July 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 5.18 39/54] ASoC: wm5102: Fix event generation for output compensation
by Sasha Levin 20 Jul '22

20 Jul '22

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 71b5ab96ffe6589abe7a2e302b83f7a426ebe099 ] The output compensation controls always returns zero regardless of if the control value was updated. This results in missing notifications to user-space of the control change. Update the handling to return 1 when the value is changed. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220628153409.3266932-1-ckeepax@opensource.cirru… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm5102.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index da2f8998df87..b034df47a5ef 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -680,12 +680,17 @@ static int wm5102_out_comp_coeff_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + uint16_t dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + int ret = 0; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_coeff = get_unaligned_be16(ucontrol->value.bytes.data); + if (arizona->dac_comp_coeff != dac_comp_coeff) { + arizona->dac_comp_coeff = dac_comp_coeff; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static int wm5102_out_comp_switch_get(struct snd_kcontrol *kcontrol, @@ -706,12 +711,20 @@ static int wm5102_out_comp_switch_put(struct snd_kcontrol *kcontrol, { struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); struct arizona *arizona = dev_get_drvdata(component->dev->parent); + struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; + int ret = 0; + + if (ucontrol->value.integer.value[0] > mc->max) + return -EINVAL; mutex_lock(&arizona->dac_comp_lock); - arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + if (arizona->dac_comp_enabled != ucontrol->value.integer.value[0]) { + arizona->dac_comp_enabled = ucontrol->value.integer.value[0]; + ret = 1; + } mutex_unlock(&arizona->dac_comp_lock); - return 0; + return ret; } static const char * const wm5102_osr_text[] = { -- 2.35.1

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[PATCH AUTOSEL 5.18 38/54] ASoC: wcd9335: Use int array instead of bitmask for TX mixers
by Sasha Levin 20 Jul '22

20 Jul '22

From: Yassine Oudjana <y.oudjana(a)protonmail.com> [ Upstream commit a5d6d28e2ea38dff017cb562dfbe0259d093a851 ] Currently slim_tx_mixer_get reports all TX mixers as enabled when at least one is, due to it reading the entire tx_port_value bitmask without testing the specific bit corresponding to a TX port. Furthermore, using the same bitmask for all capture DAIs makes setting one mixer affect them all. To prevent this, and since the SLIM TX muxes effectively only connect to one of the mixers at a time, turn tx_port_value into an int array storing the DAI index each of the ports is connected to. Signed-off-by: Yassine Oudjana <y.oudjana(a)protonmail.com> Link: https://lore.kernel.org/r/20220622061745.35399-1-y.oudjana@protonmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd9335.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 1e60db4056ad..ad887c9c7573 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -342,7 +342,7 @@ struct wcd9335_codec { struct regulator_bulk_data supplies[WCD9335_MAX_SUPPLY]; unsigned int rx_port_value[WCD9335_RX_MAX]; - unsigned int tx_port_value; + unsigned int tx_port_value[WCD9335_TX_MAX]; int hph_l_gain; int hph_r_gain; u32 rx_bias_count; @@ -1328,8 +1328,13 @@ static int slim_tx_mixer_get(struct snd_kcontrol *kc, struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kc); struct wcd9335_codec *wcd = dev_get_drvdata(dapm->dev); + struct snd_soc_dapm_widget *widget = snd_soc_dapm_kcontrol_widget(kc); + struct soc_mixer_control *mixer = + (struct soc_mixer_control *)kc->private_value; + int dai_id = widget->shift; + int port_id = mixer->shift; - ucontrol->value.integer.value[0] = wcd->tx_port_value; + ucontrol->value.integer.value[0] = wcd->tx_port_value[port_id] == dai_id; return 0; } @@ -1352,12 +1357,12 @@ static int slim_tx_mixer_put(struct snd_kcontrol *kc, case AIF2_CAP: case AIF3_CAP: /* only add to the list if value not set */ - if (enable && !(wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value |= BIT(port_id); + if (enable && wcd->tx_port_value[port_id] != dai_id) { + wcd->tx_port_value[port_id] = dai_id; list_add_tail(&wcd->tx_chs[port_id].list, &wcd->dai[dai_id].slim_ch_list); - } else if (!enable && (wcd->tx_port_value & BIT(port_id))) { - wcd->tx_port_value &= ~BIT(port_id); + } else if (!enable && wcd->tx_port_value[port_id] == dai_id) { + wcd->tx_port_value[port_id] = -1; list_del_init(&wcd->tx_chs[port_id].list); } break; -- 2.35.1

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[PATCH AUTOSEL 5.18 37/54] ASoC: tlv320adcx140: Fix tx_mask check
by Sasha Levin 20 Jul '22

