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

[alsa-devel] [PATCH 1/1] conf/cards: add Audio Advantage Micro II
by Przemek Rudy 31 Jul '13

31 Jul '13

With related kernel chnages add extended suport for: - AES bits controls (via device suffix AESn= or tools like iecset) - SPDIF switch control, turning on/off the optical transmitter (via alsamixer tool) Signed-off-by: Przemek Rudy <prudy1(a)o2.pl> diff --git a/src/conf/cards/USB-Audio.conf b/src/conf/cards/USB-Audio.conf index 0d93175..8a6d9ca 100644 --- a/src/conf/cards/USB-Audio.conf +++ b/src/conf/cards/USB-Audio.conf @@ -101,6 +101,41 @@ USB-Audio."Audiophile USB (tm)".pcm.iec958 { } } +# For this card we can (and must to get IEC61937) set AES bits +USB-Audio."MicroII".pcm.iec958 "cards.USB-Audio.Audio Advantage MicroII.pcm.iec958" +USB-Audio."Audio Advantage MicroII".pcm.iec958 { + @args [ CARD AES0 AES1 AES2 AES3 ] + @args.CARD { type string } + @args.AES0 { type integer } + @args.AES1 { type integer } + @args.AES2 { type integer } + @args.AES3 { type integer } + + type hooks + slave.pcm { + type hw + card $CARD + } + + hooks.0 { + type ctl_elems + hook_args [ + { + interface PCM + name "IEC958 Playback Default" + preserve true + optional true + value [ $AES0 $AES1 $AES2 $AES3 ] + } + { + name "IEC958 Playback Switch" + lock true + preserve true + value 1 + } + ] + } +} ################################################################################ -- 1.8.3.1

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Re: [alsa-devel] [PATCH - Preliminary support for Xonar Xense (resubmitAdded support for Xonar Xense. Only speaker out and mic in working.
by Clemens Ladisch 30 Jul '13

30 Jul '13

Ian Dawes wrote: > Clemens Ladisch <clemens(a)ladisch.de> wrote: >> Ian Dawes wrote: >>> Clemens Ladisch <clemens(a)ladisch.de> wrote: >>>> Ian Dawes wrote: >>>>> >>>>> Signed-off-by: Ian Dawes <madeallup.gen(a)gmail.com> >>>> >>>> The commit message was squashed into the subject line. There's >>>> probably an empty line missing between them. >>> >>> I'm not sure what this means. >> >> Did you intend for the subject to be this long? > > Probably not. I added "resubmitted" to the subject And the "Added support ..." was supposed to be in the subject, too? > How should I rectify this? Ensure there is an empty line after the subject. Regards, Clemens

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Re: [alsa-devel] [PATCH 7/9] ASoC: tegra: move to generic DMA DT binding
by Stephen Warren 30 Jul '13

