March 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: davinci-evm: Register machine level DAPM elements with the card
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

Machine level DAPM widgets and routes should be registered in the card's DAPM context, rather than in the CODEC's context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/davinci/davinci-evm.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 5e3bc3c..8d10244 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -95,23 +95,24 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Logic for a aic3x as connected on a davinci-evm */ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_card *card = rtd->card; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct device_node *np = codec->card->dev->of_node; int ret; /* Add davinci-evm specific widgets */ - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, + snd_soc_dapm_new_controls(&card->dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); if (np) { - ret = snd_soc_of_parse_audio_routing(codec->card, - "ti,audio-routing"); + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); if (ret) return ret; } else { /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(&card->dapm, audio_map, + ARRAY_SIZE(audio_map)); } /* not connected */ @@ -120,10 +121,10 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_disable_pin(dapm, "HPRCOM"); /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Line Out"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Line In"); + snd_soc_dapm_enable_pin(&card->dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin(&card->dapm, "Line Out"); + snd_soc_dapm_enable_pin(&card->dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(&card->dapm, "Line In"); return 0; } -- 1.8.0

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[alsa-devel] [PATCH] ASoC: migor: Convert to table based DAPM setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

Use table based setup to register the DAPM widgets and routes. This on one hand makes the code a bit shorter and cleaner and on the other hand the board level DAPM elements get registered in the card's DAPM context rather than in the CODEC's DAPM context. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/sh/migor.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 5014a88..c58c252 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -136,19 +136,6 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Mic Bias", NULL, "External Microphone" }, }; -static int migor_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, migor_dapm_widgets, - ARRAY_SIZE(migor_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* migor digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link migor_dai = { .name = "wm8978", @@ -158,7 +145,6 @@ static struct snd_soc_dai_link migor_dai = { .platform_name = "siu-pcm-audio", .codec_name = "wm8978.0-001a", .ops = &migor_dai_ops, - .init = migor_dai_init, }; /* migor audio machine driver */ @@ -167,6 +153,11 @@ static struct snd_soc_card snd_soc_migor = { .owner = THIS_MODULE, .dai_link = &migor_dai, .num_links = 1, + + .dapm_widgets = migor_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(migor_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *migor_snd_device; -- 1.8.0

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[alsa-devel] integration into ASoC
by Maxime Ripard 10 Mar '14

10 Mar '14

Hi Liam, Mark, I have a sound IP that is part of an SoC that I'm willing to write a driver for. This IP is made of a few registers to control the sampling rate, if we're using mono/stereo, plus two fifos, one for playback, one for capture, that can be seed with data. The data are then taken, go through a DAC, and the outer interface of the IP are directly analog signals (so the DAC/ADC are directly in the SoC, and the only interface we have is plain registers). From what I understood from ASoC, you have mostly three components, the DAI, the codec and the platform that plumbers the two first together. Here, my understanding is that it's pretty much the whole three in a single IP. Should such a hardware block be handled into ASoC, and if yes, how? If not, which other framework should be used? Thanks! Maxime -- Maxime Ripard, Free Electrons Embedded Linux, Kernel and Android engineering http://free-electrons.com

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[alsa-devel] [PATCH] add Steinberg UR44 USB Audio Interface support to quirks-table.
by Cyrus Harmon 10 Mar '14

10 Mar '14

* The patch is based on Clemens Ladisch's patch to support the UR22 (http://linuxaudio.org/mailarchive/lad/2013/2/23/197831), with the device ID modified for the UR44. The patch was tested with kernel versions 3.12.11 and 3.12.12 and a PREEMPT RT-patched 3.12.12-rt kernel. Basic recording and playback appear to work. * NB: On one of the systems that I used to test the patch, I was never able to achieve xrun-free JACK performance at -p256 -n3 as I would invariably see xruns roughly once every minute. But this is a new box (Intel haswell NUC) and it seems there are many possible sources of the unacceptable latency. On an older i3 box, the performance was much better. Signed-off-by: Cyrus Harmon <cyrus(a)bobobeach.com> --- sound/usb/quirks-table.h | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f652b10..987ca01 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -436,6 +436,36 @@ YAMAHA_DEVICE(0x105d, NULL), } } }, +{ + USB_DEVICE(0x0499, 0x150e), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Yamaha", */ + /* .product_name = "Steinberg UR44", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 3, + .type = QUIRK_MIDI_YAMAHA + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, YAMAHA_DEVICE(0x2000, "DGP-7"), YAMAHA_DEVICE(0x2001, "DGP-5"), YAMAHA_DEVICE(0x2002, NULL), -- 1.8.3.2

