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

[PATCH v1 0/2] Fix couple of bugs reported by Canonical
by Vitaly Rodionov 25 Apr '21

25 Apr '21

CirrusLogic: Cracking noises appears in built-in speaker when output volume is set >80% BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1923557i CirrusLogic: The default input volume is "0%" BugLink: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1924997 Stefan Binding (2): ALSA: hda/cirrus: Set Initial DMIC volume for Bullseye to 16% ALSA: hda/cirrus: Use CS8409 Equalizer to fix abnormal sounds on Bullseye sound/pci/hda/patch_cirrus.c | 37 ++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) -- 2.25.1

2 4

ASoc / PCM recording-related regression between v5.4 and v5.5
by Hin-Tak Leung 24 Apr '21

24 Apr '21

Hi, I am trying to find the cause of a rather unusual regression between kernel v5.4 and v5.5, for an out-of-tree driver on the raspberrypi. Everything looks identical between v5.4 and 5.5, within accountable differences, even when I enable dynamic debugging and echo "file sound/* +p" > /sys/kernel/debug/dynamic_debug/control for the whole of sound sub-system. It records fines against v5.4, but blocks on v5.5, and on ctrl-c, does ^CAborted by signal Interrupt... arecord: pcm_read:2145: read error: Interrupted system call I have gradually replaced the include/sound and sound/ directories between the two (the pi v5.4 and v5.5 kernels are separate branches off linux-stable, so one cannot do ordinary bisects), and narrowed it down to the following 13 commits from 5 people: 4104faaeeda0e23a169c50e43f309ff7435087b1 ASoC: soc-core: disable route checks for legacy devices 4bf2e385aa59c2fae5f880aa25cfd2b470109093 ASoC: core: Init pcm runtime work early to avoid warnings fb5126778333d289b2623a7e6260beeb1ac1b819 ASoC: core: add SND_SOC_BYTES_E dc73d73aa7145f55412611f3eead1e85ae026785 ASoC: add control components management 509ba54fcfd1e45bceebe8ccea59dc496312f1a0 ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY e443c20593de9f8efd9b2935ed40eb0bbacce30b ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() 5d07519703bc2f0bf19d33652401552a480d68b8 ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() e11381f38f34789b374880c4a149e25e8d7f0cfd ASoC: soc-core: add snd_soc_unregister_dai() 4baabbf932ed4f97df8e18cf546d39b7c2138020 ASoC: soc-dpcm: tidyup for_each_dpcm_xx() macro 2b544dd7b43b19fb55ea4fbb3e30b60eb20b7828 ASoC: soc-core: add for_each_rtd_components() and replace 33536a14879515949b065721cdb7fedb276d8e8a ASoC: soc-core: remove for_each_rtdcom_safe() 8ec241c495dde3d19a0459304298c2468c60182b ASoC: soc-core: add snd_soc_pcm_lib_ioctl() b7c5bc45ee94a03a0dc45a862180e17db8ea8e9d ASoC: soc-core: merge soc_free_pcm_runtime() and soc_rtd_free() I think it is overlay / devicetree related, since another audio device of the same family and similar hardware but different overlay works fine. As I already did 'echo "file sound/* +p" > /sys/kernel/debug/dynamic_debug/control' , I likely need some additional debug statement or enable debugging at another part to go further, despite it being a regression in 'include/sound/soc*.h' and 'sound/soc/soc-*'. Building the pi kernel cleanly - from 'make mrproper', copy .config then 'make oldconnfig' - takes about an hour, so switching between the two and compare takes about 3 hours, and quite slow. I can possibly narrow the changes further from the 13 commits (9 of them from Kuninori Morimoto), but I think some more dev_dbg() is needed as the current ones in "sound/*" don't help; and it is worth bring it to alsa-devel since it is so unusual - all the dev_dbg() in kernel/sound/ and the driver side emits to the same messages, v5.4 can records while v5.5 blocks on stuck system call. More details at https://github.com/raspberrypi/linux/issues/4279 and it was initially at https://github.com/respeaker/seeed-voicecard/issues/290 . The "seeed-8mic-voicecard-overlay.dts" is the dts corresponds to the problematic device ; The one with similar hardware but okay is "seeed-4mic-voicecard-overlay.dts" . I am subscribed to alsa-devel actually, but receive it in digest form, and haven't opened the digests for a while... Thanks a lot. Hin-Tak

