December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Applied "ASoC: ti: Kconfig: Remove the deprecated options" to the asoc tree
by Mark Brown 18 Dec '18

18 Dec '18

The patch ASoC: ti: Kconfig: Remove the deprecated options has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0f4967bc21de0bb9e107f9ad3ced852daf66acd2 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon, 17 Dec 2018 14:21:42 +0200 Subject: [PATCH] ASoC: ti: Kconfig: Remove the deprecated options We no longer have these options used anywhere. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Tony Lindgren <tony(a)atomide.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/ti/Kconfig | 159 ------------------------------------------- 1 file changed, 159 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 698810c770f0..4bf3c15d4e51 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -46,14 +46,10 @@ config SND_SOC_OMAP_DMIC Say Y or M here if you want to have support for DMIC IP found in OMAP4 and OMAP5. -config SND_OMAP_SOC_MCBSP - tristate - config SND_SOC_OMAP_MCBSP tristate "Multichannel Buffered Serial Port (McBSP) support" depends on ARCH_OMAP || ARCH_OMAP1 || COMPILE_TEST select SND_SOC_TI_SDMA_PCM - select SND_OMAP_SOC_MCBSP help Say Y or M here if you want to have support for McBSP IP found in Texas Instruments OMAP1/2/3/4/5 SoCs. @@ -209,160 +205,5 @@ config SND_SOC_DM365_VOICE_CODEC Say Y if you want to add support for SoC On-chip voice codec endchoice -menu "Deprecated kconfig options:" - -config SND_AM33XX_SOC_EVM - tristate "SoC Audio for the AM33XX chip based boards" - depends on SOC_AM33XX && I2C - select SND_SOC_TLV320AIC3X - select SND_SOC_DAVINCI_MCASP - help - Say Y or M if you want to add support for SoC audio on AM33XX - boards using McASP and TLV320AIC3X codec. For example AM335X-EVM, - AM335X-EVMSK, and BeagelBone with AudioCape boards have this - setup. - -config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" - depends on ARCH_DAVINCI && I2C - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM - select SND_SOC_DAVINCI_EVM - help - Say Y if you want to add support for SoC audio on TI - DaVinci DM6446, DM355 or DM365 EVM platforms. - -choice - prompt "DM365 codec select" - depends on SND_DAVINCI_SOC_EVM - depends on MACH_DAVINCI_DM365_EVM - -config SND_DM365_AIC3X_CODEC - tristate "Audio Codec - AIC3101" - select SND_SOC_DM365_AIC3X_CODEC - help - Say Y if you want to add support for AIC3101 audio codec - -config SND_DM365_VOICE_CODEC - tristate "Voice Codec - CQ93VC" - select SND_SOC_DM365_VOICE_CODEC - help - Say Y if you want to add support for SoC On-chip voice codec -endchoice - -config SND_DM6467_SOC_EVM - tristate "SoC Audio support for DaVinci DM6467 EVM" - depends on MACH_DAVINCI_DM6467_EVM && I2C - select SND_SOC_DAVINCI_EVM - - help - Say Y if you want to add support for SoC audio on TI - -config SND_DA830_SOC_EVM - tristate "SoC Audio support for DA830/OMAP-L137 EVM" - depends on MACH_DAVINCI_DA830_EVM && I2C - select SND_SOC_DAVINCI_EVM - - help - Say Y if you want to add support for SoC audio on TI - DA830/OMAP-L137 EVM - -config SND_DA850_SOC_EVM - tristate "SoC Audio support for DA850/OMAP-L138 EVM" - depends on MACH_DAVINCI_DA850_EVM && I2C - select SND_SOC_DAVINCI_EVM - help - Say Y if you want to add support for SoC audio on TI - DA850/OMAP-L138 EVM - -config SND_OMAP_SOC_HDMI_AUDIO - tristate "HDMI audio support for OMAP4+ based SoCs" - select SND_SOC_OMAP_HDMI - help - For HDMI audio to work OMAPDSS HDMI support should be - enabled. - The hdmi audio driver implements cpu-dai component using the - callbacks provided by OMAPDSS and registers the component - under DSS HDMI device. Omap-pcm is registered for platform - component also under DSS HDMI device. Dummy codec is used as - as codec component. The hdmi audio driver implements also - the card and registers it under its own platform device. - The device for the driver is registered by OMAPDSS hdmi - driver. - -config SND_OMAP_SOC_N810 - tristate "SoC Audio support for Nokia N810" - depends on MACH_NOKIA_N810 && I2C - select SND_SOC_NOKIA_N810 - help - Say Y if you want to add support for SoC audio on Nokia N810. - -config SND_OMAP_SOC_RX51 - tristate "SoC Audio support for Nokia N900 (RX-51)" - depends on ARM && I2C && GPIOLIB && ARCH_OMAP3 - select SND_SOC_NOKIA_RX51 - help - Say Y if you want to add support for SoC audio on Nokia N900 - cellphone. - -config SND_OMAP_SOC_AMS_DELTA - tristate "SoC Audio support for Amstrad E3 (Delta) videophone" - depends on MACH_AMS_DELTA && TTY - select SND_SOC_OMAP_AMS_DELTA - help - Say Y if you want to add support for SoC audio device connected to - a handset and a speakerphone found on Amstrad E3 (Delta) videophone. - - Note that in order to get those devices fully supported, you have to - build the kernel with standard serial port driver included and - configured for at least 4 ports. Then, from userspace, you must load - a line discipline #19 on the modem (ttyS3) serial line. The simplest - way to achieve this is to install util-linux-ng and use the included - ldattach utility. This can be started automatically from udev, - a simple rule like this one should do the trick (it does for me): - ACTION=="add", KERNEL=="controlC0", \ - RUN+="/usr/sbin/ldattach 19 /dev/ttyS3" - -config SND_OMAP_SOC_OSK5912 - tristate "SoC Audio support for omap osk5912" - depends on MACH_OMAP_OSK && I2C - select SND_SOC_OMAP_OSK5912 - help - Say Y if you want to add support for SoC audio on osk5912. - -config SND_OMAP_SOC_OMAP_TWL4030 - tristate "SoC Audio support for TI SoC based boards with twl4030 codec" - depends on TWL4030_CORE - select SND_SOC_OMAP3_TWL4030 - help - Say Y if you want to add support for SoC audio on TI SoC based boards - using twl4030 as c codec. This driver currently supports: - - Beagleboard or Devkit8000 - - Gumstix Overo or CompuLab CM-T35/CM-T3730 - - IGEP v2 - - OMAP3EVM - - SDP3430 - - Zoom2 - -config SND_OMAP_SOC_OMAP_ABE_TWL6040 - tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL6040_CORE && COMMON_CLK - depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST - select SND_SOC_OMAP_ABE_TWL6040 - help - Say Y if you want to add support for SoC audio on OMAP boards using - ABE and twl6040 codec. This driver currently supports: - - SDP4430/Blaze boards - - PandaBoard (4430) - - PandaBoardES (4460) - - omap5-uevm (5432) - -config SND_OMAP_SOC_OMAP3_PANDORA - tristate "SoC Audio support for OMAP3 Pandora" - depends on TWL4030_CORE - select SND_SOC_OMAP3_PANDORA - help - Say Y if you want to add support for SoC audio on the OMAP3 Pandora. -endmenu - endmenu -- 2.19.0.rc2

