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

Re: [alsa-devel] [pulseaudio-discuss] PulseAudio miniconference 2014, Oct 15, Düsseldorf, Germany
by Arun Raghavan 11 Sep '14

11 Sep '14

Looping in the ALSA and GStreamer lists in case someone is interested. On 9 September 2014 16:52, Tanu Kaskinen <tanu.kaskinen(a)linux.intel.com> wrote: > Hi, > > In addition to the "Linux Audio Mini Summit" that Takashi announced recently > (which focuses more on ALSA topics), we will also have a "PulseAudio > miniconference" in the same conference center, day after the ALSA event. The > PulseAudio event will be on Wednesday, starting at 14:00. Registering isn't > mandatory, but if we run out of seats, those who were fastest to register > will get preference. > > More details are in the wiki: > http://www.freedesktop.org/wiki/Software/PulseAudio/Events/2014_Dusseldorf/ > > We don't yet have discussion topics for the miniconference, so if you plan > to attend, topic ideas would be very welcome! Send them to the list, and > I'll collect them to the wiki. > > If you have any questions or comments, feel free to send them to the list or > privately to me. > > See you in Düsseldorf! > > -- > Tanu > _______________________________________________ > pulseaudio-discuss mailing list > pulseaudio-discuss(a)lists.freedesktop.org > http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss

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[alsa-devel] [PATCH 0/3] ASoC/ARM: tegra: apalis/colibri t30: sgtl5000 audio
by Marcel Ziswiler 10 Sep '14

10 Sep '14

This series adds/integrates Freescale SGTL5000 analogue audio codec support. Marcel Ziswiler (3): ASoC: tegra: add tegra sgtl5000 machine driver ARM: tegra: apalis/colibri t30: integrate audio support ARM: tegra: enable sgtl5000 audio .../bindings/sound/nvidia,tegra-audio-sgtl5000.txt | 45 +++++ arch/arm/boot/dts/tegra30-apalis.dtsi | 49 +++++ arch/arm/boot/dts/tegra30-colibri.dtsi | 49 +++++ arch/arm/configs/tegra_defconfig | 1 + sound/soc/tegra/Kconfig | 10 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_sgtl5000.c | 216 +++++++++++++++++++++ 7 files changed, 372 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt create mode 100644 sound/soc/tegra/tegra_sgtl5000.c -- 1.9.3

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[alsa-devel] [PATCH RESEND] Documentation: fsl, ssi.txt: Avoid cryptic acronyms
by Fabio Estevam 10 Sep '14

10 Sep '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> It is much clearer to use the full terms here: left-justified/right-justified. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Documentation/devicetree/bindings/sound/fsl,ssi.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl,ssi.txt b/Documentation/devicetree/bindings/sound/fsl,ssi.txt index 3aa4a8f..ceca880 100644 --- a/Documentation/devicetree/bindings/sound/fsl,ssi.txt +++ b/Documentation/devicetree/bindings/sound/fsl,ssi.txt @@ -62,9 +62,9 @@ Optional properties: "i2s-slave" - I2S mode, SSI is clock slave "i2s-master" - I2S mode, SSI is clock master "lj-slave" - left-justified mode, SSI is clock slave - "lj-master" - l.j. mode, SSI is clock master + "lj-master" - left-justified mode, SSI is clock master "rj-slave" - right-justified mode, SSI is clock slave - "rj-master" - r.j., SSI is clock master + "rj-master" - right-justified mode, SSI is clock master "ac97-slave" - AC97 mode, SSI is clock slave "ac97-master" - AC97 mode, SSI is clock master -- 1.9.1

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[alsa-devel] EP93XX / EDB9315A Audio Issues
by Jeremy Moles 10 Sep '14

10 Sep '14

(I'm cross-posting this to the linux-arm-kernel and alsa-devel lists, as recommended): I'm looking for anyone who has successfully used the EP93XX/EDB9315A demo board's sound infrastructure. I have been working with Hartley Sweeten on patching some DMA coherency mask issues (successfully), and while this makes the kernel stop complaining (and allows the userspace to "appear" to work) I never actually get any sound out of the machine on newer kernels. I do, however, have an existing (binary-only) 2.6.21 kernel from the client where sound _IS_ functional; thus, I know the system can POTENTIALLY work, and that the problem is software-based. At this point, I have a mountain of information I can provide about the state of my EDB9315A board, but what I'm really looking for is someone to connect with and/or have consult with us to get sound working on newer kernels with this board exactly (although other EP93XX boards might be similar, I'm so mistrustful of everything at this point I'm looking for someone else with this hardware EXACTLY). I've tried 3.15-3.17, all with the same results. aplay/alsamixer see the device, appear to interact with it in a normal fashion (it even mutes/unmutes the device every time you touch it), but no sound is ever produced. It is possible, of course, the problem is in userspace (I'm using a custom Yocto-based image), but I'm doubtful of that. At any rate, any help would be greatly appreciate, although i realize this is fringe hardware. :)

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Re: [alsa-devel] [PATCH] ASoC:simple-card: Add multi-CODEC support
by Benoit Cousson 10 Sep '14

