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

[PATCH v6 00/14] ASoC: Intel: Catpt - Lynx and Wildcat point
by Cezary Rojewski 22 Sep '20

22 Sep '20

Implement support for Lynxpoint and Wildcat Point AudioDSP. Catpt solution deprecates existing sound/soc/intel/haswell which is removed in the following series. This cover-letter is followed by 'Developer's deep dive' message schedding light on catpt's key concepts and areas addressed. Due to high range of errors and desynchronization from recommendations set by Windows solution, re-write came as a lower-cost solution compared to refactoring /haswell/ with several series of patches. Series is dependent on linux-spi change: spi: pxa2xx: Add SSC2 and SSPSP2 SSP registers https://www.spinics.net/lists/linux-spi/msg23885.html which has been already merged and is now part of linux-spi tree. Bulk of series content is device driver core code - everything up to patch 7/14 - with fs entries and trace macros introduced right after. While each core patch is shaped in such a way that no unavailable members are ever called, until patch 10/14 is applied, no code compilation can occur as no Makefile is present. Once said patch is added, Makefile and Kconfig are implemented and driver module compiles as expected. Special thanks go to Marcin Barlik and Piotr Papierkowski for sharing their LPT/WPT AudioDSP architecture expertise as well as helping backtrack its historical background. My thanks go to Amadeusz Slawinski for reviews and improvements proposed on and off the internal list. Most of internal diff below is his contribution. Krzysztof Hejmowski helped me setup my own Xtensa environment and recompile LPT/WPT FW binary sources what sped up the development greatly. This would not have been possible without help from these champions, especially considering how quickly the catpt was written: 2 weeks features, 3 weeks optimizations. Thank you. Userspace-exposed members are compatible with what is exposed by deprecated solution as well as FW binary being re-used thus no harm is done. The only visible differences are: the newly added 'Loopback Mute' kcontrol and volume support extending to quad from stereo. On top of fixing erros and design flows, catpt also adds module reload, dynamic SRAM memory allocation during PCM runtime and exposes missing userspace API: 'Loopback Mute' kcontrol, quad volume controls and sysfs fw-version entries. Event tracing is provided to easy solution debugging. Following are not included in this update and are scheduled as later addition: - fw logging - module (library) support Note: LPT power up/down sequences might get aligned with WPT once enough testing is done as capabilities are shared for both DSPs. Note #2: Both LPT and WPT power up/down sequences may get optimized in future updates as thanks to help from the Windows team, most of nuances behind why/what/when in regard to hw registers have been backtracked and reviewed again. Link to developer's deep dive message: https://www.spinics.net/lists/alsa-devel/msg113563.html Changes in v6: - reordered and reorganized code for patches 1/13 - 8/13 of v5, so each patches makes use of no member or function which is unavailable to it. Series size increased from 13 to 14 patches: addition of base members e.g.: registers has been split from addition of device.c file which describes acpi device behavior Changes in v5: https://www.spinics.net/lists/alsa-devel/msg115621.html Basically everything below is result of Andy's review. Thank you Andy for taking time into this detailed review - catpt now makes use of common linux/pxa2xx_ssp.h header file, removing redundant SSP register declarations in the process. As stated in the opening, this is dependent upon linux-spi change: spi: pxa2xx: Add SSC2 and SSPSP2 SSP registers - updated Kconfig by removing DMADEVICES and adding COMPILE_TEST as optional depends-on - updated all register macros definitions to be more safe against common arithmetics when specifying macro's parameters - removed CONFIG_PM and CONFIG_PM_SLEEP usage in favor of __maybe_unused - all 'if (ret < 0)' converted to simple 'if (ret)' whenever possible - fixed erroneous check for platform_device_register_data within catpt_register_board() - _SLAVE/_MASTER replaced with more inclusive _CONSUMER/_PROVIDER for enum catpt_ssp_mode - catpt_acpi_probe() is now making use of high-level wrappers for ioremapping and resource assignment, reducing function's code size - due to improved catpt_acpi_probe() behavior, catpt_acpi_remove() needs not to cast dma_free_coherent() any longer - DMA source and destrination maxburst now of value 16, see: https://www.spinics.net/lists/alsa-devel/msg114394.html - simplified catpt_dsp_update_lpclock() as list_for_each_entry() is empty-safe by default - dropped '_SSP_' from all names of all CATPT_SSP_SSXXX_DEFAULT macros - catpt_updatel_pci now makes use of linux/pci.h and uapi/linux/pci.h constants such as: PCI_PM_CTRL_STATE_MASK and PCI_D3hot Changes in v4: https://www.spinics.net/lists/alsa-devel/msg113762.html - fixed compilation with i386 kconfig (conflicting names) - streamlined naming for SHIM and PCI registers to match SSP ones (SHIM_REG -> SHIM) - catpt_component_probe removed and kcontrols