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

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

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[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

[PATCH] ALSA: usb-audio: Apply implicit feedback mode for BOSS devices
by Takashi Iwai 22 Apr '21

22 Apr '21

During the recent rewrite of the implicit feedback support, we've tested to apply the implicit fb on BOSS devices, but it failed, as the capture stream didn't start without the playback. As the end result, it got another type of quirk for tying both streams but starts playback always (commit 6234fdc1cede "ALSA: usb-audio: Quirk for BOSS GT-001"). Meanwhile, Mike Oliphant has tested the real implicit feedback mode for the playback again with the latest code, and found out that it actually works if the initial feedback sync is skipped; that is, on those BOSS devices, the playback stream has to be started at first without waiting for the capture URB completions. Otherwise it gets stuck. In the rest operations after the capture stream processed, we can take them as the implicit feedback source. This patch is an attempt to improve the support for BOSS devices with the implicit feedback mode in the way described above. It adds a new flag to snd_usb_audio, playback_first, indicating that the playback stream starts without sync with the initial capture completion. This flag is set in the quirk table with the new IMPLICIT_FB_BOTH type. Reported-and-tested-by: Mike Oliphant <oliphant(a)nostatic.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 3 ++- sound/usb/implicit.c | 35 +++++++++++++++++++++++------------ sound/usb/usbaudio.h | 1 + 3 files changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 102d53515a76..f4c3d2b38abb 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -1375,7 +1375,8 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (!ep_state_update(ep, EP_STATE_STOPPED, EP_STATE_RUNNING)) goto __error; - if (snd_usb_endpoint_implicit_feedback_sink(ep)) { + if (snd_usb_endpoint_implicit_feedback_sink(ep) && + !ep->chip->playback_first) { for (i = 0; i < ep->nurbs; i++) { struct snd_urb_ctx *ctx = ep->urb + i; list_add_tail(&ctx->ready_list, &ep->ready_playback_urbs); diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 19622c58b72b..4bd9c105a529 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -21,6 +21,7 @@ enum { IMPLICIT_FB_NONE, IMPLICIT_FB_GENERIC, IMPLICIT_FB_FIXED, + IMPLICIT_FB_BOTH, /* generic playback + capture (for BOSS) */ }; struct snd_usb_implicit_fb_match { @@ -36,6 +37,9 @@ struct snd_usb_implicit_fb_match { #define IMPLICIT_FB_FIXED_DEV(vend, prod, ep, ifnum) \ { .id = USB_ID(vend, prod), .type = IMPLICIT_FB_FIXED, .ep_num = (ep),\ .iface = (ifnum) } +#define IMPLICIT_FB_BOTH_DEV(vend, prod, ep, ifnum) \ + { .id = USB_ID(vend, prod), .type = IMPLICIT_FB_BOTH, .ep_num = (ep),\ + .iface = (ifnum) } #define IMPLICIT_FB_SKIP_DEV(vend, prod) \ { .id = USB_ID(vend, prod), .type = IMPLICIT_FB_NONE } @@ -75,14 +79,14 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Implicit feedback quirk table for capture: only FIXED type */ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ - IMPLICIT_FB_FIXED_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0130, 0x0d, 0x01), /* BOSS BR-80 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0171, 0x0d, 0x01), /* BOSS RC-505 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0185, 0x0d, 0x01), /* BOSS GP-10 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ + IMPLICIT_FB_BOTH_DEV(0x0582, 0x0203, 0x0d, 0x01), /* BOSS AD-10 */ {} /* terminator */ }; @@ -268,10 +272,17 @@ static int audioformat_implicit_fb_quirk(struct snd_usb_audio *chip, } } - /* Don't apply playback quirks for the devices with capture quirk */ + /* Special handling for devices with capture quirks */ p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts); - if (p && p->type == IMPLICIT_FB_FIXED) - return 0; /* no quirk */ + if (p) { + switch (p->type) { + case IMPLICIT_FB_FIXED: + return 0; /* no quirk */ + case IMPLICIT_FB_BOTH: + chip->playback_first = 1; + return add_generic_implicit_fb(chip, fmt, alts); + } + } /* Generic UAC2 implicit feedback */ if (attr == USB_ENDPOINT_SYNC_ASYNC && @@ -321,7 +332,7 @@ static int audioformat_capture_quirk(struct snd_usb_audio *chip, const struct snd_usb_implicit_fb_match *p; p = find_implicit_fb_entry(chip, capture_implicit_fb_quirks, alts); - if (p && p->type == IMPLICIT_FB_FIXED) + if (p && (p->type == IMPLICIT_FB_FIXED || p->type == IMPLICIT_FB_BOTH)) return add_implicit_fb_sync_ep(chip, fmt, p->ep_num, 0, p->iface, NULL); return 0; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a536ee33d36e..538831cbd925 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -37,6 +37,7 @@ struct snd_usb_audio { unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ unsigned int tx_length_quirk:1; /* Put length specifier in transfers */ unsigned int need_delayed_register:1; /* warn for delayed registration */ + unsigned int playback_first:1; /* for implicit fb: don't wait for the first capture URBs */ int num_interfaces; int num_suspended_intf; int sample_rate_read_error; -- 2.26.2

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[PATCH] ASoC: rt711-sdca: add the notification when volume changed
by shumingf＠realtek.com 22 Apr '21

22 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 | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 3ab9048b4ea3..98b26bd69e19 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 +564,9 @@ 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; + for (i = 0; i < 3; i++) { /* retry 3 times at most */ /* Lch*/ regmap_write(rt711->mbq_regmap, mc->reg, gain_l_val); @@ -573,7 +580,7 @@ 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

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