10 Sep '14

Hi Jean-Francois, On 10/09/2014 13:28, Jean-Francois Moine wrote: > This patch adds multi-CODEC support to the simple-card. > > Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> > --- > .../devicetree/bindings/sound/simple-card.txt | 28 +++++++-------- > sound/soc/generic/simple-card.c | 41 +++++++++++++++++++++- > 2 files changed, 52 insertions(+), 17 deletions(-) > > diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt > index c2e9841..c217687 100644 > --- a/Documentation/devicetree/bindings/sound/simple-card.txt > +++ b/Documentation/devicetree/bindings/sound/simple-card.txt > @@ -37,6 +37,8 @@ Required dai-link subnodes: > > - cpu : CPU sub-node > - codec : CODEC sub-node > + In case of multi-CODECs, there may > + be many of such sub-nodes. > > Optional dai-link subnode properties: > > @@ -115,37 +117,31 @@ sh_fsi2: sh_fsi2@ec230000 { > interrupts = <0 146 0x4>; > }; > > -Example 2 - many DAI links: > +Example 2 - many DAI links and multi-CODECs: > > sound { > compatible = "simple-audio-card"; > simple-audio-card,name = "Cubox Audio"; > > - simple-audio-card,dai-link@0 { /* I2S - HDMI */ > + simple-audio-card,dai-link@0 { /* S/PDIF - HDMI & S/PDIF */ > format = "i2s"; > cpu { > - sound-dai = <&audio1 0>; > - }; > - codec { > - sound-dai = <&tda998x 0>; > - }; > - }; > - > - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ > - cpu { > sound-dai = <&audio1 1>; > }; > - codec { > - sound-dai = <&tda998x 1>; > + codec@0 { > + sound-dai = <&hdmi 0>; > + }; > + codec@1 { > + sound-dai = <&spdif_codec>; > }; I don't have strong opinion on that, but in my case, I was considering using a simple list instead of several nodes. I don't like having to add fake address just to ensure uniqueness. Something like that: sound-dais = <&spdif_codec 1>, <&hdmi 0>; That being said, it will require changing the name with a plural form, and ensuring we have the same number of parameters for each codec. That was just my .2 cents. Regards, Benoit > }; > > - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ > + simple-audio-card,dai-link@1 { /* I2S - HDMI */ > cpu { > - sound-dai = <&audio1 1>; > + sound-dai = <&audio1 0>; > }; > codec { > - sound-dai = <&spdif_codec>; > + sound-dai = <&hdmi 1>; > }; > }; > }; > diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c > index 4053152..bf0ce08 100644 > --- a/sound/soc/generic/simple-card.c > +++ b/sound/soc/generic/simple-card.c > @@ -179,11 +179,12 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, > struct device_node *np = NULL; > struct device_node *bitclkmaster = NULL; > struct device_node *framemaster = NULL; > + struct snd_soc_dai_link_component *component; > unsigned int daifmt; > char *name; > char prop[128]; > char *prefix = ""; > - int ret, cpu_args; > + int ret, cpu_args, num_codec_dais; > > /* For single DAI link & old style of DT node */ > if (is_top_level_node) > @@ -225,7 +226,16 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, > } > > of_node_put(np); > + > + /* count the number of codec DAIs */ > snprintf(prop, sizeof(prop), "%scodec", prefix); > + num_codec_dais = 0; > + for_each_child_of_node(node, np) { > + if (strcmp(np->name, prop) == 0) > + num_codec_dais++; > + } > + > + /* treat the first DAI */ > np = of_get_child_by_name(node, prop); > if (!np) { > ret = -EINVAL; > @@ -307,6 +317,35 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, > if (!cpu_args) > dai_link->cpu_dai_name = NULL; > > + /* handle multi-codec DAIs */ > + if (num_codec_dais == 1) > + goto out; > + dai_link->codecs = component = > + devm_kzalloc(dev, > + sizeof *component * num_codec_dais, > + GFP_KERNEL); > + dai_link->num_codecs = num_codec_dais; > + component->of_node = dai_link->codec_of_node; > + dai_link->codec_of_node = NULL; > + component->dai_name = dai_link->codec_dai_name; > + dai_link->codec_dai_name = NULL; > + for (;;) { > + np = of_get_next_child(node, np); > + if (!np) > + break; > + component++; > + component->of_node = of_parse_phandle(np, "sound-dai", 0); > + if (!component->of_node) { > + ret = -ENODEV; > + dev_err(dev, "Bad sound-dai\n"); > + goto dai_link_of_err; > + } > + ret = snd_soc_of_get_dai_name(np, &component->dai_name); > + if (ret < 0) > + goto dai_link_of_err; > + } > + > +out: > dai_link_of_err: > if (np) > of_node_put(np); > -- Benoît Cousson BayLibre Embedded Linux Technology Lab www.baylibre.com

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[alsa-devel] [PATCH] sound: aoa: printk replacement
by Sudip Mukherjee 10 Sep '14

