November 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda: Remove redundant runtime PM calls
by Takashi Iwai 19 Nov '21

19 Nov '21

The previous fix for more comprehensive runtime PM calls turned out to be not good as hoped; a few calls including pm_runtime_enable() and pm_runtime_disable() are rather utterly superfluous for PCI devices, even triggering a kernel error message. Better to drop those calls. Note that the problem we wanted to solve with that commit seems irrelevant with the fix itself; the original bug (a GPF at azx_remove()) was likely a regression by the recent PCI core cleanup, and the buggy PCI change has been already reverted. So basically we were scratching a wrong surface. OTOH, making the runtime PM calls symmetric for both probe and remove is more consistent, and maybe that's a sensible outcome. Fixes: 4f66a9ef37d3 ("ALSA: hda: intel: More comprehensive PM runtime setup for controller driver") Reported-by: Heiner Kallweit <hkallweit1(a)gmail.com> Link: https://lore.kernel.org/r/d9d76980-966a-e031-70d1-3254ba5be5eb@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 45e85180048c..221afacbc7fd 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1349,8 +1349,6 @@ static void azx_free(struct azx *chip) if (azx_has_pm_runtime(chip) && chip->running) { pm_runtime_get_noresume(&pci->dev); - pm_runtime_disable(&pci->dev); - pm_runtime_set_suspended(&pci->dev); pm_runtime_forbid(&pci->dev); pm_runtime_dont_use_autosuspend(&pci->dev); } @@ -2328,8 +2326,6 @@ static int azx_probe_continue(struct azx *chip) if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); pm_runtime_allow(&pci->dev); - pm_runtime_set_active(&pci->dev); - pm_runtime_enable(&pci->dev); pm_runtime_put_autosuspend(&pci->dev); } -- 2.31.1

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[PATCH] ALSA: usb-audio: Don't start stream for capture at prepare
by Takashi Iwai 19 Nov '21

19 Nov '21

The recent change made mistakenly the stream for capture started at prepare stage. Add the stream direction check to avoid it. Fixes: 9c9a3b9da891 ("ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 57b046e73bfe..cec6e91afea2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -640,7 +640,8 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) runtime->delay = 0; subs->lowlatency_playback = lowlatency_playback_available(runtime, subs); - if (!subs->lowlatency_playback) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + !subs->lowlatency_playback) ret = start_endpoints(subs); unlock: -- 2.26.2

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[PATCH] ALSA: usb-audio: Switch back to non-latency mode at a later point
by Takashi Iwai 19 Nov '21

19 Nov '21

The recent regression report revealed that the judgment of the low-latency playback mode based on the runtime->stop_threshold cannot work reliably at the prepare stage, as sw_params call may happen at any time, and PCM dmix actually sets it up after the prepare call. This ended up with the stall of the stream as PCM ack won't be issued at all. For addressing this, check the free-wheeling mode again at the PCM trigger right before starting the stream again, and allow switching to the non-LL mode at a late stage. Fixes: d5f871f89e21 ("ALSA: usb-audio: Improved lowlatency playback support") Reported-and-tested-by: Kirill A. Shutemov <kirill.shutemov(a)linux.intel.com> Link: https://lore.kernel.org/r/20211117161855.m45mxcqszkfcetai@box.shutemov.name Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 95ec8eec1bb0..57b046e73bfe 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -581,6 +581,12 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) return 0; } +/* free-wheeling mode? (e.g. dmix) */ +static int in_free_wheeling_mode(struct snd_pcm_runtime *runtime) +{ + return runtime->stop_threshold > runtime->buffer_size; +} + /* check whether early start is needed for playback stream */ static int lowlatency_playback_available(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) @@ -592,8 +598,7 @@ static int lowlatency_playback_available(struct snd_pcm_runtime *runtime, /* disabled via module option? */ if (!chip->lowlatency) return false; - /* free-wheeling mode? (e.g. dmix) */ - if (runtime->stop_threshold > runtime->buffer_size) + if (in_free_wheeling_mode(runtime)) return false; /* implicit feedback mode has own operation mode */ if (snd_usb_endpoint_implicit_feedback_sink(subs->data_endpoint)) @@ -1552,6 +1557,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea subs); if (subs->lowlatency_playback && cmd == SNDRV_PCM_TRIGGER_START) { + if (in_free_wheeling_mode(substream->runtime)) + subs->lowlatency_playback = false; err = start_endpoints(subs); if (err < 0) { snd_usb_endpoint_set_callback(subs->data_endpoint, -- 2.26.2

