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[alsa-devel] [PATCH v3 0/4] Add SPDIF support for rockchip
by Sjoerd Simons 19 Aug '15

19 Aug '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board. Changes in v3: - Fix typos in commit message - Resent patches to a more complete list of maintainers. Changes in v2: - Remove platform: module alias as it was unused - Call MODULE_DEVICE_TABLE(of, ) right after the of match table - use rk_spdif as a prefix consistenly throughout the driver - Check return code of regmap_update and bubble it up - Sort the spdif node properties - Drop the 0x prefix from the node name - Rename the spdif@ node to sound@ Sjoerd Simons (4): ASoC: dt-bindings: add rockchip tranceiver bindings ASoC: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock .../devicetree/bindings/sound/rockchip-spdif.txt | 41 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 + arch/arm/boot/dts/rk3188.dtsi | 22 ++ sound/soc/rockchip/Kconfig | 8 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_spdif.c | 381 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 7 files changed, 536 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.5.0

3 9

[alsa-devel] [PATCH] ALSA: hda - Add dock support for Thinkpad W541 (17aa:2211)
by Rick Sherman 19 Aug '15

19 Aug '15

Tested with W541 and Ultra Dock 170w Signed-off-by: Rick Sherman <rick(a)shermdog.com> --- 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 0b9847a..39a0342 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5291,6 +5291,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.4.3

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix runtime PM unbalance
by Takashi Iwai 19 Aug '15

19 Aug '15

The fix for deadlock in PM in commit [1ee23fe07ee8: ALSA: usb-audio: Fix deadlocks at resuming] introduced a new check of in_pm flag. However, the brainless patch author evaluated it in a wrong way (logical AND instead of logical OR), thus usb_autopm_get_interface() is never called, leading to unbalance of runtime PM refcount. This patch fixes it by correcting the logic. Reported-by: Hans Yang <hansy(a)nvidia.com> Fixes: 1ee23fe07ee8 ('ALSA: usb-audio: Fix deadlocks at resuming') Cc: <stable(a)vger.kernel.org> [v3.15+] Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 1fab9778807a..0450593980fd 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -638,7 +638,7 @@ int snd_usb_autoresume(struct snd_usb_audio *chip) int err = -ENODEV; down_read(&chip->shutdown_rwsem); - if (chip->probing && chip->in_pm) + if (chip->probing || chip->in_pm) err = 0; else if (!chip->shutdown) err = usb_autopm_get_interface(chip->pm_intf); -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: wm8741: Drop misleading readable_reg callback implementation
by Axel Lin 18 Aug '15

18 Aug '15

regmap_readable() returns false if map->format.format_write is set. For .reg_bits = 7, .val_bits = 9, setting, map->format.format_write = regmap_format_7_9_write; Even current code has implemented map->readable_reg, regmap_readable() still returns false anyway. Thus drop the misleading readable_reg callback implementation. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8741.c | 21 --------------------- 1 file changed, 21 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 9723e44..de42c03 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -61,25 +61,6 @@ static const struct reg_default wm8741_reg_defaults[] = { { 32, 0x0002 }, /* R32 - ADDITONAL_CONTROL_1 */ }; -static bool wm8741_readable(struct device *dev, unsigned int reg) -{ - switch (reg) { - case WM8741_DACLLSB_ATTENUATION: - case WM8741_DACLMSB_ATTENUATION: - case WM8741_DACRLSB_ATTENUATION: - case WM8741_DACRMSB_ATTENUATION: - case WM8741_VOLUME_CONTROL: - case WM8741_FORMAT_CONTROL: - case WM8741_FILTER_CONTROL: - case WM8741_MODE_CONTROL_1: - case WM8741_MODE_CONTROL_2: - case WM8741_ADDITIONAL_CONTROL_1: - return true; - default: - return false; - } -} - static int wm8741_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, WM8741_RESET, 0); @@ -541,8 +522,6 @@ static const struct regmap_config wm8741_regmap = { .reg_defaults = wm8741_reg_defaults, .num_reg_defaults = ARRAY_SIZE(wm8741_reg_defaults), .cache_type = REGCACHE_RBTREE, - - .readable_reg = wm8741_readable, }; static int wm8741_set_pdata(struct device *dev, struct wm8741_priv *wm8741) -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: topology: Disable use from userspace
by Mark Brown 18 Aug '15

18 Aug '15

Since the topology API is still in sufficient flux for changes to be identified disable the use of the userspace ABI by adding #error statements to the code, ensuring that nobody relies on the headers as currently defined. It is expected that this change will be reverted for v4.3. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/asoc.h | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 51b8066..247c50b 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -18,6 +18,12 @@ #include <linux/types.h> #include <sound/asound.h> +#ifndef __KERNEL__ +#error This API is an early revision and not enabled in the current +#error kernel release, it will be enabled in a future kernel version +#error with incompatible changes to what is here. +#endif + /* * Maximum number of channels topology kcontrol can represent. */ -- 2.5.0

1 1

[alsa-devel] [PATCH] ASoC: tegra: Convert to managed resources
by Vaishali Thakkar 18 Aug '15

18 Aug '15

Use managed resource functions devm_clk_put and devm_snd_soc_register_component to simplify error handling. To be compatible with the change various gotos are replaced with direct returns, and unneeded labels are dropped. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/tegra/tegra20_spdif.c | 37 +++++++++++++------------------------ 1 file changed, 13 insertions(+), 24 deletions(-) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 9141477..f69b2e4 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -273,45 +273,40 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) GFP_KERNEL); if (!spdif) { dev_err(&pdev->dev, "Can't allocate tegra20_spdif\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } dev_set_drvdata(&pdev->dev, spdif); - spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); + spdif->clk_spdif_out = devm_clk_get(&pdev->dev, "spdif_out"); if (IS_ERR(spdif->clk_spdif_out)) { pr_err("Can't retrieve spdif clock\n"); ret = PTR_ERR(spdif->clk_spdif_out); - goto err; + return ret; } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; + return -ENODEV; } dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmareq) { dev_err(&pdev->dev, "No DMA resource\n"); - ret = -ENODEV; - goto err_clk_put; + return -ENODEV; } memregion = devm_request_mem_region(&pdev->dev, mem->start, resource_size(mem), DRV_NAME); if (!memregion) { dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; + return -EBUSY; } regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); if (!regs) { dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; - goto err_clk_put; + return -ENOMEM; } spdif->regmap = devm_regmap_init_mmio(&pdev->dev, regs, @@ -319,7 +314,7 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) if (IS_ERR(spdif->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); ret = PTR_ERR(spdif->regmap); - goto err_clk_put; + return ret; } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; @@ -334,8 +329,9 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_pm_disable; } - ret = snd_soc_register_component(&pdev->dev, &tegra20_spdif_component, - &tegra20_spdif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &tegra20_spdif_component, + &tegra20_spdif_dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -345,21 +341,17 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) ret = tegra_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - goto err_unregister_component; + return ret; } return 0; -err_unregister_component: - snd_soc_unregister_component(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) tegra20_spdif_runtime_suspend(&pdev->dev); err_pm_disable: pm_runtime_disable(&pdev->dev); -err_clk_put: - clk_put(spdif->clk_spdif_out); -err: + return ret; } @@ -372,9 +364,6 @@ static int tegra20_spdif_platform_remove(struct platform_device *pdev) tegra20_spdif_runtime_suspend(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - - clk_put(spdif->clk_spdif_out); return 0; } -- 1.9.1

3 8

[alsa-devel] Segmentation Fault in snd_pcm_rate_hw_free()
by Alexandru Costache 18 Aug '15

18 Aug '15

Hi, I was reproducing the crash that Valentin Corfu reported earlier with his test. Valgrind and gdb show that expand_linear_s16() is somehow corrupting the area used for memory management by libc for rate->pareas[0].addr More exactly, the line that seems to corrupt the heap in static void expand_linear_s16() is: *dst = (old_sample * old_weight + new_sample * new_weight) >> 16; As Valgrind says: ==11161== Invalid write of size 2 ==11161== at 0x40A8A82: linear_expand_s16 (pcm_rate_linear.c:179) ==11161== by 0x40A85A9: linear_convert (pcm_rate_linear.c:320) ==11161== by 0x40A6915: do_convert (pcm_rate.c:537) ==11161== by 0x40A6C7D: snd_pcm_rate_write_areas1 (pcm_rate.c:550) ==11161== by 0x40A6C7D: snd_pcm_rate_commit_area (pcm_rate.c:749) ==11161== by 0x40A7014: snd_pcm_rate_drain (pcm_rate.c:1105) ==11161== by 0x4084511: snd_pcm_drain (pcm.c:1122) ==11161== by 0x8049288: main (in /root/pcm_min_daisuke) ==11161== Address 0x43d6258 is 0 bytes after a block of size 86,016 alloc'd ==11161== at 0x40261B1: malloc (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==11161== by 0x4099801: snd_pcm_mmap (pcm_mmap.c:425) ==11161== by 0x4093CA7: sndrv_pcm_hw_params (pcm_params.c:2366) ==11161== by 0x409A59C: snd1_pcm_generic_hw_params (pcm_generic.c:104) ==11161== by 0x4091E7F: snd1_pcm_hw_params_slave (pcm_params.c:2250) ==11161== by 0x40A54AC: snd_pcm_rate_hw_params (pcm_rate.c:246) ==11161== by 0x4093AE3: sndrv_pcm_hw_params (pcm_params.c:2326) ==11161== by 0x40ABC37: snd_pcm_plug_hw_params (pcm_plug.c:1045) ==11161== by 0x4093AE3: sndrv_pcm_hw_params (pcm_params.c:2326) ==11161== by 0x4084281: snd_pcm_hw_params (pcm.c:830) ==11161== by 0x8048F84: main (in /root/pcm_min_daisuke) ==11161== ==11161== Invalid free() / delete / delete[] / realloc() ==11161== at 0x402726D: free (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==11161== by 0x40A53C0: snd_pcm_rate_hw_free (pcm_rate.c:344) ==11161== by 0x4083CDA: snd_pcm_hw_free (pcm.c:853) ==11161== by 0x40A9E8D: snd_pcm_plug_hw_free (pcm_plug.c:1061) ==11161== by 0x4083CDA: snd_pcm_hw_free (pcm.c:853) ==11161== by 0x80492AC: main (in /root/pcm_min_daisuke) ==11161== Address 0xe705004b is not stack'd, malloc'd or (recently) free'd I'm not very familiar with alsa in general and was wondering if someone who knows more on it can drop a look on this one? Thank you, Alexandru

2 1

[alsa-devel] [PATCH v4 1/4] drm/i915: Add audio sync_audio_rate callback
by libin.yang＠intel.com 18 Aug '15

18 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the sync_audio_rate callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS or N/M based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..aabebcb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,7 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*sync_audio_rate)(struct device *, int port, int rate); } *ops; }; -- 1.9.1

1 3

[alsa-devel] [PATCH] ASoC: topology: Add Kconfig option for topology
by Mark Brown 18 Aug '15

18 Aug '15

Allow the topology code to be compiled out so that users who don't need topology don't need to havve the code compiled in, saving them some memory. Some more configuration could be added to remove some of the hooks into the core data structures but that is probably best done with some refactoring to use functions to do the updates of the data structures rather than ifdefing in the code as we'd need to do at the minute. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-topology.h | 12 ++++++++++++ sound/soc/Kconfig | 3 +++ sound/soc/Makefile | 3 +++ 3 files changed, 18 insertions(+) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 865a141..427bc41 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -141,6 +141,8 @@ struct snd_soc_tplg_ops { int io_ops_count; }; +#ifdef CONFIG_SND_SOC_TOPOLOGY + /* gets a pointer to data from the firmware block header */ static inline const void *snd_soc_tplg_get_data(struct snd_soc_tplg_hdr *hdr) { @@ -165,4 +167,14 @@ int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, const struct snd_soc_tplg_widget_events *events, int num_events, u16 event_type); +#else + +static inline int snd_soc_tplg_component_remove(struct snd_soc_component *comp, + u32 index) +{ + return 0; +} + +#endif + #endif diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 467ee1a..225bfda 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -30,6 +30,9 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_TOPOLOGY + bool + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 19fd711..134aca1 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,9 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o + +ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) snd-soc-core-objs += soc-topology.o +endif ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o -- 2.5.0

2 4

[alsa-devel] [PATCH v2 00/10] ASoC: support adding PCM dynamically from topology
by mengdong.lin＠intel.com 18 Aug '15

18 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This series allows the topology core to create PCM devices dynamically. The user can define different DAIs in the topology files for different versions of firmware, but share a generic platform and machine driver. A dummy DAI and DAI link can be used to register the soc card and specify the platform with topology. Then real DAIs are created in platform probing phase by the topology core, and the machine driver will be notified to create relavant DAI links. We tested using topology to create FE DAI/DAI links on Broadwell. Mengdong Lin (10): ASoC: Change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: Change 2nd argument of soc_bind_dai_link() to DAI link pointer ASoC: Implement DAI links in a list ASoC: Add support for dummy DAI links and PCM runtimes ASoC: Bind new DAI links after probing components ASoC: Support adding a DAI dynamically ASoC: topology: Change pass number of DAI smaller than graph ASoC: topology: Change stream formats to bitwise flag ASOC: topology: Add PCM DAIs dynamically when loading them include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 1 - include/sound/soc.h | 28 +- include/uapi/sound/asoc.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 10 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/soc-core.c | 642 +++++++++++++++++++-------- sound/soc/soc-dapm.c | 7 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 99 ++++- sound/soc/tegra/tegra_wm8903.c | 3 +- 22 files changed, 701 insertions(+), 291 deletions(-) -- 1.9.1

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Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel driver
by Takashi Iwai 18 Aug '15

18 Aug '15

On Tue, 18 Aug 2015 01:03:31 +0200, Paulo Roberto de Oliveira Castro wrote: > > Hello again, > > I am using Arch now, and today kernel 4.1.5 was released on Arch's repo. > However, the sound isn't working yet. Am I missing something? The patch is included in 4.1.5. Make sure that you tested really properly. If not, check again whether 4.2 works in the same condition. Takashi > > 2015-07-29 14:40 GMT-03:00 Takashi Iwai <tiwai(a)suse.de>: > > > On Wed, 29 Jul 2015 18:48:29 +0200, > > Paulo Roberto de Oliveira Castro wrote: > > > > > > Thanks! Is there a way to track these changes? > > > > Track git tree :) > > > > At first, it's found in my sound git tree, then it'll be merged to > > Linus git tree, and then to each stable tree. > > > > > > Takashi > > > > > > > > 2015-07-29 4:08 GMT-03:00 Takashi Iwai <tiwai(a)suse.de>: > > > > > > > On Wed, 29 Jul 2015 00:32:08 +0200, > > > > Paulo Roberto de Oliveira Castro wrote: > > > > > > > > > > I'm not an expert on applying patches to the kernel (never did that > > > > > before), therefore, I don't know if I did everything right. > > > > > > > > > > What I did was: downloaded 4.2-rc4 from kernel.org[1], added the > > line by > > > > > hand (couldn't apply the patch using patch), copied the config file > > from > > > > > /boot/ (the config fedora 22 uses for its kernel) to .config on the > > root > > > > of > > > > > the kernel directory, runned "make oldconfig" (typing ENTER for every > > > > > question, is that right?), "make", "make modules_install", "make > > > > install". > > > > > Rebooted, and chose the new kernel I just installed. > > > > > > > > > > The sound worked like a charm, the speakers worked as expected, and > > the > > > > > volume was as loud as on Windows. > > > > > > > > Good to hear, I queued the fix now. > > > > > > > > > Is the patch going to be applied on master? Any chance that fix > > reaches > > > > my > > > > > Fedora 22 (with backport to older kernel versions)? > > > > > > > > The commit is with Cc to stable, so it'll be backported to stable > > > > kernels once when landed to upstream. Then Fedora will pick it up > > > > eventually later. > > > > > > > > > > > > thanks, > > > > > > > > Takashi > > > > > > > > > > > > > > Thanks for the help, > > > > > > > > > > Paulo > > > > > > > > > > [1] > > > > > > > > > > > https://www.kernel.org/pub/linux/kernel/v4.x/testing/linux-4.2-rc4.tar.xz > > > > > > > > > > 2015-07-28 10:49 GMT-03:00 Paulo Roberto de Oliveira Castro < > > > > > p.oliveira.castro(a)gmail.com>: > > > > > > > > > > > > > > > > > > > > > > > ---------- Forwarded message --------- > > > > > > From: Takashi Iwai <tiwai(a)suse.de> > > > > > > Date: Tue, Jul 28, 2015, 03:35 > > > > > > Subject: Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) > > > > kernel > > > > > > driver > > > > > > To: Paulo Roberto de Oliveira Castro <p.oliveira.castro(a)gmail.com> > > > > > > Cc: <alsa-devel(a)alsa-project.org> > > > > > > > > > > > > > > > > > > On Tue, 28 Jul 2015 05:35:34 +0200, > > > > > > Paulo Roberto de Oliveira Castro wrote: > > > > > > > > > > > > > > Hi everyone, > > > > > > > > > > > > > > After installing Fedora 22 on my Dell Vostro 5480, the speakers > > don't > > > > > > work, > > > > > > > and the sound only comes out of the subwoofer. Here is the > > output of > > > > > > > alsa-info.sh. After talking to "debianuser" on IRC, we found this > > > > > > > workaround: > > > > > > > > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_PIN_WIDGET_CONTROL 0x40 # > > > > enable, > > > > > > > set it to OUT > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_CONNECT_SEL 0 # switch to > > 0x0c > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0xa000 # > > unmute > > > > left > > > > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0x9000 # > > unmute > > > > > > right > > > > > > > > > > > > > > We suspect that it's a kernel driver bug. This makes the speakers > > > > work > > > > > > > alongside the headphones (both at the same time) but I need help > > > > making > > > > > > it > > > > > > > working when headphones are not plugged in. > > > > > > > > > > > > Does the patch below work? Try with 4.2-rc kernel at best. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > --- > > > > > > diff --git a/sound/pci/hda/patch_realtek.c > > > > b/sound/pci/hda/patch_realtek.c > > > > > > index 4ae877c3b6a1..18ae17ebb356 100644 > > > > > > --- a/sound/pci/hda/patch_realtek.c > > > > > > +++ b/sound/pci/hda/patch_realtek.c > > > > > > @@ -5185,6 +5185,7 @@ static const struct snd_pci_quirk > > > > alc269_fixup_tbl[] > > > > > > = { > > > > > > SND_PCI_QUIRK(0x1028, 0x064a, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x064b, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", > > > > > > ALC288_FIXUP_DELL_XPS_13), > > > > > > + SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", > > > > > > ALC290_FIXUP_SUBWOOFER_HSJACK), > > > > > > SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", > > > > > > ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > SND_PCI_QUIRK(0x1028, 0x06da, "Dell", > > > > > > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > > > > > > > > > > > _______________________________________________ > > > > > Alsa-devel mailing list > > > > > Alsa-devel(a)alsa-project.org > > > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > > > > > > > > > [2 <text/html; UTF-8 (quoted-printable)>] > > > > >

1 0

[alsa-devel] [PATCH] ASoC: davinci-vcif: Use devm_snd_soc_register_component
by Vaishali Thakkar 17 Aug '15

17 Aug '15

Use resource managed function devm_snd_soc_register_component for component registration instead of snd_soc_register_component. Also, remove davinci_vcif_remove as it is now redundant. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/davinci/davinci-vcif.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index fabd05f..c77d921 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -231,8 +231,9 @@ static int davinci_vcif_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, davinci_vcif_dev); - ret = snd_soc_register_component(&pdev->dev, &davinci_vcif_component, - &davinci_vcif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &davinci_vcif_component, + &davinci_vcif_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "could not register dai\n"); return ret; @@ -241,23 +242,14 @@ static int davinci_vcif_probe(struct platform_device *pdev) ret = edma_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - snd_soc_unregister_component(&pdev->dev); return ret; } return 0; } -static int davinci_vcif_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; -} - static struct platform_driver davinci_vcif_driver = { .probe = davinci_vcif_probe, - .remove = davinci_vcif_remove, .driver = { .name = "davinci-vcif", }, -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: mediatek: Remove AIF widgets for backend DAIs
by Koro Chen 17 Aug '15

17 Aug '15

DAPM core already creates widgets for DAIs. It is not necessary to declare them by SND_SOC_DAPM_AIF_IN/SND_SOC_DAPM_AIF_OUT. Furthermore, original codes use backend DAI's stream name to be the AIF widget name. It causes the same widget to be created twice, and after commit 92fa12426741 ("ASoC: dapm: Add new widgets to the end of the widget list") the first created widget (by snd_soc_dapm_new_controls) is used, not the 2nd created one (by snd_soc_dapm_new_dai_widgets), so audio path is broken. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 11 ----------- 1 file changed, 11 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 55471ee..d190fe0 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -897,10 +897,6 @@ static const struct snd_kcontrol_new mtk_afe_o10_mix[] = { }; static const struct snd_soc_dapm_widget mtk_afe_pcm_widgets[] = { - /* Backend DAIs */ - SND_SOC_DAPM_AIF_IN("I2S Capture", NULL, 0, SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_AIF_OUT("I2S Playback", NULL, 0, SND_SOC_NOPM, 0, 0), - /* inter-connections */ SND_SOC_DAPM_MIXER("I05", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("I06", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -932,11 +928,6 @@ static const struct snd_soc_dapm_route mtk_afe_pcm_routes[] = { { "O10", "I18 Switch", "I18" }, }; -static const struct snd_soc_dapm_widget mtk_afe_hdmi_widgets[] = { - /* Backend DAIs */ - SND_SOC_DAPM_AIF_OUT("HDMIO Playback", NULL, 0, SND_SOC_NOPM, 0, 0), -}; - static const struct snd_soc_dapm_route mtk_afe_hdmi_routes[] = { {"HDMIO Playback", NULL, "HDMI"}, }; @@ -951,8 +942,6 @@ static const struct snd_soc_component_driver mtk_afe_pcm_dai_component = { static const struct snd_soc_component_driver mtk_afe_hdmi_dai_component = { .name = "mtk-afe-hdmi-dai", - .dapm_widgets = mtk_afe_hdmi_widgets, - .num_dapm_widgets = ARRAY_SIZE(mtk_afe_hdmi_widgets), .dapm_routes = mtk_afe_hdmi_routes, .num_dapm_routes = ARRAY_SIZE(mtk_afe_hdmi_routes), }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH RFC v3 0/7] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 17 Aug '15

17 Aug '15

This is my third RFC patch series on the subject. This time also the tda998x driver patches are a serious attempt to come up with an initial implementation for HDMI audio ASoC support. Russell's "drm/edid: add function to help find SADs" is here just to produce a working patch-set on top of mainline. It should be getting in trough drm-next at some point. The binding for tda998x is taken from Jean Francois' patch series[1] on the same subject. The implementation of the of-node parsing has some minor changes from my self. Here is what I think at least could or should still be done, but non of that stuff does not sounds critical right now. Missing from tda998x driver side - hdmi_codec_ops audio_startup() implementation for audio abort support - multi channel audio support (I would need specs and preferably some HW to test for this). Missing from ASoC side generic implementation: - channel_allocation handling is completely left for the video side driver, see if ASoC side could help in any way - snd_soc_jack functionality to handle hdmi cable plug/unplug events [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095596.html Best regards, Jyri Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports Jyri Sarha (5): ASoC: hdmi: Remove obsolete dummy HDMI codec ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders drm/i2c: tda998x: Remove include/sound/tda998x.h and fix graph parsing drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality ARM: dts: am335x-boneblack: Add HDMI audio support Russell King - ARM Linux (1): drm/edid: add function to help find SADs, DO NOT MERGE .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++ arch/arm/boot/dts/am335x-boneblack.dts | 90 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 336 +++++++++++++-- include/drm/drm_edid.h | 19 + include/sound/hdmi-codec.h | 99 +++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/hdmi-codec.c | 467 +++++++++++++++++++++ sound/soc/codecs/hdmi.c | 109 ----- 10 files changed, 1026 insertions(+), 153 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c delete mode 100644 sound/soc/codecs/hdmi.c -- 1.9.1

3 20

[alsa-devel] [PATCH v2] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 17 Aug '15

17 Aug '15

V2: Update after comments from Lars-Peter yesterday: set rate to CONTINUOUS and let ALSA figure out the details, also removing hw_params as it is then not needed. Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../devicetree/bindings/sound/ics43432.txt | 17 ++++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 101 ++++++++++++++++++++ 5 files changed, 125 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..fe0af4a --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,101 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rate_min = ICS43432_RATE_MIN, + .rate_max = ICS43432_RATE_MAX, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPL2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 5

[alsa-devel] [PATCH v2] ASoC: ssm2518: Add explicit device tree support
by Ricard Wanderlof 17 Aug '15

17 Aug '15

Add OF match table to SSM2518 to allow direct matching without going through I2C subsystem. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/ssm2518.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 67ea55a..2d199b1 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -811,6 +811,14 @@ static int ssm2518_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id ssm2518_dt_ids[] = { + { .compatible = "adi,ssm2518", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); +#endif + static const struct i2c_device_id ssm2518_i2c_ids[] = { { "ssm2518", 0 }, { } @@ -821,6 +829,7 @@ static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ssm2518_dt_ids), }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 5

Re: [alsa-devel] probe issue for Dice/OXFW drivers
by Clemens Ladisch 17 Aug '15

17 Aug '15

Takashi Sakamoto wrote: > Recently, I received an issue from a user of Mackie Onyx-i series. > According to the user, snd-dice is not applied to his Dice model. > [...] According to the log, I cannot find the first > transaction which snd-dice performs. Additionally, no error messages in > syslog. There are some checks performed before the first actual transaction. The kernel loads the driver if an entry in dice_id_table[] matches, i.e., if version==1. This values indeed is in the unit directory: > ROM header and bus information block > ----------------------------------------------------------------- > 400 0404f507 bus_info_length 4, crc_length 4, crc 62727 > 404 31333934 bus_name "1394" > 408 e0008102 irmc 1, cmc 1, isc 1, bmc 0, cyc_clk_acc 0, max_rec 8 (512) > 40c 000ff210 company_id 000ff2 | > 410 01800af7 device_id 1001800af7 | EUI-64 000ff21001800af7 > > root directory > ----------------------------------------------------------------- > 414 000639d3 directory_length 6, crc 14803 > 418 03000ff2 vendor > 41c 8100000a --> descriptor leaf at 444 > 420 17000006 model > 424 8100000d --> descriptor leaf at 458 > 428 0c0087c0 node capabilities per IEEE 1394 > 42c d1000001 --> unit directory at 430 > > unit directory at 430 > ----------------------------------------------------------------- > 430 00045246 directory_length 4, crc 21062 > 434 12000ff2 specifier id > 438 13000001 version > 43c 17000006 model > 440 8100000b --> descriptor leaf at 46c The dice_interface_check() function then makes other checks: /* * Check that GUID and unit directory are constructed according to DICE * rules, i.e., that the specifier ID is the GUID's OUI, and that the * GUID chip ID consists of the 8-bit category ID, the 10-bit product * ID, and a 22-bit serial number. */ These values are correct, except for the category ID. The driver already has an category ID exception for Weiss devices. It would be possible to add another exception, but given that firmware writers apparently like to change this value without good reason, it might be a better idea to just remove the category ID check. Regards, Clemens

2 1

[alsa-devel] [PATCH 00/25] ALSA: support AMDTP variants
by Takashi Sakamoto 17 Aug '15

17 Aug '15

This patchset updates my previous RFC, excluding control/status message for TASCAM FireWire series and a MOTU driver, [alsa-devel] [RFC][PATCH 00/37] ALSA: firewire: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/094789.html Unfortunately, I've got no dedicated testers to this patchset. While, the supported models have little time to live. I think it's time to merge them. This patchset supports playbacking/capturing PCM samples/MIDI messages. Model specific functionalities such as MIDI mapping to physical controls should be done in userspace, with propper asynchronous transactions. I hope users to remind that my work is done by reasoning and assuming with a result of investigating packets on IEEE 1394 bus. There's no published specification, thus these drivers cannot fully satisfies users' expectation which comes from Windows or OS X. Changes: * Fix minor bugs, mainly for asynchronous transaction helper for MIDI port. * Regain PCM hardware constraint (I missed it). * Fix TASCAM quirk for MIDI transactions, as much as possible. Rest of work (but may not be done in this developing period): * Handling control/status messages for TASCAM driver. * Supporting MIDI exclusive message for outgoing packets. * Researching a reason to de-synchronize for Digi00x family. * Improving MIDI throttle for asynchronous transaction helper Takashi Sakamoto (25): ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' for functional separation ALSA: firewire-lib: functional separation for packet transmission layer and data processing layer ALSA: firewire-lib: add helper functions for asynchronous MIDI port ALSA: firewire-lib: add a restriction for a transaction at once ALSA: firewire-lib: schedule tasklet again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error ALSA: firewire-digi00x: add skeleton for Digi 002/003 family ALSA: firewire-digi00x: add data processing layer ALSA: firewire-digi00x: add stream functionality ALSA: firewire-digi00x: add proc node to show clock status ALSA: firewire-digi00x: add PCM functionality ALSA: firewire-digi00x: add MIDI functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging ALSA: firewire-digi00x: add support for MIDI ports for machine control ALSA: firewire-tascam: add skeleton for TASCAM FireWire series ALSA: firewire-tascam: add a structure for model-dependent parameters. ALSA: firewire-tascam: add proc node to show firmware information ALSA: firewire-tascam: add data processing layer ALSA: firewire-tascam: add streaming functionality ALSA: firewire-tascam: add PCM functionality ALSA: firewire-tascam: add transaction functionality ALSA: firewire-tascam: add MIDI functionality ALSA: firewire-tascam: add hwdep interface include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 9 + sound/firewire/Kconfig | 27 ++ sound/firewire/Makefile | 4 +- sound/firewire/amdtp-am824.c | 451 ++++++++++++++++++++++++ sound/firewire/amdtp-am824.h | 47 +++ sound/firewire/{amdtp.c => amdtp-stream.c} | 375 ++++---------------- sound/firewire/{amdtp.h => amdtp-stream.h} | 113 ++---- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_midi.c | 16 +- sound/firewire/bebob/bebob_pcm.c | 12 +- sound/firewire/bebob/bebob_stream.c | 34 +- sound/firewire/dice/dice-midi.c | 16 +- sound/firewire/dice/dice-pcm.c | 12 +- sound/firewire/dice/dice-stream.c | 41 ++- sound/firewire/dice/dice.h | 2 +- sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 436 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++ sound/firewire/digi00x/digi00x-midi.c | 198 +++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 353 +++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 ++++++ sound/firewire/digi00x/digi00x-stream.c | 422 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 137 ++++++++ sound/firewire/digi00x/digi00x.c | 174 ++++++++++ sound/firewire/digi00x/digi00x.h | 157 +++++++++ sound/firewire/fcp.c | 2 +- sound/firewire/fireworks/fireworks.c | 12 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_midi.c | 16 +- sound/firewire/fireworks/fireworks_pcm.c | 10 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/lib.c | 129 +++++++ sound/firewire/lib.h | 52 +++ sound/firewire/oxfw/oxfw-midi.c | 16 +- sound/firewire/oxfw/oxfw-pcm.c | 10 +- sound/firewire/oxfw/oxfw-stream.c | 11 +- sound/firewire/oxfw/oxfw.h | 2 +- sound/firewire/tascam/Makefile | 4 + sound/firewire/tascam/amdtp-tascam.c | 248 +++++++++++++ sound/firewire/tascam/tascam-hwdep.c | 202 +++++++++++ sound/firewire/tascam/tascam-midi.c | 162 +++++++++ sound/firewire/tascam/tascam-pcm.c | 298 ++++++++++++++++ sound/firewire/tascam/tascam-proc.c | 88 +++++ sound/firewire/tascam/tascam-stream.c | 497 +++++++++++++++++++++++++++ sound/firewire/tascam/tascam-transaction.c | 291 ++++++++++++++++ sound/firewire/tascam/tascam.c | 230 +++++++++++++ sound/firewire/tascam/tascam.h | 140 ++++++++ 48 files changed, 5277 insertions(+), 498 deletions(-) create mode 100644 sound/firewire/amdtp-am824.c create mode 100644 sound/firewire/amdtp-am824.h rename sound/firewire/{amdtp.c => amdtp-stream.c} (70%) rename sound/firewire/{amdtp.h => amdtp-stream.h} (77%) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/amdtp-dot.c create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-midi.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x-transaction.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h create mode 100644 sound/firewire/tascam/Makefile create mode 100644 sound/firewire/tascam/amdtp-tascam.c create mode 100644 sound/firewire/tascam/tascam-hwdep.c create mode 100644 sound/firewire/tascam/tascam-midi.c create mode 100644 sound/firewire/tascam/tascam-pcm.c create mode 100644 sound/firewire/tascam/tascam-proc.c create mode 100644 sound/firewire/tascam/tascam-stream.c create mode 100644 sound/firewire/tascam/tascam-transaction.c create mode 100644 sound/firewire/tascam/tascam.c create mode 100644 sound/firewire/tascam/tascam.h -- 2.1.4

3 37

[alsa-devel] [PATCH 0/4] Minor cleanups to hda code
by Takashi Iwai 17 Aug '15

17 Aug '15

Hi, this is a series of trivial cleanup patches for HDA driver, mostly for procfs code. Takashi Iwai (4): ALSA: hda - Make some helper functions local ALSA: hda/proc - Add const to possible places ALSA: hda/proc - Fix racy string access for power states ALSA: hda/eld - Add const to possible places sound/pci/hda/hda_codec.c | 70 ------------------------------ sound/pci/hda/hda_codec.h | 7 --- sound/pci/hda/hda_eld.c | 10 ++--- sound/pci/hda/hda_proc.c | 107 +++++++++++++++++++++++++++++++++++----------- 4 files changed, 86 insertions(+), 108 deletions(-) -- 2.5.0

1 4

[alsa-devel] [PATCH] ASoC: topology: Add support for TLV bytes control
by mengdong.lin＠intel.com 17 Aug '15

17 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Allow vendor drivers to define bespoke bytes ext handlers and IDs for TLV bytes controls. And the topology core will bind these handlers by matching IDs defined by the vendor driver and user space topology data file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 865a141..3e412c6 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -89,6 +89,13 @@ struct snd_soc_tplg_kcontrol_ops { struct snd_ctl_elem_info *uinfo); }; +/* Bytes ext operations, for TLV byte controls */ +struct snd_soc_tplg_bytes_ext_ops { + u32 id; + int (*get)(unsigned int __user *bytes, unsigned int size); + int (*put)(const unsigned int __user *bytes, unsigned int size); +}; + /* * DAPM widget event handlers - used to map handlers onto widgets. */ @@ -139,6 +146,10 @@ struct snd_soc_tplg_ops { /* bespoke kcontrol handlers available for binding */ const struct snd_soc_tplg_kcontrol_ops *io_ops; int io_ops_count; + + /* bespoke bytes ext handlers available for binding */ + const struct snd_soc_tplg_bytes_ext_ops *bytes_ext_ops; + int bytes_ext_ops_count; }; /* gets a pointer to data from the firmware block header */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 607f98b..b5a2868 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -70,6 +70,10 @@ struct soc_tplg { const struct snd_soc_tplg_kcontrol_ops *io_ops; int io_ops_count; + /* bespoke bytes ext handlers, for TLV bytes controls */ + const struct snd_soc_tplg_bytes_ext_ops *bytes_ext_ops; + int bytes_ext_ops_count; + /* optional fw loading callbacks to component drivers */ struct snd_soc_tplg_ops *ops; }; @@ -513,6 +517,15 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, { int i; + /* TLV bytes controls don't need get/put ops */ + if (k->iface & SNDRV_CTL_ELEM_IFACE_MIXER + && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE + && k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK + && hdr->ops.info == SND_SOC_TPLG_CTL_BYTES) { + k->info = snd_soc_bytes_info_ext; + return 0; + } + /* try and map standard kcontrols handler first */ for (i = 0; i < num_ops; i++) { @@ -547,6 +560,32 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, return -EINVAL; } +/* bind bytes ext get/put handlers for TLV bytes controls */ +static int soc_tplg_bytes_ext_bind_io(struct soc_tplg *tplg, + struct snd_soc_tplg_io_ops *ops_id, + struct soc_bytes_ext *sbe) +{ + const struct snd_soc_tplg_bytes_ext_ops *ops = tplg->bytes_ext_ops; + int num_ops = tplg->bytes_ext_ops_count; + int i; + + /* try bespoke handlers */ + for (i = 0; i < num_ops; i++) { + + if (sbe->put == NULL && ops[i].id == ops_id->put) + sbe->put = ops[i].put; + if (sbe->get == NULL && ops[i].id == ops_id->get) + sbe->get = ops[i].get; + } + + /* bespoke handlers found ? */ + if (sbe->put && sbe->get) + return 0; + + /* nothing to bind */ + return -EINVAL; +} + /* bind a widgets to it's evnt handlers */ int snd_soc_tplg_widget_bind_event(struct snd_soc_dapm_widget *w, const struct snd_soc_tplg_widget_events *events, @@ -690,6 +729,14 @@ static int soc_tplg_dbytes_create(struct soc_tplg *tplg, unsigned int count, continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc, &be->hdr); + /* map bytes ext io handlers for TLV bytes controls */ + if (kc.access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + soc_tplg_bytes_ext_bind_io(tplg, + &be->ext_ops, sbe); + } + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc, (struct snd_soc_tplg_ctl_hdr *)be); @@ -1315,6 +1362,14 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( continue; } + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc[i], &be->hdr); + /* map bytes ext io handlers for TLV bytes controls */ + if (kc[i].access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { + soc_tplg_bytes_ext_bind_io(tplg, + &be->ext_ops, sbe); + } + /* pass control to driver for optional further init */ err = soc_tplg_init_kcontrol(tplg, &kc[i], (struct snd_soc_tplg_ctl_hdr *)be); @@ -1736,6 +1791,8 @@ int snd_soc_tplg_component_load(struct snd_soc_component *comp, tplg.req_index = id; tplg.io_ops = ops->io_ops; tplg.io_ops_count = ops->io_ops_count; + tplg.bytes_ext_ops = ops->bytes_ext_ops; + tplg.bytes_ext_ops_count = ops->bytes_ext_ops_count; return soc_tplg_load(&tplg); } -- 1.9.1

3 5

[alsa-devel] [PATCH v2 0/2] USB audio fixes
by Pierre-Louis Bossart 16 Aug '15

16 Aug '15

While trying to understand how asynchronous endpoints are supported, I identified two major issues. On my M-Audio transit, the following script tries to play using the 3 different endpoints supported and fails (error and no audio) after the first playback without the patches. With the patches on top of 4.1.5 or Takashi's for-next (4.2.0-rc5), I stopped the audio after an hour. while true do aplay -d5 -Dhw:2,0 ~/16_48.wav # play 48kHz, 16 bit on async ep aplay -d5 -Dhw:2,0 ~/24_48.wav # play 48kHz, 24 bit on 'none'/async ep aplay -d5 -Dhw:2,0 ~/24_96.wav # play 96 kHz, 24 bit on adaptive ep aplay -d5 -Dhw:2,0 ~/24_48.wav aplay -d5 -Dhw:2,0 ~/16_48.wav aplay -d5 -Dhw:2,0 ~/24_96.wav done The first issue is pretty bad with corrupted pointers. When transitioning from a alternate setting with a sync_endpoint to another which doesn't have one by construction (synchronous, adaptive, async on capture), the pointers are not reset and the code tries to set the parameters of a non-existent endpoint. The second issue is that the M-audio Transit descriptors are illegal but the logic can be changed without harm. I sometimes hear noise on playback start, probably the code needs to be changed to add enough silent samples to let the PLL lock. I haven't had time to work on this and this is more of an enhancement. v2: applied Takashi's comments unconditional clean-up of sync_endpoint pointers better logical tests and removal of goto Pierre-Louis Bossart (2): ALSA: usb: fix corrupted pointers due to interface setting change ALSA: usb: handle descriptor with SYNC_NONE illegal value sound/usb/pcm.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) -- 1.9.1

2 3

[alsa-devel] BugReport: Dice works partially [Broken]
by Stanislav Sergiienko 16 Aug '15

16 Aug '15

Hi, I am a user of TC Electronic DesktopKonnekt6 and latest kernel update 3.19.2 add partial support for my device and thus broke my usual sound workflow. So i want to make good bug report to make device usable on latest kernels. With 3.19.2 kernel device is successfully recognized by system and even sometimes can play sound(!). But always, after some time sound is breaks (usually after 30-90 seconds) and to restore it i must restart pulseaudio(v6.0) and application. So device becomes unusable on latests kernels. Earlier i use my sound card through ffado + jackdbus + pulseaudio. Now it's impossible, so i have to downgrade kernel. Please request any additional info that you need to fix problem and i will post it here soon. Also i can test any "bleeding edge" changes thru dkms as described here: https://github.com/takaswie/snd-firewire-improve Related bug(not mine): https://bugs.archlinux.org/task/44258 Some useful info: cat /proc/asound/cards 0 [DesktopKonnekt6]: DICE - DesktopKonnekt6 TC Electronic DesktopKonnekt6 (serial 12941) at fw1.0, S400 cat /proc/asound/card0/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc1:000100000000 notification: 00000040 nick name: DesktopKonnekt6 clock select: internal 44100 enable: 1 status: locked 44100 ext status: 00000000 sample rate: 44100 version: 1.0.4.0 clock caps: 44100 48000 88200 96000 176400 192000 arx1 arx2 internal clock source names: Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\Unused\INTERNAL\\ tx 0: iso channel: 0 audio channels: 6 midi ports: 0 speed: S400 names: ch 1\ch 2\ - \ - \ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 rx 0: iso channel: 1 sequence start: 0 audio channels: 6 midi ports: 0 names: main out 1\main out 2\phones out 1\phones out 2\ - \ - \\ ac3 caps: 00000000 ac3 enable: 00000000 ext status: clock source: internal locked: 1 rate: 44100 adat user data: 0

