July 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH v2] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 31 Jul '24

31 Jul '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. As the ALSA controls for the firmware coefficients are not used in normal operation they can all be hidden, but add a kernel parameter so that they can be re-enabled for debugging. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- Changes in v2: - v1 was accidentally the backport version for older kernels instead of the one for v6.11 sound/pci/hda/cs35l56_hda.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 96d3f13c5abf..f2f623f713d1 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -23,6 +23,10 @@ #include "hda_cs_dsp_ctl.h" #include "hda_generic.h" +static bool expose_dsp_controls; +module_param(expose_dsp_controls, bool, 0444); +MODULE_PARM_DESC(expose_dsp_controls, "Expose firmware controls as ALSA controls 0=no (default), 1=yes"); + /* * The cs35l56_hda_dai_config[] reg sequence configures the device as * ASP1_BCLK_FREQ = 3.072 MHz @@ -613,7 +617,7 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) */ if (cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); - else + else if (expose_dsp_controls) add_dsp_controls_required = true; cs35l56->base.fw_patched = false; -- 2.43.0

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[PATCH][LINUX][ALSA][USB AUDIO CLASS 1] wrong channel ids for surround
by Sylvain BERTRAND 31 Jul '24

31 Jul '24

Hi, In the USB audio class 1 specifications, page 33 (3.7.2.3 Audio Channel Cluster Format): Surround is spatially defined as [rear], hence left surround should be SNDRV_CHMAP_RL and not SND_CHMAP_SL (Side). Same for right surround channel. Right? I am not a mailing list subscriber, CC me please. regards, -- Sylvain BERTRAND --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -244,8 +244,8 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, SNDRV_CHMAP_FR, /* right front */ SNDRV_CHMAP_FC, /* center front */ SNDRV_CHMAP_LFE, /* LFE */ - SNDRV_CHMAP_SL, /* left surround */ - SNDRV_CHMAP_SR, /* right surround */ + SNDRV_CHMAP_RL, /* left surround */ + SNDRV_CHMAP_RR, /* right surround */ SNDRV_CHMAP_FLC, /* left of center */ SNDRV_CHMAP_FRC, /* right of center */ SNDRV_CHMAP_RC, /* surround */

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[PATCH -next] ASoC: fsl: lpc3xxx-i2s: Remove set but not used variable 'savedbitclkrate'
by Yue Haibing 31 Jul '24

31 Jul '24

The variable savedbitclkrate is assigned and never used, so can be removed. sound/soc/fsl/lpc3xxx-i2s.c:42:13: warning: variable ‘savedbitclkrate’ set but not used [-Wunused-but-set-variable] Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/lpc3xxx-i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index af995ca081a3..62ef624d6dd4 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -39,7 +39,7 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 { u32 i2srate; u32 idxx, idyy; - u32 savedbitclkrate, diff, trate, baseclk; + u32 diff, trate, baseclk; /* Adjust rate for sample size (bits) and 2 channels and offset for * divider in clock output @@ -53,14 +53,12 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 /* Find the best divider */ *clkx = *clky = 0; - savedbitclkrate = 0; diff = ~0; for (idxx = 1; idxx < 0xFF; idxx++) { for (idyy = 1; idyy < 0xFF; idyy++) { trate = (baseclk * idxx) / idyy; if (abs(trate - i2srate) < diff) { diff = abs(trate - i2srate); - savedbitclkrate = trate; *clkx = idxx; *clky = idyy; } -- 2.34.1

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[PATCH] ASoC: rsnd: use pcm_dmaengine code
by Kuninori Morimoto 31 Jul '24