10 Sep '14

as pr_* macros are more preffered over printk, so printk replaced with corresponding pr_* macros. this patch will generate warning from checkpatch as it only did printk replacement and didnot fixed the other warning of : Alignment should match open parenthesis and Possible unnecessary 'out of memory' message Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- The replacement was done by a bash script to avoid copy paste error. The script is as follows : OLD1="printk(KERN_DEBUG \?" OLD2="printk(KERN_ERR \?" OLD3="printk(KERN_INFO \?" OLD4="printk(KERN_WARNING \?" OLD5="printk(KERN_ALERT \?" NEW1="pr_debug(" NEW2="pr_err(" NEW3="pr_info(" NEW4="pr_warn(" NEW5="pr_alert(" find . -type d -mindepth 0 -maxdepth 2 | while read DIR do for f in "${DIR}"/*.c do sed -i -e "s/$OLD1/$NEW1/g" -e "s/$OLD2/$NEW2/g" -e "s/$OLD3/$NEW3/g" -e "s/$OLD4/$NEW4/g" -e "s/$OLD5/$NEW5/g" "$f" done done Takashi : thanks for the -i option. for the codecs section another future patch can define proper pr_fmt and eliminate the use of PFX while printing. this patch is not build tested and i understand that you can reject the patch without even seeing at it. sound/aoa/codecs/onyx.c | 16 ++++++++-------- sound/aoa/codecs/tas.c | 10 +++++----- sound/aoa/codecs/toonie.c | 6 +++--- sound/aoa/core/alsa.c | 10 +++++----- sound/aoa/core/core.c | 6 +++--- sound/aoa/core/gpio-pmf.c | 6 +++--- sound/aoa/fabrics/layout.c | 16 ++++++++-------- sound/aoa/soundbus/core.c | 2 +- sound/aoa/soundbus/i2sbus/control.c | 4 ++-- sound/aoa/soundbus/i2sbus/core.c | 8 ++++---- sound/aoa/soundbus/i2sbus/pcm.c | 28 ++++++++++++++-------------- 11 files changed, 56 insertions(+), 56 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 401107b..55aa69f 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -873,7 +873,7 @@ static int onyx_init_codec(struct aoa_codec *codec) int err; if (!onyx->codec.gpio || !onyx->codec.gpio->methods) { - printk(KERN_ERR PFX "gpios not assigned!!\n"); + pr_err(PFX "gpios not assigned!!\n"); return -EINVAL; } @@ -885,12 +885,12 @@ static int onyx_init_codec(struct aoa_codec *codec) msleep(1); if (onyx_register_init(onyx)) { - printk(KERN_ERR PFX "failed to initialise onyx registers\n"); + pr_err(PFX "failed to initialise onyx registers\n"); return -ENODEV; } if (aoa_snd_device_new(SNDRV_DEV_CODEC, onyx, &ops)) { - printk(KERN_ERR PFX "failed to create onyx snd device!\n"); + pr_err(PFX "failed to create onyx snd device!\n"); return -ENODEV; } @@ -925,7 +925,7 @@ static int onyx_init_codec(struct aoa_codec *codec) if (onyx->codec.soundbus_dev->attach_codec(onyx->codec.soundbus_dev, aoa_get_card(), ci, onyx)) { - printk(KERN_ERR PFX "error creating onyx pcm\n"); + pr_err(PFX "error creating onyx pcm\n"); return -ENODEV; } #define ADDCTL(n) \ @@ -977,7 +977,7 @@ static int onyx_init_codec(struct aoa_codec *codec) } } #undef ADDCTL - printk(KERN_INFO PFX "attached to onyx codec via i2c\n"); + pr_info(PFX "attached to onyx codec via i2c\n"); return 0; error: @@ -991,7 +991,7 @@ static void onyx_exit_codec(struct aoa_codec *codec) struct onyx *onyx = codec_to_onyx(codec); if (!onyx->codec.soundbus_dev) { - printk(KERN_ERR PFX "onyx_exit_codec called without soundbus_dev!\n"); + pr_err(PFX "onyx_exit_codec called without soundbus_dev!\n"); return; } onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); @@ -1016,7 +1016,7 @@ static int onyx_i2c_probe(struct i2c_client *client, /* we try to read from register ONYX_REG_CONTROL * to check if the codec is present */ if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) { - printk(KERN_ERR PFX "failed to read control register\n"); + pr_err(PFX "failed to read control register\n"); goto fail; } @@ -1029,7 +1029,7 @@ static int onyx_i2c_probe(struct i2c_client *client, if (aoa_codec_register(&onyx->codec)) { goto fail; } - printk(KERN_DEBUG PFX "created and attached onyx instance\n"); + pr_debug(PFX "created and attached onyx instance\n"); return 0; fail: kfree(onyx); diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index cf3c630..abf6e8a 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -806,13 +806,13 @@ static int tas_init_codec(struct aoa_codec *codec) int err; if (!tas->codec.gpio || !tas->codec.gpio->methods) { - printk(KERN_ERR PFX "gpios not assigned!!\n"); + pr_err(PFX "gpios not assigned!!