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

[alsa-devel] Fwd: regression: from 3.8 to 3.9: headphones output no sound on Intel HDA, codec VIA VT1802
by Evan 31 May '13

31 May '13

I am having this same issue (with the same hardware). I found a temporary work around for the problem by disabling "Auto-Mute Mode" with alsamixer and then muting my "Speaker" output. If I don't mute the "Speaker" I get audio through both my headphones and my speakers, but when I mute the speakers I get the audio just through the headphones. The output of alsa-info for 3.9.4 is here[1]. [1] http://www.alsa-project.org/db/?f=6120303f74b8e46930e15a544ba11b859de25c04 -E

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[alsa-devel] [PATCH 1/3] ALSA: hdspm - Add support for 128-192kHz WordClock input
by Adrian Knoth 31 May '13

31 May '13

Allow WordClock input rates of 128, 176.4 and 192kHz. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ef3cbc0..e070ea8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -400,8 +400,8 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_freq0 (1<<5) /* input freq detected via autosync */ #define HDSPM_wc_freq1 (1<<6) /* 001=32, 010==44.1, 011=48, */ -#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, */ -/* missing Bit for 111=128, 1000=176.4, 1001=192 */ +#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, 111=128 */ +#define HDSPM_wc_freq3 0x800 /* 1000=176.4, 1001=192 */ #define HDSPM_SyncRef0 0x10000 /* Sync Reference */ #define HDSPM_SyncRef1 0x20000 @@ -412,13 +412,17 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_valid (HDSPM_wcLock|HDSPM_wcSync) -#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2|\ + HDSPM_wc_freq3) #define HDSPM_wcFreq32 (HDSPM_wc_freq0) #define HDSPM_wcFreq44_1 (HDSPM_wc_freq1) #define HDSPM_wcFreq48 (HDSPM_wc_freq0|HDSPM_wc_freq1) #define HDSPM_wcFreq64 (HDSPM_wc_freq2) #define HDSPM_wcFreq88_2 (HDSPM_wc_freq0|HDSPM_wc_freq2) #define HDSPM_wcFreq96 (HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq128 (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq176_4 (HDSPM_wc_freq3) +#define HDSPM_wcFreq192 (HDSPM_wc_freq0|HDSPM_wc_freq3) #define HDSPM_status1_F_0 0x0400000 #define HDSPM_status1_F_1 0x0800000 @@ -1181,6 +1185,15 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) case HDSPM_wcFreq96: rate = 96000; break; + case HDSPM_wcFreq128: + rate = 128000; + break; + case HDSPM_wcFreq176_4: + rate = 176400; + break; + case HDSPM_wcFreq192: + rate = 192000; + break; default: rate = 0; break; -- 1.7.10.4

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[alsa-devel] [PATCH] ALSA: hda - Add headset quirk for two Dell machines
by David Henningsson 31 May '13

31 May '13

This quirk is required for the headset mic to work on these two machines. BugLink: https://bugs.launchpad.net/bugs/1186170 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59d2e91..aa4bf82 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3493,6 +3493,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.7.9.5

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[alsa-devel] [PATCH] [resend] snd-es1968: Add radio support for MediaForte M56VAP
by Ondrej Zary 31 May '13