31 Jul '24

rsnd is implementing own DMAEngine code, but we can replace it with pcm_dmaengine code, because these are almost same. Let's use existing and stable code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/core.c | 17 --------- sound/soc/sh/rcar/dma.c | 75 ++++------------------------------------ sound/soc/sh/rcar/rsnd.h | 1 - sound/soc/sh/rcar/ssi.c | 2 +- 5 files changed, 9 insertions(+), 87 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 7bddfd5e38d6c..426632996a0a3 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_RCAR depends on COMMON_CLK depends on OF select SND_SIMPLE_CARD_UTILS + select SND_DMAENGINE_PCM select REGMAP_MMIO help This option enables R-Car SRU/SCU/SSIU/SSI sound support diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6bc7027ed4dbf..233128c8ff145 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -660,23 +660,6 @@ static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) return rsnd_rdai_get(priv, dai->id); } -/* - * rsnd_soc_dai functions - */ -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io) -{ - struct snd_pcm_substream *substream = io->substream; - - /* - * this function should be called... - * - * - if rsnd_dai_pointer_update() returns true - * - without spin lock - */ - - snd_pcm_period_elapsed(substream); -} - static void rsnd_dai_stream_init(struct rsnd_dai_stream *io, struct snd_pcm_substream *substream) { diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7b499eee50806..2342bbb6fe92e 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -7,6 +7,7 @@ #include <linux/delay.h> #include <linux/of_dma.h> +#include <sound/dmaengine_pcm.h> #include "rsnd.h" /* @@ -22,8 +23,6 @@ struct rsnd_dmaen { struct dma_chan *chan; - dma_cookie_t cookie; - unsigned int dma_len; }; struct rsnd_dmapp { @@ -66,20 +65,6 @@ static struct rsnd_mod mem = { /* * Audio DMAC */ -static void __rsnd_dmaen_complete(struct rsnd_mod *mod, - struct rsnd_dai_stream *io) -{ - if (rsnd_io_is_working(io)) - rsnd_dai_period_elapsed(io); -} - -static void rsnd_dmaen_complete(void *data) -{ - struct rsnd_mod *mod = data; - - rsnd_mod_interrupt(mod, __rsnd_dmaen_complete); -} - static struct dma_chan *rsnd_dmaen_request_channel(struct rsnd_dai_stream *io, struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) @@ -98,13 +83,7 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - - if (dmaen->chan) - dmaengine_terminate_async(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_STOP); } static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, @@ -120,7 +99,7 @@ static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, * Let's call it under prepare */ if (dmaen->chan) - dma_release_channel(dmaen->chan); + snd_dmaengine_pcm_close_release_chan(io->substream); dmaen->chan = NULL; @@ -153,7 +132,7 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, return -EIO; } - return 0; + return snd_dmaengine_pcm_open(io->substream, dmaen->chan); } static int rsnd_dmaen_start(struct rsnd_mod *mod, @@ -162,12 +141,9 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, { struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct snd_pcm_substream *substream = io->substream; struct device *dev = rsnd_priv_to_dev(priv); - struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - int is_play = rsnd_io_is_play(io); int ret; /* @@ -195,7 +171,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, } } - cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + cfg.direction = snd_pcm_substream_to_dma_direction(io->substream); cfg.src_addr = dma->src_addr; cfg.dst_addr = dma->dst_addr; cfg.src_addr_width = buswidth; @@ -209,32 +185,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - substream->runtime->dma_addr, - snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), - is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - - if (!desc) { - dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); - return -EIO; - } - - desc->callback = rsnd_dmaen_complete; - desc->callback_param = rsnd_mod_get(dma); - - dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); - - dmaen->cookie = dmaengine_submit(desc); - if (dmaen->cookie < 0) { - dev_err(dev, "dmaengine_submit() fail\n"); - return -EIO; - } - - dma_async_issue_pending(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_START); } struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, @@ -307,19 +258,7 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod, struct rsnd_dai_stream *io, snd_pcm_uframes_t *pointer) { - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct dma_tx_state state; - enum dma_status status; - unsigned int pos = 0; - - status = dmaengine_tx_status(dmaen->chan, dmaen->cookie, &state); - if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { - if (state.residue > 0 && state.residue <= dmaen->dma_len) - pos = dmaen->dma_len - state.residue; - } - *pointer = bytes_to_frames(runtime, pos); + *pointer = snd_dmaengine_pcm_pointer(io->substream); return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ff294aa2d6407..dba684e4334b1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -576,7 +576,6 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai, #define rsnd_rdai_width_get(rdai) \ rsnd_rdai_width_ctrl(rdai, 0) int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width); -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io); int rsnd_dai_connect(struct rsnd_mod *mod, struct rsnd_dai_stream *io, enum rsnd_mod_type type); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 8d2a86383ae01..b3d4e8ae07eff 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -706,7 +706,7 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, spin_unlock(&priv->lock); if (elapsed) - rsnd_dai_period_elapsed(io); + snd_pcm_period_elapsed(io->substream); if (stop) snd_pcm_stop_xrun(io->substream); -- 2.43.0

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[PATCH] ASoC: rsnd: remove rsnd_mod_confirm_ssi() under DEBUG
by Kuninori Morimoto 31 Jul '24

31 Jul '24

rsnd_mod_confirm_ssi() confirms mod sanity, it should always be confirmed, not only when DEBUG. This patch tidyup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 9 --------- 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index afd69c6eb6544..0f190abf00e75 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -262,7 +262,7 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, int id = rsnd_mod_id(src_mod); int shift = (id % 2) ? 16 : 0; - rsnd_mod_confirm_src(src_mod); + rsnd_mod_make_sure(src_mod, RSND_MOD_SRC); rsnd_adg_get_timesel_ratio(priv, io, in_rate, out_rate, @@ -291,7 +291,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) int shift = (id % 4) * 8; u32 mask = 0xFF << shift; - rsnd_mod_confirm_ssi(ssi_mod); + rsnd_mod_make_sure(ssi_mod, RSND_MOD_SSI); val = val << shift; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ff294aa2d6407..8cf5d9001f437 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -871,15 +871,6 @@ void rsnd_cmd_remove(struct rsnd_priv *priv); int rsnd_cmd_attach(struct rsnd_dai_stream *io, int id); void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type); -#ifdef DEBUG -#define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI) -#define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC) -#define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC) -#else -#define rsnd_mod_confirm_ssi(mssi) -#define rsnd_mod_confirm_src(msrc) -#define rsnd_mod_confirm_dvc(mdvc) -#endif /* * If you don't need interrupt status debug message, -- 2.43.0

