_BIAS_FLYB_BUFF WCD9335_REG(0x6, 0xC7) -#define WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK GENMASK(7, 4) -#define WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK GENMASK(0, 3) -#define WCD9XXX_HPH_L_EN WCD9335_REG(0x6, 0xD3) -#define WCD9XXX_HPH_CONST_SEL_L_MASK GENMASK(7, 3) -#define WCD9XXX_HPH_CONST_SEL_BYPASS 0 -#define WCD9XXX_HPH_CONST_SEL_LP_PATH 0x40 -#define WCD9XXX_HPH_CONST_SEL_HQ_PATH 0x80 -#define WCD9XXX_HPH_R_EN WCD9335_REG(0x6, 0xD6) -#define WCD9XXX_HPH_REFBUFF_UHQA_CTL WCD9335_REG(0x6, 0xDD) -#define WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK GENMASK(2, 0) -#define WCD9XXX_CLASSH_CTRL_VCL_2 WCD9335_REG(0x6, 0x9B) -#define WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK GENMASK(5, 4) -#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM 0x20 -#define WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM 0x0 -#define WCD9XXX_CDC_RX1_RX_PATH_CTL WCD9335_REG(0xB, 0x55) -#define WCD9XXX_CDC_RX2_RX_PATH_CTL WCD9335_REG(0xB, 0x69) -#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_CONTROL WCD9335_REG(0xD, 0x41) -#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_EN_MASK BIT(0) -#define WCD9XXX_CDC_CLK_RST_CTRL_MCLK_11P3_EN_MASK BIT(1) -#define WCD9XXX_CLASSH_CTRL_CCL_1 WCD9335_REG(0x6, 0x9C) -#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK GENMASK(7, 4) -#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA 0x50 -#define WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA 0x30 - -#define CLSH_REQ_ENABLE true -#define CLSH_REQ_DISABLE false -#define WCD_USLEEP_RANGE 50 - -enum { - DAC_GAIN_0DB = 0, - DAC_GAIN_0P2DB, - DAC_GAIN_0P4DB, - DAC_GAIN_0P6DB, - DAC_GAIN_0P8DB, - DAC_GAIN_M0P2DB, - DAC_GAIN_M0P4DB, - DAC_GAIN_M0P6DB, -}; - -static inline void wcd_enable_clsh_block(struct wcd_clsh_ctrl *ctrl, - bool enable) -{ - struct snd_soc_component *comp = ctrl->comp; - - if ((enable && ++ctrl->clsh_users == 1) || - (!enable && --ctrl->clsh_users == 0)) - snd_soc_component_update_bits(comp, WCD9XXX_A_CDC_CLSH_CRC, - WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK, - enable); - if (ctrl->clsh_users < 0) - ctrl->clsh_users = 0; -} - -static inline bool wcd_clsh_enable_status(struct snd_soc_component *comp) -{ - return snd_soc_component_read32(comp, WCD9XXX_A_CDC_CLSH_CRC) & - WCD9XXX_A_CDC_CLSH_CRC_CLK_EN_MASK; -} - -static inline void wcd_clsh_set_buck_mode(struct snd_soc_component *comp, - int mode) -{ - /* set to HIFI */ - if (mode == CLS_H_HIFI) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_UHQA); - else - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VPOS_PWR_LVL_DEFAULT); -} - -static inline void wcd_clsh_set_flyback_mode(struct snd_soc_component *comp, - int mode) -{ - /* set to HIFI */ - if (mode == CLS_H_HIFI) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_UHQA); - else - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_MASK, - WCD9XXX_A_ANA_RX_VNEG_PWR_LVL_DEFAULT); -} - -static void wcd_clsh_buck_ctrl(struct wcd_clsh_ctrl *ctrl, - int mode, - bool enable) -{ - struct snd_soc_component *comp = ctrl->comp; - - /* enable/disable buck */ - if ((enable && (++ctrl->buck_users == 1)) || - (!enable && (--ctrl->buck_users == 0))) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VPOS_EN_MASK, - enable << WCD9XXX_A_ANA_RX_VPOS_EN_SHIFT); - /* - * 500us sleep is required after buck enable/disable - * as per HW requirement - */ - usleep_range(500, 500 + WCD_USLEEP_RANGE); -} - -static void wcd_clsh_flyback_ctrl(struct wcd_clsh_ctrl *ctrl, - int mode, - bool enable) -{ - struct snd_soc_component *comp = ctrl->comp; - - int vneg[] = {0x00, 0x40}; - - /* enable/disable flyback */ - if ((enable && (++ctrl->flyback_users == 1)) || - (!enable && (--ctrl->flyback_users == 0))) { - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_VNEG_EN_MASK, - enable << WCD9XXX_A_ANA_RX_VNEG_EN_SHIFT); - /* 100usec delay is needed as per HW requirement */ - usleep_range(100, 110); - - if (enable && (WCD9335_IS_1_1(ctrl->codec_version))) { - wcd_clsh_set_flyback_mode(comp, CLS_H_HIFI); - snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, - WCD9XXX_FLYBACK_EN_DELAY_SEL_MASK, - WCD9XXX_FLYBACK_EN_DELAY_26P25_US); - snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, - WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, - WCD9XXX_FLYBACK_EN_PWDN_WITHOUT_DELAY); - vneg[0] = snd_soc_component_read32(comp, - WCD9XXX_A_ANA_RX_SUPPLIES); - vneg[0] &= ~(0x40); - vneg[1] = vneg[0] | 0x40; - - snd_soc_component_write(comp, - WCD9XXX_A_ANA_RX_SUPPLIES, vneg[0]); - snd_soc_component_write(comp, - WCD9XXX_A_ANA_RX_SUPPLIES, vneg[1]); - /* 500usec delay is needed as per HW requirement */ - usleep_range(500, 510); - snd_soc_component_update_bits(comp, WCD9XXX_FLYBACK_EN, - WCD9XXX_FLYBACK_EN_RESET_BY_EXT_MASK, - WCD9XXX_FLYBACK_EN_PWDN_WITH_DELAY); - wcd_clsh_set_flyback_mode(comp, mode); - } - - } - /* - * 500us sleep is required after flyback enable/disable - * as per HW requirement - */ - usleep_range(500, 500 + WCD_USLEEP_RANGE); -} - -static void wcd_clsh_set_gain_path(struct wcd_clsh_ctrl *ctrl, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - int val = 0; - - switch (mode) { - case CLS_H_NORMAL: - case CLS_AB: - val = WCD9XXX_HPH_CONST_SEL_BYPASS; - break; - case CLS_H_HIFI: - val = WCD9XXX_HPH_CONST_SEL_HQ_PATH; - break; - case CLS_H_LP: - val = WCD9XXX_HPH_CONST_SEL_LP_PATH; - break; - }; - - snd_soc_component_update_bits(comp, WCD9XXX_HPH_L_EN, - WCD9XXX_HPH_CONST_SEL_L_MASK, - val); - - snd_soc_component_update_bits(comp, WCD9XXX_HPH_R_EN, - WCD9XXX_HPH_CONST_SEL_L_MASK, - val); -} - -static void wcd_clsh_set_hph_mode(struct snd_soc_component *comp, - int mode) -{ - int val = 0, gain = 0, res_val; - int ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - - res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_0KOHM; - switch (mode) { - case CLS_H_NORMAL: - res_val = WCD9XXX_CLASSH_CTRL_VCL_VREF_FILT_R_50KOHM; - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; - gain = DAC_GAIN_0DB; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - break; - case CLS_AB: - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_NORMAL; - gain = DAC_GAIN_0DB; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - break; - case CLS_H_HIFI: - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_UHQA; - gain = DAC_GAIN_M0P2DB; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_50MA; - break; - case CLS_H_LP: - val = WCD9XXX_A_ANA_HPH_PWR_LEVEL_LP; - ipeak = WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_30MA; - break; - }; - - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_HPH, - WCD9XXX_A_ANA_HPH_PWR_LEVEL_MASK, val); - snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_VCL_2, - WCD9XXX_CLASSH_CTRL_VCL_2_VREF_FILT_1_MASK, - res_val); - if (mode != CLS_H_LP) - snd_soc_component_update_bits(comp, - WCD9XXX_HPH_REFBUFF_UHQA_CTL, - WCD9XXX_HPH_REFBUFF_UHQA_GAIN_MASK, - gain); - snd_soc_component_update_bits(comp, WCD9XXX_CLASSH_CTRL_CCL_1, - WCD9XXX_CLASSH_CTRL_CCL_1_DELTA_IPEAK_MASK, - ipeak); -} - -static void wcd_clsh_set_flyback_current(struct snd_soc_component *comp, - int mode) -{ - - snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, - WCD9XXX_RX_BIAS_FLYB_VPOS_5_UA_MASK, 0x0A); - snd_soc_component_update_bits(comp, WCD9XXX_RX_BIAS_FLYB_BUFF, - WCD9XXX_RX_BIAS_FLYB_VNEG_5_UA_MASK, 0x0A); - /* Sleep needed to avoid click and pop as per HW requirement */ - usleep_range(100, 110); -} - -static void wcd_clsh_set_buck_regulator_mode(struct snd_soc_component *comp, - int mode) -{ - if (mode == CLS_AB) - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_AB); - else - snd_soc_component_update_bits(comp, WCD9XXX_A_ANA_RX_SUPPLIES, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_MASK, - WCD9XXX_A_ANA_RX_REGULATOR_MODE_CLS_H); -} - -static void wcd_clsh_state_lo(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode != CLS_AB) { - dev_err(comp->dev, "%s: LO cannot be in this mode: %d\n", - __func__, mode); - return; - } - - if (is_enable) { - wcd_clsh_set_buck_regulator_mode(comp, mode); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - } else { - wcd_clsh_buck_ctrl(ctrl, mode, false); - wcd_clsh_flyback_ctrl(ctrl, mode, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); - } -} - -static void wcd_clsh_state_hph_r(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode == CLS_H_NORMAL) { - dev_err(comp->dev, "%s: Normal mode not applicable for hph_r\n", - __func__); - return; - } - - if (is_enable) { - if (mode != CLS_AB) { - wcd_enable_clsh_block(ctrl, true); - /* - * These K1 values depend on the Headphone Impedance - * For now it is assumed to be 16 ohm - */ - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_MSB, - WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, - 0x00); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_LSB, - WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, - 0xC0); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); - } - wcd_clsh_set_buck_regulator_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - wcd_clsh_set_hph_mode(comp, mode); - wcd_clsh_set_gain_path(ctrl, mode); - } else { - wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); - - if (mode != CLS_AB) { - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX2_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); - wcd_enable_clsh_block(ctrl, false); - } - /* buck and flyback set to default mode and disable */ - wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); - } -} - -static void wcd_clsh_state_hph_l(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode == CLS_H_NORMAL) { - dev_err(comp->dev, "%s: Normal mode not applicable for hph_l\n", - __func__); - return; - } - - if (is_enable) { - if (mode != CLS_AB) { - wcd_enable_clsh_block(ctrl, true); - /* - * These K1 values depend on the Headphone Impedance - * For now it is assumed to be 16 ohm - */ - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_MSB, - WCD9XXX_A_CDC_CLSH_K1_MSB_COEF_MASK, - 0x00); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_CLSH_K1_LSB, - WCD9XXX_A_CDC_CLSH_K1_LSB_COEF_MASK, - 0xC0); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); - } - wcd_clsh_set_buck_regulator_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - wcd_clsh_set_hph_mode(comp, mode); - wcd_clsh_set_gain_path(ctrl, mode); - } else { - wcd_clsh_set_hph_mode(comp, CLS_H_NORMAL); - - if (mode != CLS_AB) { - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX1_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); - wcd_enable_clsh_block(ctrl, false); - } - /* set buck and flyback to Default Mode */ - wcd_clsh_buck_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_flyback_ctrl(ctrl, CLS_H_NORMAL, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_regulator_mode(comp, CLS_H_NORMAL); - } -} - -static void wcd_clsh_state_ear(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (mode != CLS_H_NORMAL) { - dev_err(comp->dev, "%s: mode: %d cannot be used for EAR\n", - __func__, mode); - return; - } - - if (is_enable) { - wcd_enable_clsh_block(ctrl, true); - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_ENABLE); - wcd_clsh_set_buck_mode(comp, mode); - wcd_clsh_set_flyback_mode(comp, mode); - wcd_clsh_flyback_ctrl(ctrl, mode, true); - wcd_clsh_set_flyback_current(comp, mode); - wcd_clsh_buck_ctrl(ctrl, mode, true); - } else { - snd_soc_component_update_bits(comp, - WCD9XXX_A_CDC_RX0_RX_PATH_CFG0, - WCD9XXX_A_CDC_RX_PATH_CLSH_EN_MASK, - WCD9XXX_A_CDC_RX_PATH_CLSH_DISABLE); - wcd_enable_clsh_block(ctrl, false); - wcd_clsh_buck_ctrl(ctrl, mode, false); - wcd_clsh_flyback_ctrl(ctrl, mode, false); - wcd_clsh_set_flyback_mode(comp, CLS_H_NORMAL); - wcd_clsh_set_buck_mode(comp, CLS_H_NORMAL); - } -} - -static int _wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, int req_state, - bool is_enable, int mode) -{ - switch (req_state) { - case WCD_CLSH_STATE_EAR: - wcd_clsh_state_ear(ctrl, req_state, is_enable, mode); - break; - case WCD_CLSH_STATE_HPHL: - wcd_clsh_state_hph_l(ctrl, req_state, is_enable, mode); - break; - case WCD_CLSH_STATE_HPHR: - wcd_clsh_state_hph_r(ctrl, req_state, is_enable, mode); - break; - break; - case WCD_CLSH_STATE_LO: - wcd_clsh_state_lo(ctrl, req_state, is_enable, mode); - break; - default: - break; - } - - return 0; -} - -/* - * Function: wcd_clsh_is_state_valid - * Params: state - * Description: - * Provides information on valid states of Class H configuration - */ -static bool wcd_clsh_is_state_valid(int state) -{ - switch (state) { - case WCD_CLSH_STATE_IDLE: - case WCD_CLSH_STATE_EAR: - case WCD_CLSH_STATE_HPHL: - case WCD_CLSH_STATE_HPHR: - case WCD_CLSH_STATE_LO: - return true; - default: - return false; - }; -} - -/* - * Function: wcd_clsh_fsm - * Params: ctrl, req_state, req_type, clsh_event - * Description: - * This function handles PRE DAC and POST DAC conditions of different devices - * and updates class H configuration of different combination of devices - * based on validity of their states. ctrl will contain current - * class h state information - */ -int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, - enum wcd_clsh_event clsh_event, - int nstate, - enum wcd_clsh_mode mode) -{ - struct snd_soc_component *comp = ctrl->comp; - - if (nstate == ctrl->state) - return 0; - - if (!wcd_clsh_is_state_valid(nstate)) { - dev_err(comp->dev, "Class-H not a valid new state:\n"); - return -EINVAL; - } - - switch (clsh_event) { - case WCD_CLSH_EVENT_PRE_DAC: - _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_ENABLE, mode); - break; - case WCD_CLSH_EVENT_POST_PA: - _wcd_clsh_ctrl_set_state(ctrl, nstate, CLSH_REQ_DISABLE, mode); - break; - }; - - ctrl->state = nstate; - ctrl->mode = mode; - - return 0; -} - -int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl) -{ - return ctrl->state; -} - -struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp, - int version) -{ - struct wcd_clsh_ctrl *ctrl; - - ctrl = kzalloc(sizeof(*ctrl), GFP_KERNEL); - if (!ctrl) - return ERR_PTR(-ENOMEM); - - ctrl->state = WCD_CLSH_STATE_IDLE; - ctrl->comp = comp; - - return ctrl; -} - -void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl) -{ - kfree(ctrl); -} diff --git a/sound/soc/codecs/wcd-clsh.h b/sound/soc/codecs/wcd-clsh.h deleted file mode 100644 index a902f9893467..000000000000 --- a/sound/soc/codecs/wcd-clsh.h +++ /dev/null @@ -1,49 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ - -#ifndef _WCD_CLSH_V2_H_ -#define _WCD_CLSH_V2_H_ -#include <sound/soc.h> - -enum wcd_clsh_event { - WCD_CLSH_EVENT_PRE_DAC = 1, - WCD_CLSH_EVENT_POST_PA, -}; - -/* - * Basic states for Class H state machine. - * represented as a bit mask within a u8 data type - * bit 0: EAR mode - * bit 1: HPH Left mode - * bit 2: HPH Right mode - * bit 3: Lineout mode - */ -#define WCD_CLSH_STATE_IDLE 0 -#define WCD_CLSH_STATE_EAR BIT(0) -#define WCD_CLSH_STATE_HPHL BIT(1) -#define WCD_CLSH_STATE_HPHR BIT(2) -#define WCD_CLSH_STATE_LO BIT(3) -#define WCD_CLSH_STATE_MAX 4 -#define NUM_CLSH_STATES_V2 BIT(WCD_CLSH_STATE_MAX) - -enum wcd_clsh_mode { - CLS_H_NORMAL = 0, /* Class-H Default */ - CLS_H_HIFI, /* Class-H HiFi */ - CLS_H_LP, /* Class-H Low Power */ - CLS_AB, /* Class-AB */ - CLS_H_LOHIFI, /* LoHIFI */ - CLS_NONE, /* None of the above modes */ -}; - -struct wcd_clsh_ctrl; - -extern struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc( - struct snd_soc_component *component, - int version); -extern void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl); -extern int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl); -extern int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, - enum wcd_clsh_event event, - int state, - enum wcd_clsh_mode mode); - -#endif /* _WCD_CLSH_V2_H_ */ diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 06c73699f16f..bd9de5d45fa9 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -21,7 +21,6 @@ #include <dt-bindings/mfd/wcd9335.h> #include <linux/mfd/wcd9335/wcd9335.h> #include <linux/mfd/wcd9335/registers.h> -#include "wcd-clsh.h" #define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ @@ -182,7 +181,6 @@ struct wcd9335_codec { int sido_ccl_cnt; enum wcd_clock_type clk_type; - struct wcd_clsh_ctrl *clsh_ctrl; u32 hph_mode; }; @@ -1068,13 +1066,6 @@ static int wcd9335_codec_probe(struct snd_soc_component *component) int i; snd_soc_component_init_regmap(component, wcd->regmap); - /* Class-H Init*/ - wcd->clsh_ctrl = wcd_clsh_ctrl_alloc(component, wcd->version); - if (IS_ERR(wcd->clsh_ctrl)) - return PTR_ERR(wcd->clsh_ctrl); - - /* Default HPH Mode to Class-H HiFi */ - wcd->hph_mode = CLS_H_HIFI; wcd->component = component; wcd9335_codec_init(component); @@ -1089,7 +1080,6 @@ static void wcd9335_codec_remove(struct snd_soc_component *comp) { struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - wcd_clsh_ctrl_free(wcd->clsh_ctrl); free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); } -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: wcd9335: Fix build" to the asoc tree
by Mark Brown 01 Aug '18