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[alsa-devel] [PATCHv3 0/3] Simplify the CODEC ASoC probe code.
by Xiubo Li 10 Mar '14

10 Mar '14

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[alsa-devel] [PATCH 0/9] Update FLL calculations on Arizona devices
by Charles Keepax 10 Mar '14

10 Mar '14

Newer versions of the Arizona IP have several differences in the FLL configuration. The reference and synchroniser paths have slightly difference FRATIO encodings and the expanded encoding on the reference path should be used to avoid integer mode operation. This series of patches does some refactoring on the FLL configuration code, previously calculation of the FLL configuration was done before we knew whether this would be applied to reference or synchroniser path. These patches add some checks to make sure the requested clock can be accomodated and then does the calculations once we knew where they will be applied. Additionally, we add support for the encodings on the newer IP. Thanks, Charles Charles Keepax (9): ASoC: arizona: An OUTDIV of 1 is not valid, avoid this ASoC: arizona: Add defines for FLL configuration constants ASoC: arizona: Move set of OUTDIV in to arizona_apply_fll ASoC: arizona: Move calculation of FLL configuration ASoC: arizona: Don't pass Fout into arizona_calc_fll ASoC: arizona: Calculate OUTDIV first ASoC: arizona: Calculate FLL gain last mfd: arizona: Add support for new fratio encoding ASoC: arizona: Support new fratio encoding on the wm5110 rev D include/linux/mfd/arizona/registers.h | 6 +- sound/soc/codecs/arizona.c | 254 ++++++++++++++++++++++----------- 2 files changed, 171 insertions(+), 89 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH v2 1/3] ASoC: Consolidate snd_soc_register_dai() and snd_soc_register_dais()
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