30 Jul '13

On 07/30/2013 04:12 AM, Richard Zhao wrote: > On Sat, Jul 27, 2013 at 03:58:21AM +0800, Stephen Warren wrote: >> On 07/23/2013 10:10 PM, Richard Zhao wrote: >>> - add tegra_dma_filter_data to specify dma info >>> DMA DT binding needs the device that raise dma request and dma name >>> to request a dma channel. tegra30_i2s is a special case. It should be ahub >>> device and it also has dma name that cannot handled by ASoC dmaengine code. >>> So we pass the info using filter data in snd_dmaengine_dai_dma_data. >>> - change i2s/ac97 drivers to use generic DT binding >>> - tegra30_i2s: alloc ahub tx/rx fifo at driver probe time >> >>> diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c >> >> I don't understand why the FIFO alloc/free had to move from >> startup()/shutdown() to probe()/remove(). > > The call path is: > [<c0520e38>] (dump_stack+0x78/0xbc) from [<c03bb854>] (tegra_pcm_request_chan+0x10/0x34) > [<c03bb854>] (tegra_pcm_request_chan+0x10/0x34) from [<c03b6c2c>] (dmaengine_pcm_new+0xe0/0x134) > [<c03b6c2c>] (dmaengine_pcm_new+0xe0/0x134) from [<c03b5824>] (soc_new_pcm+0x2f0/0x3cc) > [<c03b5824>] (soc_new_pcm+0x2f0/0x3cc) from [<c03aa1a0>] (soc_probe_link_dais+0x358/0x39c) > [<c03aa1a0>] (soc_probe_link_dais+0x358/0x39c) from [<c03ab334>] (snd_soc_instantiate_card+0x35c/0x7a0) > [<c03ab334>] (snd_soc_instantiate_card+0x35c/0x7a0) from [<c03ab9dc>] (snd_soc_register_card+0x264/0x330) > [<c03ab9dc>] (snd_soc_register_card+0x264/0x330) from [<c03be350>] (tegra_rt5640_probe+0xf4/0x17c) > [<c03be350>] (tegra_rt5640_probe+0xf4/0x17c) from [<c0294290>] (platform_drv_probe+0x18/0x1c) > > So I need to request dma channel at probe time. The AHUB driver should look up and cache the DMA channel information during its probe() call, and defer probe() there if necessary. The FIFO alloc/free calls should simply use the cached information for the allocated channel. That way, the changes are isolated in the AHUB driver and shouldn't require any changes in the I2S driver. The DT parsing and DMA request lookup will then happen well before the snd_soc_instantiate_card() backtrace you mention above, since it will prevent the card from being instantiated until all the ASoC components are fully probed. >> This will prevent later >> changes from making the connectivity between the AHUB FIFOs and I2S >> modules more dynamic, which is kind of the whole point of the AHUB HW >> module. > > That's because it's i2s driver to register pcm device rather not ahub. > Ideally dma channel is only bound to ahub. Long-term, this should change. However, I believe if you make the changes I mention above, this won't be an issue. >>> diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h >> >>> +#define TEGRA_DMA_NAME_MAX_LEN 10 >>> + >>> +/* >>> + * Generic DMA DT binding needs the device that raise dma request and dma name >>> + * to request a dma channel. tegra30_i2s is a special case. It should be ahub >>> + * device and it also has dma name that cannot handled by ASoC dmaengine code. >>> + * So we pass the info using filter data in snd_dmaengine_dai_dma_data. >>> + */ >>> +struct tegra_dma_filter_data { >>> + struct device *dma_dev; >>> + char dma_name[TEGRA_DMA_NAME_MAX_LEN]; >>> +}; >> >> I agree with Lars-Peter's comment that it'd be better to enhance the >> ASoC dmaengine support to allow the channel name to be specified so we >> don't need to much custom code. > > Not only dma-name but also dma client device for this case. > >> And that should use "char *" not a >> fixed-length array. > > Ok, then there' two solutions. > > struct tegra30_i2s { > ... > struct snd_dmaengine_dai_dma_data capture_dma_data; > char dma_rx_name[10]; > struct tegra_dma_filter_data filter_data_rx; > ... > struct snd_dmaengine_dai_dma_data playback_dma_data; > char dma_tx_name[10]; > struct tegra_dma_filter_data filter_data_tx; > }; > > probe: i2s->filter_data_rx.dma_name = i2s.dma_rx_name > > or > > probe: i2s->filter_data_rx.dma_name = kasprintf(GFP_KERNEL, "channel%d" ...); > remove: kfree(i2s->filter_data_rx.dma.name) > > Which one do you like? Either is fine, although the first option has less dynamic allocation, so is probably better. The AHUB driver should be constructing the channel names though.

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[alsa-devel] [PATCH 1/2] ASoC: bf5xx-ac97: Fix compile error
by Lars-Peter Clausen 30 Jul '13

30 Jul '13

If CONFIG_SND_BF5XX_HAVE_COLD_RESET is enabled building the blackfin ac97 driver fails with the following compile error: sound/soc/blackfin/bf5xx-ac97.c: In function ‘asoc_bfin_ac97_probe’: sound/soc/blackfin/bf5xx-ac97.c:297: error: expected ‘;’ before ‘{’ token sound/soc/blackfin/bf5xx-ac97.c:302: error: label ‘gpio_err’ used but not defined The issue was introduced in commit 6dab2fd7 ("ASoC: bf5xx-ac97: Convert to devm_gpio_request_one()"). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ac97.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index efb1dae..e82eb37 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -294,11 +294,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) /* Request PB3 as reset pin */ ret = devm_gpio_request_one(&pdev->dev, CONFIG_SND_BF5XX_RESET_GPIO_NUM, - GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") { + GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET"); + if (ret) { dev_err(&pdev->dev, "Failed to request GPIO_%d for reset: %d\n", CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); - goto gpio_err; + return ret; } #endif -- 1.8.0

