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

[alsa-devel] [PATCH] ALSA: ASoC: au1x: fix ac97c module build
by Manuel Lauss 15 Jul '13

15 Jul '13

The recent conversion to module_platform_driver left a stray ampersand in there preventing a build. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- Against 3.11-rc1 sound/soc/au1x/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694..c8a2de1 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); -- 1.8.3.2

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[alsa-devel] [PATCH] ARM: EXYNOS: remove references to non-DT SoCs
by Paul Bolle 15 Jul '13

15 Jul '13

Commit 383ffda2fa ("ARM: EXYNOS: no more support non-DT for EXYNOS SoCs") removed the Kconfig symbols MACH_SMDKV310 and MACH_SMDKC210. Now remove the references to both of these symbols from the Kconfig entry for SND_SOC_SAMSUNG_SMDK_WM9713. They're not needed anymore. Signed-off-by: Paul Bolle <pebolle(a)tiscali.nl> --- Untested. sound/soc/samsung/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 9855dfc..3d76659 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -136,7 +136,7 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help -- 1.8.1.4

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[alsa-devel] [PATCH] ASoC: omap-abe-twl6040: Remove support for pdata (legacy boot)
by Peter Ujfalusi 15 Jul '13

15 Jul '13

Just recently OMAP4 legacy boot support has been removed. No reason to keep the code used by the legacy boot (pdata based) since neither OMAP4 or OMAP5 can boot in this mode. The platform_data header file will be removed at later point to avoid build issues. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- include/linux/platform_data/omap-abe-twl6040.h | 2 + sound/soc/omap/omap-abe-twl6040.c | 133 ++++++++----------------- 2 files changed, 43 insertions(+), 92 deletions(-) diff --git a/include/linux/platform_data/omap-abe-twl6040.h b/include/linux/platform_data/omap-abe-twl6040.h index 5d298ac..065abbe 100644 --- a/include/linux/platform_data/omap-abe-twl6040.h +++ b/include/linux/platform_data/omap-abe-twl6040.h @@ -21,6 +21,8 @@ * 02110-1301 USA */ +/* TODO: This file can be removed in 3.12 since it is no longer needed */ + #ifndef _OMAP_ABE_TWL6040_H_ #define _OMAP_ABE_TWL6040_H_ diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 70cd5c7..ebb1390 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -23,7 +23,6 @@ #include <linux/clk.h> #include <linux/platform_device.h> #include <linux/mfd/twl6040.h> -#include <linux/platform_data/omap-abe-twl6040.h> #include <linux/module.h> #include <linux/of.h> @@ -166,19 +165,10 @@ static const struct snd_soc_dapm_route audio_map[] = { {"AFMR", NULL, "Line In"}, }; -static inline void twl6040_disconnect_pin(struct snd_soc_dapm_context *dapm, - int connected, char *pin) -{ - if (!connected) - snd_soc_dapm_disable_pin(dapm, pin); -} - static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; - struct omap_abe_twl6040_data *pdata = dev_get_platdata(card->dev); struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); int hs_trim; int ret = 0; @@ -203,24 +193,6 @@ static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd) twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); } - /* - * NULL pdata means we booted with DT. In this case the routing is - * provided and the card is fully routed, no need to mark pins. - */ - if (!pdata) - return ret; - - /* Disable not connected paths if not used */ - twl6040_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone"); - twl6040_disconnect_pin(dapm, pdata->has_hf, "Ext Spk"); - twl6040_disconnect_pin(dapm, pdata->has_ep, "Earphone Spk"); - twl6040_disconnect_pin(dapm, pdata->has_aux, "Line Out"); - twl6040_disconnect_pin(dapm, pdata->has_vibra, "Vibrator"); - twl6040_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_mainmic, "Main Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_submic, "Sub Handset Mic"); - twl6040_disconnect_pin(dapm, pdata->has_afm, "Line In"); - return ret; } @@ -274,13 +246,18 @@ static struct snd_soc_card omap_abe_card = { static int omap_abe_probe(struct platform_device *pdev) { - struct omap_abe_twl6040_data *pdata = dev_get_platdata(&pdev->dev); struct device_node *node = pdev->dev.of_node; struct snd_soc_card *card = &omap_abe_card; + struct device_node *dai_node; struct abe_twl6040 *priv; int num_links = 0; int ret = 0; + if (!node) { + dev_err(&pdev->dev, "of node is missing.\n"); + return -ENODEV; + } + card->dev = &pdev->dev; priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL); @@ -289,78 +266,50 @@ static int omap_abe_probe(struct platform_device *pdev) priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (node) { - struct device_node *dai_node; - - if (snd_soc_of_parse_card_name(card, "ti,model")) { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } + if (snd_soc_of_parse_card_name(card, "ti,model")) { + dev_err(&pdev->dev, "Card name is not provided\n"); + return -ENODEV; + } - ret = snd_soc_of_parse_audio_routing(card, - "ti,audio-routing"); - if (ret) { - dev_err(&pdev->dev, - "Error while parsing DAPM routing\n"); - return ret; - } + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); + if (ret) { + dev_err(&pdev->dev, "Error while parsing DAPM routing\n"); + return ret; + } - dai_node = of_parse_phandle(node, "ti,mcpdm", 0); - if (!dai_node) { - dev_err(&pdev->dev, "McPDM node is not provided\n"); - return -EINVAL; - } - abe_twl6040_dai_links[0].cpu_dai_name = NULL; - abe_twl6040_dai_links[0].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,mcpdm", 0); + if (!dai_node) { + dev_err(&pdev->dev, "McPDM node is not provided\n"); + return -EINVAL; + } + abe_twl6040_dai_links[0].cpu_dai_name = NULL; + abe_twl6040_dai_links[0].cpu_of_node = dai_node; - dai_node = of_parse_phandle(node, "ti,dmic", 0); - if (dai_node) { - num_links = 2; - abe_twl6040_dai_links[1].cpu_dai_name = NULL; - abe_twl6040_dai_links[1].cpu_of_node = dai_node; + dai_node = of_parse_phandle(node, "ti,dmic", 0); + if (dai_node) { + num_links = 2; + abe_twl6040_dai_links[1].cpu_dai_name = NULL; + abe_twl6040_dai_links[1].cpu_of_node = dai_node; - priv->dmic_codec_dev = platform_device_register_simple( + priv->dmic_codec_dev = platform_device_register_simple( "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, - "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } - } else { - num_links = 1; - } - - priv->jack_detection = of_property_read_bool(node, - "ti,jack-detection"); - of_property_read_u32(node, "ti,mclk-freq", - &priv->mclk_freq); - if (!priv->mclk_freq) { - dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + if (IS_ERR(priv->dmic_codec_dev)) { + dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); + return PTR_ERR(priv->dmic_codec_dev); } - - omap_abe_card.fully_routed = 1; - } else if (pdata) { - if (pdata->card_name) { - card->name = pdata->card_name; - } else { - dev_err(&pdev->dev, "Card name is not provided\n"); - return -ENODEV; - } - - if (pdata->has_dmic) - num_links = 2; - else - num_links = 1; - - priv->jack_detection = pdata->jack_detection; - priv->mclk_freq = pdata->mclk_freq; } else { - dev_err(&pdev->dev, "Missing pdata\n"); - return -ENODEV; + num_links = 1; + } + + priv->jack_detection = of_property_read_bool(node, "ti,jack-detection"); + of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); + if (!priv->mclk_freq) { + dev_err(&pdev->dev, "MCLK frequency not provided\n"); + ret = -EINVAL; + goto err_unregister; } + card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); -- 1.8.2.1

