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

[alsa-devel] [PATCH] alsa/usb-audio: Generic DSD detection for Thesycon-based implementations
by Yue Wang 23 Jul '18

23 Jul '18

Thesycon provides solutions to XMOS chips, and has its own device vendor id. In this patch, we use generic method to detect DSD capability of Thesycon-based UAC2 implementations in order to support a wide range of current and future devices. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..06ae3f5adf4b 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1443,6 +1443,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, */ switch (USB_ID_VENDOR(chip->usb_id)) { case 0x20b1: /* XMOS based devices */ + case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; -- 2.18.0

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[alsa-devel] [PATCH] alsa/usb-audio: Generic DSD detection for Thesycon-based implementations
by Yue Wang 23 Jul '18

23 Jul '18

Thesycon provides solutions to XMOS chips, and has its own device vendor id. In this patch, we use generic method to detect DSD capability of Thesycon-based UAC2 implementations in order to support a wide range of current and future devices. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..c51e2dee3075 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- 2.18.0

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[alsa-devel] [PATCH] alsa/usb-audio: Topping DX7s quirk for DSD interface
by Yue Wang 23 Jul '18

23 Jul '18

This patch adds the support for native DSD for Topping DX7s DAC. The product description is on http://en.tpdz.net/products_detail/productId=32.html and it's available on https://www.amazon.com/Balanced-Headphone-Amplifier-Topping-2ES9038Q2M/dp/B… The DAC uses XMOS chip but uses a special vendor id and product id. The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC hence enable native DSD playback up to DSD512 format. Signed-off-by: Yue Wang <yuleopen(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 02b6cc02767f..c51e2dee3075 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1374,6 +1374,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x152a, 0x8750): /* Topping DX7s */ case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ -- 2.18.0

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Re: [alsa-devel] [PATCH] alsa/usb-audio: Topping DX7s quirk for DSD interface
by Takashi Iwai 23 Jul '18

23 Jul '18

On Mon, 23 Jul 2018 09:25:34 +0200, <yuleopen(a)gmail.com> wrote: > > From: Yue Wang <yuleopen(a)gmail.com> > > Signed-off-by: Yue Wang <yuleopen(a)gmail.com> Please give more explanation. No commit message already indicates that something is very wrong, no matter what the change is. thanks, Takashi

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[alsa-devel] [PATCH] ALSA: core: fix unsigned int pages overflow when comapred
by He, Bo 23 Jul '18

23 Jul '18

we see the below kernel panic on stress suspend resume test in snd_malloc_sgbuf_pages(), snd_dma_alloc_pages_fallback() alloc chunk maybe larger than the left pages due to the pages alignment, which will cause the pages overflow. while (pages > 0) { ... pages -= chunk; } the patch is change the pages from unsigned int to int to fix the issue. BUG: unable to handle kernel paging request at ffff88000deb4000 IP: [<ffffffff81404fa9>] memset_erms+0x9/0x10 Call Trace: [<ffffffff818f222f>] snd_dma_alloc_pages+0xff/0x210 [<ffffffff818f23af>] snd_dma_alloc_pages_fallback+0x6f/0x90 [<ffffffff818f2b85>] snd_malloc_sgbuf_pages+0x145/0x370 [<ffffffff818f229e>] snd_dma_alloc_pages+0x16e/0x210 [<ffffffffc011930d>] hdac_ext_dma_alloc_pages+0x1d/0x40 [snd_hda_ext_core] [<ffffffffc010729a>] snd_hdac_dsp_prepare+0xca/0x1c0 [snd_hda_core] [<ffffffffc01880f9>] skl_dsp_prepare+0x99/0xf0 [snd_soc_skl] [<ffffffffc0162a7e>] bxt_load_base_firmware+0x9e/0x5c0 [snd_soc_skl_ipc] [<ffffffffc01630ec>] bxt_set_dsp_D0+0x14c/0x300 [snd_soc_skl_ipc] [<ffffffffc015f9c3>] skl_dsp_get_core+0x43/0xd0 [snd_soc_skl_ipc] [<ffffffffc015fa60>] skl_dsp_wake+0x10/0x20 [snd_soc_skl_ipc] [<ffffffffc0188e3e>] skl_resume_dsp+0x7e/0x140 [snd_soc_skl] [<ffffffffc0183c4a>] skl_resume+0xda/0x170 [snd_soc_skl] [<ffffffff81452726>] pci_pm_resume+0x76/0xe0 [<ffffffff816616da>] dpm_run_callback+0x5a/0x180 [<ffffffff81661e3c>] device_resume+0xdc/0x2c0 [<ffffffff81663818>] dpm_resume+0x118/0x310 [<ffffffff81663e11>] dpm_resume_end+0x11/0x20 [<ffffffff810f8bcc>] suspend_devices_and_enter+0x11c/0x2b0 [<ffffffff810f90bd>] pm_suspend+0x35d/0x3d0 [<ffffffff810f78a6>] state_store+0x66/0x90 [<ffffffff813f80e2>] kobj_attr_store+0x12/0x20 [<ffffffff812a37bc>] sysfs_kf_write+0x3c/0x50 [<ffffffff812a2cbd>] kernfs_fop_write+0x11d/0x1a0 [<ffffffff8121dfaa>] __vfs_write+0x3a/0x150 [<ffffffff8121f2b1>] vfs_write+0xb1/0x1a0 [<ffffffff81220898>] SyS_write+0x58/0xc0 [<ffffffff81001fca>] do_syscall_64+0x6a/0xe0 [<ffffffff81b06560>] entry_SYSCALL_64_after_swapgs+0x5d/0xd7 Signed-off-by: he, bo <bo.he(a)intel.com> Signed-off-by: zhang jun <jun.zhang(a)intel.com> --- sound/core/sgbuf.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 84fffab..33449ee 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -68,7 +68,8 @@ void *snd_malloc_sgbuf_pages(struct device *device, size_t *res_size) { struct snd_sg_buf *sgbuf; - unsigned int i, pages, chunk, maxpages; + unsigned int i, chunk, maxpages; + int pages; struct snd_dma_buffer tmpb; struct snd_sg_page *table; struct page **pgtable; -- 2.7.4