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[alsa-devel] [PATCH] ASoC: dma-sh7760: cleanup a debug printk
by Dan Carpenter 18 Dec '18

18 Dec '18

The intent was to print the address as a hexadecimal but there is an extra "u" in the "0x%08ulx" format specification so it is displayed as decimal. Fixes: aef3b06ac697 ("[ALSA] SH7760 ASoC support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sh/dma-sh7760.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 922fb6aa3ed1..5aee11c94f2a 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -202,7 +202,7 @@ static int camelot_prepare(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct camelot_pcm *cam = &cam_pcm_data[rtd->cpu_dai->id]; - pr_debug("PCM data: addr 0x%08ulx len %d\n", + pr_debug("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, runtime->dma_bytes); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 2.17.1

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[alsa-devel] [PATCH v2 1/5] ASoC: qcom: sdm845: Add board specific dapm widgets
by Cheng-Yi Chiang 18 Dec '18

18 Dec '18

Add board specific dapm widgets so these widgets can be used in the route. Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/sdm845.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 58593db2ab151..95d8d4422dae0 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -212,6 +212,14 @@ static int sdm845_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static const struct snd_soc_dapm_widget sdm845_snd_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + static void sdm845_add_be_ops(struct snd_soc_card *card) { struct snd_soc_dai_link *link; @@ -243,6 +251,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) goto data_alloc_fail; } + card->dapm_widgets = sdm845_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); -- 2.20.0.rc0.387.gc7a69e6b6c-goog

