August 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rt5665: fix wrong register for bclk ratio control
by Bard Liao 01 Aug '17

01 Aug '17

The register of setting back ratio should be RT5665_ADDA_CLK_2 instead of RT5665_ADDA_CLK_1. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5665.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index d0a9490..db2e5b9 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4411,12 +4411,12 @@ static int rt5665_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) switch (dai->id) { case RT5665_AIF2_1: case RT5665_AIF2_2: - snd_soc_update_bits(codec, RT5665_ADDA_CLK_1, + snd_soc_update_bits(codec, RT5665_ADDA_CLK_2, RT5665_I2S_BCLK_MS2_MASK, RT5665_I2S_BCLK_MS2_64); break; case RT5665_AIF3: - snd_soc_update_bits(codec, RT5665_ADDA_CLK_1, + snd_soc_update_bits(codec, RT5665_ADDA_CLK_2, RT5665_I2S_BCLK_MS3_MASK, RT5665_I2S_BCLK_MS3_64); break; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt5663: Add the delay time to correct the calibration
by Oder Chiou 01 Aug '17

01 Aug '17

The patch adds the delay time to correct the calibration. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 9704eed..7bbfb18 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3058,6 +3058,7 @@ static void rt5663_calibrate(struct rt5663_priv *rt5663) regmap_write(rt5663->regmap, RT5663_PWR_ANLG_3, 0x018c); regmap_write(rt5663->regmap, RT5663_HP_CHARGE_PUMP_1, 0x1e32); regmap_write(rt5663->regmap, RT5663_DACREF_LDO, 0x3b0b); + msleep(40); regmap_write(rt5663->regmap, RT5663_STO_DAC_MIXER, 0x0000); regmap_write(rt5663->regmap, RT5663_BYPASS_STO_DAC, 0x000c); regmap_write(rt5663->regmap, RT5663_HP_BIAS, 0xafaa); -- 2.7.4

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[alsa-devel] [PATCH 1/4] ASoC: Intel: Skylake: Reset the controller in probe
by Guneshwor Singh 01 Aug '17

01 Aug '17

Controller can be in reset state by default. Capability structure traversal requires the controller to be out of reset else it results in broken capability parsing. Hence make sure that controller is out of reset before parsing capabilities by doing a full reset. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 334917ee41cf..a56dfde3928c 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -702,6 +702,8 @@ static int skl_first_init(struct hdac_ext_bus *ebus) return -ENXIO; } + skl_init_chip(bus, true); + snd_hdac_bus_parse_capabilities(bus); if (skl_acquire_irq(ebus, 0) < 0) -- 2.13.0

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[alsa-devel] [PATCH 1/2] ASoC: rt5514: Add the sanity checks of the buffer related address
by Oder Chiou 01 Aug '17

01 Aug '17

The patch add the sanity checks of the buffer related address to make sure the addresses are valid. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 950d1ff..6f513cc 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -173,9 +173,14 @@ static irqreturn_t rt5514_spi_irq(int irq, void *data) rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | buf[3] << 24; + if (rt5514_dsp->buf_rp % 8) + rt5514_dsp->buf_rp = (rt5514_dsp->buf_rp / 8) * 8; + rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; - schedule_delayed_work(&rt5514_dsp->copy_work, 0); + if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && + rt5514_dsp->buf_rp && rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 0); return IRQ_HANDLED; } -- 2.7.4

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[alsa-devel] [PATCH] device property: Fix usecount for of_graph_get_port_parent()
by Tony Lindgren 01 Aug '17

01 Aug '17

Fix inconsistent use of of_graph_get_port_parent() where asoc_simple_card_parse_graph_dai() does of_node_get() before calling it while other callers do not. We can fix this by not trashing the node passed to of_graph_get_port_parent(). Let's make sure the users have correct refcounts and remove related incorrect of_node_put() calls for of_for_each_phandle as that's done by of_phandle_iterator_next(). Let's fix both issues with a single patch to avoid kobject refcounts getting messed up more if two patches are merged separately. Otherwise strange issues can happen caused by memory corruption caused by too many kobject_del() calls such as: BUG: sleeping function called from invalid context at kernel/locking/mutex.c:747 ... (___might_sleep) (__mutex_lock) (mutex_lock_nested) (kernfs_remove) (kobject_del) (kobject_put) (of_get_next_parent) (of_graph_get_port_parent) (asoc_simple_card_parse_graph_dai [snd_soc_simple_card_utils]) (asoc_graph_card_probe [snd_soc_audio_graph_card]) Fixes: 0ef472a973eb ("of_graph: add of_graph_get_port_parent()") Fixes: 2692c1c63c29 ("ASoC: add audio-graph-card support") Fixes: 1689333f8311 ("ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai()") Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- drivers/of/property.c | 9 +++++++++ sound/soc/generic/audio-graph-card.c | 5 +---- sound/soc/generic/simple-card-utils.c | 5 ----- 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/drivers/of/property.c b/drivers/of/property.c --- a/drivers/of/property.c +++ b/drivers/of/property.c @@ -708,6 +708,15 @@ struct device_node *of_graph_get_port_parent(struct device_node *node) { unsigned int depth; + if (!node) + return NULL; + + /* + * Preserve usecount for passed in node as of_get_next_parent() + * will do of_node_put() on it. + */ + of_node_get(node); + /* Walk 3 levels up only if there is 'ports' node. */ for (depth = 3; depth && node; depth--) { node = of_get_next_parent(node); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -224,7 +224,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); - of_node_put(it.node); if (ret < 0) return ret; } @@ -239,10 +238,8 @@ static int asoc_graph_get_dais_count(struct device *dev) int count = 0; int rc; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) count++; - of_node_put(it.node); - } return count; } diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -282,11 +282,6 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, if (!dai_name) return 0; - /* - * of_graph_get_port_parent() will call - * of_node_put(). So, call of_node_get() here - */ - of_node_get(ep); node = of_graph_get_port_parent(ep); /* Get dai->name */ -- 2.13.2