\n"); return -EINVAL; } mutex_lock(&tas->mtx); if (tas_reset_init(tas)) { - printk(KERN_ERR PFX "tas failed to initialise\n"); + pr_err(PFX "tas failed to initialise\n"); mutex_unlock(&tas->mtx); return -ENXIO; } @@ -822,12 +822,12 @@ static int tas_init_codec(struct aoa_codec *codec) if (tas->codec.soundbus_dev->attach_codec(tas->codec.soundbus_dev, aoa_get_card(), &tas_codec_info, tas)) { - printk(KERN_ERR PFX "error attaching tas to soundbus\n"); + pr_err(PFX "error attaching tas to soundbus\n"); return -ENODEV; } if (aoa_snd_device_new(SNDRV_DEV_CODEC, tas, &ops)) { - printk(KERN_ERR PFX "failed to create tas snd device!\n"); + pr_err(PFX "failed to create tas snd device!\n"); return -ENODEV; } err = aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas)); @@ -910,7 +910,7 @@ static int tas_i2c_probe(struct i2c_client *client, if (aoa_codec_register(&tas->codec)) { goto fail; } - printk(KERN_DEBUG + pr_debug( "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n", (unsigned int)client->addr, node->full_name); return 0; diff --git a/sound/aoa/codecs/toonie.c b/sound/aoa/codecs/toonie.c index 7e8c341..f382d7f 100644 --- a/sound/aoa/codecs/toonie.c +++ b/sound/aoa/codecs/toonie.c @@ -93,14 +93,14 @@ static int toonie_init_codec(struct aoa_codec *codec) return -ENOTCONN; if (aoa_snd_device_new(SNDRV_DEV_CODEC, toonie, &ops)) { - printk(KERN_ERR PFX "failed to create toonie snd device!\n"); + pr_err(PFX "failed to create toonie snd device!\n"); return -ENODEV; } if (toonie->codec.soundbus_dev->attach_codec(toonie->codec.soundbus_dev, aoa_get_card(), &toonie_codec_info, toonie)) { - printk(KERN_ERR PFX "error creating toonie pcm\n"); + pr_err(PFX "error creating toonie pcm\n"); snd_device_free(aoa_get_card(), toonie); return -ENODEV; } @@ -113,7 +113,7 @@ static void toonie_exit_codec(struct aoa_codec *codec) struct toonie *toonie = codec_to_toonie(codec); if (!toonie->codec.soundbus_dev) { - printk(KERN_ERR PFX "toonie_exit_codec called without soundbus_dev!\n"); + pr_err(PFX "toonie_exit_codec called without soundbus_dev!\n"); return; } toonie->codec.soundbus_dev->detach_codec(toonie->codec.soundbus_dev, toonie); diff --git a/sound/aoa/core/alsa.c b/sound/aoa/core/alsa.c index 4a7e4e6..65367c3 100644 --- a/sound/aoa/core/alsa.c +++ b/sound/aoa/core/alsa.c @@ -35,7 +35,7 @@ int aoa_alsa_init(char *name, struct module *mod, struct device *dev) strlcpy(alsa_card->mixername, name, sizeof(alsa_card->mixername)); err = snd_card_register(aoa_card->alsa_card); if (err < 0) { - printk(KERN_ERR "snd-aoa: couldn't register alsa card\n"); + pr_err("snd-aoa: couldn't register alsa card\n"); snd_card_free(aoa_card->alsa_card); aoa_card = NULL; return err; @@ -69,14 +69,14 @@ int aoa_snd_device_new(enum snd_device_type type, err = snd_device_new(card, type, device_data, ops); if (err) { - printk(KERN_ERR "snd-aoa: failed to create snd device (%d)\n", err); + pr_err("snd-aoa: failed to create snd device (%d)\n", err); return err; } err = snd_device_register(card, device_data); if (err) { - printk(KERN_ERR "snd-aoa: failed to register " + pr_err("snd-aoa: failed to register " "snd device (%d)\n", err); - printk(KERN_ERR "snd-aoa: have you forgotten the " + pr_err("snd-aoa: have you forgotten the " "dev_register callback?\n"); snd_device_free(card, device_data); } @@ -92,7 +92,7 @@ int aoa_snd_ctl_add(struct snd_kcontrol* control) err = snd_ctl_add(aoa_card->alsa_card, control); if (err) - printk(KERN_ERR "snd-aoa: failed to add alsa control (%d)\n", + pr_err("snd-aoa: failed to add alsa control (%d)\n", err); return err; } diff --git a/sound/aoa/core/core.c b/sound/aoa/core/core.c index 10bec6c..263ea80 100644 --- a/sound/aoa/core/core.c +++ b/sound/aoa/core/core.c @@ -33,7 +33,7 @@ static int attach_codec_to_fabric(struct aoa_codec *c) err = fabric->found_codec(c); if (err) { module_put(c->owner); - printk(KERN_ERR "snd-aoa: fabric didn't like codec %s\n", + pr_err("snd-aoa: fabric didn't like codec %s\n", c->name); return err; } @@ -43,7 +43,7 @@ static int attach_codec_to_fabric(struct aoa_codec *c) if (c->init) err = c->init(c); if (err) { - printk(KERN_ERR "snd-aoa: codec %s didn't init\n", c->name); + pr_err("snd-aoa: codec %s didn't init\n", c->name); c->fabric = NULL; if (fabric->remove_codec) fabric->remove_codec(c); @@ -134,7 +134,7 @@ EXPORT_SYMBOL_GPL(aoa_fabric_unregister); void aoa_fabric_unlink_codec(struct aoa_codec *codec) { if (!codec->fabric) { - printk(KERN_ERR "snd-aoa: fabric unassigned " + pr_err("snd-aoa: fabric unassigned " "in aoa_fabric_unlink_codec\n"); dump_stack(); return; diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c index c8d8a1a..4a34f7b6 100644 --- a/sound/aoa/core/gpio-pmf.c +++ b/sound/aoa/core/gpio-pmf.c @@ -20,7 +20,7 @@ static void pmf_gpio_set_##name(struct gpio_runtime *rt, int on)\ if (unlikely(!rt)) return; \ rc = pmf_call_function(rt->node, #name "-mute", &args); \ if (rc && rc != -ENODEV) \ - printk(KERN_WARNING "pmf_gpio_set_" #name \ + pr_warn("pmf_gpio_set_" #name \ " failed, rc: %d\n", rc); \ rt->implementation_private &= ~(1<<bit); \ rt->implementation_private |= (!!on << bit); \ @@ -43,7 +43,7 @@ static void pmf_gpio_set_hw_reset(struct gpio_runtime *rt, int on) if (unlikely(!rt)) return; rc = pmf_call_function(rt->node, "hw-reset", &args); if (rc) - printk(KERN_WARNING "pmf_gpio_set_hw_reset" + pr_warn("pmf_gpio_set_hw_reset" " failed, rc: %d\n", rc); } @@ -190,7 +190,7 @@ static int pmf_set_notify(struct gpio_runtime *rt, name, irq_client); if (err) { - printk(KERN_ERR "snd-aoa: gpio layer failed to" + pr_err("snd-aoa: gpio layer failed to" " register %s irq (%d)\n", name, err); kfree(irq_client); goto out_unlock; diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 9dc5806..c9d1956 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -773,18 +773,18 @@ static int check_codec(struct aoa_codec *codec, "platform-%s-codec-ref", codec->name); ref = of_get_property(ldev->sound, propname, NULL); if (!ref) { - printk(KERN_INFO "snd-aoa-fabric-layout: " + pr_info("snd-aoa-fabric-layout: " "required property %s not present\n", propname); return -ENODEV; } if (*ref != codec->node->phandle) { - printk(KERN_INFO "snd-aoa-fabric-layout: " + pr_info("snd-aoa-fabric-layout: " "%s doesn't match!\n", propname); return -ENODEV; } } else { if (layouts_list_items != 1) { - printk(KERN_INFO "snd-aoa-fabric-layout: " + pr_info("snd-aoa-fabric-layout: " "more than one soundbus, but no references.\n"); return -ENODEV; } @@ -796,7 +796,7 @@ static int check_codec(struct aoa_codec *codec, if (!cc) return -EINVAL; - printk(KERN_INFO "snd-aoa-fabric-layout: can use this codec\n"); + pr_info("snd-aoa-fabric-layout: can use this codec\n"); codec->connected = 0; codec->fabric_data = cc; @@ -1017,7 +1017,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) } if (!layout) { - printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n"); + pr_err("snd-aoa-fabric-layout: unknown layout\n"); goto outnodev; } @@ -1036,12 +1036,12 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) case 51: /* PowerBook5,4 */ case 58: /* Mac Mini */ ldev->gpio.methods = ftr_gpio_methods; - printk(KERN_DEBUG + pr_debug( "snd-aoa-fabric-layout: Using direct GPIOs\n"); break; default: ldev->gpio.methods = pmf_gpio_methods; - printk(KERN_DEBUG + pr_debug( "snd-aoa-fabric-layout: Using PMF GPIOs\n"); } ldev->selfptr_headphone.ptr = ldev; @@ -1064,7 +1064,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) err = aoa_fabric_register(&layout_fabric, &sdev->ofdev.dev); if (err && err != -EALREADY) { - printk(KERN_INFO "snd-aoa-fabric-layout: can't use," + pr_info("snd-aoa-fabric-layout: can't use," " another fabric is active!\n"); goto outlistdel; } diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 7487eb7..c1438eb 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -172,7 +172,7 @@ int soundbus_add_one(struct soundbus_dev *dev) !dev->ofdev.dev.of_node || dev->pcmname || dev->pcmid != -1) { - printk(KERN_ERR "soundbus: adding device failed sanity check!\n"); + pr_err("soundbus: adding device failed sanity check!\n"); return -EINVAL; } diff --git a/sound/aoa/soundbus/i2sbus/control.c b/sound/aoa/soundbus/i2sbus/control.c index 4dc9b49..2672d1c 100644 --- a/sound/aoa/soundbus/i2sbus/control.c +++ b/sound/aoa/soundbus/i2sbus/control.c @@ -124,7 +124,7 @@ int i2sbus_control_cell(struct i2sbus_control *c, return pmf_call_one(i2sdev->cell_enable, &args); break; default: - printk(KERN_ERR "i2sbus: INVALID CELL ENABLE VALUE\n"); + pr_err("i2sbus: INVALID CELL ENABLE VALUE\n"); return -ENODEV; } @@ -167,7 +167,7 @@ int i2sbus_control_clock(struct i2sbus_control *c, return pmf_call_one(i2sdev->clock_enable, &args); break; default: - printk(KERN_ERR "i2sbus: INVALID CLOCK ENABLE VALUE\n"); + pr_err("i2sbus: INVALID CLOCK ENABLE VALUE\n"); return -ENODEV; } diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index a80d5ea..d7bb0de 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -264,7 +264,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, resource_size(&dev->resources[i]), dev->rnames[i]); if (!dev->allocated_resource[i]) { - printk(KERN_ERR "i2sbus: failed to claim resource %d!\n", i); + pr_err("i2sbus: failed to claim resource %d!