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[alsa-devel] [PATCH 0/9] ASoC: audio/simple card tidyup/cleanup
by Kuninori Morimoto 18 Dec '18

18 Dec '18

Hi Mark We could merged normal sound and DPCM sound on audio/simple card. This is nice, but I noticed few issues. 1) : for "reg" property issue, again. 2) - 5) : missing 1CPU-1Codec DPCM case, and cleanup for audio-graph 6) - 9) : missing 1CPU-1Codec DPCM case, and cleanup for simple-card Kuninori Morimoto (9): 1) ASoC: simple-card-utils: check "reg" property on asoc_simple_card_get_dai_id() 2) ASoC: audio-graph-card: add asoc_graph_card_get_conversion() 3) ASoC: audio-graph-card: add 1 CPU : 1 Codec support again 4) ASoC: audio-graph-card: cleanup DAI link loop method 5) ASoC: audio-graph-card: reduce naming prefix 6) ASoC: simple-card: add asoc_simple_card_get_conversion() 7) ASoC: simple-card: add 1 CPU : 1 Codec support again 8) ASoC: simple-card: cleanup DAI link loop method 9) ASoC: simple-card: reduce naming prefix sound/soc/generic/audio-graph-card.c | 461 +++++++++++++++++++--------------- sound/soc/generic/simple-card-utils.c | 14 +- sound/soc/generic/simple-card.c | 453 +++++++++++++++++++-------------- 3 files changed, 544 insertions(+), 384 deletions(-) -- 2.7.4

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[alsa-devel] PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX433FN variant with ALC294
by Wandrille RONCE 17 Dec '18

17 Dec '18

From 5a244e278c7bb626c901d668d359f47681e40647 Mon Sep 17 00:00:00 2001 From: Wandrille RONCE <w(a)ndrille.fr> Date: Mon, 17 Dec 2018 19:30:00 +0100 Subject: [PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS UX433FN variant with ALC294 --- 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 90f30e769a42..6f34f81125e0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6584,6 +6584,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x14a1, "ASUS UX533FD", ALC294_FIXUP_ASUS_SPK), + SND_PCI_QUIRK(0x1043, 0x16ce, "ASUS UX433FN", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), -- 2.19.1

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Re: [alsa-devel] [Sound-open-firmware] [PATCH v3 13/14] ASoC: SOF: Add xtensa support
by rander.wang 17 Dec '18

17 Dec '18

在 12/13/2018 2:15 PM, rander.wang 写道: > > 在 12/12/2018 8:00 AM, Pierre-Louis Bossart 写道: >> >> On 12/11/18 5:08 PM, Andy Shevchenko wrote: >>> On Tue, Dec 11, 2018 at 03:23:17PM -0600, Pierre-Louis Bossart wrote: >>>> Add common directory for xtensa architecture >>>> + for (i = 0; i <= stack_words - 4; i += 4) { >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x 0x%8.8x 0x%8.8x >>>> 0x%8.8x\n", >>>> + stack_ptr + i, stack[i], stack[i + 1], stack[i + 2], >>>> + stack[i + 3]); >>>> + } >>>> + >>>> + /* deal with any remaining words */ >>>> + switch (stack_words - i) { >>>> + case 0: >>>> + break; >>>> + case 1: >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x\n", >>>> + stack_ptr + stack_words - 1, stack[stack_words - 1]); >>>> + break; >>>> + case 2: >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x 0x%8.8x\n", >>>> + stack_ptr + stack_words - 2, stack[stack_words - 2], >>>> + stack[stack_words - 1]); >>>> + break; >>>> + case 3: >>>> + dev_err(sdev->dev, "0x%8.8x: 0x%8.8x 0x%8.8x 0x%8.8x\n", >>>> + stack_ptr + stack_words - 3, stack[stack_words - 3], >>>> + stack[stack_words - 2], stack[stack_words - 1]); >>>> + break; >>>> + default: >>>> + break; >>>> + } >>> hex_dump_to_buffer(). >> I vaguely recall there was a reason not to do it, we'll relook at this. >> > we don't want to convert a blob of data to "hex ASCII" in memory > (hex_dump_to_buffer). > > Here, we just want to dump the content of call stack, like: > > [ 3.250628] sof-audio sof-audio: 0x0049fbb0: 0x8000f2d0 0x0049fc00 > 0x6f6c6c61 0x00632e63 > [ 3.250670] sof-audio sof-audio: 0x0049fbb4: 0x00ffffff 0x0010b5d4 > 0x00000000 0x2000004c > [ 3.250710] sof-audio sof-audio: 0x0049fbb8: 0x0040d000 0x00000000 > 0x0000024c 0x2000930c > [ 3.250747] sof-audio sof-audio: 0x0049fbbc: 0x8000454c 0x0049fc30 > 0x00401e20 0x00000001 > [ 3.250787] sof-audio sof-audio: 0x0049fbc0: 0x8000454c 0x0049fc30 > 0x00401e20 0x00000001 > [ 3.250827] sof-audio sof-audio: 0x0049fbc4: 0x00000000 0x00000001 > 0x00000010 0x0049fcb8 > [ 3.250865] sof-audio sof-audio: 0x0049fbc8: 0x0040a000 0x00401e98 > 0x00000001 0x00000002 > [ 3.250889] sof-audio sof-audio: 0x0049fbcc: 0x80005ac5 0x0049fc60 > 0x00000000 0x00000000 Add more details: we use another tool to analysis the hex numbers with a ELF file to get the correct information, so we do it by ourself. Thanks! Rander >> _______________________________________________ >> Sound-open-firmware mailing list >> Sound-open-firmware(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/sound-open-firmware