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[alsa-devel] Question about soc_bind_dai_link()
by Kuninori Morimoto 01 Aug '17

01 Aug '17

Hi ALSA ML I noticed a little bit strange operation on soc_bind_dai_link(). To research platform, it is using list_for_each_entry(). After finding platform, it sets rtd->platform, but it doesn't use "break". This means, rtd->platform might be overwrote. Is this on purpose ? or bug ? static int soc_bind_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { ... /* find one from the set of registered platforms */ list_for_each_entry(platform, &platform_list, list) { platform_of_node = platform->dev->of_node; if (!platform_of_node && platform->dev->parent->of_node) platform_of_node = platform->dev->parent->of_node; if (dai_link->platform_of_node) { if (platform_of_node != dai_link->platform_of_node) continue; } else { if (strcmp(platform->component.name, platform_name)) continue; } => rtd->platform = platform; } ... } Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: max98090 - Expose HP out enable.
by Enric Balletbo i Serra 01 Aug '17

01 Aug '17

From: Hsin-Yu Chao <hychao(a)chromium.org> Expose the headphone output enables to user space. This will allow userspace to ensure that there is stable output before the headphones are turned on, avoiding a pop on insertion. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- sound/soc/codecs/max98090.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6682848..f38c051 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1074,6 +1074,15 @@ static SOC_ENUM_SINGLE_DECL(mixhprsel_mux_enum, static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); +/* HP output enables. */ +static const struct snd_kcontrol_new max98090_hpl_enable = + SOC_DAPM_SINGLE("Switch", M98090_REG_OUTPUT_ENABLE, + M98090_HPLEN_SHIFT, 1, 0); + +static const struct snd_kcontrol_new max98090_hpr_enable = + SOC_DAPM_SINGLE("Switch", M98090_REG_OUTPUT_ENABLE, + M98090_HPREN_SHIFT, 1, 0); + static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), @@ -1218,10 +1227,10 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), - SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, - M98090_HPLEN_SHIFT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_OUTPUT_ENABLE, - M98090_HPREN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("HP Left Out", SND_SOC_NOPM, 0, 0, + &max98090_hpl_enable), + SND_SOC_DAPM_SWITCH("HP Right Out", SND_SOC_NOPM, 0, 0, + &max98090_hpr_enable), SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_OUTPUT_ENABLE, M98090_SPLEN_SHIFT, 0, NULL, 0), @@ -1405,8 +1414,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { * Disable this for lowest power if bypassing * the DAC with an analog signal */ - {"HP Left Out", NULL, "DACL"}, - {"HP Left Out", NULL, "MIXHPLSEL Mux"}, + {"HP Left Out", "Switch", "DACL"}, + {"HP Left Out", "Switch", "MIXHPLSEL Mux"}, {"MIXHPRSEL Mux", "HP Mixer", "Right Headphone Mixer"}, @@ -1414,8 +1423,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { * Disable this for lowest power if bypassing * the DAC with an analog signal */ - {"HP Right Out", NULL, "DACR"}, - {"HP Right Out", NULL, "MIXHPRSEL Mux"}, + {"HP Right Out", "Switch", "DACR"}, + {"HP Right Out", "Switch", "MIXHPRSEL Mux"}, {"SPK Left Out", NULL, "Left Speaker Mixer"}, {"SPK Right Out", NULL, "Right Speaker Mixer"}, -- 2.9.3

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[alsa-devel] [PATCH] Set sample rate for Digidesign Mbox 2
by Viktor Radnai 01 Aug '17