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[alsa-devel] [PATCH v6 00/10] soundwire: Add multi link support
by Sanyog Kale 23 Jul '18

23 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Fix to avoid duplicate stream state assignment - Avoid incorrect stream release after configuring Master - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch - Fix to avoid bus lock acquired twice changes in v6: - Add patch to fix acquiring bus lock twice during master release - In sdw_stream_add_master(), perform multi link check at start of function as suggested by Pierre - Minor comment fixes changes in v5: - Add patch to fix duplicate stream state assignment - fix incorrect exit which was releasing the stream after configuring master - In sdw_release_bus_lock(), release the mutex in reverse order as suggested by Takashi changes in v4: - Added changes in sdw_stream_runtime structure to track Masters added to stream - Add changes to support ml_bankswitch only if both the hardware supports and the stream is handled by multiple masters Sanyog Kale (3): Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances soundwire: Fix acquiring bus lock twice during master release Shreyas NC (6): soundwire: Initialize completion for defer messages soundwire: Fix duplicate stream state assignment soundwire: fix incorrect exit after configuring stream soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 511 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 489 insertions(+), 137 deletions(-) -- 2.13.0

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[alsa-devel] [PATCH] ASoC: wm9712: fix replace codec to component
by Marcel Ziswiler 23 Jul '18