01 Aug '18

The patch ASoC: wcd9335: Fix build has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b74fd69043262003c5b7ce545e59f5a7234ab290 Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Wed, 1 Aug 2018 14:22:56 +0100 Subject: [PATCH] ASoC: wcd9335: Fix build This reverts commit e57d4ca882e28 (ASoC: wcd9335: add support to wcd9335 codec) due to build failures caused by missing dependencies. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 5 - sound/soc/codecs/Makefile | 2 - sound/soc/codecs/wcd9335.c | 1154 ------------------------------------ 3 files changed, 1161 deletions(-) delete mode 100644 sound/soc/codecs/wcd9335.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb09abf18dde..efb095dbcd71 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1066,11 +1066,6 @@ config SND_SOC_UDA1380 tristate depends on I2C -config SND_SOC_WCD9335 - tristate "WCD9335 Codec" - depends on MFD_WCD9335 - tristate - config SND_SOC_WL1273 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 01410b63daac..7ae7c85e8219 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -192,7 +192,6 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o -snd-soc-wcd9335-objs := wcd9335.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o snd-soc-wm0010-objs := wm0010.o @@ -452,7 +451,6 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o -obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c deleted file mode 100644 index bd9de5d45fa9..000000000000 --- a/sound/soc/codecs/wcd9335.c +++ /dev/null @@ -1,1154 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0 -// Copyright (c) 2015-2016, The Linux Foundation. All rights reserved. -// Copyright (c) 2017-2018, Linaro Limited - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/device.h> -#include <linux/wait.h> -#include <linux/bitops.h> -#include <linux/regulator/consumer.h> -#include <linux/clk.h> -#include <linux/delay.h> -#include <linux/kernel.h> -#include <linux/slimbus.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/soc-dapm.h> -#include <sound/tlv.h> -#include <sound/info.h> -#include <dt-bindings/mfd/wcd9335.h> -#include <linux/mfd/wcd9335/wcd9335.h> -#include <linux/mfd/wcd9335/registers.h> - -#define WCD9335_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) -/* Fractional Rates */ -#define WCD9335_FRAC_RATES_MASK (SNDRV_PCM_RATE_44100) -#define WCD9335_FORMATS_S16_S24_LE (SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE) - -/* slave port water mark level - * (0: 6bytes, 1: 9bytes, 2: 12 bytes, 3: 15 bytes) - */ -#define SLAVE_PORT_WATER_MARK_6BYTES 0 -#define SLAVE_PORT_WATER_MARK_9BYTES 1 -#define SLAVE_PORT_WATER_MARK_12BYTES 2 -#define SLAVE_PORT_WATER_MARK_15BYTES 3 -#define SLAVE_PORT_WATER_MARK_SHIFT 1 -#define SLAVE_PORT_ENABLE 1 -#define SLAVE_PORT_DISABLE 0 -#define WCD9335_SLIM_WATER_MARK_VAL \ - ((SLAVE_PORT_WATER_MARK_12BYTES << SLAVE_PORT_WATER_MARK_SHIFT) | \ - (SLAVE_PORT_ENABLE)) - -#define WCD9335_SLIM_NUM_PORT_REG 3 -#define WCD9335_SLIM_PGD_PORT_INT_TX_EN0 (WCD9335_SLIM_PGD_PORT_INT_EN0 + 2) - -#define WCD9335_MCLK_CLK_12P288MHZ 12288000 -#define WCD9335_MCLK_CLK_9P6MHZ 9600000 - -#define WCD9335_SLIM_CLOSE_TIMEOUT 1000 -#define WCD9335_SLIM_IRQ_OVERFLOW (1 << 0) -#define WCD9335_SLIM_IRQ_UNDERFLOW (1 << 1) -#define WCD9335_SLIM_IRQ_PORT_CLOSED (1 << 2) - -#define WCD9335_NUM_INTERPOLATORS 9 -#define WCD9335_RX_START 16 -#define WCD9335_SLIM_CH_START 128 - -#define WCD9335_SLIM_RX_CH(p) \ - {.port = p + WCD9335_RX_START, .shift = p,} - -/* vout step value */ -#define WCD9335_CALCULATE_VOUT_D(req_mv) (((req_mv - 650) * 10) / 25) - -enum { - WCD9335_RX0 = 0, - WCD9335_RX1, - WCD9335_RX2, - WCD9335_RX3, - WCD9335_RX4, - WCD9335_RX5, - WCD9335_RX6, - WCD9335_RX7, - WCD9335_RX8, - WCD9335_RX9, - WCD9335_RX10, - WCD9335_RX11, - WCD9335_RX12, - WCD9335_RX_MAX, -}; - -enum { - SIDO_SOURCE_INTERNAL = 0, - SIDO_SOURCE_RCO_BG, -}; - -enum wcd9335_sido_voltage { - SIDO_VOLTAGE_SVS_MV = 950, - SIDO_VOLTAGE_NOMINAL_MV = 1100, -}; - -enum { - AIF1_PB = 0, - AIF1_CAP, - AIF2_PB, - AIF2_CAP, - AIF3_PB, - AIF3_CAP, - AIF4_PB, - NUM_CODEC_DAIS, -}; - -enum { - INTn_2_INP_SEL_ZERO = 0, - INTn_2_INP_SEL_RX0, - INTn_2_INP_SEL_RX1, - INTn_2_INP_SEL_RX2, - INTn_2_INP_SEL_RX3, - INTn_2_INP_SEL_RX4, - INTn_2_INP_SEL_RX5, - INTn_2_INP_SEL_RX6, - INTn_2_INP_SEL_RX7, - INTn_2_INP_SEL_PROXIMITY, -}; - -enum { - INTn_1_MIX_INP_SEL_ZERO = 0, - INTn_1_MIX_INP_SEL_DEC0, - INTn_1_MIX_INP_SEL_DEC1, - INTn_1_MIX_INP_SEL_IIR0, - INTn_1_MIX_INP_SEL_IIR1, - INTn_1_MIX_INP_SEL_RX0, - INTn_1_MIX_INP_SEL_RX1, - INTn_1_MIX_INP_SEL_RX2, - INTn_1_MIX_INP_SEL_RX3, - INTn_1_MIX_INP_SEL_RX4, - INTn_1_MIX_INP_SEL_RX5, - INTn_1_MIX_INP_SEL_RX6, - INTn_1_MIX_INP_SEL_RX7, - -}; - -enum wcd_clock_type { - WCD_CLK_OFF, - WCD_CLK_RCO, - WCD_CLK_MCLK, -}; - -struct wcd9335_slim_ch { - u32 ch_num; - u16 port; - u16 shift; - struct list_head list; -}; - -struct wcd_slim_codec_dai_data { - struct list_head slim_ch_list; - struct slim_stream_config sconfig; - struct slim_stream_runtime *sruntime; -}; - -struct wcd9335_codec { - struct device *dev; - struct clk *mclk; - struct clk *native_clk; - u32 mclk_rate; - u8 intf_type; - u8 version; - - struct slim_device *slim; - struct slim_device *slim_ifd; - struct regmap *regmap; - struct regmap *if_regmap; - struct regmap_irq_chip_data *irq_data; - - struct wcd9335_slim_ch rx_chs[WCD9335_RX_MAX]; - u32 num_rx_port; - - int sido_input_src; - enum wcd9335_sido_voltage sido_voltage; - - struct wcd_slim_codec_dai_data dai[NUM_CODEC_DAIS]; - struct snd_soc_component *component; - - int master_bias_users; - int clk_mclk_users; - int clk_rco_users; - int sido_ccl_cnt; - enum wcd_clock_type clk_type; - - u32 hph_mode; -}; - -static const struct wcd9335_slim_ch wcd9335_rx_chs[WCD9335_RX_MAX] = { - WCD9335_SLIM_RX_CH(0), /* 16 */ - WCD9335_SLIM_RX_CH(1), /* 17 */ - WCD9335_SLIM_RX_CH(2), - WCD9335_SLIM_RX_CH(3), - WCD9335_SLIM_RX_CH(4), - WCD9335_SLIM_RX_CH(5), - WCD9335_SLIM_RX_CH(6), - WCD9335_SLIM_RX_CH(7), - WCD9335_SLIM_RX_CH(8), - WCD9335_SLIM_RX_CH(9), - WCD9335_SLIM_RX_CH(10), - WCD9335_SLIM_RX_CH(11), - WCD9335_SLIM_RX_CH(12), -}; - -struct interp_sample_rate { - int rate; - int rate_val; -}; - -static struct interp_sample_rate int_mix_rate_val[] = { - {48000, 0x4}, /* 48K */ - {96000, 0x5}, /* 96K */ - {192000, 0x6}, /* 192K */ -}; - -static struct interp_sample_rate int_prim_rate_val[] = { - {8000, 0x0}, /* 8K */ - {16000, 0x1}, /* 16K */ - {24000, -EINVAL},/* 24K */ - {32000, 0x3}, /* 32K */ - {48000, 0x4}, /* 48K */ - {96000, 0x5}, /* 96K */ - {192000, 0x6}, /* 192K */ - {384000, 0x7}, /* 384K */ - {44100, 0x8}, /* 44.1K */ -}; - -struct wcd9335_reg_mask_val { - u16 reg; - u8 mask; - u8 val; -}; - -static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_val_2_0[] = { - {WCD9335_RCO_CTRL_2, 0x0F, 0x08}, - {WCD9335_RX_BIAS_FLYB_MID_RST, 0xF0, 0x10}, - {WCD9335_FLYBACK_CTRL_1, 0x20, 0x20}, - {WCD9335_HPH_OCP_CTL, 0xFF, 0x5A}, - {WCD9335_HPH_L_TEST, 0x01, 0x01}, - {WCD9335_HPH_R_TEST, 0x01, 0x01}, - {WCD9335_CDC_BOOST0_BOOST_CFG1, 0x3F, 0x12}, - {WCD9335_CDC_BOOST0_BOOST_CFG2, 0x1C, 0x08}, - {WCD9335_CDC_COMPANDER7_CTL7, 0x1E, 0x18}, - {WCD9335_CDC_BOOST1_BOOST_CFG1, 0x3F, 0x12}, - {WCD9335_CDC_BOOST1_BOOST_CFG2, 0x1C, 0x08}, - {WCD9335_CDC_COMPANDER8_CTL7, 0x1E, 0x18}, - {WCD9335_CDC_TX0_TX_PATH_SEC7, 0xFF, 0x45}, - {WCD9335_CDC_RX0_RX_PATH_SEC0, 0xFC, 0xF4}, - {WCD9335_HPH_REFBUFF_LP_CTL, 0x08, 0x08}, - {WCD9335_HPH_REFBUFF_LP_CTL, 0x06, 0x02}, -}; - -static const struct wcd9335_reg_mask_val wcd9335_codec_reg_init_common_val[] = { - /* Rbuckfly/R_EAR(32) */ - {WCD9335_CDC_CLSH_K2_MSB, 0x0F, 0x00}, - {WCD9335_CDC_CLSH_K2_LSB, 0xFF, 0x60}, - {WCD9335_CPE_SS_DMIC_CFG, 0x80, 0x00}, - {WCD9335_CDC_BOOST0_BOOST_CTL, 0x70, 0x50}, - {WCD9335_CDC_BOOST1_BOOST_CTL, 0x70, 0x50}, - {WCD9335_CDC_RX7_RX_PATH_CFG1, 0x08, 0x08}, - {WCD9335_CDC_RX8_RX_PATH_CFG1, 0x08, 0x08}, - {WCD9335_ANA_LO_1_2, 0x3C, 0X3C}, - {WCD9335_DIFF_LO_COM_SWCAP_REFBUF_FREQ, 0x70, 0x00}, - {WCD9335_DIFF_LO_COM_PA_FREQ, 0x70, 0x40}, - {WCD9335_SOC_MAD_AUDIO_CTL_2, 0x03, 0x03}, - {WCD9335_CDC_TOP_TOP_CFG1, 0x02, 0x02}, - {WCD9335_CDC_TOP_TOP_CFG1, 0x01, 0x01}, - {WCD9335_EAR_CMBUFF, 0x08, 0x00}, - {WCD9335_CDC_TX9_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_TX10_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_TX11_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_TX12_SPKR_PROT_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_COMPANDER7_CTL3, 0x80, 0x80}, - {WCD9335_CDC_COMPANDER8_CTL3, 0x80, 0x80}, - {WCD9335_CDC_COMPANDER7_CTL7, 0x01, 0x01}, - {WCD9335_CDC_COMPANDER8_CTL7, 0x01, 0x01}, - {WCD9335_CDC_RX0_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX1_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX2_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX3_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX4_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX5_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX6_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX7_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX8_RX_PATH_CFG0, 0x01, 0x01}, - {WCD9335_CDC_RX0_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX1_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX2_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX3_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX4_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX5_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX6_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX7_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_CDC_RX8_RX_PATH_MIX_CFG, 0x01, 0x01}, - {WCD9335_VBADC_IBIAS_FE, 0x0C, 0x08}, -}; - -static int wcd9335_set_mix_interpolator_rate(struct snd_soc_dai *dai, - int rate_val, - u32 rate) -{ - struct snd_soc_component *component = dai->component; - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - struct wcd9335_slim_ch *ch; - int val, j; - - list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { - for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { - val = snd_soc_component_read32(component, - WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)) & - WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - - if (val == (ch->shift + INTn_2_INP_SEL_RX0)) - snd_soc_component_update_bits(component, - WCD9335_CDC_RX_PATH_MIX_CTL(j), - WCD9335_CDC_MIX_PCM_RATE_MASK, - rate_val); - } - } - - return 0; -} - -static int wcd9335_set_prim_interpolator_rate(struct snd_soc_dai *dai, - u8 rate_val, - u32 rate) -{ - struct snd_soc_component *comp = dai->component; - struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - struct wcd9335_slim_ch *ch; - u8 cfg0, cfg1, inp0_sel, inp1_sel, inp2_sel; - int inp, j; - - list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) { - inp = ch->shift + INTn_1_MIX_INP_SEL_RX0; - /* - * Loop through all interpolator MUX inputs and find out - * to which interpolator input, the slim rx port - * is connected - */ - for (j = 0; j < WCD9335_NUM_INTERPOLATORS; j++) { - cfg0 = snd_soc_component_read32(comp, - WCD9335_CDC_RX_INP_MUX_RX_INT_CFG0(j)); - cfg1 = snd_soc_component_read32(comp, - WCD9335_CDC_RX_INP_MUX_RX_INT_CFG1(j)); - - inp0_sel = cfg0 & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - inp1_sel = (cfg0 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - inp2_sel = (cfg1 >> 4) & WCD9335_CDC_RX_INP_MUX_RX_INT_SEL_MASK; - - if ((inp0_sel == inp) || (inp1_sel == inp) || - (inp2_sel == inp)) { - /* rate is in Hz */ - if ((j == 0) && (rate == 44100)) - dev_info(wcd->dev, - "Cannot set 44.1KHz on INT0\n"); - else - snd_soc_component_update_bits(comp, - WCD9335_CDC_RX_PATH_CTL(j), - WCD9335_CDC_MIX_PCM_RATE_MASK, - rate_val); - } - } - } - - return 0; -} - -static int wcd9335_set_interpolator_rate(struct snd_soc_dai *dai, u32 rate) -{ - int i; - - /* set mixing path rate */ - for (i = 0; i < ARRAY_SIZE(int_mix_rate_val); i++) { - if (rate == int_mix_rate_val[i].rate) { - wcd9335_set_mix_interpolator_rate(dai, - int_mix_rate_val[i].rate_val, rate); - break; - } - } - - /* set primary path sample rate */ - for (i = 0; i < ARRAY_SIZE(int_prim_rate_val); i++) { - if (rate == int_prim_rate_val[i].rate) { - wcd9335_set_prim_interpolator_rate(dai, - int_prim_rate_val[i].rate_val, rate); - break; - } - } - - return 0; -} - -static int wcd9335_slim_set_hw_params(struct wcd9335_codec *wcd, - struct wcd_slim_codec_dai_data *dai_data, - int direction) -{ - struct list_head *slim_ch_list = &dai_data->slim_ch_list; - struct slim_stream_config *cfg = &dai_data->sconfig; - struct wcd9335_slim_ch *ch; - u16 payload = 0; - int ret, i; - - cfg->ch_count = 0; - cfg->direction = direction; - cfg->port_mask = 0; - - /* Configure slave interface device */ - list_for_each_entry(ch, slim_ch_list, list) { - cfg->ch_count++; - payload |= 1 << ch->shift; - cfg->port_mask |= BIT(ch->port); - } - - cfg->chs = kcalloc(cfg->ch_count, sizeof(unsigned int), GFP_KERNEL); - if (!cfg->chs) - return -ENOMEM; - - i = 0; - list_for_each_entry(ch, slim_ch_list, list) { - cfg->chs[i++] = ch->ch_num; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) { - /* write to interface device */ - ret = regmap_write(wcd->if_regmap, - WCD9335_SLIM_PGD_RX_PORT_MULTI_CHNL_0(ch->port), - payload); - - if (ret < 0) - goto err; - - /* configure the slave port for water mark and enable*/ - ret = regmap_write(wcd->if_regmap, - WCD9335_SLIM_PGD_RX_PORT_CFG(ch->port), - WCD9335_SLIM_WATER_MARK_VAL); - if (ret < 0) - goto err; - } - } - - dai_data->sruntime = slim_stream_allocate(wcd->slim, "WCD9335-SLIM"); - slim_stream_prepare(dai_data->sruntime, cfg); - - return 0; - -err: - dev_err(wcd->dev, "Error Setting slim hw params\n"); - kfree(cfg->chs); - cfg->chs = NULL; - - return ret; -} - -static int wcd9335_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct wcd9335_codec *wcd; - int ret; - - wcd = snd_soc_component_get_drvdata(dai->component); - - switch (substream->stream) { - case SNDRV_PCM_STREAM_PLAYBACK: - ret = wcd9335_set_interpolator_rate(dai, params_rate(params)); - if (ret) { - dev_err(wcd->dev, "cannot set sample rate: %u\n", - params_rate(params)); - return ret; - } - switch (params_width(params)) { - case 16 ... 24: - wcd->dai[dai->id].sconfig.bps = params_width(params); - break; - default: - dev_err(wcd->dev, "%s: Invalid format 0x%x\n", - __func__, params_width(params)); - return -EINVAL; - } - break; - default: - dev_err(wcd->dev, "Invalid stream type %d\n", - substream->stream); - return -EINVAL; - }; - - wcd->dai[dai->id].sconfig.rate = params_rate(params); - wcd9335_slim_set_hw_params(wcd, &wcd->dai[dai->id], substream->stream); - - return 0; -} - -static int wcd9335_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct wcd_slim_codec_dai_data *dai_data; - struct wcd9335_codec *wcd; - - wcd = snd_soc_component_get_drvdata(dai->component); - dai_data = &wcd->dai[dai->id]; - slim_stream_enable(dai_data->sruntime); - - return 0; -} - -static int wcd9335_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) -{ - struct wcd9335_codec *wcd; - int i; - - wcd = snd_soc_component_get_drvdata(dai->component); - - if (!tx_slot || !rx_slot) { - dev_err(wcd->dev, "Invalid tx_slot=%p, rx_slot=%p\n", - tx_slot, rx_slot); - return -EINVAL; - } - - if (wcd->rx_chs) { - wcd->num_rx_port = rx_num; - for (i = 0; i < rx_num; i++) { - wcd->rx_chs[i].ch_num = rx_slot[i]; - INIT_LIST_HEAD(&wcd->rx_chs[i].list); - } - } - - return 0; -} - -static int wcd9335_get_channel_map(struct snd_soc_dai *dai, - unsigned int *tx_num, unsigned int *tx_slot, - unsigned int *rx_num, unsigned int *rx_slot) -{ - struct wcd9335_slim_ch *ch; - struct wcd9335_codec *wcd; - int i = 0; - - wcd = snd_soc_component_get_drvdata(dai->component); - - switch (dai->id) { - case AIF1_PB: - case AIF2_PB: - case AIF3_PB: - case AIF4_PB: - if (!rx_slot || !rx_num) { - dev_err(wcd->dev, "Invalid rx_slot %p or rx_num %p\n", - rx_slot, rx_num); - return -EINVAL; - } - - list_for_each_entry(ch, &wcd->dai[dai->id].slim_ch_list, list) - rx_slot[i++] = ch->ch_num; - - *rx_num = i; - break; - default: - dev_err(wcd->dev, "Invalid DAI ID %x\n", dai->id); - break; - } - - return 0; -} - -static struct snd_soc_dai_ops wcd9335_dai_ops = { - .hw_params = wcd9335_hw_params, - .prepare = wcd9335_prepare, - .set_channel_map = wcd9335_set_channel_map, - .get_channel_map = wcd9335_get_channel_map, -}; - -static struct snd_soc_dai_driver wcd9335_slim_dais[] = { - [0] = { - .name = "wcd9335_rx1", - .id = AIF1_PB, - .playback = { - .stream_name = "AIF1 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_max = 192000, - .rate_min = 8000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, - [1] = { - .name = "wcd9335_tx1", - .id = AIF1_CAP, - .capture = { - .stream_name = "AIF1 Capture", - .rates = WCD9335_RATES_MASK, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 4, - }, - .ops = &wcd9335_dai_ops, - }, - [2] = { - .name = "wcd9335_rx2", - .id = AIF2_PB, - .playback = { - .stream_name = "AIF2 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, - [3] = { - .name = "wcd9335_tx2", - .id = AIF2_CAP, - .capture = { - .stream_name = "AIF2 Capture", - .rates = WCD9335_RATES_MASK, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 4, - }, - .ops = &wcd9335_dai_ops, - }, - [4] = { - .name = "wcd9335_rx3", - .id = AIF3_PB, - .playback = { - .stream_name = "AIF3 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, - [5] = { - .name = "wcd9335_tx3", - .id = AIF3_CAP, - .capture = { - .stream_name = "AIF3 Capture", - .rates = WCD9335_RATES_MASK, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 4, - }, - .ops = &wcd9335_dai_ops, - }, - [6] = { - .name = "wcd9335_rx4", - .id = AIF4_PB, - .playback = { - .stream_name = "AIF4 Playback", - .rates = WCD9335_RATES_MASK | WCD9335_FRAC_RATES_MASK, - .formats = WCD9335_FORMATS_S16_S24_LE, - .rate_min = 8000, - .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, - }, - .ops = &wcd9335_dai_ops, - }, -}; - -static irqreturn_t wcd9335_slimbus_irq(int irq, void *data) -{ - struct wcd9335_codec *wcd = data; - unsigned long status = 0; - int i, j, port_id; - unsigned int val, int_val = 0; - bool tx; - unsigned short reg = 0; - - for (i = WCD9335_SLIM_PGD_PORT_INT_STATUS_RX_0, j = 0; - i <= WCD9335_SLIM_PGD_PORT_INT_STATUS_TX_1; i++, j++) { - regmap_read(wcd->if_regmap, i, &val); - status |= ((u32)val << (8 * j)); - } - - for_each_set_bit(j, &status, 32) { - tx = (j >= 16 ? true : false); - port_id = (tx ? j - 16 : j); - regmap_read(wcd->if_regmap, - WCD9335_SLIM_PGD_PORT_INT_RX_SOURCE0 + j, &val); - if (val) { - if (!tx) - reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + - (port_id / 8); - else - reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + - (port_id / 8); - regmap_read( - wcd->if_regmap, reg, &int_val); - /* - * Ignore interrupts for ports for which the - * interrupts are not specifically enabled. - */ - if (!(int_val & (1 << (port_id % 8)))) - continue; - } - if (val & WCD9335_SLIM_IRQ_OVERFLOW) - dev_err_ratelimited(wcd->dev, - "%s: overflow error on %s port %d, value %x\n", - __func__, (tx ? "TX" : "RX"), port_id, val); - if (val & WCD9335_SLIM_IRQ_UNDERFLOW) - dev_err_ratelimited(wcd->dev, - "%s: underflow error on %s port %d, value %x\n", - __func__, (tx ? "TX" : "RX"), port_id, val); - if ((val & WCD9335_SLIM_IRQ_OVERFLOW) || - (val & WCD9335_SLIM_IRQ_UNDERFLOW)) { - if (!tx) - reg = WCD9335_SLIM_PGD_PORT_INT_EN0 + - (port_id / 8); - else - reg = WCD9335_SLIM_PGD_PORT_INT_TX_EN0 + - (port_id / 8); - regmap_read( - wcd->if_regmap, reg, &int_val); - if (int_val & (1 << (port_id % 8))) { - int_val = int_val ^ (1 << (port_id % 8)); - regmap_write(wcd->if_regmap, - reg, int_val); - } - } - - regmap_write(wcd->if_regmap, - WCD9335_SLIM_PGD_PORT_INT_CLR_RX_0 + (j / 8), - BIT(j % 8)); - } - - return IRQ_HANDLED; -} - -static int wcd9335_setup_irqs(struct wcd9335_codec *wcd) -{ - int slim_irq = regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS); - int i, ret = 0; - - ret = request_threaded_irq(slim_irq, NULL, wcd9335_slimbus_irq, - IRQF_TRIGGER_RISING, "SLIMBus Slave", wcd); - if (ret) { - dev_err(wcd->dev, "Failed to request SLIMBUS irq\n"); - return ret; - } - - /* enable interrupts on all slave ports */ - for (i = 0; i < WCD9335_SLIM_NUM_PORT_REG; i++) - regmap_write(wcd->if_regmap, WCD9335_SLIM_PGD_PORT_INT_EN0 + i, - 0xFF); - - return ret; -} - -static void wcd9335_cdc_sido_ccl_enable(struct wcd9335_codec *wcd, - bool ccl_flag) -{ - struct snd_soc_component *comp = wcd->component; - - if (ccl_flag) { - if (++wcd->sido_ccl_cnt == 1) - snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, - WCD9335_SIDO_SIDO_CCL_DEF_VALUE); - } else { - if (wcd->sido_ccl_cnt == 0) { - dev_err(wcd->dev, "sido_ccl already disabled\n"); - return; - } - if (--wcd->sido_ccl_cnt == 0) - snd_soc_component_write(comp, WCD9335_SIDO_SIDO_CCL_10, - WCD9335_SIDO_SIDO_CCL_10_ICHARG_PWR_SEL_C320FF); - } -} - -static int wcd9335_enable_master_bias(struct wcd9335_codec *wcd) -{ - wcd->master_bias_users++; - if (wcd->master_bias_users == 1) { - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_EN_MASK, - WCD9335_ANA_BIAS_ENABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_EN_MASK, - WCD9335_ANA_BIAS_PRECHRG_ENABLE); - /* - * 1ms delay is required after pre-charge is enabled - * as per HW requirement - */ - usleep_range(1000, 1100); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_EN_MASK, - WCD9335_ANA_BIAS_PRECHRG_DISABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); - } - - return 0; -} - -static int wcd9335_enable_mclk(struct wcd9335_codec *wcd) -{ - /* Enable mclk requires master bias to be enabled first */ - if (wcd->master_bias_users <= 0) - return -EINVAL; - - if (((wcd->clk_mclk_users == 0) && (wcd->clk_type == WCD_CLK_MCLK)) || - ((wcd->clk_mclk_users > 0) && (wcd->clk_type != WCD_CLK_MCLK))) { - dev_err(wcd->dev, "Error enabling MCLK, clk_type: %d\n", - wcd->clk_type); - return -EINVAL; - } - - if (++wcd->clk_mclk_users == 1) { - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, - WCD9335_ANA_CLK_EXT_CLKBUF_ENABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_SRC_MASK, - WCD9335_ANA_CLK_MCLK_SRC_EXTERNAL); - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_EN_MASK, - WCD9335_ANA_CLK_MCLK_ENABLE); - regmap_update_bits(wcd->regmap, - WCD9335_CDC_CLK_RST_CTRL_FS_CNT_CONTROL, - WCD9335_CDC_CLK_RST_CTRL_FS_CNT_EN_MASK, - WCD9335_CDC_CLK_RST_CTRL_FS_CNT_ENABLE); - regmap_update_bits(wcd->regmap, - WCD9335_CDC_CLK_RST_CTRL_MCLK_CONTROL, - WCD9335_CDC_CLK_RST_CTRL_MCLK_EN_MASK, - WCD9335_CDC_CLK_RST_CTRL_MCLK_ENABLE); - /* - * 10us sleep is required after clock is enabled - * as per HW requirement - */ - usleep_range(10, 15); - } - - wcd->clk_type = WCD_CLK_MCLK; - - return 0; -} - -static int wcd9335_disable_mclk(struct wcd9335_codec *wcd) -{ - if (wcd->clk_mclk_users <= 0) - return -EINVAL; - - if (--wcd->clk_mclk_users == 0) { - if (wcd->clk_rco_users > 0) { - /* MCLK to RCO switch */ - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_SRC_MASK, - WCD9335_ANA_CLK_MCLK_SRC_RCO); - wcd->clk_type = WCD_CLK_RCO; - } else { - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_MCLK_EN_MASK, - WCD9335_ANA_CLK_MCLK_DISABLE); - wcd->clk_type = WCD_CLK_OFF; - } - - regmap_update_bits(wcd->regmap, WCD9335_ANA_CLK_TOP, - WCD9335_ANA_CLK_EXT_CLKBUF_EN_MASK, - WCD9335_ANA_CLK_EXT_CLKBUF_DISABLE); - } - - return 0; -} - -static int wcd9335_disable_master_bias(struct wcd9335_codec *wcd) -{ - if (wcd->master_bias_users <= 0) - return -EINVAL; - - wcd->master_bias_users--; - if (wcd->master_bias_users == 0) { - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_EN_MASK, - WCD9335_ANA_BIAS_DISABLE); - regmap_update_bits(wcd->regmap, WCD9335_ANA_BIAS, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE, - WCD9335_ANA_BIAS_PRECHRG_CTL_MODE_MANUAL); - } - return 0; -} - -static int wcd9335_cdc_req_mclk_enable(struct wcd9335_codec *wcd, - bool enable) -{ - int ret = 0; - - if (enable) { - wcd9335_cdc_sido_ccl_enable(wcd, true); - ret = clk_prepare_enable(wcd->mclk); - if (ret) { - dev_err(wcd->dev, "%s: ext clk enable failed\n", - __func__); - goto err; - } - /* get BG */ - wcd9335_enable_master_bias(wcd); - /* get MCLK */ - wcd9335_enable_mclk(wcd); - - } else { - /* put MCLK */ - wcd9335_disable_mclk(wcd); - /* put BG */ - wcd9335_disable_master_bias(wcd); - clk_disable_unprepare(wcd->mclk); - wcd9335_cdc_sido_ccl_enable(wcd, false); - } -err: - return ret; -} - -static void wcd9335_codec_apply_sido_voltage(struct wcd9335_codec *wcd, - enum wcd9335_sido_voltage req_mv) -{ - struct snd_soc_component *comp = wcd->component; - int vout_d_val; - - if (req_mv == wcd->sido_voltage) - return; - - /* compute the vout_d step value */ - vout_d_val = WCD9335_CALCULATE_VOUT_D(req_mv) & - WCD9335_ANA_BUCK_VOUT_MASK; - snd_soc_component_write(comp, WCD9335_ANA_BUCK_VOUT_D, vout_d_val); - snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, - WCD9335_ANA_BUCK_CTL_RAMP_START_ENABLE); - - /* 1 msec sleep required after SIDO Vout_D voltage change */ - usleep_range(1000, 1100); - wcd->sido_voltage = req_mv; - snd_soc_component_update_bits(comp, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_RAMP_START_MASK, - WCD9335_ANA_BUCK_CTL_RAMP_START_DISABLE); -} - -static int wcd9335_codec_update_sido_voltage(struct wcd9335_codec *wcd, - enum wcd9335_sido_voltage req_mv) -{ - int ret = 0; - - /* enable mclk before setting SIDO voltage */ - ret = wcd9335_cdc_req_mclk_enable(wcd, true); - if (ret) { - dev_err(wcd->dev, "Ext clk enable failed\n"); - goto err; - } - - wcd9335_codec_apply_sido_voltage(wcd, req_mv); - wcd9335_cdc_req_mclk_enable(wcd, false); - -err: - return ret; -} - -static int _wcd9335_codec_enable_mclk(struct snd_soc_component *component, - int enable) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - int ret; - - if (enable) { - ret = wcd9335_cdc_req_mclk_enable(wcd, true); - if (ret) - return ret; - - wcd9335_codec_apply_sido_voltage(wcd, - SIDO_VOLTAGE_NOMINAL_MV); - } else { - wcd9335_codec_update_sido_voltage(wcd, - wcd->sido_voltage); - wcd9335_cdc_req_mclk_enable(wcd, false); - } - - return 0; -} - -static void wcd9335_enable_sido_buck(struct snd_soc_component *component) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - - snd_soc_component_update_bits(component, WCD9335_ANA_RCO, - WCD9335_ANA_RCO_BG_EN_MASK, - WCD9335_ANA_RCO_BG_ENABLE); - snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_MASK, - WCD9335_ANA_BUCK_CTL_VOUT_D_IREF_EXT); - /* 100us sleep needed after IREF settings */ - usleep_range(100, 110); - snd_soc_component_update_bits(component, WCD9335_ANA_BUCK_CTL, - WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_MASK, - WCD9335_ANA_BUCK_CTL_VOUT_D_VREF_EXT); - /* 100us sleep needed after VREF settings */ - usleep_range(100, 110); - wcd->sido_input_src = SIDO_SOURCE_RCO_BG; -} - -static int wcd9335_enable_efuse_sensing(struct snd_soc_component *comp) -{ - _wcd9335_codec_enable_mclk(comp, true); - snd_soc_component_update_bits(comp, - WCD9335_CHIP_TIER_CTRL_EFUSE_CTL, - WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK, - WCD9335_CHIP_TIER_CTRL_EFUSE_ENABLE); - /* - * 5ms sleep required after enabling efuse control - * before checking the status. - */ - usleep_range(5000, 5500); - - if (!(snd_soc_component_read32(comp, - WCD9335_CHIP_TIER_CTRL_EFUSE_STATUS) & - WCD9335_CHIP_TIER_CTRL_EFUSE_EN_MASK)) - WARN(1, "%s: Efuse sense is not complete\n", __func__); - - wcd9335_enable_sido_buck(comp); - _wcd9335_codec_enable_mclk(comp, false); - - return 0; -} - -static void wcd9335_codec_init(struct snd_soc_component *component) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - int i; - - /* ungate MCLK and set clk rate */ - regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_GATE, - WCD9335_CODEC_RPM_CLK_GATE_MCLK_GATE_MASK, 0); - - regmap_update_bits(wcd->regmap, WCD9335_CODEC_RPM_CLK_MCLK_CFG, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); - - for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_common_val); i++) - snd_soc_component_update_bits(component, - wcd9335_codec_reg_init_common_val[i].reg, - wcd9335_codec_reg_init_common_val[i].mask, - wcd9335_codec_reg_init_common_val[i].val); - - if (WCD9335_IS_2_0(wcd->version)) { - for (i = 0; i < ARRAY_SIZE(wcd9335_codec_reg_init_val_2_0); i++) - snd_soc_component_update_bits(component, - wcd9335_codec_reg_init_val_2_0[i].reg, - wcd9335_codec_reg_init_val_2_0[i].mask, - wcd9335_codec_reg_init_val_2_0[i].val); - } - - wcd9335_enable_efuse_sensing(component); -} - -static int wcd9335_codec_probe(struct snd_soc_component *component) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(component->dev); - int i; - - snd_soc_component_init_regmap(component, wcd->regmap); - wcd->component = component; - - wcd9335_codec_init(component); - - for (i = 0; i < NUM_CODEC_DAIS; i++) - INIT_LIST_HEAD(&wcd->dai[i].slim_ch_list); - - return wcd9335_setup_irqs(wcd); -} - -static void wcd9335_codec_remove(struct snd_soc_component *comp) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - - free_irq(regmap_irq_get_virq(wcd->irq_data, WCD9335_IRQ_SLIMBUS), wcd); -} - -static int wcd9335_codec_set_sysclk(struct snd_soc_component *comp, - int clk_id, int source, - unsigned int freq, int dir) -{ - struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - - wcd->mclk_rate = freq; - - if (wcd->mclk_rate == WCD9335_MCLK_CLK_12P288MHZ) - snd_soc_component_update_bits(comp, - WCD9335_CODEC_RPM_CLK_MCLK_CFG, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_12P288MHZ); - else if (wcd->mclk_rate == WCD9335_MCLK_CLK_9P6MHZ) - snd_soc_component_update_bits(comp, - WCD9335_CODEC_RPM_CLK_MCLK_CFG, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_MCLK_MASK, - WCD9335_CODEC_RPM_CLK_MCLK_CFG_9P6MHZ); - - return clk_set_rate(wcd->mclk, freq); -} - -static const struct snd_soc_component_driver wcd9335_component_drv = { - .probe = wcd9335_codec_probe, - .remove = wcd9335_codec_remove, - .set_sysclk = wcd9335_codec_set_sysclk, -}; - -static int wcd9335_probe(struct platform_device *pdev) -{ - struct wcd9335 *pdata = dev_get_drvdata(pdev->dev.parent); - struct device *dev = &pdev->dev; - struct wcd9335_codec *wcd; - - wcd = devm_kzalloc(dev, sizeof(*wcd), GFP_KERNEL); - if (!wcd) - return -ENOMEM; - - dev_set_drvdata(dev, wcd); - - memcpy(wcd->rx_chs, wcd9335_rx_chs, sizeof(wcd9335_rx_chs)); - - wcd->regmap = pdata->regmap; - wcd->if_regmap = pdata->ifd_regmap; - wcd->slim = pdata->slim; - wcd->slim_ifd = pdata->slim_ifd; - wcd->irq_data = pdata->irq_data; - wcd->version = pdata->version; - wcd->intf_type = pdata->intf_type; - wcd->dev = dev; - wcd->mclk = pdata->mclk; - wcd->native_clk = pdata->native_clk; - wcd->sido_input_src = SIDO_SOURCE_INTERNAL; - wcd->sido_voltage = SIDO_VOLTAGE_NOMINAL_MV; - - return devm_snd_soc_register_component(dev, &wcd9335_component_drv, - wcd9335_slim_dais, - ARRAY_SIZE(wcd9335_slim_dais)); -} - -static struct platform_driver wcd9335_codec_driver = { - .probe = wcd9335_probe, - .driver = { - .name = "wcd9335-codec", - }, -}; -module_platform_driver(wcd9335_codec_driver); -MODULE_DESCRIPTION("WCD9335 Codec driver"); -MODULE_LICENSE("GPL v2"); -- 2.18.0

1 0

[alsa-devel] [PATCH v2] ASoC: soc-pcm: Use delay set in component pointer function
by Akshu Agrawal 01 Aug '18

01 Aug '18

Take into account the base delay set in pointer callback. There are cases where a pointer function populates runtime->delay, such as: ./sound/pci/hda/hda_controller.c ./sound/soc/intel/atom/sst-mfld-platform-pcm.c This delay was getting lost and was overwritten by delays from codec or cpu dai delay function if exposed. Now, Total delay = base delay + cpu_dai delay + codec_dai delay Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/soc-pcm.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 98be04b..e8b98bfd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1179,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) snd_pcm_sframes_t codec_delay = 0; int i; + /* clearing the previous total delay */ + runtime->delay = 0; + for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -1190,6 +1193,8 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) offset = component->driver->ops->pointer(substream); break; } + /* base delay if assigned in pointer callback */ + delay = runtime->delay; if (cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); -- 1.9.1

3 2

Re: [alsa-devel] [PATCH 3/9] ALSA: emu10k1: remove redundant variable attn
by Walter Harms 01 Aug '18

01 Aug '18

From: Colin Ian King <colin.king(a)canonical.com> Variable attn is being assigned but is never used hence it is redundant and can be removed. Cleans up clang warning: warning: variable 'attn' set but not used [-Wunused-but-set-variable] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/emu10k1/emupcm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 69f9b100bd24..26f6eda3e766 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -290,7 +290,7 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, struct snd_pcm_runtime *runtime = substream->runtime; unsigned int silent_page, tmp; int voice, stereo, w_16; - unsigned char attn, send_amount[8]; + unsigned char send_amount[8]; unsigned char send_routing[8]; unsigned long flags; unsigned int pitch_target; @@ -313,7 +313,6 @@ static void snd_emu10k1_pcm_init_voice(struct snd_emu10k1 *emu, /* volume parameters */ if (extra) { - attn = 0; memset(send_routing, 0, sizeof(send_routing)); send_routing[0] = 0; send_routing[1] = 1; IMHO, this code is a bit fishy. The memset() sould go in the init section and in the else branch tmp is set, maybe its an idea to set it here also (for symmetry) just my 2 cents, re, wh -- 2.17.1 -- To unsubscribe from this list: send the line "unsubscribe kernel-janitors" 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] Applied "ASoC: qcom: dt-bindings: Add sdm845 machine bindings" to the asoc tree
by Mark Brown 01 Aug '18

01 Aug '18

The patch ASoC: qcom: dt-bindings: Add sdm845 machine bindings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 00bc22e3eea0346b594442331df53d0d1e7104dd Mon Sep 17 00:00:00 2001 From: Rohit kumar <rohitkr(a)codeaurora.org> Date: Wed, 1 Aug 2018 14:31:06 +0530 Subject: [PATCH] ASoC: qcom: dt-bindings: Add sdm845 machine bindings Add devicetree bindings documentation file for SDM845 sound card. Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/qcom,sdm845.txt | 80 +++++++++++++++++++ 1 file changed, 80 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt diff --git a/Documentation/devicetree/bindings/sound/qcom,sdm845.txt b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt new file mode 100644 index 000000000000..408c4837e6d5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,sdm845.txt @@ -0,0 +1,80 @@ +* Qualcomm Technologies Inc. SDM845 ASoC sound card driver + +This binding describes the SDM845 sound card, which uses qdsp for audio. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,sdm845-sndcard" + +- audio-routing: + Usage: Optional + Value type: <stringlist> + Definition: A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names could be power supplies, MicBias + of codec and the jacks on the board. + +- model: + Usage: required + Value type: <stringlist> + Definition: The user-visible name of this sound card. + += dailinks +Each subnode of sndcard represents either a dailink, and subnodes of each +dailinks would be cpu/codec/platform dais. + +- link-name: + Usage: required + Value type: <string> + Definition: User friendly name for dai link + += CPU, PLATFORM, CODEC dais subnodes +- cpu: + Usage: required + Value type: <subnode> + Definition: cpu dai sub-node + +- codec: + Usage: required + Value type: <subnode> + Definition: codec dai sub-node + +- platform: + Usage: Optional + Value type: <subnode> + Definition: platform dai sub-node + +- sound-dai: + Usage: required + Value type: <phandle> + Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. + +Example: + +audio { + compatible = "qcom,sdm845-sndcard"; + model = "sdm845-snd-card"; + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&pri_mi2s_active &pri_mi2s_ws_active>; + pinctrl-1 = <&pri_mi2s_sleep &pri_mi2s_ws_sleep>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + pri-mi2s-dai-link { + link-name = "PRI MI2S Playback"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + + platform { + sound-dai = <&q6routing>; + }; + }; +}; -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: dt-bindings: Update dt binding name for apq8096" to the asoc tree
by Mark Brown 01 Aug '18

01 Aug '18

The patch ASoC: dt-bindings: Update dt binding name for apq8096 has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0c901e8cea15ba1e318d8f4342f7dc27a80b5978 Mon Sep 17 00:00:00 2001 From: Rohit kumar <rohitkr(a)codeaurora.org> Date: Wed, 1 Aug 2018 14:31:07 +0530 Subject: [PATCH] ASoC: dt-bindings: Update dt binding name for apq8096 Remove qcom prefix from machine driver dt bindings of apq8096 SoC. Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/qcom,apq8096.txt | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt index aa54e49fc8a2..20bc034f6c01 100644 --- a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt +++ b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt @@ -7,7 +7,7 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio. Value type: <stringlist> Definition: must be "qcom,apq8096-sndcard" -- qcom,audio-routing: +- audio-routing: Usage: Optional Value type: <stringlist> Definition: A list of the connections between audio components. @@ -49,6 +49,12 @@ This binding describes the APQ8096 sound card, which uses qdsp for audio. "DMIC1" "DMIC2" "DMIC3" + +- model: + Usage: required + Value type: <stringlist> + Definition: The user-visible name of this sound card. + = dailinks Each subnode of sndcard represents either a dailink, and subnodes of each dailinks would be cpu/codec/platform dais. @@ -79,11 +85,16 @@ dailinks would be cpu/codec/platform dais. Value type: <phandle with arguments> Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node. +Obsolete: + qcom,model: String for soundcard name (Use model instead) + qcom,audio-routing: A list of the connections between audio components. + (Use audio-routing instead) + Example: audio { compatible = "qcom,apq8096-sndcard"; - qcom,model = "DB820c"; + model = "DB820c"; mm1-dai-link { link-name = "MultiMedia1"; -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: qcom: Add support to parse common audio device nodes" to the asoc tree
by Mark Brown 01 Aug '18

