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[PATCH] ALSA: control: Fix an out-of-bounds bug in get_ctl_id_hash()
by Zheyu Ma 24 Aug '22

24 Aug '22

Since the user can control the arguments provided to the kernel by the ioctl() system call, an out-of-bounds bug occurs when the 'id->name' provided by the user does not end with '\0'. The following log can reveal it: [ 10.002313] BUG: KASAN: stack-out-of-bounds in snd_ctl_find_id+0x36c/0x3a0 [ 10.002895] Read of size 1 at addr ffff888109f5fe28 by task snd/439 [ 10.004934] Call Trace: [ 10.007140] snd_ctl_find_id+0x36c/0x3a0 [ 10.007489] snd_ctl_ioctl+0x6cf/0x10e0 Fix this by checking the bound of 'id->name' in the loop. Fixes: c27e1efb61c5 ("ALSA: control: Use xarray for faster lookups") Signed-off-by: Zheyu Ma <zheyuma97(a)gmail.com> --- sound/core/control.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index f3e893715369..e8fc4c511e5f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -385,14 +385,14 @@ static bool elem_id_matches(const struct snd_kcontrol *kctl, #define MULTIPLIER 37 static unsigned long get_ctl_id_hash(const struct snd_ctl_elem_id *id) { + int i; unsigned long h; - const unsigned char *p; h = id->iface; h = MULTIPLIER * h + id->device; h = MULTIPLIER * h + id->subdevice; - for (p = id->name; *p; p++) - h = MULTIPLIER * h + *p; + for (i = 0; id->name[i] && i < SNDRV_CTL_ELEM_ID_NAME_MAXLEN; i++) + h = MULTIPLIER * h + id->name[i]; h = MULTIPLIER * h + id->index; h &= LONG_MAX; return h; -- 2.25.1

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arcam_av.c: Include missing string.h
by GitHub pull_request - opened 24 Aug '22

24 Aug '22

alsa-project/alsa-plugins pull request #47 was opened from kraj: bzero() function needs this header to be included Signed-off-by: Khem Raj <raj.khem(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-plugins/pull/47 Patch URL : https://github.com/alsa-project/alsa-plugins/pull/47.patch Repository URL: https://github.com/alsa-project/alsa-plugins

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[PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 23 Aug '22

23 Aug '22

Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10) for example if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,16 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending of HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

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[PATCH 0/3] ASoC: SOF: Intel: hda: Correct Firmware State Register use
by Peter Ujfalusi 23 Aug '22

23 Aug '22

Hi, The FSR (Firmware State Register) holds the ROM state information, it does not contain error information. The FSR itself is a bit more complicated as well as the state depends on the module currently in use. The error code from ROM or the status code from the firmware is located at the next register. Fix the handling of the FSR in order to provide usable and human readable (in most cases) report on the status and error. Regards, Peter --- Peter Ujfalusi (3): ASoC: SOF: Intel: hda: Correct the ROM/FW state reporting code ASoC: SOF: Intel: hda-loader: Use the FSR state definitions during bootup ASoC: SOF: Intel: hda: Drop no longer used ROM state definitions sound/soc/sof/intel/hda-loader.c | 10 +-- sound/soc/sof/intel/hda.c | 147 ++++++++++++++++++++++++++----- sound/soc/sof/intel/hda.h | 69 +++++++++++++-- 3 files changed, 194 insertions(+), 32 deletions(-) -- 2.37.0

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[PATCH] ASoC: mediatek: mt8186: rename sound card name
by Chunxu Li 23 Aug '22

23 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> The field 'topology_shortname' in 'snd_soc_card' is defined as char[32], Current card name will be truncated when SOF is enabled, so rename the sound card name. Signed-off-by: chunxu.li <chunxu.li(a)mediatek.com> --- sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c index 84ee5d95a9f0..17a15bec41da 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c @@ -962,7 +962,7 @@ mt8186_mt6366_da7219_max98357_controls[] = { }; static struct snd_soc_card mt8186_mt6366_da7219_max98357_soc_card = { - .name = "mt8186_mt6366_da7219_max98357", + .name = "mt8186_da7219_max98357", .owner = THIS_MODULE, .dai_link = mt8186_mt6366_da7219_max98357_dai_links, .num_links = ARRAY_SIZE(mt8186_mt6366_da7219_max98357_dai_links), diff --git a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c index 6c41706a5621..393d179d61de 100644 --- a/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c +++ b/sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c @@ -938,7 +938,7 @@ mt8186_mt6366_rt1019_rt5682s_controls[] = { }; static struct snd_soc_card mt8186_mt6366_rt1019_rt5682s_soc_card = { - .name = "mt8186_mt6366_rt1019_rt5682s", + .name = "mt8186_rt1019_rt5682s", .owner = THIS_MODULE, .dai_link = mt8186_mt6366_rt1019_rt5682s_dai_links, .num_links = ARRAY_SIZE(mt8186_mt6366_rt1019_rt5682s_dai_links), -- 2.25.1

