September 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: arizona: Improve handling of setting REFCLK to 0
by Charles Keepax 17 Sep '13

17 Sep '13

This patch suppresses calculation of REFCLK parameters when the REFCLK source frequency is set to zero, additionally it will consider a source frequency of zero as the REFCLK being disabled and switch to using the SYNCCLK. Reported-by: Kyung Kwee Ryu <Kyung-Kwee.Ryu(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- This patch missed getting upstreamed at the time it was written, sorry about that. Thanks, Charles sound/soc/codecs/arizona.c | 23 ++++++++++++++--------- 1 files changed, 14 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 657808b..6f05b17 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1477,21 +1477,25 @@ static void arizona_enable_fll(struct arizona_fll *fll, { struct arizona *arizona = fll->arizona; int ret; + bool use_sync = false; /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. */ - if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + if (fll->ref_src >= 0 && fll->ref_freq && + fll->ref_src != fll->sync_src) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); - if (fll->sync_src >= 0) + if (fll->sync_src >= 0) { arizona_apply_fll(arizona, fll->base + 0x10, sync, fll->sync_src, true); + use_sync = true; + } } else if (fll->sync_src >= 0) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, @@ -1511,7 +1515,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, * Increase the bandwidth if we're not using a low frequency * sync source. */ - if (fll->sync_src >= 0 && fll->sync_freq > 100000) + if (use_sync && fll->sync_freq > 100000) regmap_update_bits(arizona->regmap, fll->base + 0x17, ARIZONA_FLL1_SYNC_BW, 0); else @@ -1526,8 +1530,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->ref_src >= 0 && fll->sync_src >= 0 && - fll->ref_src != fll->sync_src) + if (use_sync) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1561,10 +1564,12 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout && Fref > 0) { - ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); - if (ret != 0) - return ret; + if (fll->fout) { + if (Fref > 0) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); + if (ret != 0) + return ret; + } if (fll->sync_src >= 0) { ret = arizona_calc_fll(fll, &sync, fll->sync_freq, -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: bells: Add missing route to power up DSP clock
by Charles Keepax 17 Sep '13

17 Sep '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/bells.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 29e2468..84f5d8b 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -356,6 +356,7 @@ static struct snd_soc_dapm_widget bells_widgets[] = { static struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, + { "CLKIN", NULL, "OPCLK" }, { "DMIC", NULL, "MICBIAS2" }, { "IN2L", NULL, "DMIC" }, -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: core: Add API for configuration of DAI BCLK ratio
by Liam Girdwood 17 Sep '13

17 Sep '13

Some codec drivers when running in slave mode require that BCLK to sample rate ratio is explicitly set by the machine driver as it may not be exactly rate * frame size. Extend the DAI API by adding :- int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dai.h | 3 +++ sound/soc/soc-core.c | 16 ++++++++++++++++ 2 files changed, 19 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227..d8acf0c 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -105,6 +105,8 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); + /* Digital Audio interface formatting */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); @@ -131,6 +133,7 @@ struct snd_soc_dai_ops { int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); + int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); /* * DAI format configuration diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 09e341d..8014839 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3380,6 +3380,22 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** + * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. + * @dai: DAI + * @ratio Ratio of BCLK to Sample rate. + * + * Configures the DAI for a preset BCLK to sample rate ratio. + */ +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + if (dai->driver && dai->driver->ops->set_bclk_ratio) + return dai->driver->ops->set_bclk_ratio(dai, ratio); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); + +/** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI * @fmt: SND_SOC_DAIFMT_ format value. -- 1.8.1.2

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[alsa-devel] USB Audio initialization race
by David Henningsson 17 Sep '13

17 Sep '13

I'm sometimes seeing that USB audio devices are not initialized correctly by PulseAudio, but I can't reproduce this myself. I have, however, spotted what I think is a race condition in the driver code, if a device has more than one interface. snd_card_register is called once for every interface, which means that it will be shown to userspace after the first interface has been probed. So the race could look like: 1) The first interface is probed, which might contain the mixer controls 2) snd_card_register is called, which sends a signal to userspace 3) PulseAudio probes the device and notices that it has no PCMs, so it ignores the device 4) The second interface is probed, which adds the PCMs 5) snd_card_register is again called, which is a no-op since the card already exist 6) User is unhappy because his plugged in sound card did not show up in PulseAudio. So, assuming all this is right, it seems like we need some type of callback from the usb driver when all the interfaces for the sound card has been probed, so we can call snd_card_register at that point instead. Thoughts? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH 1/2] ASoC: core: Add devm_snd_soc_register_card()
by Mark Brown 16 Sep '13

16 Sep '13

From: Mark Brown <broonie(a)linaro.org> Simplify error handling and remove repetitive (and rarely executed) code for unregistration by providing a devm_snd_soc_register() card. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- include/sound/soc.h | 1 + sound/soc/soc-devres.c | 34 ++++++++++++++++++++++++++++++++++ 2 files changed, 35 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index f126676..7ab5d2b 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -364,6 +364,7 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); int snd_soc_poweroff(struct device *dev); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index 13fe86f..b1d7322 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -50,3 +50,37 @@ int devm_snd_soc_register_component(struct device *dev, return ret; } EXPORT_SYMBOL_GPL(devm_snd_soc_register_component); + +static void devm_card_release(struct device *dev, void *res) +{ + snd_soc_unregister_card(*(struct snd_soc_card **)res); +} + +/** + * devm_snd_soc_register_card - resource managed card registration + * @dev: Device used to manage card + * @card: Card to register + * + * Register a card with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_card(card); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_card); -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: core: Only add platform DAI widgets once.
by Liam Girdwood 16 Sep '13