3 4

[alsa-devel] [PATCH 0/9] Add DSP topology management for SKL
by Subhransu S. Prusty 15 Aug '15

15 Aug '15

[RESEND of the series after removing the extra patches] The SKL driver does not code DSP topology in driver. With this series the soc topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, compute resources for modules, initialize and bind the pipe modules. It uses the SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules. Last patch in this series removes the unused dais. Jeeja KP (9): ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Remove CPU dai that is not used sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-pcm.c | 173 ++-- sound/soc/intel/skylake/skl-topology.c | 1264 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 21 + sound/soc/intel/skylake/skl-tplg-interface.h | 78 ++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 7 files changed, 1511 insertions(+), 68 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c -- 2.4.3

3 28

[alsa-devel] [asoc:topic/fsl-card 2/2] sound/soc/fsl/fsl-asoc-card.c:486:29: error: 'WM8960_SYSCLK_AUTO' undeclared
by kbuild test robot 15 Aug '15

15 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/fsl-card head: 50e0ee01382b8e08289d3db209738c5856fd25cf commit: 50e0ee01382b8e08289d3db209738c5856fd25cf [2/2] ASoC: fsl-asoc-card: add wm8960 support config: xtensa-allyesconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 50e0ee01382b8e08289d3db209738c5856fd25cf # save the attached .config to linux build tree make.cross ARCH=xtensa All error/warnings (new ones prefixed by >>): sound/soc/fsl/fsl-asoc-card.c: In function 'fsl_asoc_card_probe': >> sound/soc/fsl/fsl-asoc-card.c:486:29: error: 'WM8960_SYSCLK_AUTO' undeclared (first use in this function) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; ^ sound/soc/fsl/fsl-asoc-card.c:486:29: note: each undeclared identifier is reported only once for each function it appears in vim +/WM8960_SYSCLK_AUTO +486 sound/soc/fsl/fsl-asoc-card.c 480 priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; 481 priv->codec_priv.pll_id = WM8962_FLL; 482 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 483 } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { 484 codec_dai_name = "wm8960-hifi"; 485 priv->card.set_bias_level = fsl_asoc_card_set_bias_level; > 486 priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; 487 priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; 488 priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; 489 } else { --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: fsl: fix typos for sound/soc/fsl/*
by Xiubo Li 15 Aug '15

15 Aug '15

There are too much noise about the typos for fsl's drivers. So I fix all the typos here in this patch in almost every file I touched. Signed-off-by: Xiubo Li <lixiubo(a)cmss.chinamobile.com> --- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_sai.h | 12 ++++++------ sound/soc/fsl/fsl_spdif.c | 6 +++--- sound/soc/fsl/fsl_ssi.c | 4 ++-- 4 files changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e1aa3834..883087f 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -182,7 +182,7 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else { if (np) { - /* The eukrea,asoc-tlv320 driver was explicitely + /* The eukrea,asoc-tlv320 driver was explicitly * requested (through the device tree). */ dev_err(&pdev->dev, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..b4666fd 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -45,7 +45,7 @@ #define FSL_SAI_xFR(tx) (tx ? FSL_SAI_TFR : FSL_SAI_RFR) #define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) -/* SAI Transmit/Recieve Control Register */ +/* SAI Transmit/Receive Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) #define FSL_SAI_CSR_SR BIT(24) @@ -67,10 +67,10 @@ #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) -/* SAI Transmit and Recieve Configuration 1 Register */ +/* SAI Transmit and Receive Configuration 1 Register */ #define FSL_SAI_CR1_RFW_MASK 0x1f -/* SAI Transmit and Recieve Configuration 2 Register */ +/* SAI Transmit and Receive Configuration 2 Register */ #define FSL_SAI_CR2_SYNC BIT(30) #define FSL_SAI_CR2_MSEL_MASK (0x3 << 26) #define FSL_SAI_CR2_MSEL_BUS 0 @@ -82,12 +82,12 @@ #define FSL_SAI_CR2_BCD_MSTR BIT(24) #define FSL_SAI_CR2_DIV_MASK 0xff -/* SAI Transmit and Recieve Configuration 3 Register */ +/* SAI Transmit and Receive Configuration 3 Register */ #define FSL_SAI_CR3_TRCE BIT(16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f -/* SAI Transmit and Recieve Configuration 4 Register */ +/* SAI Transmit and Receive Configuration 4 Register */ #define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) @@ -97,7 +97,7 @@ #define FSL_SAI_CR4_FSP BIT(1) #define FSL_SAI_CR4_FSD_MSTR BIT(0) -/* SAI Transmit and Recieve Configuration 5 Register */ +/* SAI Transmit and Receive Configuration 5 Register */ #define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) #define FSL_SAI_CR5_WNW_MASK (0x1f << 24) #define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8e93221..400d60e 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -707,7 +707,7 @@ static int fsl_spdif_subcode_get(struct snd_kcontrol *kcontrol, return ret; } -/* Q-subcode infomation. The byte size is SPDIF_UBITS_SIZE/8 */ +/* Q-subcode information. The byte size is SPDIF_UBITS_SIZE/8 */ static int fsl_spdif_qinfo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -739,7 +739,7 @@ static int fsl_spdif_qget(struct snd_kcontrol *kcontrol, return ret; } -/* Valid bit infomation */ +/* Valid bit information */ static int fsl_spdif_vbit_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -767,7 +767,7 @@ static int fsl_spdif_vbit_get(struct snd_kcontrol *kcontrol, return 0; } -/* DPLL lock infomation */ +/* DPLL lock information */ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c0b940e..b759448 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -156,7 +156,7 @@ struct fsl_ssi_soc_data { * * @dbg_stats: Debugging statistics * - * @soc: SoC specifc data + * @soc: SoC specific data */ struct fsl_ssi_private { struct regmap *regs; @@ -1210,7 +1210,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, } } - /* For those SLAVE implementations, we ingore non-baudclk cases + /* For those SLAVE implementations, we ignore non-baudclk cases * and, instead, abandon MASTER mode that needs baud clock. */ ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: adav80x: Remove .read_flag_mask setting from adav80x_regmap_config
by Axel Lin 15 Aug '15

15 Aug '15

Don't set .read_flag_mask for adav803, it's for adav801 only. Fixes: 0c2d69645628 ("ASoC: adav80x: Split SPI and I2C code into different modules") Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/adav80x.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 88fd37c..198c924 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -865,7 +865,6 @@ const struct regmap_config adav80x_regmap_config = { .val_bits = 8, .pad_bits = 1, .reg_bits = 7, - .read_flag_mask = 0x01, .max_register = ADAV80X_PLL_OUTE, -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: ssm4567: Add sense support
by Lars-Peter Clausen 15 Aug '15

15 Aug '15

The ssm4567 has sensing circuitry that can be used to monitor the current and voltage on the speaker amplifier output has well as the VBAT input. This data can be output over the I2S interface so it can be processed by a DSP or similar. This patch adds the sense capture output stream to the CODEC DAI as well as DAPM widgets that ensure that the sensing circuitry is powered up when the capture stream is active. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm4567.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/codecs/ssm4567.c b/sound/soc/codecs/ssm4567.c index ac242b2..e619d56 100644 --- a/sound/soc/codecs/ssm4567.c +++ b/sound/soc/codecs/ssm4567.c @@ -174,6 +174,12 @@ static const struct snd_soc_dapm_widget ssm4567_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("Amplifier Boost", SSM4567_REG_POWER_CTRL, 3, 1, &ssm4567_amplifier_boost_control), + SND_SOC_DAPM_SIGGEN("Sense"), + + SND_SOC_DAPM_PGA("Current Sense", SSM4567_REG_POWER_CTRL, 4, 1, NULL, 0), + SND_SOC_DAPM_PGA("Voltage Sense", SSM4567_REG_POWER_CTRL, 5, 1, NULL, 0), + SND_SOC_DAPM_PGA("VBAT Sense", SSM4567_REG_POWER_CTRL, 6, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("OUT"), }; @@ -181,6 +187,13 @@ static const struct snd_soc_dapm_route ssm4567_routes[] = { { "OUT", NULL, "Amplifier Boost" }, { "Amplifier Boost", "Switch", "DAC" }, { "OUT", NULL, "DAC" }, + + { "Current Sense", NULL, "Sense" }, + { "Voltage Sense", NULL, "Sense" }, + { "VBAT Sense", NULL, "Sense" }, + { "Capture Sense", NULL, "Current Sense" }, + { "Capture Sense", NULL, "Voltage Sense" }, + { "Capture Sense", NULL, "VBAT Sense" }, }; static int ssm4567_hw_params(struct snd_pcm_substream *substream, @@ -388,6 +401,14 @@ static struct snd_soc_dai_driver ssm4567_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32, }, + .capture = { + .stream_name = "Capture Sense", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32, + }, .ops = &ssm4567_dai_ops, }; -- 2.1.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: cs35l32: Use case range at appropriate places
by Axel Lin 15 Aug '15

15 Aug '15

Use case range syntax makes the code shorter with better readability when we have a large number of continuous switch cases. Below are the summary of readable/volatile/precious registers. The readable registers: 0x01 ~ 0x0D, 0x0F ~ 0x1C The volatile registers: 0x01 ~ 0x05, 0x15 ~ 0x18 The precious registers: 0x15 ~ 0x18 No functional change with this patch. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs35l32.c | 45 +++++---------------------------------------- 1 file changed, 5 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 203c720..44c30fe 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -74,33 +74,8 @@ static const struct reg_default cs35l32_reg_defaults[] = { static bool cs35l32_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_PWRCTL1: - case CS35L32_PWRCTL2: - case CS35L32_CLK_CTL: - case CS35L32_BATT_THRESHOLD: - case CS35L32_VMON: - case CS35L32_BST_CPCP_CTL: - case CS35L32_IMON_SCALING: - case CS35L32_AUDIO_LED_MNGR: - case CS35L32_ADSP_CTL: - case CS35L32_CLASSD_CTL: - case CS35L32_PROTECT_CTL: - case CS35L32_INT_MASK_1: - case CS35L32_INT_MASK_2: - case CS35L32_INT_MASK_3: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: - case CS35L32_FLASH_MODE: - case CS35L32_MOVIE_MODE: - case CS35L32_FLASH_TIMER: - case CS35L32_FLASH_INHIBIT: + case CS35L32_DEVID_AB ... CS35L32_AUDIO_LED_MNGR: + case CS35L32_ADSP_CTL ... CS35L32_FLASH_INHIBIT: return true; default: return false; @@ -110,15 +85,8 @@ static bool cs35l32_readable_register(struct device *dev, unsigned int reg) static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_DEVID_AB: - case CS35L32_DEVID_CD: - case CS35L32_DEVID_E: - case CS35L32_FAB_ID: - case CS35L32_REV_ID: - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_DEVID_AB ... CS35L32_REV_ID: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; @@ -128,10 +96,7 @@ static bool cs35l32_volatile_register(struct device *dev, unsigned int reg) static bool cs35l32_precious_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS35L32_INT_STATUS_1: - case CS35L32_INT_STATUS_2: - case CS35L32_INT_STATUS_3: - case CS35L32_LED_STATUS: + case CS35L32_INT_STATUS_1 ... CS35L32_LED_STATUS: return true; default: return false; -- 2.1.0

3 7

[alsa-devel] Applied "ASoC: cs42l56: Use case range at appropriate place" to the asoc tree
by Mark Brown 15 Aug '15

15 Aug '15

The patch ASoC: cs42l56: Use case range at appropriate place has been applied to the asoc tree at git://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 7f325bfc933d140337e55507821d6bb021321059 Mon Sep 17 00:00:00 2001 From: Axel Lin <axel.lin(a)ingics.com> Date: Wed, 12 Aug 2015 11:09:39 +0800 Subject: [PATCH] ASoC: cs42l56: Use case range at appropriate place The readable registers are in continuous range: 0x01 ~ 0x2e. Use case range syntax makes the code shorter with better readability when we have a large number of continuous switch cases. No functional change with this patch. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cs42l56.c | 47 +--------------------------------------------- 1 file changed, 1 insertion(+), 46 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 1e11ba4..98a68c6 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -115,52 +115,7 @@ static const struct reg_default cs42l56_reg_defaults[] = { static bool cs42l56_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS42L56_CHIP_ID_1: - case CS42L56_CHIP_ID_2: - case CS42L56_PWRCTL_1: - case CS42L56_PWRCTL_2: - case CS42L56_CLKCTL_1: - case CS42L56_CLKCTL_2: - case CS42L56_SERIAL_FMT: - case CS42L56_CLASSH_CTL: - case CS42L56_MISC_CTL: - case CS42L56_INT_STATUS: - case CS42L56_PLAYBACK_CTL: - case CS42L56_DSP_MUTE_CTL: - case CS42L56_ADCA_MIX_VOLUME: - case CS42L56_ADCB_MIX_VOLUME: - case CS42L56_PCMA_MIX_VOLUME: - case CS42L56_PCMB_MIX_VOLUME: - case CS42L56_ANAINPUT_ADV_VOLUME: - case CS42L56_DIGINPUT_ADV_VOLUME: - case CS42L56_MASTER_A_VOLUME: - case CS42L56_MASTER_B_VOLUME: - case CS42L56_BEEP_FREQ_ONTIME: - case CS42L56_BEEP_FREQ_OFFTIME: - case CS42L56_BEEP_TONE_CFG: - case CS42L56_TONE_CTL: - case CS42L56_CHAN_MIX_SWAP: - case CS42L56_AIN_REFCFG_ADC_MUX: - case CS42L56_HPF_CTL: - case CS42L56_MISC_ADC_CTL: - case CS42L56_GAIN_BIAS_CTL: - case CS42L56_PGAA_MUX_VOLUME: - case CS42L56_PGAB_MUX_VOLUME: - case CS42L56_ADCA_ATTENUATOR: - case CS42L56_ADCB_ATTENUATOR: - case CS42L56_ALC_EN_ATTACK_RATE: - case CS42L56_ALC_RELEASE_RATE: - case CS42L56_ALC_THRESHOLD: - case CS42L56_NOISE_GATE_CTL: - case CS42L56_ALC_LIM_SFT_ZC: - case CS42L56_AMUTE_HPLO_MUX: - case CS42L56_HPA_VOLUME: - case CS42L56_HPB_VOLUME: - case CS42L56_LOA_VOLUME: - case CS42L56_LOB_VOLUME: - case CS42L56_LIM_THRESHOLD_CTL: - case CS42L56_LIM_CTL_RELEASE_RATE: - case CS42L56_LIM_ATTACK_RATE: + case CS42L56_CHIP_ID_1 ... CS42L56_LIM_ATTACK_RATE: return true; default: return false; -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: Intel: fix runtime pm imbalance on error
by Junjie Mao 15 Aug '15

15 Aug '15

pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Junjie Mao <junjie.mao(a)enight.me> --- sound/soc/intel/atom/sst/sst_drv_interface.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_drv_interface.c b/sound/soc/intel/atom/sst/sst_drv_interface.c index 620da1d1b9e3..27a9653aa3d9 100644 --- a/sound/soc/intel/atom/sst/sst_drv_interface.c +++ b/sound/soc/intel/atom/sst/sst_drv_interface.c @@ -152,6 +152,7 @@ static int sst_power_control(struct device *dev, bool state) dev_dbg(ctx->dev, "Enable: pm usage count: %d\n", usage_count); if (ret < 0) { + pm_runtime_put_sync(dev); dev_err(ctx->dev, "Runtime get failed with err: %d\n", ret); return ret; } @@ -204,8 +205,10 @@ static int sst_cdev_open(struct device *dev, struct intel_sst_drv *ctx = dev_get_drvdata(dev); retval = pm_runtime_get_sync(ctx->dev); - if (retval < 0) + if (retval < 0) { + pm_runtime_put_sync(ctx->dev); return retval; + } str_id = sst_get_stream(ctx, str_params); if (str_id > 0) { @@ -672,8 +675,10 @@ static int sst_send_byte_stream(struct device *dev, if (NULL == bytes) return -EINVAL; ret_val = pm_runtime_get_sync(ctx->dev); - if (ret_val < 0) + if (ret_val < 0) { + pm_runtime_put_sync(ctx->dev); return ret_val; + } ret_val = sst_send_byte_stream_mrfld(ctx, bytes); sst_pm_runtime_put(ctx); -- 1.9.3

2 1

[alsa-devel] [PATCH 1/6] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c7647e0..9c46c7d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

2 1

[alsa-devel] [PATCH 2/6] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 9c46c7d..2ce9e1d 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

2 1

[alsa-devel] [PATCH 3/6] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 2ce9e1d..f3034b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

3 3

[alsa-devel] [PATCH 4/6] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f3034b9..0b4fcd9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

2 1

[alsa-devel] [PATCH 5/6] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0b4fcd9..e79dc16 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

2 1

[alsa-devel] [PATCH 6/6] ASoC: fsl_ssi: adjust set DAI format in AC'97 mode
by Maciej S. Szmigiero 15 Aug '15

15 Aug '15

Adjust set DAI format function in fsl_ssi driver so it doesn't fail and clears RXDIR in AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8e5ff5e..37aabe3 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -900,14 +900,16 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + if (!fsl_ssi_is_ac97(ssi_private)) + return -EINVAL; } stcr |= strcr; srcr |= strcr; - if (ssi_private->cpu_dai_drv.symmetric_rates) { - /* Need to clear RXDIR when using SYNC mode */ + if (ssi_private->cpu_dai_drv.symmetric_rates + || fsl_ssi_is_ac97(ssi_private)) { + /* Need to clear RXDIR when using SYNC or AC97 mode */ srcr &= ~CCSR_SSI_SRCR_RXDIR; scr |= CCSR_SSI_SCR_SYN; }

3 9

[alsa-devel] [RFC PATCH 08/10] ASoC: topology: change pass number of DAI smaller than graph
by mengdong.lin＠intel.com 15 Aug '15

15 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The PCM DAIs need to be loaded and added to ASoC core ealier than the graph (route). Otherwise, adding routes will fail for missing DAIs. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 1b6728b..0f8fa0d 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -45,12 +45,12 @@ #define SOC_TPLG_PASS_VENDOR 1 #define SOC_TPLG_PASS_MIXER 2 #define SOC_TPLG_PASS_WIDGET 3 -#define SOC_TPLG_PASS_GRAPH 4 -#define SOC_TPLG_PASS_PINS 5 -#define SOC_TPLG_PASS_PCM_DAI 6 +#define SOC_TPLG_PASS_PCM_DAI 4 +#define SOC_TPLG_PASS_GRAPH 5 +#define SOC_TPLG_PASS_PINS 6 #define SOC_TPLG_PASS_START SOC_TPLG_PASS_MANIFEST -#define SOC_TPLG_PASS_END SOC_TPLG_PASS_PCM_DAI +#define SOC_TPLG_PASS_END SOC_TPLG_PASS_PINS struct soc_tplg { const struct firmware *fw; -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] USB audio fixes
by Pierre-Louis Bossart 14 Aug '15

14 Aug '15

While trying to understand how asynchronous endpoints are supported, I identified two major issues. On my M-Audio transit, the following script tries to play using the 3 different endpoints supported and fails (error and no audio) after the first playback without the patches. With the patches on top of 4.1.5 or Takashi's for-next (4.2.0-rc5), I stopped the audio after an hour. while true do aplay -d5 -Dhw:2,0 ~/16_48.wav # play 48kHz, 16 bit on async ep aplay -d5 -Dhw:2,0 ~/24_48.wav # play 48kHz, 24 bit on 'none'/async ep aplay -d5 -Dhw:2,0 ~/24_96.wav # play 96 kHz, 24 bit on adaptive ep aplay -d5 -Dhw:2,0 ~/24_48.wav aplay -d5 -Dhw:2,0 ~/16_48.wav aplay -d5 -Dhw:2,0 ~/24_96.wav done The first issue is pretty bad with corrupted pointers. When transitioning from a alternate setting with a sync_endpoint to another which doesn't have one by construction (synchronous, adaptive, async on capture), the pointers are not reset and the code tries to set the parameters of a non-existent endpoint. The second issue is that the M-audio Transit descriptors are illegal but the logic can be changed without harm. I sometimes hear noise on playback start, probably the code needs to be changed to add enough silent samples to let the PLL lock. I haven't had time to work on this and this is more of an enhancement. Comments welcome. Pierre-Louis Bossart (2): ALSA: usb: fix corrupted pointers due to interface setting change ALSA: usb: handle descriptor with SYNC_NONE illegal value sound/usb/pcm.c | 32 ++++++++++++++++++++++++++++++-- 1 file changed, 30 insertions(+), 2 deletions(-) -- 1.9.1

2 10

[alsa-devel] [PATCH] ASoC: ics43432: Add codec driver for InvenSense ICS-43432
by Ricard Wanderlof 14 Aug '15

14 Aug '15

Add support for the InvenSense ICS-43432 I2S MEMS microphone. This is a non-software-configurable MEMS microphone with I2S output. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../devicetree/bindings/sound/ics43432.txt | 17 +++ .../devicetree/bindings/vendor-prefixes.txt | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ics43432.c | 111 ++++++++++++++++++++ 5 files changed, 135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 sound/soc/codecs/ics43432.c diff --git a/Documentation/devicetree/bindings/sound/ics43432.txt b/Documentation/devicetree/bindings/sound/ics43432.txt new file mode 100644 index 0000000..b02e3a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ics43432.txt @@ -0,0 +1,17 @@ +Invensense ICS-43432 MEMS microphone with I2S output. + +There are no software configuration options for this device, indeed, the only +host connection is the I2S interface. Apart from requirements on clock +frequency (460 kHz to 3.379 MHz according to the data sheet) there must be +64 clock cycles in each stereo output frame; 24 of the 32 available bits +contain audio data. A hardware pin determines if the device outputs data +on the left or right channel of the I2S frame. + +Required properties: + - compatible : Must be "invensense,ics43432" + +Example: + + ics43432: ics43432 { + compatible = "invensense,ics43432"; + }; diff --git a/Documentation/devicetree/bindings/vendor-prefixes.txt b/Documentation/devicetree/bindings/vendor-prefixes.txt index 8033919..7966637 100644 --- a/Documentation/devicetree/bindings/vendor-prefixes.txt +++ b/Documentation/devicetree/bindings/vendor-prefixes.txt @@ -103,6 +103,7 @@ img Imagination Technologies Ltd. innolux Innolux Corporation intel Intel Corporation intercontrol Inter Control Group +invensense InvenSense Inc. isee ISEE 2007 S.L. isil Intersil karo Ka-Ro electronics GmbH diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..d696c85 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -62,6 +62,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_ES8328_SPI if SPI_MASTER select SND_SOC_ES8328_I2C if I2C + select SND_SOC_ICS43432 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -444,6 +445,9 @@ config SND_SOC_ES8328_SPI tristate select SND_SOC_ES8328 +config SND_SOC_ICS43432 + tristate + config SND_SOC_ISABELLE tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index abe2d7e..23a2caf 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o +snd-soc-ics43432-objs := ics43432.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -240,6 +241,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o +obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c new file mode 100644 index 0000000..902daae --- /dev/null +++ b/sound/soc/codecs/ics43432.c @@ -0,0 +1,111 @@ +/* + * I2S MEMS microphone driver for InvenSense ICS-43432 + * + * - Non configurable. + * - I2S interface, 64 BCLs per frame, 32 bits per channel, 24 bit data + * + * Copyright (c) 2015 Axis Communications AB + * + * Licensed under GPL2. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#define ICS43432_RATE_MIN 7190 /* Hz, from data sheet */ +#define ICS43432_RATE_MAX 52800 /* Hz, from data sheet */ + +static int ics43432_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + unsigned int rate = params_rate(params); + + if (rate < ICS43432_RATE_MIN || rate > ICS43432_RATE_MAX) + return -EINVAL; + + return 0; +} + +static int ics43432_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + default: + return -EINVAL; + } + + return 0; +} + +#define ICS43432_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32) + +static const struct snd_soc_dai_ops ics43432_dai_ops = { + .hw_params = ics43432_hw_params, + .set_fmt = ics43432_set_dai_fmt, +}; + +static struct snd_soc_dai_driver ics43432_dai = { + .name = "ics43432-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ICS43432_FORMATS, + }, + .ops = &ics43432_dai_ops, +}; + +static struct snd_soc_codec_driver ics43432_codec_driver = { +}; + +static int ics43432_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &ics43432_codec_driver, + &ics43432_dai, 1); +} + +static int ics43432_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +#ifdef CONFIG_OF +static const struct of_device_id ics43432_ids[] = { + { .compatible = "invensense,ics43432", }, + { } +}; +MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +#endif + +static struct platform_driver ics43432_driver = { + .driver = { + .name = "ics43432", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ics43432_ids), + }, + .probe = ics43432_probe, + .remove = ics43432_remove, +}; + +module_platform_driver(ics43432_driver); + +MODULE_DESCRIPTION("ASoC ICS43432 driver"); +MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); +MODULE_LICENSE("GPL2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 2

[alsa-devel] [PATCH v2] ALSA: usb-audio: Fix parameter block size for UAC2 control requests
by Julian Scheel 14 Aug '15

14 Aug '15

USB Audio Class version 2.0 supports three different parameter block sizes for CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes (5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter Block). Use the correct size according to the specific control as it was already done for UACv1. The allocated block size for control requests is increased to support the 4 byte worst case. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- Changes in v2: - Removed superfluous int32 checks - Check control value length in function - Compare against 0 - Use a tmp variable to access audio_feature_info[control-1] Thanks for review by Takashi Iwai :) --- sound/usb/mixer.c | 64 ++++++++++++++++++++++++++++++++++++++----------------- sound/usb/mixer.h | 2 ++ 2 files changed, 46 insertions(+), 20 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4a49944..7046f54 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -282,6 +282,21 @@ static int get_abs_value(struct usb_mixer_elem_info *cval, int val) return val; } +static int uac2_ctl_value_size(int val_type) +{ + switch (val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + return 4; + case USB_MIXER_S16: + case USB_MIXER_U16: + return 2; + default: + return 1; + } + return 0; /* unreachable */ +} + /* * retrieve a mixer value @@ -328,14 +343,14 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = sizeof(__u16); + size = uac2_ctl_value_size(cval->val_type); } else { bRequest = UAC2_CS_RANGE; size = sizeof(buf); @@ -446,7 +461,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2]; + unsigned char buf[4]; int idx = 0, val_len, err, timeout = 10; validx += cval->idx_off; @@ -454,8 +469,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + val_len = uac2_ctl_value_size(cval->val_type); /* FIXME */ if (request != UAC_SET_CUR) { @@ -469,6 +483,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; + buf[2] = (value_set >> 16) & 0xff; + buf[3] = (value_set >> 24) & 0xff; err = snd_usb_autoresume(chip); if (err < 0) return -EIO; @@ -799,24 +815,25 @@ static int check_input_term(struct mixer_build *state, int id, /* feature unit control information */ struct usb_feature_control_info { const char *name; - unsigned int type; /* control type (mute, volume, etc.) */ + int type; /* data type for uac1 */ + int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, - { "Tone Control - Bass", USB_MIXER_S8 }, - { "Tone Control - Mid", USB_MIXER_S8 }, - { "Tone Control - Treble", USB_MIXER_S8 }, - { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Mute", USB_MIXER_INV_BOOLEAN, -1 }, + { "Volume", USB_MIXER_S16, -1 }, + { "Tone Control - Bass", USB_MIXER_S8, -1 }, + { "Tone Control - Mid", USB_MIXER_S8, -1 }, + { "Tone Control - Treble", USB_MIXER_S8, -1 }, + { "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemeted yet */ + { "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, + { "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, + { "Bass Boost", USB_MIXER_BOOLEAN, -1 }, + { "Loudness", USB_MIXER_BOOLEAN, -1 }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_S16 }, - { "Input Gain Pad Control", USB_MIXER_S16 }, - { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16, -1 }, + { "Input Gain Pad Control", USB_MIXER_S16, -1 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; /* private_free callback */ @@ -1216,6 +1233,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, int readonly_mask) { struct uac_feature_unit_descriptor *desc = raw_desc; + struct usb_feature_control_info *ctl_info; unsigned int len = 0; int mapped_name = 0; int nameid = uac_feature_unit_iFeature(desc); @@ -1241,7 +1259,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + ctl_info = &audio_feature_info[control-1]; + if (state->mixer->protocol == UAC_VERSION_1) + cval->val_type = ctl_info->type; + else /* UAC_VERSION_2 */ + cval->val_type = ctl_info->type_uac2 >= 0 ? + ctl_info->type_uac2 : ctl_info->type; + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index d3268f0..3417ef3 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -33,6 +33,8 @@ enum { USB_MIXER_U8, USB_MIXER_S16, USB_MIXER_U16, + USB_MIXER_S32, + USB_MIXER_U32, }; typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, -- 2.5.0

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Fix parameter block size for UAC2 control requests
by Julian Scheel 14 Aug '15

14 Aug '15

USB Audio Class version 2.0 supports three different parameter block sizes for CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes (5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter Block). Use the correct size according to the specific control as it was already done for UACv1. The allocated block size for control requests is increased to support the 4 byte worst case. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/usb/mixer.c | 83 +++++++++++++++++++++++++++++++++++++++++-------------- sound/usb/mixer.h | 2 ++ 2 files changed, 65 insertions(+), 20 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4a49944..675725b 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -233,6 +233,14 @@ static int convert_signed_value(struct usb_mixer_elem_info *cval, int val) if (val >= 0x8000) val -= 0x10000; break; + case USB_MIXER_U32: + val &= 0xffffffff; + break; + case USB_MIXER_S32: + val &= 0xffffffff; + if (val >= 0x80000000) + val -= 0x100000000; + break; } return val; } @@ -253,6 +261,9 @@ static int convert_bytes_value(struct usb_mixer_elem_info *cval, int val) case USB_MIXER_S16: case USB_MIXER_U16: return val & 0xffff; + case USB_MIXER_S32: + case USB_MIXER_U32: + return val & 0xffffffff; } return 0; /* not reached */ } @@ -328,14 +339,26 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2 + 3 * sizeof(__u16)]; /* enough space for one range */ + unsigned char buf[4 + 3 * sizeof(__u32)]; /* enough space for one range */ unsigned char *val; int idx = 0, ret, size; __u8 bRequest; if (request == UAC_GET_CUR) { bRequest = UAC2_CS_CUR; - size = sizeof(__u16); + switch (cval->val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + size = 4; + break; + case USB_MIXER_S16: + case USB_MIXER_U16: + size = 2; + break; + default: + size = 1; + break; + } } else { bRequest = UAC2_CS_RANGE; size = sizeof(buf); @@ -446,7 +469,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, int request, int validx, int value_set) { struct snd_usb_audio *chip = cval->head.mixer->chip; - unsigned char buf[2]; + unsigned char buf[4]; int idx = 0, val_len, err, timeout = 10; validx += cval->idx_off; @@ -454,8 +477,19 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, if (cval->head.mixer->protocol == UAC_VERSION_1) { val_len = cval->val_type >= USB_MIXER_S16 ? 2 : 1; } else { /* UAC_VERSION_2 */ - /* audio class v2 controls are always 2 bytes in size */ - val_len = sizeof(__u16); + switch (cval->val_type) { + case USB_MIXER_S32: + case USB_MIXER_U32: + val_len = 4; + break; + case USB_MIXER_S16: + case USB_MIXER_U16: + val_len = 2; + break; + default: + val_len = 1; + break; + } /* FIXME */ if (request != UAC_SET_CUR) { @@ -469,6 +503,8 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, value_set = convert_bytes_value(cval, value_set); buf[0] = value_set & 0xff; buf[1] = (value_set >> 8) & 0xff; + buf[2] = (value_set >> 16) & 0xff; + buf[3] = (value_set >> 24) & 0xff; err = snd_usb_autoresume(chip); if (err < 0) return -EIO; @@ -799,24 +835,25 @@ static int check_input_term(struct mixer_build *state, int id, /* feature unit control information */ struct usb_feature_control_info { const char *name; - unsigned int type; /* control type (mute, volume, etc.) */ + int type; /* data type for uac1 */ + int type_uac2; /* data type for uac2 if different from uac1, else -1 */ }; static struct usb_feature_control_info audio_feature_info[] = { - { "Mute", USB_MIXER_INV_BOOLEAN }, - { "Volume", USB_MIXER_S16 }, - { "Tone Control - Bass", USB_MIXER_S8 }, - { "Tone Control - Mid", USB_MIXER_S8 }, - { "Tone Control - Treble", USB_MIXER_S8 }, - { "Graphic Equalizer", USB_MIXER_S8 }, /* FIXME: not implemeted yet */ - { "Auto Gain Control", USB_MIXER_BOOLEAN }, - { "Delay Control", USB_MIXER_U16 }, /* FIXME: U32 in UAC2 */ - { "Bass Boost", USB_MIXER_BOOLEAN }, - { "Loudness", USB_MIXER_BOOLEAN }, + { "Mute", USB_MIXER_INV_BOOLEAN, -1 }, + { "Volume", USB_MIXER_S16, -1 }, + { "Tone Control - Bass", USB_MIXER_S8, -1 }, + { "Tone Control - Mid", USB_MIXER_S8, -1 }, + { "Tone Control - Treble", USB_MIXER_S8, -1 }, + { "Graphic Equalizer", USB_MIXER_S8, -1 }, /* FIXME: not implemeted yet */ + { "Auto Gain Control", USB_MIXER_BOOLEAN, -1 }, + { "Delay Control", USB_MIXER_U16, USB_MIXER_U32 }, + { "Bass Boost", USB_MIXER_BOOLEAN, -1 }, + { "Loudness", USB_MIXER_BOOLEAN, -1 }, /* UAC2 specific */ - { "Input Gain Control", USB_MIXER_S16 }, - { "Input Gain Pad Control", USB_MIXER_S16 }, - { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16, -1 }, + { "Input Gain Pad Control", USB_MIXER_S16, -1 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN, -1 }, }; /* private_free callback */ @@ -1241,7 +1278,13 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + if (state->mixer->protocol == UAC_VERSION_1) + cval->val_type = audio_feature_info[control-1].type; + else /* UAC_VERSION_2 */ + cval->val_type = audio_feature_info[control-1].type_uac2 > -1 ? + audio_feature_info[control-1].type_uac2 : + audio_feature_info[control-1].type; + if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index d3268f0..3417ef3 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -33,6 +33,8 @@ enum { USB_MIXER_U8, USB_MIXER_S16, USB_MIXER_U16, + USB_MIXER_S32, + USB_MIXER_U32, }; typedef void (*usb_mixer_elem_dump_func_t)(struct snd_info_buffer *buffer, -- 2.5.0

2 4

[alsa-devel] [PATCH v2 1/3] ASoC: dapm: dapm_dai_get_connected_widgets: Fix missing mutex unlock
by Lars-Peter Clausen 14 Aug '15

14 Aug '15

Make sure to unlock the DAPM mutex when dapm_widget_list_create() fails. This means the function will now generate a trace_snd_soc_dapm_connected event, even if the creation of the list fails. But that was the behavior before the patch that introduced the unlock issue, so that should be fine. Fixes: 1ce43acff0c0 ("ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets()") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- New in v2 --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7482c5d..c391bfe 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1191,7 +1191,7 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, ret = dapm_widget_list_create(list, &widgets); if (ret) - return ret; + paths = ret; trace_snd_soc_dapm_connected(paths, stream); mutex_unlock(&card->dapm_mutex); -- 2.1.4

2 4

[alsa-devel] [PATCH 1/5] ASoC: dapm: Drop always true checks
by Lars-Peter Clausen 14 Aug '15

14 Aug '15

list_first_entry() always returns non NULL and since the code previously checked that list is not empty it will also be a valid pointer. Furthermore a path has always a sink or a source widget. So both checks are redundant and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-dapm.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7f834a0..10f7c7c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3485,11 +3485,6 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, sink_p = list_first_entry(&w->sinks, struct snd_soc_dapm_path, list_source); - if (WARN_ON(!source_p || !sink_p) || - WARN_ON(!sink_p->source || !source_p->sink) || - WARN_ON(!source_p->source || !sink_p->sink)) - return -EINVAL; - source = source_p->source->priv; sink = sink_p->sink->priv; -- 2.1.4

2 11

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: Specify codec_dai_name for DAI links
by Nicolin Chen 14 Aug '15

14 Aug '15

The dev->name of CODEC might not be identical to its codec_dai_name, so using dev->name to probe the CODEC dai is not a correct approach. This patch specifies each supporting codec_dai_name instead of using dev->name any more. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl-asoc-card.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..040362f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -407,6 +407,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; struct clk *codec_clk; + const char *codec_dai_name; u32 width; int ret; @@ -459,6 +460,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { + codec_dai_name = "cs42888"; priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; @@ -467,9 +469,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.slot_width = 32; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; } else if (of_device_is_compatible(np, "fsl,imx-audio-sgtl5000")) { + codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { + codec_dai_name = "wm8962"; priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; @@ -521,7 +525,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; - priv->dai_link[0].codec_dai_name = codec_dev->name; + priv->dai_link[0].codec_dai_name = codec_dai_name; priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -530,7 +534,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exsits */ priv->dai_link[1].cpu_of_node = asrc_np; priv->dai_link[1].platform_of_node = asrc_np; - priv->dai_link[2].codec_dai_name = codec_dev->name; + priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; -- 1.9.1

2 1

[alsa-devel] [PATCH] Add rudimentary device tree support for SSM2518
by Ricard Wanderlof 13 Aug '15

13 Aug '15

For some reason the SSM2518 codec driver doesn't have DT support, even though there are bindings described for it in Documentation/devicetree/sound/ssm2518.txt . This patch adds rudimentary DT support for the compatible property which is a step forward. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/ssm2518.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 67ea55a..ecf9996 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -811,6 +810,14 @@ static int ssm2518_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id ssm2518_dt_ids[] = { + { .compatible = "adi,ssm2518", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); +#endif + static const struct i2c_device_id ssm2518_i2c_ids[] = { { "ssm2518", 0 }, { } @@ -821,6 +828,9 @@ static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = of_match_ptr(ssm2518_dt_ids), +#endif }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH] Add rudimentary DT support to SSM2518 codec driver
by Ricard Wanderlof 13 Aug '15

13 Aug '15

For some reason, the SSM2518 codec driver has device tree bindings specified (Documentation/devicetree/bindings/sound/ssm2518.txt) but the actual driver doesn't seem to support DT. This patch adds rudimentary support for the compatible property so that the driver can be used in a DT based setup. --- sound/soc/codecs/ssm2518.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index 67ea55a..ecf9996 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -811,6 +810,14 @@ static int ssm2518_i2c_remove(struct i2c_client *client) return 0; } +#ifdef CONFIG_OF +static const struct of_device_id ssm2518_dt_ids[] = { + { .compatible = "adi,ssm2518", }, + { } +}; +MODULE_DEVICE_TABLE(of, ssm2518_dt_ids); +#endif + static const struct i2c_device_id ssm2518_i2c_ids[] = { { "ssm2518", 0 }, { } @@ -821,6 +828,9 @@ static struct i2c_driver ssm2518_driver = { .driver = { .name = "ssm2518", .owner = THIS_MODULE, +#ifdef CONFIG_OF + .of_match_table = of_match_ptr(ssm2518_dt_ids), +#endif }, .probe = ssm2518_i2c_probe, .remove = ssm2518_i2c_remove, -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[alsa-devel] [PATCH 2/2] topology: Add API to set a vendor specific version number
by mengdong.lin＠intel.com 13 Aug '15

13 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This vendor-specific version number is optional. It will be written to the 'version' field of each block header of the binary toplogy data file. The vendor driver can check this number for further processing in kernel. The topology ABI version number is still stored in the 'abi' field of block headers. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/topology.h b/include/topology.h index 6ff8c5f..9b84bd9 100644 --- a/include/topology.h +++ b/include/topology.h @@ -690,6 +690,14 @@ int snd_tplg_build(snd_tplg_t *tplg, const char *outfile); */ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len); +/** + * \brief Set an optional vendor specific version number. + * \param tplg Topology instance. + * \param version Vendor specific version number. + * \return Zero on success, otherwise a negative error code + */ +int snd_tplg_set_version(snd_tplg_t *tplg, unsigned int version); + /* \} */ #ifdef __cplusplus diff --git a/src/topology/builder.c b/src/topology/builder.c index 3bccd44..2f01137 100644 --- a/src/topology/builder.c +++ b/src/topology/builder.c @@ -89,7 +89,7 @@ static int write_data_block(snd_tplg_t *tplg, int size, int tplg_type, /* write the header for this block */ ret = write_block_header(tplg, tplg_type, 0, - SND_SOC_TPLG_ABI_VERSION, 0, size, 1); + tplg->version, 0, size, 1); if (ret < 0) { SNDERR("error: failed to write %s block %d\n", obj_name, ret); return ret; @@ -125,7 +125,7 @@ static int write_elem_block(snd_tplg_t *tplg, } ret = write_block_header(tplg, tplg_type, vendor_type, - SND_SOC_TPLG_ABI_VERSION, 0, size, count); + tplg->version, 0, size, count); if (ret < 0) { SNDERR("error: failed to write %s block %d\n", obj_name, ret); @@ -243,7 +243,7 @@ static int write_manifest_data(snd_tplg_t *tplg) /* write the header for this block */ ret = write_block_header(tplg, SND_SOC_TPLG_TYPE_MANIFEST, 0, - SND_SOC_TPLG_ABI_VERSION, 0, + tplg->version, 0, sizeof(tplg->manifest) + tplg->manifest.priv.size, 1); if (ret < 0) { SNDERR("error: failed to write manifest block %d\n", ret); diff --git a/src/topology/parser.c b/src/topology/parser.c index 7ca244c..c0ab5e4 100644 --- a/src/topology/parser.c +++ b/src/topology/parser.c @@ -362,6 +362,13 @@ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len) return 0; } +int snd_tplg_set_version(snd_tplg_t *tplg, unsigned int version) +{ + tplg->version = version; + + return 0; +} + void snd_tplg_verbose(snd_tplg_t *tplg, int verbose) { tplg->verbose = verbose; -- 1.9.1