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[PATCH 0/2] ASoC: fsl_micfil: Check the difference for i.MX8 and i.MX9
by Shengjiu Wang 31 Jul '24

31 Jul '24

There are some register difference for i.MX8 and i.MX9 REG_MICFIL_FIFO_CTRL definition is updated. REG_MICFIL_FSYNC_CTRL, REG_MICFIL_VERID, REG_MICFIL_PARAM are added from i.MX9 Shengjiu Wang (2): ASoC: fsl_micfil: Expand the range of FIFO watermark mask ASoC: fsl_micfil: Differentiate register access permission for platforms sound/soc/fsl/fsl_micfil.c | 20 +++++++++++++++----- sound/soc/fsl/fsl_micfil.h | 2 +- 2 files changed, 16 insertions(+), 6 deletions(-) -- 2.34.1

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Re: [PATCH v1] ALSA: hda: cs35l41: Stop creating ALSA Controls for firmware coefficients
by Jaroslav Kysela 31 Jul '24

31 Jul '24

On 30. 07. 24 16:37, Stefan Binding wrote: > Add a kernel parameter to allow coefficients to be exposed as ALSA controls. > > When the CS35L41 loads its firmware, it has a number of controls to > affect its behaviour. Currently, these controls are exposed as ALSA > Controls by default. > > However, nothing in userspace currently uses them, and is unlikely to > do so in the future, therefore we don't need to create ASLA controls > for them. > > These controls can be useful for debug, so we can add a kernel > parameter to re-enable them if necessary. > > Disabling these controls would prevent userspace from trying to read > these controls when the CS35L41 is hibernating, which ordinarily > would result in an error message. This is probably not a right argument to add this code. The codec should be powered up when those controls are accessed or those controls should be cached by the driver. Although the controls have not been used yet, exposing them in this way is not ideal. Could you fix the driver (no I/O errors)? Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[PATCH alsa-lib 0/4] Add API helper functions for creating UMP Endpoint and Blocks
by Takashi Iwai 31 Jul '24

31 Jul '24

Hi, this is a patch set to alsa-lib to add the new API functions for creating a virtual UMP Endpoint and UMP Blocks arbitrarily from a user-space, that is, to create a virtual UMP device. An example program is included in test directory. Takashi === Takashi Iwai (4): ump: Add missing *_set variants for snd_ump_endpoint_info and snd_ump_block_info seq: Add API helper functions for creating UMP Endpoint and Blocks test: Add an example program to create a virtual UMP Endpoint Add test/seq-ump-example to .gitignore .gitignore | 1 + include/seqmid.h | 7 + include/ump.h | 36 ++++- src/rawmidi/ump.c | 304 +++++++++++++++++++++++++++++++++++++++-- src/seq/seq.c | 6 +- src/seq/seq_local.h | 4 + src/seq/seqmid.c | 249 +++++++++++++++++++++++++++++++++ test/Makefile.am | 3 +- test/seq-ump-example.c | 187 +++++++++++++++++++++++++ 9 files changed, 783 insertions(+), 14 deletions(-) create mode 100644 test/seq-ump-example.c -- 2.43.0

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[PATCH v4 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 31 Jul '24

31 Jul '24

This patchset serves to prevent an AB/BA deadlock: thread 0: * (lock A) acquire substream lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) wait for tasklet to finish by calling tasklet_unlock_spin_wait() in tasklet_disable_in_atomic() in ohci_flush_iso_completions() of ohci.c thread 1: * (lock B) enter tasklet * (lock A) attempt to acquire substream lock, waiting for it to be released: snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() in update_pcm_pointers() in process_ctx_payloads() in process_rx_packets() of amdtp-stream.c ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Cc: stable(a)vger.kernel.org Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): Revert "ALSA: firewire-lib: obsolete workqueue for period update" Revert "ALSA: firewire-lib: operate for period elapse event in process context" sound/firewire/amdtp-stream.c | 38 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 25 insertions(+), 14 deletions(-) -- 2.45.2

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Bug in tlv320adc3101 driver
by Terry Sanders 31 Jul '24

31 Jul '24

Hi, There is a bug in the code for the tlv320adc3101 driver when setting up the D value for PLL. The D value is set via I2C using page 0 registers 7 (MSB) and 8 (LSB). The data sheet has the following information for the registers: "Page 0 / Register 7 will be updated when Page 0 / Register 8 is written immediately after Page 0 / Register 7 is written" and: "Page 0 / Register 8 must be written immediately after writing to Page 0 / Register 7" The function that implements this is adc3xxx_setup_pll() in file soc/codecs/tlv320adc3xxx.c line 1172 - 1189. The function is shown below: static void adc3xxx_setup_pll(struct snd_soc_component *component, int div_entry) { int i = div_entry; /* P & R values */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_PR, (adc3xxx_divs[i].pll_p << ADC3XXX_PLLP_SHIFT) | (adc3xxx_divs[i].pll_r << ADC3XXX_PLLR_SHIFT)); /* J value */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_J, adc3xxx_divs[i].pll_j & ADC3XXX_PLLJ_MASK); /* D value */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_LSB, adc3xxx_divs[i].pll_d & ADC3XXX_PLLD_LSB_MASK); snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_MSB, (adc3xxx_divs[i].pll_d >> 8) & ADC3XXX_PLLD_MSB_MASK); } In the function the LSB (register 8) is written first followed by the MSB (register 7). This is the wrong way around and should be swapped so that the MSB is written first. Regards Terry Sanders