\n", i); goto err; } } @@ -298,12 +298,12 @@ static int i2sbus_add_dev(struct macio_dev *macio, goto err; if (i2sbus_control_add_dev(dev->control, dev)) { - printk(KERN_ERR "i2sbus: control layer didn't like bus\n"); + pr_err("i2sbus: control layer didn't like bus\n"); goto err; } if (soundbus_add_one(&dev->sound)) { - printk(KERN_DEBUG "i2sbus: device registration error!\n"); + pr_debug("i2sbus: device registration error!\n"); goto err; } @@ -340,7 +340,7 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) if (err) return err; if (!control) { - printk(KERN_ERR "i2sbus_control_init API breakage\n"); + pr_err("i2sbus_control_init API breakage\n"); return -ENODEV; } diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 7b74a4b..f9e272c 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -268,7 +268,7 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev, pi->stop_completion = NULL; if (timeout == 0) { /* timeout expired, stop dbdma forcefully */ - printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n"); + pr_err("i2sbus_wait_for_stop: timed out\n"); /* make sure RUN, PAUSE and S0 bits are cleared */ out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16); pi->dbdma_ring.stopping = 0; @@ -681,7 +681,7 @@ static inline void handle_interrupt(struct i2sbus_dev *i2sdev, int in) if (!(status & ACTIVE) && (!in || (status & 0x80))) break; if (--timeout <= 0) { - printk(KERN_ERR "i2sbus: timed out " + pr_err("i2sbus: timed out " "waiting for DMA to stop!\n"); break; } @@ -868,7 +868,7 @@ static void i2sbus_private_free(struct snd_pcm *pcm) i2sdev->out.created = 0; i2sdev->in.created = 0; list_for_each_entry_safe(p, tmp, &i2sdev->sound.codec_list, list) { - printk(KERN_ERR "i2sbus: a codec didn't unregister!\n"); + pr_err("i2sbus: a codec didn't unregister!\n"); list_del(&p->list); module_put(p->codec->owner); kfree(p); @@ -887,7 +887,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, struct codec_info_item *cii; if (!dev->pcmname || dev->pcmid == -1) { - printk(KERN_ERR "i2sbus: pcm name and id must be set!\n"); + pr_err("i2sbus: pcm name and id must be set!\n"); return -EINVAL; } @@ -910,12 +910,12 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, * sysclock/busclock stuff above to depend on which is usable */ list_for_each_entry(cii, &dev->codec_list, list) { if (cii->codec->sysclock_factor != ci->sysclock_factor) { - printk(KERN_DEBUG + pr_debug( "cannot yet handle multiple different sysclocks!\n"); return -EINVAL; } if (cii->codec->bus_factor != ci->bus_factor) { - printk(KERN_DEBUG + pr_debug( "cannot yet handle multiple different bus clocks!\n"); return -EINVAL; } @@ -932,7 +932,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL); if (!cii) { - printk(KERN_DEBUG "i2sbus: failed to allocate cii\n"); + pr_debug("i2sbus: failed to allocate cii\n"); return -ENOMEM; } @@ -942,20 +942,20 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, cii->codec_data = data; if (!cii->sdev) { - printk(KERN_DEBUG + pr_debug( "i2sbus: failed to get soundbus dev reference\n"); err = -ENODEV; goto out_free_cii; } if (!try_module_get(THIS_MODULE)) { - printk(KERN_DEBUG "i2sbus: failed to get module reference!\n"); + pr_debug("i2sbus: failed to get module reference!\n"); err = -EBUSY; goto out_put_sdev; } if (!try_module_get(ci->owner)) { - printk(KERN_DEBUG + pr_debug( "i2sbus: failed to get module reference to codec owner!\n"); err = -EBUSY; goto out_put_this_module; @@ -965,7 +965,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, err = snd_pcm_new(card, dev->pcmname, dev->pcmid, 0, 0, &dev->pcm); if (err) { - printk(KERN_DEBUG "i2sbus: failed to create pcm\n"); + pr_debug("i2sbus: failed to create pcm\n"); goto out_put_ci_module; } dev->pcm->dev = &dev->ofdev.dev; @@ -978,7 +978,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, if (!i2sdev->out.created && out) { if (dev->pcm->card != card) { /* eh? */ - printk(KERN_ERR + pr_err( "Can't attach same bus to different cards!\n"); err = -EINVAL; goto out_put_ci_module; @@ -993,7 +993,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, if (!i2sdev->in.created && in) { if (dev->pcm->card != card) { - printk(KERN_ERR + pr_err( "Can't attach same bus to different cards!\n"); err = -EINVAL; goto out_put_ci_module; @@ -1014,7 +1014,7 @@ i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, */ err = snd_device_register(card, dev->pcm); if (err) { - printk(KERN_ERR "i2sbus: error registering new pcm\n"); + pr_err("i2sbus: error registering new pcm\n"); goto out_put_ci_module; } /* no errors any more, so let's add this to our list */ -- 1.8.1.2