31 May '13

Add support for TEA5757 tuner on MediaForte M56VAP sound+modem+radio card. The GPIO connection type is automatically detected (like snd-fm801 driver). Also add a safety subsystem vendor check to skip radio detection if vendor differs from ESS (so we don't touch GPIOs on laptops). Tested with SF64-PCE2 and M56VAP cards. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/es1968.c | 73 +++++++++++++++++++++++++++++++++++---------------- 1 files changed, 50 insertions(+), 23 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a1f32b5..db08641 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -564,6 +564,7 @@ struct es1968 { #ifdef CONFIG_SND_ES1968_RADIO struct v4l2_device v4l2_dev; struct snd_tea575x tea; + unsigned int tea575x_tuner; #endif }; @@ -2557,37 +2558,47 @@ static int snd_es1968_input_register(struct es1968 *chip) bits 1=unmask write to given bit */ #define IO_DIR 8 /* direction register offset from GPIO_DATA bits 0/1=read/write direction */ -/* mask bits for GPIO lines */ -#define STR_DATA 0x0040 /* GPIO6 */ -#define STR_CLK 0x0080 /* GPIO7 */ -#define STR_WREN 0x0100 /* GPIO8 */ -#define STR_MOST 0x0200 /* GPIO9 */ + +/* GPIO to TEA575x maps */ +struct snd_es1968_tea575x_gpio { + u8 data, clk, wren, most; + char *name; +}; + +static struct snd_es1968_tea575x_gpio snd_es1968_tea575x_gpios[] = { + { .data = 6, .clk = 7, .wren = 8, .most = 9, .name = "SF64-PCE2" }, + { .data = 7, .clk = 8, .wren = 6, .most = 10, .name = "M56VAP" }, +}; + +#define get_tea575x_gpio(chip) \ + (&snd_es1968_tea575x_gpios[(chip)->tea575x_tuner]) + static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); u16 val = 0; - val |= (pins & TEA575X_DATA) ? STR_DATA : 0; - val |= (pins & TEA575X_CLK) ? STR_CLK : 0; - val |= (pins & TEA575X_WREN) ? STR_WREN : 0; + val |= (pins & TEA575X_DATA) ? (1 << gpio.data) : 0; + val |= (pins & TEA575X_CLK) ? (1 << gpio.clk) : 0; + val |= (pins & TEA575X_WREN) ? (1 << gpio.wren) : 0; - outw(val, io); + outw(val, chip->io_port + GPIO_DATA); } static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = inw(io); - u8 ret; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); + u16 val = inw(chip->io_port + GPIO_DATA); + u8 ret = 0; - ret = 0; - if (val & STR_DATA) + if (val & (1 << gpio.data)) ret |= TEA575X_DATA; - if (val & STR_MOST) + if (val & (1 << gpio.most)) ret |= TEA575X_MOST; + return ret; } @@ -2596,13 +2607,18 @@ static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool outpu struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 odir = inw(io + IO_DIR); + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); if (output) { - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren)), + io + IO_MASK); + outw(odir | (1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren), + io + IO_DIR); } else { - outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); - outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.clk) | (1 << gpio.wren) | (1 << gpio.data) | (1 << gpio.most)), + io + IO_MASK); + outw((odir & ~((1 << gpio.data) | (1 << gpio.most))) + | (1 << gpio.clk) | (1 << gpio.wren), io + IO_DIR); } } @@ -2772,6 +2788,9 @@ static int snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO + /* don't play with GPIOs on laptops */ + if (chip->pci->subsystem_vendor != 0x125d) + goto no_radio; err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_es1968_free(chip); @@ -2781,10 +2800,18 @@ static int snd_es1968_create(struct snd_card *card, chip->tea.private_data = chip; chip->tea.radio_nr = radio_nr; chip->tea.ops = &snd_es1968_tea_ops; - strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) - printk(KERN_INFO "es1968: detected TEA575x radio\n"); + for (i = 0; i < ARRAY_SIZE(snd_es1968_tea575x_gpios); i++) { + chip->tea575x_tuner = i; + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { + snd_printk(KERN_INFO "es1968: detected TEA575x radio type %s\n", + get_tea575x_gpio(chip)->name); + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + sizeof(chip->tea.card)); + break; + } + } +no_radio: #endif *chip_ret = chip; -- Ondrej Zary

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[alsa-devel] [PATCH 1/1] ALSA: hda - add dock support for Thinkpad T431s
by Ebben Aries 31 May '13

31 May '13

Add a model/fixup string "lenovo-dock", for Thinkpad T431s, to allow sound in docking station. Tested on Lenovo T431s with ThinkPad Mini Dock Plus Series 3 Signed-off-by: Ebben Aries <earies(a)dscp.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59d2e91..9658faf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3530,6 +3530,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 1.7.12.1

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[alsa-devel] [PATCH] ALSA: sis7019: fix error return code in sis_chip_create()
by Wei Yongjun 31 May '13

31 May '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fix to return a negative error code in the pci_set_dma_mask() error handling case instead of 0, as done elsewhere in this function. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/pci/sis7019.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index f2639a0..e413b4e 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1341,7 +1341,8 @@ static int sis_chip_create(struct snd_card *card, if (rc) goto error_out; - if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { + rc = pci_set_dma_mask(pci, DMA_BIT_MASK(30)); + if (rc < 0) { dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA"); goto error_out_enabled; }

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Re: [alsa-devel] [PATCH] ALSA: hda - Enable runtime pm for Haswell
by Wang, Xingchao 31 May '13