16 Sep '13

Currently platform CPU DAI widgets are created in soc_probe_platform and soc_probe_link_dais. Remove the extra call in soc_probe_link_dais(). Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-core.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8e86aba..09e341d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1365,7 +1365,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return -ENODEV; list_add(&cpu_dai->dapm.list, &card->dapm_list); - snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); } if (cpu_dai->driver->probe) { -- 1.8.1.2

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Re: [alsa-devel] No sound from internal speaker on 2013 Macbook Air 6, 1
by Ben Whitten 16 Sep '13

16 Sep '13

> > > OK, then the normal jack detection doesn't work with this machine, as > > > it seems. You can try the above to all pins for checking if any of > > > other pins reacts to the headphone jack. > > > > No luck I'm afraid - none of the other pins seem to react to the > > headphones. Is there anything else you can think of that I can try > > tomorrow? > > Well, it might be notified via GPIO. Try to read GPIO pins except for > GPIO0. That is, set the all unmasked, the direction to READ. We have found that jack sense comes in on the headphone pin in a windows bootcamp install. However this bit isn't set in linux using either the default chip, bootloader, or windows pin configs. No other GPIO's are enabled in windows. Ben

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[alsa-devel] Roland UA-55 (snd-usb-audio)
by Michael Findlater 15 Sep '13

15 Sep '13

Hello there! I am attempting to use the Roland UA-55, which apparently should work after recent changes were made for support for Roland devices by Clemens Ladisch. I am using the 3.11 kernel upon Debian sid. The device shows in /proc/asound/cards but does not seem to work. The GNOME volume mixer it shows the device as a "QUAD-CAPTURE Analog Surround 4.0", but I am not sure whether this is correct. When I select the card as the output device and try to play to it, nothing will play, 'aplay' does not work either (I hear nothing). If I turn the headphone output knob on the device up I hear a very soft static in my right ear. 'alsamixer' reports that 'This sound device does not have any controls.' I do not know where to go from here, but have ran the "Alsa Information Script", an output of this is available at: http://www.alsa-project.org/db/?f=8f62ce460ecc5cac394c0b014b2e2aa4d28ae2dc I am stuck with how to proceed to debug the issue. If at all possible, I would be grateful if someone is able to point me in the right direction. Best regards, Michael Findlater

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[alsa-devel] [PATCH RFC 0/8] Fix AM335x-evm analog audio support
by oku＠iki.fi 13 Sep '13

13 Sep '13

From: Jyri Sarha <jsarha(a)ti.com> This set of patches fixes the basic audio support for am335x-evm. It should also be relatively simple to add the necessary nodes to relevant dts files to get BeagleBone + AudioCape and am335x-evmsk working too. The patch set depends on following patches: [PATCH v2] ARM: dts: add AM33XX EDMA support https://lkml.org/lkml/2013/8/26/57 [PATCH v11 5/8] ARM: dts: add AM33XX SPI DMA support https://lkml.org/lkml/2013/6/18/55 [PATCH v2] ARM: EDMA: Fix clearing of unused list for DT DMA resources https://lkml.org/lkml/2013/7/22/441 I have tried my best not to break the existing support for older davinci boards, but since I do not have those boards I can not be sure. Some commit comments refer to a dmaengine based davinci audio implementation which is planned for but nothing has been done yet. Best regards, Jyri Darren Etheridge (1): ARM/dts: am335x-evm: Add audio support for am335x-evm.dts Hebbar, Gururaja (2): ASoC: davinci-evm: Add device tree binding ASoC: davinci: Add support for AM33xx SoC Audio Joel A Fernandes (1): ASoC: davinci-mcasp: Add pinctrl support Jyri Sarha (3): ASoC: davinci-mcasp: Get DMA related properties from DT ASoC: davinci-mcasp: Extract DMA channels directly from DT ARM/dts: am33xx: mcasp: Add new dma related properties Pantelis Antoniou (1): ARM/dts: am33xx: Add mcasp0 and mcasp1 device tree entries .../bindings/sound/davinci-evm-audio.txt | 53 ++++++ .../bindings/sound/davinci-mcasp-audio.txt | 13 +- arch/arm/boot/dts/am335x-evm.dts | 56 ++++++ arch/arm/boot/dts/am33xx.dtsi | 35 ++++ include/linux/platform_data/davinci_asp.h | 2 + sound/soc/davinci/Kconfig | 8 + sound/soc/davinci/Makefile | 3 + sound/soc/davinci/davinci-evm.c | 181 +++++++++++++++++--- sound/soc/davinci/davinci-mcasp.c | 82 +++++++-- 9 files changed, 391 insertions(+), 42 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt -- 1.7.9.5

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[alsa-devel] [patch] ASoC: ab8500-codec: info leak in anc_status_control_put()
by Dan Carpenter 13 Sep '13

13 Sep '13

If the user passes an invalid value it leads to an info leak when we print the error message or it could oops. This is called with user supplied data from snd_ctl_elem_write(). Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b8ba0ad..80555d7 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); struct device *dev = codec->dev; bool apply_fir, apply_iir; - int req, status; + unsigned int req; + int status; dev_dbg(dev, "%s: Enter.\n", __func__); mutex_lock(&drvdata->anc_lock); req = ucontrol->value.integer.value[0]; + if (req >= ARRAY_SIZE(enum_anc_state)) { + status = -EINVAL; + goto cleanup; + } if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",

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