snd_soc_register_dais() has basically the same code as snd_soc_register_dai(), but running in a loop. The only difference is that snd_soc_register_dai() calls fmt_single_name() to generate the DAIs name and snd_soc_register_dais() calls fmt_multiple_name(). This patch pushes the check in __snd_soc_register_component() which decides whether to call snd_soc_register_dai() or snd_soc_register_dais() to snd_soc_register_dais() to decide which naming scheme to use. This allows us to remove snd_soc_register_dai(). The patch also updates snd_soc_register_dais() to unregister every DAI it finds for the component rather than trying to unregister one DAI for each DAI that was registered. Both have the same result since there won't be more DAIs than what have been registered, but the former is easier to implement. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Fixed compile error that slipped into patch 1, but was fixed in patch 3 --- sound/soc/soc-core.c | 154 ++++++++++++++------------------------------------- 1 file changed, 42 insertions(+), 112 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8ddb15c..ae2b7cc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3897,100 +3897,47 @@ static inline char *fmt_multiple_name(struct device *dev, } /** - * snd_soc_register_dai - Register a DAI with the ASoC core + * snd_soc_unregister_dai - Unregister DAIs from the ASoC core * - * @component: The component the DAIs are registered for - * @dai_drv: DAI driver to use for the DAIs + * @component: The component for which the DAIs should be unregistered */ -static int snd_soc_register_dai(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv) +static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct device *dev = component->dev; - struct snd_soc_codec *codec; - struct snd_soc_dai *dai; - - dev_dbg(dev, "ASoC: dai register %s\n", dev_name(dev)); - - dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); - if (dai == NULL) - return -ENOMEM; - - /* create DAI component name */ - dai->name = fmt_single_name(dev, &dai->id); - if (dai->name == NULL) { - kfree(dai); - return -ENOMEM; - } - - dai->component = component; - dai->dev = dev; - dai->driver = dai_drv; - dai->dapm.dev = dev; - if (!dai->driver->ops) - dai->driver->ops = &null_dai_ops; + struct snd_soc_dai *dai, *_dai; mutex_lock(&client_mutex); + list_for_each_entry_safe(dai, _dai, &dai_list, list) { + if (dai->dev != component->dev) + continue; - list_for_each_entry(codec, &codec_list, list) { - if (codec->dev == dev) { - dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", - dai->name, codec->name); - dai->codec = codec; - break; - } - } - - if (!dai->codec) - dai->dapm.idle_bias_off = 1; - - list_add(&dai->list, &dai_list); - - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); - - return 0; -} - -/** - * snd_soc_unregister_dai - Unregister a DAI from the ASoC core - * - * @dai: DAI to unregister - */ -static void snd_soc_unregister_dai(struct device *dev) -{ - struct snd_soc_dai *dai; + list_del(&dai->list); - list_for_each_entry(dai, &dai_list, list) { - if (dev == dai->dev) - goto found; + dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n", + dai->name); + kfree(dai->name); + kfree(dai); } - return; - -found: - mutex_lock(&client_mutex); - list_del(&dai->list); mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Unregistered DAI '%s'\n", dai->name); - kfree(dai->name); - kfree(dai); } /** - * snd_soc_register_dais - Register multiple DAIs with the ASoC core + * snd_soc_register_dais - Register a DAI with the ASoC core * * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs + * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the + * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count) + struct snd_soc_dai_driver *dai_drv, size_t count, + bool legacy_dai_naming) { struct device *dev = component->dev; struct snd_soc_codec *codec; struct snd_soc_dai *dai; - int i, ret = 0; + unsigned int i; + int ret; dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count); @@ -4002,21 +3949,32 @@ static int snd_soc_register_dais(struct snd_soc_component *component, goto err; } - /* create DAI component name */ - dai->name = fmt_multiple_name(dev, &dai_drv[i]); + /* + * Back in the old days when we still had component-less DAIs, + * instead of having a static name, component-less DAIs would + * inherit the name of the parent device so it is possible to + * register multiple instances of the DAI. We still need to keep + * the same naming style even though those DAIs are not + * component-less anymore. + */ + if (count == 1 && legacy_dai_naming) { + dai->name = fmt_single_name(dev, &dai->id); + } else { + dai->name = fmt_multiple_name(dev, &dai_drv[i]); + if (dai_drv[i].id) + dai->id = dai_drv[i].id; + else + dai->id = i; + } if (dai->name == NULL) { kfree(dai); - ret = -EINVAL; + ret = -ENOMEM; goto err; } dai->component = component; dai->dev = dev; dai->driver = &dai_drv[i]; - if (dai->driver->id) - dai->id = dai->driver->id; - else - dai->id = i; dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; @@ -4025,8 +3983,7 @@ static int snd_soc_register_dais(struct snd_soc_component *component, list_for_each_entry(codec, &codec_list, list) { if (codec->dev == dev) { - dev_dbg(dev, - "ASoC: Mapped DAI %s to CODEC %s\n", + dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", dai->name, codec->name); dai->codec = codec; break; @@ -4040,33 +3997,18 @@ static int snd_soc_register_dais(struct snd_soc_component *component, mutex_unlock(&client_mutex); - dev_dbg(dai->dev, "ASoC: Registered DAI '%s'\n", dai->name); + dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); } return 0; err: - for (i--; i >= 0; i--) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(component); return ret; } /** - * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core - * - * @dai: Array of DAIs to unregister - * @count: Number of DAIs - */ -static void snd_soc_unregister_dais(struct device *dev, size_t count) -{ - int i; - - for (i = 0; i < count; i++) - snd_soc_unregister_dai(dev); -} - -/** * snd_soc_register_component - Register a component with the ASoC core * */ @@ -4097,19 +4039,7 @@ __snd_soc_register_component(struct device *dev, cmpnt->dai_drv = dai_drv; cmpnt->num_dai = num_dai; - /* - * snd_soc_register_dai() uses fmt_single_name(), and - * snd_soc_register_dais() uses fmt_multiple_name() - * for dai->name which is used for name based matching - * - * this function is used from cpu/codec. - * allow_single_dai flag can ignore "codec" driver reworking - * since it had been used snd_soc_register_dais(), - */ - if ((1 == num_dai) && allow_single_dai) - ret = snd_soc_register_dai(cmpnt, dai_drv); - else - ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai); + ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai, allow_single_dai); if (ret < 0) { dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); goto error_component_name; @@ -4164,7 +4094,7 @@ void snd_soc_unregister_component(struct device *dev) return; found: - snd_soc_unregister_dais(dev, cmpnt->num_dai); + snd_soc_unregister_dais(cmpnt); mutex_lock(&client_mutex); list_del(&cmpnt->list); -- 1.8.0