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[PATCH linux-next] ASoC: atmel_ssc_dai: Remove the unneeded result variable
by cgel.zte＠gmail.com 23 Aug '22

23 Aug '22

From: ye xingchen <ye.xingchen(a)zte.com.cn> Return the value from asoc_ssc_init() directly instead of storing it in another redundant variable. Reported-by: Zeal Robot <zealci(a)zte.com.cn> Signed-off-by: ye xingchen <ye.xingchen(a)zte.com.cn> --- sound/soc/atmel/atmel_ssc_dai.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e868b7e028d6..3763454436c1 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -891,7 +891,6 @@ static int asoc_ssc_init(struct device *dev) int atmel_ssc_set_audio(int ssc_id) { struct ssc_device *ssc; - int ret; /* If we can grab the SSC briefly to parent the DAI device off it */ ssc = ssc_request(ssc_id); @@ -903,9 +902,7 @@ int atmel_ssc_set_audio(int ssc_id) ssc_info[ssc_id].ssc = ssc; } - ret = asoc_ssc_init(&ssc->pdev->dev); - - return ret; + return asoc_ssc_init(&ssc->pdev->dev); } EXPORT_SYMBOL_GPL(atmel_ssc_set_audio); -- 2.25.1

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[RESEND PATCH] ASoC: fsl_sai: Add support multi fifo sdma script
by Shengjiu Wang 23 Aug '22

23 Aug '22

With disabling combine mode, the multiple successive FIFO registers or non successive FIFO registers of SAI module can work with the sdma multi fifo script. This patch is to configure the necessary information to the SDMA engine driver for support multi fifo script. 'words_per_fifo' is the channels for each dataline 'n_fifos_src' and 'n_fifos_dst' are the fifo number 'stride_fifos_src' and 'stride_fifos_dst' are the stride between enable FIFOs 'maxburst' is the multiply of datalines Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in resend: - rebase to latest for-6.1 sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_sai.h | 3 +++ 2 files changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a7fa6f0bf83d..0f92906d7a29 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -527,6 +527,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slot_width = word_width; int adir = tx ? RX : TX; u32 pins, bclk; + u32 watermark; int ret, i; if (sai->slots) @@ -619,7 +620,15 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } - if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1) + /* + * Combine mode has limation: + * - Can't used for singel dataline/FIFO case except the FIFO0 + * - Can't used for multi dataline/FIFO case except the enabled FIFOs + * are successive and start from FIFO0 + * + * So for common usage, all multi fifo case disable the combine mode. + */ + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1 || sai->is_multi_fifo_dma) regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_FCOMB_MASK, 0); else @@ -630,6 +639,26 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, dma_params->addr = sai->res->start + FSL_SAI_xDR0(tx) + dl_cfg[dl_cfg_idx].start_off[tx] * 0x4; + if (sai->is_multi_fifo_dma) { + sai->audio_config[tx].words_per_fifo = min(slots, channels); + if (tx) { + sai->audio_config[tx].n_fifos_dst = pins; + sai->audio_config[tx].stride_fifos_dst = dl_cfg[dl_cfg_idx].next_off[tx]; + } else { + sai->audio_config[tx].n_fifos_src = pins; + sai->audio_config[tx].stride_fifos_src = dl_cfg[dl_cfg_idx].next_off[tx]; + } + dma_params->maxburst = sai->audio_config[tx].words_per_fifo * pins; + dma_params->peripheral_config = &sai->audio_config[tx]; + dma_params->peripheral_size = sizeof(sai->audio_config[tx]); + + watermark = tx ? (sai->soc_data->fifo_depth - dma_params->maxburst) : + (dma_params->maxburst - 1); + regmap_update_bits(sai->regmap, FSL_SAI_xCR1(tx, ofs), + FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), + watermark); + } + /* Find a proper tcre setting */ for (i = 0; i < sai->soc_data->pins; i++) { trce_mask = (1 << (i + 1)) - 1; @@ -1257,6 +1286,7 @@ static int fsl_sai_probe(struct platform_device *pdev) char tmp[8]; int irq, ret, i; int index; + u32 dmas[4]; sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -1313,6 +1343,11 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_asoc_get_pll_clocks(&pdev->dev, &sai->pll8k_clk, &sai->pll11k_clk); + /* Use Multi FIFO mode depending on the support from SDMA script */ + ret = of_property_read_u32_array(np, "dmas", dmas, 4); + if (!sai->soc_data->use_edma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI) + sai->is_multi_fifo_dma = true; + /* read dataline mask for rx and tx*/ ret = fsl_sai_read_dlcfg(sai); if (ret < 0) { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 17956b5731dc..697f6690068c 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -6,6 +6,7 @@ #ifndef __FSL_SAI_H #define __FSL_SAI_H +#include <linux/dma/imx-dma.h> #include <sound/dmaengine_pcm.h> #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ @@ -281,6 +282,7 @@ struct fsl_sai { bool is_lsb_first; bool is_dsp_mode; bool is_pdm_mode; + bool is_multi_fifo_dma; bool synchronous[2]; struct fsl_sai_dl_cfg *dl_cfg; unsigned int dl_cfg_cnt; @@ -300,6 +302,7 @@ struct fsl_sai { struct pm_qos_request pm_qos_req; struct pinctrl *pinctrl; struct pinctrl_state *pins_state; + struct sdma_peripheral_config audio_config[2]; }; #define TX 1 -- 2.34.1