again initializzed statically via snd_kcontrol_new array: this is to remove kctl->id.device shenanigans - renamed catpt_set_ctlvol to catpt_set_dspvol - function name wasn't matching its purpose Changes in v3: - fixed IRAM mask usage in lpt_dsp_power_up (dsp.c) - updated dbg message formatting in catpt_restore_fwimage as suggested by Andy - fixed alignment for struct catpt_ssp_device_format - catpt_set_ctlvol now verifies all-equal scenario based on all channels rather than just first two as requested by Amadeo - fixed SPDX for registers.h Changes in v2: https://www.spinics.net/lists/alsa-devel/msg113660.html - fixed SPDX formatting for all header files as well as pcm.c - fixed size provided to memcpy() in fw_build_read() as reported by Mark - renamed struct catpt_pdata to struct catpt_spec (cosmetic) - fixed erroneous path in catpt_load_block: region is properly released - trace.h events for updating registers have been removed and usages replaced by dev_dbg (SRAMPGE/ LPCS) - as requested by Andy, struct resource has replaced struct catpt_mbank and struct catpt_mregion. This change cascaded into: - catpt_mbank_size and catpt_mregion_size replaced by resource_size - catpt_mregion_intersects replaced by resource_overlaps - all catpt_mbank_ and catpt_mregion_ handlers found in loader.c (_request, _reserve, _release, _extract, _split, _join) have been removed - __request_region and __release_region have been enlisted in their place - catpt_mregion_intersecting renamed to catpt_resource_overlapping - catpt_request_region helper has been provided to deal with -size based requests o haven't found direct replacements in resource.c/ ioport.h for both functions - catpt_mbank_create and catpt_mbank_remove renamed to catpt_sram_init and catpt_sram_free respectively - catpt_sram_init now returns void instead of int and has been converted to simple initialized. This change ultimately cascaded into: o both SRAM banks initialization being moved to catpt_dev_init from catpt_acpi_probe (device.c) o catpt_dev::spec is now initialized first, with catpt_dev_init following it soon after o catpt_acpi_probe erroneous path has been simplified as SRAM banks no longer need to be freed - catpt_sram_free now frees all resources via child -> sibling enumeration rather than region_list iteration - catpt_dsp_update_srampge and catpt_dsp_set_srampge now accept new argument: unsigned long mask. Caused by removal of catpt_mbank - mask is taken directly from catpt_dev::spec::d/iram_mask - trace.h events for catpt_mbank and catpt_mregion have been removed Diff against last drop on internal list: https://www.spinics.net/lists/alsa-devel/msg113549.html - replaced spinlock with mutex for mregion allocation and release to address sleeping in atomic context warnings - fixed coredump fw_hash dumping - kcontrol values are now always stored regardless of stream of interest is running or not - kcontrol values are now applied after stream is prepared instead of ignoring what has been set by user initially - catpt_pdata instances have been renamed from hsw_ and bdw_ to lpt_ and wpt_ respectively - reordered Makefile .o(s) (cosmetic) Cezary Rojewski (14): ASoC: Intel: Add catpt base members ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC message handlers ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: PCM operations ASoC: Intel: catpt: Device driver lifecycle ASoC: Intel: catpt: Event tracing ASoC: Intel: catpt: Simple sysfs attributes ASoC: Intel: Select catpt and deprecate haswell ASoC: Intel: haswell: Remove haswell-solution specific code ASoC: Intel: broadwell: Remove haswell-solution specific code ASoC: Intel: bdw-5650: Remove haswell-solution specific code ASoC: Intel: bdw-5677: Remove haswell-solution specific code sound/soc/intel/Kconfig | 24 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/boards/Kconfig | 8 +- sound/soc/intel/boards/bdw-rt5650.c | 36 - sound/soc/intel/boards/bdw-rt5677.c | 33 - sound/soc/intel/boards/broadwell.c | 33 - sound/soc/intel/boards/haswell.c | 28 +- sound/soc/intel/catpt/Makefile | 6 + sound/soc/intel/catpt/core.h | 189 +++++ sound/soc/intel/catpt/device.c | 354 ++++++++ sound/soc/intel/catpt/dsp.c | 572 +++++++++++++ sound/soc/intel/catpt/fs.c | 79 ++ sound/soc/intel/catpt/ipc.c | 298 +++++++ sound/soc/intel/catpt/loader.c | 671 +++++++++++++++ sound/soc/intel/catpt/messages.c | 313 +++++++ sound/soc/intel/catpt/messages.h | 401 +++++++++ sound/soc/intel/catpt/pcm.c | 1212 +++++++++++++++++++++++++++ sound/soc/intel/catpt/registers.h | 178 ++++ sound/soc/intel/catpt/trace.h | 83 ++ 19 files changed, 4377 insertions(+), 143 deletions(-) create mode 100644 sound/soc/intel/catpt/Makefile create mode 100644 sound/soc/intel/catpt/core.h create mode 100644 sound/soc/intel/catpt/device.c create mode 100644 sound/soc/intel/catpt/dsp.c create mode 100644 sound/soc/intel/catpt/fs.c create mode 100644 sound/soc/intel/catpt/ipc.c create mode 100644 sound/soc/intel/catpt/loader.c create mode 100644 sound/soc/intel/catpt/messages.c create mode 100644 sound/soc/intel/catpt/messages.h create mode 100644 sound/soc/intel/catpt/pcm.c create mode 100644 sound/soc/intel/catpt/registers.h create mode 100644 sound/soc/intel/catpt/trace.h -- 2.17.1