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[PATCH v1] ASoc: tas2781: Move tas2563_dvc_table into a separate Header file
by Shenghao Ding 30 Jul '24

30 Jul '24

Move tas2563_dvc_table into a separate Header file, as only tas2781 codec driver use this table, and hda side codec driver won't use it. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - Create tas2563-tlv.h into include/sound - Move tas2563_dvc_table from tas2781-tlv.h to tas2563-tlv.h - Add "#include <sound/tas2563-tlv.h>" into tas2781-i2c.c --- include/sound/tas2563-tlv.h | 279 +++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 260 ------------------------------ sound/soc/codecs/tas2781-i2c.c | 1 + 3 files changed, 280 insertions(+), 260 deletions(-) create mode 100644 include/sound/tas2563-tlv.h diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h new file mode 100644 index 000000000000..cf6eedb98ad2 --- /dev/null +++ b/include/sound/tas2563-tlv.h @@ -0,0 +1,279 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2563_TLV_H__ +#define __TAS2563_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); + +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 99c41bfc7827..5b003843752b 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -17,265 +17,5 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); -/* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { - { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ - { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ - { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ - { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ - { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ - { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ - { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ - { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ - { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ - { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ - { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ - { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ - { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ - { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ - { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ - { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ - { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ - { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ - { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ - { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ - { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ - { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ - { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ - { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ - { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ - { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ - { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ - { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ - { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ - { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ - { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ - { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ - { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ - { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ - { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ - { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ - { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ - { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ - { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ - { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ - { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ - { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ - { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ - { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ - { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ - { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ - { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ - { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ - { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ - { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ - { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ - { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ - { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ - { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ - { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ - { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ - { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ - { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ - { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ - { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ - { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ - { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ - { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ - { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ - { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ - { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ - { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ - { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ - { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ - { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ - { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ - { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ - { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ - { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ - { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ - { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ - { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ - { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ - { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ - { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ - { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ - { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ - { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ - { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ - { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ - { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ - { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ - { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ - { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ - { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ - { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ - { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ - { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ - { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ - { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ - { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ - { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ - { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ - { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ - { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ - { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ - { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ - { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ - { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ - { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ - { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ - { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ - { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ - { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ - { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ - { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ - { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ - { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ - { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ - { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ - { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ - { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ - { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ - { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ - { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ - { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ - { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ - { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ - { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ - { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ - { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ - { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ - { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ - { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ - { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ - { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ - { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ - { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ - { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ - { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ - { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ - { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ - { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ - { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ - { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ - { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ - { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ - { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ - { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ - { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ - { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ - { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ - { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ - { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ - { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ - { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ - { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ - { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ - { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ - { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ - { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ - { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ - { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ - { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ - { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ - { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ - { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ - { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ - { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ - { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ - { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ - { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ - { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ - { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ - { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ - { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ - { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ - { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ - { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ - { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ - { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ - { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ - { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ - { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ - { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ - { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ - { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ - { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ - { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ - { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ - { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ - { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ - { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ - { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ - { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ - { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ - { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ - { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ - { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ - { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ - { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ - { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ - { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ - { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ - { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ - { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ - { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ - { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ - { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ - { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ - { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ - { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ - { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ - { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ - { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ - { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ - { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ - { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ - { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ - { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ - { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ - { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ - { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ - { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ - { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ - { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ - { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ - { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ - { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ - { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ - { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ - { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ - { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ - { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ - { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ - { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ - { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ - { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ - { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ - { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ - { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ - { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ - { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ - { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ - { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ - { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ - { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ - { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ - { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ - { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ - { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ - { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ - { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ - { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ - { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ - { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ - { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ - { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ - { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ - { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ - { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ -}; #endif diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e79d613745b4..8a57c045afdd 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include <sound/soc.h> #include <sound/tas2781.h> #include <sound/tlv.h> +#include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> -- 2.34.1

3 4

[PATCH v1] ASoc: tas2781: Add TAS2563 into the Header
by Shenghao Ding 30 Jul '24

30 Jul '24

Add TAS2563 into the Header in case of misunderstanding and add channel No information for error debug in tasdevice_dev_read. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add TAS2563 into the Header - Add channel no into the log in tasdevice_dev_read for error debug --- sound/soc/codecs/tas2781-comlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1fbf4560f5cc..58abbc098a91 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// TAS2781 Common functions for HDA and ASoC Audio drivers +// TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // // Copyright 2023 - 2024 Texas Instruments, Inc. // @@ -64,8 +64,8 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, */ ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); + dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", + __func__, ret, chn); goto out; } } -- 2.34.1