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[alsa-devel] ASoC: codec: SPDIF only works with Tx/Rx seperate codec drivers?
by Nicolin Chen 15 Jul '13

15 Jul '13

Hi Mark, I've a little question about SPDIF codec drivers. I found there're spdif_receiver.c and spdif_transmitter.c, two files, in sound/soc/codecs directory. So it looks like SPDIF module would work separately as one IN device and one OUT device, for example: dts/spear1340.dtsi: spdif-in@d0100000 { dts/spear1340.dtsi: compatible = "st,spdif-in"; dts/spear1340.dtsi: spdif-out@d0000000 { dts/spear1340.dtsi: compatible = "st,spdif-out"; But the question is what if a SPDIF module have both Tx and Rx function in one simple CPU DAI. Is that okay to add a new codec driver for this situation? Actually I'm gonna prepare some patches to add Freescale SPDIF drivers. But this needs to create a brand new dummy SPDIF codec driver, which's some kinda same as spdif_receiver/transmitter.c but quite different in the supports of SAMPLE_RATE/FORMAT and IN/OUT-in-one integration. So I just want to make sure if this'll be okay. Thank you. And looking forward to your reply. Nicolin Chen

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[alsa-devel] Alsa USB Audio regression
by Ulrich Günther 15 Jul '13

