August 2018 - 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

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[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

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[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.

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[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

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[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

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Re: [alsa-devel] : Kernal+debug installed: Help with TASCAM US-122
by Takashi Iwai 09 Aug '18

09 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

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[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

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[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

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[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

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