20 Jul '22

From: Sascha Hauer <s.hauer(a)pengutronix.de> [ Upstream commit 7d90c8e6396ba245da16bedd789df6d669375408 ] The tx_mask check doesn't reflect what the driver and the chip support. The check currently checks for exactly two slots being enabled. The tlv320adcx140 supports anything between one and eight channels, so relax the check accordingly. The tlv320adcx140 supports arbitrary tx_mask settings, but the driver currently only supports adjacent slots beginning with the first slot, so extend the check to check that the first slot is being used and that there are no holes in the tx_mask. Leave a comment to make it's the driver that limits the tx_mask settings, not the chip itself. While at it remove the set-but-unused struct adcx140p_priv::tdm_delay field. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Link: https://lore.kernel.org/r/20220624105716.2579539-1-s.hauer@pengutronix.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320adcx140.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 32b120d624b2..bf18fcfcaeb9 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -33,7 +33,6 @@ struct adcx140_priv { bool micbias_vg; unsigned int dai_fmt; - unsigned int tdm_delay; unsigned int slot_width; }; @@ -792,12 +791,13 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - unsigned int lsb; - /* TDM based on DSP mode requires slots to be adjacent */ - lsb = __ffs(tx_mask); - if ((lsb + 1) != __fls(tx_mask)) { - dev_err(component->dev, "Invalid mask, slots must be adjacent\n"); + /* + * The chip itself supports arbitrary masks, but the driver currently + * only supports adjacent slots beginning at the first slot. + */ + if (tx_mask != GENMASK(__fls(tx_mask), 0)) { + dev_err(component->dev, "Only lower adjacent slots are supported\n"); return -EINVAL; } @@ -812,7 +812,6 @@ static int adcx140_set_dai_tdm_slot(struct snd_soc_dai *codec_dai, return -EINVAL; } - adcx140->tdm_delay = lsb; adcx140->slot_width = slot_width; return 0; -- 2.35.1

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[PATCH AUTOSEL 5.18 36/54] ASoC: ti: omap-mcbsp: duplicate sysfs error
by Sasha Levin 20 Jul '22

20 Jul '22

From: David Owens <daowens01(a)gmail.com> [ Upstream commit f0d96937d31c4615a6418e4bed5cee50a952040e ] Convert to managed versions of sysfs and clk allocation to simplify unbinding and error handling in probe. Managed sysfs node creation specifically addresses the following error seen the second time probe is attempted after sdma_pcm_platform_register() previously requsted probe deferral: sysfs: cannot create duplicate filename '/devices/platform/68000000.ocp/49022000.mcbsp/max_tx_thres' Signed-off-by: David Owens <dowens(a)precisionplanting.com> Link: https://lore.kernel.org/r/20220620183744.3176557-1-dowens@precisionplanting… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/omap-mcbsp-priv.h | 2 -- sound/soc/ti/omap-mcbsp-st.c | 14 ++------------ sound/soc/ti/omap-mcbsp.c | 19 ++----------------- 3 files changed, 4 insertions(+), 31 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h index 7865cda4bf0a..da519ea1f303 100644 --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -316,8 +316,6 @@ static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg, /* Sidetone specific API */ int omap_mcbsp_st_init(struct platform_device *pdev); -void omap_mcbsp_st_cleanup(struct platform_device *pdev); - int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp); int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp); diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c index 0bc7d26c660a..7e8179cae92e 100644 --- a/sound/soc/ti/omap-mcbsp-st.c +++ b/sound/soc/ti/omap-mcbsp-st.c @@ -347,7 +347,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data) return -ENOMEM; - st_data->mcbsp_iclk = clk_get(mcbsp->dev, "ick"); + st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick"); if (IS_ERR(st_data->mcbsp_iclk)) { dev_warn(mcbsp->dev, "Failed to get ick, sidetone might be broken\n"); @@ -359,7 +359,7 @@ int omap_mcbsp_st_init(struct platform_device *pdev) if (!st_data->io_base_st) return -ENOMEM; - ret = sysfs_create_group(&mcbsp->dev->kobj, &sidetone_attr_group); + ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group); if (ret) return ret; @@ -368,16 +368,6 @@ int omap_mcbsp_st_init(struct platform_device *pdev) return 0; } -void omap_mcbsp_st_cleanup(struct platform_device *pdev) -{ - struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev); - - if (mcbsp->st_data) { - sysfs_remove_group(&mcbsp->dev->kobj, &sidetone_attr_group); - clk_put(mcbsp->st_data->mcbsp_iclk); - } -} - static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 4479d74f0a45..9933b33c80ca 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -702,8 +702,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10; mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10; - ret = sysfs_create_group(&mcbsp->dev->kobj, - &additional_attr_group); + ret = devm_device_add_group(mcbsp->dev, &additional_attr_group); if (ret) { dev_err(mcbsp->dev, "Unable to create additional controls\n"); @@ -711,16 +710,7 @@ static int omap_mcbsp_init(struct platform_device *pdev) } } - ret = omap_mcbsp_st_init(pdev); - if (ret) - goto err_st; - - return 0; - -err_st: - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - return ret; + return omap_mcbsp_st_init(pdev); } /* @@ -1431,11 +1421,6 @@ static int asoc_mcbsp_remove(struct platform_device *pdev) if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req)) cpu_latency_qos_remove_request(&mcbsp->pm_qos_req); - if (mcbsp->pdata->buffer_size) - sysfs_remove_group(&mcbsp->dev->kobj, &additional_attr_group); - - omap_mcbsp_st_cleanup(pdev); - return 0; } -- 2.35.1