3 6

[alsa-devel] [PATCH] ASoC:simple-card: Add multi-CODEC support
by Jean-Francois Moine 10 Sep '14

10 Sep '14

This patch adds multi-CODEC support to the simple-card. Signed-off-by: Jean-Francois Moine <moinejf(a)free.fr> --- .../devicetree/bindings/sound/simple-card.txt | 28 +++++++-------- sound/soc/generic/simple-card.c | 41 +++++++++++++++++++++- 2 files changed, 52 insertions(+), 17 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index c2e9841..c217687 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -37,6 +37,8 @@ Required dai-link subnodes: - cpu : CPU sub-node - codec : CODEC sub-node + In case of multi-CODECs, there may + be many of such sub-nodes. Optional dai-link subnode properties: @@ -115,37 +117,31 @@ sh_fsi2: sh_fsi2@ec230000 { interrupts = <0 146 0x4>; }; -Example 2 - many DAI links: +Example 2 - many DAI links and multi-CODECs: sound { compatible = "simple-audio-card"; simple-audio-card,name = "Cubox Audio"; - simple-audio-card,dai-link@0 { /* I2S - HDMI */ + simple-audio-card,dai-link@0 { /* S/PDIF - HDMI & S/PDIF */ format = "i2s"; cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - cpu { sound-dai = <&audio1 1>; }; - codec { - sound-dai = <&tda998x 1>; + codec@0 { + sound-dai = <&hdmi 0>; + }; + codec@1 { + sound-dai = <&spdif_codec>; }; }; - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + simple-audio-card,dai-link@1 { /* I2S - HDMI */ cpu { - sound-dai = <&audio1 1>; + sound-dai = <&audio1 0>; }; codec { - sound-dai = <&spdif_codec>; + sound-dai = <&hdmi 1>; }; }; }; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 4053152..bf0ce08 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -179,11 +179,12 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, struct device_node *np = NULL; struct device_node *bitclkmaster = NULL; struct device_node *framemaster = NULL; + struct snd_soc_dai_link_component *component; unsigned int daifmt; char *name; char prop[128]; char *prefix = ""; - int ret, cpu_args; + int ret, cpu_args, num_codec_dais; /* For single DAI link & old style of DT node */ if (is_top_level_node) @@ -225,7 +226,16 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, } of_node_put(np); + + /* count the number of codec DAIs */ snprintf(prop, sizeof(prop), "%scodec", prefix); + num_codec_dais = 0; + for_each_child_of_node(node, np) { + if (strcmp(np->name, prop) == 0) + num_codec_dais++; + } + + /* treat the first DAI */ np = of_get_child_by_name(node, prop); if (!np) { ret = -EINVAL; @@ -307,6 +317,35 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, if (!cpu_args) dai_link->cpu_dai_name = NULL; + /* handle multi-codec DAIs */ + if (num_codec_dais == 1) + goto out; + dai_link->codecs = component = + devm_kzalloc(dev, + sizeof *component * num_codec_dais, + GFP_KERNEL); + dai_link->num_codecs = num_codec_dais; + component->of_node = dai_link->codec_of_node; + dai_link->codec_of_node = NULL; + component->dai_name = dai_link->codec_dai_name; + dai_link->codec_dai_name = NULL; + for (;;) { + np = of_get_next_child(node, np); + if (!np) + break; + component++; + component->of_node = of_parse_phandle(np, "sound-dai", 0); + if (!component->of_node) { + ret = -ENODEV; + dev_err(dev, "Bad sound-dai\n"); + goto dai_link_of_err; + } + ret = snd_soc_of_get_dai_name(np, &component->dai_name); + if (ret < 0) + goto dai_link_of_err; + } + +out: dai_link_of_err: if (np) of_node_put(np); -- 2.1.0

1 0

[alsa-devel] [PATCH] ASoC: simple-card: fix regression in clock rate lookup
by Arnd Bergmann 10 Sep '14

10 Sep '14

Commit 7c7b9cf53d284f ("ASoC: simple-card: fixup cpu_dai_name clear case") changed the way that "sound-dai" properties are handled, which leads to the clock frequency not being picked up from the node that the phandle points to, as correctly identified by gcc with this warning: sound/soc/generic/simple-card.c: In function 'asoc_simple_card_sub_parse_of': sound/soc/generic/simple-card.c:165:7: warning: 'node' may be used uninitialized in this function [-Wmaybe-uninitialized] This restores the previous behavior by using the node from of_parse_phandle_with_args() that was previously being returned from of_parse_phandle(). Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index b563850c43f4..a0440bd533d5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -116,7 +116,6 @@ asoc_simple_card_sub_parse_of(struct device_node *np, int *args_count) { struct of_phandle_args args; - struct device_node *node; struct clk *clk; u32 val; int ret; @@ -162,7 +161,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, } else if (!of_property_read_u32(np, "system-clock-frequency", &val)) { dai->sysclk = val; } else { - clk = of_clk_get(node, 0); + clk = of_clk_get(args.np, 0); if (!IS_ERR(clk)) dai->sysclk = clk_get_rate(clk); }