1 0

[alsa-devel] [PATCH 1/2] topology: Fix missing return value for snd_tplg_set_manifest_data()
by mengdong.lin＠intel.com 13 Aug '15

13 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/src/topology/parser.c b/src/topology/parser.c index ca7de06..7ca244c 100644 --- a/src/topology/parser.c +++ b/src/topology/parser.c @@ -358,6 +358,8 @@ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len) { tplg->manifest.priv.size = len; tplg->manifest_pdata = data; + + return 0; } void snd_tplg_verbose(snd_tplg_t *tplg, int verbose) -- 1.9.1

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix the white noise on Dell laptop
by woodrow.shen＠canonical.com 13 Aug '15

13 Aug '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> Dell laptop causes the white noise by login screen and headphone, and the fixup function ALC292_FIXUP_DISABLE_AAMIX can eliminate this noise. Codec: Realtek ALC3235 Vendor Id: 0x10ec0293 Subsystem Id: 0x102806db Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1484334 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- 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 0e3fe01..1de427a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5190,6 +5190,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.1.4

2 1

[alsa-devel] M-Audio transit: illegal sync type and noise on startup?
by Pierre-Louis Bossart 13 Aug '15

13 Aug '15

While playing with my M-Audio Transit USB device (0763:2006) I saw a synchronization type that doesn't exist in the USB audio class. I get audio out but often the start is garbled with white noise and cracks. Wondering what's going on. more /proc/asound/card1/stream0 M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (ADAPTIVE) Rates: 48001 - 96000 (continuous)u Interface 1 Altset 2 Format: S24_3LE Channels: 2 Endpoint: 3 OUT (NONE) <<<<<<< illegal Rates: 8000 - 48000 (continuous) Interface 1 Altset 3 Format: S16_LE Channels: 2 Endpoint: 3 OUT (ASYNC) Rates: 8000 - 48000 (continuous) The only definitions in the spec are Asynchronous, Adaptive or Synchronous... lsusb tells me that there seems to be a synch endpoint for this setting: Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0126 1x 294 bytes bInterval 1 bRefresh 0 bSynchAddress 131 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 1 Milliseconds wLockDelay 100 Milliseconds Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 2 bSynchAddress 0 I couldn't figure out what the code does in this case. I didn't find any sync URBs being retired so wondering if this is considered as a synchronous or asynchronous endpoint based on the additional sync endpoint? Any idea on how to debug further? It looks like there could be a second error in the descriptors: if this is a synchronous endpoint then the sync endpoint is meaningless, and if it is asynchronous then the information on locking time is irrelevant... Regarding the noise on startup, should a delay be inserted after the mode is set? the spec recommends that the host sends silence periods for adaptive and synchronous modes. I see a set of mdelays being added in the quirks, should a new one be added? Thanks for any pointers, -Pierre

1 0

[alsa-devel] Questions about ASoC driver implementation
by Matt Campbell 12 Aug '15

12 Aug '15

Hi All, I'm diving into adding support for a new codec (AK4621) targeting IMX6 SoC and as this is my first dive into ALSA driver code (and first deep dive in to Linux driver development) I've come up with a few questions. I've done a lot of digging in available documentation as well as reading a lot of code, but some things remain unclear. First, I'm a little confused with what appears to be inconsistent terminology. The SoC overview.txt lists three types of component drivers: codec, platform and machine. However, in the code there are four types of snd_soc_register_* functions: card, platform, codec and component. From looking at the underlying structs and how these are used in existing code this is my best guess but please correct me if I'm wrong: codec -> snd_soc_register_codec machine -> snd_soc_register_card platform -> snd_soc_register_component and snd_soc_register_platform? (not sure about this, from what I can gather at some point there was an API switch to using component rather than platforms for the DAIs) My second question is around adding support for an amplifier external to the codec. The amplifier is a PGA controlled over SPI. One possible way I can see to adding this is to create a SPI based codec driver for the PGA that exposes an ALSA control for the gain level and give it a dummy DAI then add it to the card as an aux_dev. This feels a little kludgy as the PGA isn't a perfect fit for the codec class. Another option would be to have PGA driver separate from the ASoC framework, add the controls for the gain level directly in the ASoC card and put the glue code to interact with the device in the machine driver. This feels somewhat better to me, but has two problems. First, I don't know the best way to interact with the external PGA driver from the machine driver. Second, this leaves the PGA driver outside of the ASoC framework, If it was in ASoC it could be useful for others that come across this chip when writing machine drivers. Sorry for the long email, but there has been a bit of a learning curve up front. I tried to do my best with documentation and code reading before coming to the mailing list. I'll greatly appreciate any help the community can offer and look forward to contributing to the ALSA project. Best, ~Matt

2 2

[alsa-devel] [PATCH v2 1/4] drm/i915: Add audio set_ncts callback
by libin.yang＠intel.com 12 Aug '15

12 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the set_ncts callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..7305881 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,8 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*set_ncts)(struct device *, int port, int dev_entry, + int rate); } *ops; }; -- 1.9.1

5 35

[alsa-devel] [PATCH v3 0/3] topology: Add external API for building topology data
by Liam Girdwood 12 Aug '15

12 Aug '15

Currently we can build topology binary data files from topology text files. However it is sometimes necessary for DSP FW vendors to build topology binary data within a toolset and create topology binary data directly from within vendor tools. This series adds an API to the alsa-lib topology core so that vendor tools can create topology data directly. Changes since V2:- o Fixed elem memory leaks on error handling paths o Fixed indentation. o Moved container_of and ARRAY_SIZE to local.h o remove use of unlink() and fixed file permissions. Changes since V1:- o Split out refactoring of OBJECT_TYPE to SND_TPLG_TYPE_ o Removed inclusing of type_compat.h from asoc.h Liam Girdwood (1): core: add convenience macros to local.h Mengdong Lin (2): topology: Add C templates structure for building topology from C programs topology: A API calls to directly build topology data from templates include/local.h | 7 ++ include/topology.h | 202 ++++++++++++++++++++++++++++++++ src/topology/ctl.c | 292 ++++++++++++++++++++++++++++++++++++++++++++++ src/topology/dapm.c | 183 ++++++++++++++++++++++++++--- src/topology/elem.c | 17 +++ src/topology/parser.c | 59 +++++++++- src/topology/tplg_local.h | 26 ++--- 7 files changed, 749 insertions(+), 37 deletions(-) -- 2.1.4

2 4

[alsa-devel] [PATCH v3] ASoC: fsl-sai: add 32 bit word length support
by Zidan Wang 11 Aug '15

11 Aug '15

Add 32 bit word length support. There are no code changes required in the SAI driver since it has already wirten the word width to the corresponding register. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..d137290 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -13,7 +13,8 @@ #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) /* SAI Register Map Register */ #define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rt5640: fix line out no sound issue
by John Lin 11 Aug '15

11 Aug '15

The power for line out was not turned on when line out is enabled. So we add "LOUT amp" widget to turn on the power for line out. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5640.c | 40 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 37 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 7d488d8..3c5b496 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -985,6 +985,35 @@ static int rt5640_hp_event(struct snd_soc_dapm_widget *w, return 0; } +static int rt5640_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LM, RT5640_PWR_LM); + snd_soc_update_bits(codec, RT5640_OUTPUT, + RT5640_L_MUTE | RT5640_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5640_OUTPUT, + RT5640_L_MUTE | RT5640_R_MUTE, + RT5640_L_MUTE | RT5640_R_MUTE); + snd_soc_update_bits(codec, RT5640_PWR_ANLG1, + RT5640_PWR_LM, 0); + break; + + default: + return 0; + } + + return 0; +} + static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1180,13 +1209,16 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_event, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, @@ -1501,8 +1533,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HP R Playback", "Switch", "HP Amp"}, {"HPOL", NULL, "HP L Playback"}, {"HPOR", NULL, "HP R Playback"}, - {"LOUTL", NULL, "LOUT MIX"}, - {"LOUTR", NULL, "LOUT MIX"}, + + {"LOUT amp", NULL, "LOUT MIX"}, + {"LOUTL", NULL, "LOUT amp"}, + {"LOUTR", NULL, "LOUT amp"}, }; static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v5] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 11 Aug '15

11 Aug '15

Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is available for clock configure, using MCLK to provide sysclk directly, otherwise, search a available pll out frequcncy and set pll. Configure clock in hw_params may cause problems when using bypass style paths without hw_params in machine driver getting called. So add configure clock to set_bias_level. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 220 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8960.h | 1 + 2 files changed, 180 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 1ed0720..e3b7d0c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -126,9 +129,12 @@ struct wm8960_priv { struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; bool deemph; - int playback_fs; + int lrclk; int bclk; int sysclk; + int clk_id; + int freq_in; + bool is_stream_in_use[2]; struct wm8960_data pdata; }; @@ -164,8 +170,8 @@ static int wm8960_set_deemph(struct snd_soc_codec *codec) if (wm8960->deemph) { best = 1; for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - wm8960->playback_fs) < - abs(deemph_settings[best] - wm8960->playback_fs)) + if (abs(deemph_settings[i] - wm8960->lrclk) < + abs(deemph_settings[best] - wm8960->lrclk)) best = i; } @@ -565,6 +571,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +583,110 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int sysclk, bclk, lrclk, freq_out, freq_in; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); - u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; + } + + if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; } - if (!wm8960->sysclk) { - dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + freq_in = wm8960->freq_in; + bclk = wm8960->bclk; + lrclk = wm8960->lrclk; + /* + * If it's sysclk auto mode, check if the MCLK can provide sysclk or + * not. If MCLK can provide sysclk, using MCLK to provide sysclk + * directly. Otherwise, auto select a available pll out frequency + * and set PLL. + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) { + /* disable the PLL and using MCLK to provide sysclk */ + wm8960_set_pll(codec, 0, 0); + freq_out = freq_in; + } else if (wm8960->sysclk) { + freq_out = wm8960->sysclk; + } else { + dev_err(codec->dev, "No SYSCLK configured\n"); + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk == dac_divs[j] * lrclk) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + if (i != ARRAY_SIZE(sysclk_divs)) { + goto configure_clock; + } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } + /* get a available pll out frequency and set pll */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = lrclk * dac_divs[j]; + freq_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, freq_out)) { + wm8960_set_pll(codec, + freq_in, freq_out); break; + } else { + continue; + } } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } - /* - * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC - * pin is used as a frame clock for ADCs and DACs. - */ - if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); +configure_clock: + /* configure sysclk clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); + + /* configure frame clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -667,9 +725,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8960->lrclk = params_rate(params); /* Update filters for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - wm8960->playback_fs = params_rate(params); + if (tx) { wm8960_set_deemph(codec); } else { for (i = 0; i < ARRAY_SIZE(alc_rates); i++) @@ -682,7 +740,23 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); + wm8960->is_stream_in_use[tx] = true; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON && + !wm8960->is_stream_in_use[!tx]) + return wm8960_configure_clocking(codec); + + return 0; +} + +static int wm8960_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + wm8960->is_stream_in_use[tx] = false; return 0; } @@ -702,6 +776,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int ret; switch (level) { @@ -721,11 +796,22 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, } } + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + /* Set VMID to 2x50k */ snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); break; @@ -780,6 +866,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int reg, ret; switch (level) { @@ -831,9 +918,21 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, return ret; } } + + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); @@ -892,6 +991,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -943,10 +1064,9 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec = codec_dai->codec; u16 reg; static struct _pll_div pll_div; int ret; @@ -986,6 +1106,20 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + if (pll_id == WM8960_SYSCLK_AUTO) + return 0; + + return wm8960_set_pll(codec, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1043,11 +1177,14 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, WM8960_SYSCLK_PLL); break; + case WM8960_SYSCLK_AUTO: + break; default: return -EINVAL; } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } @@ -1060,6 +1197,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, + .hw_free = wm8960_hw_free, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index 2d8163d..ab3220d 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -82,6 +82,7 @@ #define WM8960_SYSCLK_MCLK (0 << 0) #define WM8960_SYSCLK_PLL (1 << 0) +#define WM8960_SYSCLK_AUTO (2 << 0) #define WM8960_DAC_DIV_1 (0 << 3) #define WM8960_DAC_DIV_1_5 (1 << 3) -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: rsnd: Silence DMA slave ID compile warning on 64-bit
by Geert Uytterhoeven 11 Aug '15

11 Aug '15

On arm64: sound/soc/sh/rcar/dma.c: In function 'rsnd_dmaen_init': sound/soc/sh/rcar/dma.c:180:9: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] (void *)id); ^ include/linux/dmaengine.h:1185:75: note: in definition of macro 'dma_request_channel' #define dma_request_channel(mask, x, y) __dma_request_channel(&(mask), x, y) ^ Add an intermediate cast to "uintptr_t" to kill the compile warning. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/rcar/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 305b129298538850..bfbb8a5e93bdce6c 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -177,7 +177,7 @@ static int rsnd_dmaen_init(struct rsnd_dai_stream *io, dma_cap_set(DMA_SLAVE, mask); dmaen->chan = dma_request_channel(mask, shdma_chan_filter, - (void *)id); + (void *)(uintptr_t)id); } if (IS_ERR_OR_NULL(dmaen->chan)) { dmaen->chan = NULL; -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/3] topology: Add external API for building topology data
by Liam Girdwood 11 Aug '15

11 Aug '15

Currently we can build topology binary data files from topology text files. However it is sometimes necessary for DSP FW vendors to build topology binary data within a toolset and create topology binary data directly from within vendor tools. This series adds an API to the alsa-lib topology core so that vendor tools can create topology data directly. Changes since V1:- o Split out refactoring of OBJECT_TYPE to SND_TPLG_TYPE_ o Removed inclusing of type_compat.h from asoc.h Liam Girdwood (1): topology: rename OBJECT_TYPE_ to SND_TPLG_TYPE_ Mengdong Lin (2): topology: Add C templates structure for building topology from C programs topology: A API calls to directly build topology data from templates include/topology.h | 202 ++++++++++++++++++++++++++++ src/topology/builder.c | 38 +++--- src/topology/ctl.c | 329 +++++++++++++++++++++++++++++++++++++++++++--- src/topology/dapm.c | 223 ++++++++++++++++++++++++++----- src/topology/data.c | 10 +- src/topology/elem.c | 43 ++++-- src/topology/parser.c | 63 ++++++++- src/topology/pcm.c | 34 ++--- src/topology/text.c | 2 +- src/topology/tplg_local.h | 33 ++--- 10 files changed, 847 insertions(+), 130 deletions(-) -- 2.1.4

3 9

[alsa-devel] [RESEND PATCH] mfd: wm8994: Fix NULL pointer exception on missing pdata
by Inha Song 11 Aug '15

11 Aug '15

If we use the dev_get_platdata, pdata can be NULL in DT environments. Actually, NULL pointer exception is occurs with error messages like this: [ 1.226583] Unable to handle kernel NULL pointer dereference at virtual address 00000044 [ 1.467910] [<c02f1e38>] (wm8994_irq_init) from [<c02f1bd4>] (wm8994_i2c_probe+0x684/0x858) [ 1.476241] [<c02f1bd4>] (wm8994_i2c_probe) from [<c0371c78>] (i2c_device_probe+0x11c/0x17c) In wm8994-core, we already checked whether platform data is NULL and set the wm8994->pdata to platform data if platform data is not NULL. So, we can use wm8994->pdata instead of platform data. Signed-off-by: Inha Song <ideal.song(a)samsung.com> Acked-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm8994-irq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/mfd/wm8994-irq.c b/drivers/mfd/wm8994-irq.c index 55c380a..55c59d9 100644 --- a/drivers/mfd/wm8994-irq.c +++ b/drivers/mfd/wm8994-irq.c @@ -193,7 +193,7 @@ int wm8994_irq_init(struct wm8994 *wm8994) { int ret; unsigned long irqflags; - struct wm8994_pdata *pdata = dev_get_platdata(wm8994->dev); + struct wm8994_pdata *pdata = &wm8994->pdata; if (!wm8994->irq) { dev_warn(wm8994->dev, -- 2.0.0.390.gcb682f8

2 1

[alsa-devel] [PATCH] topology: fix output file unlinking
by Liam Girdwood 11 Aug '15

11 Aug '15

Unlinking is unecessary as the old file will be overwritten with the new as intended. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/parser.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/src/topology/parser.c b/src/topology/parser.c index ed25bb8..5544517 100644 --- a/src/topology/parser.c +++ b/src/topology/parser.c @@ -266,9 +266,6 @@ int snd_tplg_build_file(snd_tplg_t *tplg, const char *infile, snd_config_t *cfg = NULL; int err = 0; - /* delete any old output files */ - unlink(outfile); - tplg->out_fd = open(outfile, O_RDWR | O_CREAT, S_IRWXU | S_IRWXG | S_IRWXO); if (tplg->out_fd < 0) { -- 2.1.4

2 2

[alsa-devel] [PATCH 0/2] ALSA: fireworks/firewire-lib: add support for recent firmware quirk
by Takashi Sakamoto 11 Aug '15

11 Aug '15

Some users have reported current ALSA Fireworks driver stops packet streaming due to detecting discontinuity. [alsa-devel] ALSA snd-fireworks Echo AudioFire 2 http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080379.html [alsa-devel] AudioFire4 quirk http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095697.html Linux 3.16 or later have this issue. As long as I investigate, this issue depends on firmware version installed in these devices. While, it's not clear that which version of firmwares for which model has this quirk. At least, firmware version 5.7.0, 5.7.3 and 5.8.0 for AudioFire2/AudioFire4/AudioFirePre8 have this quirk. This patchset applies a solution for this issue, with my assumption that 'the same firmware binary has the same quirk'. These models, AudioFire8 (since Jul 2009) and Gibson Robot Interface Pack (RIP) series uses the same firmware binary in driver package. If possible, the second patch should be applied to all of stable kernels. This patchset updates my previous RFC: [alsa-devel] [RFC][PATCH 0/2] ALSA: fireworks/firewire-lib: add support for recent firmware quirk http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/095849.html Takashi Sakamoto (2): Revert "ALSA: fireworks: add support for AudioFire2 quirk" ALSA: fireworks/firewire-lib: add support for recent firmware quirk sound/firewire/amdtp.c | 5 +++-- sound/firewire/amdtp.h | 2 ++ sound/firewire/fireworks/fireworks.c | 10 ++++++++-- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_stream.c | 12 +++++++++--- 5 files changed, 23 insertions(+), 8 deletions(-) -- 2.1.4

2 7

[alsa-devel] [PATCH v4] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 11 Aug '15

11 Aug '15

Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is available for clock configure, using MCLK to provide sysclk directly, otherwise, search a available pll out frequcncy and set pll. Configure clock in hw_params may cause problems when using bypass style paths without hw_params in machine driver getting called. So add configure clock to set_bias_level. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 220 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8960.h | 1 + 2 files changed, 180 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 1ed0720..e7ee49a 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -126,9 +129,12 @@ struct wm8960_priv { struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; bool deemph; - int playback_fs; + int lrclk; int bclk; int sysclk; + int clk_id; + int freq_in; + bool is_stream_in_use[2]; struct wm8960_data pdata; }; @@ -164,8 +170,8 @@ static int wm8960_set_deemph(struct snd_soc_codec *codec) if (wm8960->deemph) { best = 1; for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - wm8960->playback_fs) < - abs(deemph_settings[best] - wm8960->playback_fs)) + if (abs(deemph_settings[i] - wm8960->lrclk) < + abs(deemph_settings[best] - wm8960->lrclk)) best = i; } @@ -565,6 +571,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +583,110 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int sysclk, bclk, lrclk, freq_out, freq_in; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); - u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; + } + + if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; } - if (!wm8960->sysclk) { - dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + freq_in = wm8960->freq_in; + bclk = wm8960->bclk; + lrclk = wm8960->lrclk; + /* + * If it's sysclk auto mode, check if the MCLK can provide sysclk or + * not. If MCLK can provide sysclk, using MCLK to provide sysclk + * directly. Otherwise, auto select a available pll out frequency + * and set PLL. + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) { + /* disable the PLL and using MCLK to provide sysclk */ + wm8960_set_pll(codec, 0, 0); + freq_out = freq_in; + } else if (wm8960->sysclk) { + freq_out = wm8960->sysclk; + } else { + dev_err(codec->dev, "No SYSCLK configured\n"); + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk == dac_divs[j] * lrclk) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + if (i != ARRAY_SIZE(sysclk_divs)) { + goto configure_clock; + } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } + /* get a available pll out frequency and set pll */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = lrclk * dac_divs[j]; + freq_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, freq_out)) { + wm8960_set_pll(codec, + freq_in, freq_out); break; + } else { + continue; + } } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } - /* - * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC - * pin is used as a frame clock for ADCs and DACs. - */ - if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); +configure_clock: + /* configure sysclk clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); + + /* configure frame clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -667,9 +725,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8960->lrclk = params_rate(params); /* Update filters for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - wm8960->playback_fs = params_rate(params); + if (tx) { wm8960_set_deemph(codec); } else { for (i = 0; i < ARRAY_SIZE(alc_rates); i++) @@ -682,7 +740,23 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); + wm8960->is_stream_in_use[tx] = true; + + if (codec->dapm.bias_level == SND_SOC_BIAS_ON && + !wm8960->is_stream_in_use[!tx]) + return wm8960_configure_clocking(codec); + + return 0; +} + +static int wm8960_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + wm8960->is_stream_in_use[tx] = false; return 0; } @@ -702,6 +776,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int ret; switch (level) { @@ -721,11 +796,22 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, } } + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + /* Set VMID to 2x50k */ snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); break; @@ -780,6 +866,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int reg, ret; switch (level) { @@ -831,9 +918,21 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, return ret; } } + + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); @@ -892,6 +991,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -943,10 +1064,9 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec = codec_dai->codec; u16 reg; static struct _pll_div pll_div; int ret; @@ -986,6 +1106,20 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + if (pll_id == WM8960_SYSCLK_AUTO) + return 0; + + return wm8960_set_pll(codec, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1043,11 +1177,14 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, WM8960_SYSCLK_PLL); break; + case WM8960_SYSCLK_AUTO: + break; default: return -EINVAL; } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } @@ -1060,6 +1197,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, + .hw_free = wm8960_hw_free, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index 2d8163d..ab3220d 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -82,6 +82,7 @@ #define WM8960_SYSCLK_MCLK (0 << 0) #define WM8960_SYSCLK_PLL (1 << 0) +#define WM8960_SYSCLK_AUTO (2 << 0) #define WM8960_DAC_DIV_1 (0 << 3) #define WM8960_DAC_DIV_1_5 (1 << 3) -- 1.9.1

2 2

[alsa-devel] [PATCH 1/3] ASoC: fsl-sai: add 32 bit word length support
by Zidan Wang 11 Aug '15

11 Aug '15

Add 32 bit word length support. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..40280ce 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -1,5 +1,5 @@ /* - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 Freescale Semiconductor, Inc. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -13,7 +13,8 @@ #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) /* SAI Register Map Register */ #define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ -- 1.9.1

3 8

[alsa-devel] [PATCH] jack: Fixing a small memory leak
by Valentin Corfu 11 Aug '15

11 Aug '15

A small memory leak can be observed when using alsa-plugins (jack). Check the following valgrind log: ... ==1034== 8 bytes in 1 blocks are definitely lost in loss record 7 of 108 ==1034== at 0x4027E5A: calloc (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==1034== by 0x4035102: snd_pcm_jack_prepare (in /usr/lib/alsa-lib/libasound_module_pcm_jack.so) ==1034== by 0x40D07AB: snd_pcm_ioplug_prepare (pcm_ioplug.c:145) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x4084A1D: snd_pcm_hw_params (pcm.c:838) ==1034== by 0x4090013: snd_pcm_set_params (pcm.c:8053) ==1034== by 0x80486EB: main (in /root/pcm_min_ok) ... ==1034== LEAK SUMMARY: ==1034== definitely lost: 8 bytes in 1 blocks ==1034== indirectly lost: 0 bytes in 0 blocks ==1034== possibly lost: 31,311 bytes in 1,630 blocks ==1034== still reachable: 5,619 bytes in 24 blocks ==1034== suppressed: 0 bytes in 0 blocks ... Signed-off-by: Valentin Corfu <corfuvalentin(a)gmail.com> --- jack/pcm_jack.c | 1 + 1 file changed, 1 insertion(+) diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 568ec43..de9c46e 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -103,6 +103,7 @@ static void snd_pcm_jack_free(snd_pcm_jack_t *jack) if (jack->io.poll_fd >= 0) close(jack->io.poll_fd); free(jack->areas); + free(jack->ports); free(jack); } } -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: nau8825: Add driver for headset chip Nuvoton 8825
by Anatol Pomozov 11 Aug '15

11 Aug '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/nau8825.txt | 68 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 944 +++++++++++++++++++++ sound/soc/codecs/nau8825.h | 276 ++++++ 5 files changed, 1294 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 0000000..46e1868 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,68 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-pullup: Pull-up JKDET pin. Boolean. + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-threshold-num: Number of buttons supported + - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) + where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold calculation. + + - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. + + - nuvoton,sar-voltage: Reference voltage for button impedance measurement. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,short-key-debounce: Button short key press debounce time. + 0 - 30ms + 1 - 50ms + 2 - 100ms + 3 - 30ms + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + +Example: + + headset: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-pullup; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedane according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0x14 0x29 0x40 0x78>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7fcbc22..1a3639c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,6 +78,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -892,6 +893,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b46c8f..4874881 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -262,6 +263,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..96eed3a --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,944 @@ +/* + * Nuvoton NAU8825 audio codec driver + * + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * Copyright 2015 Nuvoton Technology Corp. + * Co-author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "nau8825.h" + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + int irq; + struct snd_soc_jack *jack; + int button_pressed; + int micbias_voltage; + bool jkdet_pullup; + int sar_threshold_num; + int sar_threshold[8]; + int sar_hysteresis; + int sar_voltage; + int key_debounce; + int jack_insert_debounce; + int jack_eject_debounce; +}; + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static const char * const nau8825_adc_decimation[] = { + "32", "64", "128", "256" +}; + +static const struct soc_enum nau8825_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8825_REG_ADC_RATE, NAU8825_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8825_adc_decimation), nau8825_adc_decimation); + +static const char * const nau8825_dac_oversampl[] = { + "64", "256", "128", "", "32" +}; + +static const struct soc_enum nau8825_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8825_dac_oversampl), nau8825_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); + +static const struct snd_kcontrol_new nau8825_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Sidetone Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Gain", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum), +}; + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, + 15, 1), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, + 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Change Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("DACL Power", NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, + 0), + SND_SOC_DAPM_SUPPLY("DACR Power", NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, + 0), + + SND_SOC_DAPM_OUT_DRV("Output Driver R Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver L Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver R Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver L Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver R Stage 3", + NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Output Driver L Stage 3", + NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"ADC", NULL, "ADC Power"}, + {"AIFTX", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DDACL", NULL, "DACL Power"}, + {"DDACR", NULL, "DACR Power"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "Change Pump"}, + {"HP amp R", NULL, "Change Pump"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver L Stage 1", NULL, "ADACL"}, + {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"HPOL", NULL, "Output Driver L Stage 3"}, + {"HPOR", NULL, "Output Driver R Stage 3"}, +}; + +static int nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int reg_val; + + reg_val = mute ? NAU8825_HP_MUTE : 0; + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HP_MUTE, reg_val); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .digital_mute = nau8825_mute, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->jack = jack; + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons; + + /* The chip supports up to 8 buttons, but ALSA defines only 6 buttons */ + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + if (value & BIT(4)) + buttons |= SND_JACK_BTN_4; + if (value & BIT(5)) + buttons |= SND_JACK_BTN_5; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + lower 8 bits - for long pressed buttons */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_TRACKING_GAIN_MASK, + nau8825->sar_voltage << NAU8825_SAR_TRACKING_GAIN_SFT); + + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_LEVELS_NR_MASK, + (nau8825->sar_threshold_num - 1) << NAU8825_KEYDET_LEVELS_NR_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_HYSTERESIS_MASK, + nau8825->sar_hysteresis << NAU8825_KEYDET_HYSTERESIS_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK, + nau8825->key_debounce << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT); + + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, + (nau8825->sar_threshold[0] << 8) | nau8825->sar_threshold[1]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, + (nau8825->sar_threshold[2] << 8) | nau8825->sar_threshold[3]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_3, + (nau8825->sar_threshold[4] << 8) | nau8825->sar_threshold[5]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_4, + (nau8825->sar_threshold[6] << 8) | nau8825->sar_threshold[7]); + + /* Enable short press and release interruptions */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_KEY_SHORT_PRESS_EN | NAU8825_IRQ_KEY_RELEASE_EN, 0); + + /* Enable DDACR needed for interrupts + * It is the same as force_enable_pin("DDACR") we do later */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, 0x0400, 0x0400); +} + +static void nau8825_init_regs(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* VMID Enable and Tieoff */ + regmap_write(regmap, NAU8825_REG_BIAS_ADJ, 0x0060); + /* Analog Bias Enable, Disable Boost Driver, + Automatic Short circuit protection enable */ + regmap_write(regmap, NAU8825_REG_BOOST, 0x3140); + /* Ground HP Outputs[1:0], + Enable Automatic Mic/Gnd switching reading on insert interrupt[6] */ + regmap_write(regmap, NAU8825_REG_HSD_CTRL, 0x004f); + + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_SELECT, + nau8825->jkdet_pullup ? NAU8825_JKDET_PULL_SELECT : 0); + + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_INSERT_DEBOUNCE_MASK, + nau8825->jack_insert_debounce << NAU8825_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_EJECT_DEBOUNCE_MASK, + nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT); + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + /* Unmask needed IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_VOLTAGE_MASK, nau8825->micbias_voltage); + + if (nau8825->sar_threshold_num) + nau8825_setup_buttons(nau8825); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->dapm = &codec->dapm; + + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "DDACR"); + snd_soc_dapm_sync(&codec->dapm); + + /* Unmask interruptions. Handler uses dapm object so we can enable + * interruptions only after dapm is fully initialized. */ + regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, + unsigned int freq) +{ + struct regmap *regmap = nau8825->regmap; + + switch (clk_id) { + case NAU8825_CLK_MCLK: + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + break; + case NAU8825_CLK_INTERNAL: + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, + NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + break; + default: + dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8825->dev, "Sysclk is %dHz and clock id is %d\n", freq, + clk_id); + return 0; +} + +int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, + unsigned int freq, int dir) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + return nau8825_configure_sysclk(nau8825, clk_id, freq); +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + .set_sysclk = nau8825_set_sysclk, + + .controls = nau8825_controls, + .num_controls = ARRAY_SIZE(nau8825_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825; + int ret, value; + + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->irq = i2c->irq; + nau8825->dev = dev; + + nau8825->jkdet_pullup = device_property_read_bool(dev, + "nuvoton,jkdet-pullup"); + device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8825->micbias_voltage); + device_property_read_u32(dev, "nuvoton,sar-threshold-num", + &nau8825->sar_threshold_num); + device_property_read_u32_array(dev, "nuvoton,sar-threshold", + nau8825->sar_threshold, nau8825->sar_threshold_num); + device_property_read_u32(dev, "nuvoton,sar-hysteresis", + &nau8825->sar_hysteresis); + device_property_read_u32(dev, "nuvoton,sar-voltage", + &nau8825->sar_voltage); + device_property_read_u32(dev, "nuvoton,short-key-debounce", + &nau8825->key_debounce); + device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8825->jack_insert_debounce); + device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8825->jack_eject_debounce); + + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + nau8825_init_regs(nau8825); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + * Enables master mode and correctly divides BCLK and FSCLK from MCLK. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + if (nau8825->irq) { + int ret = devm_request_threaded_irq(dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + } + + return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..bb103f0 --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,276 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov(a)chrominium.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* FLL6 (0x9) */ +#define NAU8825_DCO_EN (1 << 15) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +#define NAU8825_SPKR_DWN1R_SHORT (1 << 1) +#define NAU8825_SPKR_DWN1L_SHORT (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8825_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_EJECT_DEBOUNCE_SFT) + +/* INTERRUPT_MASK (0xf) */ +#define NAU8825_IRQ_OUTPUT_EN (1 << 11) +#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) +#define NAU8825_IRQ_EJECT_EN (1 << 2) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) +#define NAU8825_SAR_TRACKING_GAIN_SFT 8 +#define NAU8825_SAR_TRACKING_GAIN_MASK (0x7 << NAU8825_SAR_TRACKING_GAIN_SFT) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT 12 +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK (0x3 << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT) +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) +#define NAU8825_KEYDET_HYSTERESIS_SFT 0 +#define NAU8825_KEYDET_HYSTERESIS_MASK 0xf + +/* GPIO12_CTRL (0x1a) */ +/* 0 - pull down, 1 - pull up */ +#define NAU8825_JKDET_PULL_SELECT (1 << 11) + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - Normal mode, 1 - Output Hi-Z */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SYNC_DOWN_SFT 0 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_SFT 0 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_MASK (3 << 4) +#define NAU8825_BIAS_VMID_SEL_OPEN (0 << 4) +#define NAU8825_BIAS_VMID_SEL_25KOHM (1 << 4) +#define NAU8825_BIAS_VMID_SEL_125KOHM (2 << 4) +#define NAU8825_BIAS_VMID_SEL_2_5KOHM (3 << 4) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 +#define NAU8825_MICBIAS_VOLTAGE_SFT 0 +#define NAU8825_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_DIS (1 << 12) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_INTEGR_R (1 << 5) + +/* CHARGE_PUMP_AND_POWER_DOWN_CONTROL (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- 2.5.0

1 0

[alsa-devel] [PATCH 1/4] drm/i915: Add audio set_ncts callback
by libin.yang＠intel.com 11 Aug '15

11 Aug '15

From: Libin Yang <libin.yang(a)intel.com> Add the set_ncts callback. With the callback, audio driver can trigger i915 driver to set the proper N/CTS based on different sample rates. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/drm/i915_component.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index c9a8b64..7305881 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -33,6 +33,8 @@ struct i915_audio_component { void (*put_power)(struct device *); void (*codec_wake_override)(struct device *, bool enable); int (*get_cdclk_freq)(struct device *); + int (*set_ncts)(struct device *, int port, int dev_entry, + int rate); } *ops; }; -- 1.9.1

4 11

[alsa-devel] [PATCH v2] ASoC: fsl-sai: add 32 bit word length support
by Zidan Wang 11 Aug '15

11 Aug '15

Add 32 bit word length support. There are no code changes required in the SAI driver since it has already wirten the word width to the corresponding register. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/fsl/fsl_sai.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 0662809..40280ce 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -1,5 +1,5 @@ /* - * Copyright 2012-2013 Freescale Semiconductor, Inc. + * Copyright 2012-2015 Freescale Semiconductor, Inc. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -13,7 +13,8 @@ #define FSL_SAI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) /* SAI Register Map Register */ #define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ -- 1.9.1

2 2

[alsa-devel] jack: Fixing a small memory leak
by Valentin Corfu 10 Aug '15

10 Aug '15

Hello Alsa developers, I observed a small memory leak in alsa-plugins (jack): ... ==1034== ==1034== 8 bytes in 1 blocks are definitely lost in loss record 7 of 108 ==1034== at 0x4027E5A: calloc (in /usr/lib/valgrind/vgpreload_memcheck-x86-linux.so) ==1034== by 0x4035102: snd_pcm_jack_prepare (in /usr/lib/alsa-lib/libasound_module_pcm_jack.so) ==1034== by 0x40D07AB: snd_pcm_ioplug_prepare (pcm_ioplug.c:145) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x409DDB4: snd_pcm_plugin_prepare (pcm_plugin.c:162) ==1034== by 0x4084961: snd_pcm_prepare (pcm.c:1052) ==1034== by 0x4084A1D: snd_pcm_hw_params (pcm.c:838) ==1034== by 0x4090013: snd_pcm_set_params (pcm.c:8053) ==1034== by 0x80486EB: main (in /root/pcm_min_ok) ... ==1034== LEAK SUMMARY: ==1034== definitely lost: 8 bytes in 1 blocks ==1034== indirectly lost: 0 bytes in 0 blocks ==1034== possibly lost: 31,311 bytes in 1,630 blocks ==1034== still reachable: 5,619 bytes in 24 blocks ==1034== suppressed: 0 bytes in 0 blocks ==1034== Reachable blocks (those to which a pointer was found) are not shown. ==1034== To see them, rerun with: --leak-check=full --show-leak-kinds=all ... Best Regards, Valentin Corfu

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[alsa-devel] [PATCH] ALSA: hda - Fix pin config and mapping on Alienware 15
by Takashi Iwai 10 Aug '15

10 Aug '15

Alienware 15 has CA0132 codec for its onboard sound, but the pin config and mapping seem quite different from other Creative boards. This patch corrects them, at least, for providing the right headphone and mic jack notification, as well as removing the non-existing SPDIF pins. Even with this fix, not all stuff works perfectly yet, mainly because of the badly written ca0132 driver code -- it has too many implicit assumptions of pin configs and maps. Nevertheless, this is a small good step forward. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101981 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_ca0132.c | 87 ++++++++++++++++++++++++++++++-------------- 1 file changed, 59 insertions(+), 28 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 0f039abe9673..186792fe226e 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -763,6 +763,20 @@ enum { QUIRK_ALIENWARE, }; +static const struct hda_pintbl alienware_pincfgs[] = { + { 0x0b, 0x90170110 }, /* Builtin Speaker */ + { 0x0c, 0x411111f0 }, /* N/A */ + { 0x0d, 0x411111f0 }, /* N/A */ + { 0x0e, 0x411111f0 }, /* N/A */ + { 0x0f, 0x0321101f }, /* HP */ + { 0x10, 0x411111f0 }, /* Headset? disabled for now */ + { 0x11, 0x03a11021 }, /* Mic */ + { 0x12, 0xd5a30140 }, /* Builtin Mic */ + { 0x13, 0x411111f0 }, /* N/A */ + { 0x18, 0x411111f0 }, /* N/A */ + {} +}; + static const struct snd_pci_quirk ca0132_quirks[] = { SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15", QUIRK_ALIENWARE), {} @@ -3147,7 +3161,7 @@ static int ca0132_select_out(struct hda_codec *codec) auto_jack = spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID]; if (auto_jack) - jack_present = snd_hda_jack_detect(codec, spec->out_pins[1]); + jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_hp); else jack_present = spec->vnode_lswitch[VNID_HP_SEL - VNODE_START_NID]; @@ -3309,7 +3323,7 @@ static int ca0132_select_mic(struct hda_codec *codec) auto_jack = spec->vnode_lswitch[VNID_AMIC1_ASEL - VNODE_START_NID]; if (auto_jack) - jack_present = snd_hda_jack_detect(codec, spec->input_pins[0]); + jack_present = snd_hda_jack_detect(codec, spec->unsol_tag_amic1); else jack_present = spec->vnode_lswitch[VNID_AMIC1_SEL - VNODE_START_NID]; @@ -4617,37 +4631,54 @@ static void ca0132_config(struct hda_codec *codec) spec->multiout.num_dacs = 3; spec->multiout.max_channels = 2; - spec->num_outputs = 2; - spec->out_pins[0] = 0x0b; /* speaker out */ if (spec->quirk == QUIRK_ALIENWARE) { codec_dbg(codec, "ca0132_config: QUIRK_ALIENWARE applied.\n"); + snd_hda_apply_pincfgs(codec, alienware_pincfgs); + + spec->num_outputs = 2; + spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x0f; - } else{ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = 0x0f; + + spec->adcs[0] = 0x7; /* digital mic / analog mic1 */ + spec->adcs[1] = 0x8; /* analog mic2 */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 3; + spec->input_pins[0] = 0x12; + spec->input_pins[1] = 0x11; + spec->input_pins[2] = 0x13; + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = 0x11; + } else { + spec->num_outputs = 2; + spec->out_pins[0] = 0x0b; /* speaker out */ spec->out_pins[1] = 0x10; /* headphone out */ + spec->shared_out_nid = 0x2; + spec->unsol_tag_hp = spec->out_pins[1]; + + spec->adcs[0] = 0x7; /* digital mic / analog mic1 */ + spec->adcs[1] = 0x8; /* analog mic2 */ + spec->adcs[2] = 0xa; /* what u hear */ + + spec->num_inputs = 3; + spec->input_pins[0] = 0x12; + spec->input_pins[1] = 0x11; + spec->input_pins[2] = 0x13; + spec->shared_mic_nid = 0x7; + spec->unsol_tag_amic1 = spec->input_pins[0]; + + /* SPDIF I/O */ + spec->dig_out = 0x05; + spec->multiout.dig_out_nid = spec->dig_out; + cfg->dig_out_pins[0] = 0x0c; + cfg->dig_outs = 1; + cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; + spec->dig_in = 0x09; + cfg->dig_in_pin = 0x0e; + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } - spec->shared_out_nid = 0x2; - spec->unsol_tag_hp = spec->out_pins[1]; - - spec->adcs[0] = 0x7; /* digital mic / analog mic1 */ - spec->adcs[1] = 0x8; /* analog mic2 */ - spec->adcs[2] = 0xa; /* what u hear */ - - spec->num_inputs = 3; - spec->input_pins[0] = 0x12; - spec->input_pins[1] = 0x11; - spec->input_pins[2] = 0x13; - spec->shared_mic_nid = 0x7; - spec->unsol_tag_amic1 = spec->input_pins[0]; - - /* SPDIF I/O */ - spec->dig_out = 0x05; - spec->multiout.dig_out_nid = spec->dig_out; - cfg->dig_out_pins[0] = 0x0c; - cfg->dig_outs = 1; - cfg->dig_out_type[0] = HDA_PCM_TYPE_SPDIF; - spec->dig_in = 0x09; - cfg->dig_in_pin = 0x0e; - cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; } static int ca0132_prepare_verbs(struct hda_codec *codec) -- 2.5.0