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[alsa-devel] [PATCH v6 0/8] Sound support for at91sam9x5-wm8731 based boards
by Richard Genoud 30 Jul '13

30 Jul '13

Hi, This patchset add sound on the at91sam9x5ek board. It's based on Nicolas Ferre's work in the 2.6.39 atmel patch for sam9x5: https://github.com/linux4sam/linux-at91/commit/0aa157c9e71ccccf3abc30fa37eb… It needs Bo Chen's patchset on the generic dmaengine (topic/atmel branch on Mark's sound tree) and the SSC interrupts disabled (cf http://www.spinics.net/lists/arm-kernel/msg256901.html) As the at91 chips are on their way to the common clock framework, I thought it may be better to retrieve the codec MCLK via this framework. So it needs Boris Brezillon patch set "ARM: at91: move to common clk framework" I've separated this patch because I don't know what is better: * applying the patchset with a hard coded clock, and once at91 has moved to common clock framework, update the code and change the DT binding. * wait for the common clock framework. Outstanding issue: If this machine driver will support more SoCs maybe we should name it differently. Maybe atmel_wm8731.c ? Since it could be merged with sam9g20_wm8731.c when the g20 is DT only (There will be some more work though). It has been tested on at91sam9g35ek, and should work on g15, g25, x25 and x35 also. patches applies on next-20130729 + "ARM: at91: move to common clk framework" Changes from v5: * Update the wm8731 DT binding documentation with available audio endpoints. * Remove Uwe's signed-off since it seems to be "an integration signed-off" * Remove rate constrains if freq==0 in wm8731. * Allocate dynamically card, dai and private structures. * Add a separate patch to set the oscillator value via DT. NB: this last patch needs Boris Brezillon patch set "ARM: at91: move to common clk framework" Changes from v4: * Move the rates constraints into the codec * Hard code the DAI format since it cannot be changed anyway (and remove the DT bindings accordingly) * Add some consistency in the functions naming (sam9x5ek/at91sam9x5ek/sam9x5) * Retrieve the SSC id from the SSC controller phandle * Clarify the atmel-ssc.txt documentation * Develop audio-routing documentation * Tidy up the includes in the machine driver Changes from v3: * Remove the use of macros in device tree documentation. Changes from v2: * Change atmel,sam9x5-audio-wm8731 to atmel,sam9x5-wm8731-audio for consistency with sam9g20 machine driver. * Use the snd_pcm_hw_constraint() API as suggested by Mark and Lars-Peter. * Remove the unnecessary snd_soc_dapm_sync() call as Lars-Peter suggested. * Add documentation for machine driver DT binding. * Update atmel-ssc DT binding documentation. * Add GPL license instead of just the "GPL" word (hope that's ok !) Changes from v1-RFC: * drop patches on SSC since Bo Chen already did the work * reorder patches to have the machine driver first * drop code already handled by the ASoC framework * drop unneeded SND_ATMEL_SOC_PDC config selection * use static snd_soc_card and snd_soc_dai_link structures. Best regards, Richard. Nicolas Ferre (1): ASoC: atmel: machine driver for at91sam9x5-wm8731 boards Richard Genoud (7): ASoC: wm8731: add rates constraints Documentation: DT: update atmel SSC with DMA binding ARM: AT91: DTS: sam9x5: add SSC DMA parameters ARM: AT91: DTS: sam9x5ek: add WM8731 codec ARM: AT91: DTS: sam9x5ek: enable SSC ARM: AT91: DTS: sam9x5ek: add sound configuration ASoC: sam9x5: get codec MCLK via device tree .../devicetree/bindings/misc/atmel-ssc.txt | 23 ++- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 39 ++++ Documentation/devicetree/bindings/sound/wm8731.txt | 9 + arch/arm/boot/dts/at91sam9x5.dtsi | 3 + arch/arm/boot/dts/at91sam9x5ek.dtsi | 32 +++ sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/sam9x5_wm8731.c | 218 ++++++++++++++++++++ sound/soc/codecs/wm8731.c | 60 +++++- 9 files changed, 392 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt create mode 100644 sound/soc/atmel/sam9x5_wm8731.c -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: ASoC: uda134x: fix codec driver
by Russell King 30 Jul '13