01 Aug '18

The patch ASoC: qcom: Add support to parse common audio device nodes has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c25e295cd77b37903ddc9ee27384e17aad08f27c Mon Sep 17 00:00:00 2001 From: Rohit kumar <rohitkr(a)codeaurora.org> Date: Wed, 1 Aug 2018 14:31:08 +0530 Subject: [PATCH] ASoC: qcom: Add support to parse common audio device nodes This adds support to parse cpu, platform and codec device nodes and add them in dai-links. Also, update apq8096 machine driver to use the common API. Acked-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/Makefile | 2 +- sound/soc/qcom/apq8096.c | 111 ++++---------------------------------- sound/soc/qcom/common.c | 112 +++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/common.h | 12 +++++ 4 files changed, 136 insertions(+), 101 deletions(-) create mode 100644 sound/soc/qcom/common.c create mode 100644 sound/soc/qcom/common.h diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 206945bb9ba1..fefecc072265 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -13,7 +13,7 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o # Machine snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o -snd-soc-apq8096-objs := apq8096.o +snd-soc-apq8096-objs := apq8096.o common.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index a56156281c8d..1e4a90d59228 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -9,6 +9,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/pcm.h> +#include "common.h" static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) @@ -24,109 +25,16 @@ static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int apq8096_sbc_parse_of(struct snd_soc_card *card) +static void apq8096_add_be_ops(struct snd_soc_card *card) { - struct device_node *np; - struct device_node *codec = NULL; - struct device_node *platform = NULL; - struct device_node *cpu = NULL; - struct device *dev = card->dev; - struct snd_soc_dai_link *link; - int ret, num_links; - - ret = snd_soc_of_parse_card_name(card, "qcom,model"); - if (ret) { - dev_err(dev, "Error parsing card name: %d\n", ret); - return ret; - } - - /* DAPM routes */ - if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) { - ret = snd_soc_of_parse_audio_routing(card, - "qcom,audio-routing"); - if (ret) - return ret; - } - - /* Populate links */ - num_links = of_get_child_count(dev->of_node); - - /* Allocate the DAI link array */ - card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); - if (!card->dai_link) - return -ENOMEM; + struct snd_soc_dai_link *link = card->dai_link; + int i, num_links = card->num_links; - card->num_links = num_links; - link = card->dai_link; - - for_each_child_of_node(dev->of_node, np) { - cpu = of_get_child_by_name(np, "cpu"); - if (!cpu) { - dev_err(dev, "Can't find cpu DT node\n"); - ret = -EINVAL; - goto err; - } - - link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); - if (!link->cpu_of_node) { - dev_err(card->dev, "error getting cpu phandle\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); - if (ret) { - dev_err(card->dev, "error getting cpu dai name\n"); - goto err; - } - - platform = of_get_child_by_name(np, "platform"); - codec = of_get_child_by_name(np, "codec"); - if (codec && platform) { - link->platform_of_node = of_parse_phandle(platform, - "sound-dai", - 0); - if (!link->platform_of_node) { - dev_err(card->dev, "platform dai not found\n"); - ret = -EINVAL; - goto err; - } - - ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); - if (ret < 0) { - dev_err(card->dev, "codec dai not found\n"); - goto err; - } - link->no_pcm = 1; - link->ignore_pmdown_time = 1; + for (i = 0; i < num_links; i++) { + if (link->no_pcm == 1) link->be_hw_params_fixup = apq8096_be_hw_params_fixup; - } else { - link->platform_of_node = link->cpu_of_node; - link->codec_dai_name = "snd-soc-dummy-dai"; - link->codec_name = "snd-soc-dummy"; - link->dynamic = 1; - } - - link->ignore_suspend = 1; - ret = of_property_read_string(np, "link-name", &link->name); - if (ret) { - dev_err(card->dev, "error getting codec dai_link name\n"); - goto err; - } - - link->dpcm_playback = 1; - link->dpcm_capture = 1; - link->stream_name = link->name; link++; } - - return 0; -err: - of_node_put(cpu); - of_node_put(codec); - of_node_put(platform); - kfree(card->dai_link); - return ret; } static int apq8096_platform_probe(struct platform_device *pdev) @@ -142,18 +50,21 @@ static int apq8096_platform_probe(struct platform_device *pdev) card->dev = dev; card->auto_bind = true; dev_set_drvdata(dev, card); - ret = apq8096_sbc_parse_of(card); + ret = qcom_snd_parse_of(card); if (ret) { dev_err(dev, "Error parsing OF data\n"); goto err; } + apq8096_add_be_ops(card); ret = snd_soc_register_card(card); if (ret) - goto err; + goto err_card_register; return 0; +err_card_register: + kfree(card->dai_link); err: kfree(card); return ret; diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c new file mode 100644 index 000000000000..eb1b9da05dd4 --- /dev/null +++ b/sound/soc/qcom/common.c @@ -0,0 +1,112 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Linaro Limited. +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#include <linux/module.h> +#include "common.h" + +int qcom_snd_parse_of(struct snd_soc_card *card) +{ + struct device_node *np; + struct device_node *codec = NULL; + struct device_node *platform = NULL; + struct device_node *cpu = NULL; + struct device *dev = card->dev; + struct snd_soc_dai_link *link; + int ret, num_links; + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret) { + dev_err(dev, "Error parsing card name: %d\n", ret); + return ret; + } + + /* DAPM routes */ + if (of_property_read_bool(dev->of_node, "audio-routing")) { + ret = snd_soc_of_parse_audio_routing(card, + "audio-routing"); + if (ret) + return ret; + } + + /* Populate links */ + num_links = of_get_child_count(dev->of_node); + + /* Allocate the DAI link array */ + card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL); + if (!card->dai_link) + return -ENOMEM; + + card->num_links = num_links; + link = card->dai_link; + for_each_child_of_node(dev->of_node, np) { + cpu = of_get_child_by_name(np, "cpu"); + if (!cpu) { + dev_err(dev, "Can't find cpu DT node\n"); + ret = -EINVAL; + goto err; + } + + link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); + if (!link->cpu_of_node) { + dev_err(card->dev, "error getting cpu phandle\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name); + if (ret) { + dev_err(card->dev, "error getting cpu dai name\n"); + goto err; + } + + platform = of_get_child_by_name(np, "platform"); + codec = of_get_child_by_name(np, "codec"); + if (codec && platform) { + link->platform_of_node = of_parse_phandle(platform, + "sound-dai", + 0); + if (!link->platform_of_node) { + dev_err(card->dev, "platform dai not found\n"); + ret = -EINVAL; + goto err; + } + + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + if (ret < 0) { + dev_err(card->dev, "codec dai not found\n"); + goto err; + } + link->no_pcm = 1; + link->ignore_pmdown_time = 1; + } else { + link->platform_of_node = link->cpu_of_node; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + link->dynamic = 1; + } + + link->ignore_suspend = 1; + ret = of_property_read_string(np, "link-name", &link->name); + if (ret) { + dev_err(card->dev, "error getting codec dai_link name\n"); + goto err; + } + + link->dpcm_playback = 1; + link->dpcm_capture = 1; + link->stream_name = link->name; + link++; + } + + return 0; +err: + of_node_put(cpu); + of_node_put(codec); + of_node_put(platform); + kfree(card->dai_link); + return ret; +} +EXPORT_SYMBOL(qcom_snd_parse_of); + +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h new file mode 100644 index 000000000000..ad5d2cf27459 --- /dev/null +++ b/sound/soc/qcom/common.h @@ -0,0 +1,12 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#ifndef __QCOM_SND_COMMON_H__ +#define __QCOM_SND_COMMON_H__ + +#include <linux/component.h> +#include <sound/soc.h> + +int qcom_snd_parse_of(struct snd_soc_card *card); + +#endif -- 2.18.0

1 0

[alsa-devel] [PATCH v5 0/4] Add support for audio on SDM845 SoC
by Rohit kumar 01 Aug '18

01 Aug '18

This provides initial patchset to support audio on Qualcomm Techonologies Inc. SDM845 SoC. Currently, it supports audio playback/capture over Primary MI2S and Quaternary TDM ports. Changes since v4: - Used snd_mask_set_format() instead of snd_mask_set() to avoid sparse warning. Rohit kumar (4): ASoC: qcom: dt-bindings: Add sdm845 machine bindings ASoC: dt-bindings: Update dt binding name for apq8096 ASoC: qcom: Add support to parse common audio device nodes ASoC: qcom: add sdm845 sound card support .../devicetree/bindings/sound/qcom,apq8096.txt | 15 +- .../devicetree/bindings/sound/qcom,sdm845.txt | 80 ++++++ sound/soc/qcom/Kconfig | 8 + sound/soc/qcom/Makefile | 4 +- sound/soc/qcom/apq8096.c | 111 +------- sound/soc/qcom/common.c | 112 ++++++++ sound/soc/qcom/common.h | 12 + sound/soc/qcom/sdm845.c | 286 +++++++++++++++++++++ 8 files changed, 525 insertions(+), 103 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt create mode 100644 sound/soc/qcom/common.c create mode 100644 sound/soc/qcom/common.h create mode 100644 sound/soc/qcom/sdm845.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 6

[alsa-devel] [PATCH v4 0/4] Add support for audio on SDM845 SoC
by Rohit kumar 01 Aug '18

01 Aug '18

This provides initial patchset to support audio on Qualcomm Techonologies Inc. SDM845 SoC. Currently, it supports audio playback/capture over Primary MI2S and Quaternary TDM ports. Changes since v3: - Added Module_license in common.c - Merged apq8096.c machine driver change with common api patch - Addressed comments by Srinivas, Mark and Vinod. Rohit kumar (4): ASoC: qcom: dt-bindings: Add sdm845 machine bindings ASoC: dt-bindings: Update dt binding name for apq8096 ASoC: qcom: Add support to parse common audio device nodes ASoC: qcom: add sdm845 sound card support .../devicetree/bindings/sound/qcom,apq8096.txt | 15 +- .../devicetree/bindings/sound/qcom,sdm845.txt | 80 ++++++ sound/soc/qcom/Kconfig | 8 + sound/soc/qcom/Makefile | 4 +- sound/soc/qcom/apq8096.c | 111 +------- sound/soc/qcom/common.c | 112 ++++++++ sound/soc/qcom/common.h | 12 + sound/soc/qcom/sdm845.c | 286 +++++++++++++++++++++ 8 files changed, 525 insertions(+), 103 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt create mode 100644 sound/soc/qcom/common.c create mode 100644 sound/soc/qcom/common.h create mode 100644 sound/soc/qcom/sdm845.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 9

Re: [alsa-devel] A bug about cache inconsistency report
by Hans Hu(SH-RD) 01 Aug '18

01 Aug '18

I did more test base on different Intel platform, and result is the same. Maybe I made some mistake? Can you do the test in yours platform? Thank you! Hans -----邮件原件----- 发件人: Takashi Iwai [mailto:tiwai@suse.de] 发送时间: 31 July 2018 21:55 收件人: Hans Hu(SH-RD) <HansHu(a)zhaoxin.com> 抄送: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org> 主题: Re: 答复: [alsa-devel] A bug about cache inconsistency report On Tue, 31 Jul 2018 15:51:06 +0200, Hans Hu(SH-RD) wrote: > > I did the test and Oops report when install driver( both snoop and > non-snoop mode) I don't understand. Did my one-liner change the behavior or not? thanks, Takashi > > -----邮件原件----- > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > 发送时间: 31 July 2018 19:59 > 收件人: Hans Hu(SH-RD) <HansHu(a)zhaoxin.com> > 抄送: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org> > 主题: Re: [alsa-devel] A bug about cache inconsistency report > > On Tue, 31 Jul 2018 12:52:04 +0200, > Hans Hu(SH-RD) wrote: > > > > Hi All, > > > > Try send an email. > > > > Appearance: > > when use non-snoop + ioctl path combination, an issue occurs: > > usage: aplay -D plughw:0,0 44100_S16_LE_2c.wav Ubuntu, it > > performance intermittently noise; > > > > Root case: > > Function set_pages_array_wc marked the PTE of the original address; Function set_memory_wc marked the PTE both original and after vmaped address. > > [cid:image002.png@01D428FF.92621520] > > However, cache or not in transmission process is based on the PTE of the later, and the set_pages_array_wc did't mark it, so a cache inconsistency issue occurs. > > > > Related bug: > > SHA1 ID: 9ddf1aeb2134e72275c97a2c6ff2e3eb04f2f27a > > OK, so did you confirm that dropping the SG buffer support "fixes" the noise problem on your system? The easiest test would be like: > > --- a/sound/pci/hda/hda_controller.c > +++ b/sound/pci/hda/hda_controller.c > @@ -770,7 +770,7 @@ int snd_hda_attach_pcm_stream(struct hda_bus > *_bus, struct hda_codec *codec, size = CONFIG_SND_HDA_PREALLOC_SIZE * > 1024; if (size > MAX_PREALLOC_SIZE) size = MAX_PREALLOC_SIZE; > -snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, > +snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > chip->card->dev, > size, MAX_PREALLOC_SIZE); > return 0; > > > thanks, > > Takashi > > > 保密声明： > 本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 > CONFIDENTIAL NOTE: > This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited. 保密声明：本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 CONFIDENTIAL NOTE: This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

2 1

[alsa-devel] [PATCH] ASoC: soc-pcm: Use delay set in pointer function
by Akshu Agrawal 01 Aug '18

01 Aug '18

There are cases where a pointer function populates runtime->delay, such as: ./sound/pci/hda/hda_controller.c ./sound/soc/intel/atom/sst-mfld-platform-pcm.c Also, in some cases cpu dai used is generic and the pcm driver needs to set delay. This delay was getting lost and was overwritten by delays from codec or cpu dai delay function if exposed. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/soc-pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 98be04b..b1a2bc2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1179,6 +1179,9 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) snd_pcm_sframes_t codec_delay = 0; int i; + /* clearing the previous delay */ + runtime->delay = 0; + for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; @@ -1203,7 +1206,7 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) } delay += codec_delay; - runtime->delay = delay; + runtime->delay += delay; return offset; } -- 1.9.1

5 19

[alsa-devel] [asoc:for-4.19 199/204] sound/soc//meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'; did you mean 'clk_set_rate'?
by kbuild test robot 01 Aug '18

01 Aug '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.19 head: a8e43c21a8a32a3af4abc605b6ebcab039f28e00 commit: c41c2a355b86368608377eaf3df442ec0f342f1e [199/204] ASoC: meson: add tdm output driver config: m68k-allyesconfig (attached as .config) compiler: m68k-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c41c2a355b86368608377eaf3df442ec0f342f1e # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=m68k All errors (new ones prefixed by >>): sound/soc//meson/axg-tdm-interface.c: In function 'axg_tdm_iface_set_lrclk': >> sound/soc//meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'; did you mean 'clk_set_rate'? [-Werror=implicit-function-declaration] ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); ^~~~~~~~~~~~~~~~~~ clk_set_rate cc1: some warnings being treated as errors vim +225 sound/soc//meson/axg-tdm-interface.c d60e4f1e Jerome Brunet 2018-07-17 188 d60e4f1e Jerome Brunet 2018-07-17 189 static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, d60e4f1e Jerome Brunet 2018-07-17 190 struct snd_pcm_hw_params *params) d60e4f1e Jerome Brunet 2018-07-17 191 { d60e4f1e Jerome Brunet 2018-07-17 192 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); d60e4f1e Jerome Brunet 2018-07-17 193 unsigned int ratio_num; d60e4f1e Jerome Brunet 2018-07-17 194 int ret; d60e4f1e Jerome Brunet 2018-07-17 195 d60e4f1e Jerome Brunet 2018-07-17 196 ret = clk_set_rate(iface->lrclk, params_rate(params)); d60e4f1e Jerome Brunet 2018-07-17 197 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 198 dev_err(dai->dev, "setting sample clock failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 199 return ret; d60e4f1e Jerome Brunet 2018-07-17 200 } d60e4f1e Jerome Brunet 2018-07-17 201 d60e4f1e Jerome Brunet 2018-07-17 202 switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { d60e4f1e Jerome Brunet 2018-07-17 203 case SND_SOC_DAIFMT_I2S: d60e4f1e Jerome Brunet 2018-07-17 204 case SND_SOC_DAIFMT_LEFT_J: d60e4f1e Jerome Brunet 2018-07-17 205 case SND_SOC_DAIFMT_RIGHT_J: d60e4f1e Jerome Brunet 2018-07-17 206 /* 50% duty cycle ratio */ d60e4f1e Jerome Brunet 2018-07-17 207 ratio_num = 1; d60e4f1e Jerome Brunet 2018-07-17 208 break; d60e4f1e Jerome Brunet 2018-07-17 209 d60e4f1e Jerome Brunet 2018-07-17 210 case SND_SOC_DAIFMT_DSP_A: d60e4f1e Jerome Brunet 2018-07-17 211 case SND_SOC_DAIFMT_DSP_B: d60e4f1e Jerome Brunet 2018-07-17 212 /* d60e4f1e Jerome Brunet 2018-07-17 213 * A zero duty cycle ratio will result in setting the mininum d60e4f1e Jerome Brunet 2018-07-17 214 * ratio possible which, for this clock, is 1 cycle of the d60e4f1e Jerome Brunet 2018-07-17 215 * parent bclk clock high and the rest low, This is exactly d60e4f1e Jerome Brunet 2018-07-17 216 * what we want here. d60e4f1e Jerome Brunet 2018-07-17 217 */ d60e4f1e Jerome Brunet 2018-07-17 218 ratio_num = 0; d60e4f1e Jerome Brunet 2018-07-17 219 break; d60e4f1e Jerome Brunet 2018-07-17 220 d60e4f1e Jerome Brunet 2018-07-17 221 default: d60e4f1e Jerome Brunet 2018-07-17 222 return -EINVAL; d60e4f1e Jerome Brunet 2018-07-17 223 } d60e4f1e Jerome Brunet 2018-07-17 224 d60e4f1e Jerome Brunet 2018-07-17 @225 ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); d60e4f1e Jerome Brunet 2018-07-17 226 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 227 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 228 "setting sample clock duty cycle failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 229 return ret; d60e4f1e Jerome Brunet 2018-07-17 230 } d60e4f1e Jerome Brunet 2018-07-17 231 d60e4f1e Jerome Brunet 2018-07-17 232 /* Set sample clock inversion */ d60e4f1e Jerome Brunet 2018-07-17 233 ret = clk_set_phase(iface->lrclk, d60e4f1e Jerome Brunet 2018-07-17 234 axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); d60e4f1e Jerome Brunet 2018-07-17 235 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 236 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 237 "setting sample clock phase failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 238 return ret; d60e4f1e Jerome Brunet 2018-07-17 239 } d60e4f1e Jerome Brunet 2018-07-17 240 d60e4f1e Jerome Brunet 2018-07-17 241 return 0; d60e4f1e Jerome Brunet 2018-07-17 242 } d60e4f1e Jerome Brunet 2018-07-17 243 :::::: The code at line 225 was first introduced by commit :::::: d60e4f1e4be5e2dfb55fb084b119aed094227a35 ASoC: meson: add tdm interface driver :::::: TO: Jerome Brunet <jbrunet(a)baylibre.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

4 4

Re: [alsa-devel] 答复: A bug about cache inconsistency report
by Takashi Iwai 31 Jul '18

31 Jul '18

On Tue, 31 Jul 2018 15:51:06 +0200, Hans Hu(SH-RD) wrote: > > I did the test and Oops report when install driver( both snoop and non-snoop mode) I don't understand. Did my one-liner change the behavior or not? thanks, Takashi > > -----邮件原件----- > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > 发送时间: 31 July 2018 19:59 > 收件人: Hans Hu(SH-RD) <HansHu(a)zhaoxin.com> > 抄送: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org> > 主题: Re: [alsa-devel] A bug about cache inconsistency report > > On Tue, 31 Jul 2018 12:52:04 +0200, > Hans Hu(SH-RD) wrote: > > > > Hi All, > > > > Try send an email. > > > > Appearance: > > when use non-snoop + ioctl path combination, an issue occurs: > > usage: aplay -D plughw:0,0 44100_S16_LE_2c.wav Ubuntu, it performance > > intermittently noise; > > > > Root case: > > Function set_pages_array_wc marked the PTE of the original address; Function set_memory_wc marked the PTE both original and after vmaped address. > > [cid:image002.png@01D428FF.92621520] > > However, cache or not in transmission process is based on the PTE of the later, and the set_pages_array_wc did't mark it, so a cache inconsistency issue occurs. > > > > Related bug: > > SHA1 ID: 9ddf1aeb2134e72275c97a2c6ff2e3eb04f2f27a > > OK, so did you confirm that dropping the SG buffer support "fixes" the noise problem on your system? The easiest test would be like: > > --- a/sound/pci/hda/hda_controller.c > +++ b/sound/pci/hda/hda_controller.c > @@ -770,7 +770,7 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, > size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024; > if (size > MAX_PREALLOC_SIZE) > size = MAX_PREALLOC_SIZE; > -snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, > +snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, > chip->card->dev, > size, MAX_PREALLOC_SIZE); > return 0; > > > thanks, > > Takashi > > > 保密声明： > 本邮件含有保密或专有信息，仅供指定收件人使用。严禁对本邮件或其内容做任何未经授权的查阅、使用、复制或转发。 > CONFIDENTIAL NOTE: > This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

1 0

[alsa-devel] [PATCH 0/4] usb-audio: Add UAC3 Power Domains
by Jorge Sanjuan 31 Jul '18

31 Jul '18

This patchset add support for UAC3 Power Domains. This feature of the USB audio class 3 allows the host to notify the device what it is making use of so power comsumption can be optimized. This proposal implements this feature for Power Domains that include an Input/Output Terminal associated to an audio Streaming interface. This is the main usage of this feature according to the spec. For that reason, the logic for the Power Domain state change has been implemented within the ALSA PCMs logic and the suspend/resume callbacks of the usb_driver. The behaviour would be as follows: * Power Domain State D0: A Power Domain will reach this state only when the audio substream associated to that domain is being used (i,e. Audio playback/capture is happening). * Power Domain State D1: This is the Idle state where the driver is going to always want to be in order to reduce power consumption. * Power Domain State D2: This state is only set when the usb driver asumes the device is not going to be used anymore and hence, it wont care about getting any interrupts from the device. This will only happen when power level is set to "auto" in sysfs so the usb driver gets suspended when the interfaces are not in use. NOTE: The way this has been implemented will always try to put the Power Domain in state D1 if the Power Domain exists so there is not a way a user could disable this feature. It may be worth getting a control exposed to userland that enables/disables this feature (?). Power Domains affecting other units independently are required to be bypassed via a Selector Unit first before the host can change the power state. This sceneario is not covered in this patchset. based on next-20180719 Jorge Sanjuan (4): ALSA: usb-audio: Initial Power Domain support ALSA: usb-audio: AudioStreaming Power Domain parsing ALSA: usb-audio: Operate UAC3 Power Domains in PCM callbacks ALSA: usb-audio: Add UAC3 Power Domains to suspend/resume include/linux/usb/audio-v3.h | 4 ++ sound/usb/Makefile | 1 + sound/usb/card.c | 9 ++++ sound/usb/card.h | 2 + sound/usb/pcm.c | 64 +++++++++++++++++++++-- sound/usb/pcm.h | 2 + sound/usb/power.c | 117 +++++++++++++++++++++++++++++++++++++++++++ sound/usb/power.h | 19 +++++++ sound/usb/stream.c | 70 +++++++++++++++++++++++--- 9 files changed, 277 insertions(+), 11 deletions(-) create mode 100644 sound/usb/power.c -- 2.11.0

5 32

[alsa-devel] A bug about cache inconsistency report
by Hans Hu(SH-RD) 31 Jul '18

31 Jul '18

Hi All, Try send an email. Appearance: when use non-snoop + ioctl path combination, an issue occurs: usage: aplay -D plughw:0,0 44100_S16_LE_2c.wav Ubuntu, it performance intermittently noise; Root case: Function set_pages_array_wc marked the PTE of the original address; Function set_memory_wc marked the PTE both original and after vmaped address. [cid:image002.png@01D428FF.92621520] However, cache or not in transmission process is based on the PTE of the later, and the set_pages_array_wc did't mark it, so a cache inconsistency issue occurs. Related bug: SHA1 ID: 9ddf1aeb2134e72275c97a2c6ff2e3eb04f2f27a Best Regards! Hans Hu ????? ????????????????????????????????????????????????????? CONFIDENTIAL NOTE: This email contains confidential or legally privileged information and is for the sole use of its intended recipient. Any unauthorized review, use, copying or forwarding of this email or the content of this email is strictly prohibited.

2 1

[alsa-devel] More snd_pcm_ioplug_avail_update() questions
by Rob Duncan 31 Jul '18

31 Jul '18

Is it required that all available data has been committed between calls to snd_pcm_ioplug_avail_update() on an IO plugin capture stream? If this is not a requirement then there can still be uncommitted data in the mmap after the current appl_ptr. This data will get overwritten by the transfer call. If it is a requirement then snd_pcm_rate_avail_update() seems to be doing the wrong thing, because it only commits in units of the period size, which means that there could be a partial period left in the mmap. Can someone clarify what the contract is for these calls? Thanks, Rob.

2 7

[alsa-devel] [asoc:for-4.19 199/244] sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'
by kbuild test robot 31 Jul '18

31 Jul '18

Hi Jerome, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.19 head: d8504acca759ae672c6d34d49a33d54ace094cbb commit: c41c2a355b86368608377eaf3df442ec0f342f1e [199/244] ASoC: meson: add tdm output driver config: openrisc-allyesconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c41c2a355b86368608377eaf3df442ec0f342f1e # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/meson/axg-tdm-interface.c: In function 'axg_tdm_iface_set_lrclk': >> sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle' [-Werror=implicit-function-declaration] ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); ^~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/clk_set_duty_cycle +225 sound/soc/meson/axg-tdm-interface.c d60e4f1e Jerome Brunet 2018-07-17 188 d60e4f1e Jerome Brunet 2018-07-17 189 static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, d60e4f1e Jerome Brunet 2018-07-17 190 struct snd_pcm_hw_params *params) d60e4f1e Jerome Brunet 2018-07-17 191 { d60e4f1e Jerome Brunet 2018-07-17 192 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); d60e4f1e Jerome Brunet 2018-07-17 193 unsigned int ratio_num; d60e4f1e Jerome Brunet 2018-07-17 194 int ret; d60e4f1e Jerome Brunet 2018-07-17 195 d60e4f1e Jerome Brunet 2018-07-17 196 ret = clk_set_rate(iface->lrclk, params_rate(params)); d60e4f1e Jerome Brunet 2018-07-17 197 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 198 dev_err(dai->dev, "setting sample clock failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 199 return ret; d60e4f1e Jerome Brunet 2018-07-17 200 } d60e4f1e Jerome Brunet 2018-07-17 201 d60e4f1e Jerome Brunet 2018-07-17 202 switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { d60e4f1e Jerome Brunet 2018-07-17 203 case SND_SOC_DAIFMT_I2S: d60e4f1e Jerome Brunet 2018-07-17 204 case SND_SOC_DAIFMT_LEFT_J: d60e4f1e Jerome Brunet 2018-07-17 205 case SND_SOC_DAIFMT_RIGHT_J: d60e4f1e Jerome Brunet 2018-07-17 206 /* 50% duty cycle ratio */ d60e4f1e Jerome Brunet 2018-07-17 207 ratio_num = 1; d60e4f1e Jerome Brunet 2018-07-17 208 break; d60e4f1e Jerome Brunet 2018-07-17 209 d60e4f1e Jerome Brunet 2018-07-17 210 case SND_SOC_DAIFMT_DSP_A: d60e4f1e Jerome Brunet 2018-07-17 211 case SND_SOC_DAIFMT_DSP_B: d60e4f1e Jerome Brunet 2018-07-17 212 /* d60e4f1e Jerome Brunet 2018-07-17 213 * A zero duty cycle ratio will result in setting the mininum d60e4f1e Jerome Brunet 2018-07-17 214 * ratio possible which, for this clock, is 1 cycle of the d60e4f1e Jerome Brunet 2018-07-17 215 * parent bclk clock high and the rest low, This is exactly d60e4f1e Jerome Brunet 2018-07-17 216 * what we want here. d60e4f1e Jerome Brunet 2018-07-17 217 */ d60e4f1e Jerome Brunet 2018-07-17 218 ratio_num = 0; d60e4f1e Jerome Brunet 2018-07-17 219 break; d60e4f1e Jerome Brunet 2018-07-17 220 d60e4f1e Jerome Brunet 2018-07-17 221 default: d60e4f1e Jerome Brunet 2018-07-17 222 return -EINVAL; d60e4f1e Jerome Brunet 2018-07-17 223 } d60e4f1e Jerome Brunet 2018-07-17 224 d60e4f1e Jerome Brunet 2018-07-17 @225 ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); d60e4f1e Jerome Brunet 2018-07-17 226 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 227 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 228 "setting sample clock duty cycle failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 229 return ret; d60e4f1e Jerome Brunet 2018-07-17 230 } d60e4f1e Jerome Brunet 2018-07-17 231 d60e4f1e Jerome Brunet 2018-07-17 232 /* Set sample clock inversion */ d60e4f1e Jerome Brunet 2018-07-17 233 ret = clk_set_phase(iface->lrclk, d60e4f1e Jerome Brunet 2018-07-17 234 axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); d60e4f1e Jerome Brunet 2018-07-17 235 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 236 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 237 "setting sample clock phase failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 238 return ret; d60e4f1e Jerome Brunet 2018-07-17 239 } d60e4f1e Jerome Brunet 2018-07-17 240 d60e4f1e Jerome Brunet 2018-07-17 241 return 0; d60e4f1e Jerome Brunet 2018-07-17 242 } d60e4f1e Jerome Brunet 2018-07-17 243 :::::: The code at line 225 was first introduced by commit :::::: d60e4f1e4be5e2dfb55fb084b119aed094227a35 ASoC: meson: add tdm interface driver :::::: TO: Jerome Brunet <jbrunet(a)baylibre.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] how could unsubscribe ?
by fulgor fulgor 31 Jul '18

31 Jul '18

how could unsubscribe ?