1 0

[alsa-devel] [PATCH] ALSA: echoaudio: Use standard C definitions of true and false
by Mark Brown 10 Aug '15

10 Aug '15

The echoaudio locally defines TRUE and FALSE. Not only is this redundant given that C now has a boolean type it results in lots of warnings as other headers also define these macros, causing duplicate definitions. Fix this by removing the local defines and converting all local users to use the standard C true and false instead, simply removing the macros is less safe due to implicit inclusion of the other definitons. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/pci/echoaudio/darla20_dsp.c | 6 +++--- sound/pci/echoaudio/darla24_dsp.c | 6 +++--- sound/pci/echoaudio/echo3g_dsp.c | 16 ++++++++-------- sound/pci/echoaudio/echoaudio.c | 2 +- sound/pci/echoaudio/echoaudio.h | 7 ++----- sound/pci/echoaudio/echoaudio_3g.c | 12 ++++++------ sound/pci/echoaudio/echoaudio_dsp.c | 26 +++++++++++++------------- sound/pci/echoaudio/gina20_dsp.c | 8 ++++---- sound/pci/echoaudio/gina24_dsp.c | 22 +++++++++++----------- sound/pci/echoaudio/indigo_dsp.c | 6 +++--- sound/pci/echoaudio/indigodj_dsp.c | 6 +++--- sound/pci/echoaudio/indigodjx_dsp.c | 6 +++--- sound/pci/echoaudio/indigoio_dsp.c | 6 +++--- sound/pci/echoaudio/indigoiox_dsp.c | 6 +++--- sound/pci/echoaudio/layla20_dsp.c | 12 ++++++------ sound/pci/echoaudio/layla24_dsp.c | 26 +++++++++++++------------- sound/pci/echoaudio/mia_dsp.c | 8 ++++---- sound/pci/echoaudio/mona_dsp.c | 20 ++++++++++---------- 18 files changed, 99 insertions(+), 102 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index febee5b..320837b 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -44,18 +44,18 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_DARLA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index 7b4f6fd..8736b5e 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -44,17 +44,17 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_DARLA24_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_ESYNC; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index ae11ce1..6deb80c 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -59,8 +59,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2); chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; - chip->has_midi = TRUE; + chip->bad_board = true; + chip->has_midi = true; chip->dsp_code_to_load = FW_ECHO3G_DSP; /* Load the DSP code and the ASIC on the PCI card and get @@ -78,8 +78,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->px_analog_in = chip->bx_analog_in = 14; chip->px_digital_in = chip->bx_digital_in = 16; chip->px_num = chip->bx_num = 24; - chip->has_phantom_power = TRUE; - chip->hasnt_input_nominal_level = TRUE; + chip->has_phantom_power = true; + chip->hasnt_input_nominal_level = true; } else if (err == E3G_LAYLA3G_BOX_TYPE) { chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | @@ -106,10 +106,10 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->non_audio_spdif = FALSE; - chip->bad_board = FALSE; - chip->phantom_power = FALSE; + chip->professional_spdif = false; + chip->non_audio_spdif = false; + chip->bad_board = false; + chip->phantom_power = false; return init_line_levels(chip); } diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 862db9a..1cb85ae 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2245,7 +2245,7 @@ static int snd_echo_resume(struct device *dev) #ifdef ECHOCARD_HAS_MIDI if (chip->midi_input_enabled) - enable_midi_input(chip, TRUE); + enable_midi_input(chip, true); if (chip->midi_out) snd_echo_midi_output_trigger(chip->midi_out, 1); #endif diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 3251522..152ce15 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -153,9 +153,6 @@ #define _ECHOAUDIO_H_ -#define TRUE 1 -#define FALSE 0 - #include "echoaudio_dsp.h" @@ -378,8 +375,8 @@ struct echoaudio { */ u8 output_clock; /* Layla20 only */ char meters_enabled; /* VU-meters status */ - char asic_loaded; /* Set TRUE when ASIC loaded */ - char bad_board; /* Set TRUE if DSP won't load */ + char asic_loaded; /* Set true when ASIC loaded */ + char bad_board; /* Set true if DSP won't load */ char professional_spdif; /* 0 = consumer; 1 = professional */ char non_audio_spdif; /* 3G - only */ char digital_in_automute; /* Gina24, Layla24, Mona - only */ diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index 2fa66dc..22c786b 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -41,7 +41,7 @@ static int check_asic_status(struct echoaudio *chip) return -EIO; chip->comm_page->ext_box_status = cpu_to_le32(E3G_ASIC_NOT_LOADED); - chip->asic_loaded = FALSE; + chip->asic_loaded = false; clear_handshake(chip); send_vector(chip, DSP_VC_TEST_ASIC); @@ -55,7 +55,7 @@ static int check_asic_status(struct echoaudio *chip) if (box_status == E3G_ASIC_NOT_LOADED) return -ENODEV; - chip->asic_loaded = TRUE; + chip->asic_loaded = true; return box_status & E3G_BOX_TYPE_MASK; } @@ -243,7 +243,7 @@ static int load_asic(struct echoaudio *chip) * 48 kHz, internal clock, S/PDIF RCA mode */ if (box_type >= 0) { err = write_control_reg(chip, E3G_48KHZ, - E3G_FREQ_REG_DEFAULT, TRUE); + E3G_FREQ_REG_DEFAULT, true); if (err < 0) return err; } @@ -378,16 +378,16 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) int err, incompatible_clock; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; break; default: dev_err(chip->card->dev, diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 1a9427a..15aae2f 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -103,8 +103,8 @@ static int write_dsp(struct echoaudio *chip, u32 data) cond_resched(); } - chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */ - dev_dbg(chip->card->dev, "write_dsp: Set bad_board to TRUE\n"); + chip->bad_board = true; /* Set true until DSP re-loaded */ + dev_dbg(chip->card->dev, "write_dsp: Set bad_board to true\n"); return -EIO; } @@ -126,8 +126,8 @@ static int read_dsp(struct echoaudio *chip, u32 *data) cond_resched(); } - chip->bad_board = TRUE; /* Set TRUE until DSP re-loaded */ - dev_err(chip->card->dev, "read_dsp: Set bad_board to TRUE\n"); + chip->bad_board = true; /* Set true until DSP re-loaded */ + dev_err(chip->card->dev, "read_dsp: Set bad_board to true\n"); return -EIO; } @@ -166,7 +166,7 @@ static int read_sn(struct echoaudio *chip) /* This card has no ASIC, just return ok */ static inline int check_asic_status(struct echoaudio *chip) { - chip->asic_loaded = TRUE; + chip->asic_loaded = true; return 0; } @@ -341,11 +341,11 @@ static int load_dsp(struct echoaudio *chip, u16 *code) dev_warn(chip->card->dev, "DSP is already loaded!\n"); return 0; } - chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ + chip->bad_board = true; /* Set true until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->asic_loaded = FALSE; /* Loading the DSP code will reset the ASIC */ + chip->asic_loaded = false; /* Loading the DSP code will reset the ASIC */ - dev_dbg(chip->card->dev, "load_dsp: Set bad_board to TRUE\n"); + dev_dbg(chip->card->dev, "load_dsp: Set bad_board to true\n"); /* If this board requires a resident loader, install it. */ #ifdef DSP_56361 @@ -471,7 +471,7 @@ static int load_dsp(struct echoaudio *chip, u16 *code) } chip->dsp_code = code; /* Show which DSP code loaded */ - chip->bad_board = FALSE; /* DSP OK */ + chip->bad_board = false; /* DSP OK */ return 0; } udelay(100); @@ -951,10 +951,10 @@ static int rest_in_peace(struct echoaudio *chip) /* Stops all active pipes (just to be sure) */ stop_transport(chip, chip->active_mask); - set_meters_on(chip, FALSE); + set_meters_on(chip, false); #ifdef ECHOCARD_HAS_MIDI - enable_midi_input(chip, FALSE); + enable_midi_input(chip, false); #endif /* Go to sleep */ @@ -981,9 +981,9 @@ static int init_dsp_comm_page(struct echoaudio *chip) /* Init all the basic stuff */ chip->card_name = ECHOCARD_NAME; - chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ + chip->bad_board = true; /* Set true until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->asic_loaded = FALSE; + chip->asic_loaded = false; memset(chip->comm_page, 0, sizeof(struct comm_page)); /* Init the comm page */ diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index 5dafe92..b237757 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -48,19 +48,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_GINA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -69,7 +69,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { - chip->professional_spdif = FALSE; + chip->professional_spdif = false; return init_line_levels(chip); } diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 6971766..8eff2b4 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -52,7 +52,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 | @@ -76,7 +76,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -86,8 +86,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; + chip->professional_spdif = false; + chip->digital_in_automute = true; return init_line_levels(chip); } @@ -152,7 +152,7 @@ static int load_asic(struct echoaudio *chip) 48 kHz, internal clock, S/PDIF RCA mode */ if (!err) { control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); } return err; } @@ -226,7 +226,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) chip->sample_rate = rate; dev_dbg(chip->card->dev, "set_sample_rate: %d clock %d\n", rate, clock); - return write_control_reg(chip, control_reg, FALSE); + return write_control_reg(chip, control_reg, false); } @@ -274,7 +274,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) } chip->input_clock = clock; - return write_control_reg(chip, control_reg, TRUE); + return write_control_reg(chip, control_reg, true); } @@ -285,17 +285,17 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) int err, incompatible_clock; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_CDROM: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; break; default: dev_err(chip->card->dev, @@ -333,7 +333,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) break; } - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); spin_unlock_irq(&chip->lock); if (err < 0) return err; diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 54edd67..c97dc83 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -49,16 +49,16 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 2cf5cc0..2428b35 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -49,16 +49,16 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_DJ_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 5252863..5fbd4a3 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -47,17 +47,17 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_DJX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; err = load_firmware(chip); if (err < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 4e81787..79b68ba 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -49,16 +49,16 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_IO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 6de3f9b..1ae394e 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -47,17 +47,17 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_INDIGO_IOX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; err = load_firmware(chip); if (err < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index f2024a3..5e5b6e2 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -51,8 +51,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; - chip->has_midi = TRUE; + chip->bad_board = true; + chip->has_midi = true; chip->dsp_code_to_load = FW_LAYLA20_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | @@ -62,7 +62,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -71,7 +71,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { - chip->professional_spdif = FALSE; + chip->professional_spdif = false; return init_line_levels(chip); } @@ -113,7 +113,7 @@ static int check_asic_status(struct echoaudio *chip) u32 asic_status; int goodcnt, i; - chip->asic_loaded = FALSE; + chip->asic_loaded = false; for (i = goodcnt = 0; i < 5; i++) { send_vector(chip, DSP_VC_TEST_ASIC); @@ -127,7 +127,7 @@ static int check_asic_status(struct echoaudio *chip) if (asic_status == ASIC_ALREADY_LOADED) { if (++goodcnt == 3) { - chip->asic_loaded = TRUE; + chip->asic_loaded = true; return 0; } } diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index 4f11e81..df28e51 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -51,8 +51,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; - chip->has_midi = TRUE; + chip->bad_board = true; + chip->has_midi = true; chip->dsp_code_to_load = FW_LAYLA24_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | @@ -64,7 +64,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; if ((err = init_line_levels(chip)) < 0) return err; @@ -77,8 +77,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; + chip->professional_spdif = false; + chip->digital_in_automute = true; return init_line_levels(chip); } @@ -135,7 +135,7 @@ static int load_asic(struct echoaudio *chip) err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, FW_LAYLA24_2S_ASIC); if (err < 0) - return FALSE; + return false; /* Now give the external ASIC a little time to set up */ mdelay(10); @@ -147,7 +147,7 @@ static int load_asic(struct echoaudio *chip) 48 kHz, internal clock, S/PDIF RCA mode */ if (!err) err = write_control_reg(chip, GML_CONVERTER_ENABLE | GML_48KHZ, - TRUE); + true); return err; } @@ -241,7 +241,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) dev_dbg(chip->card->dev, "set_sample_rate: %d clock %d\n", rate, control_reg); - return write_control_reg(chip, control_reg, FALSE); + return write_control_reg(chip, control_reg, false); } @@ -287,7 +287,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) } chip->input_clock = clock; - return write_control_reg(chip, control_reg, TRUE); + return write_control_reg(chip, control_reg, true); } @@ -334,17 +334,17 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) short asic; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; asic = FW_LAYLA24_2S_ASIC; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; asic = FW_LAYLA24_2A_ASIC; break; default: @@ -383,7 +383,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) break; } - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); spin_unlock_irq(&chip->lock); if (err < 0) return err; diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index fdad079..8f612a0 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -52,19 +52,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->dsp_code_to_load = FW_MIA_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ - chip->asic_loaded = TRUE; + chip->asic_loaded = true; if ((subdevice_id & 0x0000f) == MIA_MIDI_REV) - chip->has_midi = TRUE; + chip->has_midi = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF; if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index 843c7a5..dce9e57 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -52,7 +52,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; - chip->bad_board = TRUE; + chip->bad_board = true; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT; @@ -69,7 +69,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = load_firmware(chip)) < 0) return err; - chip->bad_board = FALSE; + chip->bad_board = false; return err; } @@ -79,8 +79,8 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) static int set_mixer_defaults(struct echoaudio *chip) { chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; + chip->professional_spdif = false; + chip->digital_in_automute = true; return init_line_levels(chip); } @@ -148,7 +148,7 @@ static int load_asic(struct echoaudio *chip) 48 kHz, internal clock, S/PDIF RCA mode */ if (!err) { control_reg = GML_CONVERTER_ENABLE | GML_48KHZ; - err = write_control_reg(chip, control_reg, TRUE); + err = write_control_reg(chip, control_reg, true); } return err; @@ -356,7 +356,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) } chip->input_clock = clock; - return write_control_reg(chip, control_reg, TRUE); + return write_control_reg(chip, control_reg, true); } @@ -367,16 +367,16 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) int err, incompatible_clock; /* Set clock to "internal" if it's not compatible with the new mode */ - incompatible_clock = FALSE; + incompatible_clock = false; switch (mode) { case DIGITAL_MODE_SPDIF_OPTICAL: case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) - incompatible_clock = TRUE; + incompatible_clock = true; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) - incompatible_clock = TRUE; + incompatible_clock = true; break; default: dev_err(chip->card->dev, @@ -415,7 +415,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) break; } - err = write_control_reg(chip, control_reg, FALSE); + err = write_control_reg(chip, control_reg, false); spin_unlock_irq(&chip->lock); if (err < 0) return err; -- 2.5.0

2 2

[alsa-devel] [RFC PATCH 00/10] ASoC: support adding PCM dynamically from topology
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch set allows the topology core to create PCM devices dynamically. The user can define different DAIs in the topology files for different versions of firmware, but share a generic platform and machine driver. A dummy DAI and DAI link can be used to register the soc card and specify the platform with topology. Then real DAIs are created in platform probing phase by the topology core, and the machine driver will be notified to create relavant DAI links. We tested using topology to create FE DAI/DAI links on Broadwell. Mengdong Lin (10): ASoC: change the PCM runtime array to a list ASoC: Define soc_init_dai_link() to wrap link intialization. ASoC: soc_bind_dai_link() change 2nd argument to DAI link pointer ASoC: implement DAI links in a list. ASoC: Add support for dummy DAI links and PCM runtimes ASoC: bind new DAI links after probing components. ASoC: support adding a DAI dynamically ASoC: topology: change pass number of DAI smaller than graph ASoC: Topology: change stream formats to bitwise flag ASOC: topology: Add PCM DAIs dynamically when loading them include/sound/soc-dai.h | 1 + include/sound/soc-topology.h | 1 - include/sound/soc.h | 28 +- include/uapi/sound/asoc.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 10 +- sound/soc/fsl/imx-wm8962.c | 10 +- sound/soc/generic/simple-card.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 12 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 7 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 7 +- sound/soc/pxa/mioa701_wm9713.c | 6 +- sound/soc/samsung/bells.c | 40 +- sound/soc/samsung/littlemill.c | 32 +- sound/soc/samsung/odroidx2_max98090.c | 9 +- sound/soc/samsung/snow.c | 9 +- sound/soc/samsung/speyside.c | 12 +- sound/soc/samsung/tobermory.c | 21 +- sound/soc/soc-core.c | 642 +++++++++++++++++++-------- sound/soc/soc-dapm.c | 7 +- sound/soc/soc-pcm.c | 22 +- sound/soc/soc-topology.c | 99 ++++- sound/soc/tegra/tegra_wm8903.c | 3 +- 22 files changed, 701 insertions(+), 291 deletions(-) -- 1.9.1

3 2

[alsa-devel] writing an alsa driver
by Johnny 10 Aug '15

10 Aug '15

Hi, I have been following your progress with great anticipation. > SSP2 > http://permalink.gmane.org/gmane.linux.alsa.devel/140328 > future versions of the firmware should support SSP1 So, I'm guessing, until the firmware supports SSP1, there will not be any sound? I also tried using the firmware that supports SSP0, which Vinod posted in another thread, and get no sound. Regards, John

1 0

[alsa-devel] [PATCH v3] ASoC: wm8960: update pll and clock setting function
by Zidan Wang 10 Aug '15

10 Aug '15

Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is available for clock configure, using MCLK to provide sysclk directly, otherwise, search a available pll out frequcncy and set pll. Configure clock in hw_params may cause problems when using bypass style paths without hw_params in machine driver getting called. So add configure clock to set_bias_level. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 217 +++++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/wm8960.h | 1 + 2 files changed, 177 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 94c5c46..e9f8e41 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -48,6 +48,9 @@ #define WM8960_DISOP 0x40 #define WM8960_DRES_MASK 0x30 +static bool is_pll_freq_available(unsigned int source, unsigned int target); +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out); /* * wm8960 register cache * We can't read the WM8960 register space when we are @@ -126,9 +129,12 @@ struct wm8960_priv { struct snd_soc_dapm_widget *rout1; struct snd_soc_dapm_widget *out3; bool deemph; - int playback_fs; + int lrclk; int bclk; int sysclk; + int clk_id; + int freq_in; + bool is_stream_in_use[2]; struct wm8960_data pdata; }; @@ -164,8 +170,8 @@ static int wm8960_set_deemph(struct snd_soc_codec *codec) if (wm8960->deemph) { best = 1; for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) { - if (abs(deemph_settings[i] - wm8960->playback_fs) < - abs(deemph_settings[best] - wm8960->playback_fs)) + if (abs(deemph_settings[i] - wm8960->lrclk) < + abs(deemph_settings[best] - wm8960->lrclk)) best = i; } @@ -565,6 +571,9 @@ static struct { { 8000, 5 }, }; +/* -1 for reserved value */ +static const int sysclk_divs[] = { 1, -1, 2, -1 }; + /* Multiply 256 for internal 256 div */ static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 }; @@ -574,61 +583,109 @@ static const int bclk_divs[] = { 120, 160, 220, 240, 320, 320, 320 }; -static void wm8960_configure_clocking(struct snd_soc_codec *codec, - bool tx, int lrclk) +static int wm8960_configure_clocking(struct snd_soc_codec *codec) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int sysclk, bclk, lrclk, freq_out, freq_in; u16 iface1 = snd_soc_read(codec, WM8960_IFACE1); - u16 iface2 = snd_soc_read(codec, WM8960_IFACE2); - u32 sysclk; - int i, j; + int i, j, k; if (!(iface1 & (1<<6))) { dev_dbg(codec->dev, "Codec is slave mode, no need to configure clock\n"); - return; + return 0; + } + + if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; } - if (!wm8960->sysclk) { - dev_dbg(codec->dev, "No SYSCLK configured\n"); - return; + freq_in = wm8960->freq_in; + bclk = wm8960->bclk; + lrclk = wm8960->lrclk; + /* + * If it's sysclk auto mode, check if the MCLK can provide sysclk or + * not. If MCLK can provide sysclk, using MCLK to provide sysclk + * directly. Otherwise, auto select a available pll out frequency + * and set PLL. + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO) { + /* disable the PLL and using MCLK to provide sysclk */ + wm8960_set_pll(codec, 0, 0); + freq_out = freq_in; + } else if (wm8960->sysclk) { + freq_out = wm8960->sysclk; + } else { + dev_err(codec->dev, "No SYSCLK configured\n"); + return -EINVAL; } - if (!wm8960->bclk || !lrclk) { - dev_dbg(codec->dev, "No audio clocks configured\n"); - return; + /* check if the sysclk frequency is available. */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + sysclk = freq_out / sysclk_divs[i]; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + if (sysclk == dac_divs[j] * lrclk) { + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) + if (sysclk == bclk * bclk_divs[k] / 10) + break; + if (k != ARRAY_SIZE(bclk_divs)) + break; + } + } + if (j != ARRAY_SIZE(dac_divs)) + break; } - for (i = 0; i < ARRAY_SIZE(dac_divs); ++i) { - if (wm8960->sysclk == lrclk * dac_divs[i]) { - for (j = 0; j < ARRAY_SIZE(bclk_divs); ++j) { - sysclk = wm8960->bclk * bclk_divs[j] / 10; - if (wm8960->sysclk == sysclk) + if (i != ARRAY_SIZE(sysclk_divs)) { + goto configure_clock; + } else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; + } + /* get a available pll out frequency and set pll */ + for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) { + if (sysclk_divs[i] == -1) + continue; + for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) { + sysclk = lrclk * dac_divs[j]; + freq_out = sysclk * sysclk_divs[i]; + + for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) { + if (sysclk == bclk * bclk_divs[k] / 10 && + is_pll_freq_available(freq_in, freq_out)) { + wm8960_set_pll(codec, freq_in, freq_out); break; + } else { + continue; + } } - if(j != ARRAY_SIZE(bclk_divs)) + if (k != ARRAY_SIZE(bclk_divs)) break; } + if (j != ARRAY_SIZE(dac_divs)) + break; } - if (i == ARRAY_SIZE(dac_divs)) { - dev_err(codec->dev, "Unsupported sysclk %d\n", wm8960->sysclk); - return; + if (i == ARRAY_SIZE(sysclk_divs)) { + dev_err(codec->dev, "failed to configure clock\n"); + return -EINVAL; } - /* - * configure frame clock. If ADCLRC configure as GPIO pin, DACLRC - * pin is used as a frame clock for ADCs and DACs. - */ - if (iface2 & (1<<6)) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, i << 3); - else if (!tx) - snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, i << 6); +configure_clock: + /* configure sysclk clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 3 << 1, i << 1); + + /* configure frame clock */ + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 3, j << 3); + snd_soc_update_bits(codec, WM8960_CLOCK1, 0x7 << 6, j << 6); /* configure bit clock */ - snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, j); + snd_soc_update_bits(codec, WM8960_CLOCK2, 0xf, k); + + return 0; } static int wm8960_hw_params(struct snd_pcm_substream *substream, @@ -667,9 +724,9 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + wm8960->lrclk = params_rate(params); /* Update filters for the new rate */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - wm8960->playback_fs = params_rate(params); + if (tx) { wm8960_set_deemph(codec); } else { for (i = 0; i < ARRAY_SIZE(alc_rates); i++) @@ -682,7 +739,23 @@ static int wm8960_hw_params(struct snd_pcm_substream *substream, /* set iface */ snd_soc_write(codec, WM8960_IFACE1, iface); - wm8960_configure_clocking(codec, tx, params_rate(params)); + wm8960->is_stream_in_use[tx] = true; + + if (codec->dapm.bias_level == SND_SOC_BIAS_ON && + !wm8960->is_stream_in_use[!tx]) + return wm8960_configure_clocking(codec); + + return 0; +} + +static int wm8960_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + wm8960->is_stream_in_use[tx] = false; return 0; } @@ -702,6 +775,7 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int ret; switch (level) { @@ -711,6 +785,9 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: switch (snd_soc_codec_get_bias_level(codec)) { case SND_SOC_BIAS_STANDBY: + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; if (!IS_ERR(wm8960->mclk)) { ret = clk_prepare_enable(wm8960->mclk); if (ret) { @@ -726,6 +803,13 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); break; @@ -780,6 +864,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + u16 pm2 = snd_soc_read(codec, WM8960_POWER2); int reg, ret; switch (level) { @@ -822,6 +907,10 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, msleep(100); + ret = wm8960_configure_clocking(codec); + if (ret) + return ret; + if (!IS_ERR(wm8960->mclk)) { ret = clk_prepare_enable(wm8960->mclk); if (ret) { @@ -834,6 +923,13 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_ON: + /* + * If it's sysclk auto mode, and the pll is enabled, + * disable the pll + */ + if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1)) + wm8960_set_pll(codec, 0, 0); + if (!IS_ERR(wm8960->mclk)) clk_disable_unprepare(wm8960->mclk); @@ -892,6 +988,28 @@ struct _pll_div { u32 k:24; }; +static bool is_pll_freq_available(unsigned int source, unsigned int target) +{ + unsigned int Ndiv; + + if (source == 0 || target == 0) + return false; + + /* Scale up target to PLL operating frequency */ + target *= 4; + Ndiv = target / source; + + if (Ndiv < 6) { + source >>= 1; + Ndiv = target / source; + } + + if ((Ndiv < 6) || (Ndiv > 12)) + return false; + + return true; +} + /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) @@ -943,10 +1061,9 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_pll(struct snd_soc_codec *codec, + unsigned int freq_in, unsigned int freq_out) { - struct snd_soc_codec *codec = codec_dai->codec; u16 reg; static struct _pll_div pll_div; int ret; @@ -986,6 +1103,20 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, return 0; } +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + wm8960->freq_in = freq_in; + + if (pll_id == WM8960_SYSCLK_AUTO) + return 0; + + return wm8960_set_pll(codec, freq_in, freq_out); +} + static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) { @@ -1043,11 +1174,14 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, snd_soc_update_bits(codec, WM8960_CLOCK1, 0x1, WM8960_SYSCLK_PLL); break; + case WM8960_SYSCLK_AUTO: + break; default: return -EINVAL; } wm8960->sysclk = freq; + wm8960->clk_id = clk_id; return 0; } @@ -1060,6 +1194,7 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, static const struct snd_soc_dai_ops wm8960_dai_ops = { .hw_params = wm8960_hw_params, + .hw_free = wm8960_hw_free, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, .set_clkdiv = wm8960_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h index 2d8163d..ab3220d 100644 --- a/sound/soc/codecs/wm8960.h +++ b/sound/soc/codecs/wm8960.h @@ -82,6 +82,7 @@ #define WM8960_SYSCLK_MCLK (0 << 0) #define WM8960_SYSCLK_PLL (1 << 0) +#define WM8960_SYSCLK_AUTO (2 << 0) #define WM8960_DAC_DIV_1 (0 << 3) #define WM8960_DAC_DIV_1_5 (1 << 3) -- 1.9.1

2 5

[alsa-devel] Support for snd_pcm_link() on the rate plugin
by Christian Gruber 10 Aug '15

10 Aug '15

Dear ALSA community, I have a question regarding the usage of the function snd_pcm_link() on the rate plugin. As can be seen from its implementation (see pcm.c, line 1368), the function snd_pcm_link() calls the fast_op "link" of the first pcm device/plugin to be linked, which is a function pointer, whose destination depends on the type of pcm device/plugin to be linked. If the device is a HW-device, the function snd_pcm_hw_link() is called for instance (see pcm_hw.c). Several other plugins use the generic function snd_pcm_generic_link() (see pcm_generic.c, line 212), which forwards the function call to their slave devices/plugins, e.g. the plugins mmap_emul, linear, lfloat, mulaw, alaw, route, etc. The rate plugin instead does not support snd_pcm_link(), since the function pointer "link" is not assigned in the definition of snd_pcm_rate_fast_ops (see pcm_rate.c, line 1153). Why does the rate plugin not support snd_pcm_link()? Is their any reason to not use the generic function snd_pcm_generic_link() within the rate plugin? If yes, what's the reason for it? Thanks in advance for answering this question, Christian

1 0

[alsa-devel] [PATCH 1/2] ASoC: extend seq_notifier callback interface
by yang.a.fang＠intel.com 10 Aug '15

10 Aug '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> This is needed for some codec driver to know whether it is power up/down sequence call from dapm Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> --- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/soc-dapm.c | 5 ++--- 6 files changed, 7 insertions(+), 8 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 37d95a8..1d017a3 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -607,7 +607,7 @@ struct snd_soc_dapm_context { /* Go to BIAS_OFF in suspend if the DAPM context is idle */ unsigned int suspend_bias_off:1; void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type, int); + enum snd_soc_dapm_type, int, bool); struct device *dev; /* from parent - for debug */ struct snd_soc_component *component; /* parent component */ diff --git a/include/sound/soc.h b/include/sound/soc.h index add5097..0072e21 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -875,7 +875,7 @@ struct snd_soc_codec_driver { bool suspend_bias_off; void (*seq_notifier)(struct snd_soc_dapm_context *, - enum snd_soc_dapm_type, int); + enum snd_soc_dapm_type, int, bool); bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */ }; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 8120b0c..6f4e6c3 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -733,7 +733,7 @@ WM5100_MIXER_CONTROLS("LHPF4", WM5100_HPLP4MIX_INPUT_1_SOURCE), }; static void wm5100_seq_notifier(struct snd_soc_dapm_context *dapm, - enum snd_soc_dapm_type event, int subseq) + enum snd_soc_dapm_type event, int subseq, bool power_up) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b011253..12da3b1 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -280,7 +280,7 @@ static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, #define WM8903_DCS_MODE_START_STOP 2 static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, - enum snd_soc_dapm_type event, int subseq) + enum snd_soc_dapm_type event, int subseq, bool power_up) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 2a696db..f12a900 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -686,7 +686,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) } static void wm8996_seq_notifier(struct snd_soc_dapm_context *dapm, - enum snd_soc_dapm_type event, int subseq) + enum snd_soc_dapm_type event, int subseq, bool power_up) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1992568..ebea2d3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1528,8 +1528,7 @@ static void dapm_seq_run(struct snd_soc_card *card, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i, - cur_subseq); + i, cur_subseq, power_up); } if (cur_dapm && w->dapm != cur_dapm) @@ -1591,7 +1590,7 @@ static void dapm_seq_run(struct snd_soc_card *card, for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) if (sort[i] == cur_sort) cur_dapm->seq_notifier(cur_dapm, - i, cur_subseq); + i, cur_subseq, power_up); } list_for_each_entry(d, &card->dapm_list, list) { -- 1.7.9.5

4 7

[alsa-devel] [PATCH 00/12] Add ASoC support for AMD APUs
by Alex Deucher 10 Aug '15

10 Aug '15

Hi, This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create an amd gnb (graphics northbridge) bus which we can hang the i2s codec on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the gnb bus and the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. Patch 3 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream Thanks, Alex Chunming Zhou (2): drm/amdgpu: add amd_gnb_bus support drm/amd: add ACP 2.x register headers Maruthi Bayyavarapu (1): drm/amd: add ACP driver support (v4) Maruthi Srinivas Bayyavarapu (9): drm/amd: remove amd gnb bus default runtime pm ops drm/amd: modify ACP DMA buffer position update logic (v2) drm/amd: add ACP suspend/resume functionality drm/amd: change ACP SRAM banks used for audio drm/amd: shutdown unused ACP SRAM memory banks drm/amd: remove bitfield usage in ACP IP module ASoC: AMD: add AMD ASoC ACP-I2S driver (v2) ASoC: AMD: add suspend/resume for ACP PCM driver ASoC: AMD: add ACP PCM driver runtime PM drivers/gpu/drm/Kconfig | 5 + drivers/gpu/drm/Makefile | 1 + drivers/gpu/drm/amd/acp/Kconfig | 9 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 1134 ++++++++++ drivers/gpu/drm/amd/acp/acp_hw.h | 91 + drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/acp/include/amd_acp.h | 214 ++ drivers/gpu/drm/amd/amdgpu/Makefile | 14 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 11 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 201 ++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 40 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/bus/Kconfig | 7 + drivers/gpu/drm/amd/bus/Makefile | 4 + drivers/gpu/drm/amd/bus/amd_gnb_bus.c | 261 +++ drivers/gpu/drm/amd/include/amd_shared.h | 1 + .../gpu/drm/amd/include/asic_reg/acp/acp_2_1_d.h | 437 ++++ .../drm/amd/include/asic_reg/acp/acp_2_1_enum.h | 1198 ++++++++++ .../drm/amd/include/asic_reg/acp/acp_2_1_sh_mask.h | 1568 +++++++++++++ .../gpu/drm/amd/include/asic_reg/acp/acp_2_2_d.h | 609 ++++++ .../drm/amd/include/asic_reg/acp/acp_2_2_enum.h | 1068 +++++++++ .../drm/amd/include/asic_reg/acp/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++ .../gpu/drm/amd/include/asic_reg/acp/acp_2_3_d.h | 582 +++++ .../drm/amd/include/asic_reg/acp/acp_2_3_enum.h | 1079 +++++++++ .../drm/amd/include/asic_reg/acp/acp_2_3_sh_mask.h | 2192 +++++++++++++++++++ drivers/gpu/drm/amd/include/bus/amd_gnb_bus.h | 78 + sound/soc/Kconfig | 2 +- sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 13 + sound/soc/amd/Makefile | 11 + sound/soc/amd/acp-pcm-dma.c | 759 +++++++ sound/soc/amd/acp-rt286.c | 264 +++ 33 files changed, 14214 insertions(+), 2 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.h create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/acp/include/amd_acp.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 drivers/gpu/drm/amd/bus/Kconfig create mode 100644 drivers/gpu/drm/amd/bus/Makefile create mode 100644 drivers/gpu/drm/amd/bus/amd_gnb_bus.c create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_1_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_2_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_d.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_enum.h create mode 100644 drivers/gpu/drm/amd/include/asic_reg/acp/acp_2_3_sh_mask.h create mode 100644 drivers/gpu/drm/amd/include/bus/amd_gnb_bus.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp-rt286.c -- 1.8.3.1

5 24

[alsa-devel] [PATCH 0/3] topology: Add external API for building topology data.
by Liam Girdwood 10 Aug '15

10 Aug '15

Currently we can build topology binary data files from topology text files. However it is sometimes necessary for DSP FW vendors to build topology binary data within a toolset and create topology binary data directly from within vendor tools. This series adds an API to the alsa-lib topology core so that vendor tools can create topology data directly. Mengdong Lin (3): topology: Add C templates structure for building topology from C programs topology: A API calls to directly build topology data from templates topology: add support for adding bytes controls to widgets include/sound/asoc.h | 2 + include/topology.h | 202 ++++++++++++++++++++++++++++ src/topology/builder.c | 38 +++--- src/topology/ctl.c | 329 +++++++++++++++++++++++++++++++++++++++++++--- src/topology/dapm.c | 226 ++++++++++++++++++++++++++----- src/topology/data.c | 10 +- src/topology/elem.c | 43 ++++-- src/topology/parser.c | 63 ++++++++- src/topology/pcm.c | 34 ++--- src/topology/text.c | 2 +- src/topology/tplg_local.h | 33 ++--- 11 files changed, 849 insertions(+), 133 deletions(-) -- 2.1.4

2 5

[alsa-devel] ALSA: es1968: Use monotonic clock
by Abhilash Jindal 10 Aug '15

10 Aug '15

Wall time obtained from do_gettimeofday is susceptible to sudden jumps due to user setting the time or due to NTP. Monotonic time is constantly increasing time better suited for comparing two timestamps. --- sound/pci/es1968.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index b0e3d92..6830c31 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -1710,7 +1710,7 @@ static void es1968_measure_clock(struct es1968 *chip) int i, apu; unsigned int pa, offset, t; struct esm_memory *memory; - struct timeval start_time, stop_time; + struct timespec start_time, stop_time; if (chip->clock == 0) chip->clock = 48000; /* default clock value */ @@ -1759,12 +1759,12 @@ static void es1968_measure_clock(struct es1968 *chip) snd_es1968_bob_inc(chip, ESM_BOB_FREQ); __apu_set_register(chip, apu, 5, pa & 0xffff); snd_es1968_trigger_apu(chip, apu, ESM_APU_16BITLINEAR); - do_gettimeofday(&start_time); + ktime_get_ts(&start_time); spin_unlock_irq(&chip->reg_lock); msleep(50); spin_lock_irq(&chip->reg_lock); offset = __apu_get_register(chip, apu, 5); - do_gettimeofday(&stop_time); + ktime_get_ts(&stop_time); snd_es1968_trigger_apu(chip, apu, 0); /* stop */ snd_es1968_bob_dec(chip); chip->in_measurement = 0; @@ -1776,11 +1776,11 @@ static void es1968_measure_clock(struct es1968 *chip) offset += chip->measure_count * (CLOCK_MEASURE_BUFSIZE/2); t = stop_time.tv_sec - start_time.tv_sec; - t *= 1000000; - if (stop_time.tv_usec < start_time.tv_usec) - t -= start_time.tv_usec - stop_time.tv_usec; + t *= USEC_PER_SEC; + if (stop_time.tv_nsec < start_time.tv_nsec) + t -= (start_time.tv_nsec - stop_time.tv_nsec)/NSEC_PER_USEC; else - t += stop_time.tv_usec - start_time.tv_usec; + t += (stop_time.tv_nsec - start_time.tv_nsec)/NSEC_PER_USEC; if (t == 0) { snd_printk(KERN_ERR "?? calculation error..\n"); } else { -- 1.7.9.5

2 1

[alsa-devel] RME MADIFX mainline inclusion
by Adrian Knoth 10 Aug '15

10 Aug '15

Hi Takashi! Back in the days (late 2012 I guess), I wrote the MADIFX driver based on the existing RME hdspm.c. I never had physical access to this 1399 Euro audio interface, only the OSX driver and a shell login to one of IOSONO's machines which temporarily featured such a card. I know there are other users like University of Washington (DXARTS) and probably some more (the usual WFS suspects). Long story short: I don't have such a card and since I no longer work in Linux audio, I guess I won't have access in the foreseeable future. That said, I think the driver should be part of the mainline kernel, so it gets all the love the other drivers get (static code analysis, API changes etc.) I think I've ported everything you guys have done to hdspm: I've read the commitlog for hdspm and applied those changes to the madifx driver. Code is here: https://github.com/adiknoth/madifx/commits/tokernel Status: * PCM playback/capture working (SS and DS tested, QS untested) * MIDI working * All card settings working (e.g. TX64, SMUX, AESpro, WC-Term, WC-singlespeed...) * Slave Mode/External clock selection working * Mirror-MADI1-to-Out2+3 maybe working (untested) * Redundancy mode maybe working (untested) * ioctls implemented (see ioctl.c) * Static mixer working (fixed 1:1 mapping) * DSP NOT working. RME doesn't intend to release any information regarding the DSP. * Adjustable mixer NOT working (needs new userspace tools) * Levelmetering NOT working (maybe wrong, needs new userspace tools) I have no idea who could write the new mixer code. Apparently, demand is low, maybe zero. checkpatch reports four warnings of which one is a false-positive, the other one doesn't make sense and only two are real but somewhat ugly to fix (line over 80 with really long array indices). In its current state, can I send you three patches (Kconfig, madifx.h, madifx.c) that add said driver to sound/pci/rme9652? Cheers

6 16

[alsa-devel] DP MST audio support
by Dave Airlie 10 Aug '15

10 Aug '15

Hi, So I have a branch that makes no sound, http://cgit.freedesktop.org/~airlied/linux/log/?h=dp-mst-audio and I'm not sure where I need to turn to next, The Intel docs I've read are kinda vague, assuming you know lots of things I clearly don't. so in theory my branch, sets up the SDP stream to the monitor in the payload creation, enables the codec in the intel GPU driver, and passes the ELD to the audio driver. The audio driver uses the device list to get the presence/valid bits per device, and manages to retrieve the ELD. I even create ELD files in /proc/asound/HDMI/ that have sensible values in them So it looks like I'm just missing some routing somewhere, most likely in the audio driver, then again I could be missing a lot more than that. Just looking for any ideas or knowledge people may have locked in their brains or inside their firewalls. Dave.