5 23

[PATCH v2 0/2] DAI driver for new XCVR IP
by Viorel Suman (OSS) 21 Sep '20

21 Sep '20

From: Viorel Suman <viorel.suman(a)nxp.com> DAI driver for new XCVR IP found in i.MX8MP. Viorel Suman (2): ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver ASoC: dt-bindings: fsl_xcvr: Add document for XCVR Changes since v1: - improved 6- and 12-ch layout comment - used regmap polling function, improved clocks handling in runtime_resume - added FW size check in FW load function, improved IRQ handler, removed dummy IRQ handlers - fixed yaml file .../devicetree/bindings/sound/fsl,xcvr.yaml | 103 ++ sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_xcvr.c | 1343 ++++++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 266 ++++ 5 files changed, 1724 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,xcvr.yaml create mode 100644 sound/soc/fsl/fsl_xcvr.c create mode 100644 sound/soc/fsl/fsl_xcvr.h -- 2.7.4

1 2

[PATCH] ALSA: usb-audio: Add mixer support for Pioneer DJ DJM-250MK2
by František Kučera 21 Sep '20

21 Sep '20

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows mapping playback and capture channels to available sources. Configures the card through USB commands. --- sound/usb/mixer_quirks.c | 216 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 216 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199cdbfdc761..b7816a71c42d 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2602,6 +2602,219 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * Pioneer DJ DJM-250MK2 and maybe other DJM models + * + * For playback, no duplicate mapping should be set. + * There are three mixer stereo channels (CH1, CH2, AUX) + * and three stereo sources (Playback 1-2, Playback 3-4, Playback 5-6). + * Each channel should be mapped just once to one source. + * If mapped multiple times, only one source will play on given channel + * (sources are not mixed together). + * + * For recording, duplicate mapping is OK. We will get the same signal multiple times. + * + * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN. + * + * See also notes in the quirks-table.h file. + */ + +typedef struct { + const u16 wIndex; + const u16 wValue; + const char *name; +} snd_pioneer_djm_option; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_LEVEL[] = { + { .name = "-5 dB", .wValue = 0x0300, .wIndex = 0x8003 }, + { .name = "-10 dB", .wValue = 0x0200, .wIndex = 0x8003 }, + { .name = "-15 dB", .wValue = 0x0100, .wIndex = 0x8003 }, + { .name = "-19 dB", .wValue = 0x0000, .wIndex = 0x8003 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH12[] = { + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0109, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x010d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x010a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH34[] = { + { .name = "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0209, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x020d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x020a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH56[] = { + { .name = "REC OUT", .wValue = 0x030a, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0311, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0312, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0309, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x030d, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_12[] = { + { .name = "CH1", .wValue = 0x0100, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0101, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0104, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_34[] = { + { .name = "CH1", .wValue = 0x0200, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0201, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0204, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_56[] = { + { .name = "CH1", .wValue = 0x0300, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0301, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0304, .wIndex = 0x8016 } +}; + +typedef struct { + const char *name; + const snd_pioneer_djm_option *options; + const size_t count; + const u16 default_value; +} snd_pioneer_djm_option_group; + +#define SND_PIONEER_DJM_OPTION_GROUP_ITEM(_name, suffix, _default_value) { \ + .name = _name, \ + .options = SND_PIONEER_DJM_OPTIONS_##suffix, \ + .count = ARRAY_SIZE(SND_PIONEER_DJM_OPTIONS_##suffix), \ + .default_value = _default_value } + +static const snd_pioneer_djm_option_group SND_PIONEER_DJM_OPTION_GROUPS[] = { + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Master Capture Level Capture Switch", CAPTURE_LEVEL, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 1-2 Capture Switch", CAPTURE_CH12, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 3-4 Capture Switch", CAPTURE_CH34, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 5-6 Capture Switch", CAPTURE_CH56, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 1-2 Playback Switch", PLAYBACK_12, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 3-4 Playback Switch", PLAYBACK_34, 1), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 5-6 Playback Switch", PLAYBACK_56, 2) +}; + +// layout of the kcontrol->private_value: +#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff +#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000 +#define SND_PIONEER_DJM_GROUP_SHIFT 16 + +static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *info) +{ + u16 group_index = kctl->private_value >> SND_PIONEER_DJM_GROUP_SHIFT; + size_t count; + const char *name; + const snd_pioneer_djm_option_group *group; + + if (group_index < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS)) { + group = &SND_PIONEER_DJM_OPTION_GROUPS[group_index]; + count = group->count; + if (info->value.enumerated.item >= count) + info->value.enumerated.item = count - 1; + name = group->options[info->value.enumerated.item].name; + strlcpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name)); + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = count; + return 0; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value) +{ + int err; + + if (group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS) + && value < SND_PIONEER_DJM_OPTION_GROUPS[group].count) { + err = snd_usb_lock_shutdown(mixer->chip); + if (err) + return err; + + err = snd_usb_ctl_msg( + mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), + USB_REQ_SET_FEATURE, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wValue, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wIndex, + NULL, 0); + + snd_usb_unlock_shutdown(mixer->chip); + return err; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + elem->value.enumerated.item[0] = kctl->private_value & SND_PIONEER_DJM_VALUE_MASK; + return 0; +} + +static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + unsigned long private_value = kctl->private_value; + + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = elem->value.enumerated.item[0]; + + kctl->private_value = group << SND_PIONEER_DJM_GROUP_SHIFT | value; + + return snd_pioneer_djm_controls_update(mixer, group, value); +} + +static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list) +{ + unsigned long private_value = list->kctl->private_value; + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = (private_value & SND_PIONEER_DJM_VALUE_MASK); + + return snd_pioneer_djm_controls_update(list->mixer, group, value); +} + +static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + const snd_pioneer_djm_option_group *group; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_pioneer_djm_controls_info, + .get = snd_pioneer_djm_controls_get, + .put = snd_pioneer_djm_controls_put + }; + + for (i = 0; i < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS); i++) { + group = &SND_PIONEER_DJM_OPTION_GROUPS[i]; + knew.name = group->name; + knew.private_value = i << SND_PIONEER_DJM_GROUP_SHIFT | group->default_value; + err = snd_pioneer_djm_controls_update(mixer, i, group->default_value); + if (err) + return err; + err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume, + &knew, NULL); + if (err) + return err; + } + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2706,6 +2919,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ + err = snd_pioneer_djm_controls_create(mixer); + break; } return err; -- 2.20.1

1 0

[PATCH] ASoC: SOF: pm: Fix prepare callback behavior for OF usecase
by Kai Vehmanen 21 Sep '20