23 Jul '18

From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Since commit 143b44845d87 ("ASoC: wm9712: replace codec to component") "wm9712-codec" got renamed to "wm9712-component", however, this change never got propagated down to the actual board/platform drivers. E.g. on Colibri T20 this lead to the following spew upon boot with sound/touch being broken: [ 2.214121] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.222137] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.344384] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.351885] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 2.668339] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 2.675811] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.208408] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.216312] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 3.235397] tegra-snd-wm9712 sound: ASoC: CODEC DAI wm9712-hifi not registered [ 3.248938] tegra-snd-wm9712 sound: snd_soc_register_card failed (-517) ... [ 14.970443] ALSA device list: [ 14.996628] No soundcards found. This commit finally fixes this again. Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> --- arch/arm/mach-pxa/tosa.c | 2 +- arch/mips/alchemy/devboards/db1300.c | 2 +- drivers/mfd/wm97xx-core.c | 6 +++--- sound/soc/au1x/db1200.c | 4 ++-- sound/soc/fsl/pcm030-audio-fabric.c | 6 +++--- sound/soc/fsl/phycore-ac97.c | 6 +++--- sound/soc/pxa/e800_wm9712.c | 4 ++-- sound/soc/pxa/em-x270.c | 4 ++-- sound/soc/pxa/palm27x.c | 4 ++-- sound/soc/pxa/tosa.c | 4 ++-- sound/soc/tegra/tegra_wm9712.c | 4 ++-- 11 files changed, 23 insertions(+), 23 deletions(-) diff --git a/arch/arm/mach-pxa/tosa.c b/arch/arm/mach-pxa/tosa.c index cb5cd8e78c94..32011e600496 100644 --- a/arch/arm/mach-pxa/tosa.c +++ b/arch/arm/mach-pxa/tosa.c @@ -848,7 +848,7 @@ static struct platform_device sharpsl_rom_device = { }; static struct platform_device wm9712_device = { - .name = "wm9712-codec", + .name = "wm9712-component", .id = -1, }; diff --git a/arch/mips/alchemy/devboards/db1300.c b/arch/mips/alchemy/devboards/db1300.c index efb318e03e0a..841a27736eb1 100644 --- a/arch/mips/alchemy/devboards/db1300.c +++ b/arch/mips/alchemy/devboards/db1300.c @@ -638,7 +638,7 @@ static struct platform_device db1300_sd0_dev = { /**********************************************************************/ static struct platform_device db1300_wm9715_dev = { - .name = "wm9712-codec", + .name = "wm9712-component", .id = 1, /* ID of PSC for AC97 audio, see asoc glue! */ }; diff --git a/drivers/mfd/wm97xx-core.c b/drivers/mfd/wm97xx-core.c index f5a8347f837f..9fec11fc0410 100644 --- a/drivers/mfd/wm97xx-core.c +++ b/drivers/mfd/wm97xx-core.c @@ -109,7 +109,7 @@ static const struct regmap_config wm9705_regmap_config = { }; static struct mfd_cell wm9705_cells[] = { - { .name = "wm9705-codec", }, + { .name = "wm9705-component", }, { .name = "wm97xx-ts", }, }; @@ -177,7 +177,7 @@ static const struct regmap_config wm9712_regmap_config = { }; static struct mfd_cell wm9712_cells[] = { - { .name = "wm9712-codec", }, + { .name = "wm9712-component", }, { .name = "wm97xx-ts", }, }; @@ -248,7 +248,7 @@ static const struct regmap_config wm9713_regmap_config = { }; static struct mfd_cell wm9713_cells[] = { - { .name = "wm9713-codec", }, + { .name = "wm9713-component", }, { .name = "wm97xx-ts", }, }; diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 301e1fc9a377..d452478d62b4 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link db1200_ac97_dai = { .codec_dai_name = "ac97-hifi", .cpu_dai_name = "au1xpsc_ac97.1", .platform_name = "au1xpsc-pcm.1", - .codec_name = "ac97-codec.1", + .codec_name = "ac97-component.1", }; static struct snd_soc_card db1200_ac97_machine = { @@ -68,7 +68,7 @@ static struct snd_soc_dai_link db1300_ac97_dai = { .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "au1xpsc_ac97.1", .platform_name = "au1xpsc-pcm.1", - .codec_name = "wm9712-codec.1", + .codec_name = "wm9712-component.1", }; static struct snd_soc_card db1300_ac97_machine = { diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index ec731223cab3..869f4b226ccd 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -33,14 +33,14 @@ static struct snd_soc_dai_link pcm030_fabric_dai[] = { .stream_name = "AC97 Analog", .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "mpc5200-psc-ac97.0", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97.1", .stream_name = "AC97 IEC958", .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; @@ -85,7 +85,7 @@ static int pcm030_fabric_probe(struct platform_device *op) if (ret) dev_err(&op->dev, "request_module returned: %d\n", ret); - pdata->codec_device = platform_device_alloc("wm9712-codec", -1); + pdata->codec_device = platform_device_alloc("wm9712-component", -1); if (!pdata->codec_device) dev_err(&op->dev, "platform_device_alloc() failed\n"); diff --git a/sound/soc/fsl/phycore-ac97.c b/sound/soc/fsl/phycore-ac97.c index 66fb6c4614d2..dbd81d3b7c2c 100644 --- a/sound/soc/fsl/phycore-ac97.c +++ b/sound/soc/fsl/phycore-ac97.c @@ -30,8 +30,8 @@ static struct snd_soc_dai_link imx_phycore_dai_ac97[] = { { .name = "HiFi", .stream_name = "HiFi", - .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_dai_name = "wm9712-hifi", + .codec_name = "wm9712-component", .cpu_dai_name = "imx-ssi.0", .platform_name = "imx-ssi.0", .ops = &imx_phycore_hifi_ops, @@ -88,7 +88,7 @@ static int __init imx_phycore_init(void) if (ret) goto fail1; - imx_phycore_snd_device = platform_device_alloc("wm9712-codec", -1); + imx_phycore_snd_device = platform_device_alloc("wm9712-component", -1); if (!imx_phycore_snd_device) { ret = -ENOMEM; goto fail2; diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 1ed8aa2348f1..937746d1e23e 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -75,7 +75,7 @@ static struct snd_soc_dai_link e800_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97 Aux", @@ -83,7 +83,7 @@ static struct snd_soc_dai_link e800_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index e046770ce70e..e958c4c241f6 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -37,7 +37,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, { .name = "AC97 Aux", @@ -45,7 +45,7 @@ static struct snd_soc_dai_link em_x270_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", }, }; diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 97167048572d..040db695270e 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -93,7 +93,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { .stream_name = "AC97 HiFi", .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .platform_name = "pxa-pcm-audio", .init = palm27x_ac97_init, }, @@ -102,7 +102,7 @@ static struct snd_soc_dai_link palm27x_dai[] = { .stream_name = "AC97 Aux", .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .platform_name = "pxa-pcm-audio", }, }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index ae9c12e1ea2a..af42417ccec5 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -189,7 +189,7 @@ static struct snd_soc_dai_link tosa_dai[] = { .cpu_dai_name = "pxa2xx-ac97", .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .ops = &tosa_ops, }, { @@ -198,7 +198,7 @@ static struct snd_soc_dai_link tosa_dai[] = { .cpu_dai_name = "pxa2xx-ac97-aux", .codec_dai_name = "wm9712-aux", .platform_name = "pxa-pcm-audio", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .ops = &tosa_ops, }, }; diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 864a3345972e..aa12be76d15a 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -53,7 +53,7 @@ static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", .codec_dai_name = "wm9712-hifi", - .codec_name = "wm9712-codec", + .codec_name = "wm9712-component", .init = tegra_wm9712_init, }; @@ -83,7 +83,7 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) card->dev = &pdev->dev; snd_soc_card_set_drvdata(card, machine); - machine->codec = platform_device_alloc("wm9712-codec", -1); + machine->codec = platform_device_alloc("wm9712-component", -1); if (!machine->codec) { dev_err(&pdev->dev, "Can't allocate wm9712 platform device\n"); return -ENOMEM; -- 2.14.4