3 10

Warnings on startup
by GitHub issues - opened 23 Apr '21

23 Apr '21

alsa-project/alsa-plugins issue #20 was opened from bramtayl: I've been trying to get Julia's PortAudio to load alsa-plugins (it's been a process). I thought I might as well see if I can cut down on the warnings on start up (`using PortAudio` from Julia) and some of them I've been able to knock out but some of them not so much: These ones ``` ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ``` I got rid of by commenting out lines in my alsa.conf file: ``` # pcm.rear cards.pcm.rear # pcm.center_lfe cards.pcm.center_lfe # pcm.side cards.pcm.side ``` This one ``` ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ``` I got rid of by commening out: ``` # pcm.surround21 cards.pcm.surround21 # pcm.surround40 cards.pcm.surround40 # pcm.surround41 cards.pcm.surround41 # pcm.surround50 cards.pcm.surround50 # pcm.surround51 cards.pcm.surround51 # pcm.surround71 cards.pcm.surround71 ``` I'm not sure if these work arounds are the best solutions...shouldn't alsa be able to tell that my laptop doesn't have surround sound? This one is expected, I haven't yet figured out how to build jack on Julia's BinaryBuilder: ``` ALSA lib dlmisc.c:339:(snd_dlobj_cache_get0) Cannot open shared library libasound_module_pcm_jack.so (/home/brandon/.julia/artifacts/d587f48a6815c72dec4b083bf297caec7a8e3f4b/lib/alsa-lib/libasound_module_pcm_jack.so: cannot open shared object file: No such file or directory) ``` These ones are a bit confusing; ``` ALSA lib pcm_oss.c:377:(_snd_pcm_oss_open) Unknown field port ALSA lib pcm_usb_stream.c:486:(_snd_pcm_usb_stream_open) Invalid type for card ``` Is it possible that `oss/50-oss.conf` and `usb_stream/98-usb-stream.conf` need to get updated? Issue URL : https://github.com/alsa-project/alsa-plugins/issues/20 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v2] ASoC: rt711-sdca: add the notification when volume changed
by shumingf＠realtek.com 23 Apr '21

23 Apr '21

From: Shuming Fan <shumingf(a)realtek.com> This patch adds the return value when the volume settings were changed. The userspace application might monitor the kcontrols to check which control changed. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 39848d2641c3..cc36739f7fcf 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -509,12 +509,16 @@ static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); unsigned int read_l, read_r, gain_l_val, gain_r_val; - unsigned int i, adc_vol_flag = 0; + unsigned int i, adc_vol_flag = 0, changed = 0; + unsigned int lvalue, rvalue; if (strstr(ucontrol->id.name, "FU1E Capture Volume") || strstr(ucontrol->id.name, "FU0F Capture Volume")) adc_vol_flag = 1; + regmap_read(rt711->mbq_regmap, mc->reg, &lvalue); + regmap_read(rt711->mbq_regmap, mc->rreg, &rvalue); + /* control value to 2's complement value */ /* L Channel */ gain_l_val = ucontrol->value.integer.value[0]; @@ -560,6 +564,11 @@ static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, gain_r_val &= 0xffff; } + if (lvalue != gain_l_val || rvalue != gain_r_val) + changed = 1; + else + return 0; + for (i = 0; i < 3; i++) { /* retry 3 times at most */ /* Lch*/ regmap_write(rt711->mbq_regmap, mc->reg, gain_l_val); @@ -573,7 +582,7 @@ static int rt711_sdca_set_gain_put(struct snd_kcontrol *kcontrol, break; } - return i == 3 ? -EIO : 0; + return i == 3 ? -EIO : changed; } static int rt711_sdca_set_gain_get(struct snd_kcontrol *kcontrol, -- 2.29.0

2 1

[PATCH v4] ASoC: rt711-sdca: change capture switch controls
by shumingf＠realtek.com 23 Apr '21