21 Sep '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> On i.MX platforms PM is not managed via ACPI although CONFIG_ACPI can be set. So, in order to correctly set the system target state we introduce a flag for platforms that require to use acpi target states. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/sof.h | 2 ++ sound/soc/sof/pm.c | 12 +++++++----- sound/soc/sof/sof-pci-dev.c | 9 +++++++++ 3 files changed, 18 insertions(+), 5 deletions(-) diff --git a/include/sound/sof.h b/include/sound/sof.h index f3e716c8ce1c..9aa055289dcc 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -66,6 +66,8 @@ struct sof_dev_desc { /* alternate list of machines using this configuration */ struct snd_soc_acpi_mach *alt_machines; + bool use_acpi_target_states; + /* Platform resource indexes in BAR / ACPI resources. */ /* Must set to -1 if not used - add new items to end */ int resindex_lpe_base; diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index a5f7c7f024a1..c83fb6255961 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -305,15 +305,17 @@ EXPORT_SYMBOL(snd_sof_suspend); int snd_sof_prepare(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); + const struct sof_dev_desc *desc = sdev->pdata->desc; + + /* will suspend to S3 by default */ + sdev->system_suspend_target = SOF_SUSPEND_S3; + + if (!desc->use_acpi_target_states) + return 0; #if defined(CONFIG_ACPI) if (acpi_target_system_state() == ACPI_STATE_S0) sdev->system_suspend_target = SOF_SUSPEND_S0IX; - else - sdev->system_suspend_target = SOF_SUSPEND_S3; -#else - /* will suspend to S3 by default */ - sdev->system_suspend_target = SOF_SUSPEND_S3; #endif return 0; diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 8c53f6935417..8f62e3487dc1 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -77,6 +77,7 @@ static const struct dmi_system_id community_key_platforms[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) static const struct sof_dev_desc bxt_desc = { .machines = snd_soc_acpi_intel_bxt_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -94,6 +95,7 @@ static const struct sof_dev_desc bxt_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) static const struct sof_dev_desc glk_desc = { .machines = snd_soc_acpi_intel_glk_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -139,6 +141,7 @@ static const struct sof_dev_desc tng_desc = { static const struct sof_dev_desc cnl_desc = { .machines = snd_soc_acpi_intel_cnl_machines, .alt_machines = snd_soc_acpi_intel_cnl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -157,6 +160,7 @@ static const struct sof_dev_desc cnl_desc = { static const struct sof_dev_desc cfl_desc = { .machines = snd_soc_acpi_intel_cfl_machines, .alt_machines = snd_soc_acpi_intel_cfl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -175,6 +179,7 @@ static const struct sof_dev_desc cfl_desc = { static const struct sof_dev_desc cml_desc = { .machines = snd_soc_acpi_intel_cml_machines, .alt_machines = snd_soc_acpi_intel_cml_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -193,6 +198,7 @@ static const struct sof_dev_desc cml_desc = { static const struct sof_dev_desc icl_desc = { .machines = snd_soc_acpi_intel_icl_machines, .alt_machines = snd_soc_acpi_intel_icl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -211,6 +217,7 @@ static const struct sof_dev_desc icl_desc = { static const struct sof_dev_desc tgl_desc = { .machines = snd_soc_acpi_intel_tgl_machines, .alt_machines = snd_soc_acpi_intel_tgl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -244,6 +251,7 @@ static const struct sof_dev_desc tglh_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) static const struct sof_dev_desc ehl_desc = { .machines = snd_soc_acpi_intel_ehl_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -261,6 +269,7 @@ static const struct sof_dev_desc ehl_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_JASPERLAKE) static const struct sof_dev_desc jsl_desc = { .machines = snd_soc_acpi_intel_jsl_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, -- 2.27.0

2 1

[PATCH v3 1/2] dt-bindings: tlv320adcx140: Add GPIO config and drive config
by Camel Guo 21 Sep '20

21 Sep '20

From: Camel Guo <camelg(a)axis.com> Add properties for configuring the General Purpose Input Output (GPIO). There are 2 settings for GPIO, configuration and the output drive type. Signed-off-by: Camel Guo <camelg(a)axis.com> Acked-by: Dan Murphy <dmurphy(a)ti.com> --- v3: - Fix typo - Add Acked-By from Dan .../bindings/sound/tlv320adcx140.yaml | 44 +++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index f578f17f3e04..2f95ccde4dc3 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -134,6 +134,49 @@ patternProperties: 4d - Drive weak low and active high 5d - Drive Hi-Z and active high + ti,gpio-config: + description: | + Defines the configuration and output drive for the General Purpose + Input and Output pin (GPIO1). Its value is a pair, the first value is for + the configuration type and the second value is for the output drive + type. The array is defined as <GPIO1_CFG GPIO1_DRV> + + configuration for the GPIO pin can be one of the following: + 0 - disabled + 1 - GPIO1 is configured as a general-purpose output (GPO) + 2 - (default) GPIO1 is configured as a device interrupt output (IRQ) + 3 - GPIO1 is configured as a secondary ASI output (SDOUT2) + 4 - GPIO1 is configured as a PDM clock output (PDMCLK) + 8 - GPIO1 is configured as an input to control when MICBIAS turns on or + off (MICBIAS_EN) + 9 - GPIO1 is configured as a general-purpose input (GPI) + 10 - GPIO1 is configured as a master clock input (MCLK) + 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN) + 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2 + (PDMDIN1) + 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4 + (PDMDIN2) + 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6 + (PDMDIN3) + 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8 + (PDMDIN4) + + output drive type for the GPIO pin can be one of the following: + 0 - Hi-Z output + 1 - Drive active low and active high + 2 - (default) Drive active low and weak high + 3 - Drive active low and Hi-Z + 4 - Drive weak low and active high + 5 - Drive Hi-Z and active high + + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32-array + - minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + required: - compatible - reg @@ -150,6 +193,7 @@ examples: ti,mic-bias-source = <6>; ti,pdm-edge-select = <0 1 0 1>; ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; ti,gpo-config-1 = <0 0>; ti,gpo-config-2 = <0 0>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; -- 2.20.1