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[alsa-devel] [PATCH 1/3] ASoC: omap-abe-twl6040: No need to register DMIC routes seperatly
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

When using table based DAPM setup there is no need to register DAPM elements for different sub-components separately. The widgets will be registered before the first sub-component is initialized, the routes are only added after the last sub-component has been initialized, meaning everything will be available when it is needed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/omap-abe-twl6040.c | 19 ++++--------------- 1 file changed, 4 insertions(+), 15 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index ebb1390..5011bfa 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -163,6 +163,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFML", NULL, "Line In"}, {"AFMR", NULL, "Line In"}, + + /* DMIC routing */ + {"DMic", NULL, "Digital Mic"}, + {"Digital Mic", NULL, "Digital Mic1 Bias"}, }; static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) @@ -196,20 +200,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static const struct snd_soc_dapm_route dmic_audio_map[] = { - {"DMic", NULL, "Digital Mic"}, - {"Digital Mic", NULL, "Digital Mic1 Bias"}, -}; - -static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - return snd_soc_dapm_add_routes(dapm, dmic_audio_map, - ARRAY_SIZE(dmic_audio_map)); -} - /* Digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link abe_twl6040_dai_links[] = { { @@ -229,7 +219,6 @@ static struct snd_soc_dai_link abe_twl6040_dai_links[] = { .codec_dai_name = "dmic-hifi", .platform_name = "omap-pcm-audio", .codec_name = "dmic-codec", - .init = omap_abe_dmic_init, .ops = &omap_abe_dmic_ops, }, }; -- 1.8.0

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[alsa-devel] [PATCH v2 0/4] AM43xx-ePOS-EVM audio support with TLV320AIC31XX driver
by Jyri Sarha 10 Mar '14

10 Mar '14

Since the RFC version of the patches [1] I have addressed Mark Browns comments [2] as described in my mail here [3]. Besr regards, Jyri [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/073289.h… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073703.html http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073704.html http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073705.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073807.html Jyri Sarha (4): ASoC: tlv320aic31xx: Add basic codec driver implementation ASoC: davinci-evm: Add AM43xx-EPOS-EVM audio support ASoC: davinci: Add SND_AM43XX_SOC_EPOS_EVM build option ASoC: tlv320aic32x4: Sort Makefile in alphabetic order .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/tlv320aic31xx.txt | 58 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 9 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 6 +- sound/soc/codecs/tlv320aic31xx.c | 1345 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ sound/soc/davinci/Kconfig | 12 + sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 41 + 10 files changed, 1738 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: 88pm860: Fix IO setup
by Lars-Peter Clausen 10 Mar '14

10 Mar '14

The 88pm860 is a MFD device and the CODEC driver is using the regmap struct of the parent device, hence automatic IO setup will not work and we need to manually call snd_soc_codec_set_cache_io(). The issue was introduced in commit f9ded3b2e7 ("ASoC: 88pm860x: Use regmap for I/O"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/88pm860x-codec.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 697d24b..8703244 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1328,6 +1328,9 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; codec->control_data = pm860x->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret) + return ret; for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, -- 1.8.0

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