23 Apr '21

From: Shuming Fan <shumingf(a)realtek.com> The DAPM event and mixer control could mute/unmute the capture directly. That will be confused that capture still works if the user settings is unmute before the capture. Therefore, this patch uses the variables to record the capture switch status of DAPM and mixer. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca.c | 168 +++++++++++++++++++++++++--------- sound/soc/codecs/rt711-sdca.h | 2 + 2 files changed, 127 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index bfb7f1c8ec8f..39848d2641c3 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -642,6 +642,114 @@ static int rt711_sdca_set_gain_get(struct snd_kcontrol *kcontrol, return 0; } +static int rt711_sdca_set_fu0f_capture_ctl(struct rt711_sdca_priv *rt711) +{ + int err; + unsigned int ch_l, ch_r; + + ch_l = (rt711->fu0f_dapm_mute || rt711->fu0f_mixer_l_mute) ? 0x01 : 0x00; + ch_r = (rt711->fu0f_dapm_mute || rt711->fu0f_mixer_r_mute) ? 0x01 : 0x00; + + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_L), ch_l); + if (err < 0) + return err; + + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, + RT711_SDCA_CTL_FU_MUTE, CH_R), ch_r); + if (err < 0) + return err; + + return 0; +} + +static int rt711_sdca_set_fu1e_capture_ctl(struct rt711_sdca_priv *rt711) +{ + int err; + unsigned int ch_l, ch_r; + + ch_l = (rt711->fu1e_dapm_mute || rt711->fu1e_mixer_l_mute) ? 0x01 : 0x00; + ch_r = (rt711->fu1e_dapm_mute || rt711->fu1e_mixer_r_mute) ? 0x01 : 0x00; + + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_L), ch_l); + if (err < 0) + return err; + + err = regmap_write(rt711->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU1E, + RT711_SDCA_CTL_FU_MUTE, CH_R), ch_r); + if (err < 0) + return err; + + return 0; +} + +static int rt711_sdca_fu1e_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !rt711->fu1e_mixer_l_mute; + ucontrol->value.integer.value[1] = !rt711->fu1e_mixer_r_mute; + return 0; +} + +static int rt711_sdca_fu1e_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int err, changed = 0; + + if (rt711->fu1e_mixer_l_mute != !ucontrol->value.integer.value[0] || + rt711->fu1e_mixer_r_mute != !ucontrol->value.integer.value[1]) + changed = 1; + + rt711->fu1e_mixer_l_mute = !ucontrol->value.integer.value[0]; + rt711->fu1e_mixer_r_mute = !ucontrol->value.integer.value[1]; + err = rt711_sdca_set_fu1e_capture_ctl(rt711); + if (err < 0) + return err; + + return changed; +} + +static int rt711_sdca_fu0f_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !rt711->fu0f_mixer_l_mute; + ucontrol->value.integer.value[1] = !rt711->fu0f_mixer_r_mute; + return 0; +} + +static int rt711_sdca_fu0f_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); + int err, changed = 0; + + if (rt711->fu0f_mixer_l_mute != !ucontrol->value.integer.value[0] || + rt711->fu0f_mixer_r_mute != !ucontrol->value.integer.value[1]) + changed = 1; + + rt711->fu0f_mixer_l_mute = !ucontrol->value.integer.value[0]; + rt711->fu0f_mixer_r_mute = !ucontrol->value.integer.value[1]; + err = rt711_sdca_set_fu0f_capture_ctl(rt711); + if (err < 0) + return err; + + return changed; +} + static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); @@ -652,14 +760,10 @@ static const struct snd_kcontrol_new