4 6

[alsa-devel] [RFC PATCH 10/10] ASOC: topology: Add PCM DAIs dynamically when loading them
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Add PCM DAIs to ASoC core when loading them from topology. Embed dobj in the DAI to free it when unloading. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 2df96b1..e8f0404 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -207,6 +207,7 @@ struct snd_soc_dai_driver { const char *name; unsigned int id; unsigned int base; + struct snd_soc_dobj dobj; /* DAI driver callbacks */ int (*probe)(struct snd_soc_dai *dai); diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 5405f64..a49c134 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -71,7 +71,6 @@ struct snd_soc_dobj { union { struct snd_soc_dobj_control control; struct snd_soc_dobj_widget widget; - struct snd_soc_dobj_pcm_dai pcm_dai; }; void *private; /* core does not touch this */ }; diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 0f8fa0d..6c0dd3f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -499,6 +499,9 @@ static void remove_widget(struct snd_soc_component *comp, static void remove_pcm_dai(struct snd_soc_component *comp, struct snd_soc_dobj *dobj, int pass) { + struct snd_soc_dai_driver *dai = + container_of(dobj, struct snd_soc_dai_driver, dobj); + if (pass != SOC_TPLG_PASS_PCM_DAI) return; @@ -506,7 +509,7 @@ static void remove_pcm_dai(struct snd_soc_component *comp, dobj->ops->pcm_dai_unload(comp, dobj); list_del(&dobj->list); - kfree(dobj); + kfree(dai); } /* bind a kcontrol to it's IO handlers */ @@ -1558,12 +1561,75 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } +static unsigned int get_rates(unsigned int rate_min, unsigned int rate_max) +{ + static int pcm_rates[] = { + 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, + 64000, 88200, 96000, 176400, 192000, + }; + unsigned int rates = 0; + int i; + + for (i = 0; i < ARRAY_SIZE(pcm_rates); i++) { + if (pcm_rates[i] >= rate_min && pcm_rates[i] <= rate_max) + rates |= 1 << i; + } + + return rates; +} + +static int soc_tplg_pcm_dai_create(struct soc_tplg *tplg, + struct snd_soc_tplg_pcm_dai *dai) +{ + struct snd_soc_dai_driver *dai_drv; + struct snd_soc_pcm_stream *stream; + struct snd_soc_tplg_stream_caps *caps; + + dai_drv = kzalloc(sizeof(struct snd_soc_dai_driver), GFP_KERNEL); + if (dai_drv == NULL) + return -ENOMEM; + + dai_drv->name = dai->name; + dai_drv->id = dai->id; + dai_drv->compress_dai = dai->compress; + + if (dai->playback) { + stream = &dai_drv->playback; + caps = &dai->capconf[SND_SOC_TPLG_STREAM_PLAYBACK].caps; + + stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + stream->channels_min = caps->channels_min; + stream->channels_max = caps->channels_max; + stream->rates = get_rates(caps->rate_min, caps->rate_max); + stream->formats = caps->formats; + } + + if (dai->capture) { + stream = &dai_drv->capture; + caps = &dai->capconf[SND_SOC_TPLG_STREAM_CAPTURE].caps; + + stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + stream->channels_min = caps->channels_min; + stream->channels_max = caps->channels_max; + stream->rates = get_rates(caps->rate_min, caps->rate_max); + stream->formats = caps->formats; + } + + dai_drv->dobj.index = tplg->index; + dai_drv->dobj.ops = tplg->ops; + dai_drv->dobj.type = SND_SOC_DOBJ_PCM; + list_add(&dai_drv->dobj.list, &tplg->comp->dobj_list); + + return snd_soc_add_dai(tplg->comp, dai_drv); +} + static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg, struct snd_soc_tplg_hdr *hdr) { struct snd_soc_tplg_pcm_dai *pcm_dai; struct snd_soc_dobj *dobj; int count = hdr->count; + int i; int ret; if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) @@ -1579,26 +1645,21 @@ static int soc_tplg_pcm_dai_elems_load(struct soc_tplg *tplg, return -EINVAL; } - dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count); - tplg->pos += sizeof(struct snd_soc_tplg_pcm_dai) * count; - - dobj = kzalloc(sizeof(struct snd_soc_dobj), GFP_KERNEL); - if (dobj == NULL) - return -ENOMEM; - - /* Call the platform driver call back to register the dais */ + /* pass control to driver for optional further init */ ret = soc_tplg_pcm_dai_load(tplg, pcm_dai, count); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: PCM DAI loading failed\n"); goto err; } - dobj->type = get_dobj_type(hdr, NULL); - dobj->pcm_dai.count = count; - dobj->pcm_dai.pd = pcm_dai; - dobj->ops = tplg->ops; - dobj->index = tplg->index; - list_add(&dobj->list, &tplg->comp->dobj_list); + /* register the dais */ + for (i = 0; i < count; i++) { + soc_tplg_pcm_dai_create(tplg, pcm_dai); + pcm_dai++; + } + + dev_dbg(tplg->dev, "ASoC: adding %d PCM DAIs\n", count); + tplg->pos += sizeof(struct snd_soc_tplg_pcm_dai) * count; return 0; err: -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 09/10] ASoC: Topology: change stream formats to bitwise flag
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> The toplogy user space tool will generate this bitwise flag by using SNDRV_PCM_FORMAT_* exposed by asound.h, and the topology core will copy this flag when generating DAI streams. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 69383b0..eb2ad59 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -191,7 +191,7 @@ struct snd_soc_tplg_ctl_hdr { struct snd_soc_tplg_stream_caps { __le32 size; /* in bytes of this structure */ char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - __le64 formats[SND_SOC_TPLG_MAX_FORMATS]; /* supported formats SNDRV_PCM_FMTBIT_* */ + __le64 formats; /* supported formats SNDRV_PCM_FORMAT_* */ __le32 rates; /* supported rates SNDRV_PCM_RATE_* */ __le32 rate_min; /* min rate */ __le32 rate_max; /* max rate */ -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 07/10] ASoC: support adding a DAI dynamically
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> API snd_soc_add_dai() is defined to register a DAI dynamically, create its DAI widget, and notify the machine driver. This API can be used by the topology core to add DAIs dynamically. And a callback add_dai() is defined for the soc card. The machine driver can implement this callback. Adding a new DAI dynmaically will trigger this callback, and the machine driver can create relavant DAI link in the callback context. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 7e04ec0..2a474c8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1081,6 +1081,7 @@ struct snd_soc_card { int (*set_bias_level_post)(struct snd_soc_card *, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level); + int (*add_dai)(struct snd_soc_card *, struct snd_soc_dai_driver*); long pmdown_time; @@ -1634,6 +1635,9 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, void snd_soc_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); +int snd_soc_add_dai(struct snd_soc_component *component, + struct snd_soc_dai_driver *dai_drv); + #include <sound/soc-dai.h> #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2818709..b330676 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2792,6 +2792,98 @@ err: return ret; } +static struct snd_soc_dai *soc_register_dai(struct snd_soc_component *component, + struct snd_soc_dai_driver *dai_drv, + bool legacy_dai_naming) +{ + struct device *dev = component->dev; + struct snd_soc_dai *dai; + int ret; + + dev_dbg(dev, "ASoC: dynamically register DAI %s\n", dev_name(dev)); + + dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); + if (dai == NULL) { + ret = -ENOMEM; + goto err; + } + + /* + * Back in the old days when we still had component-less DAIs, + * instead of having a static name, component-less DAIs would + * inherit the name of the parent device so it is possible to + * register multiple instances of the DAI. We still need to keep + * the same naming style even though those DAIs are not + * component-less anymore. + */ + if (legacy_dai_naming) { + dai->name = fmt_single_name(dev, &dai->id); + } else { + dai->name = fmt_multiple_name(dev, dai_drv); + if (dai_drv->id) + dai->id = dai_drv->id; + else + dai->id = component->num_dai; + } + if (dai->name == NULL) { + kfree(dai); + ret = -ENOMEM; + goto err; + } + + dai->component = component; + dai->dev = dev; + dai->driver = dai_drv; + if (!dai->driver->ops) + dai->driver->ops = &null_dai_ops; + + list_add(&dai->list, &component->dai_list); + component->num_dai++; + + dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); + return dai; + +err: + kfree(dai); + return NULL; +} + +/** + * snd_soc_add_dai - Add a DAI dynamically with the ASoC core + * + * @component: The component the DAIs are registered for + * @dai_drv: DAI driver to use for the DAI + */ +int snd_soc_add_dai(struct snd_soc_component *component, + struct snd_soc_dai_driver *dai_drv) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct snd_soc_card *card = component->card; + struct snd_soc_dai *dai; + int ret; + + lockdep_assert_held(&client_mutex); + dai = soc_register_dai(component, dai_drv, false); + if (!dai) + return -ENOMEM; + + ret = snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret != 0) { + dev_err(component->dev, + "Failed to create DAI widgets %d\n", ret); + } + + /* Notify the machine driver a new DAI is added, so the machine driver + * can add new DAI links in the callback context. + */ + if (card && card->add_dai) + card->add_dai(card, dai_drv); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_add_dai); + static void snd_soc_component_seq_notifier(struct snd_soc_dapm_context *dapm, enum snd_soc_dapm_type type, int subseq) { -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 06/10] ASoC: bind new DAI links after probing components.
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Probing components can bring new DAI or DAI links based on the topology info. This patch finds the unbound DAI links and bind them. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f552fa0..2818709 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1169,6 +1169,19 @@ static void soc_remove_dai_links(struct snd_soc_card *card) } } +static bool soc_is_dai_link_bound(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + if (rtd->dai_link == dai_link) + return true; + } + + return false; +} + static int snd_soc_init_multicodec(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -1763,6 +1776,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai_link *dai_link; int ret, i, order; mutex_lock(&client_mutex); @@ -1853,6 +1867,21 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* no longer need dummy runtimes */ soc_remove_dummy_pcm_runtimes(card); + /* Find new DAI links added during probing components and bind them. + * Components with topology may bring new DAIs and DAI links. + */ + list_for_each_entry(dai_link, &card->dai_link_list, list) { + if (dai_link->dummy || soc_is_dai_link_bound(card, dai_link)) + continue; + + ret = soc_init_dai_link(card, dai_link); + if (ret) + goto probe_dai_err; + ret = soc_bind_dai_link(card, dai_link); + if (ret) + goto probe_dai_err; + } + /* probe all DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 05/10] ASoC: Add support for dummy DAI links and PCM runtimes
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Add dummy flag to a DAI link and PCM runtime. A dummy link will create a dummy PCM runtime. A machine driver can define dummy dai links to specify the expected platform and codec. And its dummy runtime is used to probe the platform and codec components, which can bring real DAI/DAI links based on topology. After components probing, all dummy PCM runtimes will be removed. Only non-dummy PCM runtimes will be assigned a valid number and create PCM streams. The ASoC core will find the platform for a dummy DAI link, assuming the cpu DAI can be dummy but the platform can bring the topology info. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 79891e2..7e04ec0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -970,6 +970,11 @@ struct snd_soc_dai_link { enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ + /* This DAI link is dummy. Machine drivers can define static dummy + * links to specify platform/codec with topology. + */ + bool dummy; + /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); @@ -1087,6 +1092,7 @@ struct snd_soc_card { struct list_head rtd_list; int num_rtd; + int num_rtd_dev; /* monotonic increase and exclude dummy ones */ /* optional codec specific configuration */ struct snd_soc_codec_conf *codec_conf; @@ -1181,6 +1187,7 @@ struct snd_soc_pcm_runtime { struct dentry *debugfs_dpcm_state; #endif + bool dummy; unsigned int num; /* 0-based and monotonic increasing */ struct list_head list; /* rtd list of the soc card */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 58eac57..f552fa0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -561,6 +561,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->card = card; rtd->dai_link = dai_link; + rtd->dummy = dai_link->dummy; rtd->codec_dais = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai *) * dai_link->num_codecs, @@ -583,10 +584,28 @@ static void soc_add_pcm_runtime(struct snd_soc_card *card, struct snd_soc_pcm_runtime *rtd) { list_add_tail(&rtd->list, &card->rtd_list); - rtd->num = card->num_rtd; + + if (!rtd->dummy) { + rtd->num = card->num_rtd_dev; + card->num_rtd_dev++; + } + card->num_rtd++; } +static void soc_remove_dummy_pcm_runtimes(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd, *_rtd; + + list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) { + if (rtd->dummy) { + list_del(&rtd->list); + soc_free_pcm_runtime(card, rtd); + card->num_rtd--; + } + } +} + static void soc_remove_pcm_runtimes(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd, *_rtd; @@ -1007,6 +1026,15 @@ static int soc_bind_dai_link(struct snd_soc_card *card, } } + /* Dummy link can use dummy cpu DAI but set platform to get topology */ + if (dai_link->dummy && dai_link->platform_name) { + if (!soc_find_component(NULL, dai_link->platform_name)) { + dev_err(card->dev, "ASoC: Platform %s not registered\n", + dai_link->platform_name); + goto _err_defer; + } + } + /* Single codec links expect codec and codec_dai in runtime data */ rtd->codec_dai = codec_dais[0]; rtd->codec = rtd->codec_dai->codec; @@ -1250,6 +1278,10 @@ static int soc_init_dai_link(struct snd_soc_card *card, void snd_soc_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { + /* Dummy links should be static, so overlook them */ + if (dai_link->dummy) + return; + lockdep_assert_held(&client_mutex); list_add_tail(&dai_link->list, &card->dai_link_list); card->num_dai_links++; @@ -1818,6 +1850,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } + /* no longer need dummy runtimes */ + soc_remove_dummy_pcm_runtimes(card); + /* probe all DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { @@ -2516,6 +2551,7 @@ int snd_soc_register_card(struct snd_soc_card *card) INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; + card->num_rtd_dev = 0; card->rtd_aux = devm_kzalloc(card->dev, sizeof(struct snd_soc_pcm_runtime) * -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 04/10] ASoC: implement DAI links in a list.
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Implement a dai link list for the soc card. And API snd_soc_add_dai_link() is defined for machine drivers to add DAI links dynamically based on topology. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index e2b7b0f..79891e2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1011,6 +1011,8 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + + struct list_head list; /* dai link list of the soc card */ }; struct snd_soc_codec_conf { @@ -1078,8 +1080,11 @@ struct snd_soc_card { long pmdown_time; /* CPU <--> Codec DAI links */ - struct snd_soc_dai_link *dai_link; - int num_links; + struct snd_soc_dai_link *dai_link; /* predefined links only */ + int num_links; /* predefined links only */ + struct list_head dai_link_list; /* all links except the dummy ones */ + int num_dai_links; + struct list_head rtd_list; int num_rtd; @@ -1619,6 +1624,9 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, struct device_node *of_node, struct snd_soc_dai_link *dai_link); +void snd_soc_add_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link); + #include <sound/soc-dai.h> #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ddfdd1d..58eac57 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1235,6 +1235,27 @@ static int soc_init_dai_link(struct snd_soc_card *card, return 0; } +/** + * snd_soc_add_dai_link - Add a DAI link dynamically + * @card: The ASoC card to which the DAI link is added + * @dai_link: The new DAI link to add + * + * This function adds a DAI link to the ASoC card, and creates a ASoC + * runtime for the link. + * + * Note: For adding DAI links dynamically by machine drivers based on + * the topology info when probing the platform component. And machine + * drivers can still define static DAI links in dai_link array. + */ +void snd_soc_add_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) +{ + lockdep_assert_held(&client_mutex); + list_add_tail(&dai_link->list, &card->dai_link_list); + card->num_dai_links++; +} +EXPORT_SYMBOL_GPL(snd_soc_add_dai_link); + static void soc_set_name_prefix(struct snd_soc_card *card, struct snd_soc_component *component) { @@ -1729,6 +1750,10 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto base_error; } + /* all predefined resources are ready, store non-dummy links */ + for (i = 0; i < card->num_links; i++) + snd_soc_add_dai_link(card, card->dai_link+i); + /* initialize the register cache for each available codec */ list_for_each_entry(codec, &codec_list, list) { if (codec->cache_init) @@ -2486,6 +2511,9 @@ int snd_soc_register_card(struct snd_soc_card *card) snd_soc_initialize_card_lists(card); + INIT_LIST_HEAD(&card->dai_link_list); + card->num_dai_links = 0; + INIT_LIST_HEAD(&card->rtd_list); card->num_rtd = 0; -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 03/10] ASoC: soc_bind_dai_link() change 2nd argument to DAI link pointer
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Just code refactoring, to reuse it if new DAI Links are added later based on topology in component probing phase. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e9e8956..ddfdd1d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -961,9 +961,9 @@ static struct snd_soc_dai *snd_soc_find_dai( return NULL; } -static int soc_bind_dai_link(struct snd_soc_card *card, int num) +static int soc_bind_dai_link(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_link) { - struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link_component *codecs = dai_link->codecs; struct snd_soc_dai_link_component cpu_dai_component; @@ -972,7 +972,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) const char *platform_name; int i; - dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (snd_soc_get_pcm_runtime(card, dai_link->name)) { dev_dbg(card->dev, "ASoC: dai link %s already bound\n", @@ -1717,7 +1717,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* bind DAIs */ for (i = 0; i < card->num_links; i++) { - ret = soc_bind_dai_link(card, i); + ret = soc_bind_dai_link(card, &card->dai_link[i]); if (ret != 0) goto base_error; } -- 1.9.1

1 0

[alsa-devel] [RFC PATCH 01/10] ASoC: change the PCM runtime array to a list
by mengdong.lin＠intel.com 10 Aug '15

10 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Currently the number of DAI links is statically defined by the machine driver at build time using an array. This makes it difficult to shrink/ grow the number of DAI links at runtime in order to reflect any changes in topology. We can change the DAI link array in the core to a list so that PCMs and FE DAI links can be added and deleted at runtime to reflect changes in use case and DSP topology. The machine driver can still register DAI links as an array. As the 1st step, this patch change the PCM runtime array to a list. A new PCM runtime is added to the list when a DAI link is bound successfully. Later patches will further implement the DAI link list. More: - define snd_soc_new/free_pcm_runtime() to create/free a runtime. - define soc_add_pcm_runtime() to add a runtime to the rtd list. - define soc_remove_pcm_runtimes() to clean up the runtime list. - traverse the rtd list to probe the link components and dais. - Add a field "num" to PCM runtime struct, used to specify the device number when creating the pcm device, and for a simple card to access its dai_props array. - The following 3rd party machine/platform drivers either iterate the rtd list or use snd_soc_get_pcm_runtime() to access a rtd: intel/atom/sst-mfld-platform-pcm.c intel/boards/cht_bsw_rt5672 intel/boards/cht_bsw_rt5645 fsl/imx-wm8962.c fsl/fsl-asoc-card.c samsung/bells.c samsung/littlemill.c samsung/odroidx2_max98090.c samsung/snow.c samsung/speyside.c samsung/tobermory.c pxa/mioa701_wm9713.c tegra/tegra_wm8903.c Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/include/sound/soc.h b/include/sound/soc.h index 4cef20e..e2b7b0f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1080,7 +1080,7 @@ struct snd_soc_card { /* CPU <--> Codec DAI links */ struct snd_soc_dai_link *dai_link; int num_links; - struct snd_soc_pcm_runtime *rtd; + struct list_head rtd_list; int num_rtd; /* optional codec specific configuration */ @@ -1175,6 +1175,9 @@ struct snd_soc_pcm_runtime { struct dentry *debugfs_dpcm_root; struct dentry *debugfs_dpcm_state; #endif + + unsigned int num; /* 0-based and monotonic increasing */ + struct list_head list; /* rtd list of the soc card */ }; /* mixer control */ diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..b80cf09 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -211,12 +211,15 @@ static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; struct codec_priv *codec_priv = &priv->codec_priv; struct device *dev = card->dev; unsigned int pll_out; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; if (dapm->dev != codec_dai->dev) return 0; @@ -383,11 +386,14 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, static int fsl_asoc_card_late_probe(struct snd_soc_card *card) { struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; struct codec_priv *codec_priv = &priv->codec_priv; struct device *dev = card->dev; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index b38b98c..201a70d 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -69,13 +69,16 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; struct imx_priv *priv = &card_priv; struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; if (dapm->dev != codec_dai->dev) return 0; @@ -135,12 +138,15 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, static int imx_wm8962_late_probe(struct snd_soc_card *card) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; struct imx_priv *priv = &card_priv; struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, data->clk_frequency, SND_SOC_CLOCK_IN); if (ret < 0) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 3ff76d4..95a2f91 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -45,7 +45,7 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = - &priv->dai_props[rtd - rtd->card->rtd]; + &priv->dai_props[rtd->num]; int ret; ret = clk_prepare_enable(dai_props->cpu_dai.clk); @@ -64,7 +64,7 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct simple_dai_props *dai_props = - &priv->dai_props[rtd - rtd->card->rtd]; + &priv->dai_props[rtd->num]; clk_disable_unprepare(dai_props->cpu_dai.clk); @@ -78,8 +78,7 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); - struct simple_dai_props *dai_props = - &priv->dai_props[rtd - rtd->card->rtd]; + struct simple_dai_props *dai_props = &priv->dai_props[rtd->num]; unsigned int mclk, mclk_fs = 0; int ret = 0; @@ -172,10 +171,9 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; struct simple_dai_props *dai_props; - int num, ret; + int ret; - num = rtd - rtd->card->rtd; - dai_props = &priv->dai_props[num]; + dai_props = &priv->dai_props[rtd->num]; ret = __asoc_simple_card_dai_init(codec, &dai_props->codec_dai); if (ret < 0) return ret; diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 683e501..5ae8ae2 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -777,15 +777,15 @@ static int sst_platform_remove(struct platform_device *pdev) static int sst_soc_prepare(struct device *dev) { struct sst_data *drv = dev_get_drvdata(dev); - int i; + struct snd_soc_pcm_runtime *rtd; /* suspend all pcms first */ snd_soc_suspend(drv->soc_card->dev); snd_soc_poweroff(drv->soc_card->dev); /* set the SSPs to idle */ - for (i = 0; i < drv->soc_card->num_rtd; i++) { - struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { + struct snd_soc_dai *dai = rtd->cpu_dai; if (dai->active) { send_ssp_cmd(dai, dai->name, 0); @@ -799,11 +799,11 @@ static int sst_soc_prepare(struct device *dev) static void sst_soc_complete(struct device *dev) { struct sst_data *drv = dev_get_drvdata(dev); - int i; + struct snd_soc_pcm_runtime *rtd; /* restart SSPs */ - for (i = 0; i < drv->soc_card->num_rtd; i++) { - struct snd_soc_dai *dai = drv->soc_card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &drv->soc_card->rtd_list, list) { + struct snd_soc_dai *dai = rtd->cpu_dai; if (dai->active) { sst_handle_vb_timer(dai, true); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index bdcaf46..fade1a2 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -47,12 +47,9 @@ struct cht_mc_private { static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) { - int i; - - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd; + struct snd_soc_pcm_runtime *rtd; - rtd = card->rtd + i; + list_for_each_entry(rtd, &card->rtd_list, list) { if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, strlen(CHT_CODEC_DAI))) return rtd->codec_dai; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2c9cc5b..fee43bd 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -46,12 +46,9 @@ static struct snd_soc_jack_pin cht_bsw_headset_pins[] = { static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) { - int i; + struct snd_soc_pcm_runtime *rtd; - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd; - - rtd = card->rtd + i; + list_for_each_entry(rtd, &card->rtd_list, list) { if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI, strlen(CHT_CODEC_DAI))) return rtd->codec_dai; diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index a9615a57..4f6cafe 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -81,8 +81,12 @@ static int rear_amp_power(struct snd_soc_codec *codec, int power) static int rear_amp_event(struct snd_soc_dapm_widget *widget, struct snd_kcontrol *kctl, int event) { - struct snd_soc_codec *codec = widget->dapm->card->rtd[0].codec; + struct snd_soc_card *card = widget->dapm->card; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_codec *codec; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec = rtd->codec; return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event)); } diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index e5f05e6..3dd246f 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -58,11 +58,16 @@ static int bells_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[DAI_DSP_CODEC].codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct snd_soc_codec *codec; struct bells_drvdata *bells = card->drvdata; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_DSP_CODEC].name); + codec_dai = rtd->codec_dai; + codec = codec_dai->codec; + if (dapm->dev != codec_dai->dev) return 0; @@ -99,11 +104,16 @@ static int bells_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[DAI_DSP_CODEC].codec_dai; - struct snd_soc_codec *codec = codec_dai->codec; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct snd_soc_codec *codec; struct bells_drvdata *bells = card->drvdata; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_DSP_CODEC].name); + codec_dai = rtd->codec_dai; + codec = codec_dai->codec; + if (dapm->dev != codec_dai->dev) return 0; @@ -137,14 +147,22 @@ static int bells_set_bias_level_post(struct snd_soc_card *card, static int bells_late_probe(struct snd_soc_card *card) { struct bells_drvdata *bells = card->drvdata; - struct snd_soc_codec *wm0010 = card->rtd[DAI_AP_DSP].codec; - struct snd_soc_codec *codec = card->rtd[DAI_DSP_CODEC].codec; - struct snd_soc_dai *aif1_dai = card->rtd[DAI_DSP_CODEC].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_codec *wm0010; + struct snd_soc_codec *codec; + struct snd_soc_dai *aif1_dai; struct snd_soc_dai *aif2_dai; struct snd_soc_dai *aif3_dai; struct snd_soc_dai *wm9081_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_AP_DSP].name); + wm0010 = rtd->codec; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_DSP_CODEC].name); + codec = rtd->codec; + aif1_dai = rtd->codec_dai; + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK, ARIZONA_CLK_SRC_FLL1, bells->sysclk_rate, @@ -181,7 +199,8 @@ static int bells_late_probe(struct snd_soc_card *card) return ret; } - aif2_dai = card->rtd[DAI_CODEC_CP].cpu_dai; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_CODEC_CP].name); + aif2_dai = rtd->cpu_dai; ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); if (ret != 0) { @@ -192,8 +211,9 @@ static int bells_late_probe(struct snd_soc_card *card) if (card->num_rtd == DAI_CODEC_SUB) return 0; - aif3_dai = card->rtd[DAI_CODEC_SUB].cpu_dai; - wm9081_dai = card->rtd[DAI_CODEC_SUB].codec_dai; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[DAI_CODEC_SUB].name); + aif3_dai = rtd->cpu_dai; + wm9081_dai = rtd->codec_dai; ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0); if (ret != 0) { diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 31a820e..7cb204e 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -23,9 +23,13 @@ static int littlemill_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *aif1_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + aif1_dai = rtd->codec_dai; + if (dapm->dev != aif1_dai->dev) return 0; @@ -66,9 +70,13 @@ static int littlemill_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *aif1_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + aif1_dai = rtd->codec_dai; + if (dapm->dev != aif1_dai->dev) return 0; @@ -168,9 +176,13 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_card *card = w->dapm->card; - struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *aif2_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[1].name); + aif2_dai = rtd->cpu_dai; + switch (event) { case SND_SOC_DAPM_PRE_PMU: ret = snd_soc_dai_set_pll(aif2_dai, WM8994_FLL2, @@ -245,11 +257,19 @@ static struct snd_soc_jack littlemill_headset; static int littlemill_late_probe(struct snd_soc_card *card) { - struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; - struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_codec *codec; + struct snd_soc_dai *aif1_dai; + struct snd_soc_dai *aif2_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec = rtd->codec; + aif1_dai = rtd->codec_dai; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[1].name); + aif2_dai = rtd->cpu_dai; + ret = snd_soc_dai_set_sysclk(aif1_dai, WM8994_SYSCLK_MCLK2, 32768, SND_SOC_CLOCK_IN); if (ret < 0) diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c index 596f118..0421727 100644 --- a/sound/soc/samsung/odroidx2_max98090.c +++ b/sound/soc/samsung/odroidx2_max98090.c @@ -25,10 +25,15 @@ static struct snd_soc_dai_link odroidx2_dai[]; static int odroidx2_late_probe(struct snd_soc_card *card) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + cpu_dai = rtd->cpu_dai; + ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK, SND_SOC_CLOCK_IN); diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index 7651dc9..0969577 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -35,10 +35,15 @@ static struct snd_soc_dai_link snow_dai[] = { static int snow_late_probe(struct snd_soc_card *card) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + cpu_dai = rtd->cpu_dai; + /* Set the MCLK rate for the codec */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, FIN_PLL_RATE, SND_SOC_CLOCK_IN); diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index d1ae21c..083ef5e 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -25,9 +25,13 @@ static int speyside_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[1].name); + codec_dai = rtd->codec_dai; + if (dapm->dev != codec_dai->dev) return 0; @@ -57,9 +61,13 @@ static int speyside_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[1].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[1].name); + codec_dai = rtd->codec_dai; + if (dapm->dev != codec_dai->dev) return 0; diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index 85ccfb7..3310eda 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -23,9 +23,13 @@ static int tobermory_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + if (dapm->dev != codec_dai->dev) return 0; @@ -62,9 +66,13 @@ static int tobermory_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + if (dapm->dev != codec_dai->dev) return 0; @@ -170,10 +178,15 @@ static struct snd_soc_jack_pin tobermory_headset_pins[] = { static int tobermory_late_probe(struct snd_soc_card *card) { - struct snd_soc_codec *codec = card->rtd[0].codec; - struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_codec *codec; + struct snd_soc_dai *codec_dai; int ret; + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec = rtd->codec; + codec_dai = rtd->codec_dai; + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, 32768, SND_SOC_CLOCK_IN); if (ret < 0) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1b63a03..fb78674 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -537,26 +537,76 @@ static inline void snd_soc_debugfs_exit(void) struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream) { - int i; + struct snd_soc_pcm_runtime *rtd; - for (i = 0; i < card->num_links; i++) { - if (card->rtd[i].dai_link->no_pcm && - !strcmp(card->rtd[i].dai_link->name, dai_link)) - return card->rtd[i].pcm->streams[stream].substream; + list_for_each_entry(rtd, &card->rtd_list, list) { + if (rtd->dai_link->no_pcm && + !strcmp(rtd->dai_link->name, dai_link)) + return rtd->pcm->streams[stream].substream; } dev_dbg(card->dev, "ASoC: failed to find dai link %s\n", dai_link); return NULL; } EXPORT_SYMBOL_GPL(snd_soc_get_dai_substream); +static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( + struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_pcm_runtime *rtd; + + rtd = devm_kzalloc(card->dev, + sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); + if (!rtd) + return NULL; + + rtd->card = card; + rtd->dai_link = dai_link; + rtd->codec_dais = devm_kzalloc(card->dev, + sizeof(struct snd_soc_dai *) * + dai_link->num_codecs, + GFP_KERNEL); + if (!rtd->codec_dais) + return NULL; + + return rtd; +} + +static void soc_free_pcm_runtime(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) +{ + if (rtd->codec_dai) + devm_kfree(card->dev, rtd->codec_dais); + devm_kfree(card->dev, rtd); +} + +static void soc_add_pcm_runtime(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) +{ + list_add_tail(&rtd->list, &card->rtd_list); + rtd->num = card->num_rtd; + card->num_rtd++; +} + +static void soc_remove_pcm_runtimes(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd, *_rtd; + + list_for_each_entry_safe(rtd, _rtd, &card->rtd_list, list) { + list_del(&rtd->list); + soc_free_pcm_runtime(card, rtd); + } + + card->num_rtd = 0; +} + struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card, const char *dai_link) { - int i; + struct snd_soc_pcm_runtime *rtd; - for (i = 0; i < card->num_links; i++) { - if (!strcmp(card->rtd[i].dai_link->name, dai_link)) - return &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) { + if (!strcmp(rtd->dai_link->name, dai_link)) + return rtd; } dev_dbg(card->dev, "ASoC: failed to find rtd %s\n", dai_link); return NULL; @@ -578,7 +628,8 @@ int snd_soc_suspend(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_codec *codec; - int i, j; + struct snd_soc_pcm_runtime *rtd; + int i; /* If the card is not initialized yet there is nothing to do */ if (!card->instantiated) @@ -595,13 +646,13 @@ int snd_soc_suspend(struct device *dev) snd_power_change_state(card->snd_card, SNDRV_CTL_POWER_D3hot); /* mute any active DACs */ - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; - for (j = 0; j < card->rtd[i].num_codecs; j++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dais[j]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; struct snd_soc_dai_driver *drv = dai->driver; if (drv->ops->digital_mute && dai->playback_active) @@ -610,20 +661,20 @@ int snd_soc_suspend(struct device *dev) } /* suspend all pcms */ - for (i = 0; i < card->num_rtd; i++) { - if (card->rtd[i].dai_link->ignore_suspend) + list_for_each_entry(rtd, &card->rtd_list, list) { + if (rtd->dai_link->ignore_suspend) continue; - snd_pcm_suspend_all(card->rtd[i].pcm); + snd_pcm_suspend_all(rtd->pcm); } if (card->suspend_pre) card->suspend_pre(card); - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; if (cpu_dai->driver->suspend && !cpu_dai->driver->bus_control) @@ -631,19 +682,19 @@ int snd_soc_suspend(struct device *dev) } /* close any waiting streams */ - for (i = 0; i < card->num_rtd; i++) - flush_delayed_work(&card->rtd[i].delayed_work); + list_for_each_entry(rtd, &card->rtd_list, list) + flush_delayed_work(&rtd->delayed_work); - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(&card->rtd[i], + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_SUSPEND); - snd_soc_dapm_stream_event(&card->rtd[i], + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_SUSPEND); } @@ -690,10 +741,10 @@ int snd_soc_suspend(struct device *dev) } } - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; if (cpu_dai->driver->suspend && cpu_dai->driver->bus_control) @@ -717,8 +768,9 @@ static void soc_resume_deferred(struct work_struct *work) { struct snd_soc_card *card = container_of(work, struct snd_soc_card, deferred_resume_work); + struct snd_soc_pcm_runtime *rtd; struct snd_soc_codec *codec; - int i, j; + int i; /* our power state is still SNDRV_CTL_POWER_D3hot from suspend time, * so userspace apps are blocked from touching us @@ -733,10 +785,10 @@ static void soc_resume_deferred(struct work_struct *work) card->resume_pre(card); /* resume control bus DAIs */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; if (cpu_dai->driver->resume && cpu_dai->driver->bus_control) @@ -751,28 +803,28 @@ static void soc_resume_deferred(struct work_struct *work) } } - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(&card->rtd[i], + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_RESUME); - snd_soc_dapm_stream_event(&card->rtd[i], + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, SND_SOC_DAPM_STREAM_RESUME); } /* unmute any active DACs */ - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(rtd, &card->rtd_list, list) { - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; - for (j = 0; j < card->rtd[i].num_codecs; j++) { - struct snd_soc_dai *dai = card->rtd[i].codec_dais[j]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *dai = rtd->codec_dais[i]; struct snd_soc_dai_driver *drv = dai->driver; if (drv->ops->digital_mute && dai->playback_active) @@ -780,10 +832,10 @@ static void soc_resume_deferred(struct work_struct *work) } } - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - if (card->rtd[i].dai_link->ignore_suspend) + if (rtd->dai_link->ignore_suspend) continue; if (cpu_dai->driver->resume && !cpu_dai->driver->bus_control) @@ -808,15 +860,14 @@ int snd_soc_resume(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); bool bus_control = false; - int i; + struct snd_soc_pcm_runtime *rtd; /* If the card is not initialized yet there is nothing to do */ if (!card->instantiated) return 0; /* activate pins from sleep state */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) { struct snd_soc_dai **codec_dais = rtd->codec_dais; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int j; @@ -837,8 +888,8 @@ int snd_soc_resume(struct device *dev) * have that problem and may take a substantial amount of time to resume * due to I/O costs and anti-pop so handle them out of line. */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + list_for_each_entry(rtd, &card->rtd_list, list) { + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; bus_control |= cpu_dai->driver->bus_control; } if (bus_control) { @@ -913,16 +964,26 @@ static struct snd_soc_dai *snd_soc_find_dai( static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai_link_component *codecs = dai_link->codecs; struct snd_soc_dai_link_component cpu_dai_component; - struct snd_soc_dai **codec_dais = rtd->codec_dais; + struct snd_soc_dai **codec_dais; struct snd_soc_platform *platform; const char *platform_name; int i; dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); + if (snd_soc_get_pcm_runtime(card, dai_link->name)) { + dev_dbg(card->dev, "ASoC: dai link %s already bound\n", + dai_link->name); + return 0; + } + + rtd = soc_new_pcm_runtime(card, dai_link); + if (!rtd) + return -ENOMEM; + cpu_dai_component.name = dai_link->cpu_name; cpu_dai_component.of_node = dai_link->cpu_of_node; cpu_dai_component.dai_name = dai_link->cpu_dai_name; @@ -930,18 +991,19 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) if (!rtd->cpu_dai) { dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); - return -EPROBE_DEFER; + goto _err_defer; } rtd->num_codecs = dai_link->num_codecs; /* Find CODEC from registered CODECs */ + codec_dais = rtd->codec_dais; for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); - return -EPROBE_DEFER; + goto _err_defer; } } @@ -973,9 +1035,12 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return -EPROBE_DEFER; } - card->num_rtd++; - + soc_add_pcm_runtime(card, rtd); return 0; + +_err_defer: + soc_free_pcm_runtime(card, rtd); + return -EPROBE_DEFER; } static void soc_remove_component(struct snd_soc_component *component) @@ -1014,9 +1079,9 @@ static void soc_remove_dai(struct snd_soc_dai *dai, int order) } } -static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) +static void soc_remove_link_dais(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, int order) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; int i; /* unregister the rtd device */ @@ -1032,10 +1097,9 @@ static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) soc_remove_dai(rtd->cpu_dai, order); } -static void soc_remove_link_components(struct snd_soc_card *card, int num, - int order) +static void soc_remove_link_components(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, int order) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_component *component; @@ -1061,21 +1125,20 @@ static void soc_remove_link_components(struct snd_soc_card *card, int num, static void soc_remove_dai_links(struct snd_soc_card *card) { - int dai, order; + int order; + struct snd_soc_pcm_runtime *rtd; for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - for (dai = 0; dai < card->num_rtd; dai++) - soc_remove_link_dais(card, dai, order); + list_for_each_entry(rtd, &card->rtd_list, list) + soc_remove_link_dais(card, rtd, order); } for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - for (dai = 0; dai < card->num_rtd; dai++) - soc_remove_link_components(card, dai, order); + list_for_each_entry(rtd, &card->rtd_list, list) + soc_remove_link_components(card, rtd, order); } - - card->num_rtd = 0; } static void soc_set_name_prefix(struct snd_soc_card *card, @@ -1220,10 +1283,10 @@ static int soc_post_component_init(struct snd_soc_pcm_runtime *rtd, return 0; } -static int soc_probe_link_components(struct snd_soc_card *card, int num, +static int soc_probe_link_components(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, int order) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_component *component; int i, ret; @@ -1319,15 +1382,15 @@ static int soc_link_dai_widgets(struct snd_soc_card *card, return 0; } -static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) +static int soc_probe_link_dais(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, int order) { - struct snd_soc_dai_link *dai_link = &card->dai_link[num]; - struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int i, ret; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", - card->name, num, order); + card->name, rtd->num, order); /* set default power off timeout */ rtd->pmdown_time = pmdown_time; @@ -1372,7 +1435,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (cpu_dai->driver->compress_dai) { /*create compress_device"*/ - ret = soc_new_compress(rtd, num); + ret = soc_new_compress(rtd, rtd->num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1382,7 +1445,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, num); + ret = soc_new_pcm(rtd, rtd->num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -1552,6 +1615,7 @@ EXPORT_SYMBOL_GPL(snd_soc_runtime_set_dai_fmt); static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; + struct snd_soc_pcm_runtime *rtd; int ret, i, order; mutex_lock(&client_mutex); @@ -1624,8 +1688,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe all components used by DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - for (i = 0; i < card->num_links; i++) { - ret = soc_probe_link_components(card, i, order); + list_for_each_entry(rtd, &card->rtd_list, list) { + ret = soc_probe_link_components(card, rtd, order); if (ret < 0) { dev_err(card->dev, "ASoC: failed to instantiate card %d\n", @@ -1638,8 +1702,8 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) /* probe all DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { - for (i = 0; i < card->num_links; i++) { - ret = soc_probe_link_dais(card, i, order); + list_for_each_entry(rtd, &card->rtd_list, list) { + ret = soc_probe_link_dais(card, rtd, order); if (ret < 0) { dev_err(card->dev, "ASoC: failed to instantiate card %d\n", @@ -1733,6 +1797,7 @@ card_probe_error: snd_card_free(card->snd_card); base_error: + soc_remove_pcm_runtimes(card); mutex_unlock(&card->mutex); mutex_unlock(&client_mutex); @@ -1763,13 +1828,12 @@ static int soc_probe(struct platform_device *pdev) static int soc_cleanup_card_resources(struct snd_soc_card *card) { + struct snd_soc_pcm_runtime *rtd; int i; /* make sure any delayed work runs */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); - } /* remove auxiliary devices */ for (i = 0; i < card->num_aux_devs; i++) @@ -1777,6 +1841,7 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove and free each DAI */ soc_remove_dai_links(card); + soc_remove_pcm_runtimes(card); soc_cleanup_card_debugfs(card); @@ -1803,29 +1868,26 @@ static int soc_remove(struct platform_device *pdev) int snd_soc_poweroff(struct device *dev) { struct snd_soc_card *card = dev_get_drvdata(dev); - int i; + struct snd_soc_pcm_runtime *rtd; if (!card->instantiated) return 0; /* Flush out pmdown_time work - we actually do want to run it * now, we're shutting down so no imminent restart. */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) flush_delayed_work(&rtd->delayed_work); - } snd_soc_dapm_shutdown(card); /* deactivate pins to sleep state */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int j; + int i; pinctrl_pm_select_sleep_state(cpu_dai->dev); - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; pinctrl_pm_select_sleep_state(codec_dai->dev); } } @@ -2337,6 +2399,7 @@ static int snd_soc_init_multicodec(struct snd_soc_card *card, int snd_soc_register_card(struct snd_soc_card *card) { int i, j, ret; + struct snd_soc_pcm_runtime *rtd; if (!card->name || !card->dev) return -EINVAL; @@ -2408,25 +2471,15 @@ int snd_soc_register_card(struct snd_soc_card *card) snd_soc_initialize_card_lists(card); - card->rtd = devm_kzalloc(card->dev, + INIT_LIST_HEAD(&card->rtd_list); + card->num_rtd = 0; + + card->rtd_aux = devm_kzalloc(card->dev, sizeof(struct snd_soc_pcm_runtime) * - (card->num_links + card->num_aux_devs), + card->num_aux_devs, GFP_KERNEL); - if (card->rtd == NULL) + if (card->rtd_aux == NULL) return -ENOMEM; - card->num_rtd = 0; - card->rtd_aux = &card->rtd[card->num_links]; - - for (i = 0; i < card->num_links; i++) { - card->rtd[i].card = card; - card->rtd[i].dai_link = &card->dai_link[i]; - card->rtd[i].codec_dais = devm_kzalloc(card->dev, - sizeof(struct snd_soc_dai *) * - (card->rtd[i].dai_link->num_codecs), - GFP_KERNEL); - if (card->rtd[i].codec_dais == NULL) - return -ENOMEM; - } for (i = 0; i < card->num_aux_devs; i++) card->rtd_aux[i].card = card; @@ -2442,8 +2495,7 @@ int snd_soc_register_card(struct snd_soc_card *card) return ret; /* deactivate pins to sleep state */ - for (i = 0; i < card->num_rtd; i++) { - struct snd_soc_pcm_runtime *rtd = &card->rtd[i]; + list_for_each_entry(rtd, &card->rtd_list, list) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int j; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7482c5d..6d1f066 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3861,13 +3861,10 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = card->rtd; - int i; + struct snd_soc_pcm_runtime *rtd; /* for each BE DAI link... */ - for (i = 0; i < card->num_rtd; i++) { - rtd = &card->rtd[i]; - + list_for_each_entry(rtd, &card->rtd_list, list) { /* * dynamic FE links have no fixed DAI mapping. * CODEC<->CODEC links have no direct connection. diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 70e4b9d..0246019 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1166,11 +1166,10 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { struct snd_soc_pcm_runtime *be; - int i, j; + int i; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - for (i = 0; i < card->num_links; i++) { - be = &card->rtd[i]; + list_for_each_entry(be, &card->rtd_list, list) { if (!be->dai_link->no_pcm) continue; @@ -1178,16 +1177,15 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (be->cpu_dai->playback_widget == widget) return be; - for (j = 0; j < be->num_codecs; j++) { - struct snd_soc_dai *dai = be->codec_dais[j]; + for (i = 0; i < be->num_codecs; i++) { + struct snd_soc_dai *dai = be->codec_dais[i]; if (dai->playback_widget == widget) return be; } } } else { - for (i = 0; i < card->num_links; i++) { - be = &card->rtd[i]; + list_for_each_entry(be, &card->rtd_list, list) { if (!be->dai_link->no_pcm) continue; @@ -1195,8 +1193,8 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (be->cpu_dai->capture_widget == widget) return be; - for (j = 0; j < be->num_codecs; j++) { - struct snd_soc_dai *dai = be->codec_dais[j]; + for (i = 0; i < be->num_codecs; i++) { + struct snd_soc_dai *dai = be->codec_dais[i]; if (dai->capture_widget == widget) return be; } @@ -2296,12 +2294,12 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) */ int soc_dpcm_runtime_update(struct snd_soc_card *card) { - int i, old, new, paths; + struct snd_soc_pcm_runtime *fe; + int old, new, paths; mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - for (i = 0; i < card->num_rtd; i++) { + list_for_each_entry(fe, &card->rtd_list, list) { struct snd_soc_dapm_widget_list *list; - struct snd_soc_pcm_runtime *fe = &card->rtd[i]; /* make sure link is FE */ if (!fe->dai_link->dynamic) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 2160400..e485278 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -199,7 +199,8 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) static int tegra_wm8903_remove(struct snd_soc_card *card) { - struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]); + struct snd_soc_pcm_runtime *rtd = + snd_soc_get_pcm_runtime(card, card->dai_link[0].name); struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_codec *codec = codec_dai->codec; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); -- 1.9.1