2 1

[PATCH v3 0/3] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 30 Jul '24

30 Jul '24

This patchset serves to prevent an AB/BA deadlock: thread 0: * (lock A) acquire substream lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) wait for tasklet to finish by calling tasklet_unlock_spin_wait() in tasklet_disable_in_atomic() in ohci_flush_iso_completions() of ohci.c thread 1: * (lock B) enter tasklet * (lock A) attempt to acquire substream lock, waiting for it to be released: snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() in update_pcm_pointers() in process_ctx_payloads() in process_rx_packets() of amdtp-stream.c ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Cc: stable(a)vger.kernel.org Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (3): Revert "ALSA: firewire-lib: obsolete workqueue for period update" Revert "ALSA: firewire-lib: operate for period elapse event in process context" ALSA: firewire-lib: amdtp-stream work queue inline description sound/firewire/amdtp-stream.c | 38 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 25 insertions(+), 14 deletions(-) -- 2.45.2

2 4

[PATCH] ASoC: dt-bindings: qcom,apq8016-sbc-sndcard: move to separate binding
by Krzysztof Kozlowski 30 Jul '24

30 Jul '24

The APQ8016 SBC and MSM8916 QDSP6 sound cards are a bit different from others: they have additional IO muxing address space and pin control. Move them to separate schema, so the original qcom,sm8250.yaml will be easier to manage. New schema is going to grow for other platforms having more of IO muxing address spaces. Cc: Adam Skladowski <a39.skl(a)gmail.com> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../sound/qcom,apq8016-sbc-sndcard.yaml | 205 ++++++++++++++++++ .../bindings/sound/qcom,sm8250.yaml | 137 ------------ 2 files changed, 205 insertions(+), 137 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml new file mode 100644 index 000000000000..6ad451549036 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml @@ -0,0 +1,205 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,apq8016-sbc-sndcard.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm APQ8016 and similar sound cards + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + - Stephan Gerhold <stephan(a)gerhold.net> + +properties: + compatible: + enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + + reg: + items: + - description: Microphone I/O mux register address + - description: Speaker I/O mux register address + + reg-names: + items: + - const: mic-iomux + - const: spkr-iomux + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names could be power supplies, + MicBias of codec and the jacks on the board. + + aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User visible long sound card name + + pin-switches: + description: List of widget names for which pin switches should be created. + $ref: /schemas/types.yaml#/definitions/string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + platform: + description: Holds subnode which indicates platform dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 8 + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - model + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound@7702000 { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "DB410c"; + audio-routing = + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS External1"; + + pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; + pinctrl-names = "default", "sleep"; + + quaternary-dai-link { + link-name = "ADV7533"; + cpu { + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; + + primary-dai-link { + link-name = "WCD"; + cpu { + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + + tertiary-dai-link { + link-name = "WCD-Capture"; + cpu { + sound-dai = <&lpass MI2S_TERTIARY>; + }; + codec { + sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; + }; + }; + }; + + - | + #include <dt-bindings/sound/qcom,q6afe.h> + #include <dt-bindings/sound/qcom,q6asm.h> + sound@7702000 { + compatible = "qcom,msm8916-qdsp6-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "msm8916"; + widgets = + "Speaker", "Speaker", + "Headphone", "Headphones"; + pin-switches = "Speaker"; + audio-routing = + "Speaker", "Speaker Amp OUT", + "Speaker Amp IN", "HPH_R", + "Headphones", "HPH_L", + "Headphones", "HPH_R", + "AMIC1", "MIC BIAS Internal1", + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS Internal3"; + aux-devs = <&speaker_amp>; + + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&cdc_pdm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + primary-dai-link { + link-name = "Primary MI2S"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + platform { + sound-dai = <&q6routing>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index c9076dcd44c1..1d3acdc0c733 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -27,9 +27,7 @@ properties: - qcom,sm8650-sndcard - const: qcom,sm8450-sndcard - enum: - - qcom,apq8016-sbc-sndcard - qcom,apq8096-sndcard - - qcom,msm8916-qdsp6-sndcard - qcom,qcm6490-idp-sndcard - qcom,qcs6490-rb3gen2-sndcard - qcom,qrb5165-rb5-sndcard @@ -58,18 +56,6 @@ properties: $ref: /schemas/types.yaml#/definitions/string description: User visible long sound card name - pin-switches: - description: List of widget names for which pin switches should be created. - $ref: /schemas/types.yaml#/definitions/string-array - - widgets: - description: User specified audio sound widgets. - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - - # Only valid for some compatibles (see allOf if below) - reg: true - reg-names: true - patternProperties: ".*-dai-link$": description: @@ -122,34 +108,6 @@ required: - compatible - model -allOf: - - if: - properties: - compatible: - contains: - enum: - - qcom,apq8016-sbc-sndcard - - qcom,msm8916-qdsp6-sndcard - then: - properties: - reg: - items: - - description: Microphone I/O mux register address - - description: Speaker I/O mux register address - reg-names: - items: - - const: mic-iomux - - const: spkr-iomux - required: - - compatible - - model - - reg - - reg-names - else: - properties: - reg: false - reg-names: false - additionalProperties: false examples: @@ -231,98 +189,3 @@ examples: }; }; }; - - - | - #include <dt-bindings/sound/qcom,lpass.h> - sound@7702000 { - compatible = "qcom,apq8016-sbc-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "DB410c"; - audio-routing = - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS External1"; - - pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; - pinctrl-names = "default", "sleep"; - - quaternary-dai-link { - link-name = "ADV7533"; - cpu { - sound-dai = <&lpass MI2S_QUATERNARY>; - }; - codec { - sound-dai = <&adv_bridge 0>; - }; - }; - - primary-dai-link { - link-name = "WCD"; - cpu { - sound-dai = <&lpass MI2S_PRIMARY>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - - tertiary-dai-link { - link-name = "WCD-Capture"; - cpu { - sound-dai = <&lpass MI2S_TERTIARY>; - }; - codec { - sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; - }; - }; - }; - - - | - #include <dt-bindings/sound/qcom,q6afe.h> - #include <dt-bindings/sound/qcom,q6asm.h> - sound@7702000 { - compatible = "qcom,msm8916-qdsp6-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "msm8916"; - widgets = - "Speaker", "Speaker", - "Headphone", "Headphones"; - pin-switches = "Speaker"; - audio-routing = - "Speaker", "Speaker Amp OUT", - "Speaker Amp IN", "HPH_R", - "Headphones", "HPH_L", - "Headphones", "HPH_R", - "AMIC1", "MIC BIAS Internal1", - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS Internal3"; - aux-devs = <&speaker_amp>; - - pinctrl-names = "default", "sleep"; - pinctrl-0 = <&cdc_pdm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus>; - - mm1-dai-link { - link-name = "MultiMedia1"; - cpu { - sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; - }; - }; - - primary-dai-link { - link-name = "Primary MI2S"; - cpu { - sound-dai = <&q6afedai PRIMARY_MI2S_RX>; - }; - platform { - sound-dai = <&q6routing>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - }; -- 2.43.0