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[PATCH v2 0/5] ASoC: SOF: Add PM support for i.MX8/i.MX8X/i.MX8M
by Daniel Baluta 19 Nov '21

19 Nov '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds support for System PM and Runtime PM with SOF for i.MX8 platforms. First patch adds common code to manage clocks, patch 2/3 adds the actual PM support, patch 4/5 implement the DSP start / reset callbacks for i.mx8m. Changes since v1: * rebased on top of Mark's for-next branch Daniel Baluta (5): ASoC: SOF: imx: Add code to manage DSP related clocks ASoC: SOF: imx8: Add runtime PM / System PM support ASoC: SOF: imx8m: Add runtime PM / System PM support ASoC: SOF: imx8m: Implement DSP start ASoC: SOF: imx8m: Implement reset callback sound/soc/sof/imx/imx-common.c | 24 +++++ sound/soc/sof/imx/imx-common.h | 11 ++ sound/soc/sof/imx/imx8.c | 137 +++++++++++++++++++++++ sound/soc/sof/imx/imx8m.c | 191 ++++++++++++++++++++++++++++++++- 4 files changed, 362 insertions(+), 1 deletion(-) -- 2.27.0

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[PATCH v2 0/3] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 19 Nov '21

19 Nov '21

This patch series is to make lpass varient independent pin control functions common and to add lpass sc7280 pincontrol support. It also includes dt-bindings for lpass sc7280 lpi compatible. Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (3): pinctrl: qcom: Update lpass variant independent functions as generic dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl compatible pinctrl: qcom: Add SC7280 lpass pin configuration .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 4 +- drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 56 ++++++++++++++++++---- 2 files changed, 51 insertions(+), 9 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH] sound/oss/dmasound: fix build when some drivers are =m and others are =y
by Randy Dunlap 19 Nov '21

19 Nov '21

When CONFIG_DMASOUND_ATARI=y and CONFIG_DMASOUND_Q40=m, dmasound_atari.o is built first (listed first in the Makefile), so dmasound_core.o is built as builtin, not for use by loadable modules. Then dmasound_q40.o is built and linked with the already-built dmasound_core.o, but the latter does not support use by loadable modules. This causes the missing symbol to be undefined. Fixes this build error: ERROR: modpost: "dmasound_deinit" [sound/oss/dmasound/dmasound_q40.ko] undefined! Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") # "forever" Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- There may be some Makefile solution to this but I don't know what it is. I didn't want to spend lots of time on these old drivers. sound/oss/dmasound/Makefile | 6 +++--- sound/oss/dmasound/dmasound_atari.c | 2 +- sound/oss/dmasound/dmasound_core.c | 16 ---------------- sound/oss/dmasound/dmasound_paula.c | 2 +- sound/oss/dmasound/dmasound_q40.c | 2 +- 5 files changed, 6 insertions(+), 22 deletions(-) --- linux-next-20211117.orig/sound/oss/dmasound/dmasound_atari.c +++ linux-next-20211117/sound/oss/dmasound/dmasound_atari.c @@ -27,7 +27,7 @@ #include <asm/atariints.h> #include <asm/atari_stram.h> -#include "dmasound.h" +#include "dmasound_core.c" #define DMASOUND_ATARI_REVISION 0 #define DMASOUND_ATARI_EDITION 3 --- linux-next-20211117.orig/sound/oss/dmasound/dmasound_paula.c +++ linux-next-20211117/sound/oss/dmasound/dmasound_paula.c @@ -30,7 +30,7 @@ #include <asm/amigaints.h> #include <asm/machdep.h> -#include "dmasound.h" +#include "dmasound_core.c" #define DMASOUND_PAULA_REVISION 0 #define DMASOUND_PAULA_EDITION 4 --- linux-next-20211117.orig/sound/oss/dmasound/dmasound_q40.c +++ linux-next-20211117/sound/oss/dmasound/dmasound_q40.c @@ -25,7 +25,7 @@ #include <asm/q40ints.h> #include <asm/q40_master.h> -#include "dmasound.h" +#include "dmasound_core.c" #define DMASOUND_Q40_REVISION 0 #define DMASOUND_Q40_EDITION 3 --- linux-next-20211117.orig/sound/oss/dmasound/Makefile +++ linux-next-20211117/sound/oss/dmasound/Makefile @@ -3,6 +3,6 @@ # Makefile for the DMA sound driver # -obj-$(CONFIG_DMASOUND_ATARI) += dmasound_core.o dmasound_atari.o -obj-$(CONFIG_DMASOUND_PAULA) += dmasound_core.o dmasound_paula.o -obj-$(CONFIG_DMASOUND_Q40) += dmasound_core.o dmasound_q40.o +obj-$(CONFIG_DMASOUND_ATARI) += dmasound_atari.o +obj-$(CONFIG_DMASOUND_PAULA) += dmasound_paula.o +obj-$(CONFIG_DMASOUND_Q40) += dmasound_q40.o --- linux-next-20211117.orig/sound/oss/dmasound/dmasound_core.c +++ linux-next-20211117/sound/oss/dmasound/dmasound_core.c @@ -1570,19 +1570,3 @@ char dmasound_alaw2dma8[] = { 3, 3, 3, 3, 3, 3, 3, 3 }; #endif /* HAS_8BIT_TABLES */ - - /* - * Visible symbols for modules - */ - -EXPORT_SYMBOL(dmasound); -EXPORT_SYMBOL(dmasound_init); -#ifdef MODULE -EXPORT_SYMBOL(dmasound_deinit); -#endif -EXPORT_SYMBOL(dmasound_write_sq); -EXPORT_SYMBOL(dmasound_catchRadius); -#ifdef HAS_8BIT_TABLES -EXPORT_SYMBOL(dmasound_ulaw2dma8); -EXPORT_SYMBOL(dmasound_alaw2dma8); -#endif