1 0

Re: [alsa-devel] [SOLVED] [pulseaudio-discuss] No sound Lubuntu 14.02
by Raymond Yau 08 Aug '15

08 Aug '15

>>>> >> > >>>> >> > sl@max:~$ lspci -nnk | grep -iA2 audio >>>> >> > 00:1b.0 Audio device [0403]: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller [8086:2668] (rev 04) >>>> >> > Subsystem: QUANTA Computer Inc Device [152d:0748] >>>> >> > Kernel driver in use: snd_hda_intel >>>> >> > >>>> >> > sl@max:~$ aplay -l >>>> >> > **** Liste der Hardware-Geräte (PLAYBACK) **** >>>> >> > Karte 0: Intel [HDA Intel], Gerät 6: Si3054 Modem [Si3054 Modem] >>>> >> > Sub-Geräte: 1/1 >>>> >> > Sub-Gerät #0: subdevice #0 >>>> >> > >>>> >> >>>> >> https://lists.fedoraproject.org/pipermail/test/2014-November/123662.html >>>> >> >>>> >> If bios did not setup pin default of alc880 codec, you need to find out the pin of hp and mic jacks by hdajacksensetest , speaker and internal mic need to find out by trial and err with hdajackretask >>>> >> >>>> > >>>> > This is the output of hda-jack-sense-test. What do you recommend? >>>> > >>>> > root@max:~# hda-jack-sense-test >>>> > Pin 0x14 (Unknown Line Out): present = No >>>> > Pin 0x15 (Unknown Line Out): present = No >>>> > Pin 0x16 (Unknown Line Out): present = No >>>> > Pin 0x17 (Unknown Line Out): present = No >>>> > Pin 0x18 (Unknown Line Out): present = No >>>> > Pin 0x19 (Unknown Line Out): present = No >>>> > Pin 0x1a (Unknown Line Out): present = No >>>> > Pin 0x1b (Unknown Line Out): present = No >>>> > Pin 0x1c (Unknown Line Out): present = No >>>> > Pin 0x1d (Unknown Line Out): present = No >>>> > Pin 0x1e (Unknown Line Out): present = No >>>> > Pin 0x1f (Unknown Line Out): present = No >>>> > >>>> >>>> Post the output of alsa-info.sh >>>> >>>> You have to use >>>> >>>> hdajacksensetest -a >>> >>> It gives the same output. >>>> >>>> Did you plug and unplug headphone or mic jack during test ? >>> >>> No. >> >> you have to post output of >> >> hdajacksensetest -a >> >> when you plug headphone , unplug headphone , plug mic and unplug mic >> >> >> >>>> >>>> Since pin default of all pin complex are zero which is Line Out >>> >>> It is a laptop. I want to use its internal speakers, if that info helps. >> >> All pins have to be fixed by hdajackretask >> >> !!Sysfs Files >> !!----------- >> >> /sys/class/sound/hwC0D0/init_pin_configs: >> 0x14 0x00000000 >> 0x15 0x00000000 >> 0x16 0x00000000 >> 0x17 0x00000000 >> 0x18 0x00000000 >> 0x19 0x00000000 >> 0x1a 0x00000000 >> 0x1b 0x00000000 >> 0x1c 0x00000000 >> 0x1d 0x00000000 >> 0x1e 0x01000000 >> 0x1f 0x00000000 >> >> >>>> >>>> Any redundant pins must set to [N/A] >>>> >>>> alc880 only have models for desktop >>>> >>>> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… >>>> >>>> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/… >>> >>> >> >> >> In the past, there was a "test" model which setup all pins >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> you can try >> >> options snd-hda-intel model=6stack-dig >> >> to find out whether the speaker is node 0x14 and Headphone and 0x1b >> you have to manually find front mic, rear mic and line in is your laptop internal mic and mic jacks >> >> >> Do windows driver contain info about the pins since it also need to setup the pin complex too ? >> >> >> >> sl@max:~$ cat /tmp/alsa-info.txt.5lh8RkGutE >> upload=true&script=true&cardinfo= >> !!################################ >> !!ALSA Information Script v 0.4.64 >> !!################################ >> >> !!Script ran on: Fri Aug 7 11:57:07 UTC 2015 >> >> >> !!Linux Distribution >> !!------------------ >> >> Ubuntu 14.04.3 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 14.04.3 LTS" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 14.04.3 LTS" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL=" http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" >> >> >> !!DMI Information >> !!--------------- >> >> Manufacturer: MAXDATA >> Product Name: PRO8100IS58 >> Product Version: 5123660003 >> Firmware Version: M3B61 >> >> >> !!Kernel Information >> !!------------------ >> >> Kernel release: 3.16.0-45-generic >> Operating System: GNU/Linux >> Architecture: i686 >> Processor: i686 >> SMP Enabled: Yes >> >> >> !!ALSA Version >> !!------------ >> >> Driver version: k3.16.0-45-generic >> Library version: 1.0.27.2 >> Utilities version: 1.0.27.2 >> >> >> !!Loaded ALSA modules >> !!------------------- >> >> snd_hda_intel >> >> >> !!Sound Servers on this system >> !!---------------------------- >> >> Pulseaudio: >> Installed - Yes (/usr/bin/pulseaudio) >> Running - Yes >> >> >> !!Soundcards recognised by ALSA >> !!----------------------------- >> >> 0 [Intel ]: HDA-Intel - HDA Intel >> HDA Intel at 0xc0000000 irq 46 >> >> >> !!PCI Soundcards installed in the system >> !!-------------------------------------- >> >> 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 04) >> >> >> !!Advanced information - PCI Vendor/Device/Subsystem ID's >> !!------------------------------------------------------- >> >> 00:1b.0 0403: 8086:2668 (rev 04) >> Subsystem: 152d:0748 >> -- >> Prefetchable memory behind bridge: 0000000040000000-00000000401fffff >> Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- <SERR- <PERR- >> >> >> >> >> !!HDA-Intel Codec information >> !!--------------------------- >> --startcollapse-- >> >> Codec: Realtek ALC880 >> Address: 0 >> AFG Function Id: 0x1 (unsol 0) >> Vendor Id: 0x10ec0880 >> Subsystem Id: 0x08800000 >> Revision Id: 0x100800 >> No Modem Function Group found >> Default PCM: >> rates [0x560]: 44100 48000 96000 192000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Default Amp-In caps: N/A >> Default Amp-Out caps: N/A >> State of AFG node 0x01: >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 >> IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 >> IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 >> Node 0x02 [Audio Output] wcaps 0x411: Stereo >> Converter: stream=0, channel=0 >> PCM: >> rates [0x560]: 44100 48000 96000 192000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Node 0x03 [Audio Output] wcaps 0x411: Stereo >> Converter: stream=0, channel=0 >> PCM: >> rates [0x560]: 44100 48000 96000 192000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Node 0x04 [Audio Output] wcaps 0x411: Stereo >> Converter: stream=0, channel=0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Node 0x05 [Audio Output] wcaps 0x411: Stereo >> Converter: stream=0, channel=0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0xe]: 16 20 24 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital >> Converter: stream=0, channel=0 >> Digital: >> Digital category: 0x0 >> IEC Coding Type: 0x0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x1e]: 16 20 24 32 >> formats [0x1]: PCM >> Node 0x07 [Audio Input] wcaps 0x10051b: Stereo Amp-In >> Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 >> Amp-In vals: [0x00 0x00] >> Converter: stream=0, channel=0 >> SDI-Select: 0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x6]: 16 20 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Connection: 7 >> 0x18* 0x19 0x1a 0x1b 0x1c 0x14 0x15 >> Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In >> Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 >> Amp-In vals: [0x00 0x00] >> Converter: stream=0, channel=0 >> SDI-Select: 0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x6]: 16 20 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Connection: 7 >> 0x18 0x19 0x1a* 0x1b 0x1c 0x14 0x15 >> Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In >> Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 >> Amp-In vals: [0x00 0x00] >> Converter: stream=0, channel=0 >> SDI-Select: 0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x6]: 16 20 >> formats [0x1]: PCM >> Power states: D0 D1 D2 D3 >> Power: setting=D0, actual=D0 >> Connection: 10 >> 0x18 0x19 0x1a 0x1b 0x1c* 0x0b 0x14 0x15 0x16 0x17 >> Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital >> Converter: stream=0, channel=0 >> SDI-Select: 0 >> Digital: >> Digital category: 0x0 >> IEC Coding Type: 0x0 >> PCM: >> rates [0x160]: 44100 48000 96000 >> bits [0x1e]: 16 20 24 32 >> formats [0x1]: PCM >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x1f >> Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In >> Control: name="Beep Playback Volume", index=0, device=0 >> ControlAmp: chs=3, dir=In, idx=5, ofs=0 >> Control: name="Beep Playback Switch", index=0, device=0 >> ControlAmp: chs=3, dir=In, idx=5, ofs=0 >> Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 >> Amp-In vals: [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0x80 0x80] [0xa3 0xa3] [0xa3 0xa3] >> Connection: 8 >> 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 >> Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x80 0x80] >> Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> Amp-Out vals: [0x40 0x40] >> Connection: 2 >> 0x02 0x0b >> Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x80 0x80] >> Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> Amp-Out vals: [0x40 0x40] >> Connection: 2 >> 0x03 0x0b >> Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x80 0x80] >> Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> Amp-Out vals: [0x40 0x40] >> Connection: 2 >> 0x04 0x0b >> Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out >> Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-In vals: [0x00 0x00] [0x80 0x80] >> Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 >> Amp-Out vals: [0x40 0x40] >> Connection: 2 >> 0x05 0x0b >> Node 0x10 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 4 >> 0x0c* 0x0d 0x0e 0x0f >> Node 0x11 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 4 >> 0x0c* 0x0d 0x0e 0x0f >> Node 0x12 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 4 >> 0x0c* 0x0d 0x0e 0x0f >> Node 0x13 [Audio Selector] wcaps 0x300101: Stereo >> Connection: 4 >> 0x0c* 0x0d 0x0e 0x0f >> Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0c >> Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0d >> Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0e >> Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x0f >> Node 0x18 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense >> Vref caps: HIZ 50 80 >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x21: IN VREF_50 >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x10 >> Node 0x19 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense >> Vref caps: HIZ 50 80 >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN VREF_HIZ >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x11 >> Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense >> Vref caps: HIZ 50 80 >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN VREF_HIZ >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x12 >> Node 0x1b [Pin Complex] wcaps 0x40018d: Stereo Amp-Out >> Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 >> Amp-Out vals: [0x80 0x80] >> Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense >> Vref caps: HIZ 50 80 >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x20: IN VREF_HIZ >> Unsolicited: tag=00, enabled=0 >> Connection: 1 >> 0x13 >> Node 0x1c [Pin Complex] wcaps 0x400001: Stereo >> Pincap 0x00000020: IN >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Node 0x1d [Pin Complex] wcaps 0x400000: Mono >> Pincap 0x00000020: IN >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital >> Control: name="Line Out Phantom Jack", index=0, device=0 >> Pincap 0x00000010: OUT >> Pin Default 0x01000000: [Jack] Line Out at Ext Rear >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Connection: 1 >> 0x06 >> Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital >> Pincap 0x00000020: IN >> Pin Default 0x00000000: [Jack] Line Out at Ext N/A >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono >> Processing caps: benign=0, ncoeff=10 >> Node 0x21 [Volume Knob Widget] wcaps 0x600080: Mono >> Volume-Knob: delta=0, steps=64, direct=0, val=25 >> Unsolicited: tag=00, enabled=0 >> Connection: 0 >> Codec: Motorola Si3054 >> Address: 1 >> MFG Function Id: 0x2 (unsol 1) >> Vendor Id: 0x10573055 >> Subsystem Id: 0x10573055 >> Revision Id: 0x100700 >> Modem Function Group: 0x1 >> --endcollapse-- >> >> >> !!ALSA Device nodes >> !!----------------- >> >> crw-rw----+ 1 root audio 116, 2 Aug 7 12:03 /dev/snd/controlC0 >> crw-rw----+ 1 root audio 116, 5 Aug 7 12:03 /dev/snd/hwC0D0 >> crw-rw----+ 1 root audio 116, 6 Aug 7 12:03 /dev/snd/hwC0D1 >> crw-rw----+ 1 root audio 116, 4 Aug 7 12:03 /dev/snd/pcmC0D6c >> crw-rw----+ 1 root audio 116, 3 Aug 7 12:03 /dev/snd/pcmC0D6p >> crw-rw----+ 1 root audio 116, 1 Aug 7 12:03 /dev/snd/seq >> crw-rw----+ 1 root audio 116, 33 Aug 7 12:03 /dev/snd/timer >> >> /dev/snd/by-path: >> total 0 >> drwxr-xr-x 2 root root 60 Aug 7 12:03 . >> drwxr-xr-x 3 root root 200 Aug 7 12:03 .. >> lrwxrwxrwx 1 root root 12 Aug 7 12:03 pci-0000:00:1b.0 -> ../controlC0 >> >> >> !!Aplay/Arecord output >> !!-------------------- >> >> APLAY >> >> **** List of PLAYBACK Hardware Devices **** >> card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> ARECORD >> >> **** List of CAPTURE Hardware Devices **** >> card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] >> Subdevices: 1/1 >> Subdevice #0: subdevice #0 >> >> !!Amixer output >> !!------------- >> >> !!-------Mixer controls for card 0 [Intel] >> >> Card hw:0 'Intel'/'HDA Intel at 0xc0000000 irq 46' >> Mixer name : 'Realtek ALC880' >> Components : 'HDA:10ec0880,08800000,00100800 HDA:10573055,10573055,00100700' >> Controls : 8 >> Simple ctrls : 4 >> Simple mixer control 'PCM',0 >> Capabilities: pvolume >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 255 >> Mono: >> Front Left: Playback 255 [100%] [0.00dB] >> Front Right: Playback 255 [100%] [0.00dB] >> Simple mixer control 'Beep',0 >> Capabilities: pvolume pswitch >> Playback channels: Front Left - Front Right >> Limits: Playback 0 - 65 >> Mono: >> Front Left: Playback 0 [0%] [-35.00dB] [off] >> Front Right: Playback 0 [0%] [-35.00dB] [off] >> Simple mixer control 'Caller ID',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [off] >> Simple mixer control 'Off-hook',0 >> Capabilities: pswitch pswitch-joined >> Playback channels: Mono >> Mono: Playback [off] >> >> >> !!Alsactl output >> !!-------------- >> >> --startcollapse-- >> state.Intel { >> control.1 { >> iface CARD >> name 'Line Out Phantom Jack' >> value true >> comment { >> access read >> type BOOLEAN >> count 1 >> } >> } >> control.2 { >> iface MIXER >> name 'Beep Playback Volume' >> value.0 0 >> value.1 0 >> comment { >> access 'read write' >> type INTEGER >> count 2 >> range '0 - 65' >> dbmin -3500 >> dbmax 3000 >> dbvalue.0 -3500 >> dbvalue.1 -3500 >> } >> } >> control.3 { >> iface MIXER >> name 'Beep Playback Switch' >> value.0 false >> value.1 false >> comment { >> access 'read write' >> type BOOLEAN >> count 2 >> } >> } >> control.4 { >> iface MIXER >> name 'Off-hook Switch' >> value false >> comment { >> access 'read write' >> type BOOLEAN >> count 1 >> } >> } >> control.5 { >> iface MIXER >> name 'Caller ID Switch' >> value false >> comment { >> access 'read write' >> type BOOLEAN >> count 1 >> } >> } >> control.6 { >> iface PCM >> device 6 >> name 'Playback Channel Map' >> value 0 >> comment { >> access read >> type INTEGER >> count 1 >> range '0 - 36' >> } >> } >> control.7 { >> iface PCM >> device 6 >> name 'Capture Channel Map' >> value 0 >> comment { >> access read >> type INTEGER >> count 1 >> range '0 - 36' >> } >> } >> control.8 { >> iface MIXER >> name 'PCM Playback Volume' >> value.0 255 >> value.1 255 >> comment { >> access 'read write user' >> type INTEGER >> count 2 >> range '0 - 255' >> tlv '0000000100000008ffffec1400000014' >> dbmin -5100 >> dbmax 0 >> dbvalue.0 0 >> dbvalue.1 0 >> } >> } >> } >> --endcollapse-- >> >> >> >> >> !!Sysfs Files >> !!----------- >> >> /sys/class/sound/hwC0D0/init_pin_configs: >> 0x14 0x00000000 >> 0x15 0x00000000 >> 0x16 0x00000000 >> 0x17 0x00000000 >> 0x18 0x00000000 >> 0x19 0x00000000 >> 0x1a 0x00000000 >> 0x1b 0x00000000 >> 0x1c 0x00000000 >> 0x1d 0x00000000 >> 0x1e 0x01000000 >> 0x1f 0x00000000 >> >> /sys/class/sound/hwC0D0/driver_pin_configs: >> >> /sys/class/sound/hwC0D0/user_pin_configs: >> >> /sys/class/sound/hwC0D0/init_verbs: >> >> /sys/class/sound/hwC0D0/hints: >> >> /sys/class/sound/hwC0D1/init_pin_configs: >> >> /sys/class/sound/hwC0D1/driver_pin_configs: >> >> /sys/class/sound/hwC0D1/user_pin_configs: >> >> /sys/class/sound/hwC0D1/init_verbs: >> >> /sys/class/sound/hwC0D1/hints: >> >> >> !!ALSA/HDA dmesg >> !!-------------- >> >> [ 29.335966] ipw2200: Detected geography ZZE (13 802.11bg channels, 19 802.11a channels) >> [ 29.378618] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X >> [ 29.673379] sound hdaudioC0D0: autoconfig: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line >> [ 29.673387] sound hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) >> [ 29.673390] sound hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) >> [ 29.673393] sound hdaudioC0D0: mono: mono_out=0x1e >> [ 29.673396] sound hdaudioC0D0: inputs: >> [ 30.010420] Bluetooth: RFCOMM TTY layer initialized >> >> >> > > > After figuring out what lines are used for which device, I was able > to set config. Now it is working. > Can you post out the correct pin config for your laptop ? Do auto mute and auto mic selection work as expected ? Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital >> Control: name="Line Out Phantom Jack", index=0, device=0 >> Pincap 0x00000010: OUT >> Pin Default 0x01000000: [Jack] Line Out at Ext Rear >> Conn = Unknown, Color = Unknown >> DefAssociation = 0x0, Sequence = 0x0 >> Pin-ctls: 0x00: >> Connection: 1 >> 0x06 There is a bug in hda_auto_parser which treat invalid digital out pin as mono out(line out) and create Line Out Phantom Jack

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[alsa-devel] [PATCH] topology: treat all DAPM controls types the same when copying
by Liam Girdwood 07 Aug '15

07 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Copy all DAPM controls types using the same method. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/dapm.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 7e26ea0..a0a8b86 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -107,8 +107,6 @@ static int tplg_parse_dapm_enums(snd_config_t *cfg, struct tplg_elem *elem) static int copy_dapm_control(struct tplg_elem *elem, struct tplg_elem *ref) { struct snd_soc_tplg_dapm_widget *widget = elem->widget; - struct snd_soc_tplg_mixer_control *mixer_ctrl = ref->mixer_ctrl; - struct snd_soc_tplg_enum_control *enum_ctrl = ref->enum_ctrl; tplg_dbg("Control '%s' used by '%s'\n", ref->id, elem->id); tplg_dbg("\tparent size: %d + %d -> %d, priv size -> %d\n", @@ -121,13 +119,10 @@ static int copy_dapm_control(struct tplg_elem *elem, struct tplg_elem *ref) elem->widget = widget; - /* copy new widget at the end */ - if (ref->type == OBJECT_TYPE_MIXER) - memcpy((void*)widget + elem->size, mixer_ctrl, ref->size); - else if (ref->type == OBJECT_TYPE_ENUM) - memcpy((void*)widget + elem->size, enum_ctrl, ref->size); - + /* append the control to the end of the widget */ + memcpy((void*)widget + elem->size, ref->obj, ref->size); elem->size += ref->size; + widget->num_kcontrols++; ref->compound_elem = 1; return 0; -- 2.1.4

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[alsa-devel] [PATCH v2 0/2] topology: UAPI updates for TLV and ops.
by Liam Girdwood 07 Aug '15

07 Aug '15

This series updates the topology UAPI to allow easier addition of new TLV object types and generic operations for alsa-lib. Mengdong Lin (2): topology: update ABI to improve support for different TLV object types. topology: Add ops support to byte control objects. include/sound/asoc.h | 28 +++++++++++++++++++--------- src/topology/ctl.c | 20 ++++++++------------ 2 files changed, 27 insertions(+), 21 deletions(-) -- 2.1.4

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Re: [alsa-devel] [pulseaudio-discuss] No sound Lubuntu 14.02
by Raymond Yau 07 Aug '15

07 Aug '15

> > >> > >> > > >> > I tried a lot already. No success to get the sound working on my old > maxdata laptop. > >> > This is what I got: > >> > > >> > sl@max:~$ lspci -nnk | grep -iA2 audio > >> > 00:1b.0 Audio device [0403]: Intel Corporation 82801FB/FBM/FR/FW/FRW > (ICH6 Family) High Definition Audio Controller [8086:2668] (rev 04) > >> > Subsystem: QUANTA Computer Inc Device [152d:0748] > >> > Kernel driver in use: snd_hda_intel > >> > > >> > sl@max:~$ aplay -l > >> > **** Liste der Hardware-Geräte (PLAYBACK) **** > >> > Karte 0: Intel [HDA Intel], Gerät 6: Si3054 Modem [Si3054 Modem] > >> > Sub-Geräte: 1/1 > >> > Sub-Gerät #0: subdevice #0 > >> > > >> > sl@max:~$ aplay /usr/share/sounds/alsa/Front_Center.wav > >> > Wiedergabe: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed > 16 bit Little Endian, Rate: 48000 Hz, mono > >> > > >> > Please let me know if/what further information I should provide to > give you a better understanding in what is wrong. > >> > > >> > >> > https://lists.fedoraproject.org/pipermail/test/2014-November/123662.html > >> > >> If bios did not setup pin default of alc880 codec, you need to find out > the pin of hp and mic jacks by hdajacksensetest , speaker and internal mic > need to find out by trial and err with hdajackretask > >> > >> _______________________________________________ > >> pulseaudio-discuss mailing list > >> pulseaudio-discuss(a)lists.freedesktop.org > >> http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss > > > > This is the output of hda-jack-sense-test. What do you recommend? > > > > root@max:~# hda-jack-sense-test > > Pin 0x14 (Unknown Line Out): present = No > > Pin 0x15 (Unknown Line Out): present = No > > Pin 0x16 (Unknown Line Out): present = No > > Pin 0x17 (Unknown Line Out): present = No > > Pin 0x18 (Unknown Line Out): present = No > > Pin 0x19 (Unknown Line Out): present = No > > Pin 0x1a (Unknown Line Out): present = No > > Pin 0x1b (Unknown Line Out): present = No > > Pin 0x1c (Unknown Line Out): present = No > > Pin 0x1d (Unknown Line Out): present = No > > Pin 0x1e (Unknown Line Out): present = No > > Pin 0x1f (Unknown Line Out): present = No > > > > Post the output of alsa-info.sh > > You have to use > > hdajacksensetest -a > > It gives the same output. > > Did you plug and unplug headphone or mic jack during test ? > > No. > you have to post output of hdajacksensetest -a when you plug headphone , unplug headphone , plug mic and unplug mic > > Since pin default of all pin complex are zero which is Line Out > > It is a laptop. I want to use its internal speakers, if that info helps. > All pins have to be fixed by hdajackretask !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x00000000 0x15 0x00000000 0x16 0x00000000 0x17 0x00000000 0x18 0x00000000 0x19 0x00000000 0x1a 0x00000000 0x1b 0x00000000 0x1c 0x00000000 0x1d 0x00000000 0x1e 0x01000000 0x1f 0x00000000 > Any redundant pins must set to [N/A] > > alc880 only have models for desktop > > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/Documenta… > > > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/… > > > In the past, there was a "test" model which setup all pins https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… you can try options snd-hda-intel model=6stack-dig to find out whether the speaker is node 0x14 and Headphone and 0x1b you have to manually find front mic, rear mic and line in is your laptop internal mic and mic jacks Do windows driver contain info about the pins since it also need to setup the pin complex too ? sl@max:~$ cat /tmp/alsa-info.txt.5lh8RkGutE upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.64 !!################################ !!Script ran on: Fri Aug 7 11:57:07 UTC 2015 !!Linux Distribution !!------------------ Ubuntu 14.04.3 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 14.04.3 LTS" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 14.04.3 LTS" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL=" http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" !!DMI Information !!--------------- Manufacturer: MAXDATA Product Name: PRO8100IS58 Product Version: 5123660003 Firmware Version: M3B61 !!Kernel Information !!------------------ Kernel release: 3.16.0-45-generic Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: k3.16.0-45-generic Library version: 1.0.27.2 Utilities version: 1.0.27.2 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xc0000000 irq 46 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 04) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!------------------------------------------------------- 00:1b.0 0403: 8086:2668 (rev 04) Subsystem: 152d:0748 -- Prefetchable memory behind bridge: 0000000040000000-00000000401fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- <SERR- <PERR- !!Modprobe options (Sound related) !!-------------------------------- snd_atiixp_modem: index=-2 snd_intel8x0m: index=-2 snd_via82xx_modem: index=-2 snd_usb_audio: index=-2 snd_usb_caiaq: index=-2 snd_usb_ua101: index=-2 snd_usb_us122l: index=-2 snd_usb_usx2y: index=-2 snd_cmipci: mpu_port=0x330 fm_port=0x388 snd_pcsp: index=-2 snd_usb_audio: index=-2 !!Loaded sound module options !!--------------------------- !!Module: snd_hda_intel align_buffer_size : -1 bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N snoop : Y !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC880 Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x10ec0880 Subsystem Id: 0x08800000 Revision Id: 0x100800 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x04 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x05 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 7 0x18* 0x19 0x1a 0x1b 0x1c 0x14 0x15 Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 7 0x18 0x19 0x1a* 0x1b 0x1c 0x14 0x15 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x23, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 10 0x18 0x19 0x1a 0x1b 0x1c* 0x0b 0x14 0x15 0x16 0x17 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x23, nsteps=0x41, stepsize=0x03, mute=1 Amp-In vals: [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0xa3 0xa3] [0x80 0x80] [0xa3 0xa3] [0xa3 0xa3] Connection: 8 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x05 0x0b Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0c* 0x0d 0x0e 0x0f Node 0x11 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0c* 0x0d 0x0e 0x0f Node 0x12 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0c* 0x0d 0x0e 0x0f Node 0x13 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0c* 0x0d 0x0e 0x0f Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x17 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0f Node 0x18 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x21: IN VREF_50 Unsolicited: tag=00, enabled=0 Connection: 1 0x10 Node 0x19 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x11 Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x12 Node 0x1b [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000133f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 80 Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x13 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Control: name="Line Out Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x01000000: [Jack] Line Out at Ext Rear Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x00000000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=10 Node 0x21 [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=64, direct=0, val=25 Unsolicited: tag=00, enabled=0 Connection: 0 Codec: Motorola Si3054 Address: 1 MFG Function Id: 0x2 (unsol 1) Vendor Id: 0x10573055 Subsystem Id: 0x10573055 Revision Id: 0x100700 Modem Function Group: 0x1 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 2 Aug 7 12:03 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 5 Aug 7 12:03 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 6 Aug 7 12:03 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 4 Aug 7 12:03 /dev/snd/pcmC0D6c crw-rw----+ 1 root audio 116, 3 Aug 7 12:03 /dev/snd/pcmC0D6p crw-rw----+ 1 root audio 116, 1 Aug 7 12:03 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Aug 7 12:03 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Aug 7 12:03 . drwxr-xr-x 3 root root 200 Aug 7 12:03 .. lrwxrwxrwx 1 root root 12 Aug 7 12:03 pci-0000:00:1b.0 -> ../controlC0 !!Aplay/Arecord output !!-------------------- APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xc0000000 irq 46' Mixer name : 'Realtek ALC880' Components : 'HDA:10ec0880,08800000,00100800 HDA:10573055,10573055,00100700' Controls : 8 Simple ctrls : 4 Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 65 Mono: Front Left: Playback 0 [0%] [-35.00dB] [off] Front Right: Playback 0 [0%] [-35.00dB] [off] Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!Alsactl output !!-------------- --startcollapse-- state.Intel { control.1 { iface CARD name 'Line Out Phantom Jack' value true comment { access read type BOOLEAN count 1 } } control.2 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 65' dbmin -3500 dbmax 3000 dbvalue.0 -3500 dbvalue.1 -3500 } } control.3 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.4 { iface MIXER name 'Off-hook Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.5 { iface MIXER name 'Caller ID Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.6 { iface PCM device 6 name 'Playback Channel Map' value 0 comment { access read type INTEGER count 1 range '0 - 36' } } control.7 { iface PCM device 6 name 'Capture Channel Map' value 0 comment { access read type INTEGER count 1 range '0 - 36' } } control.8 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x00000000 0x15 0x00000000 0x16 0x00000000 0x17 0x00000000 0x18 0x00000000 0x19 0x00000000 0x1a 0x00000000 0x1b 0x00000000 0x1c 0x00000000 0x1d 0x00000000 0x1e 0x01000000 0x1f 0x00000000 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC0D0/hints: /sys/class/sound/hwC0D1/init_pin_configs: /sys/class/sound/hwC0D1/driver_pin_configs: /sys/class/sound/hwC0D1/user_pin_configs: /sys/class/sound/hwC0D1/init_verbs: /sys/class/sound/hwC0D1/hints: !!ALSA/HDA dmesg !!-------------- [ 29.335966] ipw2200: Detected geography ZZE (13 802.11bg channels, 19 802.11a channels) [ 29.378618] snd_hda_intel 0000:00:1b.0: irq 46 for MSI/MSI-X [ 29.673379] sound hdaudioC0D0: autoconfig: line_outs=0 (0x0/0x0/0x0/0x0/0x0) type:line [ 29.673387] sound hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 29.673390] sound hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 29.673393] sound hdaudioC0D0: mono: mono_out=0x1e [ 29.673396] sound hdaudioC0D0: inputs: [ 30.010420] Bluetooth: RFCOMM TTY layer initialized

1 0

[alsa-devel] bug report - headphones stopped working
by Tom Tromey 07 Aug '15

07 Aug '15

A recent kernel upgrade caused my headphones to stop working. I asked on #alsa about this and after a lot of help from "debianuser" (thank you very much), he suggested I report this to this list. I'm using x86-64 Fedora 22. The kernel that works is kernel-4.0.5-300.fc22.x86_64. The kernel that fails is kernel-4.1.3-200.fc22.x86_64. (These are the RPM package names in case that isn't clear.) The output from alsa-info.sh for the working case: http://www.alsa-project.org/db/?f=1da69e358935e2d4a35513fc308a74a07925a4fd ... and for the failing case: http://www.alsa-project.org/db/?f=6a1439affc507f1b83f99b5f42b7cbfe0664f9e8 I'm happy to provide other information as needed, though you'll have to tell me exactly what to do, and turnaround may be on the slow side. thanks, Tom

1 0

[alsa-devel] [PATCH v2 0/6] ASoC: Intel: Skylake: Add DSP management routines
by Vinod Koul 07 Aug '15

07 Aug '15

This is second rev of series to add DSP management routines for SKL driver The first two patches are fixes to use ACPI header and size fix on NHLT query Then we add DSP module configuration calculation helpers followed by SRC and updown calculation. These are used by next patch which adds module creation, and binding routines. Lastly modules are used by pipes so we add pipelines creation, destroy and scheduling helpers Hardik T Shah (1): ASoC: Intel: Skylake: Add helpers for SRC and converter modules Jeeja KP (5): ASoC: Intel: Skylake: Fix the NHLT rate size ASoC: Intel: Skylake: Use acpi header for NHLT header ASoC: Intel: Skylake: Add helpers for DSP module configuration ASoC: Intel: Skylake: Add DSP module init and binding routines ASoC: Intel: Skylake: Add pipe management helpers sound/soc/intel/skylake/skl-messages.c | 751 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 3 +- sound/soc/intel/skylake/skl-nhlt.h | 14 +- sound/soc/intel/skylake/skl-topology.h | 286 ++++++++++ sound/soc/intel/skylake/skl-tplg-interface.h | 88 ++++ 5 files changed, 1128 insertions(+), 14 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.h create mode 100644 sound/soc/intel/skylake/skl-tplg-interface.h -- 1.9.1

2 12

[alsa-devel] [PATCH 0/4] Add SPDIF support for rockchip
by Sjoerd Simons 07 Aug '15

07 Aug '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board. Sjoerd Simons (4): ASoC: dt-bindings: add rockchip tranceiver bindings ASoc: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock .../devicetree/bindings/sound/rockchip-spdif.txt | 41 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 ++ arch/arm/boot/dts/rk3188.dtsi | 22 ++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 3 + sound/soc/rockchip/rockchip_spdif.c | 375 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 7 files changed, 532 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.4.6

5 10

[alsa-devel] [PATCH 29/31] sound/oxfw: use kmemdup rather than duplicating its implementation
by Andrzej Hajda 07 Aug '15

07 Aug '15

The patch was generated using fixed coccinelle semantic patch scripts/coccinelle/api/memdup.cocci [1]. [1]: http://permalink.gmane.org/gmane.linux.kernel/2014320 Signed-off-by: Andrzej Hajda <a.hajda(a)samsung.com> --- sound/firewire/oxfw/oxfw-stream.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 873d40f..77ad5b9 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -512,12 +512,11 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, if (err < 0) goto end; - formats[eid] = kmalloc(*len, GFP_KERNEL); + formats[eid] = kmemdup(buf, *len, GFP_KERNEL); if (formats[eid] == NULL) { err = -ENOMEM; goto end; } - memcpy(formats[eid], buf, *len); /* apply the format for each available sampling rate */ for (i = 0; i < ARRAY_SIZE(oxfw_rate_table); i++) { @@ -531,12 +530,11 @@ assume_stream_formats(struct snd_oxfw *oxfw, enum avc_general_plug_dir dir, continue; eid++; - formats[eid] = kmalloc(*len, GFP_KERNEL); + formats[eid] = kmemdup(buf, *len, GFP_KERNEL); if (formats[eid] == NULL) { err = -ENOMEM; goto end; } - memcpy(formats[eid], buf, *len); formats[eid][2] = avc_stream_rate_table[i]; } @@ -594,12 +592,11 @@ static int fill_stream_formats(struct snd_oxfw *oxfw, if (err < 0) break; - formats[eid] = kmalloc(len, GFP_KERNEL); + formats[eid] = kmemdup(buf, len, GFP_KERNEL); if (formats[eid] == NULL) { err = -ENOMEM; break; } - memcpy(formats[eid], buf, len); /* get next entry */ len = AVC_GENERIC_FRAME_MAXIMUM_BYTES; -- 1.9.1

2 1

[alsa-devel] [PATCH 30/31] sound/atom: use kmemdup rather than duplicating its implementation
by Andrzej Hajda 07 Aug '15

07 Aug '15

The patch was generated using fixed coccinelle semantic patch scripts/coccinelle/api/memdup.cocci [1]. [1]: http://permalink.gmane.org/gmane.linux.kernel/2014320 Signed-off-by: Andrzej Hajda <a.hajda(a)samsung.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 5a27861..3dc7358 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -352,10 +352,9 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, * copy from mailbox **/ if (msg_high.part.large) { - data = kzalloc(msg_low, GFP_KERNEL); + data = kmemdup((void *)msg->mailbox_data, msg_low, GFP_KERNEL); if (!data) return; - memcpy(data, (void *) msg->mailbox_data, msg_low); /* Copy command id so that we can use to put sst to reset */ dsp_hdr = (struct ipc_dsp_hdr *)data; cmd_id = dsp_hdr->cmd_id; -- 1.9.1

2 2

[alsa-devel] [PATCH 31/31] sound/soc: use kmemdup rather than duplicating its implementation
by Andrzej Hajda 07 Aug '15

07 Aug '15

The patch was generated using fixed coccinelle semantic patch scripts/coccinelle/api/memdup.cocci [1]. [1]: http://permalink.gmane.org/gmane.linux.kernel/2014320 Signed-off-by: Andrzej Hajda <a.hajda(a)samsung.com> --- sound/soc/soc-topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dadb5e..a5da91b 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -831,12 +831,12 @@ static int soc_tplg_denum_create_values(struct soc_enum *se, if (ec->items > sizeof(*ec->values)) return -EINVAL; - se->dobj.control.dvalues = - kmalloc(ec->items * sizeof(u32), GFP_KERNEL); + se->dobj.control.dvalues = kmemdup(ec->values, + ec->items * sizeof(u32), + GFP_KERNEL); if (!se->dobj.control.dvalues) return -ENOMEM; - memcpy(se->dobj.control.dvalues, ec->values, ec->items * sizeof(u32)); return 0; } -- 1.9.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.2
by Mark Brown 07 Aug '15

07 Aug '15

The following changes since commit 74d33293e467df61de1b1d8b2fbe29e550dec33b: Linux 4.2-rc5 (2015-08-02 18:34:55 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.2-rc5 for you to fetch changes up to d00a9e02178401433fc386e69c936f2039f07b57: Merge remote-tracking branches 'asoc/fix/cs4265', 'asoc/fix/intel' and 'asoc/fix/topology' into asoc-linus (2015-08-06 12:39:07 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.2 There are a couple of small driver specific fixes here but the overwhelming bulk of these changes are fixes to the topology ABI that has been newly introduced in v4.2. Once this makes it into a release we will have to firm this up but for now getting enhancements in before they've made it into a release is the most expedient thing. ---------------------------------------------------------------- Axel Lin (1): ASoC: cs4265: Fix setting dai format for Left/Right Justified Bard Liao (1): ASoC: rt5645: Fix lost pin setting for DMIC1 Jie Yang (2): ASoC: Intel: haswell: fix initialize 'NULL device *' issue ASoC: Intel: sst_byt: fix initialize 'NULL device *' issue Liam Girdwood (1): ASoC: topology: Add private data type and bump ABI version to 3 Mark Brown (3): Merge tag 'asoc-fix-v4.2-rc3' into asoc-fix-topology Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/cs4265', 'asoc/fix/intel' and 'asoc/fix/topology' into asoc-linus Mengdong Lin (3): ASoC: topology: fix typo in soc_tplg_kcontrol_bind_io() ASoC: topology: Update TLV support so we can support more TLV types ASoC: topology: Add ops support to byte controls UAPI Subhransu S. Prusty (1): ASoC: topology: Add subsequence in topology Vinod Koul (1): ASoC: topology: add private data to manifest include/uapi/sound/asoc.h | 35 +++++++++++------ sound/soc/codecs/cs4265.c | 10 ++--- sound/soc/codecs/rt5645.c | 2 + sound/soc/codecs/rt5645.h | 4 ++ sound/soc/intel/baytrail/sst-baytrail-ipc.c | 2 + sound/soc/intel/haswell/sst-haswell-ipc.c | 2 + sound/soc/soc-topology.c | 61 ++++++++++++++++++++--------- 7 files changed, 82 insertions(+), 34 deletions(-)