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[alsa-devel] [PATCH] ALSA: hda/i915: Allow delayed i915 audio component binding
by Takashi Iwai 22 Jul '18

22 Jul '18

Currently HD-audio i915 audio binding doesn't support any delayed binding, and supposes that the i915 driver registers the component immediately. This has been OK, so far, but the work-in-progress change in i915 may introduce the asynchronous binding, which effectively delays the component registration. For addressing it, implement a completion to be synced with the master binding. The timeout is set to 10 seconds which should be long enough and hopefully be not too annoying if anyone boots up a debugging session with i915 KMS turned off. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is applied on top of the current for-next or topic/drm_audio_component branch of sound git tree. sound/hda/hdac_i915.c | 24 ++++++++++++++++++++++-- 1 file changed, 22 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 8f2aa8bc1185..b5282cbbe489 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -20,6 +20,8 @@ #include <sound/hda_i915.h> #include <sound/hda_register.h> +static struct completion bind_complete; + #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ @@ -97,6 +99,19 @@ static bool i915_gfx_present(void) return pci_dev_present(ids); } +static int i915_master_bind(struct device *dev, + struct drm_audio_component *acomp) +{ + complete_all(&bind_complete); + /* clear audio_ops here as it was needed only for completion call */ + acomp->audio_ops = NULL; + return 0; +} + +static const struct drm_audio_component_audio_ops i915_init_ops = { + .master_bind = i915_master_bind +}; + /** * snd_hdac_i915_init - Initialize i915 audio component * @bus: HDA core bus @@ -117,7 +132,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!i915_gfx_present()) return -ENODEV; - err = snd_hdac_acomp_init(bus, NULL, + init_completion(&bind_complete); + + err = snd_hdac_acomp_init(bus, &i915_init_ops, i915_component_master_match, sizeof(struct i915_audio_component) - sizeof(*acomp)); if (err < 0) @@ -125,8 +142,11 @@ int snd_hdac_i915_init(struct hdac_bus *bus) acomp = bus->audio_component; if (!acomp) return -ENODEV; - if (!acomp->ops) + if (!acomp->ops) { request_module("i915"); + /* 10s timeout */ + wait_for_completion_timeout(&bind_complete, 10 * 1000); + } if (!acomp->ops) { snd_hdac_acomp_exit(bus); return -ENODEV; -- 2.18.0