15 Jul '13

Hello, i encounterd a possible alsa bug with the "ID 0d8c:0006 C-Media Electronics, Inc. Storm HP-USB500 5.1 Headset". After Ubuntu 10.04 LTS (Ubuntu 10.10 works, Ubuntu 11.04 doesn't work at all, Ubuntu 11.10 and newer have the bug), the headset only plays sound at maximum volume level if the volume is turned down just a little no sound is being played. I have filled a Ubuntu bug for 12.04 LTS at: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1200925 . I was advised to contact this mailing list. AlsaInfo for Ubuntu 12.04 LTS with the bug: http://www.alsa-project.org/db/?f=3bde76b15a15f585418a6a4a4703ca67fd4cdf42 . AlsaInfo for Ubuntu 10.04 LTS without the bug: http://www.alsa-project.org/db/?f=5824e55345c3f7128937405254485a489886b9dc . Thank You for your help.

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Re: [alsa-devel] [PATCH 00/50] USB: cleanup spin_lock in URB->complete()
by Andy Walls 14 Jul '13

14 Jul '13

On Thu, 2013-07-11 at 17:05 +0800, Ming Lei wrote: > Hi, > > As we are going to run URB->complete() in tasklet context[1][2], Hi, Please pardon my naivete, but why was it decided to use tasklets to defer work, as opposed to some other deferred work mechanism? It seems to me that getting rid of tasklets has been an objective for years: http://lwn.net/Articles/239633/ http://lwn.net/Articles/520076/ http://lwn.net/Articles/240054/ Regards, Andy > and > hard interrupt may be enabled when running URB completion handler[3], > so we might need to disable interrupt when acquiring one lock in > the completion handler for the below reasons: > > - URB->complete() holds a subsystem wide lock which may be acquired > in another hard irq context, and the subsystem wide lock is acquired > by spin_lock()/read_lock()/write_lock() in complete() > > - URB->complete() holds a private lock with spin_lock()/read_lock()/write_lock() > but driver may export APIs to make other drivers acquire the same private > lock in its interrupt handler. > > For the sake of safety and making the change simple, this patch set > converts all spin_lock()/read_lock()/write_lock() in completion handler > path into their irqsave version mechanically. > > But if you are sure the above two cases do not happen in your driver, > please let me know and I can drop the unnecessary change. > > Also if you find some conversions are missed, also please let me know so > that I can add it in the next round. > > > [1], http://marc.info/?l=linux-usb&m=137286322526312&w=2 > [2], http://marc.info/?l=linux-usb&m=137286326726326&w=2 > [3], http://marc.info/?l=linux-usb&m=137286330626363&w=2 > [snip] > > > Thanks, > -- > Ming Lei

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[alsa-devel] [PATCH v2 1/3] ASoC: sglt5000: Provide the reg_stride field
by Fabio Estevam 12 Jul '13

12 Jul '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> sgtl5000 has 16-bit registers, and only even numbers are valid for its registers addresses. Let regmap knows about this feature by specifying the 'reg_stride' field, so that it can access only the valid registers. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Fix the description in the commit log sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559..7c99f3c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1470,6 +1470,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = { static const struct regmap_config sgtl5000_regmap = { .reg_bits = 16, .val_bits = 16, + .reg_stride = 2, .max_register = SGTL5000_MAX_REG_OFFSET, .volatile_reg = sgtl5000_volatile, -- 1.8.1.2

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[alsa-devel] [PATCH] ASoC: fsl: Disable SSI in trigger() if RE/TE are both cleared
by Nicolin Chen 12 Jul '13

12 Jul '13

The code enabled SSIEN when triggered by SNDRV_PCM_TRIGGER_START, so move the disable code to SNDRV_PCM_TRIGGER_STOP for symmetric. This also allows us to use the SSI driver more flexible so that it can support some use cases like "aplay S16_LE.wav S24_LE.wav" which would call the driver in sequence like: startup()->hw_params(S16_LE)->trigger(START)->tirgger(STOP)-> hw_params(S24_LE)->trigger(START)->tirgger(STOP)->shutdown() If we disable SSIEN in shutdown(), the second hw_params() would bypass the sample bits setting while using symmetric_rate. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 12 +++--------- 1 files changed, 3 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2f2d837..b6ab341 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -510,6 +510,9 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); else write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); + + if ((read_ssi(&ssi->scr) & (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); break; default: @@ -534,15 +537,6 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, ssi_private->first_stream = ssi_private->second_stream; ssi_private->second_stream = NULL; - - /* - * If this is the last active substream, disable the SSI. - */ - if (!ssi_private->first_stream) { - struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - - write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); - } } static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) -- 1.7.1