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[PATCH] ALSA: ctxfi: Fix out-of-range access
by Takashi Iwai 18 Nov '21

18 Nov '21

The master and next_conj of rcs_ops are used for iterating the resource list entries, and currently those are supposed to return the current value. The problem is that next_conf may go over the last entry before the loop abort condition is evaluated, and it may return the "current" value that is beyond the array size. It was caught recently as a GPF, for example. Those return values are, however, never actually evaluated, hence basically we don't have to consider the current value as the return at all. By dropping those return values, the potential out-of-range access above is also fixed automatically. This patch changes the return type of master and next_conj callbacks to void and drop the superfluous code accordingly. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214985 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ctxfi/ctamixer.c | 14 ++++++-------- sound/pci/ctxfi/ctdaio.c | 16 ++++++++-------- sound/pci/ctxfi/ctresource.c | 7 +++---- sound/pci/ctxfi/ctresource.h | 4 ++-- sound/pci/ctxfi/ctsrc.c | 7 +++---- 5 files changed, 22 insertions(+), 26 deletions(-) diff --git a/sound/pci/ctxfi/ctamixer.c b/sound/pci/ctxfi/ctamixer.c index da6e6350ceaf..d074727c3e21 100644 --- a/sound/pci/ctxfi/ctamixer.c +++ b/sound/pci/ctxfi/ctamixer.c @@ -23,16 +23,15 @@ #define BLANK_SLOT 4094 -static int amixer_master(struct rsc *rsc) +static void amixer_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct amixer, rsc)->idx[0]; } -static int amixer_next_conj(struct rsc *rsc) +static void amixer_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct amixer, rsc)->idx[rsc->conj]; } static int amixer_index(const struct rsc *rsc) @@ -331,16 +330,15 @@ int amixer_mgr_destroy(struct amixer_mgr *amixer_mgr) /* SUM resource management */ -static int sum_master(struct rsc *rsc) +static void sum_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct sum, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct sum, rsc)->idx[0]; } -static int sum_next_conj(struct rsc *rsc) +static void sum_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct sum, rsc)->idx[rsc->conj]; } static int sum_index(const struct rsc *rsc) diff --git a/sound/pci/ctxfi/ctdaio.c b/sound/pci/ctxfi/ctdaio.c index f589da045342..7fc720046ce2 100644 --- a/sound/pci/ctxfi/ctdaio.c +++ b/sound/pci/ctxfi/ctdaio.c @@ -51,12 +51,12 @@ static const struct daio_rsc_idx idx_20k2[NUM_DAIOTYP] = { [SPDIFIO] = {.left = 0x05, .right = 0x85}, }; -static int daio_master(struct rsc *rsc) +static void daio_master(struct rsc *rsc) { /* Actually, this is not the resource index of DAIO. * For DAO, it is the input mapper index. And, for DAI, * it is the output time-slot index. */ - return rsc->conj = rsc->idx; + rsc->conj = rsc->idx; } static int daio_index(const struct rsc *rsc) @@ -64,19 +64,19 @@ static int daio_index(const struct rsc *rsc) return rsc->conj; } -static int daio_out_next_conj(struct rsc *rsc) +static void daio_out_next_conj(struct rsc *rsc) { - return rsc->conj += 2; + rsc->conj += 2; } -static int daio_in_next_conj_20k1(struct rsc *rsc) +static void daio_in_next_conj_20k1(struct rsc *rsc) { - return rsc->conj += 0x200; + rsc->conj += 0x200; } -static int daio_in_next_conj_20k2(struct rsc *rsc) +static void daio_in_next_conj_20k2(struct rsc *rsc) { - return rsc->conj += 0x100; + rsc->conj += 0x100; } static const struct rsc_ops daio_out_rsc_ops = { diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index 81ad26934518..be1d3e61309c 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -109,18 +109,17 @@ static int audio_ring_slot(const struct rsc *rsc) return (rsc->conj << 4) + offset_in_audio_slot_block[rsc->type]; } -static int rsc_next_conj(struct rsc *rsc) +static void rsc_next_conj(struct rsc *rsc) { unsigned int i; for (i = 0; (i < 8) && (!(rsc->msr & (0x1 << i))); ) i++; rsc->conj += (AUDIO_SLOT_BLOCK_NUM >> i); - return rsc->conj; } -static int rsc_master(struct rsc *rsc) +static void rsc_master(struct rsc *rsc) { - return rsc->conj = rsc->idx; + rsc->conj = rsc->idx; } static const struct rsc_ops rsc_generic_ops = { diff --git a/sound/pci/ctxfi/ctresource.h b/sound/pci/ctxfi/ctresource.h index fdbfd808816d..58553bda44f4 100644 --- a/sound/pci/ctxfi/ctresource.h +++ b/sound/pci/ctxfi/ctresource.h @@ -39,8 +39,8 @@ struct rsc { }; struct rsc_ops { - int (*master)(struct rsc *rsc); /* Move to master resource */ - int (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */ + void (*master)(struct rsc *rsc); /* Move to master resource */ + void (*next_conj)(struct rsc *rsc); /* Move to next conjugate resource */ int (*index)(const struct rsc *rsc); /* Return the index of resource */ /* Return the output slot number */ int (*output_slot)(const struct rsc *rsc); diff --git a/sound/pci/ctxfi/ctsrc.c b/sound/pci/ctxfi/ctsrc.c index bd4697b44233..4a94b4708a77 100644 --- a/sound/pci/ctxfi/ctsrc.c +++ b/sound/pci/ctxfi/ctsrc.c @@ -590,16 +590,15 @@ int src_mgr_destroy(struct src_mgr *src_mgr) /* SRCIMP resource manager operations */ -static int srcimp_master(struct rsc *rsc) +static void srcimp_master(struct rsc *rsc) { rsc->conj = 0; - return rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0]; + rsc->idx = container_of(rsc, struct srcimp, rsc)->idx[0]; } -static int srcimp_next_conj(struct rsc *rsc) +static void srcimp_next_conj(struct rsc *rsc) { rsc->conj++; - return container_of(rsc, struct srcimp, rsc)->idx[rsc->conj]; } static int srcimp_index(const struct rsc *rsc) -- 2.26.2

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[GIT PULL] ASoC fixes for v5.16-rc1
by Mark Brown 18 Nov '21