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[alsa-devel] [PATCH] ASoC: da7219: Allow pdata to specify a VDDIO
by Daniel Kurtz 21 Jul '18

21 Jul '18

Some systems do not have software controllable regulators driving the DA7219's supplies, nor can they use device tree to create "always-on fixed regulators" to easily pretend like they do. On these systems the call to devm_regulator_bulk_get() just creates a set of dummy registers. Calling regulator_get_voltage() on a dummy regulator just returns -EINVAL, in which case the DA7219 is always set up to use the default VDDIO voltage range of 2.5-3.6V. Provide a new device property to let such systems specify a different VDDIO if needed (e.g., 1.8V). Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- This patch tries to solve the same problem as in the following patches, but does so in a way that doesn't require registering fixed regulators in the audio machine-driver: https://patchwork.kernel.org/patch/10531097 & 10531099 https://patchwork.kernel.org/patch/10536023/ include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 19 +++++++++++++------ 2 files changed, 15 insertions(+), 6 deletions(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16f2d10ab..16ab125ad4adbf 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + u32 vddio; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8bf56d38..d8f202c57f958d 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1634,6 +1634,9 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone else pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF; + if (!device_property_read_string(dev, "dlg,vddio", &of_val32)) + pdata->vddio = of_val32; + return pdata; } @@ -1717,8 +1720,12 @@ static int da7219_handle_supplies(struct snd_soc_component *component) /* Determine VDDIO voltage provided */ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer; ret = regulator_get_voltage(vddio); + /* If regulator_get_voltage() fails, try to use vddio from pdata. */ + if (ret < 0 && da7219->pdata) + ret = da7219->pdata->vddio; if (ret < 1200000) - dev_warn(component->dev, "Invalid VDDIO voltage\n"); + dev_warn(component->dev, "Invalid VDDIO voltage: %d mV\n", + ret); else if (ret < 2800000) io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V; @@ -1872,6 +1879,11 @@ static int da7219_probe(struct snd_soc_component *component) mutex_init(&da7219->ctrl_lock); mutex_init(&da7219->pll_lock); + /* Handle DT/ACPI/Platform data */ + da7219->pdata = dev_get_platdata(component->dev); + if (!da7219->pdata) + da7219->pdata = da7219_fw_to_pdata(component); + /* Regulator configuration */ ret = da7219_handle_supplies(component); if (ret) @@ -1897,11 +1909,6 @@ static int da7219_probe(struct snd_soc_component *component) break; } - /* Handle DT/ACPI/Platform data */ - da7219->pdata = dev_get_platdata(component->dev); - if (!da7219->pdata) - da7219->pdata = da7219_fw_to_pdata(component); - da7219_handle_pdata(component); /* Check if MCLK provided */ -- 2.18.0.233.g985f88cf7e-goog

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[alsa-devel] [asoc:for-4.19 199/204] sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle'
by kbuild test robot 21 Jul '18