30 Jul '13

For some reason, the DAC/ADCs are not being powered up when I try and use the UDA1341 driver; this used to work. Looking back in the git history, I don't see anything obvious which would cause this regression. However, from dumping the register writes, it seems that the codec is powered down, and nothing calls set_bias_level to wake the codec up. Moreover, this driver hasn't had DAPM support added to it, which prevents platform drivers from taking advantage of DAPMs facilities. So, let's add DAPM support to the driver. As we move the power control for the DAC/ADC into DAPM, we no longer need it in set_bias_level() - this function just becomes a way to manipulate the power control and sync the register cache with the hardware at the appropriate point. Signed-off-by: Russell King <rmk+kernel(a)arm.linux.org.uk> --- I noticed this while checking the Assabet audio, which uses the UDA1341. I've also added the corresponding support to my Assabet audio driver. If this looks like the correct thing to do to fix this, please apply. sound/soc/codecs/uda134x.c | 88 ++++++++++++++++++++++++++----------------- 1 files changed, 53 insertions(+), 35 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 6d0aa44..c94d4c1 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -325,7 +325,6 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; struct uda134x_platform_data *pd = codec->control_data; int i; u8 *cache = codec->reg_cache; @@ -334,23 +333,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - /* ADC, DAC on */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg | 0x03); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_PREPARE: /* power on */ @@ -362,23 +344,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - /* ADC, DAC power off */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03)); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_OFF: /* power off */ @@ -450,6 +415,37 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; +/* UDA1341 has the DAC/ADC power down in STATUS1 */ +static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0), +}; + +/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */ +static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0), +}; + +/* Common DAPM widgets */ +static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("VINL1"), + SND_SOC_DAPM_INPUT("VINR1"), + SND_SOC_DAPM_INPUT("VINL2"), + SND_SOC_DAPM_INPUT("VINR2"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route uda134x_dapm_routes[] = { + { "ADC", NULL, "VINL1" }, + { "ADC", NULL, "VINR1" }, + { "ADC", NULL, "VINL2" }, + { "ADC", NULL, "VINR2" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, +}; + static const struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, @@ -485,6 +481,8 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; struct uda134x_platform_data *pd = codec->card->dev->platform_data; + const struct snd_soc_dapm_widget *widgets; + unsigned num_widgets; int ret; @@ -526,6 +524,22 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) else uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pd->model == UDA134X_UDA1341) { + widgets = uda1341_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1341_dapm_widgets); + } else { + widgets = uda1340_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1340_dapm_widgets); + } + + ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets); + if (ret) { + printk(KERN_ERR "%s failed to register dapm controls: %d", + __func__, ret); + kfree(uda134x); + return ret; + } + switch (pd->model) { case UDA134X_UDA1340: case UDA134X_UDA1344: @@ -599,6 +613,10 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = { .read = uda134x_read_reg_cache, .write = uda134x_write, .set_bias_level = uda134x_set_bias_level, + .dapm_widgets = uda134x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets), + .dapm_routes = uda134x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes), }; static int uda134x_codec_probe(struct platform_device *pdev) -- 1.7.4.4

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[alsa-devel] [PATCH 1/3] ALSA: hda - Clearing jackpoll_interval avoid pending work
by Wang Xingchao 30 Jul '13

30 Jul '13

Clearing jackpoll_interval before calling cancel_delayed_work_sync(), otherwise the work will be triggered again and cause impact in hda_jackpoll_work(). The next patch will poll jack once even with jackpoll_interval=0. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_via.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e2481ba..0bc20ef 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec) return; if (spec->hp_work_active) { snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1); + codec->jackpoll_interval = 0; cancel_delayed_work_sync(&codec->jackpoll_work); spec->hp_work_active = false; - codec->jackpoll_interval = 0; } } -- 1.8.3.2

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[alsa-devel] [PATCH - microII 1/1] ALSA: usb-audio: Add Audio Advantage Micro II
by Przemek Rudy 30 Jul '13