18 Nov '21

The following changes since commit 6195eb15f6d60dd92d1644dc11f1c1c2e84ebfeb: Merge series "Multiple headphone codec driver support" from Brent Lu <brent.lu(a)intel.com>: (2021-10-29 21:53:41 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.16-rc1 for you to fetch changes up to 424fe7edbed18d47f7b97f7e1322a6f8969b77ae: ASoC: stm32: i2s: fix 32 bits channel length without mclk (2021-11-17 13:04:38 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.16 A relatively large collection of fixes that came in since the merge window, though a lot of this is just a collection of new machine quirks for x86 platforms. ---------------------------------------------------------------- AngeloGioacchino Del Regno (2): ASoC: mediatek: mt8173-rt5650: Rename Speaker control to Ext Spk ASoC: mediatek: mt8173: Fix debugfs registration for components Arnd Bergmann (1): ASoC: SOF: build compression interface into snd_sof.ko Charles Keepax (1): ASoC: cs35l41: Change monitor widgets to siggens ChiYuan Huang (3): ASoC: rt9120: Update internal ocp level to the correct value ASoC: rt9120: Fix clock auto sync issue when fs is the multiple of 48 ASoC: rt9120: Add the compatibility with rt9120s Derek Fang (2): ASoC: rt5682: Avoid the unexpected IRQ event during going to suspend ASoC: rt5682: Re-detect the combo jack after resuming Gongjun Song (9): ASoC: Intel: sof_sdw: Add support for SKU 0AF3 product ASoC: Intel: soc-acpi: add SKU 0AF3 SoundWire configuration ASoC: Intel: sof_sdw: Add support for SKU 0B00 and 0B01 products ASoC: Intel: sof_sdw: Add support for SKU 0B11 product ASoC: Intel: sof_sdw: Add support for SKU 0B13 product ASoC: Intel: soc-acpi: add SKU 0B13 SoundWire configuration ASoC: Intel: sof_sdw: Add support for SKU 0B29 product ASoC: Intel: soc-acpi: add SKU 0B29 SoundWire configuration ASoC: Intel: sof_sdw: Add support for SKU 0B12 product Jack Yu (1): ASoC: rt1011: revert 'I2S Reference' to SOC_ENUM_EXT Kai Vehmanen (1): ASoC: SOF: Intel: hda: fix hotplug when only codec is suspended Kuninori Morimoto (1): ASoC: rsnd: fixup DMAEngine API Mark Brown (1): ASoC: Merge rt9120 series from ChiYuan Huang: Olivier Moysan (1): ASoC: stm32: i2s: fix 32 bits channel length without mclk Peter Ujfalusi (1): ASoC: SOF:control: Fix variable type in snd_sof_refresh_control() Simon Trimmer (1): ASoC: wm_adsp: wm_adsp_control_add() error: uninitialized symbol 'ret' Srinivas Kandagatla (8): ASoC: qdsp6: qdsp6: q6prm: handle clk disable correctly ASoC: qdsp6: q6routing: Conditionally reset FrontEnd Mixer ASoC: qdsp6: q6asm: fix q6asm_dai_prepare error handling ASoC: qdsp6: q6adm: improve error reporting ASoC: qdsp6: q6routing: validate port id before setting up route ASoC: codecs: wcd938x: fix volatile register range ASoC: codecs: wcd934x: return error code correctly from hw_params ASoC: codecs: lpass-rx-macro: fix HPHR setting CLSH mask Takashi Iwai (2): ASoC: DAPM: Cover regression by kctl change notification fix ASoC: topology: Add missing rwsem around snd_ctl_remove() calls sound/soc/codecs/cs35l41.c | 14 +-- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/rt1011.c | 55 ++++++++++-- sound/soc/codecs/rt1011.h | 7 ++ sound/soc/codecs/rt5682-i2c.c | 1 + sound/soc/codecs/rt5682.c | 38 ++++++-- sound/soc/codecs/rt5682.h | 1 + sound/soc/codecs/rt9120.c | 58 +++++++++--- sound/soc/codecs/wcd934x.c | 3 +- sound/soc/codecs/wcd938x.c | 3 + sound/soc/codecs/wm_adsp.c | 5 +- sound/soc/intel/boards/sof_sdw.c | 69 ++++++++++++++ sound/soc/intel/common/soc-acpi-intel-adl-match.c | 105 ++++++++++++++++++++++ sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 51 +++++++++-- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 8 +- sound/soc/qcom/qdsp6/audioreach.h | 4 + sound/soc/qcom/qdsp6/q6adm.c | 4 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 19 ++-- sound/soc/qcom/qdsp6/q6prm.c | 53 ++++++++++- sound/soc/qcom/qdsp6/q6routing.c | 12 ++- sound/soc/sh/rcar/dma.c | 2 +- sound/soc/soc-dapm.c | 29 ++++-- sound/soc/soc-topology.c | 3 + sound/soc/sof/Kconfig | 2 +- sound/soc/sof/control.c | 8 +- sound/soc/sof/intel/hda-bus.c | 17 ++++ sound/soc/sof/intel/hda-dsp.c | 3 +- sound/soc/sof/intel/hda.c | 16 ++++ sound/soc/stm/stm32_i2s.c | 2 +- 29 files changed, 519 insertions(+), 75 deletions(-)

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[PATCH] ASoC: codecs: MBHC: Remove useless condition check
by Srinivasa Rao Mandadapu 18 Nov '21