01 Aug '17

Hi all, I'm re-submitting this as I didn't get a response on it for about two weeks. Please see the old thread called "Patch for Digidesign Mbox 2 (set sample rate)" for prior discussion on it. Let me know if you need anything else from me to get this applied. Cheers, Vik diff -ur linux-source-4.11-orig/sound/usb/quirks.c linux-source-4.11/sound/usb/quirks.c --- linux-source-4.11-orig/sound/usb/quirks.c 2017-06-17 05:47:27.000000000 +0100 +++ linux-source-4.11/sound/usb/quirks.c 2017-07-18 22:37:58.522858262 +0100 @@ -768,23 +768,32 @@ static void mbox2_setup_48_24_magic(struct usb_device *dev) { - u8 srate[3]; - u8 temp[12]; - /* Choose 48000Hz permanently */ - srate[0] = 0x80; - srate[1] = 0xbb; - srate[2] = 0x00; + u8 srate[3] = { 0x80, 0xbb, 0x00 }; + u8 temp[3]; + int err; /* Send the magic! */ - snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), - 0x01, 0x22, 0x0100, 0x0085, &temp, 0x0003); - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003); - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003); - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003); + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0085, &temp, sizeof(temp))) < 0) + dev_err(&dev->dev, "mbox2: magic setup step #1 failed\n"); + + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0085, &srate, sizeof(srate))) < 0) + dev_err(&dev->dev, "mbox2: magic setup step #2 failed\n"); + + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0086, &srate, sizeof(srate))) < 0) + dev_err(&dev->dev, "mbox2: magic setup step #3 failed\n"); + + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0003, &srate, sizeof(srate))) < 0) + dev_err(&dev->dev, "mbox2: magic setup step #4 failed\n"); + return; } @@ -815,16 +824,16 @@ count = 0; bootresponse[0] = MBOX2_BOOT_LOADING; - while ((bootresponse[0] == MBOX2_BOOT_LOADING) && (count < 10)) { - msleep(500); /* 0.5 second delay */ + do { snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), /* Control magic - load onboard firmware */ - 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012); + UAC_GET_MEM, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x0001, 0x0000, &bootresponse, sizeof(bootresponse)); if (bootresponse[0] == MBOX2_BOOT_READY) break; + msleep(500); /* 0.5 second delay */ dev_dbg(&dev->dev, "device not ready, resending boot sequence...\n"); count++; - } + } while ((bootresponse[0] == MBOX2_BOOT_LOADING) && (count < 10)); if (bootresponse[0] != MBOX2_BOOT_READY) { dev_err(&dev->dev, "Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse[0]); @@ -847,7 +856,7 @@ mbox2_setup_48_24_magic(dev); - dev_info(&dev->dev, "Digidesign Mbox 2: 24bit 48kHz"); + dev_info(&dev->dev, "Digidesign Mbox 2 configured"); return 0; /* Successful boot */ } @@ -1111,6 +1120,26 @@ subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } +static void set_format_mbox2_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt) +{ + struct usb_device *dev = subs->dev; + u8 srate[3]; + int err; + + srate[0] = subs->cur_rate; + srate[1] = subs->cur_rate >> 8; + srate[2] = subs->cur_rate >> 16; + + /* Set the sample rate for endpoint IN 5 (0x0085) until we succeed */ + while ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, 0x0085, srate, sizeof(srate))) < 0) + dev_err(&dev->dev, "mbox2: set sample rate failed, retrying\n"); + + dev_info(&subs->dev->dev, "mbox2: set sample rate to %dHz\n", subs->cur_rate); +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt) { @@ -1121,6 +1150,9 @@ case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x0dba, 0x3000): /* Digidesign Mbox 2 */ + set_format_mbox2_quirk(subs, fmt); + break; } } diff -ur linux-source-4.11-orig/sound/usb/quirks-table.h linux-source-4.11/sound/usb/quirks-table.h --- linux-source-4.11-orig/sound/usb/quirks-table.h 2017-06-17 05:47:27.000000000 +0100 +++ linux-source-4.11/sound/usb/quirks-table.h 2017-07-18 17:33:57.620424012 +0100 @@ -3020,12 +3020,13 @@ .attributes = 0x00, .endpoint = 0x03, .ep_attr = USB_ENDPOINT_SYNC_ASYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000, + .rate_min = 44100, .rate_max = 48000, - .nr_rates = 1, + .nr_rates = 2, .rate_table = (unsigned int[]) { - 48000 + 44100, 48000 } } }, @@ -3045,12 +3046,13 @@ .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x85, .ep_attr = USB_ENDPOINT_SYNC_SYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000, + .rate_min = 44100, .rate_max = 48000, - .nr_rates = 1, + .nr_rates = 2, .rate_table = (unsigned int[]) { - 48000 + 44100, 48000 } } },

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