2 1

[alsa-devel] [PATCHv4] ASoC: simple-card: Adjust the comments of simple card.
by Xiubo Li 10 Sep '14

10 Sep '14

Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Changes in V4: - To apply against current code. sound/soc/generic/simple-card.c | 48 ++++++++++++++++++++--------------------- 1 file changed, 24 insertions(+), 24 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 4053152..f20feb4 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -122,7 +122,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, int ret; /* - * get node via "sound-dai = <&phandle port>" + * Get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() */ ret = of_parse_phandle_with_args(np, "sound-dai", @@ -135,19 +135,19 @@ asoc_simple_card_sub_parse_of(struct device_node *np, if (args_count) *args_count = args.args_count; - /* get dai->name */ + /* Get dai->name */ ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) return ret; - /* parse TDM slot */ + /* Parse TDM slot */ ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); if (ret) return ret; /* - * dai->sysclk come from - * "clocks = <&xxx>" (if system has common clock) + * Parse dai->sysclk come from "clocks = <&xxx>" + * (if system has common clock) * or "system-clock-frequency = <xxx>" * or device's module clock. */ @@ -240,9 +240,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, goto dai_link_of_err; if (strlen(prefix) && !bitclkmaster && !framemaster) { - /* No dai-link level and master setting was not found from - sound node level, revert back to legacy DT parsing and - take the settings from codec node. */ + /* + * No DAI link level and master setting was found + * from sound node level, revert back to legacy DT + * parsing and take the settings from codec node. + */ dev_dbg(dev, "%s: Revert to legacy daifmt parsing\n", __func__); dai_props->cpu_dai.fmt = dai_props->codec_dai.fmt = @@ -271,10 +273,10 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, goto dai_link_of_err; } - /* simple-card assumes platform == cpu */ + /* Simple Card assumes platform == cpu */ dai_link->platform_of_node = dai_link->cpu_of_node; - /* Link name is created from CPU/CODEC dai name */ + /* DAI link name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, strlen(dai_link->cpu_dai_name) + strlen(dai_link->codec_dai_name) + 2, @@ -296,11 +298,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, dai_props->codec_dai.sysclk); /* - * soc_bind_dai_link() will check cpu name - * after of_node matching if dai_link has cpu_dai_name. - * but, it will never match if name was created by fmt_single_name() - * remove cpu_dai_name if cpu_args was 0. - * see + * In soc_bind_dai_link() will check cpu name after + * of_node matching if dai_link has cpu_dai_name. + * but, it will never match if name was created by + * fmt_single_name() remove cpu_dai_name if cpu_args + * was 0. See: * fmt_single_name() * fmt_multiple_name() */ @@ -329,10 +331,10 @@ static int asoc_simple_card_parse_of(struct device_node *node, if (!node) return -EINVAL; - /* parsing the card name from DT */ + /* Parse the card name from DT */ snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name"); - /* off-codec widgets */ + /* The off-codec widgets */ if (of_property_read_bool(node, "simple-audio-card,widgets")) { ret = snd_soc_of_parse_audio_simple_widgets(&priv->snd_card, "simple-audio-card,widgets"); @@ -387,7 +389,7 @@ static int asoc_simple_card_parse_of(struct device_node *node, return 0; } -/* update the reference count of the devices nodes at end of probe */ +/* Decrease the reference count of the device nodes */ static int asoc_simple_card_unref(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); @@ -416,29 +418,27 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; int num_links, ret; - /* get the number of DAI links */ + /* Get the number of DAI links */ if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) num_links = of_get_child_count(np); else num_links = 1; - /* allocate the private data and the DAI link array */ + /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, sizeof(*priv) + sizeof(*dai_link) * num_links, GFP_KERNEL); if (!priv) return -ENOMEM; - /* - * init snd_soc_card - */ + /* Init snd_soc_card */ priv->snd_card.owner = THIS_MODULE; priv->snd_card.dev = dev; dai_link = priv->dai_link; priv->snd_card.dai_link = dai_link; priv->snd_card.num_links = num_links; - /* get room for the other properties */ + /* Get room for the other properties */ priv->dai_props = devm_kzalloc(dev, sizeof(*priv->dai_props) * num_links, GFP_KERNEL); -- 2.1.0.27.g96db324

2 1

[alsa-devel] [PATCH v2] ASoC: soc-pcm: fix dpcm_path_get error handling
by Qiao Zhou 10 Sep '14

10 Sep '14

per Mark's suggestion, add braces to the else if branch Qiao Zhou (1): ASoC: soc-pcm: fix dpcm_path_get error handling sound/soc/soc-compress.c | 6 +++++- sound/soc/soc-pcm.c | 6 +++++- 2 files changed, 10 insertions(+), 2 deletions(-)

2 3