30 Jul '13

Signed-off-by: Przemek Rudy <prudy1(a)o2.pl> diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index ebe9144..d42a584 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -9,6 +9,8 @@ * Alan Cox (alan(a)lxorguk.ukuu.org.uk) * Thomas Sailer (sailer(a)ife.ee.ethz.ch) * + * Audio Advantage Micro II support added by: + * Przemek Rudy (prudy1(a)o2.pl) * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -30,6 +32,7 @@ #include <linux/usb.h> #include <linux/usb/audio.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/control.h> #include <sound/hwdep.h> @@ -1315,6 +1318,211 @@ static struct std_mono_table ebox44_table[] = { {} }; +/* Audio Advantage Micro II findings: + * + * Mapping spdif AES bits to vendor register.bit: + * AES0: [0 0 0 0 2.3 2.2 2.1 2.0] - default 0x00 + * AES1: [3.3 3.2.3.1.3.0 2.7 2.6 2.5 2.4] - default: 0x01 + * AES2: [0 0 0 0 0 0 0 0] + * AES3: [0 0 0 0 0 0 x 0] - 'x' bit is set basing on standard usb request + * (UAC_EP_CS_ATTR_SAMPLE_RATE) for Audio Devices + * + * power on values: + * r2: 0x10 + * r3: 0x20 (b7 is zeroed just before playback (except IEC61937) and set + * just after it to 0xa0, presumably it disables/mutes some analog + * parts when there is no audio.) + * r9: 0x28 + * + * Optical transmitter on/off: + * vendor register.bit: 9.1 + * 0 - on (0x28 register value) + * 1 - off (0x2a register value) + * + */ +static int snd_microii_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_microii_spdif_default_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + struct usb_interface *iface; + struct usb_host_interface *alts; + unsigned int ep; + unsigned char data[3]; + int rate; + + ucontrol->value.iec958.status[0] = kcontrol->private_value & 0xff; + ucontrol->value.iec958.status[1] = (kcontrol->private_value >> 8) & 0xff; + ucontrol->value.iec958.status[2] = 0x00; + + /* use known values for that card: interface#1 altsetting#1 */ + iface = usb_ifnum_to_if(mixer->chip->dev, 1); + alts = &iface->altsetting[1]; + ep = get_endpoint(alts, 0)->bEndpointAddress; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_rcvctrlpipe(mixer->chip->dev, 0), + UAC_GET_CUR, + USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, + ep, + data, + sizeof(data)); + if (err < 0) + goto end; + + rate = data[0] | (data[1] << 8) | (data[2] << 16); + ucontrol->value.iec958.status[3] = (rate == 48000) ? + IEC958_AES3_CON_FS_48000 : IEC958_AES3_CON_FS_44100; + + err = 0; +end: + return err; +} + +static int snd_microii_spdif_default_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg; + unsigned long priv_backup = kcontrol->private_value; + + reg = ((ucontrol->value.iec958.status[1] & 0x0f) << 4) | + (ucontrol->value.iec958.status[0] & 0x0f); + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 2, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xfffff0f0; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0x0f) << 8; + kcontrol->private_value |= (ucontrol->value.iec958.status[0] & 0x0f); + + reg = (ucontrol->value.iec958.status[0] & IEC958_AES0_NONAUDIO) ? + 0xa0 : 0x20; + reg |= (ucontrol->value.iec958.status[1] >> 4) & 0x0f; + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 3, + NULL, + 0); + if (err < 0) + goto end; + + kcontrol->private_value &= 0xffff0fff; + kcontrol->private_value |= (ucontrol->value.iec958.status[1] & 0xf0) << 8; + + /* The frequency bits in AES3 cannot be set via register access. */ + + /* Silently ignore any bits from the request that cannot be set. */ + + err = (priv_backup != kcontrol->private_value); +end: + return err; +} + +static int snd_microii_spdif_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.iec958.status[0] = 0x0f; + ucontrol->value.iec958.status[1] = 0xff; + ucontrol->value.iec958.status[2] = 0x00; + ucontrol->value.iec958.status[3] = 0x00; + + return 0; +} + +static int snd_microii_spdif_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = !(kcontrol->private_value & 0x02); + + return 0; +} + +static int snd_microii_spdif_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_interface *mixer = snd_kcontrol_chip(kcontrol); + int err; + u8 reg = ucontrol->value.integer.value[0] ? 0x28 : 0x2a; + + err = snd_usb_ctl_msg(mixer->chip->dev, + usb_sndctrlpipe(mixer->chip->dev, 0), + UAC_SET_CUR, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + reg, + 9, + NULL, + 0); + + if (!err) { + err = (reg != (kcontrol->private_value & 0x0ff)); + if (err) + kcontrol->private_value = reg; + } + + return err; +} + +static struct snd_kcontrol_new snd_microii_mixer_spdif[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_default_get, + .put = snd_microii_spdif_default_put, + .private_value = 0x00000100UL,/* reset value */ + }, + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK), + .info = snd_microii_spdif_info, + .get = snd_microii_spdif_mask_get, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), + .info = snd_ctl_boolean_mono_info, + .get = snd_microii_spdif_switch_get, + .put = snd_microii_spdif_switch_put, + .private_value = 0x00000028UL,/* reset value */ + } +}; + +static int snd_microii_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_microii_mixer_spdif); ++i) { + err = snd_ctl_add(mixer->chip->card, + snd_ctl_new1(&snd_microii_mixer_spdif[i], mixer)); + if (err < 0) + return err; + } + + return err; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -1353,6 +1561,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) err = snd_xonar_u1_controls_create(mixer); break; + case USB_ID(0x0d8c, 0x0103): /* Audio Advantage Micro II */ + err = snd_microii_controls_create(mixer); + break; + case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ err = snd_nativeinstruments_create_mixer(mixer, snd_nativeinstruments_ta6_mixers, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8b75bcf..2be7410 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3434,4 +3434,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* + * The original product_name is "USB Sound Device", however this name + * is also used by the CM106 based cards, so make it unique. + */ + USB_DEVICE(0x0d8c, 0x0103), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .product_name = "Audio Advantage MicroII", + .ifnum = QUIRK_NO_INTERFACE + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- 1.8.1.4