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Re: [alsa-devel] [PATCH 01/21] ASoC: SOF: Intel: Add BYT, CHT and BSW DSP HW support.
by Yang, Libin 17 Dec '18

17 Dec '18

Hi Sakamoto, > >-------- Forwarded Message -------- >Subject: Re: [alsa-devel] [PATCH 01/21] ASoC: SOF: Intel: Add BYT, CHT and >BSW DSP HW support. >Date: Wed, 12 Dec 2018 13:08:48 +0900 >From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> >To: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>, >alsa-devel(a)alsa-project.org >CC: Daniel Baluta <daniel.baluta(a)gmail.com>, andriy.shevchenko(a)intel.com, >tiwai(a)suse.de, Pan Xiuli <xiuli.pan(a)linux.intel.com>, >liam.r.girdwood(a)linux.intel.com, vkoul(a)kernel.org, broonie(a)kernel.org, >sound-open-firmware(a)alsa-project.org, Rander Wang ><rander.wang(a)linux.intel.com>, Alan Cox <alan(a)linux.intel.com> > >Hi, > >I have small nitpicking. > >On 2018/12/12 6:30, Pierre-Louis Bossart wrote: >> From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> >> >> Add support for the audio DSP hardware found on Intel Baytrail, >> Cherrytrail and Braswell based devices. >> >> Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> >> Signed-off-by: Pan Xiuli <xiuli.pan(a)linux.intel.com> >> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> >> Signed-off-by: Pierre-Louis Bossart >> <pierre-louis.bossart(a)linux.intel.com> >> --- >> sound/soc/sof/intel/byt.c | 805 >+++++++++++++++++++++++++++++++++++++ >> sound/soc/sof/intel/shim.h | 159 ++++++++ >> 2 files changed, 964 insertions(+) >> create mode 100644 sound/soc/sof/intel/byt.c >> create mode 100644 sound/soc/sof/intel/shim.h .. >> diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h >> new file mode 100644 index 000000000000..2fdcb2d262f4 >> --- /dev/null >> +++ b/sound/soc/sof/intel/shim.h >> @@ -0,0 +1,159 @@ >> ... >> +extern struct snd_sof_dsp_ops sof_byt_ops; extern struct >> +snd_sof_dsp_ops sof_cht_ops; > >These two symbols are found in this patch. > >> +extern struct snd_sof_dsp_ops sof_hsw_ops; > >This symbol is added in second patch and this first patch has no reference to it. > >> +extern struct snd_sof_dsp_ops sof_bdw_ops; > >This symbol is added in third patch and this first patch has no reference to it. > >> + >> +#endif > >I think it better to add these extern declarations when adding corresponding >symbols (Of course no issues as is). > >Furthermore, if you have no plan to change contents of these symbols in >kernel run time, it's better to have 'const' qualifier to locate the symbol to >readonly section. However, user of these symbols is machine driver and you >have a plan to implement it later. They can get 'const' >when you work for arrangement of existent/new codes. > >In this meaning, it might be good to add a new member to 'struct >snd_soc_acpi_mach' for new drivers to refer to the const ops, then 'pdata' >member is used from the existent drivers. I made a patch based on your suggestion. Could you please help review to see if my understanding is right? https://github.com/thesofproject/linux/pull/457 Regards, Libin > > >Regards > >Takashi Sakamoto >_______________________________________________ >Alsa-devel mailing list >Alsa-devel(a)alsa-project.org >http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] ASoC: pcm512x: Implement the digital_mute interface
by Dimitris Papavasiliou 17 Dec '18