2 1

[alsa-devel] [PATCH 0/4] ASoC: Tegra30 TDM support
by Jorge Sanjuan 30 Jul '18

30 Jul '18

This patchset adds support for TDM audio on Tegra30 hardware. It adds the DAI's `set_tdm_slot` callback and enables a tegra pcm to have up to 8 channels. It also includes support for other audio formats supported by the Tegra30 HW and fixes a broken macro needed for setting the TDM on the registers. Based on Linux 4.18-rc3 tag. Edward Cragg (4): ASoC: tegra: i2s: Fix typo/broken macro ASoC: tegra: Add a TDM configuration callback ASoC: tegra: Allow 32-bit and 24-bit samples ASoC: tegra: i2s: Add support for more than 2 channels sound/soc/tegra/tegra30_i2s.c | 72 ++++++++++++++++++++++++++++++++++++------- sound/soc/tegra/tegra30_i2s.h | 2 +- 2 files changed, 62 insertions(+), 12 deletions(-) -- 2.11.0

6 19

[alsa-devel] [PATCH 1/2] ALSA: seq: virmidi: Offload the output event processing
by Takashi Iwai 30 Jul '18

30 Jul '18

The virmidi sequencer stuff tries to translate the rawmidi bytes to sequencer events and deliver the packets at trigger callback. The amount of the whole process of these translations and deliveries depends on the incoming rawmidi bytes, and we have no limit for that; this was the cause of a CPU soft lockup that had been reported and fixed recently. Although we've fixed the soft lockup by putting the temporary unlock and cond_resched(), it's rather a quick band aid. In this patch, meanwhile, the event parsing and delivery process is offloaded to a dedicated work, and the trigger callback just kicks it off. It has three merits, at least: - The processing is always done in a sleepable context, which can assure the event delivery with non-atomic flag without hackish is_atomic() usage. - Other relevant codes can be simplified, reducing the lines - It makes me happier Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/seq_virmidi.h | 1 + sound/core/seq/seq_virmidi.c | 100 ++++++++++++++++------------------- 2 files changed, 47 insertions(+), 54 deletions(-) diff --git a/include/sound/seq_virmidi.h b/include/sound/seq_virmidi.h index 695257ae64ac..d488dcfa3a4e 100644 --- a/include/sound/seq_virmidi.h +++ b/include/sound/seq_virmidi.h @@ -41,6 +41,7 @@ struct snd_virmidi { struct snd_seq_event event; struct snd_virmidi_dev *rdev; struct snd_rawmidi_substream *substream; + struct work_struct output_work; }; #define SNDRV_VIRMIDI_SUBSCRIBE (1<<0) diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 8ebbca554e99..67ea5d62cebc 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -154,70 +154,58 @@ static void snd_virmidi_input_trigger(struct snd_rawmidi_substream *substream, i } } -/* - * trigger rawmidi stream for output +/* process rawmidi bytes and send events; + * we need no lock here for vmidi->event since it's handled only in this work */ -static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) +static void snd_vmidi_output_work(struct work_struct *work) { - struct snd_virmidi *vmidi = substream->runtime->private_data; - int count, res; - unsigned char buf[32], *pbuf; - unsigned long flags; - bool check_resched = !in_atomic(); + struct snd_virmidi *vmidi; + struct snd_rawmidi_substream *substream; + unsigned char input; + int ret; + + vmidi = container_of(work, struct snd_virmidi, output_work); + substream = vmidi->substream; + + /* discard the outputs in dispatch mode unless subscribed */ + if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && + !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { + while (!snd_rawmidi_transmit_empty(substream)) + snd_rawmidi_transmit_ack(substream, 1); + return; + } - if (up) { - vmidi->trigger = 1; - if (vmidi->seq_mode == SNDRV_VIRMIDI_SEQ_DISPATCH && - !(vmidi->rdev->flags & SNDRV_VIRMIDI_SUBSCRIBE)) { - while (snd_rawmidi_transmit(substream, buf, - sizeof(buf)) > 0) { - /* ignored */ - } - return; - } - spin_lock_irqsave(&substream->runtime->lock, flags); + while (vmidi->trigger) { + if (snd_rawmidi_transmit(substream, &input, 1) != 1) + break; + if (snd_midi_event_encode_byte(vmidi->parser, input, + &vmidi->event) <= 0) + continue; if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) - goto out; + ret = snd_seq_kernel_client_dispatch(vmidi->client, + &vmidi->event, + false, 0); vmidi->event.type = SNDRV_SEQ_EVENT_NONE; - } - while (1) { - count = __snd_rawmidi_transmit_peek(substream, buf, sizeof(buf)); - if (count <= 0) + if (ret < 0) break; - pbuf = buf; - while (count > 0) { - res = snd_midi_event_encode(vmidi->parser, pbuf, count, &vmidi->event); - if (res < 0) { - snd_midi_event_reset_encode(vmidi->parser); - continue; - } - __snd_rawmidi_transmit_ack(substream, res); - pbuf += res; - count -= res; - if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) - goto out; - vmidi->event.type = SNDRV_SEQ_EVENT_NONE; - } - } - if (!check_resched) - continue; - /* do temporary unlock & cond_resched() for avoiding - * CPU soft lockup, which may happen via a write from - * a huge rawmidi buffer - */ - spin_unlock_irqrestore(&substream->runtime->lock, flags); - cond_resched(); - spin_lock_irqsave(&substream->runtime->lock, flags); } - out: - spin_unlock_irqrestore(&substream->runtime->lock, flags); - } else { - vmidi->trigger = 0; + /* rawmidi input might be huge, allow to have a break */ + cond_resched(); } } +/* + * trigger rawmidi stream for output + */ +static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct snd_virmidi *vmidi = substream->runtime->private_data; + + vmidi->trigger = !!up; + if (up) + queue_work(system_highpri_wq, &vmidi->output_work); +} + /* * open rawmidi handle for input */ @@ -270,6 +258,7 @@ static int snd_virmidi_output_open(struct snd_rawmidi_substream *substream) vmidi->port = rdev->port; snd_virmidi_init_event(vmidi, &vmidi->event); vmidi->rdev = rdev; + INIT_WORK(&vmidi->output_work, snd_vmidi_output_work); runtime->private_data = vmidi; return 0; } @@ -299,6 +288,9 @@ static int snd_virmidi_input_close(struct snd_rawmidi_substream *substream) static int snd_virmidi_output_close(struct snd_rawmidi_substream *substream) { struct snd_virmidi *vmidi = substream->runtime->private_data; + + vmidi->trigger = 0; /* to be sure */ + cancel_work_sync(&vmidi->output_work); snd_midi_event_free(vmidi->parser); substream->runtime->private_data = NULL; kfree(vmidi); -- 2.18.0

1 1

[alsa-devel] [PATCH v2 0/6] arm64: dts: rockchip: add basic dts file for PX30 SoCs
by cl＠rock-chips.com 30 Jul '18

30 Jul '18

From: Liang Chen <cl(a)rock-chips.com> Changes in V2: 1. change the subject to format "dt-bindings: module: ..." for the documents. 2. use new property <num-interpolated-steps> for backlight. 3. remove pinctrl of rk809, because the driver is not ready. Liang Chen (6): dt-bindings: usb: dwc2: add description for px30 dt-bindings: mmc: rockchip-dw-mshc: add description for px30 dt-bindings: sound: rockchip-i2s: add description for px30 dt-bindings: rockchip: grf: add grf and pmugrf description for px30 arm64: dts: rockchip: add core dtsi file for PX30 SoCs arm64: dts: rockchip: add PX30 eavluation board devicetree Documentation/devicetree/bindings/arm/rockchip.txt | 4 + .../devicetree/bindings/mmc/rockchip-dw-mshc.txt | 1 + .../devicetree/bindings/soc/rockchip/grf.txt | 2 + .../devicetree/bindings/sound/rockchip-i2s.txt | 1 + Documentation/devicetree/bindings/usb/dwc2.txt | 1 + arch/arm64/boot/dts/rockchip/Makefile | 1 + arch/arm64/boot/dts/rockchip/px30-evb.dts | 473 +++++ arch/arm64/boot/dts/rockchip/px30.dtsi | 2043 ++++++++++++++++++++ 8 files changed, 2526 insertions(+) create mode 100644 arch/arm64/boot/dts/rockchip/px30-evb.dts create mode 100644 arch/arm64/boot/dts/rockchip/px30.dtsi -- 1.9.1

3 8

[alsa-devel] [PATCH] ASoC: wm8988: fix typo in rate constraints
by Ladislav Michl 30 Jul '18

30 Jul '18

Remove duplicated entry and add missing zero in rate constraints. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- sound/soc/codecs/wm8988.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 62200117444b..6e52c6a8bab3 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -522,7 +522,7 @@ static inline int get_coeff(int mclk, int rate) /* The set of rates we can generate from the above for each SYSCLK */ static const unsigned int rates_12288[] = { - 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, + 8000, 12000, 16000, 24000, 32000, 48000, 96000, }; static const struct snd_pcm_hw_constraint_list constraints_12288 = { @@ -540,7 +540,7 @@ static const struct snd_pcm_hw_constraint_list constraints_112896 = { }; static const unsigned int rates_12[] = { - 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 41100, 48000, 48000, 88235, 96000, }; -- 2.18.0

2 1

[alsa-devel] [PATCH] ALSA: memalloc: Fix missing PAGE_SIZE definition
by Takashi Iwai 30 Jul '18

30 Jul '18

The recent fix moved the inline snd_sgbuf_aligned_pages() outside the ifdef, and this triggered a build error on some architectures due to the undefined PAGE_SIZE, as spotted by 0day bot. Fix it by adding the missing header inclusion. Fixes: 4cae99d9b530 ("ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally") Reported-by: kbuild test robot <lkp(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/memalloc.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index c669900e6cbe..67561b997915 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -24,6 +24,8 @@ #ifndef __SOUND_MEMALLOC_H #define __SOUND_MEMALLOC_H +#include <asm/page.h> + struct device; /* -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: ak4642: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> As original license mentioned, it is GPL-2.0 in SPDX. Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL". See ${LINUX}/include/linux/module.h "GPL" [GNU Public License v2 or later] "GPL v2" [GNU Public License v2] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 26 +++++++++++--------------- 1 file changed, 11 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 6050559..3532370 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -1,17 +1,13 @@ -/* - * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Kuninori Morimoto <morimoto.kuninori(a)renesas.com> - * - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * 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. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Kuninori Morimoto <morimoto.kuninori(a)renesas.com> +// +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood /* ** CAUTION ** * @@ -709,4 +705,4 @@ module_i2c_driver(ak4642_i2c_driver); MODULE_DESCRIPTION("Soc AK4642 driver"); MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori(a)renesas.com>"); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: ak4554: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> As original license mentioned, it is GPL-2.0 in SPDX. Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL". See ${LINUX}/include/linux/module.h "GPL" [GNU Public License v2 or later] "GPL v2" [GNU Public License v2] Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4554.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index b7ee1340..2fa83a1 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -1,13 +1,8 @@ -/* - * ak4554.c - * - * Copyright (C) 2013 Renesas Solutions Corp. - * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ +// SPDX-License-Identifier: GPL-2.0 +// ak4554.c +// +// Copyright (C) 2013 Renesas Solutions Corp. +// Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> #include <linux/module.h> #include <sound/soc.h> @@ -97,6 +92,6 @@ static struct platform_driver ak4554_driver = { }; module_platform_driver(ak4554_driver); -MODULE_LICENSE("GPL"); +MODULE_LICENSE("GPL v2"); MODULE_DESCRIPTION("SoC AK4554 driver"); MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: da7210: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/da7210.c | 27 +++++++++++---------------- 1 file changed, 11 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a664111..e172913 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1,19 +1,14 @@ -/* - * DA7210 ALSA Soc codec driver - * - * Copyright (c) 2009 Dialog Semiconductor - * Written by David Chen <Dajun.chen(a)diasemi.com> - * - * Copyright (C) 2009 Renesas Solutions Corp. - * Cleanups by Kuninori Morimoto <morimoto.kuninori(a)renesas.com> - * - * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S - * - * This program is free software; you can redistribute it and/or modify it - * under the terms of the GNU General Public License as published by the - * Free Software Foundation; either version 2 of the License, or (at your - * option) any later version. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// DA7210 ALSA Soc codec driver +// +// Copyright (c) 2009 Dialog Semiconductor +// Written by David Chen <Dajun.chen(a)diasemi.com> +// +// Copyright (C) 2009 Renesas Solutions Corp. +// Cleanups by Kuninori Morimoto <morimoto.kuninori(a)renesas.com> +// +// Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S #include <linux/delay.h> #include <linux/i2c.h> -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: ak4613: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 26 +++++++++++--------------- 1 file changed, 11 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 8523ff9..c1181a2 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -1,18 +1,14 @@ -/* - * ak4613.c -- Asahi Kasei ALSA Soc Audio driver - * - * Copyright (C) 2015 Renesas Electronics Corporation - * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * Based on ak4642.c by Kuninori Morimoto - * Based on wm8731.c by Richard Purdie - * Based on ak4535.c by Richard Purdie - * Based on wm8753.c by Liam Girdwood - * - * 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. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// ak4613.c -- Asahi Kasei ALSA Soc Audio driver +// +// Copyright (C) 2015 Renesas Electronics Corporation +// Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> +// +// Based on ak4642.c by Kuninori Morimoto +// Based on wm8731.c by Richard Purdie +// Based on ak4535.c by Richard Purdie +// Based on wm8753.c by Liam Girdwood #include <linux/clk.h> #include <linux/delay.h> -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: sh: Kconfig: convert to SPDX identifiers
by Kuninori Morimoto 30 Jul '18

30 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> By default all files without license information are under the default license of the kernel, which is GPL version 2. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 0ae0800..dc20f0f 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,3 +1,4 @@ +# SPDX-License-Identifier: GPL-2.0 menu "SoC Audio support for Renesas SoCs" depends on SUPERH || ARCH_RENESAS || COMPILE_TEST -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: tegra_alc5632: fix device_node refcounting" to the asoc tree
by Mark Brown 30 Jul '18

30 Jul '18

The patch ASoC: tegra_alc5632: fix device_node refcounting has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4321723648b0abb456f7a9af51bb09a4ec60799d Mon Sep 17 00:00:00 2001 From: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Date: Sat, 28 Jul 2018 00:06:59 +0300 Subject: [PATCH] ASoC: tegra_alc5632: fix device_node refcounting tegra_alc5632_probe() increments reference count of device nodes with of_parse_phandle(), but there is no code decrementing them in the driver. The patch adds of_node_put() to tegra_alc5632_remove() and to error handling paths in the probe. Found by Linux Driver Verification project (linuxtesting.org). Signed-off-by: Alexey Khoroshilov <khoroshilov(a)ispras.ru> Acked-by: Jon Hunter <jonathanh(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/tegra/tegra_alc5632.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 5197d6b18cb6..98d87801d57a 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -190,14 +190,14 @@ static int tegra_alc5632_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -210,6 +210,13 @@ static int tegra_alc5632_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&alc5632->util_data); +err_put_cpu_of_node: + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; err: return ret; } @@ -223,6 +230,12 @@ static int tegra_alc5632_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_alc5632_dai.cpu_of_node); + tegra_alc5632_dai.cpu_of_node = NULL; + tegra_alc5632_dai.platform_of_node = NULL; + of_node_put(tegra_alc5632_dai.codec_of_node); + tegra_alc5632_dai.codec_of_node = NULL; + return 0; } -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: sirf: Fix potential NULL pointer dereference
by Gustavo A. R. Silva 30 Jul '18

30 Jul '18

There is a potential execution path in which function platform_get_resource() returns NULL. If this happens, we will end up having a NULL pointer dereference. Fix this by replacing devm_ioremap with devm_ioremap_resource, which has the NULL check and the memory region request. This code was detected with the help of Coccinelle. Cc: stable(a)vger.kernel.org Fixes: 2bd8d1d5cf89 ("ASoC: sirf: Add audio usp interface driver") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/sirf/sirf-usp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/sirf/sirf-usp.c b/sound/soc/sirf/sirf-usp.c index 77e7dcf..d70fcd4 100644 --- a/sound/soc/sirf/sirf-usp.c +++ b/sound/soc/sirf/sirf-usp.c @@ -370,10 +370,9 @@ static int sirf_usp_pcm_probe(struct platform_device *pdev) platform_set_drvdata(pdev, usp); mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - base = devm_ioremap(&pdev->dev, mem_res->start, - resource_size(mem_res)); - if (base == NULL) - return -ENOMEM; + base = devm_ioremap_resource(&pdev->dev, mem_res); + if (IS_ERR(base)) + return PTR_ERR(base); usp->regmap = devm_regmap_init_mmio(&pdev->dev, base, &sirf_usp_regmap_config); if (IS_ERR(usp->regmap)) -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: uniphier: add support for multichannel output
by Katsuhiro Suzuki 30 Jul '18

30 Jul '18

This patch adds multichannel PCM output support for LD11/LD20. Currently driver tested and supported only 2ch, 6ch, and 8ch. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 78 ++++++++++++++++++++++++++++++++--- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-reg.h | 1 + sound/soc/uniphier/aio.h | 3 ++ 4 files changed, 78 insertions(+), 6 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 638cb3fc5f7b..8b09bbb0f8d0 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -264,6 +264,57 @@ void aio_port_reset(struct uniphier_aio_sub *sub) } } +/** + * aio_port_set_ch - set channels of LPCM + * @sub: the AIO substream pointer, PCM substream only + * @ch : count of channels + * + * Set suitable slot selecting to input/output port block of AIO. + * + * This function may return error if non-PCM substream. + * + * Return: Zero if successful, otherwise a negative value on error. + */ +static int aio_port_set_ch(struct uniphier_aio_sub *sub) +{ + struct regmap *r = sub->aio->chip->regmap; + u32 slotsel_2ch[] = { + 0, 0, 0, 0, 0, + }; + u32 slotsel_multi[] = { + OPORTMXTYSLOTCTR_SLOTSEL_SLOT0, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT1, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT2, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT3, + OPORTMXTYSLOTCTR_SLOTSEL_SLOT4, + }; + u32 mode, *slotsel; + int i; + + switch (params_channels(&sub->params)) { + case 8: + case 6: + mode = OPORTMXTYSLOTCTR_MODE; + slotsel = slotsel_multi; + break; + case 2: + mode = 0; + slotsel = slotsel_2ch; + break; + default: + return -EINVAL; + } + + for (i = 0; i < AUD_MAX_SLOTSEL; i++) { + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_MODE, mode); + regmap_update_bits(r, OPORTMXTYSLOTCTR(sub->swm->oport.map, i), + OPORTMXTYSLOTCTR_SLOTSEL_MASK, slotsel[i]); + } + + return 0; +} + /** * aio_port_set_rate - set sampling rate of LPCM * @sub: the AIO substream pointer, PCM substream only @@ -575,6 +626,10 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, rate = params_rate(params); } + ret = aio_port_set_ch(sub); + if (ret) + return ret; + ret = aio_port_set_rate(sub, rate); if (ret) return ret; @@ -731,15 +786,28 @@ void aio_port_set_volume(struct uniphier_aio_sub *sub, int vol) int aio_if_set_param(struct uniphier_aio_sub *sub, int pass_through) { struct regmap *r = sub->aio->chip->regmap; - u32 v; + u32 memfmt, v; if (sub->swm->dir == PORT_DIR_OUTPUT) { - if (pass_through) + if (pass_through) { v = PBOUTMXCTR0_ENDIAN_0123 | PBOUTMXCTR0_MEMFMT_STREAM; - else - v = PBOUTMXCTR0_ENDIAN_3210 | - PBOUTMXCTR0_MEMFMT_2CH; + } else { + switch (params_channels(&sub->params)) { + case 2: + memfmt = PBOUTMXCTR0_MEMFMT_2CH; + break; + case 6: + memfmt = PBOUTMXCTR0_MEMFMT_6CH; + break; + case 8: + memfmt = PBOUTMXCTR0_MEMFMT_8CH; + break; + default: + return -EINVAL; + } + v = PBOUTMXCTR0_ENDIAN_3210 | memfmt; + } regmap_write(r, PBOUTMXCTR0(sub->swm->oif.map), v); regmap_write(r, PBOUTMXCTR1(sub->swm->oif.map), 0); diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index ab04d3331be9..de962df245ba 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -286,7 +286,7 @@ static struct snd_soc_dai_driver uniphier_aio_dai_ld11[] = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .rates = SNDRV_PCM_RATE_48000, .channels_min = 2, - .channels_max = 2, + .channels_max = 8, }, .ops = &uniphier_aio_i2s_ops, }, diff --git a/sound/soc/uniphier/aio-reg.h b/sound/soc/uniphier/aio-reg.h index 45fdc6ae358a..734395dbcffb 100644 --- a/sound/soc/uniphier/aio-reg.h +++ b/sound/soc/uniphier/aio-reg.h @@ -374,6 +374,7 @@ #define OPORTMXTYVOLGAINSTATUS(n, m) (0x42108 + 0x400 * (n) + 0x20 * (m)) #define OPORTMXTYVOLGAINSTATUS_CUR_MASK GENMASK(15, 0) #define OPORTMXTYSLOTCTR(n, m) (0x42114 + 0x400 * (n) + 0x20 * (m)) +#define OPORTMXTYSLOTCTR_MODE BIT(15) #define OPORTMXTYSLOTCTR_SLOTSEL_MASK GENMASK(11, 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT0 (0x8 << 8) #define OPORTMXTYSLOTCTR_SLOTSEL_SLOT1 (0x9 << 8) diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index aa89c2f6fa24..23a5c3c68658 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -141,6 +141,9 @@ enum IEC61937_PC { #define AUD_MIN_FRAGMENT_SIZE (4 * 1024) #define AUD_MAX_FRAGMENT_SIZE (16 * 1024) +/* max 5 slots, 10 channels, 2 channel in 1 slot */ +#define AUD_MAX_SLOTSEL 5 + /* * This is a selector for virtual register map of AIO. * -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: uniphier: change functions to static
by Katsuhiro Suzuki 30 Jul '18

30 Jul '18

This patch changes some functions that are not used by other objects to static. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 6 +++--- sound/soc/uniphier/aio.h | 3 --- 2 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 8b09bbb0f8d0..9bcba06ba52e 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -327,7 +327,7 @@ static int aio_port_set_ch(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) +static int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -446,7 +446,7 @@ int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_fmt(struct uniphier_aio_sub *sub) +static int aio_port_set_fmt(struct uniphier_aio_sub *sub) { struct regmap *r = sub->aio->chip->regmap; struct device *dev = &sub->aio->chip->pdev->dev; @@ -511,7 +511,7 @@ int aio_port_set_fmt(struct uniphier_aio_sub *sub) * * Return: Zero if successful, otherwise a negative value on error. */ -int aio_port_set_clk(struct uniphier_aio_sub *sub) +static int aio_port_set_clk(struct uniphier_aio_sub *sub) { struct uniphier_aio_chip *chip = sub->aio->chip; struct device *dev = &sub->aio->chip->pdev->dev; diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index 23a5c3c68658..ca6ccbae0ee8 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -325,9 +325,6 @@ int aio_chip_set_pll(struct uniphier_aio_chip *chip, int pll_id, void aio_chip_init(struct uniphier_aio_chip *chip); int aio_init(struct uniphier_aio_sub *sub); void aio_port_reset(struct uniphier_aio_sub *sub); -int aio_port_set_rate(struct uniphier_aio_sub *sub, int rate); -int aio_port_set_fmt(struct uniphier_aio_sub *sub); -int aio_port_set_clk(struct uniphier_aio_sub *sub); int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params); void aio_port_set_enable(struct uniphier_aio_sub *sub, int enable); -- 2.18.0

2 1

[alsa-devel] [PATCH v0 0/3] ASoC: max9867: driver update
by Ladislav Michl 30 Jul '18

30 Jul '18

Hi there, after this patch serie, I'm finally able to use max9867 codec. This brings in a question how was this driver tested and whenever these tests can be repeated with this patchset applied. Anyway, this serie is intentionally marked as v0 as I'm unclear about few things described in comment section of respective patches. I'd be very happy for answers as they are missing in documentation and are not obvious from other drivers code. Thank you in advance, ladis

2 7

Re: [alsa-devel] [PATCH 10/10] ALSA: xen: Use standard pcm_format_to_bits() for ALSA format bits
by Oleksandr Andrushchenko 30 Jul '18

30 Jul '18

Hello, Takashi! > Date: Wed, 25 Jul 2018 23:19:45 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > To: alsa-devel(a)alsa-project.org > Subject: [alsa-devel] [PATCH 10/10] ALSA: xen: Use standard > pcm_format_to_bits() for ALSA format bits > Message-ID: <20180725211945.19222-11-tiwai(a)suse.de> > > The open codes with the bit shift in xen_snd_front_alsa.c give sparse > warnings as the PCM format type is with __bitwise. > There is already a standard macro to get the format bits, so let's use > it instead. > > This fixes sparse warnings like: > sound/xen/xen_snd_front_alsa.c:191:47: warning: restricted snd_pcm_format_t degrades to integer > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > --- > sound/xen/xen_snd_front_alsa.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c > index 5a2bd70a2fa1..129180e17db1 100644 > --- a/sound/xen/xen_snd_front_alsa.c > +++ b/sound/xen/xen_snd_front_alsa.c > @@ -188,7 +188,7 @@ static u64 to_sndif_formats_mask(u64 alsa_formats) > > mask = 0; > for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) > - if (1 << ALSA_SNDIF_FORMATS[i].alsa & alsa_formats) > + if (pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa) & alsa_formats) > mask |= 1 << ALSA_SNDIF_FORMATS[i].sndif; > > return mask; > @@ -202,7 +202,7 @@ static u64 to_alsa_formats_mask(u64 sndif_formats) > mask = 0; > for (i = 0; i < ARRAY_SIZE(ALSA_SNDIF_FORMATS); i++) > if (1 << ALSA_SNDIF_FORMATS[i].sndif & sndif_formats) > - mask |= 1 << ALSA_SNDIF_FORMATS[i].alsa; > + mask |= pcm_format_to_bits(ALSA_SNDIF_FORMATS[i].alsa); > > return mask; > } Thank you for your patch, Reviewed-by: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Thank you, Oleksandr

1 0

[alsa-devel] [PATCH v2] ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally
by Pierre-Louis Bossart 30 Jul '18

30 Jul '18

Make this helper inline function available for all platforms. This helps solve 0-day compilation issues when CONFIG_SND_DMA_SGBUF is not defined. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: add <asm/page.h> to solve compilation issue on ALPHA reported by 0-day include/sound/memalloc.h | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 9c3db3dce32b..67561b997915 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -24,6 +24,8 @@ #ifndef __SOUND_MEMALLOC_H #define __SOUND_MEMALLOC_H +#include <asm/page.h> + struct device; /* @@ -67,6 +69,14 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +/* + * return the pages matching with the given byte size + */ +static inline unsigned int snd_sgbuf_aligned_pages(size_t size) +{ + return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; +} + #ifdef CONFIG_SND_DMA_SGBUF /* * Scatter-Gather generic device pages @@ -90,14 +100,6 @@ struct snd_sg_buf { struct device *dev; }; -/* - * return the pages matching with the given byte size - */ -static inline unsigned int snd_sgbuf_aligned_pages(size_t size) -{ - return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; -} - /* * return the physical address at the corresponding offset */ -- 2.14.1

3 2

[alsa-devel] [PATCH 2/2] ALSA: hda - Sleep for 10ms after entering D3 on Conexant codecs
by Park Ju Hyung 28 Jul '18

28 Jul '18

On rare occasions, we are still noticing that the internal speaker spitting out spurious noises even after adding the problematic codec to the list. Adding a 10ms artificial delay before rebooting fixes the issue entirely. Patch for Realtek codecs also adds the same amount of delay after entering D3. Signed-off-by: Park Ju Hyung <qkrwngud825(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 909a880f5e01..1a8a2d440fbd 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -226,6 +226,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) snd_hda_codec_set_power_to_all(codec, codec->core.afg, AC_PWRST_D3); snd_hda_codec_write(codec, codec->core.afg, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + msleep(10); } static void cx_auto_free(struct hda_codec *codec) -- 2.18.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Turn CX8200 into D3 as well upon reboot
by Park Ju Hyung 28 Jul '18

28 Jul '18

As an equivalent codec with CX20724, CX8200 is also subject to the reboot bug. Late 2017 and 2018 LG Gram and some HP Spectre laptops are known victims to this issue, causing extremely loud noises upon reboot. Now that we know that this bug is subject to multiple codecs, fix the comment as well. Signed-off-by: Park Ju Hyung <qkrwngud825(a)gmail.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_conexant.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index f641c20095f7..909a880f5e01 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -211,6 +211,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; switch (codec->core.vendor_id) { + case 0x14f12008: /* CX8200 */ case 0x14f150f2: /* CX20722 */ case 0x14f150f4: /* CX20724 */ break; @@ -218,7 +219,7 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) return; } - /* Turn the CX20722 codec into D3 to avoid spurious noises + /* Turn the problematic codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, false); -- 2.18.0

2 1

[alsa-devel] Patch "ASoC: topology: Add missing clock gating parameter when parsing hw_configs" has been added to the 4.17-stable tree
by gregkh＠linuxfoundation.org 28 Jul '18

28 Jul '18

This is a note to let you know that I've just added the patch titled ASoC: topology: Add missing clock gating parameter when parsing hw_configs to the 4.17-stable tree which can be found at: http://www.kernel.org/git/?p=linux/kernel/git/stable/stable-queue.git;a=sum… The filename of the patch is: asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch and it can be found in the queue-4.17 subdirectory. If you, or anyone else, feels it should not be added to the stable tree, please let <stable(a)vger.kernel.org> know about it. >From foo@baz Sat Jul 28 10:14:31 CEST 2018 From: Kirill Marinushkin <k.marinushkin(a)gmail.com> Date: Wed, 4 Apr 2018 06:19:38 +0200 Subject: ASoC: topology: Add missing clock gating parameter when parsing hw_configs From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit 933e1c4a667103c4d10ebdc9505a0a6abd8c3fbd ] Clock gating parameter is a part of `dai_fmt`. It is supported by `alsa-lib` when creating a topology binary file, but ignored by kernel when loading this topology file. After applying this commit, the clock gating parameter is not ignored any more. This solution is backwards compatible. The existing behaviour is not broken, because by default the parameter value is 0 and is ignored. snd_soc_tplg_hw_config.clock_gated = 0 => no effect snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. pm_gate_clocks "true" # clock can be gated } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" ] default_hw_conf_id "1" } ~~~~ Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pan Xiuli <xiuli.pan(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 7 ++++++- sound/soc/soc-topology.c | 7 +++++++ 2 files changed, 13 insertions(+), 1 deletion(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -139,6 +139,11 @@ #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) +/* DAI clock gating */ +#define SND_SOC_TPLG_DAI_CLK_GATE_UNDEFINED 0 +#define SND_SOC_TPLG_DAI_CLK_GATE_GATED 1 +#define SND_SOC_TPLG_DAI_CLK_GATE_CONT 2 + /* DAI physical PCM data formats. * Add new formats to the end of the list. */ @@ -324,7 +329,7 @@ struct snd_soc_tplg_hw_config { __le32 size; /* in bytes of this structure */ __le32 id; /* unique ID - - used to match */ __le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */ - __u8 clock_gated; /* 1 if clock can be gated to save power */ + __u8 clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2006,6 +2006,13 @@ static void set_link_hw_format(struct sn link->dai_fmt = hw_config->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + /* clock gating */ + if (hw_config->clock_gated == SND_SOC_TPLG_DAI_CLK_GATE_GATED) + link->dai_fmt |= SND_SOC_DAIFMT_GATED; + else if (hw_config->clock_gated == + SND_SOC_TPLG_DAI_CLK_GATE_CONT) + link->dai_fmt |= SND_SOC_DAIFMT_CONT; + /* clock signal polarity */ invert_bclk = hw_config->invert_bclk; invert_fsync = hw_config->invert_fsync; Patches currently in stable-queue which might be from k.marinushkin(a)gmail.com are queue-4.17/asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch queue-4.17/asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch

1 0

[alsa-devel] Patch "ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()" has been added to the 4.17-stable tree
by gregkh＠linuxfoundation.org 28 Jul '18

28 Jul '18

This is a note to let you know that I've just added the patch titled ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() to the 4.17-stable tree which can be found at: http://www.kernel.org/git/?p=linux/kernel/git/stable/stable-queue.git;a=sum… The filename of the patch is: asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch and it can be found in the queue-4.17 subdirectory. If you, or anyone else, feels it should not be added to the stable tree, please let <stable(a)vger.kernel.org> know about it. >From foo@baz Sat Jul 28 10:14:31 CEST 2018 From: Kirill Marinushkin <k.marinushkin(a)gmail.com> Date: Wed, 4 Apr 2018 06:19:37 +0200 Subject: ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit a941e2fab3207cb0d57dc4ec47b1b12c8ea78b84 ] The values of bclk and fsync are inverted WRT the codec. But the existing solution already works for Broadwell, see the alsa-lib config: `alsa-lib/src/conf/topology/broadwell/broadwell.conf` This commit provides the backwards-compatible solution to fix this misuse. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 16 ++++++++++++++-- sound/soc/soc-topology.c | 12 +++++++----- 2 files changed, 21 insertions(+), 7 deletions(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -160,6 +160,18 @@ #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) +/* DAI topology BCLK parameter + * For the backwards capability, by default codec is bclk master + */ +#define SND_SOC_TPLG_BCLK_CM 0 /* codec is bclk master */ +#define SND_SOC_TPLG_BCLK_CS 1 /* codec is bclk slave */ + +/* DAI topology FSYNC parameter + * For the backwards capability, by default codec is fsync master + */ +#define SND_SOC_TPLG_FSYNC_CM 0 /* codec is fsync master */ +#define SND_SOC_TPLG_FSYNC_CS 1 /* codec is fsync slave */ + /* * Block Header. * This header precedes all object and object arrays below. @@ -315,8 +327,8 @@ struct snd_soc_tplg_hw_config { __u8 clock_gated; /* 1 if clock can be gated to save power */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ - __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ - __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */ + __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* 0 for input, 1 for output */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2019,13 +2019,15 @@ static void set_link_hw_format(struct sn link->dai_fmt |= SND_SOC_DAIFMT_IB_IF; /* clock masters */ - bclk_master = hw_config->bclk_master; - fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + bclk_master = (hw_config->bclk_master == + SND_SOC_TPLG_BCLK_CM); + fsync_master = (hw_config->fsync_master == + SND_SOC_TPLG_FSYNC_CM); + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; Patches currently in stable-queue which might be from k.marinushkin(a)gmail.com are queue-4.17/asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch queue-4.17/asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch

1 0

[alsa-devel] Patch "ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()" has been added to the 4.14-stable tree
by gregkh＠linuxfoundation.org 28 Jul '18