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Behringer UMC204HD HiFi "Default" profile crackling audio
by GitHub issues - opened 20 Jul '22

20 Jul '22

alsa-project/alsa-ucm-conf issue #192 was opened from acepukas: Since the recent update to 1.2.7.2, the Behringer UMC204HD "Default" profile has crackling audio through "Line A" and "Line B" ports. Switching to the "Direct" profile allows for clean audio but it's output over 4 speakers (front left and right, rear left and right). Not really sure how to trouble shoot this. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/192 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ASoC: Is this bug ? DAPM source vs sink
by Kuninori Morimoto 20 Jul '22

20 Jul '22

Hi ASoC Cc Lars-Peter I have noticed that DAPM source vs sink are strange. ASoC had used source/sink or input/output before, but it was exchanged by this patch a3423b02cf745c1418f1f199646c450d6fc9ca4e ("ASoC: dapm: Consolidate input and output path handling") I think source/sink pair are converted to SND_SOC_DAPM_DIR_IN/OUT by this patch. source <=> IN sink <=> OUT But, its for_each are inverted #define snd_soc_dapm_widget_for_each_sink_path(w, p) \ snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_IN, p) #define snd_soc_dapm_widget_for_each_source_path(w, p) \ snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_OUT, p) I have thought this for_each is for special handling, but, the original was sink is for sink, source is for source after all. #define snd_soc_dapm_widget_for_each_sink_path(w, p) \ list_for_each_entry(p, &w->sinks, list_source) #define snd_soc_dapm_widget_for_each_source_path(w, p) \ list_for_each_entry(p, &w->sources, list_sink) So, I guess current for_each is wrong, but I'm not 100% sure #define snd_soc_dapm_widget_for_each_sink_path(w, p) \ - snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_IN, p) + snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_OUT, p) #define snd_soc_dapm_widget_for_each_source_path(w, p) \ - snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_OUT, p) + snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_IN, p) I'm happy to create patch if I could get your ack. Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH 00/11] ASoC: SOF: Intel: updates and cleanups
by Pierre-Louis Bossart 19 Jul '22

19 Jul '22

Set of updates for IPC3, IPC4, MTL support and cleanups for the topology filename override which was broken for HDaudio platforms. It may be too late for 5.20, it's ok if these patches are applied in the next cycle. The topology override depends on the patch "ASoC: SOF: Intel: enable dmic handling with 2 or fewer SoundWire links" shared in a separate patchset. Bard Liao (1): ASoC: SOF: ipc4-topology: move ida allocate/free to widget_setup/free Peter Ujfalusi (2): ASoC: SOF: ipc3-loader: Print out the non matching ext_man magic number ASoC: SOF: Intel: atom: don't keep a temporary string in fixup_tplg_name Pierre-Louis Bossart (4): ASoC: SOF: Intel: hda: don't set fw_filename ASoC: SOF: Intel: hda: only fixup topology name if not set already ASoC: SOF: probes: rename assign/free callbacks as startup/shutdown ASoC: SOF: Intel: hda-stream: test DMA buffer first in hw_params Rander Wang (3): ASoC: SOF: ipc4-pcm: set pcm rate to dai setting ASoC: SOF: add ipc4 SRC module support ASoC: SOF: ipc4-topology: set domain bit based on dp domain type Yong Zhi (1): ASoC: SOF: Intel: MTL: remove redundant snd_sof_dsp_read() call sound/soc/sof/intel/atom.c | 16 +-- sound/soc/sof/intel/hda-probes.c | 16 +-- sound/soc/sof/intel/hda-stream.c | 15 ++- sound/soc/sof/intel/hda.c | 71 +++++++++----- sound/soc/sof/intel/mtl.c | 13 +-- sound/soc/sof/ipc3-loader.c | 6 +- sound/soc/sof/ipc4-pcm.c | 4 + sound/soc/sof/ipc4-topology.c | 156 +++++++++++++++++++++++++----- sound/soc/sof/ipc4-topology.h | 27 +++++- sound/soc/sof/sof-client-probes.c | 8 +- sound/soc/sof/sof-client-probes.h | 8 +- 11 files changed, 250 insertions(+), 90 deletions(-) -- 2.34.1

