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[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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