28 Jul '18

This is a note to let you know that I've just added the patch titled ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() to the 4.14-stable tree which can be found at: http://www.kernel.org/git/?p=linux/kernel/git/stable/stable-queue.git;a=sum… The filename of the patch is: asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch and it can be found in the queue-4.14 subdirectory. If you, or anyone else, feels it should not be added to the stable tree, please let <stable(a)vger.kernel.org> know about it. >From foo@baz Sat Jul 28 10:25:26 CEST 2018 From: Kirill Marinushkin <k.marinushkin(a)gmail.com> Date: Wed, 4 Apr 2018 06:19:37 +0200 Subject: ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit a941e2fab3207cb0d57dc4ec47b1b12c8ea78b84 ] The values of bclk and fsync are inverted WRT the codec. But the existing solution already works for Broadwell, see the alsa-lib config: `alsa-lib/src/conf/topology/broadwell/broadwell.conf` This commit provides the backwards-compatible solution to fix this misuse. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 16 ++++++++++++++-- sound/soc/soc-topology.c | 12 +++++++----- 2 files changed, 21 insertions(+), 7 deletions(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -160,6 +160,18 @@ #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) +/* DAI topology BCLK parameter + * For the backwards capability, by default codec is bclk master + */ +#define SND_SOC_TPLG_BCLK_CM 0 /* codec is bclk master */ +#define SND_SOC_TPLG_BCLK_CS 1 /* codec is bclk slave */ + +/* DAI topology FSYNC parameter + * For the backwards capability, by default codec is fsync master + */ +#define SND_SOC_TPLG_FSYNC_CM 0 /* codec is fsync master */ +#define SND_SOC_TPLG_FSYNC_CS 1 /* codec is fsync slave */ + /* * Block Header. * This header precedes all object and object arrays below. @@ -315,8 +327,8 @@ struct snd_soc_tplg_hw_config { __u8 clock_gated; /* 1 if clock can be gated to save power */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ - __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ - __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */ + __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* 0 for input, 1 for output */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2010,13 +2010,15 @@ static void set_link_hw_format(struct sn link->dai_fmt |= SND_SOC_DAIFMT_IB_IF; /* clock masters */ - bclk_master = hw_config->bclk_master; - fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + bclk_master = (hw_config->bclk_master == + SND_SOC_TPLG_BCLK_CM); + fsync_master = (hw_config->fsync_master == + SND_SOC_TPLG_FSYNC_CM); + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; Patches currently in stable-queue which might be from k.marinushkin(a)gmail.com are queue-4.14/asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch queue-4.14/asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch

1 0

[alsa-devel] Patch "ASoC: topology: Add missing clock gating parameter when parsing hw_configs" has been added to the 4.14-stable tree
by gregkh＠linuxfoundation.org 28 Jul '18

28 Jul '18

This is a note to let you know that I've just added the patch titled ASoC: topology: Add missing clock gating parameter when parsing hw_configs to the 4.14-stable tree which can be found at: http://www.kernel.org/git/?p=linux/kernel/git/stable/stable-queue.git;a=sum… The filename of the patch is: asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch and it can be found in the queue-4.14 subdirectory. If you, or anyone else, feels it should not be added to the stable tree, please let <stable(a)vger.kernel.org> know about it. >From foo@baz Sat Jul 28 10:25:26 CEST 2018 From: Kirill Marinushkin <k.marinushkin(a)gmail.com> Date: Wed, 4 Apr 2018 06:19:38 +0200 Subject: ASoC: topology: Add missing clock gating parameter when parsing hw_configs From: Kirill Marinushkin <k.marinushkin(a)gmail.com> [ Upstream commit 933e1c4a667103c4d10ebdc9505a0a6abd8c3fbd ] Clock gating parameter is a part of `dai_fmt`. It is supported by `alsa-lib` when creating a topology binary file, but ignored by kernel when loading this topology file. After applying this commit, the clock gating parameter is not ignored any more. This solution is backwards compatible. The existing behaviour is not broken, because by default the parameter value is 0 and is ignored. snd_soc_tplg_hw_config.clock_gated = 0 => no effect snd_soc_tplg_hw_config.clock_gated = 1 => SND_SOC_DAIFMT_GATED snd_soc_tplg_hw_config.clock_gated = 2 => SND_SOC_DAIFMT_CONT For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is now supported: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. pm_gate_clocks "true" # clock can be gated } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" ] default_hw_conf_id "1" } ~~~~ Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pan Xiuli <xiuli.pan(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <alexander.levin(a)microsoft.com> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- include/uapi/sound/asoc.h | 7 ++++++- sound/soc/soc-topology.c | 7 +++++++ 2 files changed, 13 insertions(+), 1 deletion(-) --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -139,6 +139,11 @@ #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS (1 << 1) #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) +/* DAI clock gating */ +#define SND_SOC_TPLG_DAI_CLK_GATE_UNDEFINED 0 +#define SND_SOC_TPLG_DAI_CLK_GATE_GATED 1 +#define SND_SOC_TPLG_DAI_CLK_GATE_CONT 2 + /* DAI physical PCM data formats. * Add new formats to the end of the list. */ @@ -324,7 +329,7 @@ struct snd_soc_tplg_hw_config { __le32 size; /* in bytes of this structure */ __le32 id; /* unique ID - - used to match */ __le32 fmt; /* SND_SOC_DAI_FORMAT_ format value */ - __u8 clock_gated; /* 1 if clock can be gated to save power */ + __u8 clock_gated; /* SND_SOC_TPLG_DAI_CLK_GATE_ value */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1997,6 +1997,13 @@ static void set_link_hw_format(struct sn link->dai_fmt = hw_config->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + /* clock gating */ + if (hw_config->clock_gated == SND_SOC_TPLG_DAI_CLK_GATE_GATED) + link->dai_fmt |= SND_SOC_DAIFMT_GATED; + else if (hw_config->clock_gated == + SND_SOC_TPLG_DAI_CLK_GATE_CONT) + link->dai_fmt |= SND_SOC_DAIFMT_CONT; + /* clock signal polarity */ invert_bclk = hw_config->invert_bclk; invert_fsync = hw_config->invert_fsync; Patches currently in stable-queue which might be from k.marinushkin(a)gmail.com are queue-4.14/asoc-topology-fix-bclk-and-fsync-inversion-in-set_link_hw_format.patch queue-4.14/asoc-topology-add-missing-clock-gating-parameter-when-parsing-hw_configs.patch

1 0

[alsa-devel] Support for Focusrite 18i20 audio interface
by debian＠callistix.net 27 Jul '18

27 Jul '18

Hi everyone, I own a Focusrite Scarlett 18i20 2nd Gen audio interface. The inputs and outputs are supported by alsa out of the box, with jack all of them are visible and patchable. But there are some functions which are not exposed by alsa, which include: - low latency monitoring, apparently you are supposed to be able to patch any input to any headphone output - selection of sample rates Is someone working on adding this functionality? As an owner of this hardware I would be able to support any dev who's willing to work on this. User manuals: https://us.focusrite.com/downloads?product=Scarlett+18i20 Best regards, vitaminx

1 0

[alsa-devel] [PATCH 00/12] mfd: Add support to WCD9335 Audio Codec
by Srinivas Kandagatla 27 Jul '18

27 Jul '18

Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC. It is integrated in multiple Qualcomm SoCs like: MSM8996, MSM8976, and MSM8956 chipsets. WCD9335 had multiple functional blocks, like: Soundwire controller, interrupt mux, pin controller, Audio codec, MBHC, MAD(Mic activity Detection), Ultrasonic proximity and pen detection, Battery-voltage monitoring and Codec processing engine. Currently this patchset has been only tested with SLIMbus interface due to hardware avaiablity, but it can be easily made to work with both SLIMbus and I2C/I2S. This patchset adds very basic playback and capture support witout much fancy features. New features will be added once the basic support is in. This patchset is tested on top of linux-next on DB820c for both playback and capture paths. This patchset also depends on the SLIMbus Stream apis which is already merged via char-misc tree for 4.19. Some parts of the code has been inherited from Qualcomm andriod kernel, so credits to various authors. WCD9335 can be interfaced via I2S/I2C or SLIMbus. Here is my test branch incase someone want to try this out: https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=wcd9335 Thanks, Srini Srinivas Kandagatla (12): mfd: dt-bindings: Add wcd9335 mfd bindings mfd: wcd9335: add support to wcd9335 core mfd: wcd9335: add wcd irq support ASoC: dt-bindings: add dt bindings for wcd9335 audio codec ASoC: core: add support to snd_soc_dai_get_channel_map() ASoC: wcd9335: add support to wcd9335 codec ASoC: wcd9335: add CLASS-H Controller support ASoC: wcd9335: add basic controls ASoC: wcd9335: add playback dapm widgets ASoC: wcd9335: add capture dapm widgets ASoC: wcd9335: add audio routings ASoC: apq8096: Add support to Analog audio via WCD9335 slim .../devicetree/bindings/mfd/qcom,wcd9335.txt | 130 + .../devicetree/bindings/sound/qcom,wcd9335.txt | 21 + drivers/mfd/Kconfig | 18 + drivers/mfd/Makefile | 4 + drivers/mfd/wcd9335-core.c | 278 ++ drivers/mfd/wcd9335-irq.c | 172 + include/dt-bindings/mfd/wcd9335.h | 43 + include/linux/mfd/wcd9335/registers.h | 580 +++ include/linux/mfd/wcd9335/wcd9335.h | 43 + include/sound/soc-dai.h | 8 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-clsh.c | 1010 ++++ sound/soc/codecs/wcd-clsh.h | 69 + sound/soc/codecs/wcd9335.c | 4981 ++++++++++++++++++++ sound/soc/qcom/apq8096.c | 68 + sound/soc/soc-core.c | 22 + 17 files changed, 7454 insertions(+) create mode 100644 Documentation/devicetree/bindings/mfd/qcom,wcd9335.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd9335.txt create mode 100644 drivers/mfd/wcd9335-core.c create mode 100644 drivers/mfd/wcd9335-irq.c create mode 100644 include/dt-bindings/mfd/wcd9335.h create mode 100644 include/linux/mfd/wcd9335/registers.h create mode 100644 include/linux/mfd/wcd9335/wcd9335.h create mode 100644 sound/soc/codecs/wcd-clsh.c create mode 100644 sound/soc/codecs/wcd-clsh.h create mode 100644 sound/soc/codecs/wcd9335.c -- 2.16.2

3 23

[alsa-devel] [PATCH] sound: pci: cthw20k2: Replace mdelay() with msleep() and usleep_range()
by Jia-Ju Bai 27 Jul '18

27 Jul '18

hw_pll_init(), hw_dac_stop(), hw_dac_start() and hw_adc_init() are never called in atomic context. They call mdelay() to busily wait, which is not necessary. mdelay() can be replaced with msleep(). This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/pci/ctxfi/cthw20k2.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index b866d6b2c923..3c966fafc754 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1316,12 +1316,12 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 4 : 147 - 4); set_field(&pllctl, PLLCTL_RD, 48000 == rsr ? 1 - 1 : 10 - 1); hw_write_20kx(hw, PLL_CTL, pllctl); - mdelay(40); + msleep(40); pllctl = hw_read_20kx(hw, PLL_CTL); set_field(&pllctl, PLLCTL_FD, 48000 == rsr ? 16 - 2 : 147 - 2); hw_write_20kx(hw, PLL_CTL, pllctl); - mdelay(40); + msleep(40); for (i = 0; i < 1000; i++) { pllstat = hw_read_20kx(hw, PLL_STAT); @@ -1584,7 +1584,7 @@ static void hw_dac_stop(struct hw *hw) data = hw_read_20kx(hw, GPIO_DATA); data &= 0xFFFFFFFD; hw_write_20kx(hw, GPIO_DATA, data); - mdelay(10); + usleep_range(10000, 11000); } static void hw_dac_start(struct hw *hw) @@ -1593,7 +1593,7 @@ static void hw_dac_start(struct hw *hw) data = hw_read_20kx(hw, GPIO_DATA); data |= 0x2; hw_write_20kx(hw, GPIO_DATA, data); - mdelay(50); + msleep(50); } static void hw_dac_reset(struct hw *hw) @@ -1864,11 +1864,11 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) hw_write_20kx(hw, GPIO_DATA, data); } - mdelay(10); + usleep_range(10000, 11000); /* Return the ADC to normal operation. */ data |= (0x1 << 15); hw_write_20kx(hw, GPIO_DATA, data); - mdelay(50); + msleep(50); /* I2C write to register offset 0x0B to set ADC LRCLK polarity */ /* invert bit, interface format to I2S, word length to 24-bit, */ -- 2.17.0

2 1

[alsa-devel] [PATCH] sound: pci: cthw20k1: Replace mdelay() with msleep()
by Jia-Ju Bai 27 Jul '18

27 Jul '18

hw_pll_init(), hw_reset_dac() and hw_card_init() are never called in atomic context. They calls mdelay() to busily wait, which is not necessary. mdelay() can be replaced with msleep(). This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/pci/ctxfi/cthw20k1.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 8e6eb9d7984b..6a051a1c3724 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1319,7 +1319,7 @@ static int hw_pll_init(struct hw *hw, unsigned int rsr) break; hw_write_20kx(hw, PLLCTL, pllctl); - mdelay(40); + msleep(40); } if (i >= 3) { dev_alert(hw->card->dev, "PLL initialization failed!!!\n"); @@ -1407,7 +1407,7 @@ static int hw_reset_dac(struct hw *hw) /* To be effective, need to reset the DAC twice. */ for (i = 0; i < 2; i++) { /* set gpio */ - mdelay(100); + msleep(100); gpioorg = (u16)hw_read_20kx(hw, GPIO); gpioorg &= 0xfffd; hw_write_20kx(hw, GPIO, gpioorg); @@ -2030,7 +2030,7 @@ static int hw_card_init(struct hw *hw, struct card_conf *info) hw_write_20kx(hw, GIE, 0); /* Reset all SRC pending interrupts */ hw_write_20kx(hw, SRCIP, 0); - mdelay(30); + msleep(30); /* Detect the card ID and configure GPIO accordingly. */ switch (hw->model) { -- 2.17.0

2 1

[alsa-devel] [PATCH] sound: usb: quirks: Replace mdelay() with msleep() and usleep_range()
by Jia-Ju Bai 27 Jul '18

27 Jul '18

snd_usb_select_mode_quirk(), snd_usb_set_interface_quirk() and snd_usb_ctl_msg_quirk() are never called in atomic context. They call mdelay() to busily wait, which is not necessary. mdelay() can be replaced with msleep() and usleep_range(). This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/usb/quirks.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index acbeb52f6fd6..9eed650c54d4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1181,7 +1181,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, if (err < 0) return err; - mdelay(20); /* Delay needed after setting the interface */ + msleep(20); /* Delay needed after setting the interface */ /* Vendor mode switch cmd is required. */ if (fmt->formats & SNDRV_PCM_FMTBIT_DSD_U32_BE) { @@ -1202,7 +1202,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, return err; } - mdelay(20); + msleep(20); } return 0; } @@ -1249,7 +1249,7 @@ void snd_usb_set_interface_quirk(struct usb_device *dev) switch (USB_ID_VENDOR(chip->usb_id)) { case 0x23ba: /* Playback Design */ case 0x0644: /* TEAC Corp. */ - mdelay(50); + msleep(50); break; } } @@ -1269,7 +1269,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, */ if (USB_ID_VENDOR(chip->usb_id) == 0x23ba && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* * "TEAC Corp." products need a 20ms delay after each @@ -1277,14 +1277,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, */ if (USB_ID_VENDOR(chip->usb_id) == 0x0644 && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* ITF-USB DSD based DACs functionality need a delay * after each class compliant request */ if (is_itf_usb_dsd_dac(chip->usb_id) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + msleep(20); /* Zoom R16/24, Logitech H650e, Jabra 550a needs a tiny delay here, * otherwise requests like get/set frequency return as failed despite @@ -1294,7 +1294,7 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, chip->usb_id == USB_ID(0x046d, 0x0a46) || chip->usb_id == USB_ID(0x0b0e, 0x0349)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(1); + usleep_range(1000, 2000); } /* -- 2.17.0

2 1

[alsa-devel] BUG: soft lockup in snd_virmidi_output_trigger
by Dae R. Jeong 27 Jul '18

27 Jul '18

Reporting the crash: BUG: soft lockup in snd_virmidi_output_trigger This crash has been found in v4.18-rc3 using RaceFuzzer (a modified version of Syzkaller), which we describe more at the end of this report. Note that this bug is previously reported by Syzkaller a few month ago. (https://syzkaller.appspot.com/bug?id=642c927972318775e0ea1b4779ccd8624ce9e6…) Nonetheless, the crash still can be detected. We guess that the crash has not fixed yet. We report the crash log and the attached reproducer with a hope that they are helpful to diagnose the crash and to fix a bug. C Reproducer: https://kiwi.cs.purdue.edu/static/race-fuzzer/repro-dab082.c Kernel config: https://kiwi.cs.purdue.edu/static/race-fuzzer/config-dab082 (Please disregard all code related to "should_hypercall" in the C repro, as this is only for our debugging purposes using our own hypervisor.) Crash log: ================================================================== watchdog: BUG: soft lockup - CPU#1 stuck for 22s! [syz-executor0:24261] Modules linked in: irq event stamp: 6692 hardirqs last enabled at (6691): [<ffffffff84eb1f12>] __raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] hardirqs last enabled at (6691): [<ffffffff84eb1f12>] _raw_spin_unlock_irqrestore+0x62/0x90 kernel/locking/spinlock.c:184 hardirqs last disabled at (6692): [<ffffffff85000905>] interrupt_entry+0xb5/0xf0 arch/x86/entry/entry_64.S:625 softirqs last enabled at (1468): [<ffffffff852005ff>] __do_softirq+0x5ff/0x7f4 kernel/softirq.c:314 softirqs last disabled at (1455): [<ffffffff8130de56>] invoke_softirq kernel/softirq.c:368 [inline] softirqs last disabled at (1455): [<ffffffff8130de56>] irq_exit+0x126/0x130 kernel/softirq.c:408 CPU: 1 PID: 24261 Comm: syz-executor0 Not tainted 4.18.0-rc3 #1 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.8.2-0-g33fbe13 by qemu-project.org 04/01/2014 RIP: 0010:arch_local_irq_restore arch/x86/include/asm/paravirt.h:783 [inline] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] RIP: 0010:_raw_spin_unlock_irqrestore+0x7d/0x90 kernel/locking/spinlock.c:184 Code: 0d c8 ce 16 7b 5b 41 5c 5d c3 e8 3e 0f 52 fc 48 c7 c7 68 b0 71 86 e8 62 4c 8d fc 48 83 3d 42 91 86 01 00 74 0e 48 89 df 57 9d <0f> 1f 44 00 00 eb cd 0f 0b 0f 0b 0f 1f 84 00 00 00 00 00 55 48 89 RSP: 0018:ffff8801b8e4f7e8 EFLAGS: 00000282 ORIG_RAX: ffffffffffffff13 RAX: 0000000000000000 RBX: 0000000000000282 RCX: ffffffff84eb1f1e RDX: dffffc0000000000 RSI: dffffc0000000000 RDI: 0000000000000282 RBP: ffff8801b8e4f7f8 R08: ffffed002573a06a R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: ffff88012b9d0348 R13: ffff8801e6417580 R14: ffff8801b8e4f868 R15: ffff8801e64175a8 FS: 00007f3447a49700(0000) GS:ffff8801f6d00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007f3447a48db8 CR3: 00000001ee58d000 CR4: 00000000000006e0 Call Trace: spin_unlock_irqrestore include/linux/spinlock.h:365 [inline] snd_virmidi_output_trigger+0x37d/0x420 sound/core/seq/seq_virmidi.c:205 snd_rawmidi_output_trigger sound/core/rawmidi.c:150 [inline] snd_rawmidi_kernel_write1+0x383/0x410 sound/core/rawmidi.c:1288 snd_rawmidi_write+0x249/0xa50 sound/core/rawmidi.c:1338 __vfs_write+0xf9/0x5d0 fs/read_write.c:485 vfs_write+0x195/0x380 fs/read_write.c:549 ksys_write+0xdb/0x1d0 fs/read_write.c:598 __do_sys_write fs/read_write.c:610 [inline] __se_sys_write fs/read_write.c:607 [inline] __x64_sys_write+0x43/0x50 fs/read_write.c:607 do_syscall_64+0x182/0x540 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x44b939 Code: 8d 6b fc ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 5b 6b fc ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007f3447a48b48 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 000000000071bfa0 RCX: 000000000044b939 RDX: 000000002000010b RSI: 0000000020000100 RDI: 0000000000000013 RBP: 000000000000b7a8 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007f3447a496d4 R13: 00000000ffffffff R14: 00000000006f7848 R15: 0000000000000000 = About RaceFuzzer RaceFuzzer is a customized version of Syzkaller, specifically tailored to find race condition bugs in the Linux kernel. While we leverage many different technique, the notable feature of RaceFuzzer is in leveraging a custom hypervisor (QEMU/KVM) to interleave the scheduling. In particular, we modified the hypervisor to intentionally stall a per-core execution, which is similar to supporting per-core breakpoint functionality. This allows RaceFuzzer to force the kernel to deterministically trigger racy condition (which may rarely happen in practice due to randomness in scheduling).

3 7

Re: [alsa-devel] 答复: [PATCH 4/4] drm/amdgpu: Add audio component support
by Takashi Iwai 27 Jul '18

27 Jul '18

On Wed, 25 Jul 2018 07:38:37 +0200, Qu, Jim wrote: > > Jim: Just like Alex said, we want driver can get eld info when hotplug in new device. amdgpu driver is a bit difference from radeon driver, it is not a suitable place to call notify() function in *_audio_enable() , since they are not in the hotplug process context like them in radeon driver, but the mode setting. > > IMO, the right place to call notify() function is also in the amdgpu_connector_xxxx_detect() in amdgpu_connector.c. Hm, but by the modesetting, it actually enables / disables the audio as well, no? If so, the notifier is exactly for that purpose. The audio driver needs to know not only about the physical connection but whether the audio can be actually driven. That is, even if the monitor is connected, the audio won't come out if the mode is off. That is equivalent with the unplugged state for the audio driver. The i915 driver deals with the notifier just like the above, so the behavior is intentional. thanks, Takashi > With None-DC driver, the detect() of connectors will to be preformed in drm_helper_hpd_irq_event() when there is device hotplug. However, amdgpu_connector_xxxx_detect() maybe need to be modified to fit your change. > > Thanks > JimQu > ************************************************************************************************************ > --- > drivers/gpu/drm/Kconfig | 1 + > drivers/gpu/drm/amd/amdgpu/Makefile | 2 +- > drivers/gpu/drm/amd/amdgpu/amdgpu.h | 4 + > drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c | 97 +++++++++++++++++++++++ > drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h | 3 + > drivers/gpu/drm/amd/amdgpu/dce_v10_0.c | 6 ++ > drivers/gpu/drm/amd/amdgpu/dce_v11_0.c | 6 ++ > drivers/gpu/drm/amd/amdgpu/dce_v6_0.c | 6 ++ > drivers/gpu/drm/amd/amdgpu/dce_v8_0.c | 6 ++ > 9 files changed, 130 insertions(+), 1 deletion(-) > create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c > > diff --git a/drivers/gpu/drm/Kconfig b/drivers/gpu/drm/Kconfig > index 2c7112ddfed4..fbe7216c5c56 100644 > --- a/drivers/gpu/drm/Kconfig > +++ b/drivers/gpu/drm/Kconfig > @@ -193,6 +193,7 @@ config DRM_AMDGPU > select BACKLIGHT_LCD_SUPPORT > select INTERVAL_TREE > select CHASH > + select SND_HDA_COMPONENT if SND_HDA_CORE > help > Choose this option if you have a recent AMD Radeon graphics card. > > diff --git a/drivers/gpu/drm/amd/amdgpu/Makefile b/drivers/gpu/drm/amd/amdgpu/Makefile > index bfd332c95b61..9c26facddb17 100644 > --- a/drivers/gpu/drm/amd/amdgpu/Makefile > +++ b/drivers/gpu/drm/amd/amdgpu/Makefile > @@ -52,7 +52,7 @@ amdgpu-y += amdgpu_device.o amdgpu_kms.o \ > amdgpu_ucode.o amdgpu_bo_list.o amdgpu_ctx.o amdgpu_sync.o \ > amdgpu_gtt_mgr.o amdgpu_vram_mgr.o amdgpu_virt.o amdgpu_atomfirmware.o \ > amdgpu_queue_mgr.o amdgpu_vf_error.o amdgpu_sched.o amdgpu_debugfs.o \ > - amdgpu_ids.o > + amdgpu_ids.o amdgpu_audio.o > > # add asic specific block > amdgpu-$(CONFIG_DRM_AMDGPU_CIK)+= cik.o cik_ih.o kv_smc.o kv_dpm.o \ > diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu.h b/drivers/gpu/drm/amd/amdgpu/amdgpu.h > index a59c07590cee..203d2584c989 100644 > --- a/drivers/gpu/drm/amd/amdgpu/amdgpu.h > +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu.h > @@ -1957,5 +1957,9 @@ int amdgpu_dm_display_resume(struct amdgpu_device *adev ); > static inline int amdgpu_dm_display_resume(struct amdgpu_device *adev) { return 0; } > #endif > > +int amdgpu_audio_component_init(struct amdgpu_device *adev); > +void amdgpu_audio_component_fini(struct amdgpu_device *adev); > +void amdgpu_audio_eld_notify(struct amdgpu_device *adev, int pin); > + > #include "amdgpu_object.h" > #endif > diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c b/drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c > new file mode 100644 > index 000000000000..39256e2f84b3 > --- /dev/null > +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c > @@ -0,0 +1,97 @@ > +// SPDX-License-Identifier: MIT > + > +#include <linux/component.h> > +#include "amdgpu.h" > + > +static int amdgpu_audio_component_get_eld(struct device *kdev, int port, > + int pipe, bool *enabled, > + unsigned char *buf, int max_bytes) > +{ > + struct drm_device *dev = dev_get_drvdata(kdev); > + struct drm_encoder *encoder; > + struct amdgpu_encoder *amdgpu_encoder; > + struct amdgpu_encoder_atom_dig *dig; > + struct drm_connector *connector; > + int ret = 0; > + > + *enabled = 0; > + list_for_each_entry(encoder, &dev->mode_config.encoder_list, head) { > + amdgpu_encoder = to_amdgpu_encoder(encoder); > + dig = amdgpu_encoder->enc_priv; > + if (!dig || !dig->afmt || !dig->afmt->enabled) > + continue; > + if (!dig->afmt->pin || dig->afmt->pin->id != port) > + continue; > + connector = amdgpu_get_connector_for_encoder(encoder); > + *enabled = !!connector; > + if (connector) { > + ret = drm_eld_size(connector->eld); > + memcpy(buf, connector->eld, min(max_bytes, ret)); > + break; > + } > + } > + > + return ret; > +} > + > +static const struct drm_audio_component_ops amdgpu_audio_component_ops = { > + .get_eld = amdgpu_audio_component_get_eld, > +}; > + > +static int amdgpu_audio_component_bind(struct device *kdev, > + struct device *hda_kdev, void *data) > +{ > + struct drm_device *dev = dev_get_drvdata(kdev); > + struct amdgpu_device *adev = dev->dev_private; > + struct drm_audio_component *acomp = data; > + > + acomp->ops = &amdgpu_audio_component_ops; > + acomp->dev = kdev; > + adev->mode_info.audio.component = acomp; > + return 0; > +} > + > +static void amdgpu_audio_component_unbind(struct device *kdev, > + struct device *hda_kdev, void *data) > +{ > + struct drm_device *dev = dev_get_drvdata(kdev); > + struct amdgpu_device *adev = dev->dev_private; > + struct drm_audio_component *acomp = data; > + > + acomp->ops = NULL; > + acomp->dev = NULL; > + adev->mode_info.audio.component = NULL; > +} > + > +static const struct component_ops amdgpu_audio_component_bind_ops = { > + .bind = amdgpu_audio_component_bind, > + .unbind = amdgpu_audio_component_unbind, > +}; > + > +int amdgpu_audio_component_init(struct amdgpu_device *adev) > +{ > + int err; > + > + err = component_add(adev->dev, &amdgpu_audio_component_bind_ops); > + if (err < 0) > + return err; > + adev->mode_info.audio.component_registered = true; > + return 0; > +} > + > +void amdgpu_audio_component_fini(struct amdgpu_device *adev) > +{ > + if (adev->mode_info.audio.component_registered) { > + component_del(adev->dev, &amdgpu_audio_component_bind_ops); > + adev->mode_info.audio.component_registered = false; > + } > +} > + > +void amdgpu_audio_eld_notify(struct amdgpu_device *adev, int pin) > +{ > + struct drm_audio_component *acomp = adev->mode_info.audio.component; > + > + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > + pin, -1); > +} > diff --git a/drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h b/drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h > index b9e9e8b02fb7..7cb1aa97e522 100644 > --- a/drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h > +++ b/drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h > @@ -39,6 +39,7 @@ > #include <drm/drm_fb_helper.h> > #include <drm/drm_plane_helper.h> > #include <drm/drm_fb_helper.h> > +#include <drm/drm_audio_component.h> > #include <linux/i2c.h> > #include <linux/i2c-algo-bit.h> > #include <linux/hrtimer.h> > @@ -260,6 +261,8 @@ struct amdgpu_audio { > bool enabled; > struct amdgpu_audio_pin pin[AMDGPU_MAX_AFMT_BLOCKS]; > int num_pins; > + bool component_registered; > + struct drm_audio_component *component; > }; > > struct amdgpu_display_funcs { > diff --git a/drivers/gpu/drm/amd/amdgpu/dce_v10_0.c b/drivers/gpu/drm/amd/amdgpu/dce_v10_0.c > index ada241bfeee9..c8471a59930f 100644 > --- a/drivers/gpu/drm/amd/amdgpu/dce_v10_0.c > +++ b/drivers/gpu/drm/amd/amdgpu/dce_v10_0.c > @@ -1392,6 +1392,8 @@ static void dce_v10_0_audio_enable(struct amdgpu_device *adev, > > WREG32_AUDIO_ENDPT(pin->offset, ixAZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL, > enable ? AZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL__AUDIO_ENABLED_MASK : 0); > + > + amdgpu_audio_eld_notify(adev, pin->id); > } > > static const u32 pin_offsets[] = > @@ -1430,6 +1432,8 @@ static int dce_v10_0_audio_init(struct amdgpu_device *adev) > dce_v10_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > } > > + amdgpu_audio_component_init(adev); > + > return 0; > } > > @@ -1443,6 +1447,8 @@ static void dce_v10_0_audio_fini(struct amdgpu_device *adev) > if (!adev->mode_info.audio.enabled) > return; > > + amdgpu_audio_component_fini(adev); > + > for (i = 0; i < adev->mode_info.audio.num_pins; i++) > dce_v10_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > > diff --git a/drivers/gpu/drm/amd/amdgpu/dce_v11_0.c b/drivers/gpu/drm/amd/amdgpu/dce_v11_0.c > index a5b96eac3033..49edb74725a7 100644 > --- a/drivers/gpu/drm/amd/amdgpu/dce_v11_0.c > +++ b/drivers/gpu/drm/amd/amdgpu/dce_v11_0.c > @@ -1418,6 +1418,8 @@ static void dce_v11_0_audio_enable(struct amdgpu_device *adev, > > WREG32_AUDIO_ENDPT(pin->offset, ixAZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL, > enable ? AZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL__AUDIO_ENABLED_MASK : 0); > + > + amdgpu_audio_eld_notify(adev, pin->id); > } > > static const u32 pin_offsets[] = > @@ -1472,6 +1474,8 @@ static int dce_v11_0_audio_init(struct amdgpu_device *adev) > dce_v11_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > } > > + amdgpu_audio_component_init(adev); > + > return 0; > } > > @@ -1485,6 +1489,8 @@ static void dce_v11_0_audio_fini(struct amdgpu_device *adev) > if (!adev->mode_info.audio.enabled) > return; > > + amdgpu_audio_component_fini(adev); > + > for (i = 0; i < adev->mode_info.audio.num_pins; i++) > dce_v11_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > > diff --git a/drivers/gpu/drm/amd/amdgpu/dce_v6_0.c b/drivers/gpu/drm/amd/amdgpu/dce_v6_0.c > index 394cc1e8fe20..2b7468eaacca 100644 > --- a/drivers/gpu/drm/amd/amdgpu/dce_v6_0.c > +++ b/drivers/gpu/drm/amd/amdgpu/dce_v6_0.c > @@ -1297,6 +1297,8 @@ static void dce_v6_0_audio_enable(struct amdgpu_device *adev, > > WREG32_AUDIO_ENDPT(pin->offset, ixAZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL, > enable ? AZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL__AUDIO_ENABLED_MASK : 0); > + > + amdgpu_audio_eld_notify(adev, pin->id); > } > > static const u32 pin_offsets[7] = > @@ -1343,6 +1345,8 @@ static int dce_v6_0_audio_init(struct amdgpu_device *adev) > dce_v6_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > } > > + amdgpu_audio_component_init(adev); > + > return 0; > } > > @@ -1356,6 +1360,8 @@ static void dce_v6_0_audio_fini(struct amdgpu_device *adev) > if (!adev->mode_info.audio.enabled) > return; > > + amdgpu_audio_component_fini(adev); > + > for (i = 0; i < adev->mode_info.audio.num_pins; i++) > dce_v6_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > > diff --git a/drivers/gpu/drm/amd/amdgpu/dce_v8_0.c b/drivers/gpu/drm/amd/amdgpu/dce_v8_0.c > index c9b9ab8f1b05..c2370ea1f64f 100644 > --- a/drivers/gpu/drm/amd/amdgpu/dce_v8_0.c > +++ b/drivers/gpu/drm/amd/amdgpu/dce_v8_0.c > @@ -1348,6 +1348,8 @@ static void dce_v8_0_audio_enable(struct amdgpu_device *adev, > > WREG32_AUDIO_ENDPT(pin->offset, ixAZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL, > enable ? AZALIA_F0_CODEC_PIN_CONTROL_HOT_PLUG_CONTROL__AUDIO_ENABLED_MASK : 0); > + > + amdgpu_audio_eld_notify(adev, pin->id); > } > > static const u32 pin_offsets[7] = > @@ -1395,6 +1397,8 @@ static int dce_v8_0_audio_init(struct amdgpu_device *adev) > dce_v8_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > } > > + amdgpu_audio_component_init(adev); > + > return 0; > } > > @@ -1408,6 +1412,8 @@ static void dce_v8_0_audio_fini(struct amdgpu_device *adev) > if (!adev->mode_info.audio.enabled) > return; > > + amdgpu_audio_component_fini(adev); > + > for (i = 0; i < adev->mode_info.audio.num_pins; i++) > dce_v8_0_audio_enable(adev, &adev->mode_info.audio.pin[i], false); > > -- > 2.18.0 > > _______________________________________________ > amd-gfx mailing list > amd-gfx(a)lists.freedesktop.org > https://lists.freedesktop.org/mailman/listinfo/amd-gfx