17 Dec '18

Clicks and pops of various volumes can be produced while the device is opened, closed, put into and taken out of standby, or reconfigured. Fix this, by implementing the digital_mute interface, so that the output is muted during such operations. Signed-off-by: Dimitris Papavasiliou <dpapavas(a)gmail.com> --- Notes: Although the datasheet isn't very specific about it, PCM512x_ANALOG_MUTE_DET is only updated once ramp-down to digital mute is complete (followed by analog mute, assuming PCM512x_ANALOG_MUTE_CTRL is left with the default value, as is now the case). I've measured the time it takes for the polling to finish and it seems consistent with that assumption, including properly tracking changes in the ramp-down parameters. Thus, polling the register ensures that the device has been completely muted before returning. When unmuting, the time required for PCM512x_ANALOG_MUTE_DET to reflect the change, doesn't depend on the ramp-up parameters (as might be expected, since analog mute is likely lifted first). Nevertheless the value is consistently substantially larger than zero, so that it seemingly reflects a particular stage in the unmuting process. It also does no harm and seems to help minimize clicks, so I've kept the delay. I've tested this patch both in my day-to-day use of my DAC for more than a month, as well as by running a script that opens the device, performs several sample rate and format changes and closes it, every one second. Without the patch, several different kinds of clicks and pops are produced. After the patch, I've run the script for several hours with the DAC being consistently virtually silent. sound/soc/codecs/pcm512x.c | 121 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/pcm512x.h | 2 + 2 files changed, 121 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index f0f2d4fd3769..6cb1653be804 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -53,6 +53,8 @@ struct pcm512x_priv { unsigned long overclock_pll; unsigned long overclock_dac; unsigned long overclock_dsp; + int mute; + struct mutex mutex; }; /* @@ -384,6 +386,61 @@ static const struct soc_enum pcm512x_veds = SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4, pcm512x_ramp_step_text); +static int pcm512x_update_mute(struct pcm512x_priv *pcm512x) +{ + return regmap_update_bits( + pcm512x->regmap, PCM512x_MUTE, PCM512x_RQML | PCM512x_RQMR, + (!!(pcm512x->mute & 0x5) << PCM512x_RQML_SHIFT) + | (!!(pcm512x->mute & 0x3) << PCM512x_RQMR_SHIFT)); +} + +static int pcm512x_digital_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + + mutex_lock(&pcm512x->mutex); + ucontrol->value.integer.value[0] = !(pcm512x->mute & 0x4); + ucontrol->value.integer.value[1] = !(pcm512x->mute & 0x2); + mutex_unlock(&pcm512x->mutex); + + return 0; +} + +static int pcm512x_digital_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + int ret, changed = 0; + + mutex_lock(&pcm512x->mutex); + + if ((pcm512x->mute & 0x4) == (ucontrol->value.integer.value[0] << 2)) { + pcm512x->mute ^= 0x4; + changed = 1; + } + if ((pcm512x->mute & 0x2) == (ucontrol->value.integer.value[1] << 1)) { + pcm512x->mute ^= 0x2; + changed = 1; + } + + if (changed) { + ret = pcm512x_update_mute(pcm512x); + if (ret != 0) { + dev_err(component->dev, + "Failed to update digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + } + + mutex_unlock(&pcm512x->mutex); + + return changed; +} + static const struct snd_kcontrol_new pcm512x_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), @@ -391,8 +448,15 @@ SOC_DOUBLE_TLV("Analogue Playback Volume", PCM512x_ANALOG_GAIN_CTRL, PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), SOC_DOUBLE_TLV("Analogue Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), -SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, - PCM512x_RQMR_SHIFT, 1, 1), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = pcm512x_digital_playback_switch_get, + .put = pcm512x_digital_playback_switch_put +}, SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), SOC_ENUM("DSP Program", pcm512x_dsp_program), @@ -1319,10 +1383,61 @@ static int pcm512x_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static int pcm512x_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + struct pcm512x_priv *pcm512x = snd_soc_component_get_drvdata(component); + int ret; + unsigned int mute_det; + + mutex_lock(&pcm512x->mutex); + + if (mute) { + pcm512x->mute |= 0x1; + ret = regmap_update_bits(pcm512x->regmap, PCM512x_MUTE, + PCM512x_RQML | PCM512x_RQMR, + PCM512x_RQML | PCM512x_RQMR); + if (ret != 0) { + dev_err(component->dev, + "Failed to set digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + + regmap_read_poll_timeout(pcm512x->regmap, + PCM512x_ANALOG_MUTE_DET, + mute_det, (mute_det & 0x3) == 0, + 200, 10000); + + mutex_unlock(&pcm512x->mutex); + } else { + pcm512x->mute &= ~0x1; + ret = pcm512x_update_mute(pcm512x); + if (ret != 0) { + dev_err(component->dev, + "Failed to update digital mute: %d\n", ret); + mutex_unlock(&pcm512x->mutex); + return ret; + } + + regmap_read_poll_timeout(pcm512x->regmap, + PCM512x_ANALOG_MUTE_DET, + mute_det, + (mute_det & 0x3) + == ((~pcm512x->mute >> 1) & 0x3), + 200, 10000); + } + + mutex_unlock(&pcm512x->mutex); + + return 0; +} + static const struct snd_soc_dai_ops pcm512x_dai_ops = { .startup = pcm512x_dai_startup, .hw_params = pcm512x_hw_params, .set_fmt = pcm512x_set_fmt, + .digital_mute = pcm512x_digital_mute, }; static struct snd_soc_dai_driver pcm512x_dai = { @@ -1388,6 +1503,8 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) if (!pcm512x) return -ENOMEM; + mutex_init(&pcm512x->mutex); + dev_set_drvdata(dev, pcm512x); pcm512x->regmap = regmap; diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index d70d9c0c2088..9dda8693498e 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -112,7 +112,9 @@ #define PCM512x_RQST_SHIFT 4 /* Page 0, Register 3 - mute */ +#define PCM512x_RQMR (1 << 0) #define PCM512x_RQMR_SHIFT 0 +#define PCM512x_RQML (1 << 4) #define PCM512x_RQML_SHIFT 4 /* Page 0, Register 4 - PLL */ -- 2.11.0