rt711_sdca_snd_controls[] = { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU05, RT711_SDCA_CTL_FU_VOLUME, CH_R), 0x57, 0x57, 0, rt711_sdca_set_gain_get, rt711_sdca_set_gain_put, out_vol_tlv), - SOC_DOUBLE_R("FU1E Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), - SOC_DOUBLE_R("FU0F Capture Switch", - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_L), - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, RT711_SDCA_CTL_FU_MUTE, CH_R), - 0, 1, 1), + SOC_DOUBLE_EXT("FU1E Capture Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt711_sdca_fu1e_capture_get, rt711_sdca_fu1e_capture_put), + SOC_DOUBLE_EXT("FU0F Capture Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt711_sdca_fu0f_capture_get, rt711_sdca_fu0f_capture_put), SOC_DOUBLE_R_EXT_TLV("FU1E Capture Volume", SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_L), SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, RT711_SDCA_CTL_FU_VOLUME, CH_R), @@ -809,28 +913,15 @@ static int rt711_sdca_fu0f_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - unsigned char unmute = 0x0, mute = 0x1; switch (event) { case SND_SOC_DAPM_POST_PMU: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_L), - unmute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_R), - unmute); + rt711->fu0f_dapm_mute = false; + rt711_sdca_set_fu0f_capture_ctl(rt711); break; case SND_SOC_DAPM_PRE_PMD: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_L), - mute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT711_SDCA_ENT_USER_FU0F, - RT711_SDCA_CTL_FU_MUTE, CH_R), - mute); + rt711->fu0f_dapm_mute = true; + rt711_sdca_set_fu0f_capture_ctl(rt711); break; } return 0; @@ -842,29 +933,16 @@ static int rt711_sdca_fu1e_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt711_sdca_priv *rt711 = snd_soc_component_get_drvdata(component); - unsigned char unmute = 0x0, mute = 0x1; switch (event) { case SND_SOC_DAPM_POST_PMU: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_L), - unmute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_R), - unmute); + rt711->fu1e_dapm_mute = false; + rt711_sdca_set_fu1e_capture_ctl(rt711); break; case SND_SOC_DAPM_PRE_PMD: - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_L), - mute); - regmap_write(rt711->regmap, - SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT711_SDCA_ENT_USER_FU1E, - RT711_SDCA_CTL_FU_MUTE, CH_R), - mute); - break; + rt711->fu1e_dapm_mute = true; + rt711_sdca_set_fu1e_capture_ctl(rt711); + break; } return 0; } @@ -1330,6 +1408,10 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, */ rt711->hw_init = false; rt711->first_hw_init = false; + rt711->fu0f_dapm_mute = true; + rt711->fu1e_dapm_mute = true; + rt711->fu0f_mixer_l_mute = rt711->fu0f_mixer_r_mute = true; + rt711->fu1e_mixer_l_mute = rt711->fu1e_mixer_r_mute = true; /* JD source uses JD2 in default */ rt711->jd_src = RT711_JD2; diff --git a/sound/soc/codecs/rt711-sdca.h b/sound/soc/codecs/rt711-sdca.h index 98a022cec0bd..43ae82b7fdb3 100644 --- a/sound/soc/codecs/rt711-sdca.h +++ b/sound/soc/codecs/rt711-sdca.h @@ -30,6 +30,8 @@ struct rt711_sdca_priv { int jack_type, jd_src; unsigned int scp_sdca_stat1, scp_sdca_stat2; int hw_ver; + bool fu0f_dapm_mute, fu0f_mixer_l_mute, fu0f_mixer_r_mute; + bool fu1e_dapm_mute, fu1e_mixer_l_mute, fu1e_mixer_r_mute; }; struct sdw_stream_data { -- 2.29.0