2 3

[alsa-devel] [PATCH v7 00/10] soundwire: Add multi link support
by Shreyas NC 27 Jul '18

27 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Fix to avoid duplicate stream state assignment - Avoid incorrect stream release after configuring Master - Fix to avoid bus lock acquired twice - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch Additionally, as suggested by Vinod, the generic fixes are arranged at the top of the series followed by the Documentation patch for multi link support and then the multi link patches. changes in v7: - Fix the git bisect issue pointed by Pierre. Now, the initialization of the m_rt_count and its modification/usage is split into 2 patches. While the declaration and initialization is in one patch, modification of the refcount and usage is in the multilink bankswitch patch. - Re-arrange the patches as suggested by Vinod changes in v6: - Add patch to fix acquiring bus lock twice during master release - In sdw_stream_add_master(), perform multi link check at start of function as suggested by Pierre - Minor comment fixes Sanyog Kale (3): soundwire: Fix acquiring bus lock twice during master release Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances Shreyas NC (6): soundwire: Fix duplicate stream state assignment soundwire: Fix incorrect exit after configuring stream soundwire: Initialize completion for defer messages soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 511 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 489 insertions(+), 137 deletions(-) -- 2.7.4

2 12

[alsa-devel] [PATCH v4 0/8] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 26 Jul '18

26 Jul '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v4 is based on Takashi's topic/drm_audio_component merged on top on Marks' for-next branch - the merge is not included here. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Both the HDaudio and HDMI outputs were tested. Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. TODO in future update: fix the HDMI jack detection which only works after the mixer values are set, which isn't practical for headless devices always connected (this is a problem in the hdac_hdmi codec that was present before this series was submitted) Changes v4: - rebase/update on Takashi's topic/drm_audio_component branch - changes in the HDaudio detection to avoid adding a fake ACPI ID - new Kconfigs to control HDaudio detection Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Pierre-Louis Bossart (2): ASoC: Intel: common: add table for HDA-based platforms ASoC: Intel: Skylake: add option to control HDAudio + DSP usage Rakesh Ughreja (6): ASoC: Intel: Skylake: fix widget handling ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: use HDAudio if ACPI enumeration fails ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers include/sound/soc-acpi-intel-match.h | 6 + sound/pci/hda/hda_bind.c | 12 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 469 ++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 ++ sound/soc/intel/Kconfig | 18 + sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 133 ++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 37 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 174 ++++++++ sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/soc-acpi-intel-hda-match.c | 40 ++ sound/soc/intel/skylake/skl-pcm.c | 70 +++- sound/soc/intel/skylake/skl-topology.c | 3 + sound/soc/intel/skylake/skl.c | 100 ++++- sound/soc/intel/skylake/skl.h | 12 +- 18 files changed, 1093 insertions(+), 26 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-hda-match.c -- 2.14.1

3 22

[alsa-devel] [PATCH] ALSA: memalloc: declare snd_sgbuf_aligned_pages() unconditionally
by Pierre-Louis Bossart 26 Jul '18

26 Jul '18

Make this helper inline function available for all platforms. This helps solve 0-day compilation issues when CONFIG_SND_DMA_SGBUF is not defined. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/memalloc.h | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 9c3db3dce32b..c669900e6cbe 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -67,6 +67,14 @@ struct snd_dma_buffer { void *private_data; /* private for allocator; don't touch */ }; +/* + * return the pages matching with the given byte size + */ +static inline unsigned int snd_sgbuf_aligned_pages(size_t size) +{ + return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; +} + #ifdef CONFIG_SND_DMA_SGBUF /* * Scatter-Gather generic device pages @@ -90,14 +98,6 @@ struct snd_sg_buf { struct device *dev; }; -/* - * return the pages matching with the given byte size - */ -static inline unsigned int snd_sgbuf_aligned_pages(size_t size) -{ - return (size + PAGE_SIZE - 1) >> PAGE_SHIFT; -} - /* * return the physical address at the corresponding offset */ -- 2.14.1

4 7

[alsa-devel] [PATCH 0/6] ASoC: PCM format type conversion fixes
by Takashi Iwai 26 Jul '18

26 Jul '18

Hi, the PCM format is a fixed type with __bitwise, and it needs cast or other trick to deal with integer. This is a series of such fixes for ASoC. Takashi === Takashi Iwai (6): ALSA: pcm: Add snd_mask_set_format() helper for standard usages ASoC: doc: Replace open code with params_set_format() ASoC: intel: Fix snd_pcm_format_t handling ASoC: fsl: Use snd_mask_set_format() ASoC: pcm186x: Declare PCM format with snd_pcm_format_t ASoC: dmaengine: Use standard pcm_format_to_bits() macro Documentation/sound/soc/dpcm.rst | 4 +--- include/sound/pcm_params.h | 10 ++++++++-- sound/soc/codecs/pcm186x.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 4 +--- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 4 ++-- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 4 ++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/soc-generic-dmaengine-pcm.c | 2 +- 14 files changed, 23 insertions(+), 21 deletions(-) -- 2.18.0

2 14

[alsa-devel] [PATCH 0/3] ASoC: meson-axg: fifo and card fixup
by Jerome Brunet 26 Jul '18

26 Jul '18

The purpose of this patchset is to solve the problems which have been raised since the initial axg sound support has been accepted. Jerome Brunet (3): ASoC: meson: use IRQ_RETVAL in the fifo irq handler ASoC: meson: update axg sound card bindings ASoC: meson: align axg card driver with DT bindings documentation .../bindings/sound/amlogic,axg-sound-card.txt | 68 +++++++++---------- sound/soc/meson/axg-card.c | 12 ++-- sound/soc/meson/axg-fifo.c | 2 +- 3 files changed, 41 insertions(+), 41 deletions(-) -- 2.17.1

2 6

[alsa-devel] Applied "ASoC: rsnd: Document R-Car M3-N support" to the asoc tree
by Mark Brown 26 Jul '18

26 Jul '18

The patch ASoC: rsnd: Document R-Car M3-N support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 150a6dc8fc764c1912fa2a46e18d1505dfd3536f Mon Sep 17 00:00:00 2001 From: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Date: Thu, 26 Jul 2018 05:40:15 +0900 Subject: [PATCH] ASoC: rsnd: Document R-Car M3-N support Document support for the sound modules in the Renesas M3-N (r8a77965) SoC. No driver update is needed. Signed-off-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Yoshihiro Kaneko <ykaneko0929(a)gmail.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Simon Horman <horms+renesas(a)verge.net.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index b86d790f630f..9e764270c36b 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -352,6 +352,7 @@ Required properties: - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) - "renesas,rcar_sound-r8a7796" (R-Car M3-W) + - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: zte: Fix incorrect PCM format bit usages
by Takashi Iwai 26 Jul '18

26 Jul '18

zx-tdm driver sets the DAI driver definitions with the format bits wrongly set with SNDRV_PCM_FORMAT_*, instead of SNDRV_PCM_FMTBIT_*. This patch corrects the definitions. Spotted by a sparse warning: sound/soc/zte/zx-tdm.c:363:35: warning: restricted snd_pcm_format_t degrades to integer Fixes: 870e0ddc4345 ("ASoC: zx-tdm: add zte's tdm controller driver") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/zte/zx-tdm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/zte/zx-tdm.c b/sound/soc/zte/zx-tdm.c index dc955272f58b..389272eeba9a 100644 --- a/sound/soc/zte/zx-tdm.c +++ b/sound/soc/zte/zx-tdm.c @@ -144,8 +144,8 @@ static void zx_tdm_rx_dma_en(struct zx_tdm_info *tdm, bool on) #define ZX_TDM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) #define ZX_TDM_FMTBIT \ - (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_MU_LAW | \ - SNDRV_PCM_FORMAT_A_LAW) + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_MU_LAW | \ + SNDRV_PCM_FMTBIT_A_LAW) static int zx_tdm_dai_probe(struct snd_soc_dai *dai) { -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: dmaengine: Fix missing __user prefix in copy_user callback
by Takashi Iwai 26 Jul '18

26 Jul '18

It seems that __user prefix was forgotten to be added to dmaengine_copy_user callback while we refactored the user-copy PCM core. This patch adds the missing prefix, remove the superfluous cast, and add the needed cast (__force is needed for downgrading from user pointer to kernel pointer), too. Spotted by a sparse warning like: sound/soc/soc-generic-dmaengine-pcm.c:397:27: warning: incorrect type in initializer (incompatible argument 4 (different address spaces)) Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 56a541b9ff9e..61027dd0ef91 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -343,7 +343,7 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer( static int dmaengine_copy_user(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, - void *buf, unsigned long bytes) + void __user *buf, unsigned long bytes) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component = @@ -359,18 +359,17 @@ static int dmaengine_copy_user(struct snd_pcm_substream *substream, int ret; if (is_playback) - if (copy_from_user(dma_ptr, (void __user *)buf, bytes)) + if (copy_from_user(dma_ptr, buf, bytes)) return -EFAULT; if (process) { - ret = process(substream, channel, hwoff, - (void __user *)buf, bytes); + ret = process(substream, channel, hwoff, (__force void *)buf, bytes); if (ret < 0) return ret; } if (!is_playback) - if (copy_to_user((void __user *)buf, dma_ptr, bytes)) + if (copy_to_user(buf, dma_ptr, bytes)) return -EFAULT; return 0; -- 2.18.0

2 1

[alsa-devel] [PATCH] ASoC: dapm: Use int for format bit position
by Takashi Iwai 26 Jul '18

26 Jul '18

fmt in snd_soc_dai_link_event() contains the format bit position, not the format bit itself. Hence it can be a simple integer instead of the explicit u64. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 229c12349803..a9d0ca4f92e1 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3662,7 +3662,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; struct snd_pcm_runtime *runtime = NULL; - u64 fmt; + unsigned int fmt; int ret; if (WARN_ON(!config) || -- 2.18.0

2 1

[alsa-devel] [PATCH 00/15] ASoC: meson: add axg audio subsystem support
by Jerome Brunet 26 Jul '18

26 Jul '18

This patchset adds support for the audio subsystem found on Amlogic's axg SoC family. The first SoC of this family is the A113D/X targeted at smart speaker application. The G12a SoC family, Amlogic next-gen STB chipset, will reuse this audio subsystem with a few changes. This patchset implements the DPCM machine driver and the different components it made of. Several components are still missing, such spdif input and pdm, and will submitted later on. The fancier part of this patchset is probably around the TDM. Properly modeling the SoC architecture and its capatilities made things a little more complex than usual (details in the related commit descriptions) If necessary, this patchset could broken into smaller series. For the intial submission, I wanted to give a global view of the subsystem. Hopefully, it will make it a bit easier to understand. Jerome Brunet (15): ASoC: meson: add axg fifos DT binding documentation ASoC: meson: add axg fifo base driver ASoC: meson: add axg frddr driver ASoC: meson: add axg toddr driver ASoC: meson: add axg spdif output DT bindings documentation ASoC: meson: add axg spdif output ASoC: meson: add axg tdm formatters DT bindings documentation ASoC: meson: add axg tdm interface DT bindings documentation ASoC: meson: add tdm formatter base driver ASoC: meson: add tdm interface driver ASoC: meson: add tdm output driver ASoC: meson: add tdm input driver ASoC: export snd_soc_of_get_slot_mask ASoC: meson: add axg sound card DT bindings documentation ASoC: meson: add axg sound card support .../devicetree/bindings/sound/amlogic,axg-fifo.txt | 23 + .../bindings/sound/amlogic,axg-sound-card.txt | 124 ++++ .../bindings/sound/amlogic,axg-spdifout.txt | 20 + .../bindings/sound/amlogic,axg-tdm-formatters.txt | 28 + .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 + include/sound/soc.h | 3 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/meson/Kconfig | 64 ++ sound/soc/meson/Makefile | 21 + sound/soc/meson/axg-card.c | 671 +++++++++++++++++++++ sound/soc/meson/axg-fifo.c | 341 +++++++++++ sound/soc/meson/axg-fifo.h | 80 +++ sound/soc/meson/axg-frddr.c | 141 +++++ sound/soc/meson/axg-spdifout.c | 456 ++++++++++++++ sound/soc/meson/axg-tdm-formatter.c | 381 ++++++++++++ sound/soc/meson/axg-tdm-formatter.h | 39 ++ sound/soc/meson/axg-tdm-interface.c | 542 +++++++++++++++++ sound/soc/meson/axg-tdm.h | 78 +++ sound/soc/meson/axg-tdmin.c | 229 +++++++ sound/soc/meson/axg-tdmout.c | 259 ++++++++ sound/soc/meson/axg-toddr.c | 199 ++++++ sound/soc/soc-core.c | 7 +- 23 files changed, 3727 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt create mode 100644 sound/soc/meson/Kconfig create mode 100644 sound/soc/meson/Makefile create mode 100644 sound/soc/meson/axg-card.c create mode 100644 sound/soc/meson/axg-fifo.c create mode 100644 sound/soc/meson/axg-fifo.h create mode 100644 sound/soc/meson/axg-frddr.c create mode 100644 sound/soc/meson/axg-spdifout.c create mode 100644 sound/soc/meson/axg-tdm-formatter.c create mode 100644 sound/soc/meson/axg-tdm-formatter.h create mode 100644 sound/soc/meson/axg-tdm-interface.c create mode 100644 sound/soc/meson/axg-tdm.h create mode 100644 sound/soc/meson/axg-tdmin.c create mode 100644 sound/soc/meson/axg-tdmout.c create mode 100644 sound/soc/meson/axg-toddr.c -- 2.14.4

4 39

[alsa-devel] Applied "ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002" to the asoc tree
by Mark Brown 26 Jul '18

26 Jul '18

The patch ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002 has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c183fec10ae6b3d1dfb77471d4158e576f6b43ae Mon Sep 17 00:00:00 2001 From: Akshu Agrawal <akshu.agrawal(a)amd.com> Date: Wed, 25 Jul 2018 17:00:59 +0800 Subject: [PATCH] ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002 DA7219 for our platform need to be configured for 1.8V. Hence, we add a volatge regulator with supplies of 1.8V in the machine driver. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 43 ++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf4d1a4..58c1dcb4d255 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f42606e5879e..cd3cf6e691a9 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -32,6 +32,8 @@ #include <linux/clk.h> #include <linux/gpio.h> #include <linux/module.h> +#include <linux/regulator/machine.h> +#include <linux/regulator/driver.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/acpi.h> @@ -320,11 +322,52 @@ static struct snd_soc_card cz_card = { .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"), +}; + +static struct regulator_init_data acp_da7219_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies), + .consumer_supplies = acp_da7219_supplies, +}; + +static struct regulator_config acp_da7219_cfg = { + .init_data = &acp_da7219_data, +}; + +static struct regulator_ops acp_da7219_ops = { +}; + +static struct regulator_desc acp_da7219_desc = { + .name = "reg-fixed-1.8V", + .type = REGULATOR_VOLTAGE, + .owner = THIS_MODULE, + .ops = &acp_da7219_ops, + .fixed_uV = 1800000, /* 1.8V */ + .n_voltages = 1, +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + struct regulator_dev *rdev; + + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + ret); + return -EINVAL; + } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); -- 2.18.0

1 0

[alsa-devel] [PATCH v2] ASoC: AMD: Fix build warning
by Akshu Agrawal 26 Jul '18

26 Jul '18

Fixes sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe': sound/soc/amd/acp-da7219-max98357a.c:367:3: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] dev_err(&pdev->dev, "Failed to register regulator: %d\n", ret); Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: closing braces missing sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index cd3cf6e..8e3275a 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -365,7 +365,7 @@ static int cz_probe(struct platform_device *pdev) &acp_da7219_cfg); if (IS_ERR(rdev)) { dev_err(&pdev->dev, "Failed to register regulator: %d\n", - ret); + (int)PTR_ERR(rdev)); return -EINVAL; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: trident: Suppress gcc string warning
by Takashi Iwai 26 Jul '18

26 Jul '18

The meddlesome gcc warns about the possible shortname string in trident driver code: sound/pci/trident/trident.c: In function ‘snd_trident_probe’: sound/pci/trident/trident.c:126:2: warning: ‘strcat’ accessing 17 or more bytes at offsets 36 and 20 may overlap 1 byte at offset 36 [-Wrestrict] strcat(card->shortname, card->driver); It happens since gcc calculates the possible string size from card->driver, but this can't be true since we did set the string just before that, and they are much shorter. For shutting it up, use the exactly same string set to card->driver for strcat() to card->shortname, too. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/trident/trident.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index cedf13b64803..2f18b1cdc2cd 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -123,7 +123,7 @@ static int snd_trident_probe(struct pci_dev *pci, } else { strcpy(card->shortname, "Trident "); } - strcat(card->shortname, card->driver); + strcat(card->shortname, str); sprintf(card->longname, "%s PCI Audio at 0x%lx, irq %d", card->shortname, trident->port, trident->irq); -- 2.18.0

1 0

[alsa-devel] [PATCH 1/2] ASoC: tegra: improve goto error label
by Marcel Ziswiler 26 Jul '18

26 Jul '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> While the two error labels "err" and "err_clk_put" goto the same place it is rather confusing that the earlier one is certainly used later again. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- sound/soc/tegra/tegra20_ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index affad46bf188..682ef33afb5f 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -377,7 +377,7 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ret = clk_prepare_enable(ac97->clk_ac97); if (ret) { dev_err(&pdev->dev, "clk_enable failed: %d\n", ret); - goto err; + goto err_clk_put; } ret = snd_soc_set_ac97_ops(&tegra20_ac97_ops); -- 2.14.4

5 12

[alsa-devel] [PATCH] Added support for Encore mDSD USB DAC
by Jeff Crukley 26 Jul '18

26 Jul '18

From: JCruk <jcrukley(a)gmail.com> This patch adds native DSD playback support for the Encore mDSD USB DAC by specifying the vendor and product ID's Signed-off-by: Jeff Crukley <jcrukley(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc0..6fbafce 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1373,6 +1373,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: AMD: Fix build warning
by Akshu Agrawal 26 Jul '18

26 Jul '18

Fixes sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe': sound/soc/amd/acp-da7219-max98357a.c:367:3: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] dev_err(&pdev->dev, "Failed to register regulator: %d\n", ret); Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index cd3cf6e..48c1d94 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -365,7 +365,7 @@ static int cz_probe(struct platform_device *pdev) &acp_da7219_cfg); if (IS_ERR(rdev)) { dev_err(&pdev->dev, "Failed to register regulator: %d\n", - ret); + (int)PTR_ERR(rdev); return -EINVAL; } -- 1.9.1

1 0

[alsa-devel] [PATCH 00/14] ALSA: Proper endian notations
by Takashi Iwai 26 Jul '18

26 Jul '18

Hi, this is a patchset for correcting the integer types to the endian-wise ones, as caught by sparse. Just cosmetic fixes for catching future bugs, no functional change by itself. Takashi === Takashi Iwai (14): ALSA: ymfpci: Proper endian notations ALSA: trident: Proper endian notations ALSA: hda: Proper endian notations for BDL pointers ALSA: riptide: Properly endian notations ALSA: mixart: Proper endian notations ALSA: lola: Proper endian notations ALSA: intel8x0: Proper endian notations ALSA: intel8x0m: Proper endian notations ALSA: maestro3: Proper endian notations ALSA: echoaudio: Proper endian notations ALSA: bt87x: Proper endian notations ALSA: atiixp: Proper endian notations ALSA: atiixp_modem: Proper endian notations ALSA: sb: Proper endian notations sound/hda/hdac_stream.c | 4 +- sound/isa/sb/sb16_csp.c | 30 +++++------ sound/pci/atiixp.c | 4 +- sound/pci/atiixp_modem.c | 4 +- sound/pci/bt87x.c | 4 +- sound/pci/echoaudio/echoaudio.h | 2 +- sound/pci/echoaudio/echoaudio_3g.c | 14 +++--- sound/pci/echoaudio/echoaudio_dsp.c | 6 +-- sound/pci/echoaudio/echoaudio_dsp.h | 50 +++++++++--------- sound/pci/echoaudio/echoaudio_gml.c | 8 +-- sound/pci/intel8x0.c | 6 +-- sound/pci/intel8x0m.c | 6 +-- sound/pci/lola/lola.c | 4 +- sound/pci/lola/lola.h | 4 +- sound/pci/lola/lola_pcm.c | 8 +-- sound/pci/maestro3.c | 6 +-- sound/pci/mixart/mixart_core.c | 4 +- sound/pci/mixart/mixart_hwdep.c | 42 ++++++++-------- sound/pci/mixart/mixart_hwdep.h | 8 +-- sound/pci/riptide/riptide.c | 8 +-- sound/pci/trident/trident.h | 2 +- sound/pci/trident/trident_main.c | 2 +- sound/pci/ymfpci/ymfpci.h | 78 ++++++++++++++--------------- sound/pci/ymfpci/ymfpci_main.c | 6 +-- 24 files changed, 157 insertions(+), 153 deletions(-) -- 2.18.0

1 14

[alsa-devel] [PATCH 00/10] ALSA: yet more PCM format type conversion fixes
by Takashi Iwai 26 Jul '18

26 Jul '18

Hi, this is a series of similar fixes as the previous patchset for ASoC, covering other ALSA drivers to deal with PCM format type properly. Takashi === Takashi Iwai (10): ALSA: hda: Fix implicit PCM format type conversion ALSA: riptide: Fix PCM format type conversion ALSA: pcm: Fix sparse warning wrt PCM format type ALSA: ad1816a: Fix sparse warning wrt PCM format type ALSA: au88x0: Fix sparse warning wrt PCM format type ALSA: asihpi: Fix PCM format notations ALSA: wss: Fix sparse warning wrt PCM format type ALSA: sb: Fix PCM format bit calculation ALSA: sb: Fix sparse warning wrt PCM format type ALSA: xen: Use standard pcm_format_to_bits() for ALSA format bits include/sound/hdaudio.h | 3 ++- include/sound/sb16_csp.h | 2 +- sound/core/oss/pcm_oss.c | 2 +- sound/hda/hdac_device.c | 2 +- sound/isa/ad1816a/ad1816a_lib.c | 3 ++- sound/isa/sb/sb16_csp.c | 6 +++--- sound/isa/sb/sb16_main.c | 9 ++++++--- sound/isa/wss/wss_lib.c | 2 +- sound/pci/asihpi/asihpi.c | 24 +++++++++++++----------- sound/pci/au88x0/au88x0.h | 2 +- sound/pci/au88x0/au88x0_core.c | 2 +- sound/pci/riptide/riptide.c | 2 +- sound/xen/xen_snd_front_alsa.c | 4 ++-- 13 files changed, 35 insertions(+), 28 deletions(-) -- 2.18.0

1 10

[alsa-devel] [PATCH 00/12] Fixes for sparse warnings and other misc bugs
by Takashi Iwai 26 Jul '18

26 Jul '18

Hi, this is a series of patches to correct bugs spotted by sparse and manual lookups, applied in a wide range of drivers. Takashi === Takashi Iwai (12): ALSA: usb-audio: Fix multiple definitions in AU0828_DEVICE() macro ALSA: msnd: Add missing __iomem annotations ALSA: msnd: Fix the default sample sizes ALSA: hda/ca0132 - Use NULL instead of 0 ALSA: msnd: Use NULL instead of 0 ALSA: hda - Fix a sparse warning about snd_ctl_elem_iface_t ALSA: opl3: Declare common variables properly ALSA: usb-audio: Declare the common variable in header file ALSA: pcm: Add __force to cast in snd_pcm_lib_read/write() ALSA: pcm: Use standard lower_32_bits() and upper_32_bits() ALSA: korg1212: Add __force annotation to cast in user-copy callbacks ALSA: emu10k1: Fix missing __force annotation for user/kernel pointer cast include/sound/pcm.h | 4 ++-- sound/core/oss/pcm_plugin.c | 8 ++++---- sound/drivers/opl3/opl3_drums.c | 2 -- sound/drivers/opl3/opl3_lib.c | 3 +-- sound/drivers/opl3/opl3_midi.c | 4 ---- sound/drivers/opl3/opl3_oss.c | 2 -- sound/drivers/opl3/opl3_synth.c | 1 + sound/drivers/opl3/opl3_voice.h | 4 ++++ sound/isa/msnd/msnd.c | 18 +++++++++--------- sound/isa/msnd/msnd.h | 2 +- sound/isa/msnd/msnd_midi.c | 2 +- sound/isa/msnd/msnd_pinnacle.c | 8 ++++---- sound/pci/emu10k1/emufx.c | 24 ++++++++++++------------ sound/pci/hda/hda_codec.c | 4 ++-- sound/pci/hda/patch_ca0132.c | 3 +-- sound/pci/korg1212/korg1212.c | 4 ++-- sound/usb/mixer.h | 2 ++ sound/usb/mixer_quirks.c | 2 -- sound/usb/quirks-table.h | 3 ++- 19 files changed, 48 insertions(+), 52 deletions(-) -- 2.18.0

1 12

[alsa-devel] [PATCH 1/3] ALSA: vx222: Fix invalid endian conversions
by Takashi Iwai 25 Jul '18

25 Jul '18

The endian conversions used in vx2_dma_read() and vx2_dma_write() are superfluous and even wrong on big-endian machines, as inl() and outl() already do conversions. Kill them. Spotted by sparse, a warning like: sound/pci/vx222/vx222_ops.c:278:30: warning: incorrect type in argument 1 (different base types) Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/vx222/vx222_ops.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index d4298af6d3ee..c0d0bf44f365 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -275,7 +275,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) { - outl(cpu_to_le32(*addr), port); + outl(*addr, port); addr++; } addr = (u32 *)runtime->dma_area; @@ -285,7 +285,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) { - outl(cpu_to_le32(*addr), port); + outl(*addr, port); addr++; } @@ -313,7 +313,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; length > 0; length--) - *addr++ = le32_to_cpu(inl(port)); + *addr++ = inl(port); addr = (u32 *)runtime->dma_area; pipe->hw_ptr = 0; } @@ -321,7 +321,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ for (; count > 0; count--) - *addr++ = le32_to_cpu(inl(port)); + *addr++ = inl(port); vx2_release_pseudo_dma(chip); } -- 2.18.0

1 2

[alsa-devel] [PATCH] ALSA: seq: Fix poll() error return
by Takashi Iwai 25 Jul '18

25 Jul '18

The sanity checks in ALSA sequencer and OSS sequencer emulation codes return falsely -ENXIO from poll callback. They should be EPOLLERR instead. This was caught thanks to the recent change to the return value. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss.c | 2 +- sound/core/seq/seq_clientmgr.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index 5f64d0d88320..e1f44fc86885 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -203,7 +203,7 @@ odev_poll(struct file *file, poll_table * wait) struct seq_oss_devinfo *dp; dp = file->private_data; if (snd_BUG_ON(!dp)) - return -ENXIO; + return EPOLLERR; return snd_seq_oss_poll(dp, file, wait); } diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 56ca78423040..6fd4b074b206 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1101,7 +1101,7 @@ static __poll_t snd_seq_poll(struct file *file, poll_table * wait) /* check client structures are in place */ if (snd_BUG_ON(!client)) - return -ENXIO; + return EPOLLERR; if ((snd_seq_file_flags(file) & SNDRV_SEQ_LFLG_INPUT) && client->data.user.fifo) { -- 2.18.0

1 0

[alsa-devel] Need some help building alsa-utils
by Jamey Kirby 25 Jul '18

25 Jul '18

I have been working with alsa for a while. We use it to archive content from our radio station. I am using arecord. Build problem: 1) I cloned alsa-utils from github 2) Spent two hours on Google trying to figure out how to build the darn thing. 3) I am running Ubuntu Server 18.0 with build-essentials installed. 4) I get errors that I do not know how to fix when I try to setup ./configure. 5) I just need to make a build. What do I need to do? Feature: I am adding a feature. In the loop for capture(), when a new file is opened, I want to close the old file (when doing continuous recording). I added a new CLI parameter -K. When this is set, the capture() loop will close the old file before opening the new file. This is an issue for me for two reasons: 1) When will i run out of file handles? I capture one hour files continuously. After 10 hours, I see that all the capture files are still open? 2) I need to be notified when the file is closed so that I can do post-processing of rht capture. I've tried inotifywait in a script, and in another script I tried lsof. lsof works great, but the files are open,. See, the main issue is that I need to know when each file completes so that I can post process. Here is my post process script outline: #!/bin/bash for f in $HOME/vboxshare/jkirby/newarc/*.raw do filename=$(basename $f) open=$(lsof -t $f | wc -w) if [ $open -lt 1 ] then echo "$filename is closed" echo "Do post processing here" else echo "$filename is open by $open processe(s)" fi done If I could close the file before or after the previous file is closed, I can do my work. Otherwise, I have no way to know if the file is open and being processed or not. So, I've made the changes, I just need to build the darn thing. Thanks for any advice or help. -- Jamey Kirby Disrupting the establishment since 1964 This is a personal email account and as such, emails are not subject to archiving. Nothing else really matters.