3 4

[PATCH v5 1/2] dt-bindings: tlv320adcx140: Add GPIO config and drive config
by Camel Guo 21 Sep '20

21 Sep '20

From: Camel Guo <camelg(a)axis.com> Add properties for configuring the General Purpose Input Output (GPIO). There are 2 settings for GPIO, configuration and the output drive type. Signed-off-by: Camel Guo <camelg(a)axis.com> Acked-by: Dan Murphy <dmurphy(a)ti.com> --- v4: - Rebase v3: - Fix typo - Add Acked-By from Dan .../bindings/sound/tlv320adcx140.yaml | 44 +++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index e79f8d1891e4..798b366fe6d1 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -140,6 +140,49 @@ patternProperties: 4d - Drive weak low and active high 5d - Drive Hi-Z and active high + ti,gpio-config: + description: | + Defines the configuration and output drive for the General Purpose + Input and Output pin (GPIO1). Its value is a pair, the first value is for + the configuration type and the second value is for the output drive + type. The array is defined as <GPIO1_CFG GPIO1_DRV> + + configuration for the GPIO pin can be one of the following: + 0 - disabled + 1 - GPIO1 is configured as a general-purpose output (GPO) + 2 - (default) GPIO1 is configured as a device interrupt output (IRQ) + 3 - GPIO1 is configured as a secondary ASI output (SDOUT2) + 4 - GPIO1 is configured as a PDM clock output (PDMCLK) + 8 - GPIO1 is configured as an input to control when MICBIAS turns on or + off (MICBIAS_EN) + 9 - GPIO1 is configured as a general-purpose input (GPI) + 10 - GPIO1 is configured as a master clock input (MCLK) + 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN) + 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2 + (PDMDIN1) + 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4 + (PDMDIN2) + 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6 + (PDMDIN3) + 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8 + (PDMDIN4) + + output drive type for the GPIO pin can be one of the following: + 0 - Hi-Z output + 1 - Drive active low and active high + 2 - (default) Drive active low and weak high + 3 - Drive active low and Hi-Z + 4 - Drive weak low and active high + 5 - Drive Hi-Z and active high + + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32-array + - minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + required: - compatible - reg @@ -156,6 +199,7 @@ examples: ti,mic-bias-source = <6>; ti,pdm-edge-select = <0 1 0 1>; ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; ti,gpo-config-1 = <0 0>; ti,gpo-config-2 = <0 0>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; -- 2.20.1

2 2

[PATCH] ASoC: SOF: imx: add missing MODULE_LICENSE() for imx-common
by Kai Vehmanen 21 Sep '20

21 Sep '20

Fix build warning: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/sof/imx/imx-common.o Fixes: 18ebffe4d043 ("ASoC: SOF: imx: Add debug support for imx platforms") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/imx/imx-common.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/imx/imx-common.c b/sound/soc/sof/imx/imx-common.c index 63d8a5d7bc44..5fee637834c2 100644 --- a/sound/soc/sof/imx/imx-common.c +++ b/sound/soc/sof/imx/imx-common.c @@ -4,6 +4,7 @@ // // Common helpers for the audio DSP on i.MX8 +#include <linux/module.h> #include <sound/sof/xtensa.h> #include "../ops.h" @@ -70,3 +71,5 @@ void imx8_dump(struct snd_sof_dev *sdev, u32 flags) IMX8_STACK_DUMP_SIZE); } EXPORT_SYMBOL(imx8_dump); + +MODULE_LICENSE("Dual BSD/GPL"); -- 2.27.0

2 1

[PATCH] ASoC: cs42l51: add additional ADC volume controls
by Guillermo Rodríguez 21 Sep '20