18 Nov '21

Remove redundant conditional check and clean code in special headset support functions. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> Co-developed-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> --- sound/soc/codecs/wcd-mbhc-v2.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index d6545e4..99586ce 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -1055,12 +1055,8 @@ static bool wcd_mbhc_check_for_spl_headset(struct wcd_mbhc *mbhc) hs_threshold = wcd_mbhc_get_spl_hs_thres(mbhc); hph_threshold = wcd_mbhc_adc_get_hph_thres(mbhc); - if (output_mv > hs_threshold || output_mv < hph_threshold) { - if (mbhc->force_linein == true) - is_spl_hs = false; - } else { + if (!(output_mv > hs_threshold || output_mv < hph_threshold)) is_spl_hs = true; - } /* Back MIC_BIAS2 to 1.8v if the type is not special headset */ if (!is_spl_hs) { @@ -1149,13 +1145,13 @@ static void wcd_correct_swch_plug(struct work_struct *work) plug_type = wcd_mbhc_get_plug_from_adc(mbhc, output_mv); is_pa_on = wcd_mbhc_read_field(mbhc, WCD_MBHC_HPH_PA_EN); - if ((output_mv > hs_threshold) && (!is_spl_hs)) { + if (output_mv > hs_threshold && !is_spl_hs) { is_spl_hs = wcd_mbhc_check_for_spl_headset(mbhc); output_mv = wcd_measure_adc_once(mbhc, MUX_CTL_IN2P); if (is_spl_hs) { - hs_threshold = (hs_threshold * wcd_mbhc_get_micbias(mbhc)) / - micbias_mv; + hs_threshold *= wcd_mbhc_get_micbias(mbhc); + hs_threshold /= micbias_mv; } } @@ -1185,7 +1181,7 @@ static void wcd_correct_swch_plug(struct work_struct *work) } /* cable is extension cable */ - if (output_mv > hs_threshold || mbhc->force_linein == true) + if (output_mv > hs_threshold || mbhc->force_linein) plug_type = MBHC_PLUG_TYPE_HIGH_HPH; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH 0/6] ASoC: SOF: Add support for Mediatek MT8195
by Daniel Baluta 18 Nov '21

18 Nov '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds initial SOF support on Mediatek mt8195 platform. MT8195 has four Cortex A78 cores paired with four Cortex A55 cores. It also has Cadence HiFi-4 DSP single core. There are shared DRAM and mailbox interrupt between AP and DSP to use for IPC communication. YC Hung (6): ASoC: SOF: mediatek: Add mt8195 hardware support ASoC: SOF: tokens: add token for Mediatek AFE ASoC: SOF: topology: Add support for Mediatek AFE DAI ASoC: SOF: mediatek: Add fw loader and mt8195 dsp ops to load firmware ASoC: SOF: Add mt8195 device descriptor ASoC: SOF: mediatek: Add dai driver dsp ops callback for mt8195 include/sound/sof/dai-mediatek.h | 23 ++ include/sound/sof/dai.h | 3 + include/uapi/sound/sof/tokens.h | 5 + sound/soc/sof/Kconfig | 3 +- sound/soc/sof/Makefile | 1 + sound/soc/sof/mediatek/Kconfig | 33 ++ sound/soc/sof/mediatek/Makefile | 2 + sound/soc/sof/mediatek/adsp_helper.h | 49 +++ sound/soc/sof/mediatek/mediatek-ops.h | 8 + sound/soc/sof/mediatek/mt8195/Makefile | 4 + sound/soc/sof/mediatek/mt8195/mt8195-loader.c | 56 +++ sound/soc/sof/mediatek/mt8195/mt8195.c | 377 ++++++++++++++++++ sound/soc/sof/mediatek/mt8195/mt8195.h | 158 ++++++++ sound/soc/sof/pcm.c | 12 + sound/soc/sof/sof-of-dev.c | 13 + sound/soc/sof/topology.c | 59 +++ 16 files changed, 805 insertions(+), 1 deletion(-) create mode 100644 include/sound/sof/dai-mediatek.h create mode 100644 sound/soc/sof/mediatek/Kconfig create mode 100644 sound/soc/sof/mediatek/Makefile create mode 100644 sound/soc/sof/mediatek/adsp_helper.h create mode 100644 sound/soc/sof/mediatek/mediatek-ops.h create mode 100644 sound/soc/sof/mediatek/mt8195/Makefile create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195-loader.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195.c create mode 100644 sound/soc/sof/mediatek/mt8195/mt8195.h -- 2.27.0

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