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[PATCH] ASoC: SOF: compress: Remove dai_posn variable
by Daniel Baluta 23 Aug '22

23 Aug '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> dai_posn is set but never used. Initial intention was to use dai_posn to shorthen one code line but it looks fine without it too. Fixes: c1a731c71359 ("ASoC: SOF: compress: Add support for computing timestamps") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/compress.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index 174b3d8e67dd..c8ae778a50d1 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -327,7 +327,6 @@ static int sof_compr_pointer(struct snd_soc_component *component, struct snd_compr_stream *cstream, struct snd_compr_tstamp *tstamp) { - u64 dai_posn; struct snd_sof_pcm *spcm; struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct sof_compr_stream *sstream = cstream->runtime->private_data; @@ -336,8 +335,6 @@ static int sof_compr_pointer(struct snd_soc_component *component, if (!spcm) return -EINVAL; - dai_posn = spcm->stream[cstream->direction].posn.dai_posn; - tstamp->sampling_rate = sstream->sampling_rate; tstamp->copied_total = sstream->copied_total; tstamp->pcm_io_frames = div_u64(spcm->stream[cstream->direction].posn.dai_posn, -- 2.27.0

2 1

[PATCH] ASoC: fsl_sai: Add support multi fifo sdma script
by Shengjiu Wang 23 Aug '22