6 6

[PATCH 0/6] ASoC: codecs: wcd93xx/wsa88xx: Correct Soundwire ports mask
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Incorrect mask of Soundwire ports - one bit too long/big - was passed. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Not adding Cc-stable because I don't think the issue can lead to real out-of-bounds read. Best regards, Krzysztof --- Krzysztof Kozlowski (6): ASoC: codecs: wcd937x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd938x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd939x-sdw: Correct Soundwire ports mask ASoC: codecs: wsa881x: Correct Soundwire ports mask ASoC: codecs: wsa883x: Correct Soundwire ports mask ASoC: codecs: wsa884x: Correct Soundwire ports mask sound/soc/codecs/wcd937x-sdw.c | 4 ++-- sound/soc/codecs/wcd938x-sdw.c | 4 ++-- sound/soc/codecs/wcd939x-sdw.c | 4 ++-- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- 6 files changed, 9 insertions(+), 9 deletions(-) --- base-commit: 864b1099d16fc7e332c3ad7823058c65f890486c change-id: 20240726-asoc-wcd-wsa-swr-ports-genmask-868239f9682f Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 8

[PATCH] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 29 Jul '24

29 Jul '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. As the ALSA controls for the firmware coefficients are not used in normal operation they can all be hidden, but add a kernel parameter so that they can be re-enabled for debugging. Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 96d3f13c5abf..1494383b22c9 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -23,6 +23,10 @@ #include "hda_cs_dsp_ctl.h" #include "hda_generic.h" +static bool expose_dsp_controls; +module_param(expose_dsp_controls, bool, 0444); +MODULE_PARM_DESC(expose_dsp_controls, "Expose firmware controls as ALSA controls 0=no (default), 1=yes"); + /* * The cs35l56_hda_dai_config[] reg sequence configures the device as * ASP1_BCLK_FREQ = 3.072 MHz @@ -758,6 +762,9 @@ static int cs35l56_hda_bind(struct device *dev, struct device *master, void *mas cs35l56_hda_create_controls(cs35l56); + if (expose_dsp_controls) + cs35l56_hda_add_dsp_controls(cs35l56); + #if IS_ENABLED(CONFIG_SND_DEBUG) cs35l56->debugfs_root = debugfs_create_dir(dev_name(cs35l56->base.dev), sound_debugfs_root); cs_dsp_init_debugfs(&cs35l56->cs_dsp, cs35l56->debugfs_root); -- 2.43.0

2 2

ASUS RoG Strix Z790-F: SPDIF (PCM) has wrong channels | how to get passthrough working?
by GitHub issues - opened 29 Jul '24