2 1

[PATCH] ASoC: simple-card: Fix breakage on kontron-sl28-var3-ads2
by Mark Brown 23 Apr '21

23 Apr '21

A KernelCI bisection identified 59c35c44a9cf89 "ASoC: simple-card: add simple_parse_node()" as causing simple-card to fail to instantiate on kontron-sl28-var3-ads2 systems. Since the merge window is expected to open over the weekend drop that commit and subsequent ones which depend on it for now in case other systems are affected too. The boot log showed the error as: <4>[ 9.948821] sysfs: cannot create duplicate filename '/devices/platform/sound/(null)-wm8904-hifi' (backtrace) <3>[ 10.191982] kobject_add_internal failed for (null)-wm8904-hifi with -EEXIST, don't try to register things with the same name in the same directory. The dropped commits are: 73371bacf0475a20ab6 "ASoC: audio-graph: tidyup graph_dai_link_of_dpcm()" 434392271afcff350fe "ASoC: simple-card: add simple_link_init()" 59c35c44a9cf89a83a9 "ASoC: simple-card: add simple_parse_node()" Reported-by: Guillaume Tucker <guillaume.tucker(a)collabora.com> Reported-by: "kernelci.org bot" <bot(a)kernelci.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/audio-graph-card.c | 30 ++--- sound/soc/generic/simple-card.c | 178 ++++++++++++++------------- 2 files changed, 106 insertions(+), 102 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 0159a4576e9c..2c8a2fcb7922 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -276,19 +276,24 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct link_info *li) { struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct device_node *top = dev->of_node; struct device_node *ep = li->cpu ? cpu_ep : codec_ep; + struct device_node *port; + struct device_node *ports; + struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); + struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); char dai_name[64]; int ret; + port = of_get_parent(ep); + ports = of_get_parent(port); + dev_dbg(dev, "link_of DPCM (%pOF)\n", ep); if (li->cpu) { - struct snd_soc_card *card = simple_priv_to_card(priv); - struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); - /* Codec is dummy */ /* FE settings */ @@ -297,7 +302,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, ret = graph_parse_node(priv, cpu_ep, li, 1); if (ret) - return ret; + goto out_put_node; snprintf(dai_name, sizeof(dai_name), "fe.%pOFP.%s", cpus->of_node, cpus->dai_name); @@ -314,10 +319,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (card->component_chaining && !soc_component_is_pcm(cpus)) dai_link->no_pcm = 1; } else { - struct snd_soc_codec_conf *cconf = simple_props_to_codec_conf(dai_props, 0); - struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); - struct device_node *port; - struct device_node *ports; + struct snd_soc_codec_conf *cconf; /* CPU is dummy */ @@ -325,25 +327,22 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; + cconf = simple_props_to_codec_conf(dai_props, 0); + ret = graph_parse_node(priv, codec_ep, li, 0); if (ret < 0) - return ret; + goto out_put_node; snprintf(dai_name, sizeof(dai_name), "be.%pOFP.%s", codecs->of_node, codecs->dai_name); /* check "prefix" from top node */ - port = of_get_parent(ep); - ports = of_get_parent(port); snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node, "prefix"); if (of_node_name_eq(ports, "ports")) snd_soc_of_parse_node_prefix(ports, cconf, codecs->of_node, "prefix"); snd_soc_of_parse_node_prefix(port, cconf, codecs->of_node, "prefix"); - - of_node_put(ports); - of_node_put(port); } graph_parse_convert(dev, ep, &dai_props->adata); @@ -352,8 +351,11 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); +out_put_node: li->link++; + of_node_put(ports); + of_node_put(port); return ret; } diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6a04632944b8..a1373be4558f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -93,11 +93,12 @@ static void simple_parse_convert(struct device *dev, } static void simple_parse_mclk_fs(struct device_node *top, - struct device_node *np, + struct device_node *cpu, + struct device_node *codec, struct simple_dai_props *props, char *prefix) { - struct device_node *node = of_get_parent(np); + struct device_node *node = of_get_parent(cpu); char prop[128]; snprintf(prop, sizeof(prop), "%smclk-fs", PREFIX); @@ -105,79 +106,12 @@ static void simple_parse_mclk_fs(struct device_node *top, snprintf(prop, sizeof(prop), "%smclk-fs", prefix); of_property_read_u32(node, prop, &props->mclk_fs); - of_property_read_u32(np, prop, &props->mclk_fs); + of_property_read_u32(cpu, prop, &props->mclk_fs); + of_property_read_u32(codec, prop, &props->mclk_fs); of_node_put(node); } -static int simple_parse_node(struct asoc_simple_priv *priv, - struct device_node *np, - struct link_info *li, - char *prefix, - int is_cpu) -{ - struct device *dev = simple_priv_to_dev(priv); - struct device_node *top = dev->of_node; - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); - struct snd_soc_dai_link_component *dlc; - struct asoc_simple_dai *dai; - int ret, single = 0; - - if (is_cpu) { - dlc = asoc_link_to_cpu(dai_link, 0); - dai = simple_props_to_dai_cpu(dai_props, 0); - } else { - dlc = asoc_link_to_codec(dai_link, 0); - dai = simple_props_to_dai_codec(dai_props, 0); - } - - simple_parse_mclk_fs(top, np, dai_props, prefix); - - ret = asoc_simple_parse_dai(np, dlc, &single); - if (ret) - return ret; - - ret = asoc_simple_parse_clk(dev, np, dai, dlc); - if (ret) - return ret; - - ret = asoc_simple_parse_tdm(np, dai); - if (ret) - return ret; - - if (is_cpu) { - struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); - - asoc_simple_canonicalize_cpu(dlc, single); - asoc_simple_canonicalize_platform(platforms, dlc); - } - - return 0; -} - -static int simple_link_init(struct