23 Aug '22

With disabling combine mode, the multiple successive FIFO registers or non successive FIFO registers of SAI module can work with the sdma multi fifo script. This patch is to configure the necessary information to the SDMA engine driver for support multi fifo script. 'words_per_fifo' is the channels for each dataline 'n_fifos_src' and 'n_fifos_dst' are the fifo number 'stride_fifos_src' and 'stride_fifos_dst' are the stride between enable FIFOs 'maxburst' is the multiply of datalines Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_sai.h | 3 +++ 2 files changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index acf76ea21138..a2b375e2b56c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -533,6 +533,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, u32 slot_width = word_width; int adir = tx ? RX : TX; u32 pins, bclk; + u32 watermark; int ret, i; if (sai->slot_width) @@ -627,7 +628,15 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR5_FBT_MASK, val_cr5); } - if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1) + /* + * Combine mode has limation: + * - Can't used for singel dataline/FIFO case except the FIFO0 + * - Can't used for multi dataline/FIFO case except the enabled FIFOs + * are successive and start from FIFO0 + * + * So for common usage, all multi fifo case disable the combine mode. + */ + if (hweight8(dl_cfg[dl_cfg_idx].mask[tx]) <= 1 || sai->is_multi_fifo_dma) regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_FCOMB_MASK, 0); else @@ -638,6 +647,26 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, dma_params->addr = sai->res->start + FSL_SAI_xDR0(tx) + dl_cfg[dl_cfg_idx].start_off[tx] * 0x4; + if (sai->is_multi_fifo_dma) { + sai->audio_config[tx].words_per_fifo = min(slots, channels); + if (tx) { + sai->audio_config[tx].n_fifos_dst = pins; + sai->audio_config[tx].stride_fifos_dst = dl_cfg[dl_cfg_idx].next_off[tx]; + } else { + sai->audio_config[tx].n_fifos_src = pins; + sai->audio_config[tx].stride_fifos_src = dl_cfg[dl_cfg_idx].next_off[tx]; + } + dma_params->maxburst = sai->audio_config[tx].words_per_fifo * pins; + dma_params->peripheral_config = &sai->audio_config[tx]; + dma_params->peripheral_size = sizeof(sai->audio_config[tx]); + + watermark = tx ? (sai->soc_data->fifo_depth - dma_params->maxburst) : + (dma_params->maxburst - 1); + regmap_update_bits(sai->regmap, FSL_SAI_xCR1(tx, ofs), + FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), + watermark); + } + /* Find a proper tcre setting */ for (i = 0; i < sai->soc_data->pins; i++) { trce_mask = (1 << (i + 1)) - 1; @@ -1265,6 +1294,7 @@ static int fsl_sai_probe(struct platform_device *pdev) char tmp[8]; int irq, ret, i; int index; + u32 dmas[4]; sai = devm_kzalloc(dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -1321,6 +1351,11 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_asoc_get_pll_clocks(&pdev->dev, &sai->pll8k_clk, &sai->pll11k_clk); + /* Use Multi FIFO mode depending on the support from SDMA script */ + ret = of_property_read_u32_array(np, "dmas", dmas, 4); + if (!sai->soc_data->use_edma && !ret && dmas[2] == IMX_DMATYPE_MULTI_SAI) + sai->is_multi_fifo_dma = true; + /* read dataline mask for rx and tx*/ ret = fsl_sai_read_dlcfg(sai); if (ret < 0) { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 17956b5731dc..697f6690068c 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -6,6 +6,7 @@ #ifndef __FSL_SAI_H #define __FSL_SAI_H +#include <linux/dma/imx-dma.h> #include <sound/dmaengine_pcm.h> #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ @@ -281,6 +282,7 @@ struct fsl_sai { bool is_lsb_first; bool is_dsp_mode; bool is_pdm_mode; + bool is_multi_fifo_dma; bool synchronous[2]; struct fsl_sai_dl_cfg *dl_cfg; unsigned int dl_cfg_cnt; @@ -300,6 +302,7 @@ struct fsl_sai { struct pm_qos_request pm_qos_req; struct pinctrl *pinctrl; struct pinctrl_state *pins_state; + struct sdma_peripheral_config audio_config[2]; }; #define TX 1 -- 2.17.1

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[PATCH] ASoC: mchp-spdiftx: Fix clang -Wbitfield-constant-conversion
by Nathan Chancellor 23 Aug '22

23 Aug '22

A recent change in clang strengthened its -Wbitfield-constant-conversion to warn when 1 is assigned to a 1-bit signed integer bitfield, as it can only be 0 or -1, not 1: sound/soc/atmel/mchp-spdiftx.c:505:20: error: implicit truncation from 'int' to bit-field changes value from 1 to -1 [-Werror,-Wbitfield-constant-conversion] dev->gclk_enabled = 1; ^ ~ 1 error generated. The actual value of the field is never checked, just that it is not zero, so there is not a real bug here. However, it is simple enough to silence the warning by making the bitfield unsigned, which matches the mchp-spdifrx driver. Fixes: 06ca24e98e6b ("ASoC: mchp-spdiftx: add driver for S/PDIF TX Controller") Link: https://github.com/ClangBuiltLinux/linux/issues/1686 Link: https://github.com/llvm/llvm-project/commit/82afc9b169a67e8b8a1862fb9c41a2c… Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/atmel/mchp-spdiftx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index 4850a177803d..ab2d7a791f39 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -196,7 +196,7 @@ struct mchp_spdiftx_dev { struct clk *pclk; struct clk *gclk; unsigned int fmt; - int gclk_enabled:1; + unsigned int gclk_enabled:1; }; static inline int mchp_spdiftx_is_running(struct mchp_spdiftx_dev *dev) base-commit: 15205c2829ca2cbb5ece5ceaafe1171a8470e62b -- 2.37.1

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