1 2

[alsa-devel] [asoc:topic/amd 1/1] sound/soc/amd/acp-da7219-max98357a.c:325:3: warning: 'ret' may be used uninitialized in this function
by kbuild test robot 25 Jul '18

25 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/amd head: 7b5317aa809f833d017196ad4a34e3b3b7dae101 commit: 7b5317aa809f833d017196ad4a34e3b3b7dae101 [1/1] ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002 config: x86_64-randconfig-x002-201829 (attached as .config) compiler: gcc-7 (Debian 7.3.0-16) 7.3.0 reproduce: git checkout 7b5317aa809f833d017196ad4a34e3b3b7dae101 # save the attached .config to linux build tree make ARCH=x86_64 Note: it may well be a FALSE warning. FWIW you are at least aware of it now. http://gcc.gnu.org/wiki/Better_Uninitialized_Warnings All warnings (new ones prefixed by >>): sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe': >> sound/soc/amd/acp-da7219-max98357a.c:325:3: warning: 'ret' may be used uninitialized in this function [-Wmaybe-uninitialized] dev_err(&pdev->dev, "Failed to register regulator: %d\n", ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ret); ~~~~ vim +/ret +325 sound/soc/amd/acp-da7219-max98357a.c 313 314 static int cz_probe(struct platform_device *pdev) 315 { 316 int ret; 317 struct snd_soc_card *card; 318 struct acp_platform_info *machine; 319 struct regulator_dev *rdev; 320 321 acp_da7219_cfg.dev = &pdev->dev; 322 rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, 323 &acp_da7219_cfg); 324 if (IS_ERR(rdev)) { > 325 dev_err(&pdev->dev, "Failed to register regulator: %d\n", 326 ret); 327 return -EINVAL; 328 } 329 330 machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), 331 GFP_KERNEL); 332 if (!machine) 333 return -ENOMEM; 334 card = &cz_card; 335 cz_card.dev = &pdev->dev; 336 platform_set_drvdata(pdev, card); 337 snd_soc_card_set_drvdata(card, machine); 338 ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); 339 if (ret) { 340 dev_err(&pdev->dev, 341 "devm_snd_soc_register_card(%s) failed: %d\n", 342 cz_card.name, ret); 343 return ret; 344 } 345 bt_uart_enable = !device_property_read_bool(&pdev->dev, 346 "bt-pad-enable"); 347 return 0; 348 } 349 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2] ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002
by Akshu Agrawal 25 Jul '18

25 Jul '18

DA7219 for our platform need to be configured for 1.8V. Hence, we add a volatge regulator with supplies of 1.8V in the machine driver. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 43 ++++++++++++++++++++++++++++++++++++ 2 files changed, 44 insertions(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf..58c1dcb 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,7 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f42606e..cd3cf6e 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -32,6 +32,8 @@ #include <linux/clk.h> #include <linux/gpio.h> #include <linux/module.h> +#include <linux/regulator/machine.h> +#include <linux/regulator/driver.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/acpi.h> @@ -320,11 +322,52 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"), +}; + +static struct regulator_init_data acp_da7219_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies), + .consumer_supplies = acp_da7219_supplies, +}; + +static struct regulator_config acp_da7219_cfg = { + .init_data = &acp_da7219_data, +}; + +static struct regulator_ops acp_da7219_ops = { +}; + +static struct regulator_desc acp_da7219_desc = { + .name = "reg-fixed-1.8V", + .type = REGULATOR_VOLTAGE, + .owner = THIS_MODULE, + .ops = &acp_da7219_ops, + .fixed_uV = 1800000, /* 1.8V */ + .n_voltages = 1, +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + struct regulator_dev *rdev; + + acp_da7219_cfg.dev = &pdev->dev; + rdev = devm_regulator_register(&pdev->dev, &acp_da7219_desc, + &acp_da7219_cfg); + if (IS_ERR(rdev)) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + ret); + return -EINVAL; + } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); -- 1.9.1

2 1

[alsa-devel] [PATCH 1/3] ASoC: rt5651: add Volume to the name of volume control
by Bard Liao 25 Jul '18

25 Jul '18

add Volume to the name of volume control Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5651.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 985852f..fe970ed 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -332,11 +332,11 @@ static const struct snd_kcontrol_new rt5651_snd_controls[] = { RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, 175, 0, dac_vol_tlv), /* IN1/IN2/IN3 Control */ - SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2, + SOC_SINGLE_TLV("IN1 Boost Volume", RT5651_IN1_IN2, RT5651_BST_SFT1, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2, + SOC_SINGLE_TLV("IN2 Boost Volume", RT5651_IN1_IN2, RT5651_BST_SFT2, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN3 Boost", RT5651_IN3, + SOC_SINGLE_TLV("IN3 Boost Volume", RT5651_IN3, RT5651_BST_SFT1, 8, 0, bst_tlv), /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL, -- 2.7.4

2 3

[alsa-devel] [PATCH 0/6] Fix Sparse warnings on sound/soc/intel
by Pierre-Louis Bossart 25 Jul '18

25 Jul '18

Since we found quite a few issues when checking the SOF code with Sparse, I figured I'd run a quick check on the rest of the Intel audio drivers. Some of the issues are trivial but in the end there are a couple of improvements identified by reducing the volume of warnings, so it's worth it overall. If there are concerns about the code readability, I would still recommend merging the last 3-4 patches that are really the interesting ones exposed by Sparse checks. Pierre-Louis Bossart (6): ASoC: Intel: boards: cleanup snd_mask_set usage ASoC: Intel: Haswell: fix endianness handling ASoC: Intel: Skylake: BDL definitions should be __le32 ASoC: Intel: common: make sst_dma functions static ASoC: Intel: Atom: simplify iomem address and casts ASoC: Intel: Atom: fix inversion between __iowrite32 and __ioread32 sound/soc/intel/atom/sst/sst_drv_interface.c | 29 ++++++------ sound/soc/intel/atom/sst/sst_loader.c | 6 +-- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 6 ++- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 6 ++- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/common/sst-firmware.c | 6 +-- sound/soc/intel/haswell/sst-haswell-dsp.c | 53 ++++++++++++---------- sound/soc/intel/skylake/skl-sst-cldma.c | 8 ++-- sound/soc/intel/skylake/skl-sst-cldma.h | 2 +- 16 files changed, 70 insertions(+), 62 deletions(-) -- 2.14.1

3 13

[alsa-devel] Applied "ASoC: Intel: Skylake: fix widget handling" to the asoc tree
by Mark Brown 25 Jul '18

25 Jul '18

The patch ASoC: Intel: Skylake: fix widget handling has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fe65324e3f5205072a2d55ac9c63ec77155fa528 Mon Sep 17 00:00:00 2001 From: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Date: Tue, 24 Jul 2018 19:50:48 -0500 Subject: [PATCH] ASoC: Intel: Skylake: fix widget handling include DAPM Mux and output widgets into the list. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 647e52aecdc3..9845d853644c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -108,6 +108,9 @@ static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w, case snd_soc_dapm_aif_out: case snd_soc_dapm_dai_out: case snd_soc_dapm_switch: + case snd_soc_dapm_output: + case snd_soc_dapm_mux: + return false; default: return true; -- 2.18.0

1 0

[alsa-devel] [PATCH] Added support for Encore mDSD and Topping DX7s XMOS USB DACs
by JCruk 25 Jul '18

25 Jul '18

Signed-off-by: JCruk <jcrukley(a)gmail.com> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc0..7683d33 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1373,6 +1373,8 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ -- 2.7.4

1 0

Re: [alsa-devel] [PATCH 1/3] ASoC: rt5651: add Volume to the name of volume control
by Mark Brown 25 Jul '18

25 Jul '18

On Wed, Jul 25, 2018 at 02:01:21AM +0000, Bard Liao wrote: > > The RT5631 doesn't appear to have any users except Arndale, certainly no > > x86 users? > As far as I know, there is no x86 users using RT5631. OK, I think it's probably safe to apply that one then as I'm not seeing any in tree users except that Arndale expansion.

1 0

[alsa-devel] [PATCH] ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002
by Akshu Agrawal 25 Jul '18

25 Jul '18

DA7219 for our platform need to be configured for 1.8V. Hence, we add a fixed volatge regulator with supplies of 1.8V in the machine driver. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/Kconfig | 2 ++ sound/soc/amd/acp-da7219-max98357a.c | 45 ++++++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 6cbf9cf..c447a51 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -8,6 +8,8 @@ config SND_SOC_AMD_CZ_DA7219MX98357_MACH select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_ADAU7002 + select REGULATOR + select REGULATOR_FIXED_VOLTAGE depends on SND_SOC_AMD_ACP && I2C help This option enables machine driver for DA7219 and MAX9835. diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index f42606e..fdf8972 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -31,7 +31,10 @@ #include <sound/jack.h> #include <linux/clk.h> #include <linux/gpio.h> +#include <linux/platform_device.h> #include <linux/module.h> +#include <linux/regulator/fixed.h> +#include <linux/regulator/machine.h> #include <linux/i2c.h> #include <linux/input.h> #include <linux/acpi.h> @@ -320,11 +323,53 @@ static void cz_dmic_shutdown(struct snd_pcm_substream *substream) .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static struct regulator_consumer_supply acp_da7219_supplies[] = { + REGULATOR_SUPPLY("VDD", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDMIC", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("VDDIO", "i2c-DLGS7219:00"), + REGULATOR_SUPPLY("IOVDD", "ADAU7002:00"), +}; + +static struct regulator_init_data acp_da7219_data = { + .constraints = { + .always_on = 1, + }, + .num_consumer_supplies = ARRAY_SIZE(acp_da7219_supplies), + .consumer_supplies = acp_da7219_supplies, +}; + +static struct fixed_voltage_config acp_da7219 = { + .supply_name = "reg-fixed-1.8V", + .microvolts = 1800000, /* 1.8V */ + .gpio = -EINVAL, + .enabled_at_boot = 1, + .init_data = &acp_da7219_data, +}; + +static struct platform_device acp_da7219_regulator = { + .name = "reg-fixed-voltage", + .id = PLATFORM_DEVID_AUTO, + .dev = { + .platform_data = &acp_da7219, + }, +}; + static int cz_probe(struct platform_device *pdev) { int ret; struct snd_soc_card *card; struct acp_platform_info *machine; + static bool regulators_registered; + + if (!regulators_registered) { + ret = platform_device_register(&acp_da7219_regulator); + if (ret) { + dev_err(&pdev->dev, "Failed to register regulator: %d\n", + ret); + return ret; + } + regulators_registered = true; + } machine = devm_kzalloc(&pdev->dev, sizeof(struct acp_platform_info), GFP_KERNEL); -- 1.9.1

3 4

Re: [alsa-devel] 答复: [PATCH 3/4] drm/radeon: Add audio component support
by Takashi Iwai 25 Jul '18

25 Jul '18

On Wed, 25 Jul 2018 05:32:52 +0200, Qu, Jim wrote: > @@ -269,6 +271,10 @@ static void radeon_audio_enable(struct radeon_device *rdev, > > if (rdev->audio.funcs->enable) > rdev->audio.funcs->enable(rdev, pin, enable_mask); > + > + if (acomp && acomp->audio_ops && acomp->audio_ops->pin_eld_notify) > + acomp->audio_ops->pin_eld_notify(acomp->audio_ops->audio_ptr, > + pin->id, -1); > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> > Jim: radeon_audio_enable() can enable audios and also disable them, so eld noitfy callback should be called when enable_mask is true. It's intentional. The notifier needs to be called at disablement time, too. The audio driver has to follow the HDMI audio disablement, and notifier receives both on and off case. The actual state is inquired via get_eld call by HD-audio followed after the notification. thanks, Takashi

2 1

[alsa-devel] Building alsa-utils, adding changes
by Jamey Kirby 25 Jul '18

25 Jul '18

I have been working with alsa for a while. We use it to archive content from our radio station. I am using arecord. Build problem: 1) I cloned alsa-utils from github 2) Spent two hours on Google trying to figure out how to build the darn thing. 3) I am running Ubuntu Server 18.0 with build-essentials installed. 4) I get errors that I do not know how to fix when I try to setup ./configure. 5) I just need to make a build. What do I need to do? Feature: I am adding a feature. In the loop for capture(), when a new file is opened, I want to close the old file (when doing continuous recording). I added a new CLI parameter -K. When this is set, the capture() loop will close the old file before opening the new file. This is an issue for me for two reasons: 1) When will i run out of file handles? I capture one hour files continuously. After 10 hours, I see that all the capture files are still open? 2) I need to be notified when the file is closed so that I can do post-processing of rht capture. I've tried inotifywait in a script, and in another script I tried lsof. lsof works great, but the files are open,. See, the main issue is that I need to know when each file completes so that I can post process. Here is my post process script outline: #!/bin/bash for f in $HOME/vboxshare/jkirby/newarc/*.raw do filename=$(basename $f) open=$(lsof -t $f | wc -w) if [ $open -lt 1 ] then echo "$filename is closed" echo "Do post processing here" else echo "$filename is open by $open processe(s)" fi done If I could close the file before or after the previous file is closed, I can do my work. Otherwise, I have no way to know if the file is open and being processed or not. So, I've made the changes, I just need to build the darn thing. Thanks for any advice or help. -- Jamey Kirby Disrupting the establishment since 1964 *This is a personal email account and as such, emails are not subject to archiving. Nothing else really matters.*

1 0

[alsa-devel] [PATCH v2] ASoC: da7219: Allow pdata to specify a VDDIO
by Daniel Kurtz 24 Jul '18

24 Jul '18

Some systems do not have software controllable regulators driving the DA7219's supplies, nor can they use device tree to create "always-on fixed regulators" to easily pretend like they do. On these systems the call to devm_regulator_bulk_get() just creates a set of dummy registers. Calling regulator_get_voltage() on a dummy regulator just returns -EINVAL, in which case the DA7219 is always set up to use the default VDDIO voltage range of 2.5-3.6V. Provide a new device property to let such systems specify a different VDDIO if needed (e.g., 1.8V). Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- Changes for v2: - fix to use device_property_read_u32() include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 19 +++++++++++++------ 2 files changed, 15 insertions(+), 6 deletions(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16f2d10ab..16ab125ad4adbf 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + u32 vddio; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d38..9893920b26f41f 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1634,6 +1634,9 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone else pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + if (device_property_read_u32(dev, "dlg,vddio", &of_val32) >= 0) + pdata->vddio = of_val32; + return pdata; } @@ -1717,8 +1720,12 @@ static int da7219_handle_supplies(struct snd_soc_component *component) /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; ret = regulator_get_voltage(vddio); + /* If regulator_get_voltage() fails, try to use vddio from pdata. */ + if (ret < 0 && da7219->pdata) + ret = da7219->pdata->vddio; if (ret < 1200000) - dev_warn(component->dev, "Invalid VDDIO voltage\n"); + dev_warn(component->dev, "Invalid VDDIO voltage: %d mV\n", + ret); else if (ret < 2800000) io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; @@ -1872,6 +1879,11 @@ static int da7219_probe(struct snd_soc_component *component) mutex_init(&da7219->ctrl_lock); mutex_init(&da7219->pll_lock); + /* Handle DT/ACPI/Platform data */ + da7219->pdata = dev_get_platdata(component->dev); + if (!da7219->pdata) + da7219->pdata = da7219_fw_to_pdata(component); + /* Regulator configuration */ ret = da7219_handle_supplies(component); if (ret) @@ -1897,11 +1909,6 @@ static int da7219_probe(struct snd_soc_component *component) break; } - /* Handle DT/ACPI/Platform data */ - da7219->pdata = dev_get_platdata(component->dev); - if (!da7219->pdata) - da7219->pdata = da7219_fw_to_pdata(component); - da7219_handle_pdata(component); /* Check if MCLK provided */ -- 2.18.0.233.g985f88cf7e-goog

3 2

[alsa-devel] [PATCH 1/4] ASoC: fsl-asoc-card: Switch to SPDX identifier
by Fabio Estevam 24 Jul '18

24 Jul '18

Adopt the SPDX license identifier headers to ease license compliance management. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 18 +++++++----------- 1 file changed, 7 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4a6750a..07808c6 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -1,14 +1,10 @@ -/* - * Freescale Generic ASoC Sound Card driver with ASRC - * - * Copyright (C) 2014 Freescale Semiconductor, Inc. - * - * Author: Nicolin Chen <nicoleotsuka(a)gmail.com> - * - * This file is licensed under the terms of the GNU General Public License - * version 2. This program is licensed "as is" without any warranty of any - * kind, whether express or implied. - */ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale Generic ASoC Sound Card driver with ASRC +// +// Copyright (C) 2014 Freescale Semiconductor, Inc. +// +// Author: Nicolin Chen <nicoleotsuka(a)gmail.com> #include <linux/clk.h> #include <linux/i2c.h> -- 2.7.4

2 7

[alsa-devel] Incorrect name of PCM
by Christopher Head 24 Jul '18

24 Jul '18

Hi folks, I have an Asus P8H77-I motherboard. On board is an Intel HDA connected to a Via VT1708S codec. There are a number of analogue ports. In addition to them, on the back panel are an HDMI port (from an integrated GPU) and a TOSLINK port. The motherboard manual isn’t totally clear on whether the HDMI port can also carry sound or only video, but Windows claims to be able to send sound there, so probably yes. I don’t use the integrated GPU, though. Here’s the problem: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: VT1708S Analog [VT1708S Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 2: VT1708S Alt Analog [VT1708S Alt Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: VT1708S Digital [VT1708S Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 OK, a little bit unexpected that there is only one digital device listed rather than two. The codec datasheet says it should have two digital output widgets. Maybe it’s due to the integrated GPU being disabled though. $ aplay -L null Discard all samples (playback) or generate zero samples (capture) default:CARD=PCH HDA Intel PCH, VT1708S Analog Default Audio Device sysdefault:CARD=PCH HDA Intel PCH, VT1708S Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=PCH,DEV=0 HDA Intel PCH, VT1708S Digital HDMI Audio Output hdmi:CARD=NVidia,DEV=0 HDA NVidia, HDMI 0 HDMI Audio Output hdmi:CARD=NVidia,DEV=1 HDA NVidia, HDMI 1 HDMI Audio Output hdmi:CARD=NVidia,DEV=2 HDA NVidia, HDMI 2 HDMI Audio Output hdmi:CARD=NVidia,DEV=3 HDA NVidia, HDMI 3 HDMI Audio Output Why is this wrong? Because when I play to the “hdmi” PCM, at least with the S/PDIF switch turned on in mixer controls, the sound comes out the TOSLINK port! It seems that the PCM naming is wrong. This is all without any configuration file. Everything works fine, just the name is wrong. I don’t know whether turning off the S/PDIF switch results in the sound coming out the HDMI port instead; maybe it does (I don’t have anything that speaks HDMI and can also play sound to test with). Either way, seems like it should be called either “spdif” (if it’s only the TOSLINK port) or some combination of the two (if it’s both and switchable by the mixer switch). If replying, please note that I’m not subscribed. Thanks! -- Christopher Head

3 10

Re: [alsa-devel] [PATCH 1/3] ASoC: rt5651: add Volume to the name of volume control
by Mark Brown 24 Jul '18

24 Jul '18

On Tue, Jul 24, 2018 at 01:49:48AM +0000, Bard Liao wrote: > Please ignore these patches. There is no significant advantage of > adding "Volume" to the control name. To prevent breaking the > configuration files on user space, we should find out all projects > which are using these controls and modify them. It will take time. > So I think it is better to leave it as it is. The RT5631 doesn't appear to have any users except Arndale, certainly no x86 users?

1 0

[alsa-devel] Applied "ASoC: doc: replace codec to component" to the asoc tree
by Mark Brown 24 Jul '18

24 Jul '18

The patch ASoC: doc: replace codec to component has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a3ac7727f3a226b0569431de62a4e9bb4c770b2b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 28 Mar 2018 02:14:22 +0000 Subject: [PATCH] ASoC: doc: replace codec to component Now we can replace Codec to Component. Let's do it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/sound/soc/codec.rst | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/soc/codec.rst b/Documentation/sound/soc/codec.rst index f87612b94812..58f625fe3d39 100644 --- a/Documentation/sound/soc/codec.rst +++ b/Documentation/sound/soc/codec.rst @@ -179,12 +179,12 @@ i.e. static int wm8974_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - u16 mute_reg = snd_soc_read(codec, WM8974_DAC) & 0xffbf; + struct snd_soc_component *component = dai->component; + u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf; if (mute) - snd_soc_write(codec, WM8974_DAC, mute_reg | 0x40); + snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40); else - snd_soc_write(codec, WM8974_DAC, mute_reg); + snd_soc_component_write(component, WM8974_DAC, mute_reg); return 0; } -- 2.18.0

1 0

[alsa-devel] Applied "ASoC: trace: remove snd_soc_codec" to the asoc tree
by Mark Brown 24 Jul '18

24 Jul '18

The patch ASoC: trace: remove snd_soc_codec has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 211b4415ecb254e167837206756dea123342689c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 28 Mar 2018 02:18:13 +0000 Subject: [PATCH] ASoC: trace: remove snd_soc_codec snd_soc_codec is replaced to snd_soc_component, and it is not used in this file. Let's remove it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/trace/events/asoc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index ccd1a3bdff46..40c300fe704d 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -12,7 +12,6 @@ #define DAPM_ARROW(dir) (((dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-") struct snd_soc_jack; -struct snd_soc_codec; struct snd_soc_card; struct snd_soc_dapm_widget; struct snd_soc_dapm_path; -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: meson: axg-spdifout: select SND_PCM_IEC958
by Arnd Bergmann 24 Jul '18

24 Jul '18

When CONFIG_SND_PCM_IEC958 is disabled, we get a link error for the new driver: sound/soc/meson/axg-spdifout.o: In function `axg_spdifout_hw_params': axg-spdifout.c:(.text+0x650): undefined reference to `snd_pcm_create_iec958_consumer_hw_params' The other users use 'select', so we should do the same here. Fixes: 53eb4b7aaa04 ("ASoC: meson: add axg spdif output") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 4cf93c05a982..8af8bc358a90 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -56,6 +56,7 @@ config SND_MESON_AXG_SOUND_CARD config SND_MESON_AXG_SPDIFOUT tristate "Amlogic AXG SPDIF Output Support" + select SND_PCM_IEC958 imply SND_SOC_SPDIF help Select Y or M to add support for SPDIF output serializer embedded -- 2.18.0

3 2

[alsa-devel] [PATCH] ASoC: rt5514: Fix the issue of the delay volume applied
by Oder Chiou 24 Jul '18

24 Jul '18

The patch fixes the issue of the delay volume applied. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 1570b91..dca82dd 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -64,8 +64,8 @@ static const struct reg_sequence rt5514_patch[] = { {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_ADCFED, 0x00000800}, {RT5514_ASRC_IN_CTRL1, 0x00000003}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, }; static const struct reg_default rt5514_reg[] = { @@ -92,10 +92,10 @@ static const struct reg_default rt5514_reg[] = { {RT5514_ASRC_IN_CTRL1, 0x00000003}, {RT5514_DOWNFILTER0_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER0_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER0_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER0_CTRL3, 0x10000352}, {RT5514_DOWNFILTER1_CTRL1, 0x00020c2f}, {RT5514_DOWNFILTER1_CTRL2, 0x00020c2f}, - {RT5514_DOWNFILTER1_CTRL3, 0x10000362}, + {RT5514_DOWNFILTER1_CTRL3, 0x10000352}, {RT5514_ANA_CTRL_LDO10, 0x00028604}, {RT5514_ANA_CTRL_LDO18_16, 0x02000345}, {RT5514_ANA_CTRL_ADC12, 0x0000a2a8}, -- 2.7.4

2 1

[alsa-devel] WARNING in port_delete
by Dae R. Jeong 24 Jul '18

24 Jul '18

Reporting the crash: WARNING in port_delete This crash has been found in v4.18-rc3 using RaceFuzzer (a modified version of Syzkaller), which we descrbie more at the end of this report. Our analysis shows that the race occurs when invoking two close syscalls concurrently. The executed program is as follows: r0 = open("/dev/snd/seq", 0x0, 0x0) ioctl(r0, SNDRV_SEQ_IOCTL_CREATE_PORT, {{0x80}, "706f72943000000000000000000000000000000000000000000000000000000000233f770800000000000000000000000000000000000000001000", 0xffffffffffffffff, 0xfffffffffffffbff}) r1 = openat(AT_FDCWD, "/dev/sequencer2", 0x8402, 0x0) close(r0) close(r1) and two threads executed the program as follows: Thread0 Thread1 open("/dev/snd/seq") ioctl(r0, SNDRV_SEQ_IOCTL_CREATE_PORT) openat("/dev/sequencer2") close(r0) close(r1) (Note that two close() syscalls were exeuted after openat() syscall, and two close() syscalls were executed concurrently.) Crash log: ================================================================== WARNING: CPU: 1 PID: 32519 at /home/daeryong/workspace/race-fuzzer/kernels_repo/kernel_v4.18-rc3/sound/core/seq/seq_ports.c:275 port_delete+0xde/0xf0 sound/core/seq/seq_ports.c:275 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 32519 Comm: syz-executor0 Not tainted 4.18.0-rc3 #1 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS rel-1.8.2-0-g33fbe13 by qemu-project.org 04/01/2014 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x16e/0x22c lib/dump_stack.c:113 panic+0x1a8/0x3a7 kernel/panic.c:184 __warn+0x191/0x1a0 kernel/panic.c:536 report_bug+0x132/0x1b0 lib/bug.c:186 fixup_bug.part.10+0x28/0x50 arch/x86/kernel/traps.c:178 fixup_bug arch/x86/kernel/traps.c:247 [inline] do_error_trap+0x284/0x2c0 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:316 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:992 RIP: 0010:port_delete+0xde/0xf0 sound/core/seq/seq_ports.c:275 Code: 00 00 e8 b5 68 d2 fd 8b 83 58 01 00 00 85 c0 75 1d e8 86 5c ae fd 48 89 df e8 3e 52 d2 fd 31 c0 5b 41 5c 5d c3 e8 72 5c ae fd <0f> 0b eb c8 e8 69 5c ae fd 0f 0b eb da 0f 1f 44 00 00 55 48 89 e5 RSP: 0018:ffff8801d635f900 EFLAGS: 00010216 RAX: 0000000000040000 RBX: ffff8801da196900 RCX: ffffffff839b86ee RDX: 0000000000005e94 RSI: ffffc90002640000 RDI: ffff8801da196978 RBP: ffff8801d635f910 R08: 0000000000000098 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: ffffffff839ba820 R13: ffff8801da196950 R14: ffff8801ed5bc080 R15: ffff8801da196958 snd_seq_delete_port+0x2b0/0x2d0 sound/core/seq/seq_ports.c:303 snd_seq_ioctl_delete_port+0x5b/0xa0 sound/core/seq/seq_clientmgr.c:1325 snd_seq_kernel_client_ctl+0xd4/0xf0 sound/core/seq/seq_clientmgr.c:2361 snd_seq_event_port_detach+0xcb/0x120 sound/core/seq/seq_ports.c:705 delete_port+0x3a/0x70 sound/core/seq/oss/seq_oss_init.c:354 snd_seq_oss_release+0x7c/0x90 sound/core/seq/oss/seq_oss_init.c:433 odev_release+0x40/0x60 sound/core/seq/oss/seq_oss.c:153 __fput+0x234/0x470 fs/file_table.c:209 ____fput+0x15/0x20 fs/file_table.c:243 task_work_run+0x15a/0x1c0 kernel/task_work.c:113 tracehook_notify_resume include/linux/tracehook.h:192 [inline] exit_to_usermode_loop+0x2a3/0x2b0 arch/x86/entry/common.c:166 prepare_exit_to_usermode arch/x86/entry/common.c:197 [inline] syscall_return_slowpath arch/x86/entry/common.c:268 [inline] do_syscall_64+0x485/0x4b0 arch/x86/entry/common.c:293 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x456469 Code: 1d ba fb ff c3 66 2e 0f 1f 84 00 00 00 00 00 66 90 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 0f 83 eb b9 fb ff c3 66 2e 0f 1f 84 00 00 00 00 RSP: 002b:00007fe9a8936b28 EFLAGS: 00000246 ORIG_RAX: 0000000000000003 RAX: 0000000000000000 RBX: 000000000072bfa0 RCX: 0000000000456469 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000014 RBP: 0000000000000054 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007fe9a89376d4 R13: 00000000ffffffff R14: 00000000006f5880 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. = About RaceFuzzer RaceFuzzer is a customized version of Syzkaller, specifically tailored to find race condition bugs in the Linux kernel. While we leverage many different technique, the notable feature of RaceFuzzer is in leveraging a custom hypervisor (QEMU/KVM) to interleave the scheduling. In particular, we modified the hypervisor to intentionally stall a per-core execution, which is similar to supporting per-core breakpoint functionality. This allows RaceFuzzer to force the kernel to deterministically trigger racy condition (which may rarely happen in practice due to randomness in scheduling). RaceFuzzer's C repro always pinpoints two racy syscalls. Since C repro's scheduling synchronization should be performed at the user space, its reproducibility is limited (reproduction may take from 1 second to 10 minutes (or even more), depending on a bug). This is because, while RaceFuzzer precisely interleaves the scheduling at the kernel's instruction level when finding this bug, C repro cannot fully utilize such a feature. Please disregard all code related to "should_hypercall" in the C repro, as this is only for our debugging purposes using our own hypervisor.

3 4

[alsa-devel] [PATCH 0/4] drm_audio_component support for AMD/ATI HDMI codecs
by Takashi Iwai 24 Jul '18

24 Jul '18

Hi, this is a patch set to add the support of drm_audio_component for AMD/ATI HDMI codecs. With the drm_audio_component, the HDMI/DP audio hotplug and ELD read-out can be achieved directly without the hardware access. The best point by that is that it makes the hotplug notification working even during runtime suspend. The support is totally optional and dynamic, hence it still works even if either HD-audio or DRM side isn't patched, and it'll fall back to the existing method. The current patch supports only radeon and a part of amdgpu; the DC support isn't included yet. Takashi === Takashi Iwai (4): ALSA: hda/hdmi: Use single mutex unlock in error paths ALSA: hda/hdmi: Allow audio component for AMD/ATI HDMI drm/radeon: Add audio component support drm/amdgpu: Add audio component support drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/amdgpu/Makefile | 2 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 4 + drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c | 97 ++++++++++ drivers/gpu/drm/amd/amdgpu/amdgpu_mode.h | 3 + drivers/gpu/drm/amd/amdgpu/dce_v10_0.c | 6 + drivers/gpu/drm/amd/amdgpu/dce_v11_0.c | 6 + drivers/gpu/drm/amd/amdgpu/dce_v6_0.c | 6 + drivers/gpu/drm/amd/amdgpu/dce_v8_0.c | 6 + drivers/gpu/drm/radeon/radeon.h | 3 + drivers/gpu/drm/radeon/radeon_audio.c | 79 ++++++++ sound/pci/hda/patch_hdmi.c | 209 +++++++++++++++++----- 12 files changed, 374 insertions(+), 49 deletions(-) create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_audio.c -- 2.18.0

2 6

[alsa-devel] Applied "ASoC: rockchip-i2s: add description for px30" to the asoc tree
by Mark Brown 23 Jul '18

23 Jul '18

The patch ASoC: rockchip-i2s: add description for px30 has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 2f562a4739602d96928de0b040ce46b00d8d3adc Mon Sep 17 00:00:00 2001 From: Liang Chen <cl(a)rock-chips.com> Date: Mon, 23 Jul 2018 17:25:21 +0800 Subject: [PATCH] ASoC: rockchip-i2s: add description for px30 Add "rockchip,px30-i2s", "rockchip,rk3066-i2s" for i2s on px30 platform. Acked-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Liang Chen <cl(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rockchip-i2s.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt index b208a752576c..54aefab71f2c 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-i2s.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.txt @@ -7,6 +7,7 @@ Required properties: - compatible: should be one of the following: - "rockchip,rk3066-i2s": for rk3066 + - "rockchip,px30-i2s", "rockchip,rk3066-i2s": for px30 - "rockchip,rk3036-i2s", "rockchip,rk3066-i2s": for rk3036 - "rockchip,rk3188-i2s", "rockchip,rk3066-i2s": for rk3188 - "rockchip,rk3228-i2s", "rockchip,rk3066-i2s": for rk3228 -- 2.18.0

1 0

[alsa-devel] [PATCH] alsa/usb-audio: Generic DSD detection for Thesycon-based implementations
by Yue Wang 23 Jul '18

23 Jul '18

Thesycon provides solutions to XMOS chips, and has its own device vendor id. In this patch, we use generic method to detect DSD capability of Thesycon-based UAC2 implementations in order to support a wide range of current and future devices. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..06ae3f5adf4b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1443,6 +1443,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, */ switch (USB_ID_VENDOR(chip->usb_id)) { case 0x20b1: /* XMOS based devices */ + case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; -- 2.18.0

2 1

[alsa-devel] [PATCH] alsa/usb-audio: Generic DSD detection for Thesycon-based implementations
by Yue Wang 23 Jul '18

23 Jul '18

Thesycon provides solutions to XMOS chips, and has its own device vendor id. In this patch, we use generic method to detect DSD capability of Thesycon-based UAC2 implementations in order to support a wide range of current and future devices. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..c51e2dee3075 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- 2.18.0

2 2

[alsa-devel] [PATCH] alsa/usb-audio: Topping DX7s quirk for DSD interface
by Yue Wang 23 Jul '18

23 Jul '18

This patch adds the support for native DSD for Topping DX7s DAC. The product description is on http://en.tpdz.net/products_detail/productId=32.html and it's available on https://www.amazon.com/Balanced-Headphone-Amplifier-Topping-2ES9038Q2M/dp/B… The DAC uses XMOS chip but uses a special vendor id and product id. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..c51e2dee3075 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- 2.18.0

2 2

Re: [alsa-devel] [PATCH] alsa/usb-audio: Topping DX7s quirk for DSD interface
by Takashi Iwai 23 Jul '18

23 Jul '18

On Mon, 23 Jul 2018 09:25:34 +0200, <yuleopen(a)gmail.com> wrote: > > From: Yue Wang <yuleopen(a)gmail.com> > > Signed-off-by: Yue Wang <yuleopen(a)gmail.com> Please give more explanation. No commit message already indicates that something is very wrong, no matter what the change is. thanks, Takashi

2 2

[alsa-devel] [PATCH] ALSA: core: fix unsigned int pages overflow when comapred
by He, Bo 23 Jul '18