asoc_simple_priv *priv, - struct device_node *node, - struct device_node *codec, - struct link_info *li, - char *prefix, - char *name) -{ - struct device *dev = simple_priv_to_dev(priv); - struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - int ret; - - ret = asoc_simple_parse_daifmt(dev, node, codec, - prefix, &dai_link->dai_fmt); - if (ret < 0) - return 0; - - dai_link->init = asoc_simple_dai_init; - dai_link->ops = &simple_ops; - - return asoc_simple_set_dailink_name(dev, dai_link, name); -} - static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct device_node *np, struct device_node *codec, @@ -187,32 +121,50 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); + struct asoc_simple_dai *dai; struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); + struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); struct device_node *top = dev->of_node; struct device_node *node = of_get_parent(np); char *prefix = ""; - char dai_name[64]; int ret; dev_dbg(dev, "link_of DPCM (%pOF)\n", np); + li->link++; + /* For single DAI link & old style of DT node */ if (is_top) prefix = PREFIX; if (li->cpu) { + int is_single_links = 0; + /* Codec is dummy */ /* FE settings */ dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; - ret = simple_parse_node(priv, np, li, prefix, 1); + dai = simple_props_to_dai_cpu(dai_props, 0); + + ret = asoc_simple_parse_dai(np, cpus, &is_single_links); + if (ret) + goto out_put_node; + + ret = asoc_simple_parse_clk(dev, np, dai, cpus); if (ret < 0) goto out_put_node; - snprintf(dai_name, sizeof(dai_name), "fe.%s", cpus->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "fe.%s", + cpus->dai_name); + if (ret < 0) + goto out_put_node; + + asoc_simple_canonicalize_cpu(cpus, is_single_links); + asoc_simple_canonicalize_platform(platforms, cpus); } else { struct snd_soc_codec_conf *cconf; @@ -222,13 +174,22 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; + dai = simple_props_to_dai_codec(dai_props, 0); cconf = simple_props_to_codec_conf(dai_props, 0); - ret = simple_parse_node(priv, np, li, prefix, 0); + ret = asoc_simple_parse_dai(np, codecs, NULL); + if (ret < 0) + goto out_put_node; + + ret = asoc_simple_parse_clk(dev, np, dai, codecs); if (ret < 0) goto out_put_node; - snprintf(dai_name, sizeof(dai_name), "be.%s", codecs->dai_name); + ret = asoc_simple_set_dailink_name(dev, dai_link, + "be.%s", + codecs->dai_name); + if (ret < 0) + goto out_put_node; /* check "prefix" from top node */ snd_soc_of_parse_node_prefix(top, cconf, codecs->of_node, @@ -240,14 +201,23 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, } simple_parse_convert(dev, np, &dai_props->adata); + simple_parse_mclk_fs(top, np, codec, dai_props, prefix); + + ret = asoc_simple_parse_tdm(np, dai); + if (ret) + goto out_put_node; + + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); + if (ret < 0) + goto out_put_node; snd_soc_dai_link_set_capabilities(dai_link); - ret = simple_link_init(priv, node, codec, li, prefix, dai_name); + dai_link->ops = &simple_ops; + dai_link->init = asoc_simple_dai_init; out_put_node: - li->link++; - of_node_put(node); return ret; } @@ -260,19 +230,23 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, { struct device *dev = simple_priv_to_dev(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); + struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); + struct asoc_simple_dai *cpu_dai = simple_props_to_dai_cpu(dai_props, 0); + struct asoc_simple_dai *codec_dai = simple_props_to_dai_codec(dai_props, 0); struct snd_soc_dai_link_component *cpus = asoc_link_to_cpu(dai_link, 0); struct snd_soc_dai_link_component *codecs = asoc_link_to_codec(dai_link, 0); struct snd_soc_dai_link_component *platforms = asoc_link_to_platform(dai_link, 0); + struct device_node *top = dev->of_node; struct device_node *cpu = NULL; struct device_node *node = NULL; struct device_node *plat = NULL; - char dai_name[64]; char prop[128]; char *prefix = ""; - int ret; + int ret, single_cpu = 0; cpu = np; node = of_get_parent(np); + li->link++; dev_dbg(dev, "link_of (%pOF)\n", node); @@ -283,11 +257,18 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, snprintf(prop, sizeof(prop), "%splat", prefix); plat = of_get_child_by_name(node, prop); - ret = simple_parse_node(priv, cpu, li, prefix, 1); + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); if (ret < 0) goto dai_link_of_err; - ret = simple_parse_node(priv, codec, li, prefix, 0); + simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); + + ret = asoc_simple_parse_dai(cpu, cpus, &single_cpu); + if (ret < 0) + goto dai_link_of_err; + + ret = asoc_simple_parse_dai(codec, codecs, NULL); if (ret < 0) goto dai_link_of_err; @@ -295,18 +276,39 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) goto dai_link_of_err; - snprintf(dai_name, sizeof(dai_name), - "%s-%s", cpus->dai_name, codecs->dai_name); - ret = simple_link_init(priv, node, codec, li, prefix, dai_name); + ret = asoc_simple_parse_tdm(cpu, cpu_dai); + if (ret < 0) + goto dai_link_of_err; + + ret = asoc_simple_parse_tdm(codec, codec_dai); if (ret < 0) goto dai_link_of_err; + ret = asoc_simple_parse_clk(dev, cpu, cpu_dai, cpus); + if (ret < 0) + goto dai_link_of_err; + + ret = asoc_simple_parse_clk(dev, codec, codec_dai, codecs); + if (ret < 0) + goto dai_link_of_err; + + ret = asoc_simple_set_dailink_name(dev, dai_link, + "%s-%s", + cpus->dai_name, + codecs->dai_name); + if (ret < 0) + goto dai_link_of_err; + + dai_link->ops = &simple_ops; + dai_link->init = asoc_simple_dai_init; + + asoc_simple_canonicalize_cpu(cpus, single_cpu); + asoc_simple_canonicalize_platform(platforms, cpus); + dai_link_of_err: of_node_put(plat); of_node_put(node); - li->link++; - return ret; } -- 2.20.1