21 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-4.19 head: a8e43c21a8a32a3af4abc605b6ebcab039f28e00 commit: c41c2a355b86368608377eaf3df442ec0f342f1e [199/204] ASoC: meson: add tdm output driver config: openrisc-allmodconfig (attached as .config) compiler: or1k-linux-gcc (GCC) 6.0.0 20160327 (experimental) reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout c41c2a355b86368608377eaf3df442ec0f342f1e # save the attached .config to linux build tree make.cross ARCH=openrisc All errors (new ones prefixed by >>): sound/soc/meson/axg-tdm-interface.c: In function 'axg_tdm_iface_set_lrclk': >> sound/soc/meson/axg-tdm-interface.c:225:8: error: implicit declaration of function 'clk_set_duty_cycle' [-Werror=implicit-function-declaration] ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); ^~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/clk_set_duty_cycle +225 sound/soc/meson/axg-tdm-interface.c d60e4f1e Jerome Brunet 2018-07-17 188 d60e4f1e Jerome Brunet 2018-07-17 189 static int axg_tdm_iface_set_lrclk(struct snd_soc_dai *dai, d60e4f1e Jerome Brunet 2018-07-17 190 struct snd_pcm_hw_params *params) d60e4f1e Jerome Brunet 2018-07-17 191 { d60e4f1e Jerome Brunet 2018-07-17 192 struct axg_tdm_iface *iface = snd_soc_dai_get_drvdata(dai); d60e4f1e Jerome Brunet 2018-07-17 193 unsigned int ratio_num; d60e4f1e Jerome Brunet 2018-07-17 194 int ret; d60e4f1e Jerome Brunet 2018-07-17 195 d60e4f1e Jerome Brunet 2018-07-17 196 ret = clk_set_rate(iface->lrclk, params_rate(params)); d60e4f1e Jerome Brunet 2018-07-17 197 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 198 dev_err(dai->dev, "setting sample clock failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 199 return ret; d60e4f1e Jerome Brunet 2018-07-17 200 } d60e4f1e Jerome Brunet 2018-07-17 201 d60e4f1e Jerome Brunet 2018-07-17 202 switch (iface->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { d60e4f1e Jerome Brunet 2018-07-17 203 case SND_SOC_DAIFMT_I2S: d60e4f1e Jerome Brunet 2018-07-17 204 case SND_SOC_DAIFMT_LEFT_J: d60e4f1e Jerome Brunet 2018-07-17 205 case SND_SOC_DAIFMT_RIGHT_J: d60e4f1e Jerome Brunet 2018-07-17 206 /* 50% duty cycle ratio */ d60e4f1e Jerome Brunet 2018-07-17 207 ratio_num = 1; d60e4f1e Jerome Brunet 2018-07-17 208 break; d60e4f1e Jerome Brunet 2018-07-17 209 d60e4f1e Jerome Brunet 2018-07-17 210 case SND_SOC_DAIFMT_DSP_A: d60e4f1e Jerome Brunet 2018-07-17 211 case SND_SOC_DAIFMT_DSP_B: d60e4f1e Jerome Brunet 2018-07-17 212 /* d60e4f1e Jerome Brunet 2018-07-17 213 * A zero duty cycle ratio will result in setting the mininum d60e4f1e Jerome Brunet 2018-07-17 214 * ratio possible which, for this clock, is 1 cycle of the d60e4f1e Jerome Brunet 2018-07-17 215 * parent bclk clock high and the rest low, This is exactly d60e4f1e Jerome Brunet 2018-07-17 216 * what we want here. d60e4f1e Jerome Brunet 2018-07-17 217 */ d60e4f1e Jerome Brunet 2018-07-17 218 ratio_num = 0; d60e4f1e Jerome Brunet 2018-07-17 219 break; d60e4f1e Jerome Brunet 2018-07-17 220 d60e4f1e Jerome Brunet 2018-07-17 221 default: d60e4f1e Jerome Brunet 2018-07-17 222 return -EINVAL; d60e4f1e Jerome Brunet 2018-07-17 223 } d60e4f1e Jerome Brunet 2018-07-17 224 d60e4f1e Jerome Brunet 2018-07-17 @225 ret = clk_set_duty_cycle(iface->lrclk, ratio_num, 2); d60e4f1e Jerome Brunet 2018-07-17 226 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 227 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 228 "setting sample clock duty cycle failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 229 return ret; d60e4f1e Jerome Brunet 2018-07-17 230 } d60e4f1e Jerome Brunet 2018-07-17 231 d60e4f1e Jerome Brunet 2018-07-17 232 /* Set sample clock inversion */ d60e4f1e Jerome Brunet 2018-07-17 233 ret = clk_set_phase(iface->lrclk, d60e4f1e Jerome Brunet 2018-07-17 234 axg_tdm_lrclk_invert(iface->fmt) ? 180 : 0); d60e4f1e Jerome Brunet 2018-07-17 235 if (ret) { d60e4f1e Jerome Brunet 2018-07-17 236 dev_err(dai->dev, d60e4f1e Jerome Brunet 2018-07-17 237 "setting sample clock phase failed: %d\n", ret); d60e4f1e Jerome Brunet 2018-07-17 238 return ret; d60e4f1e Jerome Brunet 2018-07-17 239 } d60e4f1e Jerome Brunet 2018-07-17 240 d60e4f1e Jerome Brunet 2018-07-17 241 return 0; d60e4f1e Jerome Brunet 2018-07-17 242 } d60e4f1e Jerome Brunet 2018-07-17 243 :::::: The code at line 225 was first introduced by commit :::::: d60e4f1e4be5e2dfb55fb084b119aed094227a35 ASoC: meson: add tdm interface driver :::::: TO: Jerome Brunet <jbrunet(a)baylibre.com> :::::: CC: Mark Brown <broonie(a)kernel.org> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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