21 Sep '20

Add volume controls for: - Analog programmable gain amplifier (PGA) (-3 .. +12 dB) - ADC attenuator (0 .. -96 dB) Signed-off-by: Guillermo Rodríguez <guille.rodriguez(a)gmail.com> --- sound/soc/codecs/cs42l51.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 764f2ef8f59d..097c4e8d9950 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -122,6 +122,9 @@ static const char *chan_mix[] = { "R L", }; +static const DECLARE_TLV_DB_SCALE(pga_tlv, -300, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_att_tlv, -9600, 100, 0); + static SOC_ENUM_SINGLE_EXT_DECL(cs42l51_chan_mix, chan_mix); static const struct snd_kcontrol_new cs42l51_snd_controls[] = { @@ -138,6 +141,12 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), + SOC_DOUBLE_R_SX_TLV("ADC Attenuator Volume", + CS42L51_ADCA_ATT, CS42L51_ADCB_ATT, + 0, 0xA0, 96, adc_att_tlv), + SOC_DOUBLE_R_SX_TLV("PGA Volume", + CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, + 0, 0x1A, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.26.0

3 2

[PATCH 1/2] dt-bindings: tas2562: Add TAS2564 to binding
by Dan Murphy 21 Sep '20

21 Sep '20

Add the TAS2564 compatible and data sheet link to the binding. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tas2562.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index c3b7e19a0d44..d3ceda356aaa 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -19,12 +19,14 @@ description: | Specifications about the audio amplifier can be found at: https://www.ti.com/lit/gpn/tas2562 https://www.ti.com/lit/gpn/tas2563 + https://www.ti.com/lit/gpn/tas2564 properties: compatible: enum: - ti,tas2562 - ti,tas2563 + - ti,tas2564 reg: maxItems: 1 -- 2.28.0

2 2

[PATCH] ASoC: SOF: topology: fix the process being scheduled on core0 always
by Kai Vehmanen 21 Sep '20