29 Jul '24

alsa-project/alsa-ucm-conf issue #434 was opened from Nightwing0815: Hi there, finally, [after a long journey,](https://github.com/alsa-project/alsa-ucm-conf/issues/350) yesterday I did an update from linuxmint to newest version 22, based on ubuntu 24.04 (I'll attach alsa-info and a system info). Now it's no big problem any more, to get SPDIF working: I just selected the right device and muted all others in alsamixer, except the pcm signal. But hey, there's a new problem: With a speaker-test -c6, I'm not able to hear my rear channels and the lfe channel, where they should be. Rear coming on front speakers... So, what do I have to do? Further, I'm asking for help to configure a pcm signal passthru for dts and (e)ac3 soundfiles. My teufel receiver recognices only a 5.1 stereo source, not an DD5.1 or DTS signal... [alsa-info.txt](https://github.com/user-attachments/files/16417700/alsa-info… [sysinfo.txt](https://github.com/user-attachments/files/16417724/sysinfo.txt) Thanks in advance for getting this problems solved, ToM Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/434 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/7] ASoC: codecs: wsa88xx and wcd93xx: Soundwire port non-functional cleanup
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups to make the code more robust or readable. No functional impact (compiled objects stay the same). Best regards, Krzysztof --- Krzysztof Kozlowski (7): ASoC: codecs: wsa881x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa883x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa884x: Use designator array initializers for Soundwire ports ASoC: codecs: wcd938x: Drop unused defines and enums ASoC: codecs: wcd937x: Move max port number defines to enum ASoC: codecs: wcd938x: Move max port number defines to enum ASoC: codecs: wcd939x: Move max port number defines to enum sound/soc/codecs/wcd937x.h | 34 +++++++++++++++++----------------- sound/soc/codecs/wcd938x.c | 18 ------------------ sound/soc/codecs/wcd938x.h | 4 ++-- sound/soc/codecs/wcd939x.h | 6 +++--- sound/soc/codecs/wsa881x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa883x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa884x.c | 34 ++++++++++++++++++++++------------ 7 files changed, 88 insertions(+), 92 deletions(-) --- base-commit: 66df68b9536e7465aec703733ea91ca39b787ed8 change-id: 20240725-asoc-wsa88xx-port-arrays-c58c96cd888e Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 15

[PATCH v2] ASoC: sti: add missing probe entry for player and reader
by Jerome Audu 29 Jul '24

29 Jul '24

This patch addresses a regression in the ASoC STI drivers that was introduced in Linux version 6.6.y. The issue originated from a series of patches (see https://lore.kernel.org/all/87wmy5b0wt.wl-kuninori.morimoto.gx@renesas.com/) that unintentionally omitted necessary probe functions for the player and reader components. Probe function in `sound/soc/sti/sti_uniperif.c:415` is being replaced by another probe function located at `sound/soc/sti/sti_uniperif.c:453`, which should instead be derived from the player and reader components. This patch correctly reinserts the missing probe entries, restoring the intended functionality. Fixes: 9f625f5e6cf9 ("ASoC: sti: merge DAI call back functions into ops") Signed-off-by: Jerome Audu <jau(a)free.fr> --- Changes in v2: - Update changelog - Link to v1: https://lore.kernel.org/r/20240721-sti-audio-fix-v1-1-a8b1ecf61cb4@free.fr --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 4 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index ba824f14a39c..a7956e5a4ee5 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -352,7 +352,7 @@ static int sti_uniperiph_resume(struct snd_soc_component *component) return ret; } -static int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct sti_uniperiph_dai *dai_data = &priv->dai_data; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..74e51f0ff85c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1380,6 +1380,7 @@ int uni_reader_init(struct platform_device *pdev, struct uniperif *reader); /* common */ +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai); int sti_uniperiph_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index dd9013c47664..6d1ce030963c 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -1038,6 +1038,7 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .startup = uni_player_startup, .shutdown = uni_player_shutdown, .prepare = uni_player_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 065c5f0d1f5f..05ea2b794eb9 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -401,6 +401,7 @@ static const struct snd_soc_dai_ops uni_reader_dai_ops = { .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, --- base-commit: 0c3836482481200ead7b416ca80c68a29cfdaabd change-id: 20240721-sti-audio-fix-1374fffb24e7 Best regards, -- Jerome Audu <jau(a)free.fr>

2 1

[RESEND] ASoC: codecs: ES8326: suspend issue
by Zhang Yi 29 Jul '24

29 Jul '24

We find that we need to disable micbias for the codec to enter suspend So We modify the trigger conditions for enable_micbias and disable_micbias Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b246694ebb4f..60877116c0ef 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -805,6 +805,7 @@ static void es8326_jack_button_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); button_to_report = 0; } + es8326_disable_micbias(es8326->component); } mutex_unlock(&es8326->lock); } @@ -878,7 +879,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00); - es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10); @@ -897,6 +897,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "button pressed\n"); regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + es8326_enable_micbias(es8326->component); queue_delayed_work(system_wq, &es8326->button_press_work, 10); goto exit; } @@ -1067,6 +1068,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC_MUTE, 0x0f); regmap_write(es8326->regmap, ES8326_CLK_DIV_LRCK, 0xff); + es8326_disable_micbias(es8326->component); msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