31 May '13

Hi Rafael, > -----Original Message----- > From: Wysocki, Rafael J > Sent: Wednesday, May 22, 2013 7:09 PM > To: Takashi Iwai > Cc: Wang, Xingchao; Girdwood, Liam R; Lin, Mengdong; Li, Jocelyn; > alsa-devel(a)alsa-project.org; Wang Xingchao > Subject: Re: [PATCH] ALSA: hda - Enable runtime pm for Haswell > > On 5/22/2013 7:51 AM, Takashi Iwai wrote: > > At Wed, 22 May 2013 03:56:00 +0000, > > Wang, Xingchao wrote: > >> Hi, > >> > >> Add Rafael in loop. > >> > >>> -----Original Message----- > >>> From: Takashi Iwai [mailto:tiwai@suse.de] > >>> Sent: Thursday, May 16, 2013 4:49 PM > >>> To: Wang Xingchao > >>> Cc: Girdwood, Liam R; Lin, Mengdong; Li, Jocelyn; > >>> alsa-devel(a)alsa-project.org; Wang, Xingchao > >>> Subject: Re: [PATCH] ALSA: hda - Enable runtime pm for Haswell > >>> > >>> At Thu, 16 May 2013 16:29:05 +0800, > >>> Wang Xingchao wrote: > >>>> Haswell doesnot support runtime pm by default. > >>>> This patch let haswell Display HD-A controller enter runtime > >>>> suspend, and bring more power saving whith power-well. > >>>> > >>>> Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> > >>> I don't think it's good to fiddle such a thing in the driver side. > >>> If Haswell can support runtime PM really, it should be set commonly. > >> I wonder whether it's HD-A driver's policy to only support runtime PM if the > device can support signal wakup? > >> According to Rafael, the device can support runtime PM regardless, no > matter it supports PME or not. > >> If so, we should remove the "if" condition check here. > > Well, if the decision is purely a driver issue, then we can get rid of > > PME check. > > Yes, it is. > > > But in that case, it should be simply like: > > > > azx_probe() { > > ... > > if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) > > pm_runtime_put_noidle(&pci->dev); > > ... > > } > > > > azx_remove() { > > ... > > if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) > > pm_runtime_get_noresume(&pci->dev); > > ... > > } > > > > > > AFAIU, calling pm_runtime_allow() enables the runtime PM *forcibly*. > > Usually this isn't a good thing. > > > > That's correct. Moreover, calling both pm_runtime_allow() *and* > pm_runtime_put_noidle() together would be a bug IMO. It's interesting something maybe wrong in my test: 1. withtout calling pm_runtime_allow(), azx_runteim_idle/suspend() will not be called. 2. another trick is on my Haswell ULT C stepping board, the runtime PM only work after exit from resume: echo mem > /sys/power/state if you donot let system enter suspend manually, the runtime pm will not be triggered. Is there any dependency between runtime pm suspend and normal suspend? Thanks --xingchao > > Thanks, > Rafael > > > > > >>>> --- > >>>> sound/pci/hda/hda_intel.c | 6 ++++++ > >>>> 1 file changed, 6 insertions(+) > >>>> > >>>> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > >>>> index bf27693..eb25888 100644 > >>>> --- a/sound/pci/hda/hda_intel.c > >>>> +++ b/sound/pci/hda/hda_intel.c > >>>> @@ -3755,6 +3755,12 @@ static int azx_probe(struct pci_dev *pci, > >>>> > >>>> if (pci_dev_run_wake(pci)) > >>>> pm_runtime_put_noidle(&pci->dev); > >>>> + else if (chip->driver_caps > >>>> + & AZX_DCAPS_I915_POWERWELL) { > >>>> + /* Haswell doesnot support runtime pm by default */ > >>>> + pm_runtime_put_noidle(&pci->dev); > >>>> + pm_runtime_allow(&pci->dev); > >>>> + } > >>>> > >>>> dev++; > >>>> complete_all(&chip->probe_wait); > >>>> -- > >>>> 1.7.9.5 > >>>>

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[alsa-devel] [PATCH v7 00/11] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 31 May '13

31 May '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. Regards, Markus Changes in v7: - Only one soc_ac97_ops struct for both ssi drivers (fsl-ssi and imx-ssi) to make it possible to compile a kernel with both drivers. - Other small bug/style fixes described in the changelogs of the patches. Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (11): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move soc_ac97_ops from imx-ssi to fsl_ssi ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: imx ac97 support ARM: imx: Export ac97 reset functions ASoC: phycore-ac97: Add DT support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 4 +- sound/soc/fsl/fsl_ssi.c | 450 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 15 +- sound/soc/fsl/imx-ssi.c | 14 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 236 +++++++++-- 13 files changed, 656 insertions(+), 128 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

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[alsa-devel] [PATCH] alsa-utils 1.0.27.1 still ships 90-alsa-restore.rules
by Elimar Riesebieter 30 May '13