21 Sep '20

From: Keyon Jie <yang.jie(a)linux.intel.com> In commit 783898ce68de ("ASoC: SOF: append extended data to sof_ipc_comp_process") the process components are set to run on the fixed core 0, this break us from scheduling components on any other DSP core. Since we can get the DSP core index from swidget->core, it is duplicated to pass the extra 'core' argument for those sof_widget_load_xx() functions. Here removes the duplicate 'core' argument and get component core from swidget->core directly to fix the issue mentioned above. Fixes: 783898ce68de ("ASoC: SOF: append extended data to sof_ipc_comp_process") Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/topology.c | 82 +++++++++++++++++----------------------- 1 file changed, 35 insertions(+), 47 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index eaa1122d5a68..69313fbdb636 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1465,13 +1465,11 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, * @ipc_size: IPC payload size that will be updated depending on valid * extended data. * @index: ID of the pipeline the component belongs to - * @core: index of the DSP core that the component should run on * * Return: The pointer to the new allocated component, NULL if failed. */ static struct sof_ipc_comp *sof_comp_alloc(struct snd_sof_widget *swidget, - size_t *ipc_size, int index, - int core) + size_t *ipc_size, int index) { u8 nil_uuid[SOF_UUID_SIZE] = {0}; struct sof_ipc_comp *comp; @@ -1490,7 +1488,7 @@ static struct sof_ipc_comp *sof_comp_alloc(struct snd_sof_widget *swidget, comp->hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; comp->id = swidget->comp_id; comp->pipeline_id = index; - comp->core = core; + comp->core = swidget->core; /* handle the extended data if needed */ if (total_size > *ipc_size) { @@ -1505,7 +1503,7 @@ static struct sof_ipc_comp *sof_comp_alloc(struct snd_sof_widget *swidget, } static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r, struct snd_sof_dai *dai) @@ -1517,7 +1515,7 @@ static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, int ret; comp_dai = (struct sof_ipc_comp_dai *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!comp_dai) return -ENOMEM; @@ -1571,7 +1569,7 @@ static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_buffer(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1590,7 +1588,7 @@ static int sof_widget_load_buffer(struct snd_soc_component *scomp, int index, buffer->comp.id = swidget->comp_id; buffer->comp.type = SOF_COMP_BUFFER; buffer->comp.pipeline_id = index; - buffer->comp.core = core; + buffer->comp.core = swidget->core; ret = sof_parse_tokens(scomp, buffer, buffer_tokens, ARRAY_SIZE(buffer_tokens), private->array, @@ -1642,7 +1640,7 @@ static int spcm_bind(struct snd_soc_component *scomp, struct snd_sof_pcm *spcm, */ static int sof_widget_load_pcm(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, enum sof_ipc_stream_direction dir, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) @@ -1654,7 +1652,7 @@ static int sof_widget_load_pcm(struct snd_soc_component *scomp, int index, int ret; host = (struct sof_ipc_comp_host *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!host) return -ENOMEM; @@ -1779,7 +1777,7 @@ static int sof_widget_load_pipeline(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1790,7 +1788,7 @@ static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, int ret; mixer = (struct sof_ipc_comp_mixer *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!mixer) return -ENOMEM; @@ -1824,7 +1822,7 @@ static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, * Mux topology */ static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1835,7 +1833,7 @@ static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, int ret; mux = (struct sof_ipc_comp_mux *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!mux) return -ENOMEM; @@ -1870,7 +1868,7 @@ static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1884,7 +1882,7 @@ static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, int ret; volume = (struct sof_ipc_comp_volume *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!volume) return -ENOMEM; @@ -1946,7 +1944,7 @@ static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_src(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1957,7 +1955,7 @@ static int sof_widget_load_src(struct snd_soc_component *scomp, int index, int ret; src = (struct sof_ipc_comp_src *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!src) return -ENOMEM; @@ -2003,7 +2001,7 @@ static int sof_widget_load_src(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_asrc(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -2014,7 +2012,7 @@ static int sof_widget_load_asrc(struct snd_soc_component *scomp, int index, int ret; asrc = (struct sof_ipc_comp_asrc *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!asrc) return -ENOMEM; @@ -2062,7 +2060,7 @@ static int sof_widget_load_asrc(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -2073,7 +2071,7 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, int ret; tone = (struct sof_ipc_comp_tone *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!tone) return -ENOMEM; @@ -2229,7 +2227,7 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, } process = (struct sof_ipc_comp_process *) - sof_comp_alloc(swidget, &ipc_size, index, 0); + sof_comp_alloc(swidget, &ipc_size, index); if (!process) { ret = -ENOMEM; goto out; @@ -2307,7 +2305,7 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_process(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -2322,7 +2320,7 @@ static int sof_widget_load_process(struct snd_soc_component *scomp, int index, } memset(&config, 0, sizeof(config)); - config.comp.core = core; + config.comp.core = swidget->core; /* get the process token */ ret = sof_parse_tokens(scomp, &config, process_tokens, @@ -2450,8 +2448,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, return -ENOMEM; } - ret = sof_widget_load_dai(scomp, index, swidget, comp.core, - tw, &reply, dai); + ret = sof_widget_load_dai(scomp, index, swidget, tw, &reply, dai); if (ret == 0) { sof_connect_dai_widget(scomp, w, tw, dai); list_add(&dai->list, &sdev->dai_list); @@ -2461,12 +2458,10 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, } break; case snd_soc_dapm_mixer: - ret = sof_widget_load_mixer(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_mixer(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_pga: - ret = sof_widget_load_pga(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_pga(scomp, index, swidget, tw, &reply); /* Find scontrol for this pga and set readback offset*/ list_for_each_entry(scontrol, &sdev->kcontrol_list, list) { if (scontrol->comp_id == swidget->comp_id) { @@ -2476,41 +2471,34 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, } break; case snd_soc_dapm_buffer: - ret = sof_widget_load_buffer(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_buffer(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_scheduler: - ret = sof_widget_load_pipeline(scomp, index, swidget, - tw, &reply); + ret = sof_widget_load_pipeline(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_aif_out: - ret = sof_widget_load_pcm(scomp, index, swidget, comp.core, + ret = sof_widget_load_pcm(scomp, index, swidget, SOF_IPC_STREAM_CAPTURE, tw, &reply); break; case snd_soc_dapm_aif_in: - ret = sof_widget_load_pcm(scomp, index, swidget, comp.core, + ret = sof_widget_load_pcm(scomp, index, swidget, SOF_IPC_STREAM_PLAYBACK, tw, &reply); break; case snd_soc_dapm_src: - ret = sof_widget_load_src(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_src(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_asrc: - ret = sof_widget_load_asrc(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_asrc(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_siggen: - ret = sof_widget_load_siggen(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_siggen(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_effect: - ret = sof_widget_load_process(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_process(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_mux: case snd_soc_dapm_demux: - ret = sof_widget_load_mux(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_mux(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_switch: case snd_soc_dapm_dai_link: -- 2.27.0

2 1