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[alsa-devel] [PATCH 0/3] Add Xilinx I2S driver
by Maruthi Srinivas Bayyavarapu 17 Dec '18

17 Dec '18

Xilinx I2S IP enables I2S audio. Each IP can either work in playback or capture mode exclusively. The driver supports both by reading the configuration in device tree. Patchset includes devicetree bindings documentation, driver and build enablement. Maruthi Srinivas Bayyavarapu (3): dt-bindings: ASoC: xlnx,i2s: Document i2s bindings ASoC: xlnx: Add i2s driver ASoC: xlnx: enable i2s driver build .../devicetree/bindings/sound/xlnx,i2s.txt | 28 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/xilinx/Kconfig | 8 + sound/soc/xilinx/Makefile | 2 + sound/soc/xilinx/xlnx_i2s.c | 185 +++++++++++++++++++++ 6 files changed, 225 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/xlnx,i2s.txt create mode 100644 sound/soc/xilinx/Kconfig create mode 100644 sound/soc/xilinx/Makefile create mode 100644 sound/soc/xilinx/xlnx_i2s.c -- 2.7.4

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[alsa-devel] ASoC: About graph node issue without 'reg' property
by Kunihiko Hayashi 17 Dec '18

17 Dec '18

Hello Morimoto-san, Mark, I found that your commit b6f3fc005a2c ("ASoC: simple-card-utils: fixup asoc_simple_card_get_dai_id() counting") replaced with of_graph_parse_endpoint(), that needs 'reg' property in each port that has an endpoint. Currently devicetrees for UniPhier SoCs have also audio graph nodes [1], and these graph nodes include an endpoint for each port. Although the port doesn't have 'reg' property, it has worked fine before. And after applying the commit, it doesn't work. It is written in Documentation/devicetree/bindings/graph.txt that common scheme is to include the 'reg' property. Maybe I think that we'd better add 'reg' properties to each port, however, there might be same matter in other SoCs, so I'm not sure whether your patch breaks the backword compatibility or our devicetrees are wrong. I'd be glad if I could hear something about this issue from you. Thank you, [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… --- Best Regards, Kunihiko Hayashi

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