30 May '13

Hi all, [0] tells: alsactl: remove 90-alsa-restore.rules from the tar ball but it still ships 90-alsa-restore.rules.in and installs the file as 90-alsa-restore.rules [0] http://www.alsa-project.org/main/index.php/Changes_v1.0.27_v1.0.27.1 Thanks Signed-off-by: Elimar Riesebieter <riesebie(a)lxtec.de> --- alsactl/90-alsa-restore.rules.in | 8 -------- alsactl/Makefile.am | 5 ----- 2 files changed, 13 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in deleted file mode 100644 index 88e12e0..0000000 --- a/alsactl/90-alsa-restore.rules.in +++ /dev/null @@ -1,8 +0,0 @@ -ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", GOTO="alsa_restore_go" -GOTO="alsa_restore_end" - -LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{number}" - -LABEL="alsa_restore_end" diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index bac84eb..3bf8eca 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -11,8 +11,6 @@ alsactl_SOURCES=alsactl.c state.c lock.c utils.c init_parse.c daemon.c alsactl_CFLAGS=$(AM_CFLAGS) -D__USE_GNU -DSYS_ASOUNDRC=\"$(ASOUND_STATE_DIR)/asound.state\" -DSYS_PIDFILE=\"$(ALSACTL_PIDFILE_DIR)/alsactl.pid\" noinst_HEADERS=alsactl.h list.h init_sysdeps.c init_utils_string.c init_utils_run.c init_sysfs.c -udevrules_DATA = \ - 90-alsa-restore.rules if HAVE_SYSTEMD @@ -49,20 +47,17 @@ alsa-restore.service: alsa-restore.service.in alsa-store.service: alsa-store.service.in $(edit) -90-alsa-restore.rules: 90-alsa-restore.rules.in $(edit) EXTRA_DIST += \ alsa-state.service.in \ alsa-restore.service.in \ alsa-store.service.in \ - 90-alsa-restore.rules.in CLEANFILES = \ alsa-state.service \ alsa-restore.service \ alsa-store.service \ - 90-alsa-restore.rules %.7: %.xml xmlto man $?

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[alsa-devel] [PATCH 1/5] ASoC: wm8994: Add digital loopback paths
by Mark Brown 30 May '13

30 May '13

There is loopback control within the audio interfaces, provide control of this as there are some obscure scenarios where this could be used in production. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 31 +++++++++++++++++++++++++++++-- 1 file changed, 29 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 561c2f9..2081fa1 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1499,6 +1499,24 @@ static const char *aif1dac_text[] = { "AIF1DACDAT", "AIF3DACDAT", }; +static const char *loopback_text[] = { + "None", "ADCDAT", +}; + +static const struct soc_enum aif1_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif1_loopback = + SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum); + +static const struct soc_enum aif2_loopback_enum = + SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2, + loopback_text); + +static const struct snd_kcontrol_new aif2_loopback = + SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum); + static const struct soc_enum aif1dac_enum = SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); @@ -1745,6 +1763,9 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_MUX("AIF1 Loopback", SND_SOC_NOPM, 0, 0, &aif1_loopback), +SND_SOC_DAPM_MUX("AIF2 Loopback", SND_SOC_NOPM, 0, 0, &aif2_loopback), + SND_SOC_DAPM_POST("Debug log", post_ev), }; @@ -1876,9 +1897,9 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF1DAC2L", NULL, "AIF1DAC Mux" }, { "AIF1DAC2R", NULL, "AIF1DAC Mux" }, - { "AIF1DAC Mux", "AIF1DACDAT", "AIF1DACDAT" }, + { "AIF1DAC Mux", "AIF1DACDAT", "AIF1 Loopback" }, { "AIF1DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, - { "AIF2DAC Mux", "AIF2DACDAT", "AIF2DACDAT" }, + { "AIF2DAC Mux", "AIF2DACDAT", "AIF2 Loopback" }, { "AIF2DAC Mux", "AIF3DACDAT", "AIF3DACDAT" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCL" }, { "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCR" }, @@ -1929,6 +1950,12 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" }, { "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" }, + /* Loopback */ + { "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" }, + { "AIF1 Loopback", "None", "AIF1DACDAT" }, + { "AIF2 Loopback", "ADCDAT", "AIF2ADCDAT" }, + { "AIF2 Loopback", "None", "AIF2DACDAT" }, + /* Sidetone */ { "Left Sidetone", "ADC/DMIC1", "ADCL Mux" }, { "Left Sidetone", "DMIC2", "DMIC2L" }, -- 1.7.10.4

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