2 1

[alsa-devel] [PATCH 00/42] ASoC: Replace users of TLV_DB_RANGE_HEAD
by Lars-Peter Clausen 07 Aug '15

07 Aug '15

Usage of TLV_DB_RANGE_HEAD() is deprecated in favor of DECLARE_TLV_DB_RANGE(). The reason for this is that the former requires to specify the number of items inside the range container while the later is able to calculate this automatically and hence leaves less room for error. While non of the current users of TLV_DB_RANGE_HEAD() gets it wrong it still makes sense to clean this up since existing drivers are often used as templates for new drivers. As an added bonus the code is also slightly shorter. There are also a few drivers which use TLV_DB_RANGE_HEAD() with only a single sub-item. Those are updated to not use a range container at all since it is not necessary. Most of the conversion was auto-generated using the following coccinelle script: // <smpl> @@ declarer name DECLARE_TLV_DB_RANGE; identifier tlv; constant x; @@ -unsigned int tlv[] = { - TLV_DB_RANGE_HEAD(x), +DECLARE_TLV_DB_RANGE(tlv, ... -}; +); // </smpl> - Lars Lars-Peter Clausen (42): ASoC: 88pm860x: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: ab8500: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: adau1373: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: alc5623: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: alc5632: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l52: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l56: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l73: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da7210: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da7213: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da9055: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: jz4740: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9768: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98088: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98090: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98095: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9850: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9877: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5631: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5640: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5645: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5651: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5670: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5677: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: sgtl5000: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: ssm2602: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: tpa6130a2: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: twl4030: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: uda1380: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8350: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8400: Don't use range container for TLV with one entry ASoC: wm8737: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8753: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8961: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8962: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8990: Don't use range container for TLV with one entry ASoC: wm8991: Don't use range container for TLVs with one entry ASoC: wm8993: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9081: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9090: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9713: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm_hubs: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE sound/soc/codecs/88pm860x-codec.c | 28 +++++++++------------ sound/soc/codecs/ab8500-codec.c | 7 +++--- sound/soc/codecs/adau1373.c | 21 +++++++--------- sound/soc/codecs/alc5623.c | 7 +++--- sound/soc/codecs/alc5632.c | 7 +++--- sound/soc/codecs/cs42l52.c | 7 +++--- sound/soc/codecs/cs42l56.c | 21 +++++++--------- sound/soc/codecs/cs42l73.c | 14 +++++------ sound/soc/codecs/da7210.c | 22 +++++++---------- sound/soc/codecs/da7213.c | 15 +++++------ sound/soc/codecs/da9055.c | 15 +++++------ sound/soc/codecs/jz4740.c | 7 +++--- sound/soc/codecs/max9768.c | 7 +++--- sound/soc/codecs/max98088.c | 23 ++++++++--------- sound/soc/codecs/max98090.c | 42 ++++++++++++++----------------- sound/soc/codecs/max98095.c | 35 +++++++++++--------------- sound/soc/codecs/max9850.c | 7 +++--- sound/soc/codecs/max9877.c | 14 +++++------ sound/soc/codecs/rt5631.c | 7 +++--- sound/soc/codecs/rt5640.c | 7 +++--- sound/soc/codecs/rt5645.c | 7 +++--- sound/soc/codecs/rt5651.c | 7 +++--- sound/soc/codecs/rt5670.c | 7 +++--- sound/soc/codecs/rt5677.c | 7 +++--- sound/soc/codecs/sgtl5000.c | 7 +++--- sound/soc/codecs/ssm2602.c | 7 +++--- sound/soc/codecs/tpa6130a2.c | 14 +++++------ sound/soc/codecs/twl4030.c | 7 +++--- sound/soc/codecs/uda1380.c | 14 +++++------ sound/soc/codecs/wm8350.c | 7 +++--- sound/soc/codecs/wm8400.c | 5 +--- sound/soc/codecs/wm8737.c | 7 +++--- sound/soc/codecs/wm8753.c | 7 +++--- sound/soc/codecs/wm8961.c | 7 +++--- sound/soc/codecs/wm8962.c | 14 +++++------ sound/soc/codecs/wm8990.c | 5 +--- sound/soc/codecs/wm8991.c | 52 +++++++-------------------------------- sound/soc/codecs/wm8993.c | 7 +++--- sound/soc/codecs/wm9081.c | 7 +++--- sound/soc/codecs/wm9090.c | 21 +++++++--------- sound/soc/codecs/wm9713.c | 7 +++--- sound/soc/codecs/wm_hubs.c | 7 +++--- 42 files changed, 216 insertions(+), 327 deletions(-) -- 2.1.4

6 91

[alsa-devel] External plugin question
by Mina Almasry 07 Aug '15

07 Aug '15

I'm new to alsa dev and I'm trying to write a very simple external pcm plugin. I want my pcm plugin to output sound to multiple slave pcms when sound is being writei'd to it. So: snd_pcm_myplugin_writei(pcm, buffer, size) { foreach slave pcm { snd_pcm_writei(slave_pcm, buffer, size); } } I'm having trouble implementing this plugin because the external plugin sdk does not allow me to define the writei function for my plugin, but lets me define the transfer function which has a very different signature than writei: transfer(snd_pcm_ioplug_t *io, const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, snd_pcm_uframes_t size) how do I extract buffer and size from these arguments? Any tips on how I write my plugin. Mina

2 1

[alsa-devel] External plugin question
by Mina Almasry 07 Aug '15

07 Aug '15

I'm new to alsa dev and I'm trying to write a very simple external pcm plugin. I want my pcm plugin to output sound to multiple slave pcms when sound is being writei'd to it. So: snd_pcm_myplugin_writei(pcm, buffer, size) { foreach slave pcm { snd_pcm_writei(slave_pcm, buffer, size); } } I'm having trouble implementing this plugin because the external plugin sdk does not allow me to define the writei function for my plugin, but lets me define the transfer function which has a very different signature than writei: transfer(snd_pcm_ioplug_t <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___i_o_plug.html…> *io, const snd_pcm_channel_area_t <http://www.alsa-project.org/alsa-doc/alsa-lib/structsnd__pcm__channel__area…> *areas, snd_pcm_uframes_t <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gab01fcfe9…> offset, snd_pcm_uframes_t <http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#gab01fcfe9…> size) how do I extract buffer and size from these arguments? Any tips on how I write my plugin. (I'm aware of the multi plugin but I don't wont to use that. I want to write my own and introduce custom logic later.) Mina

1 0

[alsa-devel] [PATCH 2/2] ASoC: max98090: Enforce correct device sequencing when configuring a new
by yang.a.fang＠intel.com 07 Aug '15

07 Aug '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> audio route when there is an existing active audio route(s). This patch fixed recording noise issue while playback is active. We have some registers which require the device to be in full shutdown or to enter full shutdown before the register settings will take effect. Currently the driver is not shutting down the device when a new audio route is created. If a new audio route is made active while there is already an active audio route, then the required register sequencing is violated. A hardware shutdown toggle when creating a new audio route corrects the sequencing error. The device must remain in hardware shutdown for 40ms to allow the internal hardware core to fully shutdown. Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> --- sound/soc/codecs/max98090.c | 46 +++++++++++++++++++++++++++++++++++++------ sound/soc/codecs/max98090.h | 1 + 2 files changed, 41 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 91c9f9f..1516c6b 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -850,6 +850,19 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, return 0; } +static int max98090_shdn_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + if (event & SND_SOC_DAPM_POST_PMU) + max98090->shdn_pending = TRUE; + + return 0; + +} + static const char *mic1_mux_text[] = { "IN12", "IN56" }; static SOC_ENUM_SINGLE_DECL(mic1_mux_enum, @@ -1158,9 +1171,11 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SDOEN", M98090_REG_IO_CONFIGURATION, M98090_SDOEN_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DMICL_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICL_SHIFT, 0, NULL, 0), + M98090_DIGMICL_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("DMICR_ENA", M98090_REG_DIGITAL_MIC_ENABLE, - M98090_DIGMICR_SHIFT, 0, NULL, 0), + M98090_DIGMICR_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("AHPF", M98090_REG_FILTER_CONFIG, M98090_AHPF_SHIFT, 0, NULL, 0), @@ -1205,10 +1220,12 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { &max98090_right_adc_mixer_controls[0], ARRAY_SIZE(max98090_right_adc_mixer_controls)), - SND_SOC_DAPM_ADC("ADCL", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADLEN_SHIFT, 0), - SND_SOC_DAPM_ADC("ADCR", NULL, M98090_REG_INPUT_ENABLE, - M98090_ADREN_SHIFT, 0), + SND_SOC_DAPM_ADC_E("ADCL", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADLEN_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADCR", NULL, M98090_REG_INPUT_ENABLE, + M98090_ADREN_SHIFT, 0, max98090_shdn_event, + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_AIF_OUT("AIFOUTL", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), @@ -2539,9 +2556,26 @@ static int max98090_remove(struct snd_soc_codec *codec) return 0; } +static void max98090_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq, bool power_up) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(dapm); + struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); + + if (max98090->shdn_pending && power_up) { + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, 0); + msleep(40); + snd_soc_update_bits(codec, M98090_REG_DEVICE_SHUTDOWN, + M98090_SHDNN_MASK, M98090_SHDNN_MASK); + max98090->shdn_pending = FALSE; + } +} + static struct snd_soc_codec_driver soc_codec_dev_max98090 = { .probe = max98090_probe, .remove = max98090_remove, + .seq_notifier = max98090_seq_notifier, .set_bias_level = max98090_set_bias_level, }; diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 21ff743..bc610d9 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1543,6 +1543,7 @@ struct max98090_priv { unsigned int pa2en; unsigned int sidetone; bool master; + bool shdn_pending; }; int max98090_mic_detect(struct snd_soc_codec *codec, -- 1.7.9.5

1 0

[alsa-devel] [asoc:fix/topology 5/7] sound/soc/soc-topology.c:596:14: sparse: incorrect type in assignment (different base types)
by kbuild test robot 06 Aug '15

06 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git fix/topology head: c7bcf8777a539e64dafc7417c00047aee6eb8909 commit: 28a87eebcad40101b1b273cbd4f2a02c104f9367 [5/7] ASoC: topology: Update TLV support so we can support more TLV types reproduce: # apt-get install sparse git checkout 28a87eebcad40101b1b273cbd4f2a02c104f9367 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) sound/soc/soc-topology.c:1696:45: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:190:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:191:39: sparse: incorrect type in return expression (different base types) sound/soc/soc-topology.c:191:39: expected int sound/soc/soc-topology.c:191:39: got restricted __le32 [usertype] reg sound/soc/soc-topology.c:203:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:204:39: sparse: incorrect type in return expression (different base types) sound/soc/soc-topology.c:204:39: expected int sound/soc/soc-topology.c:204:39: got restricted __le32 [usertype] shift sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:228:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:248:20: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:519:42: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:521:42: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:523:42: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:534:61: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:536:61: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:538:62: sparse: restricted __le32 degrades to integer >> sound/soc/soc-topology.c:596:14: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:596:14: expected unsigned int [unsigned] <noident> sound/soc/soc-topology.c:596:14: got restricted __le32 [usertype] min sound/soc/soc-topology.c:597:22: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:609:17: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:612:15: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:616:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:667:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:677:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:677:27: expected unsigned int [unsigned] [addressable] [assigned] access sound/soc/soc-topology.c:677:27: got restricted __le32 [usertype] access sound/soc/soc-topology.c:679:26: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:679:26: expected int [signed] max sound/soc/soc-topology.c:679:26: got restricted __le32 [usertype] max sound/soc/soc-topology.c:748:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:758:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:758:27: expected unsigned int [unsigned] [addressable] [assigned] access sound/soc/soc-topology.c:758:27: got restricted __le32 [usertype] access sound/soc/soc-topology.c:770:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:770:25: expected int [signed] max sound/soc/soc-topology.c:770:25: got restricted __le32 [usertype] max sound/soc/soc-topology.c:771:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:771:25: expected int [signed] min sound/soc/soc-topology.c:771:25: got restricted __le32 [usertype] min sound/soc/soc-topology.c:772:28: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:772:28: expected unsigned int [unsigned] invert:1 sound/soc/soc-topology.c:772:28: got restricted __le32 [usertype] invert sound/soc/soc-topology.c:773:34: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:773:34: expected int [signed] platform_max sound/soc/soc-topology.c:773:34: got restricted __le32 [usertype] platform_max sound/soc/soc-topology.c:824:44: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:828:27: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:855:15: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:859:27: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:863:9: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:887:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:905:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:905:27: expected unsigned int [unsigned] [addressable] [assigned] access sound/soc/soc-topology.c:905:27: got restricted __le32 [usertype] access sound/soc/soc-topology.c:913:27: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:913:27: expected unsigned int [unsigned] items sound/soc/soc-topology.c:913:27: got restricted __le32 [usertype] items sound/soc/soc-topology.c:914:26: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:914:26: expected unsigned int [unsigned] mask sound/soc/soc-topology.c:914:26: got restricted __le32 [usertype] mask sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:920:36: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:994:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:994:45: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1001:28: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1013:60: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:1013:60: expected unsigned long [unsigned] [usertype] size sound/soc/soc-topology.c:1013:60: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1020:59: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:1020:59: expected unsigned long [unsigned] [usertype] size sound/soc/soc-topology.c:1020:59: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1023:60: sparse: incorrect type in argument 3 (different base types) sound/soc/soc-topology.c:1023:60: expected unsigned long [unsigned] [usertype] size sound/soc/soc-topology.c:1023:60: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1005:41: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1040:24: sparse: incorrect type in initializer (different base types) sound/soc/soc-topology.c:1040:24: expected int [signed] count sound/soc/soc-topology.c:1040:24: got restricted __le32 [usertype] count sound/soc/soc-topology.c:1043:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1043:45: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1049:27: sparse: incorrect type in argument 4 (different base types) sound/soc/soc-topology.c:1049:27: expected unsigned long [unsigned] [usertype] bytes sound/soc/soc-topology.c:1049:27: got restricted __le32 [usertype] payload_size sound/soc/soc-topology.c:1107:33: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1120:30: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1120:30: expected unsigned int [unsigned] access sound/soc/soc-topology.c:1120:30: got restricted __le32 [usertype] access sound/soc/soc-topology.c:1132:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1132:25: expected int [signed] max sound/soc/soc-topology.c:1132:25: got restricted __le32 [usertype] max sound/soc/soc-topology.c:1133:25: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1133:25: expected int [signed] min sound/soc/soc-topology.c:1133:25: got restricted __le32 [usertype] min sound/soc/soc-topology.c:1134:28: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1134:28: expected unsigned int [unsigned] invert:1 sound/soc/soc-topology.c:1134:28: got restricted __le32 [usertype] invert sound/soc/soc-topology.c:1135:34: sparse: incorrect type in assignment (different base types) sound/soc/soc-topology.c:1135:34: expected int [signed] platform_max sound/soc/soc-topology.c:1135:34: got restricted __le32 [usertype] platform_max sound/soc/soc-topology.c:1179:25: sparse: restricted __le32 degrades to integer sound/soc/soc-topology.c:1200:20: sparse: incorrect type in assignment (different base types) vim +596 sound/soc/soc-topology.c 580 return 0; 581 } 582 583 584 static int soc_tplg_create_tlv_db_scale(struct soc_tplg *tplg, 585 struct snd_kcontrol_new *kc, struct snd_soc_tplg_tlv_dbscale *scale) 586 { 587 unsigned int item_len = 2 * sizeof(unsigned int); 588 unsigned int *p; 589 590 p = kzalloc(item_len + 2 * sizeof(unsigned int), GFP_KERNEL); 591 if (!p) 592 return -ENOMEM; 593 594 p[0] = SNDRV_CTL_TLVT_DB_SCALE; 595 p[1] = item_len; > 596 p[2] = scale->min; 597 p[3] = (scale->step & TLV_DB_SCALE_MASK) 598 | (scale->mute ? TLV_DB_SCALE_MUTE : 0); 599 600 kc->tlv.p = (void *)p; 601 return 0; 602 } 603 604 static int soc_tplg_create_tlv(struct soc_tplg *tplg, --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm8731: Drop wm8731_writeable function
by Axel Lin 06 Aug '15

06 Aug '15

When .max_register is set and .writeable_reg is not implement, registers between 0 and .max_register are writeable. This is the same as current implementation of wm8731_writeable(), so just drop implementation for .writeable_reg callback. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8731.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c862232..ace8645 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -79,11 +79,6 @@ static bool wm8731_volatile(struct device *dev, unsigned int reg) return reg == WM8731_RESET; } -static bool wm8731_writeable(struct device *dev, unsigned int reg) -{ - return reg <= WM8731_RESET; -} - #define wm8731_reset(m) regmap_write(m, WM8731_RESET, 0) static const char *wm8731_input_select[] = {"Line In", "Mic"}; @@ -655,7 +650,6 @@ static const struct regmap_config wm8731_regmap = { .max_register = WM8731_RESET, .volatile_reg = wm8731_volatile, - .writeable_reg = wm8731_writeable, .cache_type = REGCACHE_RBTREE, .reg_defaults = wm8731_reg_defaults, -- 2.1.0

3 5

[alsa-devel] [PATCH] ASoC: wm8741: Simplify wm8741_set_dai_sysclk implementation
by Axel Lin 06 Aug '15

06 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8741.c | 41 ++++++++++++++--------------------------- 1 file changed, 14 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 430fa7d..9723e44 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -278,51 +278,38 @@ static int wm8741_set_dai_sysclk(struct snd_soc_dai *codec_dai, switch (freq) { case 0: wm8741->sysclk_constraints = NULL; - wm8741->sysclk = freq; - return 0; - + break; case 11289600: wm8741->sysclk_constraints = &constraints_11289; - wm8741->sysclk = freq; - return 0; - + break; case 12288000: wm8741->sysclk_constraints = &constraints_12288; - wm8741->sysclk = freq; - return 0; - + break; case 16384000: wm8741->sysclk_constraints = &constraints_16384; - wm8741->sysclk = freq; - return 0; - + break; case 16934400: wm8741->sysclk_constraints = &constraints_16934; - wm8741->sysclk = freq; - return 0; - + break; case 18432000: wm8741->sysclk_constraints = &constraints_18432; - wm8741->sysclk = freq; - return 0; - + break; case 22579200: case 33868800: wm8741->sysclk_constraints = &constraints_22579; - wm8741->sysclk = freq; - return 0; - + break; case 24576000: wm8741->sysclk_constraints = &constraints_24576; - wm8741->sysclk = freq; - return 0; - + break; case 36864000: wm8741->sysclk_constraints = &constraints_36864; - wm8741->sysclk = freq; - return 0; + break; + default: + return -EINVAL; } - return -EINVAL; + + wm8741->sysclk = freq; + return 0; } static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, -- 2.1.0

3 2

[alsa-devel] query on upstream ASoC driver split across SOUND and DRM subsystems
by Bayyavarapu, Maruthi 06 Aug '15

06 Aug '15

Hi, I want to send AMD ASoC driver for review. This driver has a dependency with AMD's 'DRM' subsystem components, which are not upstreamed yet. Please suggest, whether this ASoC driver can be upstreamed via 'DRM' tree or any alternate approach is preferred. The link to ASoC driver is : http://cgit.freedesktop.org/~agd5f/linux/commit/?h=acp-wip&id=2b67a44150988… ASoC commits and dependencies are currently in 'acp-wip' branch of : git://people.freedesktop.org/~agd5f/linux Thanks, Maruthi

3 2

[alsa-devel] [PATCH v2 0/3] ASoC: topology: UAPI updates for TLV, ops and types
by Liam Girdwood 06 Aug '15

06 Aug '15

This series updates the topology UAPI to allow easier addition of new TLV object types, generic operations and added a private data type for the kernel topology core. Changes since V1 * Split into several patches for UAPI changes. * Fixed {} usage in if statement. * Added __le32 data[SND_SOC_TPLG_TLV_SIZE] back to struct snd_soc_tplg_ctl_tlv Liam Girdwood (1): ASoC: topology: Add private data type and bump ABI version to 3 Mengdong Lin (2): ASoC: topology: Update TLV support so we can support more TLV types ASoC: topology: Add ops support to byte controls UAPI include/uapi/sound/asoc.h | 33 ++++++++++++++++++--------- sound/soc/soc-topology.c | 58 +++++++++++++++++++++++++++++++++-------------- 2 files changed, 63 insertions(+), 28 deletions(-) -- 2.1.4

3 6

[alsa-devel] [PATCH] ASoC: rl6231: Simplify DMIC divider calculation expression
by Anatol Pomozov 06 Aug '15

06 Aug '15

Existing implementation checks all divider values and tracks 'red' proximity value for the frequency. But as divider array is monotonically increasing the first divider that gives DMIC rate in 3MHz range is the best one we should use. No need for 'red' zone tracking. Additionally make sure that DMIC frequency is higher 1MHz. Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/codecs/rl6231.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 57e51c1..aca479f 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -62,31 +62,31 @@ int rl6231_get_pre_div(struct regmap *map, unsigned int reg, int sft) EXPORT_SYMBOL_GPL(rl6231_get_pre_div); /** - * rl6231_calc_dmic_clk - Calculate the parameter of dmic. + * rl6231_calc_dmic_clk - Calculate the frequency divider parameter of dmic. * * @rate: base clock rate. * - * Choose dmic clock between 1MHz and 3MHz. - * It is better for clock to approximate 3MHz. + * Choose divider parameter that gives the highest possible DMIC frequency in + * 1MHz - 3MHz range. */ int rl6231_calc_dmic_clk(int rate) { - int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL; - int i, red, bound, temp; + int div[] = {2, 3, 4, 6, 8, 12}; + int i; + + if (rate < 1000000 * div[0]) { + pr_warn("Base clock rate %d is too low\n", rate); + return -EINVAL; + } - red = 3000000 * 12; for (i = 0; i < ARRAY_SIZE(div); i++) { - bound = div[i] * 3000000; - if (rate > bound) - continue; - temp = bound - rate; - if (temp < red) { - red = temp; - idx = i; - } + /* find divider that gives DMIC frequency below 3MHz */ + if (3000000 * div[i] >= rate) + return i; } - return idx; + pr_warn("Base clock rate %d is too high\n", rate); + return -EINVAL; } EXPORT_SYMBOL_GPL(rl6231_calc_dmic_clk); -- 2.5.0.rc2.392.g76e840b

3 2

[alsa-devel] [PATCH v2] ASoC: topology: Use kcalloc instead of kzalloc for array allocation
by Axel Lin 06 Aug '15

06 Aug '15

Also remove unnecessary memset. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v2: One more kcalloc conversion in soc_tplg_dapm_widget_dmixer_create sound/soc/soc-topology.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dadb5e..e4b7fc6 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1069,7 +1069,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( struct snd_soc_tplg_mixer_control *mc; int i, err; - kc = kzalloc(sizeof(*kc) * num_kcontrols, GFP_KERNEL); + kc = kcalloc(num_kcontrols, sizeof(*kc), GFP_KERNEL); if (kc == NULL) return NULL; @@ -1250,7 +1250,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( struct snd_kcontrol_new *kc; int i, err; - kc = kzalloc(sizeof(*kc) * count, GFP_KERNEL); + kc = kcalloc(count, sizeof(*kc), GFP_KERNEL); if (!kc) return NULL; @@ -1273,7 +1273,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); - memset(kc, 0, sizeof(*kc)); kc[i].name = be->hdr.name; kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; -- 2.1.0

2 1

[alsa-devel] Verb for 0x17 pin config
by Kailang 06 Aug '15

06 Aug '15

Hi Takashi, commit http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=977e… ALSA: hda/realtek - Support Dell headset mode for ALC298 Dell had use verb to do speaker preset. So, The same SSID maybe 0x17 verb was different. 0x17 0x90170110 0x17 0x90170140 0x17 0x90170150 Upper verb will show on this case. Could I change it to quirk for SSID? BR, Kailang

2 6

Re: [alsa-devel] [ALC668]: Asus N751JK - mic selection not working
by Raymond Yau 06 Aug '15

06 Aug '15

>> >> Ports: >> >> analog-output-speaker: Speakers (priority: 10000) >> analog-output-headphones: Headphones (priority: 9000, not available) >> Active Port: analog-output-speaker >> >> >> When you select model=dell-headset-multi, the driver create "Headphone Mic Jack" kctl with type 1 which is SND_JACK_HEADPHONE >> >> type 2 is SND_JACK_MICROPHONE >> >> type 3 is SND_JACK_HEADSET but hda jack does not use this type >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/include/s… >> >> JACK created Headphone Mic, type 1 >> CTRL: add: Headphone Mic Jack:0 >> JACK report Headphone Mic, status 0 >> JACK created Headset Mic Phantom, type 2 >> CTRL: add: Headset Mic Phantom Jack:0 >> JACK report Headset Mic Phantom, status 2 >> CTL Notify: Headset Mic Phantom Jack:0, mask=1 >> JACK created Internal Mic Phantom, type 2 >> CTRL: add: Internal Mic Phantom Jack:0 >> JACK report Internal Mic Phantom, status 2 >> CTL Notify: Internal Mic Phantom Jack:0, mask=1 >> JACK created Speaker Front Phantom, type 4 >> CTRL: add: Speaker Front Phantom Jack:0 >> JACK report Speaker Front Phantom, status 4 >> CTL Notify: Speaker Front Phantom Jack:0, mask=1 >> JACK created External Subwoofer, type 4 >> CTRL: add: External Subwoofer Jack:0 >> >> The problem of hda jack kctls are those retaskable I/O jack controls >> >> 1) multi io jacks in those 3 stack desktop >> 2) headphone/mic jack in asus netbook >> 3) dell-headset-multi combo jack >> >> the role of the jack in 1) change from input to output when "Channel mode" change from "2ch" to "4ch" or "6ch" >> >> the role of the jack in 2) and 3) change from Output to Input when "Capture Source" change from Internal Mic/Headset Mic to Headphone Mic >> >> In your case, Headphone Mic Jack kctl return the presence of HP jack except when the combo jack is set to support conventional mic (i.e. when the driver change pin ctl of hp pin node 0x15 to zero and set pin ctl of mic pin to IN), >> >> The driver already unmute the internal speaker when you change capture source to Headphone Mic >> >> The only point is whether pulseaudio still regard this Headphone Mic Jack as the availability of the headphone port when the capture source is Headphone Mic >> >> http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=ca4942e89cf462… >> >> > > > I really appreciate your efforts in this matter. Unfortunately I understand parts of our talks and I am unsure what I can do. > I am not a kernel / driver developer. > > Are there any steps I can do knowing my limited skills on driver development? > If so, would you mind to describe them step by step, so this beginner understands? :D Don't forget cc David since he implement dell-headset-multi https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… static void alc_update_headset_mode(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + + int new_headset_mode; if (!snd_hda_jack_detect(codec, hp_pin)) + new_headset_mode = ALC_HEADSET_MODE_UNPLUGGED; + else if (mux_pin == spec->headset_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_HEADSET; + else if (mux_pin == spec->headphone_mic_pin) + new_headset_mode = ALC_HEADSET_MODE_MIC; + else + new_headset_mode = ALC_HEADSET_MODE_HEADPHONE; + + if (new_headset_mode == spec->current_headset_mode) + return; + + switch (new_headset_mode) { + case ALC_HEADSET_MODE_UNPLUGGED: + alc_headset_mode_unplugged(codec); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADSET: + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; + break; + case ALC_HEADSET_MODE_MIC: + alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); + spec->gen.hp_jack_present = false; + break; + case ALC_HEADSET_MODE_HEADPHONE: + alc_headset_mode_default(codec); + spec->gen.hp_jack_present = true; + break; + } + if (new_headset_mode != ALC_HEADSET_MODE_MIC) { + snd_hda_set_pin_ctl_cache(codec, hp_pin, + AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN); + if (spec->headphone_mic_pin) + snd_hda_set_pin_ctl_cache(codec, spec->headphone_mic_pin, + PIN_VREFHIZ); + } + spec->current_headset_mode = new_headset_mode; + + snd_hda_gen_update_outputs(codec); +} You have to post the output of pactl list sources when the is no jack plugged, headphone plugged, headset plugged and mic jack plugged The logic is defined in alc_update_headset_mode() Although auto mic selection is disabled when the combo jack can support conventional mic and headset mic The capture source of these kind of laptop is not as same as the capture source of ac97 codec of pci sound cards, it is not really manually selected in sound preference When there is no jack plugged in, the capture source should be internal mic. As the codec cannot differentitate headset, headphone and mic when there is jack plugged in and driver use mux_pin to determine the type of the jack In pulseaudio term, [Jack Headphone Mic] can determine the availability of internal mic is YES when unplugged but unknown when plugged http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Jack Headphone Mic] The availability of headphone mic is unknown when plugged and NO when unplugged http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… [Jack Headphone Mic] The availability of headset mic is unknown when plugged and NO when unplugged http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… It is not exactly UNKNOWN but determined by the capture source when plugged

2 4

[alsa-devel] [PATCH 5/6][RESEND] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 154bcf6..8e5ff5e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

1 0

[alsa-devel] [PATCH 4/6][RESEND] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7f4f0b9..154bcf6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

1 0

[alsa-devel] [PATCH 3/6][RESEND] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a83b900..7f4f0b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

1 0

[alsa-devel] [PATCH 2/6][RESEND] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8185edc..a83b900 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

1 0

[alsa-devel] [PATCH 1/6][RESEND] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 05 Aug '15

05 Aug '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- This is a resend without changes, to keep the whole series together. sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 484ff20..8185edc 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

1 0

[alsa-devel] [PATCH] ALSA: aoa-soundbus: Switch to dev_pm_ops
by Lars-Peter Clausen 05 Aug '15

05 Aug '15

Update the aoa-soundbus framework to use dev_pm_ops rather than the deprecated legacy suspend and resume callbacks. Since there isn't anything special to do at the bus level the bus driver does not have to implement any callbacks. The device driver core will automatically pick up and execute the device's PM ops. As there is only a single aoa-soundbus driver implementing suspend and resume, update both the core and driver at the same time to avoid unnecessary code churn. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/aoa/fabrics/layout.c | 21 ++++++++++++--------- sound/aoa/soundbus/core.c | 28 ---------------------------- sound/aoa/soundbus/soundbus.h | 2 -- 3 files changed, 12 insertions(+), 39 deletions(-) diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 9dc5806..8f71f7e 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1120,10 +1120,10 @@ static int aoa_fabric_layout_remove(struct soundbus_dev *sdev) return 0; } -#ifdef CONFIG_PM -static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state) +#ifdef CONFIG_PM_SLEEP +static int aoa_fabric_layout_suspend(struct device *dev) { - struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); + struct layout_dev *ldev = dev_get_drvdata(dev); if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off) ldev->gpio.methods->all_amps_off(&ldev->gpio); @@ -1131,15 +1131,19 @@ static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t sta return 0; } -static int aoa_fabric_layout_resume(struct soundbus_dev *sdev) +static int aoa_fabric_layout_resume(struct device *dev) { - struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); + struct layout_dev *ldev = dev_get_drvdata(dev); if (ldev->gpio.methods && ldev->gpio.methods->all_amps_restore) ldev->gpio.methods->all_amps_restore(&ldev->gpio); return 0; } + +static SIMPLE_DEV_PM_OPS(aoa_fabric_layout_pm_ops, + aoa_fabric_layout_suspend, aoa_fabric_layout_resume); + #endif static struct soundbus_driver aoa_soundbus_driver = { @@ -1147,12 +1151,11 @@ static struct soundbus_driver aoa_soundbus_driver = { .owner = THIS_MODULE, .probe = aoa_fabric_layout_probe, .remove = aoa_fabric_layout_remove, -#ifdef CONFIG_PM - .suspend = aoa_fabric_layout_suspend, - .resume = aoa_fabric_layout_resume, -#endif .driver = { .owner = THIS_MODULE, +#ifdef CONFIG_PM_SLEEP + .pm = &aoa_fabric_layout_pm_ops, +#endif } }; diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c index 3edf736..70bcaa7 100644 --- a/sound/aoa/soundbus/core.c +++ b/sound/aoa/soundbus/core.c @@ -126,30 +126,6 @@ static void soundbus_device_shutdown(struct device *dev) drv->shutdown(soundbus_dev); } -#ifdef CONFIG_PM - -static int soundbus_device_suspend(struct device *dev, pm_message_t state) -{ - struct soundbus_dev * soundbus_dev = to_soundbus_device(dev); - struct soundbus_driver * drv = to_soundbus_driver(dev->driver); - - if (dev->driver && drv->suspend) - return drv->suspend(soundbus_dev, state); - return 0; -} - -static int soundbus_device_resume(struct device * dev) -{ - struct soundbus_dev * soundbus_dev = to_soundbus_device(dev); - struct soundbus_driver * drv = to_soundbus_driver(dev->driver); - - if (dev->driver && drv->resume) - return drv->resume(soundbus_dev); - return 0; -} - -#endif /* CONFIG_PM */ - /* soundbus_dev_attrs is declared in sysfs.c */ ATTRIBUTE_GROUPS(soundbus_dev); static struct bus_type soundbus_bus_type = { @@ -158,10 +134,6 @@ static struct bus_type soundbus_bus_type = { .uevent = soundbus_uevent, .remove = soundbus_device_remove, .shutdown = soundbus_device_shutdown, -#ifdef CONFIG_PM - .suspend = soundbus_device_suspend, - .resume = soundbus_device_resume, -#endif .dev_groups = soundbus_dev_groups, }; diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h index 21e756c..ae40224 100644 --- a/sound/aoa/soundbus/soundbus.h +++ b/sound/aoa/soundbus/soundbus.h @@ -188,8 +188,6 @@ struct soundbus_driver { int (*probe)(struct soundbus_dev* dev); int (*remove)(struct soundbus_dev* dev); - int (*suspend)(struct soundbus_dev* dev, pm_message_t state); - int (*resume)(struct soundbus_dev* dev); int (*shutdown)(struct soundbus_dev* dev); struct device_driver driver; -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: tegra: Use devm_clk_get
by Vaishali Thakkar 05 Aug '15

05 Aug '15

This patch introduces the use of managed resource function devm_clk_get instead of clk_get and removes corresponding calls to clk_put in the probe and remove functions. To be compatible with the change various gotos are replaced with direct returns, and unneeded labels are dropped. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/tegra/tegra30_ahub.c | 49 ++++++++++++++---------------------------- 1 file changed, 16 insertions(+), 33 deletions(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index bc94e5d..989b1e8 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -549,54 +549,51 @@ static int tegra30_ahub_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Can't get reset %s\n", configlink_mods[i].rst_name); ret = PTR_ERR(rst); - goto err; + return ret; } ret = reset_control_deassert(rst); reset_control_put(rst); if (ret) - goto err; + return ret; } ahub = devm_kzalloc(&pdev->dev, sizeof(struct tegra30_ahub), GFP_KERNEL); if (!ahub) { dev_err(&pdev->dev, "Can't allocate tegra30_ahub\n"); - ret = -ENOMEM; - goto err; + return -ENOMEM; } dev_set_drvdata(&pdev->dev, ahub); ahub->soc_data = soc_data; ahub->dev = &pdev->dev; - ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio"); + ahub->clk_d_audio = devm_clk_get(&pdev->dev, "d_audio"); if (IS_ERR(ahub->clk_d_audio)) { dev_err(&pdev->dev, "Can't retrieve ahub d_audio clock\n"); ret = PTR_ERR(ahub->clk_d_audio); - goto err; + return ret; } - ahub->clk_apbif = clk_get(&pdev->dev, "apbif"); + ahub->clk_apbif = devm_clk_get(&pdev->dev, "apbif"); if (IS_ERR(ahub->clk_apbif)) { dev_err(&pdev->dev, "Can't retrieve ahub apbif clock\n"); ret = PTR_ERR(ahub->clk_apbif); - goto err_clk_put_d_audio; + return ret; } res0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res0) { dev_err(&pdev->dev, "No apbif memory resource\n"); - ret = -ENODEV; - goto err_clk_put_apbif; + return -ENODEV; } region = devm_request_mem_region(&pdev->dev, res0->start, resource_size(res0), DRV_NAME); if (!region) { dev_err(&pdev->dev, "request region apbif failed\n"); - ret = -EBUSY; - goto err_clk_put_apbif; + return -EBUSY; } ahub->apbif_addr = res0->start; @@ -604,8 +601,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) resource_size(res0)); if (!regs_apbif) { dev_err(&pdev->dev, "ioremap apbif failed\n"); - ret = -ENOMEM; - goto err_clk_put_apbif; + return -ENOMEM; } ahub->regmap_apbif = devm_regmap_init_mmio(&pdev->dev, regs_apbif, @@ -613,31 +609,28 @@ static int tegra30_ahub_probe(struct platform_device *pdev) if (IS_ERR(ahub->regmap_apbif)) { dev_err(&pdev->dev, "apbif regmap init failed\n"); ret = PTR_ERR(ahub->regmap_apbif); - goto err_clk_put_apbif; + return ret; } regcache_cache_only(ahub->regmap_apbif, true); res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); if (!res1) { dev_err(&pdev->dev, "No ahub memory resource\n"); - ret = -ENODEV; - goto err_clk_put_apbif; + return -ENODEV; } region = devm_request_mem_region(&pdev->dev, res1->start, resource_size(res1), DRV_NAME); if (!region) { dev_err(&pdev->dev, "request region ahub failed\n"); - ret = -EBUSY; - goto err_clk_put_apbif; + return -EBUSY; } regs_ahub = devm_ioremap(&pdev->dev, res1->start, resource_size(res1)); if (!regs_ahub) { dev_err(&pdev->dev, "ioremap ahub failed\n"); - ret = -ENOMEM; - goto err_clk_put_apbif; + return -ENOMEM; } ahub->regmap_ahub = devm_regmap_init_mmio(&pdev->dev, regs_ahub, @@ -645,7 +638,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) if (IS_ERR(ahub->regmap_ahub)) { dev_err(&pdev->dev, "ahub regmap init failed\n"); ret = PTR_ERR(ahub->regmap_ahub); - goto err_clk_put_apbif; + return ret; } regcache_cache_only(ahub->regmap_ahub, true); @@ -662,12 +655,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) err_pm_disable: pm_runtime_disable(&pdev->dev); -err_clk_put_apbif: - clk_put(ahub->clk_apbif); -err_clk_put_d_audio: - clk_put(ahub->clk_d_audio); - ahub = NULL; -err: + return ret; } @@ -680,11 +668,6 @@ static int tegra30_ahub_remove(struct platform_device *pdev) if (!pm_runtime_status_suspended(&pdev->dev)) tegra30_ahub_runtime_suspend(&pdev->dev); - clk_put(ahub->clk_apbif); - clk_put(ahub->clk_d_audio); - - ahub = NULL; - return 0; } -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/4] Add SPDIF support for rockchip
by Sjoerd Simons 05 Aug '15

05 Aug '15

This patchset adds support for the Rockchip SPDIF transceiver as present on RK3066, RK3188 and RK3288 boards and enables it on a Radxa rock pro. Tested on a Radxa Rock Pro board. Changes in v2: - Remove platform: module alias as it was unused - Call MODULE_DEVICE_TABLE(of, ) right after the of match table - use rk_spdif as a prefix consistenly throughout the driver - Check return code of regmap_update and bubble it up - Sort the spdif node properties - Drop the 0x prefix from the node name - Rename the spdif@ node to sound@ Sjoerd Simons (4): ASoC: dt-bindings: add rockchip tranceiver bindings ASoc: rockchip: Add rockchip SPDIF transceiver driver ARM: dts: rockchip: Add SPDIF transceiver for RK3188 ARM: dts: rockchip: Add SPDIF optical out on Radxa Rock .../devicetree/bindings/sound/rockchip-spdif.txt | 41 +++ arch/arm/boot/dts/rk3188-radxarock.dts | 19 + arch/arm/boot/dts/rk3188.dtsi | 22 ++ sound/soc/rockchip/Kconfig | 8 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_spdif.c | 381 +++++++++++++++++++++ sound/soc/rockchip/rockchip_spdif.h | 63 ++++ 7 files changed, 536 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-spdif.txt create mode 100644 sound/soc/rockchip/rockchip_spdif.c create mode 100644 sound/soc/rockchip/rockchip_spdif.h -- 2.5.0

2 4

[alsa-devel] [PATCH] ASoC: topology: Update ABI with improvements for TLV, operations and private data.
by Liam Girdwood 05 Aug '15