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[alsa-devel] [PATCH RFC] ALSA: usb-audio: add front jack channel selector for EMU0204
by Vasily Khoruzhick 12 Jul '13

12 Jul '13

Add support for front jack channel selector which is present on EMU0204. It allows to get 4 channels out of this soundcard. Tested-by: Yury Bushmelev <jay(a)jay-tech.ru> Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- Control request looks like generic UAC_SET_CUR request, but with a bit weird values, valid values are { 0x01, 0x01} and { 0x01, 0x02 }, and there's no such control type ATM. I'm not well familiar with UAC so any suggestins how to get rid of custom control req are welcome! sound/usb/mixer.h | 1 + sound/usb/mixer_quirks.c | 77 ++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 78 insertions(+) diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index aab80df..4a5f001 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -23,6 +23,7 @@ struct usb_mixer_interface { u8 audigy2nx_leds[3]; u8 xonar_u1_status; + u8 emu0204_ch_switch; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ebe9144..dc9a397 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -430,6 +430,76 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, } } +/* EMU0204 */ +#define snd_emu0204_ch_switch_info snd_ctl_boolean_mono_info + +static int snd_emu0204_ch_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = mixer->emu0204_ch_switch; + return 0; +} + +static int snd_emu0204_ch_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int value = ucontrol->value.integer.value[0]; + int err, changed; + unsigned char buf[2]; + + if (value > 1) + return -EINVAL; + + buf[0] = 0x01; + buf[1] = value ? 0x02 : 0x01; + + changed = value != mixer->emu0204_ch_switch; + down_read(&mixer->chip->shutdown_rwsem); + if (mixer->chip->shutdown) { + err = -ENODEV; + goto out; + } + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), 0x1, + 0x21, + 0x0400, 0x0e00, buf, 2); + out: + up_read(&mixer->chip->shutdown_rwsem); + if (err < 0) + return err; + mixer->emu0204_ch_switch = value; + return changed; +} + + +static struct snd_kcontrol_new snd_emu0204_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Route channels 3/4 to front jack", + .info = snd_emu0204_ch_switch_info, + .get = snd_emu0204_ch_switch_get, + .put = snd_emu0204_ch_switch_put, + .private_value = 0, + }, +}; + +static int snd_emu0204_controls_create(struct usb_mixer_interface *mixer) +{ + int i, err; + + for (i = 0; i < ARRAY_SIZE(snd_emu0204_controls); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_emu0204_controls[i], mixer)); + if (err < 0) + return err; + } + mixer->emu0204_ch_switch = 0; /* Default is 1/2 */ + + return 0; +} /* ASUS Xonar U1 / U3 controls */ static int snd_xonar_u1_switch_get(struct snd_kcontrol *kcontrol, @@ -1337,6 +1407,13 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) snd_audigy2nx_proc_read); break; + /* EMU0204 */ + case USB_ID(0x041e, 0x3f19): + err = snd_emu0204_controls_create(mixer); + if (err < 0) + break; + break; + case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C400 */ err = snd_c400_create_mixer(mixer); -- 1.8.3.2

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[alsa-devel] Question about WM8904 regcache
by Bo Shen 12 Jul '13

12 Jul '13

Hi Mark, I am sorry to disturb you. On our EK board, it use wm8904 codec. When I enable the audio, and use alsa utils aplay to test, I meet the following issue. The audio can not playback, it show the following error. ---8>--- #aplay 44.1k_16bit_stereo.wav Playing WAVE '44.1k_16bit_stereo.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: pcm_write:1737: write error: Input/output error ---<8--- After check the register, they are not be configured correctly. When read from debug file system, found the cache_dirty always Y before playback and also failed playback. I try to add the following two line in wm8904_set_bias_level into "case SND_SOC_BIAS_ON", it doesn't work. ---8>--- @@ -1832,9 +1834,12 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + regcache_cache_only(wm8904->regmap, false); + regcache_sync(wm8904->regmap); break; ---<8--- However, If I comment the regcache_cache_only(wm8904->regmap, true) as following, then it play OK. ---8>--- @@ -2232,7 +2239,7 @@ static int wm8904_i2c_probe(struct i2c_client *i2c, WM8904_POBCTRL, 0); /* Can leave the device powered off until we need it */ - regcache_cache_only(wm8904->regmap, true); + // regcache_cache_only(wm8904->regmap, true); ---<8--- So, any clue for this issue? Thanks. Best Regards, Bo Shen

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