1 0

[PATCH v1 01/14] spi: pxa2xx: Use one point of return when ->probe() fails
by Andy Shevchenko 23 Apr '21

23 Apr '21

When we can't allocate SPI controller, jump to the error path rather than return locally. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- drivers/spi/spi-pxa2xx.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/spi/spi-pxa2xx.c b/drivers/spi/spi-pxa2xx.c index 0cc767283674..0ef22045b006 100644 --- a/drivers/spi/spi-pxa2xx.c +++ b/drivers/spi/spi-pxa2xx.c @@ -1701,8 +1701,8 @@ static int pxa2xx_spi_probe(struct platform_device *pdev) if (!controller) { dev_err(&pdev->dev, "cannot alloc spi_controller\n"); - pxa_ssp_free(ssp); - return -ENOMEM; + status = -ENOMEM; + goto out_error_controller_alloc; } drv_data = spi_controller_get_devdata(controller); drv_data->controller = controller; -- 2.30.2

2 16

New udev rules for firewire character device are added to systemd
by Takashi Sakamoto 23 Apr '21

23 Apr '21

Hi, Summary of this message: * systemd got udev rules with new database for firewire node/unit * then fw character device for audio is owned by 'audio' group with ACL * the entries are added by my investigation, thus doesn't cover all * if you have firewire audio devices not listed in README of below repository, please contact to me with image of configuration ROM: * https://github.com/takaswie/am-config-roms/ The way to create image file of configuration ROM is typically: ``` $ cat /sys/bus/firewire/devices/fw1/config_rom > filename.img ``` Here, I presuppose that Linux FireWire subsystem detects your device as 'fw1'. Well, in the past, access permission of Linux firewire character device is decided by udev rules just for video devices[1]. This was inconvenient some project such as ALSA and FFADO to produce audio application. The source code of libffado includes own file for udev rules[2] to take firewire character device owned by 'audio' group. Additionally the rules gives 'ID_FFADO' tag, and systemd includes another udev rule[3] to ACL at logging-in time according to it. As a whole, the above is not comprehensive and self-contained. I proposed patchset to systemd for better solution and today it was merged. * https://github.com/systemd/systemd/pull/19124 In the patchset, I add some udev rules, based on hwdb for new entries of node and units in IEEE 1394 bus. You can see the database[4]. The entries of database have below variables when matching to either node or unit devices: * IEEE1394_UNIT_FUNCTION_MIDI * IEEE1394_UNIT_FUNCTION_AUDIO * IEEE1394_UNIT_FUNCTION_VIDEO The added udev rules interpret the content of variables and decide group owner of fw character device(see [1]). Furthermore, the variables are used again to decide ACL in logging-in time(see [3], too). The entries of database also include below variables: * ID_VENDOR_FROM_DATABASE * ID_MODEL_FROM_DATABASE They are expected to use applications such as PipeWire and PulseAudio for better names of sound device, which binds to unit instead of node. I expect the variables can obsolete my former patch for pulseaudio[5]. I handy write the entries of database from my investigation, thus it could includes the lack of your device, or mistakes. I wish you to contact to me with image file of configuration ROM when you can not find your device in README of my collection repository[6], or when you find any mistakes in database file. Thanks for your cooperation in advance. [1] 4 rules in 'rules.d/50-udev-default.rules' https://github.com/systemd/systemd/blob/main/rules.d/50-udev-default.rules.… [2] many rules in 'libffado/60-ffado.rules' http://subversion.ffado.org/browser/trunk/libffado/libffado/60-ffado.rules?… [3] 'src/login/70-uaccess.rules.m4' https://github.com/systemd/systemd/blob/main/src/login/70-uaccess.rules.m4 [4] 'hwdb.d/80-ieee1394-unit-function.hwdb' https://github.com/systemd/systemd/blob/main/hwdb.d/80-ieee1394-unit-functi… [5] udev: use ID_MODEL/ID_VENDOR to give friendly name for FireWire devices https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/3ac73598c67c [6] https://github.com/takaswie/am-config-roms/ Takashi Sakamoto

1 0

[PATCH] ASoC: adau17x1: Avoid overwriting CHPF
by Niklas Carlsson 22 Apr '21

22 Apr '21

From: Niklas Carlsson <niklasc(a)axis.com> Configuring number of channels per LRCLK frame by using e.g. snd_soc_dai_set_tdm_slot before configuring DAI format was being overwritten by the latter due to a regmap_write which would write over the whole register. Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau17x1.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 546ee8178038..8aae7ab74091 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -553,6 +553,7 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, { struct adau *adau = snd_soc_component_get_drvdata(dai->component); unsigned int ctrl0, ctrl1; + unsigned int ctrl0_mask; int lrclk_pol; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -612,8 +613,16 @@ static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, if (lrclk_pol) ctrl0 |= ADAU17X1_SERIAL_PORT0_LRCLK_POL; - regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0); - regmap_write(adau->regmap, ADAU17X1_SERIAL_PORT1, ctrl1); + /* Set the mask to update all relevant bits in ADAU17X1_SERIAL_PORT0 */ + ctrl0_mask = ADAU17X1_SERIAL_PORT0_MASTER | + ADAU17X1_SERIAL_PORT0_LRCLK_POL | + ADAU17X1_SERIAL_PORT0_BCLK_POL | + ADAU17X1_SERIAL_PORT0_PULSE_MODE; + + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT0, ctrl0_mask, + ctrl0); + regmap_update_bits(adau->regmap, ADAU17X1_SERIAL_PORT1, + ADAU17X1_SERIAL_PORT1_DELAY_MASK, ctrl1); adau->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; -- 2.20.1