23 Jul '18

we see the below kernel panic on stress suspend resume test in snd_malloc_sgbuf_pages(), snd_dma_alloc_pages_fallback() alloc chunk maybe larger than the left pages due to the pages alignment, which will cause the pages overflow. while (pages > 0) { ... pages -= chunk; } the patch is change the pages from unsigned int to int to fix the issue. BUG: unable to handle kernel paging request at ffff88000deb4000 IP: [<ffffffff81404fa9>] memset_erms+0x9/0x10 Call Trace: [<ffffffff818f222f>] snd_dma_alloc_pages+0xff/0x210 [<ffffffff818f23af>] snd_dma_alloc_pages_fallback+0x6f/0x90 [<ffffffff818f2b85>] snd_malloc_sgbuf_pages+0x145/0x370 [<ffffffff818f229e>] snd_dma_alloc_pages+0x16e/0x210 [<ffffffffc011930d>] hdac_ext_dma_alloc_pages+0x1d/0x40 [snd_hda_ext_core] [<ffffffffc010729a>] snd_hdac_dsp_prepare+0xca/0x1c0 [snd_hda_core] [<ffffffffc01880f9>] skl_dsp_prepare+0x99/0xf0 [snd_soc_skl] [<ffffffffc0162a7e>] bxt_load_base_firmware+0x9e/0x5c0 [snd_soc_skl_ipc] [<ffffffffc01630ec>] bxt_set_dsp_D0+0x14c/0x300 [snd_soc_skl_ipc] [<ffffffffc015f9c3>] skl_dsp_get_core+0x43/0xd0 [snd_soc_skl_ipc] [<ffffffffc015fa60>] skl_dsp_wake+0x10/0x20 [snd_soc_skl_ipc] [<ffffffffc0188e3e>] skl_resume_dsp+0x7e/0x140 [snd_soc_skl] [<ffffffffc0183c4a>] skl_resume+0xda/0x170 [snd_soc_skl] [<ffffffff81452726>] pci_pm_resume+0x76/0xe0 [<ffffffff816616da>] dpm_run_callback+0x5a/0x180 [<ffffffff81661e3c>] device_resume+0xdc/0x2c0 [<ffffffff81663818>] dpm_resume+0x118/0x310 [<ffffffff81663e11>] dpm_resume_end+0x11/0x20 [<ffffffff810f8bcc>] suspend_devices_and_enter+0x11c/0x2b0 [<ffffffff810f90bd>] pm_suspend+0x35d/0x3d0 [<ffffffff810f78a6>] state_store+0x66/0x90 [<ffffffff813f80e2>] kobj_attr_store+0x12/0x20 [<ffffffff812a37bc>] sysfs_kf_write+0x3c/0x50 [<ffffffff812a2cbd>] kernfs_fop_write+0x11d/0x1a0 [<ffffffff8121dfaa>] __vfs_write+0x3a/0x150 [<ffffffff8121f2b1>] vfs_write+0xb1/0x1a0 [<ffffffff81220898>] SyS_write+0x58/0xc0 [<ffffffff81001fca>] do_syscall_64+0x6a/0xe0 [<ffffffff81b06560>] entry_SYSCALL_64_after_swapgs+0x5d/0xd7 Signed-off-by: he, bo <bo.he(a)intel.com> Signed-off-by: zhang jun <jun.zhang(a)intel.com> --- sound/core/sgbuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 84fffab..33449ee 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -68,7 +68,8 @@ void *snd_malloc_sgbuf_pages(struct device *device, size_t *res_size) { struct snd_sg_buf *sgbuf; - unsigned int i, pages, chunk, maxpages; + unsigned int i, chunk, maxpages; + int pages; struct snd_dma_buffer tmpb; struct snd_sg_page *table; struct page **pgtable; -- 2.7.4

3 7

[alsa-devel] [PATCH v6 00/10] soundwire: Add multi link support
by Sanyog Kale 23 Jul '18

23 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Fix to avoid duplicate stream state assignment - Avoid incorrect stream release after configuring Master - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch - Fix to avoid bus lock acquired twice changes in v6: - Add patch to fix acquiring bus lock twice during master release - In sdw_stream_add_master(), perform multi link check at start of function as suggested by Pierre - Minor comment fixes changes in v5: - Add patch to fix duplicate stream state assignment - fix incorrect exit which was releasing the stream after configuring master - In sdw_release_bus_lock(), release the mutex in reverse order as suggested by Takashi changes in v4: - Added changes in sdw_stream_runtime structure to track Masters added to stream - Add changes to support ml_bankswitch only if both the hardware supports and the stream is handled by multiple masters Sanyog Kale (3): Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances soundwire: Fix acquiring bus lock twice during master release Shreyas NC (6): soundwire: Initialize completion for defer messages soundwire: Fix duplicate stream state assignment soundwire: fix incorrect exit after configuring stream soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 511 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 489 insertions(+), 137 deletions(-) -- 2.13.0

3 14

[alsa-devel] [PATCH] ASoC: wm9712: fix replace codec to component
by Marcel Ziswiler 23 Jul '18

23 Jul '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Since commit 143b44845d87 ("ASoC: wm9712: replace codec to component") "wm9712-codec" got renamed to "wm9712-component", however, this change never got propagated down to the actual board/platform drivers. E.g. on Colibri T20 this lead to the following spew upon boot with sound/touch being broken: [ 2.214121] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.222137] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.344384] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.351885] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.668339] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.675811] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.208408] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.216312] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.235397] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.248938] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 14.970443] ALSA device list: [ 14.996628] No soundcards found. This commit finally fixes this again. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- arch/arm/mach-pxa/tosa.c | 2 +- arch/mips/alchemy/devboards/db1300.c | 2 +- drivers/mfd/wm97xx-core.c | 6 +++--- sound/soc/au1x/db1200.c | 4 ++-- sound/soc/fsl/pcm030-audio-fabric.c | 6 +++--- sound/soc/fsl/phycore-ac97.c | 6 +++--- sound/soc/pxa/e800_wm9712.c | 4 ++-- sound/soc/pxa/em-x270.c | 4 ++-- sound/soc/pxa/palm27x.c | 4 ++-- sound/soc/pxa/tosa.c | 4 ++-- sound/soc/tegra/tegra_wm9712.c | 4 ++-- 11 files changed, 23 insertions(+), 23 deletions(-) diff --git a/arch/arm/mach-pxa/tosa.c b/arch/arm/mach-pxa/tosa.c index cb5cd8e78c94..32011e600496 100644 --- a/arch/arm/mach-pxa/tosa.c +++ b/arch/arm/mach-pxa/tosa.c @@ -848,7 +848,7 @@ static struct platform_device sharpsl_rom_device = { }; static struct platform_device wm9712_device = { - .name = "wm9712-codec", + .name = "wm9712-component", .id = -1, }; diff --git a/arch/mips/alchemy/devboards/db1300.c b/arch/mips/alchemy/devboards/db1300.c index efb318e03e0a..841a27736eb1 100644 --- a/arch/mips/alchemy/devboards/db1300.c +++ b/arch/mips/alchemy/devboards/db1300.c @@ -638,7 +638,7 @@ static struct platform_device db1300_sd0_dev = { /**********************************************************************/ static struct platform_device db1300_wm9715_dev = { - .name = "wm9712-codec", + .name = "wm9712-component", .id = 1, /* ID of PSC for AC97 audio, see asoc glue! */ }; diff --git a/drivers/mfd/wm97xx-core.c b/drivers/mfd/wm97xx-core.c index f5a8347f837f..9fec11fc0410 100644 --- a/drivers/mfd/wm97xx-core.c +++ b/drivers/mfd/wm97xx-core.c @@ -109,7 +109,7 @@ static const struct regmap_config wm9705_regmap_config = { }; static struct mfd_cell wm9705_cells[] = { - { .name = "wm9705-codec", }, + { .name = "wm9705-component", }, { .name = "wm97xx-ts", }, }; @@ -177,7 +177,7 @@ static const struct regmap_config wm9712_regmap_config = { }; static struct mfd_cell wm9712_cells[] = { - { .name = "wm9712-codec", }, + { .name = "wm9712-component", }, { .name = "wm97xx-ts", }, }; @@ -248,7 +248,7 @@ static const struct regmap_config wm9713_regmap_config = { }; static struct mfd_cell wm9713_cells[] = { - { .name = "wm9713-codec", }, + { .name = "wm9713-component", }, { .name = "wm97xx-ts", }, }; diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 301e1fc9a377..d452478d62b4 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link db1200_ac97_dai = { .codec_dai_name = "ac97-hifi", .cpu_dai_name = "au1xpsc_ac97.1", .platform_name = "au1xpsc-pcm.1", - .codec_name = "ac97-codec.1", + .codec_name = "ac97-component.1", }; static struct snd_soc_card db1200_ac97_machine = { @@ -68,7 +68,7 @@ static struct snd_soc_dai_link db1300_ac97_dai = { .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "au1xpsc_ac97.1", .platform_name = "au1xpsc-pcm.1", - .codec_name = "wm9712-codec.1", + .codec_name = "wm9712-component.1", }; static struct snd_soc_card db1300_ac97_machine = { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index ec731223cab3..869f4b226ccd 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -33,14 +33,14 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { .stream_name = "AC97 Analog", .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "mpc5200-psc-ac97.0", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97.1", .stream_name = "AC97 IEC958", .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; @@ -85,7 +85,7 @@ static int pcm030_fabric_probe(struct platform_device *op) if (ret) dev_err(&op->dev, "request_module returned: %d\n", ret); - pdata->codec_device = platform_device_alloc("wm9712-codec", -1); + pdata->codec_device = platform_device_alloc("wm9712-component", -1); if (!pdata->codec_device) dev_err(&op->dev, "platform_device_alloc() failed\n"); diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index 66fb6c4614d2..dbd81d3b7c2c 100644 --- a/sound/soc/fsl/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c @@ -30,8 +30,8 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { { .name = "HiFi", .stream_name = "HiFi", - .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_dai_name = "wm9712-hifi", + .codec_name = "wm9712-component", .cpu_dai_name = "imx-ssi.0", .platform_name = "imx-ssi.0", .ops = &imx_phycore_hifi_ops, @@ -88,7 +88,7 @@ static int __init imx_phycore_init(void) if (ret) goto fail1; - imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1); + imx_phycore_snd_device = platform_device_alloc("wm9712-component", -1); if (!imx_phycore_snd_device) { ret = -ENOMEM; goto fail2; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 1ed8aa2348f1..937746d1e23e 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -75,7 +75,7 @@ static struct snd_soc_dai_link e800_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97 Aux", @@ -83,7 +83,7 @@ static struct snd_soc_dai_link e800_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index e046770ce70e..e958c4c241f6 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -37,7 +37,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97 Aux", @@ -45,7 +45,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 97167048572d..040db695270e 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -93,7 +93,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .platform_name = "pxa-pcm-audio", .init = palm27x_ac97_init, }, @@ -102,7 +102,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .platform_name = "pxa-pcm-audio", }, }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index ae9c12e1ea2a..af42417ccec5 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -189,7 +189,7 @@ static struct snd_soc_dai_link tosa_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .ops = &tosa_ops, }, { @@ -198,7 +198,7 @@ static struct snd_soc_dai_link tosa_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .ops = &tosa_ops, }, }; diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 864a3345972e..aa12be76d15a 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -53,7 +53,7 @@ static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .init = tegra_wm9712_init, }; @@ -83,7 +83,7 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) card->dev = &pdev->dev; snd_soc_card_set_drvdata(card, machine); - machine->codec = platform_device_alloc("wm9712-codec", -1); + machine->codec = platform_device_alloc("wm9712-component", -1); if (!machine->codec) { dev_err(&pdev->dev, "Can't allocate wm9712 platform device\n"); return -ENOMEM; -- 2.14.4

3 4

[alsa-devel] [PATCH] ALSA: hda/i915: Allow delayed i915 audio component binding
by Takashi Iwai 22 Jul '18

22 Jul '18

Currently HD-audio i915 audio binding doesn't support any delayed binding, and supposes that the i915 driver registers the component immediately. This has been OK, so far, but the work-in-progress change in i915 may introduce the asynchronous binding, which effectively delays the component registration. For addressing it, implement a completion to be synced with the master binding. The timeout is set to 10 seconds which should be long enough and hopefully be not too annoying if anyone boots up a debugging session with i915 KMS turned off. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is applied on top of the current for-next or topic/drm_audio_component branch of sound git tree. sound/hda/hdac_i915.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 8f2aa8bc1185..b5282cbbe489 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,8 @@ #include <sound/hda_i915.h> #include <sound/hda_register.h> +static struct completion bind_complete; + #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ @@ -97,6 +99,19 @@ static bool i915_gfx_present(void) return pci_dev_present(ids); } +static int i915_master_bind(struct device *dev, + struct drm_audio_component *acomp) +{ + complete_all(&bind_complete); + /* clear audio_ops here as it was needed only for completion call */ + acomp->audio_ops = NULL; + return 0; +} + +static const struct drm_audio_component_audio_ops i915_init_ops = { + .master_bind = i915_master_bind +}; + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -117,7 +132,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!i915_gfx_present()) return -ENODEV; - err = snd_hdac_acomp_init(bus, NULL, + init_completion(&bind_complete); + + err = snd_hdac_acomp_init(bus, &i915_init_ops, i915_component_master_match, sizeof(struct i915_audio_component) - sizeof(*acomp)); if (err < 0) @@ -125,8 +142,11 @@ int snd_hdac_i915_init(struct hdac_bus *bus) acomp = bus->audio_component; if (!acomp) return -ENODEV; - if (!acomp->ops) + if (!acomp->ops) { request_module("i915"); + /* 10s timeout */ + wait_for_completion_timeout(&bind_complete, 10 * 1000); + } if (!acomp->ops) { snd_hdac_acomp_exit(bus); return -ENODEV; -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: da7219: Allow pdata to specify a VDDIO
by Daniel Kurtz 21 Jul '18

21 Jul '18

Some systems do not have software controllable regulators driving the DA7219's supplies, nor can they use device tree to create "always-on fixed regulators" to easily pretend like they do. On these systems the call to devm_regulator_bulk_get() just creates a set of dummy registers. Calling regulator_get_voltage() on a dummy regulator just returns -EINVAL, in which case the DA7219 is always set up to use the default VDDIO voltage range of 2.5-3.6V. Provide a new device property to let such systems specify a different VDDIO if needed (e.g., 1.8V). Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- This patch tries to solve the same problem as in the following patches, but does so in a way that doesn't require registering fixed regulators in the audio machine-driver: https://patchwork.kernel.org/patch/10531097 & 10531099 https://patchwork.kernel.org/patch/10536023/ include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 19 +++++++++++++------ 2 files changed, 15 insertions(+), 6 deletions(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16f2d10ab..16ab125ad4adbf 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + u32 vddio; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d38..d8f202c57f958d 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1634,6 +1634,9 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone else pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + if (!device_property_read_string(dev, "dlg,vddio", &of_val32)) + pdata->vddio = of_val32; + return pdata; } @@ -1717,8 +1720,12 @@ static int da7219_handle_supplies(struct snd_soc_component *component) /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; ret = regulator_get_voltage(vddio); + /* If regulator_get_voltage() fails, try to use vddio from pdata. */ + if (ret < 0 && da7219->pdata) + ret = da7219->pdata->vddio; if (ret < 1200000) - dev_warn(component->dev, "Invalid VDDIO voltage\n"); + dev_warn(component->dev, "Invalid VDDIO voltage: %d mV\n", + ret); else if (ret < 2800000) io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; @@ -1872,6 +1879,11 @@ static int da7219_probe(struct snd_soc_component *component) mutex_init(&da7219->ctrl_lock); mutex_init(&da7219->pll_lock); + /* Handle DT/ACPI/Platform data */ + da7219->pdata = dev_get_platdata(component->dev); + if (!da7219->pdata) + da7219->pdata = da7219_fw_to_pdata(component); + /* Regulator configuration */ ret = da7219_handle_supplies(component); if (ret) @@ -1897,11 +1909,6 @@ static int da7219_probe(struct snd_soc_component *component) break; } - /* Handle DT/ACPI/Platform data */ - da7219->pdata = dev_get_platdata(component->dev); - if (!da7219->pdata) - da7219->pdata = da7219_fw_to_pdata(component); - da7219_handle_pdata(component); /* Check if MCLK provided */ -- 2.18.0.233.g985f88cf7e-goog

2 3

[alsa-devel] [asoc:for-4.19 199/204] sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'
by kbuild test robot 21 Jul '18

21 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.19 head: a8e43c21a8a32a3af4abc605b6ebcab039f28e00 commit: c41c2a355b86368608377eaf3df442ec0f342f1e [199/204] ASoC: meson: add tdm output driver config: openrisc-allmodconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c41c2a355b86368608377eaf3df442ec0f342f1e # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/meson/axg-tdm-interface.c: In function 'axg_tdm_iface_set_lrclk': >> sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle' [-Werror=implicit-function-declaration] ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); ^~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/clk_set_duty_cycle +225 sound/soc/meson/axg-tdm-interface.c d60e4f1e Jerome Brunet 2018-07-17 188 d60e4f1e Jerome Brunet 2018-07-17 189 static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, d60e4f1e Jerome Brunet 2018-07-17 190 struct snd_pcm_hw_params *params) d60e4f1e Jerome Brunet 2018-07-17 191 { d60e4f1e Jerome Brunet 2018-07-17 192 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); d60e4f1e Jerome Brunet 2018-07-17 193 unsigned int ratio_num; d60e4f1e Jerome Brunet 2018-07-17 194 int ret; d60e4f1e Jerome Brunet 2018-07-17 195 d60e4f1e Jerome Brunet 2018-07-17 196 ret = clk_set_rate(iface->lrclk, params_rate(params)); d60e4f1e Jerome Brunet 2018-07-17 197 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 198 dev_err(dai->dev, "setting sample clock failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 199 return ret; d60e4f1e Jerome Brunet 2018-07-17 200 } d60e4f1e Jerome Brunet 2018-07-17 201 d60e4f1e Jerome Brunet 2018-07-17 202 switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { d60e4f1e Jerome Brunet 2018-07-17 203 case SND_SOC_DAIFMT_I2S: d60e4f1e Jerome Brunet 2018-07-17 204 case SND_SOC_DAIFMT_LEFT_J: d60e4f1e Jerome Brunet 2018-07-17 205 case SND_SOC_DAIFMT_RIGHT_J: d60e4f1e Jerome Brunet 2018-07-17 206 /* 50% duty cycle ratio */ d60e4f1e Jerome Brunet 2018-07-17 207 ratio_num = 1; d60e4f1e Jerome Brunet 2018-07-17 208 break; d60e4f1e Jerome Brunet 2018-07-17 209 d60e4f1e Jerome Brunet 2018-07-17 210 case SND_SOC_DAIFMT_DSP_A: d60e4f1e Jerome Brunet 2018-07-17 211 case SND_SOC_DAIFMT_DSP_B: d60e4f1e Jerome Brunet 2018-07-17 212 /* d60e4f1e Jerome Brunet 2018-07-17 213 * A zero duty cycle ratio will result in setting the mininum d60e4f1e Jerome Brunet 2018-07-17 214 * ratio possible which, for this clock, is 1 cycle of the d60e4f1e Jerome Brunet 2018-07-17 215 * parent bclk clock high and the rest low, This is exactly d60e4f1e Jerome Brunet 2018-07-17 216 * what we want here. d60e4f1e Jerome Brunet 2018-07-17 217 */ d60e4f1e Jerome Brunet 2018-07-17 218 ratio_num = 0; d60e4f1e Jerome Brunet 2018-07-17 219 break; d60e4f1e Jerome Brunet 2018-07-17 220 d60e4f1e Jerome Brunet 2018-07-17 221 default: d60e4f1e Jerome Brunet 2018-07-17 222 return -EINVAL; d60e4f1e Jerome Brunet 2018-07-17 223 } d60e4f1e Jerome Brunet 2018-07-17 224 d60e4f1e Jerome Brunet 2018-07-17 @225 ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); d60e4f1e Jerome Brunet 2018-07-17 226 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 227 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 228 "setting sample clock duty cycle failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 229 return ret; d60e4f1e Jerome Brunet 2018-07-17 230 } d60e4f1e Jerome Brunet 2018-07-17 231 d60e4f1e Jerome Brunet 2018-07-17 232 /* Set sample clock inversion */ d60e4f1e Jerome Brunet 2018-07-17 233 ret = clk_set_phase(iface->lrclk, d60e4f1e Jerome Brunet 2018-07-17 234 axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); d60e4f1e Jerome Brunet 2018-07-17 235 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 236 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 237 "setting sample clock phase failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 238 return ret; d60e4f1e Jerome Brunet 2018-07-17 239 } d60e4f1e Jerome Brunet 2018-07-17 240 d60e4f1e Jerome Brunet 2018-07-17 241 return 0; d60e4f1e Jerome Brunet 2018-07-17 242 } d60e4f1e Jerome Brunet 2018-07-17 243 :::::: The code at line 225 was first introduced by commit :::::: d60e4f1e4be5e2dfb55fb084b119aed094227a35 ASoC: meson: add tdm interface driver :::::: TO: Jerome Brunet <jbrunet(a)baylibre.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v1 0/7] arm64: dts: rockchip: add basic dts file for PX30 SoCs
by cl＠rock-chips.com 20 Jul '18

20 Jul '18

From: Liang Chen <cl(a)rock-chips.com> Liang Chen (7): Documentation: usb: dwc2: add description for px30 Documentation: rockchip-dw-mshc: add description for px30 Documentation: spi-rockchip: add description for px30 Documentation: rockchip-i2s: add description for px30 Documentation: rockchip: grf: add grf and pmugrf description for px30 arm64: dts: rockchip: add core dtsi file for PX30 SoCs arm64: dts: rockchip: add PX30 eavluation board devicetree Documentation/devicetree/bindings/arm/rockchip.txt | 4 + .../devicetree/bindings/mmc/rockchip-dw-mshc.txt | 1 + .../devicetree/bindings/soc/rockchip/grf.txt | 2 + .../devicetree/bindings/sound/rockchip-i2s.txt | 1 + .../devicetree/bindings/spi/spi-rockchip.txt | 1 + Documentation/devicetree/bindings/usb/dwc2.txt | 1 + arch/arm64/boot/dts/rockchip/Makefile | 1 + arch/arm64/boot/dts/rockchip/px30-evb.dts | 539 ++++++ arch/arm64/boot/dts/rockchip/px30.dtsi | 2044 ++++++++++++++++++++ 9 files changed, 2594 insertions(+) create mode 100644 arch/arm64/boot/dts/rockchip/px30-evb.dts create mode 100644 arch/arm64/boot/dts/rockchip/px30.dtsi -- 1.9.1

3 14

[alsa-devel] [PATCH v2] ASoC: pxa: remove clock divider and pll setup from zylonite and magician
by Daniel Mack 20 Jul '18

20 Jul '18

The SSP DAI now handles the clocking setup itself, all it needs is the master clock frequency. Remove the code from Zylonite and Magician platforms. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/magician.c | 106 +-------------------------------------- sound/soc/pxa/zylonite.c | 9 ---- 2 files changed, 2 insertions(+), 113 deletions(-) diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 2fc012b06c43..935a248e5bf6 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -90,95 +90,9 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int acps, acds, width; - unsigned int div4 = PXA_SSP_CLK_SCDB_4; + unsigned int width; int ret = 0; - width = snd_pcm_format_physical_width(params_format(params)); - - /* - * rate = SSPSCLK / (2 * width(16 or 32)) - * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1) - */ - switch (params_rate(params)) { - case 8000: - /* off by a factor of 2: bug in the PXA27x audio clock? */ - acps = 32842000; - switch (width) { - case 16: - /* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_16; - break; - default: /* 32 */ - /* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_8; - } - break; - case 11025: - acps = 5622000; - switch (width) { - case 16: - /* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_4; - break; - default: /* 32 */ - /* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - } - break; - case 22050: - acps = 5622000; - switch (width) { - case 16: - /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 44100: - acps = 5622000; - switch (width) { - case 16: - /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 48000: - acps = 12235000; - switch (width) { - case 16: - /* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - break; - default: /* 32 */ - /* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - } - break; - case 96000: - default: - acps = 12235000; - switch (width) { - case 16: - /* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_1; - break; - default: /* 32 */ - /* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */ - acds = PXA_SSP_CLK_AUDIO_DIV_2; - div4 = PXA_SSP_CLK_SCDB_1; - break; - } - break; - } - /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -191,6 +105,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + width = snd_pcm_format_physical_width(params_format(params)); ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; @@ -201,23 +116,6 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* set the SSP audio system clock ACDS divider */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_ACDS, acds); - if (ret < 0) - return ret; - - /* set the SSP audio system clock SCDB divider4 */ - ret = snd_soc_dai_set_clkdiv(cpu_dai, - PXA_SSP_AUDIO_DIV_SCDB, div4); - if (ret < 0) - return ret; - - /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); - if (ret < 0) - return ret; - return 0; } diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index ba468e560dd2..230eee450f45 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -83,11 +83,9 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int pll_out = 0; unsigned int wm9713_div = 0; int ret = 0; int rate = params_rate(params); - int width = snd_pcm_format_physical_width(params_format(params)); /* Only support ratios that we can generate neatly from the AC97 * based master clock - in particular, this excludes 44.1kHz. @@ -109,17 +107,10 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Add 1 to the width for the leading clock cycle */ - pll_out = rate * (width + 1) * 8; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1); if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); - if (ret < 0) - return ret; - if (clk_pout) ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV, WM9713_PCMDIV(wm9713_div)); -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: topology: Increase SND_SOC_TPLG_NUM_TEXTS to 32
by Puneeth Prabhu 20 Jul '18

20 Jul '18

Topology supports maximum of 32 configs for a path whereas kernel can support only 16 configs of a path. So, increase SND_SOC_TPLG_NUM_TEXTS to 32 to support 32 configs for a path. Signed-off-by: Puneeth Prabhu <puneethx.prabhu(a)intel.com> Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada(a)intel.com> --- include/uapi/sound/asoc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index a74ca23..5b1411b 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -88,7 +88,7 @@ #define SND_SOC_TPLG_MAGIC 0x41536F43 /* ASoC */ /* string sizes */ -#define SND_SOC_TPLG_NUM_TEXTS 16 +#define SND_SOC_TPLG_NUM_TEXTS 32 /* ABI version */ #define SND_SOC_TPLG_ABI_VERSION 0x5 /* current version */ -- 2.1.4

3 6

[alsa-devel] No controls in alsamixer
by Riccardo Casagrande 20 Jul '18

20 Jul '18

Hello all, following this thread http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/108068.html I added a new sound card without codec. On my board the i.MX6Q chip puts audio data directly to some ST FDA2100 power amplifiers. The sound card is listed correctly in alsa but in alsamixer there are no audio controls, I need a volume control. I tried to edit asound.conf file with this part pcm.softvol { type softvol slave { pcm "hw:1,0" } control { name "SoftVol" card 1 } } ctl.softvol { type hw card 1 } This is part of logs during the boot --> ITEL imx_itel_probe --> ITEL Configuro audmux --> ITEL imx-audmux-confic ext-port port=0x00000001ptcr=0x9ce73800 pdcr=0x00006000 --> ITEL imx-audmux-config int-port port=0x00000001ptcr=0x00000800 pdcr=0x00002000 --> ITEL imx-probe registro card imx-itel sound-audio-itel: snd-soc-dummy-dai <-> 202c000.ssi mapping ok --> ITEL imx-probe of ssi --> ITEL imx-probe of codec [...] ALSA device list: #0: cs42888-audio #1: imx-audio-itel #2: imx-spdif #3: imx-hdmi-soc [...] ALSA: Restoring mixer settings... No state is present for card imxaudioitel Found hardware: "imx-audio-itel" "" "" "" "" Hardware is initialized using a generic method No state is present for card imxaudioitel No state is present for card imxspdif Found hardware: "imx-spdif" "" "" "" "" Hardware is initialized using a generic method No state is present for card imxspdif Sorry but I'm new in alsa and linux in general. Riccardo

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: fix spelling in devicetree binding document
by Marcel Ziswiler 20 Jul '18

20 Jul '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> This fixes a spelling mistake. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 0f214457476f..9c58f724396a 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -17,7 +17,7 @@ Optional properties: - VDDD-supply : the regulator provider of VDDD -- micbias-resistor-k-ohms : the bias resistor to be used in kOmhs +- micbias-resistor-k-ohms : the bias resistor to be used in kOhms The resistor can take values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not mentioned or if the value is unknown, then -- 2.14.4

2 1

[alsa-devel] [PATCH] MAINTAINERS: add entry for STI audio drivers
by Arnaud Pouliquen 20 Jul '18

20 Jul '18

Add sound/soc/sti drivers entry for STI audio drivers from ST Microelectronics. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 192d7f7..50a7a53 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -13574,6 +13574,13 @@ L: linux-block(a)vger.kernel.org S: Maintained F: drivers/block/skd*[ch] +STI AUDIO (ASoC) DRIVERS +M: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt +F: sound/soc/sti/ + STI CEC DRIVER M: Benjamin Gaignard <benjamin.gaignard(a)linaro.org> S: Maintained -- 2.7.4

2 3

[alsa-devel] [PATCH 00/11] Sound Open Firmware Core
by Liam Girdwood 20 Jul '18

20 Jul '18

Sound Open Firmware (SOF) is a host and DSP architecture agnostic audio DSP firmware. SOF is not tied to any specific host architecture or any specific physical IO communication type (it will work with on SoC DSPs, or DSP connected via SPI/I2C). SOF is also not coupled to any particular DSP architecture and has abstraction similar to Linux to allow porting to other DSP architectures. https://www.sofproject.org/ This patch series introduces the SOF core, later series will add host platform support for SOF. The SOF core manages all the core DSP services and ALSA/ASoC IO. The core is responsible for loading firmware, parsing topology, exposing PCMs and kcontrols, providing debug and trace mechanisms and performing IPC between host and DSP. The SOF core also has logic to allow reuse of existing machine drivers for other platforms/machines without any code modification. i.e. DAI links can be modified at runtime to bind with SOF and SOF topologies instead of existing hard coded DAI links and topology. Liam Girdwood (11): ASoC: SOF: Add Sound Open Firmware driver core ASoC: SOF: Add Sound Open Firmware KControl support ASoC: SOF: Add driver debug support. ASoC: SOF: Add support for IPC IO between DSP and Host ASoC: SOF: Add PCM operations support ASoC: SOF: Add support for loading topologies ASoC: SOF: Add DSP firmware trace event support ASoC: SOF: Add DSP HW abstraction operations ASoC: SOF: Add firmware loader support ASoC: SOF: Add userspace ABI support ASoC: SOF: Add Build support for SOF core include/sound/soc.h | 1 + include/sound/sof.h | 92 ++ include/uapi/sound/sof-abi.h | 29 + include/uapi/sound/sof-eq.h | 102 ++ include/uapi/sound/sof-fw.h | 69 + include/uapi/sound/sof-ipc.h | 915 +++++++++++++ include/uapi/sound/sof-tone.h | 28 + include/uapi/sound/sof-topology.h | 92 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sof/Kconfig | 38 + sound/soc/sof/Makefile | 6 + sound/soc/sof/control.c | 218 +++ sound/soc/sof/core.c | 373 ++++++ sound/soc/sof/debug.c | 147 +++ sound/soc/sof/ipc.c | 659 ++++++++++ sound/soc/sof/loader.c | 287 ++++ sound/soc/sof/ops.c | 210 +++ sound/soc/sof/ops.h | 298 +++++ sound/soc/sof/pcm.c | 641 +++++++++ sound/soc/sof/sof-priv.h | 504 +++++++ sound/soc/sof/topology.c | 2036 +++++++++++++++++++++++++++++ sound/soc/sof/trace.c | 279 ++++ 23 files changed, 7026 insertions(+) create mode 100644 include/sound/sof.h create mode 100644 include/uapi/sound/sof-abi.h create mode 100644 include/uapi/sound/sof-eq.h create mode 100644 include/uapi/sound/sof-fw.h create mode 100644 include/uapi/sound/sof-ipc.h create mode 100644 include/uapi/sound/sof-tone.h create mode 100644 include/uapi/sound/sof-topology.h create mode 100644 sound/soc/sof/Kconfig create mode 100644 sound/soc/sof/Makefile create mode 100644 sound/soc/sof/control.c create mode 100644 sound/soc/sof/core.c create mode 100644 sound/soc/sof/debug.c create mode 100644 sound/soc/sof/ipc.c create mode 100644 sound/soc/sof/loader.c create mode 100644 sound/soc/sof/ops.c create mode 100644 sound/soc/sof/ops.h create mode 100644 sound/soc/sof/pcm.c create mode 100644 sound/soc/sof/sof-priv.h create mode 100644 sound/soc/sof/topology.c create mode 100644 sound/soc/sof/trace.c -- 2.17.1

1 0