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[PATCH] ASOC: nau8821: add new widget to control system clock
by Wallace Lin 19 Jul '22

19 Jul '22

From: Wallace Lin <savagecin0(a)gmail.com> Add new widget to control system clock for power saving. Signed-off-by: Wallace Lin <savagecin0(a)gmail.com> Signed-off-by: Wallace Lin <sjlin0(a)nuvoton.com> --- sound/soc/codecs/nau8821.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 2600be250a3c..83a17c48d594 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -36,6 +36,7 @@ static int nau8821_configure_sysclk(struct nau8821 *nau8821, int clk_id, unsigned int freq); +static bool nau8821_is_jack_inserted(struct regmap *regmap); struct nau8821_fll { int mclk_src; @@ -495,7 +496,33 @@ static int nau8821_output_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int system_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + dev_dbg(nau8821->dev, "system clock control : POWER OFF\n"); + /* Set clock source to disable or internal clock before the + * playback or capture end. Codec needs clock for Jack + * detection and button press if jack inserted; otherwise, + * the clock should be closed. + */ + if (nau8821_is_jack_inserted(nau8821->regmap)) { + nau8821_configure_sysclk(nau8821, + NAU8821_CLK_INTERNAL, 0); + } else { + nau8821_configure_sysclk(nau8821, NAU8821_CLK_DIS, 0); + } + } + return 0; +} + static const struct snd_soc_dapm_widget nau8821_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("System Clock", SND_SOC_NOPM, 0, 0, + system_clock_control, SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("MICBIAS", NAU8821_R74_MIC_BIAS, NAU8821_MICBIAS_POWERUP_SFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMIC Clock", SND_SOC_NOPM, 0, 0, @@ -607,6 +634,9 @@ static const struct snd_soc_dapm_route nau8821_dapm_routes[] = { {"AIFTX", NULL, "ADCL Digital path"}, {"AIFTX", NULL, "ADCR Digital path"}, + {"AIFTX", NULL, "System Clock"}, + {"AIFRX", NULL, "System Clock"}, + {"DDACL", NULL, "AIFRX"}, {"DDACR", NULL, "AIFRX"}, -- 2.25.1

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[PATCH] ASoC: amd: yc: Decrease level of error message
by Mario Limonciello 19 Jul '22

19 Jul '22

On a number of platforms that contain acp3x controller a new ERR level message is showing up: `acp6x pci device not found` This is because ACP3x and ACP6x share same PCI ID but can be identified by PCI revision. As this is expected behavior for a system with ACP3x decrease message to debug. Fixes: b1630fcbfde6c ("ASoC: amd: yc: add new YC platform varaint support") Cc: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/yc/pci-acp6x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/pci-acp6x.c b/sound/soc/amd/yc/pci-acp6x.c index 20f7a99783f20..77c5fa1f7af14 100644 --- a/sound/soc/amd/yc/pci-acp6x.c +++ b/sound/soc/amd/yc/pci-acp6x.c @@ -159,7 +159,7 @@ static int snd_acp6x_probe(struct pci_dev *pci, case 0x6f: break; default: - dev_err(&pci->dev, "acp6x pci device not found\n"); + dev_dbg(&pci->dev, "acp6x pci device not found\n"); return -ENODEV; } if (pci_enable_device(pci)) { -- 2.25.1

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[PATCH v1 1/1] rsnd: ssiu: add missing .quit callback for gen2
by Jiada Wang 19 Jul '22

19 Jul '22

From: Wang Jiada <jiada_wang(a)mentor.com> commit 83b220cf8eb2a ("ASoC: rsnd: implement BUSIF related code in ssiu.c") moved BUSIF related code in ssiu.c which added .quit callback in ssiu.c to disable error irq when ssiu quits. But .quit callback was added only for gen1 and is missing for gen2, this causes error irq is left enabled after ssiu quits on gen2. This commit adds missing .quit callback for gen2 Signed-off-by: Wang Jiada <jiada_wang(a)mentor.com> Cc: Dean Jenkins <Dean_Jenkins(a)mentor.com> Cc: Eugeniu Rosca <erosca(a)de.adit-jv.com> --- sound/soc/sh/rcar/ssiu.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4b8a63e336c7..19a8030a660d 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -417,6 +417,7 @@ static struct rsnd_mod_ops rsnd_ssiu_ops_gen2 = { .name = SSIU_NAME, .dma_req = rsnd_ssiu_dma_req, .init = rsnd_ssiu_init_gen2, + .quit = rsnd_ssiu_quit, .start = rsnd_ssiu_start_gen2, .stop = rsnd_ssiu_stop_gen2, .get_status = rsnd_ssiu_get_status, -- 2.25.1

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