3 8

[alsa-devel] [PATCH - Preliminary support for Xonar Xense (resubmitAdded support for Xonar Xense. Only speaker out and mic in working.
by Ian Dawes 29 Jul '13

29 Jul '13

Signed-off-by: Ian Dawes <madeallup.gen(a)gmail.com> diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 64b9fda..df99ec9 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8428) }, { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c index c8c7f2c..2567810 100644 --- a/sound/pci/oxygen/xonar_pcm179x.c +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -499,6 +499,17 @@ static void xonar_stx_init(struct oxygen *chip) xonar_st_init_common(chip); } +static void xonar_xense_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_st_init(chip); +} + static void xonar_d2_cleanup(struct oxygen *chip) { xonar_disable_output(chip); @@ -1138,6 +1149,14 @@ int get_xonar_pcm179x_model(struct oxygen *chip, chip->model.resume = xonar_stx_resume; chip->model.set_dac_params = set_pcm1796_params; break; + case 0x8428: + chip->model = model_xonar_st; + chip->model.shortname = "Xonar Xense"; + chip->model.chip = "AV100"; + chip->model.init = xonar_xense_init; + chip->model.dac_channels_pcm = 8; + chip->model.dac_channels_mixer = 8; + break; default: return -EINVAL; } -- 1.8.1.2

2 1

Re: [alsa-devel] snd_seq_system_info_get_cur_ports function
by Jose B 29 Jul '13

29 Jul '13

Dear Devs, I've been working on a wrapper to oversimplify using the alsa midi sequencer, and for it I was writing a function to quickly check if any new ports were registered on the sequencer after the application using the wrapper starts up. I noticed there is no function for getting the current port count but there is one for getting the current number of sequencer clients : snd_seq_system_info_get_cur_clients(const snd_seq_system_info_t *info) Shouldnt there be a corresponding function for getting the current port count as seen below? Or is it just missing from the documentation? snd_seq_system_info_get_cur_ports(const snd_seq_system_info_t *info) -- It doesn't show in code::blocks' autocomplete so i'm guessing it wasn't written yet. Is anyone willing to check this out please? It would be really handy to have this func. Thanks, and happy dev-ing Jose'

2 1