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[PATCH] ASoC: codecs: lpass-wsa-macro: Do not hard-code dai in VI mixer
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

The wsa_macro_vi_feed_mixer_put() callback for setting VI feedback mixer value could be used for different DAIs (planned in the future CPS DAI), so make the code a bit more generic by using DAI ID from widget->shift, instead of hard-coding it. The get() callback already follows such convention. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-wsa-macro.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 8351f0974e6a..4158657914fb 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2296,36 +2296,37 @@ static int wsa_macro_vi_feed_mixer_put(struct snd_kcontrol *kcontrol, struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); u32 enable = ucontrol->value.integer.value[0]; u32 spk_tx_id = mixer->shift; + u32 dai_id = widget->shift; if (enable) { if (spk_tx_id == WSA_MACRO_TX0 && !test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } if (spk_tx_id == WSA_MACRO_TX1 && !test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } } else { if (spk_tx_id == WSA_MACRO_TX0 && test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } if (spk_tx_id == WSA_MACRO_TX1 && test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } } snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, NULL); -- 2.43.0

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[PATCH] ASoC: cs42l42: Convert comma to semicolon
by Chen Ni 29 Jul '24

29 Jul '24

Replace a comma between expression statements by a semicolon. Fixes: 90f6a2a20bd2 ("ASoC: cs42l42: Add SoundWire support") Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/codecs/cs42l42-sdw.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 94a66a325303..29891c1f6bec 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -323,15 +323,15 @@ static int cs42l42_sdw_read_prop(struct sdw_slave *peripheral) prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; /* DP1 - capture */ - ports[0].num = CS42L42_SDW_CAPTURE_PORT, - ports[0].type = SDW_DPN_FULL, - ports[0].ch_prep_timeout = 10, + ports[0].num = CS42L42_SDW_CAPTURE_PORT; + ports[0].type = SDW_DPN_FULL; + ports[0].ch_prep_timeout = 10; prop->src_dpn_prop = &ports[0]; /* DP2 - playback */ - ports[1].num = CS42L42_SDW_PLAYBACK_PORT, - ports[1].type = SDW_DPN_FULL, - ports[1].ch_prep_timeout = 10, + ports[1].num = CS42L42_SDW_PLAYBACK_PORT; + ports[1].type = SDW_DPN_FULL; + ports[1].ch_prep_timeout = 10; prop->sink_dpn_prop = &ports[1]; return 0; -- 2.25.1

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[PATCH 0/4] ASoC: codecs: wsa88xx: Few cleanups
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups around wsa88xx codecs. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: codecs: wsa881x: Drop unused version readout ASoC: codecs: wsa883x: Handle reading version failure ASoC: codecs: wsa883x: Simplify handling variant/version ASoC: codecs: wsa884x: Simplify handling variant sound/soc/codecs/wsa881x.c | 2 -- sound/soc/codecs/wsa883x.c | 33 ++++++++++++++++++--------------- sound/soc/codecs/wsa884x.c | 5 ++--- 3 files changed, 20 insertions(+), 20 deletions(-) --- base-commit: 0b58e108042b0ed28a71cd7edf5175999955b233 change-id: 20240710-asoc-wsa88xx-version-dd0039265245 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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Re: [PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by edmund.raile 29 Jul '24

29 Jul '24

> Thank you for your sending the revised patches, it looks better than the > previous one. However, I have an additional request. Allright, patch v3 it is. > [1] https://git-scm.com/docs/git-revert Should have known git has something like that, how handy! > $ git revert -s b5b519965c4c Yes, 5b5 can be removed via revert, but what is the difference in effect? Just time saving? > $ git revert -s 7ba5ca32fe6e This one I'd like to ask you about: The original inline comment in amdtp-stream.c amdtp_domain_stream_pcm_pointer() ``` // This function is called in software IRQ context of // period_work or process context. // // When the software IRQ context was scheduled by software IRQ // context of IT contexts, queued packets were already handled. // Therefore, no need to flush the queue in buffer furthermore. // // When the process context reach here, some packets will be // already queued in the buffer. These packets should be handled // immediately to keep better granularity of PCM pointer. // // Later, the process context will sometimes schedules software // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above ``` (let's call the above v1) was replaced with ``` // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. ``` on occasion of 7ba5ca32fe6e (let's call this v2). I sought to replace it with ``` // use wq to prevent deadlock between process context spin_lock // of snd_pcm_stream_lock_irq() in snd_pcm_status64() and // softIRQ context spin_lock of snd_pcm_stream_lock_irqsave() // in snd_pcm_period_elapsed() ``` to prevent this issue from occurring again (let's call this v3). Should I include v1, v3 or a combination of v1 and v3 in my next patch? > Just for safe, it is preferable to execute 'scripts/checkpatch.pl' in > kernel tree to check the patchset generated by send-email subcommand[3]. Absolutely should have done so, sorry. Thank you for your patience and guidance, Edmund Raile.

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[PATCH AUTOSEL 5.15 09/13] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 089b6c7994f9a..cfa0c3eaffea7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -258,6 +258,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

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