2 1

[PATCH] ASoC: tegra: mark runtime-pm functions as __maybe_unused
by Arnd Bergmann 22 Apr '21

22 Apr '21

From: Arnd Bergmann <arnd(a)arndb.de> A reorganization of the driver source led to two of them causing a compile time warning in some configurations: tegra/tegra20_spdif.c:36:12: error: 'tegra20_spdif_runtime_resume' defined but not used [-Werror=unused-function] 36 | static int tegra20_spdif_runtime_resume(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~ tegra/tegra20_spdif.c:27:12: error: 'tegra20_spdif_runtime_suspend' defined but not used [-Werror=unused-function] 27 | static int tegra20_spdif_runtime_suspend(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ tegra/tegra30_ahub.c:64:12: error: 'tegra30_ahub_runtime_resume' defined but not used [-Werror=unused-function] 64 | static int tegra30_ahub_runtime_resume(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~~~~~~ tegra/tegra30_ahub.c:43:12: error: 'tegra30_ahub_runtime_suspend' defined but not used [-Werror=unused-function] 43 | static int tegra30_ahub_runtime_suspend(struct device *dev) | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~ Mark these functions as __maybe_unused to avoid this kind of warning. Fixes: b5571449e618 ("ASoC: tegra30: ahub: Remove handing of disabled runtime PM") Fixes: c53b396f0dd4 ("ASoC: tegra20: spdif: Remove handing of disabled runtime PM") Fixes: 80ec4a4cb36d ("ASoC: tegra20: i2s: Remove handing of disabled runtime PM") Fixes: b5f6f781fcb2 ("ASoC: tegra30: i2s: Remove handing of disabled runtime PM") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/tegra/tegra20_i2s.c | 4 ++-- sound/soc/tegra/tegra20_spdif.c | 4 ++-- sound/soc/tegra/tegra30_ahub.c | 4 ++-- sound/soc/tegra/tegra30_i2s.c | 4 ++-- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 1b27f81c10fe..b280ebd72591 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -34,7 +34,7 @@ #define DRV_NAME "tegra20-i2s" -static int tegra20_i2s_runtime_suspend(struct device *dev) +static __maybe_unused int tegra20_i2s_runtime_suspend(struct device *dev) { struct tegra20_i2s *i2s = dev_get_drvdata(dev); @@ -45,7 +45,7 @@ static int tegra20_i2s_runtime_suspend(struct device *dev) return 0; } -static int tegra20_i2s_runtime_resume(struct device *dev) +static __maybe_unused int tegra20_i2s_runtime_resume(struct device *dev) { struct tegra20_i2s *i2s = dev_get_drvdata(dev); int ret; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 7b597ee63cb5..de698ff2a69c 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -24,7 +24,7 @@ #define DRV_NAME "tegra20-spdif" -static int tegra20_spdif_runtime_suspend(struct device *dev) +static __maybe_unused int tegra20_spdif_runtime_suspend(struct device *dev) { struct tegra20_spdif *spdif = dev_get_drvdata(dev); @@ -33,7 +33,7 @@ static int tegra20_spdif_runtime_suspend(struct device *dev) return 0; } -static int tegra20_spdif_runtime_resume(struct device *dev) +static __maybe_unused int tegra20_spdif_runtime_resume(struct device *dev) { struct tegra20_spdif *spdif = dev_get_drvdata(dev); int ret; diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d1718f3af3cd..4692c70ed933 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -40,7 +40,7 @@ static inline void tegra30_audio_write(u32 reg, u32 val) regmap_write(ahub->regmap_ahub, reg, val); } -static int tegra30_ahub_runtime_suspend(struct device *dev) +static __maybe_unused int tegra30_ahub_runtime_suspend(struct device *dev) { regcache_cache_only(ahub->regmap_apbif, true); regcache_cache_only(ahub->regmap_ahub, true); @@ -61,7 +61,7 @@ static int tegra30_ahub_runtime_suspend(struct device *dev) * stopping streams should dynamically adjust the clock as required. However, * this is not yet implemented. */ -static int tegra30_ahub_runtime_resume(struct device *dev) +static __maybe_unused int tegra30_ahub_runtime_resume(struct device *dev) { int ret; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 8730ffa0f691..36344f0a64c1 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -36,7 +36,7 @@ #define DRV_NAME "tegra30-i2s" -static int tegra30_i2s_runtime_suspend(struct device *dev) +static __maybe_unused int tegra30_i2s_runtime_suspend(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); @@ -47,7 +47,7 @@ static int tegra30_i2s_runtime_suspend(struct device *dev) return 0; } -static int tegra30_i2s_runtime_resume(struct device *dev) +static __maybe_unused int tegra30_i2s_runtime_resume(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); int ret; -- 2.29.2

2 1