05 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This is the kernel patch - a corresponding alsa-lib patch has also been sent. This patch updates the topology ABI version and adds support for private data types, specfic structures for each TLV data type and refactors the operations mapping structure to be generic (so that it can be used by other types). This code is only used within Intel atm and is not in an official alsa-lib release so impact should be minimali for the ABI update. Currently the TLV topology structure is targeted at only supporting the DB scale data. This patch extends support for the other TLV types so they can be easily added at a later stage. TLV structure is moved to common topology control header since it's a common field for controls and can be processed in a general way. Users must set a proper access flag for a control since it's used to decide if the TLV field is valid and if a TLV callback is needed. Removed the following fields from topology TLV struct: - size/count: type can decide the size. - numid: not needed to initialize TLV for kcontrol. - data: replaced by the type specific struct. Added TLV structure to generic control header and removed TLV structure from mixer control. Added generic ops to geneic control header. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/uapi/sound/asoc.h | 32 +++++++++++++++++--------- sound/soc/soc-topology.c | 58 +++++++++++++++++++++++++++++++++-------------- 2 files changed, 62 insertions(+), 28 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 2819fc1..69383b0 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -77,7 +77,7 @@ #define SND_SOC_TPLG_NUM_TEXTS 16 /* ABI version */ -#define SND_SOC_TPLG_ABI_VERSION 0x2 +#define SND_SOC_TPLG_ABI_VERSION 0x3 /* Max size of TLV data */ #define SND_SOC_TPLG_TLV_SIZE 32 @@ -97,7 +97,8 @@ #define SND_SOC_TPLG_TYPE_PCM 7 #define SND_SOC_TPLG_TYPE_MANIFEST 8 #define SND_SOC_TPLG_TYPE_CODEC_LINK 9 -#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_CODEC_LINK +#define SND_SOC_TPLG_TYPE_PDATA 10 +#define SND_SOC_TPLG_TYPE_MAX SND_SOC_TPLG_TYPE_PDATA /* vendor block IDs - please add new vendor types to end */ #define SND_SOC_TPLG_TYPE_VENDOR_FW 1000 @@ -137,11 +138,18 @@ struct snd_soc_tplg_private { /* * Kcontrol TLV data. */ +struct snd_soc_tplg_tlv_dbscale { + __le32 min; + __le32 step; + __le32 mute; +} __attribute__((packed)); + struct snd_soc_tplg_ctl_tlv { - __le32 size; /* in bytes aligned to 4 */ - __le32 numid; /* control element numeric identification */ - __le32 count; /* number of elem in data array */ - __le32 data[SND_SOC_TPLG_TLV_SIZE]; + __le32 size; /* in bytes of this structure */ + __le32 type; /* SNDRV_CTL_TLVT_*, type of TLV */ + union { + struct snd_soc_tplg_tlv_dbscale scale; + }; } __attribute__((packed)); /* @@ -155,9 +163,11 @@ struct snd_soc_tplg_channel { } __attribute__((packed)); /* - * Kcontrol Operations IDs + * Genericl Operations IDs, for binding Kcontrol or Bytes ext ops + * Kcontrol ops need get/put/info. + * Bytes ext ops need get/put. */ -struct snd_soc_tplg_kcontrol_ops_id { +struct snd_soc_tplg_io_ops { __le32 get; __le32 put; __le32 info; @@ -171,8 +181,8 @@ struct snd_soc_tplg_ctl_hdr { __le32 type; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; __le32 access; - struct snd_soc_tplg_kcontrol_ops_id ops; - __le32 tlv_size; /* non zero means control has TLV data */ + struct snd_soc_tplg_io_ops ops; + struct snd_soc_tplg_ctl_tlv tlv; } __attribute__((packed)); /* @@ -260,7 +270,6 @@ struct snd_soc_tplg_mixer_control { __le32 invert; __le32 num_channels; struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; - struct snd_soc_tplg_ctl_tlv tlv; struct snd_soc_tplg_private priv; } __attribute__((packed)); @@ -304,6 +313,7 @@ struct snd_soc_tplg_bytes_control { __le32 mask; __le32 base; __le32 num_regs; + struct snd_soc_tplg_io_ops ext_ops; struct snd_soc_tplg_private priv; } __attribute__((packed)); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dadb5e..bebd6f3 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -33,6 +33,7 @@ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/soc-topology.h> +#include <sound/tlv.h> /* * We make several passes over the data (since it wont necessarily be ordered) @@ -579,28 +580,51 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, return 0; } + +static int soc_tplg_create_tlv_db_scale(struct soc_tplg *tplg, + struct snd_kcontrol_new *kc, struct snd_soc_tplg_tlv_dbscale *scale) +{ + unsigned int item_len = 2 * sizeof(unsigned int); + unsigned int *p; + + p = kzalloc(item_len + 2 * sizeof(unsigned int), GFP_KERNEL); + if (!p) + return -ENOMEM; + + p[0] = SNDRV_CTL_TLVT_DB_SCALE; + p[1] = item_len; + p[2] = scale->min; + p[3] = (scale->step & TLV_DB_SCALE_MASK) + | (scale->mute ? TLV_DB_SCALE_MUTE : 0); + + kc->tlv.p = (void *)p; + return 0; +} + static int soc_tplg_create_tlv(struct soc_tplg *tplg, - struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_tlv *tplg_tlv) + struct snd_kcontrol_new *kc, struct snd_soc_tplg_ctl_hdr *tc) { - struct snd_ctl_tlv *tlv; - int size; + struct snd_soc_tplg_ctl_tlv *tplg_tlv; - if (tplg_tlv->count == 0) + if (!(tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE)) return 0; - size = ((tplg_tlv->count + (sizeof(unsigned int) - 1)) & - ~(sizeof(unsigned int) - 1)); - tlv = kzalloc(sizeof(*tlv) + size, GFP_KERNEL); - if (tlv == NULL) - return -ENOMEM; - - dev_dbg(tplg->dev, " created TLV type %d size %d bytes\n", - tplg_tlv->numid, size); + if (tc->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) + kc->tlv.c = snd_soc_bytes_tlv_callback; + else { + tplg_tlv = &tc->tlv; + switch (tplg_tlv->type) { + case SNDRV_CTL_TLVT_DB_SCALE: + return soc_tplg_create_tlv_db_scale(tplg, kc, + &tplg_tlv->scale); - tlv->numid = tplg_tlv->numid; - tlv->length = size; - memcpy(&tlv->tlv[0], tplg_tlv->data, size); - kc->tlv.p = (void *)tlv; + /* TODO: add support for other TLV types */ + default: + dev_dbg(tplg->dev, "Unsupported TLV type %d\n", + tplg_tlv->type); + return -EINVAL; + } + } return 0; } @@ -772,7 +796,7 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count, } /* create any TLV data */ - soc_tplg_create_tlv(tplg, &kc, &mc->tlv); + soc_tplg_create_tlv(tplg, &kc, &mc->hdr); /* register control here */ err = soc_tplg_add_kcontrol(tplg, &kc, -- 2.1.4

2 2

[alsa-devel] [PATCH] ALSA: hda - one Dell machine needs the headphone white noise fixup
by Hui Wang 05 Aug '15

05 Aug '15

The fixup ALC292_FIXUP_DISABLE_AAMIX can fix the white noise of the headphone on this Dell machine. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- 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 b20f104..dd7274d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5189,6 +5189,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.9.1

2 1

[alsa-devel] [PATCH] topology: Update ABI to improve TLV, operations and private data support.
by Liam Girdwood 05 Aug '15

05 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This is the alsa-lib patch - a corresponding kernel patch has also been sent. This patch updates the topology ABI version and adds support for private data types, specfic structures for each TLV data type and refactors the operations mapping structure to be generic (so that it can be used by other types). This code is only used within Intel atm and is not in an official alsa-lib release so impact should be minimal for the ABI update. Currently the TLV topology structure is targeted at only supporting the DB scale data. This patch extends support for the other TLV types so they can be easily added at a later stage. TLV structure is moved to common topology control header since it's a common field for controls and can be processed in a general way. Users must set a proper access flag for a control since it's used to decide if the TLV field is valid and if a TLV callback is needed. Removed the following fields from topology TLV struct: - size/count: type can decide the size. - numid: not needed to initialize TLV for kcontrol. - data: replaced by the type specific struct. Added TLV structure to generic control header and removed TLV structure from mixer control. Added generic ops to geneic control header. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/asoc.h | 27 ++++++++++++++++++--------- src/topology/ctl.c | 20 ++++++++------------ 2 files changed, 26 insertions(+), 21 deletions(-) diff --git a/include/sound/asoc.h b/include/sound/asoc.h index bb6dcf3..61b8473 100644 --- a/include/sound/asoc.h +++ b/include/sound/asoc.h @@ -135,11 +135,18 @@ struct snd_soc_tplg_private { /* * Kcontrol TLV data. */ +struct snd_soc_tplg_tlv_dbscale { + __le32 min; + __le32 step; + __le32 mute; +} __attribute__((packed)); + struct snd_soc_tplg_ctl_tlv { - __le32 size; /* in bytes aligned to 4 */ - __le32 numid; /* control element numeric identification */ - __le32 count; /* number of elem in data array */ - __le32 data[SND_SOC_TPLG_TLV_SIZE]; + __le32 size; /* in bytes of this structure */ + __le32 type; /* SNDRV_CTL_TLVT_*, type of TLV */ + union { + struct snd_soc_tplg_tlv_dbscale scale; + }; } __attribute__((packed)); /* @@ -153,9 +160,11 @@ struct snd_soc_tplg_channel { } __attribute__((packed)); /* - * Kcontrol Operations IDs + * Genericl Operations IDs, for binding Kcontrol or Bytes ext ops + * Kcontrol ops need get/put/info. + * Bytes ext ops need get/put. */ -struct snd_soc_tplg_kcontrol_ops_id { +struct snd_soc_tplg_io_ops { __le32 get; __le32 put; __le32 info; @@ -169,8 +178,8 @@ struct snd_soc_tplg_ctl_hdr { __le32 type; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; __le32 access; - struct snd_soc_tplg_kcontrol_ops_id ops; - __le32 tlv_size; /* non zero means control has TLV data */ + struct snd_soc_tplg_io_ops ops; + struct snd_soc_tplg_ctl_tlv tlv; } __attribute__((packed)); /* @@ -258,7 +267,6 @@ struct snd_soc_tplg_mixer_control { __le32 invert; __le32 num_channels; struct snd_soc_tplg_channel channel[SND_SOC_TPLG_MAX_CHAN]; - struct snd_soc_tplg_ctl_tlv tlv; struct snd_soc_tplg_private priv; } __attribute__((packed)); @@ -302,6 +310,7 @@ struct snd_soc_tplg_bytes_control { __le32 mask; __le32 base; __le32 num_regs; + struct snd_soc_tplg_io_ops ext_ops; struct snd_soc_tplg_private priv; } __attribute__((packed)); diff --git a/src/topology/ctl.c b/src/topology/ctl.c index aa06ff6..930b508 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -28,10 +28,7 @@ static int copy_tlv(struct tplg_elem *elem, struct tplg_elem *ref) tplg_dbg("TLV '%s' used by '%s\n", ref->id, elem->id); /* TLV has a fixed size */ - mixer_ctrl->tlv = *tlv; - - /* set size of TLV data */ - mixer_ctrl->hdr.tlv_size = tlv->count * sizeof(uint32_t); + mixer_ctrl->hdr.tlv = *tlv; return 0; } @@ -209,20 +206,19 @@ static int tplg_parse_tlv_dbscale(snd_config_t *cfg, struct tplg_elem *elem) snd_config_iterator_t i, next; snd_config_t *n; struct snd_soc_tplg_ctl_tlv *tplg_tlv; + struct snd_soc_tplg_tlv_dbscale *scale; const char *id = NULL, *value = NULL; - int *data; tplg_dbg(" scale: %s\n", elem->id); tplg_tlv = calloc(1, sizeof(*tplg_tlv)); if (!tplg_tlv) return -ENOMEM; - data = (int*)(tplg_tlv->data); elem->tlv = tplg_tlv; - tplg_tlv->numid = SNDRV_CTL_TLVT_DB_SCALE; - tplg_tlv->count = 8; - tplg_tlv->size = sizeof(*tplg_tlv); + tplg_tlv->size = sizeof(struct snd_soc_tplg_ctl_tlv); + tplg_tlv->type = SNDRV_CTL_TLVT_DB_SCALE; + scale = &tplg_tlv->scale; snd_config_for_each(i, next, cfg) { @@ -242,11 +238,11 @@ static int tplg_parse_tlv_dbscale(snd_config_t *cfg, struct tplg_elem *elem) /* get TLV data */ if (strcmp(id, "min") == 0) - data[0] = atoi(value); + scale->min = atoi(value); else if (strcmp(id, "step") == 0) - data[1] = atoi(value); + scale->step = atoi(value); else if (strcmp(id, "mute") == 0) - data[2] = atoi(value); + scale->mute = atoi(value); else SNDERR("error: unknown key %s\n", id); } -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: topology: Use kcalloc instead of kzalloc for array allocation
by Axel Lin 05 Aug '15

05 Aug '15

Also remove unnecessary memset. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/soc-topology.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4dadb5e..a5eb48e 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1250,7 +1250,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( struct snd_kcontrol_new *kc; int i, err; - kc = kzalloc(sizeof(*kc) * count, GFP_KERNEL); + kc = kcalloc(count, sizeof(*kc), GFP_KERNEL); if (!kc) return NULL; @@ -1273,7 +1273,6 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); - memset(kc, 0, sizeof(*kc)); kc[i].name = be->hdr.name; kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; -- 2.1.0

1 0

[alsa-devel] [PATCH] ASoC: Add function "rl6231_get_pre_div" to correct the dmic clock calculation
by Oder Chiou 05 Aug '15

05 Aug '15

Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rl6231.c | 47 +++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rl6231.h | 1 + sound/soc/codecs/rt5640.c | 7 ++++--- sound/soc/codecs/rt5645.c | 7 ++++--- sound/soc/codecs/rt5651.c | 7 ++++--- sound/soc/codecs/rt5670.c | 7 ++++--- sound/soc/codecs/rt5677.c | 5 ++++- 7 files changed, 68 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/rl6231.c b/sound/soc/codecs/rl6231.c index 96f3e90..57e51c1 100644 --- a/sound/soc/codecs/rl6231.c +++ b/sound/soc/codecs/rl6231.c @@ -11,10 +11,57 @@ */ #include <linux/module.h> +#include <linux/regmap.h> #include "rl6231.h" /** + * rl6231_get_pre_div - Return the value of pre divider. + * + * @map: map for setting. + * @reg: register. + * @sft: shift. + * + * Return the value of pre divider from given register value. + * Return negative error code for unexpected register value. + */ +int rl6231_get_pre_div(struct regmap *map, unsigned int reg, int sft) +{ + int pd, val; + + regmap_read(map, reg, &val); + + val = (val >> sft) & 0x7; + + switch (val) { + case 0: + case 1: + case 2: + case 3: + pd = val + 1; + break; + case 4: + pd = 6; + break; + case 5: + pd = 8; + break; + case 6: + pd = 12; + break; + case 7: + pd = 16; + break; + default: + pd = -EINVAL; + break; + } + + return pd; +} +EXPORT_SYMBOL_GPL(rl6231_get_pre_div); + +/** * rl6231_calc_dmic_clk - Calculate the parameter of dmic. * * @rate: base clock rate. diff --git a/sound/soc/codecs/rl6231.h b/sound/soc/codecs/rl6231.h index 0f7b057..4c77b44 100644 --- a/sound/soc/codecs/rl6231.h +++ b/sound/soc/codecs/rl6231.h @@ -30,5 +30,6 @@ int rl6231_calc_dmic_clk(int rate); int rl6231_pll_calc(const unsigned int freq_in, const unsigned int freq_out, struct rl6231_pll_code *pll_code); int rl6231_get_clk_info(int sclk, int rate); +int rl6231_get_pre_div(struct regmap *map, unsigned int reg, int sft); #endif /* __RL6231_H__ */ diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 119eca7..c557192 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -459,10 +459,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - int idx = -EINVAL; - - idx = rl6231_calc_dmic_clk(rt5640->sysclk); + int idx, rate; + rate = rt5640->sysclk / rl6231_get_pre_div(rt5640->regmap, + RT5640_ADDA_CLK1, RT5640_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9d54b34..4b01b63 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -543,10 +543,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); - int idx = -EINVAL; - - idx = rl6231_calc_dmic_clk(rt5645->sysclk); + int idx, rate; + rate = rt5645->sysclk / rl6231_get_pre_div(rt5645->regmap, + RT5645_ADDA_CLK1, RT5645_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 96295cc..8f7159b 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -378,10 +378,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); - int idx = -EINVAL; - - idx = rl6231_calc_dmic_clk(rt5651->sysclk); + int idx, rate; + rate = rt5651->sysclk / rl6231_get_pre_div(rt5651->regmap, + RT5651_ADDA_CLK1, RT5651_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index fd6ffd2..177748a 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -683,10 +683,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); - int idx = -EINVAL; - - idx = rl6231_calc_dmic_clk(rt5670->sysclk); + int idx, rate; + rate = rt5670->sysclk / rl6231_get_pre_div(rt5670->regmap, + RT5670_ADDA_CLK1, RT5670_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8d70911..3316700 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -916,8 +916,11 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct rt5677_priv *rt5677 = snd_soc_codec_get_drvdata(codec); - int idx = rl6231_calc_dmic_clk(rt5677->lrck[RT5677_AIF1] << 8); + int idx, rate; + rate = rt5677->sysclk / rl6231_get_pre_div(rt5677->regmap, + RT5677_CLK_TREE_CTRL1, RT5677_I2S_PD1_SFT); + idx = rl6231_calc_dmic_clk(rate); if (idx < 0) dev_err(codec->dev, "Failed to set DMIC clock\n"); else -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] Segmentation Fault in snd_pcm_rate_hw_free()
by Valentin Corfu 05 Aug '15

05 Aug '15

Hello ALSA developers, I observed one segmentation fault in snd_pcm_rate_hw_free() function, with the following BT: (gdb) up #1 0xb7554cc1 in raise (sig=6) at ../nptl/sysdeps/unix/sysv/linux/raise.c:64 64 return INLINE_SYSCALL (tgkill, 3, pid, selftid, sig); (gdb) #2 0xb75580ee in abort () at abort.c:92 92 raise (SIGABRT); (gdb) #3 0xb758a7dd in __libc_message (do_abort=2, fmt=0xb766053c "*** glibc detected *** %s: %s: 0x%s ***\n") at ../sysdeps/unix/sysv/linux/libc_fatal.c:189 189 abort (); (gdb) #4 0xb7594a71 in malloc_printerr (action=<value optimized out>, str=<value optimized out>, ptr=0x969ae98) at malloc.c:6283 6283 __libc_message (action & 2, (gdb) #5 0xb759636b in _int_free (av=<value optimized out>, p=0x969ae90) at malloc.c:4795 4795 malloc_printerr (check_action, errstr, chunk2mem(p)); (gdb) #6 0xb75994bd in __libc_free (mem=0x969ae98) at malloc.c:3738 3738 _int_free(ar_ptr, p); (gdb) #7 0xb76f3a81 in snd_pcm_rate_hw_free (pcm=0x9685d78) at pcm_rate.c:341 341 free(rate->pareas[0].addr); (gdb) #8 0xb76d045b in snd_pcm_hw_free (pcm=0x9685d78) at pcm.c:858 858 err = pcm->ops->hw_free(pcm->op_arg); (gdb) #9 0xb76f826e in snd_pcm_plug_hw_free (pcm=0x96856b0) at pcm_plug.c:1046 1046 int err = snd_pcm_hw_free(slave); (gdb) #10 0xb76d045b in snd_pcm_hw_free (pcm=0x96856b0) at pcm.c:858 858 err = pcm->ops->hw_free(pcm->op_arg); (gdb) #11 0x080492ad in main () Could you please give me some hints how to solve this issue? I can provide you more info or the test application, if needed. I can see the issue every time, and I also checked with latest version of alsa-lib but I got the same results. Thank you, Valentin Corfu

2 6

[alsa-devel] [PATCH] topology: Add support for writing manifest private data.
by Liam Girdwood 05 Aug '15

05 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Allow manifest to contain private data and write this data to file. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/builder.c | 37 ++++++++++++++++++++++++++++++++----- src/topology/tplg_local.h | 1 + 2 files changed, 33 insertions(+), 5 deletions(-) diff --git a/src/topology/builder.c b/src/topology/builder.c index 0066b22..a944866 100644 --- a/src/topology/builder.c +++ b/src/topology/builder.c @@ -226,9 +226,6 @@ static int write_block(snd_tplg_t *tplg, struct list_head *base, case OBJECT_TYPE_CC: return write_elem_block(tplg, base, size, SND_SOC_TPLG_TYPE_DAI_LINK, "cc"); - case OBJECT_TYPE_MANIFEST: - return write_data_block(tplg, size, SND_SOC_TPLG_TYPE_MANIFEST, - "manifest", &tplg->manifest); case OBJECT_TYPE_DATA: return write_elem_block(tplg, base, size, SND_SOC_TPLG_TYPE_PDATA, "data"); @@ -239,13 +236,43 @@ static int write_block(snd_tplg_t *tplg, struct list_head *base, return 0; } +/* write the manifest including its private data */ +static int write_manifest_data(snd_tplg_t *tplg) +{ + int ret; + + /* write the header for this block */ + ret = write_block_header(tplg, SND_SOC_TPLG_TYPE_MANIFEST, 0, + SND_SOC_TPLG_ABI_VERSION, 0, + sizeof(tplg->manifest) + tplg->manifest.priv.size, 1); + if (ret < 0) { + SNDERR("error: failed to write manifest block %d\n", ret); + return ret; + } + + verbose(tplg, "manifest : write %d bytes\n", sizeof(tplg->manifest)); + ret = write(tplg->out_fd, &tplg->manifest, sizeof(tplg->manifest)); + if (ret < 0) { + SNDERR("error: failed to write manifest %d\n", ret); + return ret; + } + + verbose(tplg, "manifest : write %d priv bytes\n", tplg->manifest.priv.size); + ret = write(tplg->out_fd, tplg->manifest_pdata, tplg->manifest.priv.size); + if (ret < 0) { + SNDERR("error: failed to write manifest priv data %d\n", ret); + return ret; + } + + return 0; +} + int tplg_write_data(snd_tplg_t *tplg) { int ret; /* write manifest */ - ret = write_data_block(tplg, sizeof(tplg->manifest), - OBJECT_TYPE_MANIFEST, "manifest", &tplg->manifest); + ret = write_manifest_data(tplg); if (ret < 0) { SNDERR("failed to write manifest %d\n", ret); return ret; diff --git a/src/topology/tplg_local.h b/src/topology/tplg_local.h index 62788e4..ad38945 100644 --- a/src/topology/tplg_local.h +++ b/src/topology/tplg_local.h @@ -77,6 +77,7 @@ struct snd_tplg { /* manifest */ struct snd_soc_tplg_manifest manifest; + const void *manifest_pdata; /* copied by builder at file write */ /* list of each element type */ struct list_head tlv_list; -- 2.1.4

2 1

[alsa-devel] [PATCH] topology: Add element ID so we can look up references by name.
by Liam Girdwood 05 Aug '15

05 Aug '15

From: Jin Yao <yao.jin(a)linux.intel.com> Add support to lookup elements by name. This is in preparation for adding some new API calls that will allow building topology data using a C API. This will allow applications to build thier own topology data directly. Signed-off-by: Jin Yao <yao.jin(a)linux.intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- src/topology/ctl.c | 9 ++++----- src/topology/dapm.c | 2 +- src/topology/data.c | 2 +- src/topology/elem.c | 15 +++++++++++---- src/topology/pcm.c | 10 +++++----- src/topology/text.c | 2 +- src/topology/tplg_local.h | 2 +- 7 files changed, 24 insertions(+), 18 deletions(-) diff --git a/src/topology/ctl.c b/src/topology/ctl.c index 9c1333c..aa06ff6 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -264,7 +264,7 @@ int tplg_parse_tlv(snd_tplg_t *tplg, snd_config_t *cfg, int err = 0; struct tplg_elem *elem; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_TLV); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_TLV); if (!elem) return -ENOMEM; @@ -298,7 +298,7 @@ int tplg_parse_control_bytes(snd_tplg_t *tplg, const char *id, *val = NULL; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_BYTES); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_BYTES); if (!elem) return -ENOMEM; @@ -403,11 +403,10 @@ int tplg_parse_control_enum(snd_tplg_t *tplg, snd_config_t *cfg, const char *id, *val = NULL; int err, j; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_ENUM); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_ENUM); if (!elem) return -ENOMEM; - /* init new mixer */ ec = elem->enum_ctrl; elem_copy_text(ec->hdr.name, elem->id, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); ec->hdr.type = SND_SOC_TPLG_TYPE_ENUM; @@ -501,7 +500,7 @@ int tplg_parse_control_mixer(snd_tplg_t *tplg, const char *id, *val = NULL; int err, j; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_MIXER); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_MIXER); if (!elem) return -ENOMEM; diff --git a/src/topology/dapm.c b/src/topology/dapm.c index 1da82ad..7e26ea0 100644 --- a/src/topology/dapm.c +++ b/src/topology/dapm.c @@ -420,7 +420,7 @@ int tplg_parse_dapm_widget(snd_tplg_t *tplg, const char *id, *val = NULL; int widget_type, err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_DAPM_WIDGET); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_DAPM_WIDGET); if (!elem) return -ENOMEM; diff --git a/src/topology/data.c b/src/topology/data.c index 13e1e2b..c768bc5 100644 --- a/src/topology/data.c +++ b/src/topology/data.c @@ -268,7 +268,7 @@ int tplg_parse_data(snd_tplg_t *tplg, snd_config_t *cfg, int err = 0; struct tplg_elem *elem; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_DATA); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_DATA); if (!elem) return -ENOMEM; diff --git a/src/topology/elem.c b/src/topology/elem.c index d7a1fd7..7fee653 100644 --- a/src/topology/elem.c +++ b/src/topology/elem.c @@ -103,20 +103,27 @@ struct tplg_elem *tplg_elem_lookup(struct list_head *base, const char* id, /* create a new common element and object */ struct tplg_elem* tplg_elem_new_common(snd_tplg_t *tplg, - snd_config_t *cfg, enum object_type type) + snd_config_t *cfg, const char *name, enum object_type type) { struct tplg_elem *elem; const char *id; int obj_size = 0; void *obj; + if (!cfg && !name) + return NULL; + elem = tplg_elem_new(); if (!elem) return NULL; - snd_config_get_id(cfg, &id); - strncpy(elem->id, id, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); - elem->id[SNDRV_CTL_ELEM_ID_NAME_MAXLEN - 1] = 0; + /* do we get name from cfg */ + if (cfg) { + snd_config_get_id(cfg, &id); + elem_copy_text(elem->id, id, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); + elem->id[SNDRV_CTL_ELEM_ID_NAME_MAXLEN - 1] = 0; + } else if (name != NULL) + elem_copy_text(elem->id, name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN); switch (type) { case OBJECT_TYPE_DATA: diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 8f23a6f..deae47b 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -228,7 +228,7 @@ int tplg_parse_pcm_config(snd_tplg_t *tplg, const char *id; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_STREAM_CONFIG); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_STREAM_CONFIG); if (!elem) return -ENOMEM; @@ -294,7 +294,7 @@ int tplg_parse_pcm_caps(snd_tplg_t *tplg, char *s; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_STREAM_CAPS); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_STREAM_CAPS); if (!elem) return -ENOMEM; @@ -461,7 +461,7 @@ int tplg_parse_pcm(snd_tplg_t *tplg, const char *id, *val = NULL; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_PCM); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_PCM); if (!elem) return -ENOMEM; @@ -524,7 +524,7 @@ int tplg_parse_be(snd_tplg_t *tplg, const char *id, *val = NULL; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_BE); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_BE); if (!elem) return -ENOMEM; @@ -587,7 +587,7 @@ int tplg_parse_cc(snd_tplg_t *tplg, const char *id, *val = NULL; int err; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_CC); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_CC); if (!elem) return -ENOMEM; diff --git a/src/topology/text.c b/src/topology/text.c index ebb6e38..7128056 100644 --- a/src/topology/text.c +++ b/src/topology/text.c @@ -64,7 +64,7 @@ int tplg_parse_text(snd_tplg_t *tplg, snd_config_t *cfg, int err = 0; struct tplg_elem *elem; - elem = tplg_elem_new_common(tplg, cfg, OBJECT_TYPE_TEXT); + elem = tplg_elem_new_common(tplg, cfg, NULL, OBJECT_TYPE_TEXT); if (!elem) return -ENOMEM; diff --git a/src/topology/tplg_local.h b/src/topology/tplg_local.h index 688c78f..62788e4 100644 --- a/src/topology/tplg_local.h +++ b/src/topology/tplg_local.h @@ -216,7 +216,7 @@ struct tplg_elem *tplg_elem_lookup(struct list_head *base, const char* id, unsigned int type); struct tplg_elem* tplg_elem_new_common(snd_tplg_t *tplg, - snd_config_t *cfg, enum object_type type); + snd_config_t *cfg, const char *name, enum object_type type); static inline void elem_copy_text(char *dest, const char *src, int len) { -- 2.1.4

2 1

[alsa-devel] [PATCH] topology: fix element object type is switch()
by Liam Girdwood 05 Aug '15

05 Aug '15

Use the correct type for this object. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/elem.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/topology/elem.c b/src/topology/elem.c index 32ba2c1..d7a1fd7 100644 --- a/src/topology/elem.c +++ b/src/topology/elem.c @@ -137,7 +137,7 @@ struct tplg_elem* tplg_elem_new_common(snd_tplg_t *tplg, list_add_tail(&elem->list, &tplg->enum_list); obj_size = sizeof(struct snd_soc_tplg_enum_control); break; - case SND_SOC_TPLG_TYPE_MIXER: + case OBJECT_TYPE_MIXER: list_add_tail(&elem->list, &tplg->mixer_list); obj_size = sizeof(struct snd_soc_tplg_mixer_control); break; -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix Dell laptop for internal mic/headset mic
by woodrow.shen＠canonical.com 05 Aug '15

05 Aug '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> The new Dell laptop with ALC3266 can't use internal microphone to record. Also, the 3 ring headset mic doesn't work on it. For src of patch_realtek.c, only the pin cfg of Nid 0x17 is different, so the Nid 0x17 is added into pin quirk, and others are defined to macro ALC298_STANDARD_PINS in common use. Codec: Realtek ALC3266 Vendor Id: 0x10ec0298 Subsystem Id: 0x102806e5 BugLink: https://bugs.launchpad.net/bugs/1481575 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b20f104..294ba67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5401,6 +5401,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x14, 0x90170110}, \ {0x15, 0x0221401f} +#define ALC298_STANDARD_PINS \ + {0x12, 0x90a60130}, \ + {0x21, 0x03211020} + static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, {0x14, 0x90170110}, @@ -5570,9 +5574,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { ALC292_STANDARD_PINS, {0x13, 0x90a60140}), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x90a60130}, - {0x17, 0x90170140}, - {0x21, 0x03211020}), + ALC298_STANDARD_PINS, + {0x17, 0x90170140}), + SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC298_STANDARD_PINS, + {0x17, 0x90170110}), {} }; -- 2.1.4

2 1

Re: [alsa-devel] When select 2.1 profile in pulseaudio, how do pulseaudio select notebook 's internal subwoofer or 5.1 external speaker ?
by Raymond Yau 05 Aug '15

05 Aug '15

> > > > Do pulseaudio 2.1 profile select the internal subwoofer or your 5.1 > > external speaker ? > > My external speaker (i.e not subwoofer but center). It always was so. > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… +/* default channel maps for 2.1 speakers; + * since HD-audio supports only stereo, odd number channels are omitted + */ +const struct snd_pcm_chmap_elem snd_pcm_2_1_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_LFE, SNDRV_CHMAP_LFE } }, + { } +}; The default 2.1 channel map for internal subwoofer does not has 5.1 channel map But your notebook have three jacks at ext front which support 5.1 In your case,there is no default 2.1 channel map when the current model has three speaker pins after remove the not exist internal surround speakers, [ALC882_FIXUP_ACER_ASPIRE_4930G] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x16, 0x99130111 }, /* CLFE speaker */ - { 0x17, 0x99130112 }, /* surround speaker */ - { } - }, you need to use the left channel version of 2.1 channel and 5.1 channel map +static const struct snd_pcm_chmap_elem acer_pcm_chmaps[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR } }, + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_LFE, SNDRV_CHMAP_NA } }, /* LFE only on left */ + { .channels = 4, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { .channels = 6, + .map = { SNDRV_CHMAP_FL, SNDRV_CHMAP_FR, + SNDRV_CHMAP_RL, SNDRV_CHMAP_RR, + SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { } +}; http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152e… <http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152e…> conf: Allow 2.1 surround to use different number of channels This way, cards that support LFE on four channels (e g laptop with internal subwoofer) can do that, and other cards on a six channel setup can use that as well. which channel map is used by alsa lib when you open surround21 pcm device since your notebook will have both channel maps ?

1 1

[alsa-devel] [RFC][PATCH 0/2] ALSA: fireworks/firewire-lib: add support for recent firmware quirk
by Takashi Sakamoto 05 Aug '15

05 Aug '15

Some users have reported current ALSA Fireworks driver stops packet streaming due to detecting discontinuity. [alsa-devel] ALSA snd-fireworks Echo AudioFire 2 http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080379.html [alsa-devel] AudioFire4 quirk http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095697.html As long as I investigate, this issue depends on firmware version installed in these devices. While, it's not clear that which version of firmwares for which model has this quirk. At least, firmware version 5.7.0, 5.7.3 and 5.8.0 for AudioFire2/AudioFire4/AudioFirePre8 have this quirk. This patchset applies a solution for this issue, with my assumption that 'the same firmware binary has the same quirk'. These models, AudioFire8 (since Jul 2009) and Gibson Robot Interface Pack (RIP) series uses the same firmware binary in driver package. Takashi Sakamoto (2): Revert "ALSA: fireworks: add support for AudioFire2 quirk" ALSA: fireworks/firewire-lib: add support for recent firmware quirk sound/firewire/amdtp.c | 5 +++-- sound/firewire/amdtp.h | 2 ++ sound/firewire/fireworks/fireworks.c | 10 ++++++++-- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_stream.c | 12 +++++++++--- 5 files changed, 23 insertions(+), 8 deletions(-) -- 2.1.4

2 5

Re: [alsa-devel] ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets()
by Dan Carpenter 04 Aug '15

04 Aug '15

Hello Lars-Peter Clausen, The patch 1ce43acff0c0: "ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets()" from Jul 26, 2015, leads to the following static checker warning: sound/soc/soc-dapm.c:1199 snd_soc_dapm_dai_get_connected_widgets() warn: inconsistent returns 'mutex:&card->dapm_mutex'. Locked on: line 1194 Unlocked on: line 1199 sound/soc/soc-dapm.c 1164 int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, 1165 struct snd_soc_dapm_widget_list **list) 1166 { 1167 struct snd_soc_card *card = dai->component->card; 1168 struct snd_soc_dapm_widget *w; 1169 LIST_HEAD(widgets); 1170 int paths; 1171 int ret; 1172 1173 mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); 1174 1175 /* 1176 * For is_connected_{output,input}_ep fully discover the graph we need 1177 * to reset the cached number of inputs and outputs. 1178 */ 1179 list_for_each_entry(w, &card->widgets, list) { 1180 w->inputs = -1; 1181 w->outputs = -1; 1182 } 1183 1184 if (stream == SNDRV_PCM_STREAM_PLAYBACK) 1185 paths = is_connected_output_ep(dai->playback_widget, &widgets); 1186 else 1187 paths = is_connected_input_ep(dai->capture_widget, &widgets); 1188 1189 /* Drop starting point */ 1190 list_del(widgets.next); 1191 1192 ret = dapm_widget_list_create(list, &widgets); 1193 if (ret) 1194 return ret; ^^^^^^^^^^^ Holding card->dapm_mutex. 1195 1196 trace_snd_soc_dapm_connected(paths, stream); 1197 mutex_unlock(&card->dapm_mutex); 1198 1199 return paths; 1200 } regards, dan carpenter

2 1

[alsa-devel] [TRIVIAL PATCH V2] ASoC: atmel_ssc_dai: Correct misuse of 0x%<decimal>
by Joe Perches 04 Aug '15

04 Aug '15

Correct misuse of 0x%d in logging message. Signed-off-by: Joe Perches <joe(a)perches.com> --- Resend with later, non-broken version of Evolution email client (3.16 -> 3.16.3) sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d059..ba8def5 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -290,7 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, int dir, dir_mask; int ret; - pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + pr_debug("atmel_ssc_startup: SSC_SR=0x%x\n", ssc_readl(ssc_p->ssc->regs, SR)); /* Enable PMC peripheral clock for this SSC */

2 1

[alsa-devel] [PATCH] ASoC: topology: fix typo in soc_tplg_kcontrol_bind_io()
by Liam Girdwood 04 Aug '15

04 Aug '15

From: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index df3ddf0..4dadb5e 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -534,7 +534,7 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, k->put = bops[i].put; if (k->get == NULL && bops[i].id == hdr->ops.get) k->get = bops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && bops[i].id == hdr->ops.info) k->info = bops[i].info; } -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: nau8825: Add driver for headset chip Nuvoton 8825
by Anatol Pomozov 04 Aug '15

04 Aug '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/nau8825.txt | 18 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 874 +++++++++++++++++++++ sound/soc/codecs/nau8825.h | 262 ++++++ 5 files changed, 1160 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 0000000..c3beb37 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,18 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-pullup: Pull-up JKDET pin. Boolean. + +Example: + + codec: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + nuvoton,jkdet-pullup = "true"; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7fcbc22..1a3639c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,6 +78,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -892,6 +893,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b46c8f..4874881 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -262,6 +263,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..db24465 --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,874 @@ +/* + * NAU8825 audio codec driver + * + * Copyright 2015 Nuvoton Technology Corp. + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/input.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "nau8825.h" + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + int irq; + struct snd_soc_jack *jack; + int button_pressed; + bool jkdet_pullup; +}; + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); + +static const struct snd_kcontrol_new nau8825_snd_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Sidetone Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Gain", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), +}; + +static int nau8825_charge_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, NAU8825_JAMNODCLOW); + } else { + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, 0); + } + + return 0; +} + +static int nau8825_output_driver_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* Power Up L and R Output Drivers */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3c, + 0x3c); + /* Power up Main Output Drivers (The main driver must be turned + on after the predriver to avoid pops) */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3, + 0x3); + } else { + /* Power Down L and R Output Drivers */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3f, + 0x0); + } + + return 0; +} + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, + &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, + &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Change Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_charge_pump_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("DACL Power", NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, + 0), + SND_SOC_DAPM_SUPPLY("DACR Power", NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, + 0), + + SND_SOC_DAPM_OUT_DRV_E("Output Driver", SND_SOC_NOPM, 0, 0, NULL, 0, + nau8825_output_driver_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"Capture", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DDACL", NULL, "DACL Power"}, + {"DDACR", NULL, "DACR Power"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "Change Pump"}, + {"HP amp R", NULL, "Change Pump"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver", NULL, "ADACL"}, + {"Output Driver", NULL, "ADACR"}, + {"HPOL", NULL, "Output Driver"}, + {"HPOR", NULL, "Output Driver"}, +}; + +static int nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int reg_val; + + reg_val = mute ? NAU8825_HP_MUTE : 0; + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HP_MUTE, reg_val); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .digital_mute = nau8825_mute, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + nau8825->jack = jack; + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons; + + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + lower 8 bits - for long pressed buttons */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct regmap *regmap) +{ + /* Setup Button Detect (Debounce, number of buttons, and Hysteresis) */ + regmap_write(regmap, NAU8825_REG_KEYDET_CTRL, 0x7311); + + /* Setup 4 buttons impedane according to Android specification + * https://source.android.com/accessories/headset-spec.html + * Button 0 - 0-70 Ohm + * Button 1 - 110-180 Ohm + * Button 2 - 210-290 Ohm + * Button 3 - 360-680 Ohm + */ + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, 0x0f1f); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, 0x325f); +} + +static void nau8825_init_regs(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + int jkdet_pullup_val; + + /* Power down DACs for power savings */ + regmap_write(regmap, NAU8825_REG_CHARGE_PUMP, 0x0300); + /* DAC Zero Crossing Enable */ + regmap_write(regmap, NAU8825_REG_MUTE_CTRL, 0x1000); + /* VMID Enable and Tieoff */ + regmap_write(regmap, NAU8825_REG_BIAS_ADJ, 0x0060); + /* Analog Bias Enable, Disable Boost Driver, + Automatic Short circuit protection enable */ + regmap_write(regmap, NAU8825_REG_BOOST, 0x3140); + /* Ground HP Outputs[1:0], + Enable Automatic Mic/Gnd switching reading on insert interrupt[6] */ + regmap_write(regmap, NAU8825_REG_HSD_CTRL, 0x004f); + + jkdet_pullup_val = nau8825->jkdet_pullup ? + NAU8825_JKDET_PULL_SELECT : 0; + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_SELECT, jkdet_pullup_val); + + regmap_write(regmap, NAU8825_REG_JACK_DET_CTRL, 0x01e0); + /* IRQ Output Enable, Mask Insert Interrupt */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_MASK, 0x0801); + + /* Setup SAR ADC */ + regmap_write(regmap, NAU8825_REG_SAR_CTRL, 0x0280); + nau8825_setup_buttons(regmap); + + /* Setup ADC x128 OSR */ + regmap_write(regmap, NAU8825_REG_ADC_RATE, 0x0002); + /* Setup DAC x128 OSR */ + regmap_write(regmap, NAU8825_REG_DAC_CTRL1, 0x0082); + /* DAC bias settings */ + regmap_write(regmap, NAU8825_REG_ANALOG_CONTROL_2, 0x1003); + /* ADC Bias Settings */ + regmap_write(regmap, NAU8825_REG_ANALOG_ADC_2, 0x0260); + /* Maximize Jack Insert Debounce */ + regmap_write(regmap, NAU8825_REG_JACK_DET_CTRL, 0x00e0); + + /* Setup clocks */ + regmap_write(regmap, NAU8825_REG_ENA_CTRL, 0x0416); + + /* Mask jack insert interruption */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0x0011); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->dapm = &codec->dapm; + + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "DDACR"); + snd_soc_dapm_sync(&codec->dapm); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + * Enables master mode and correctly divides BCLK and FSCLK from MCLK. + */ + regmap_write(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, 0x2019); + regmap_write(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, 0x0010); + + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + + .controls = nau8825_snd_controls, + .num_controls = ARRAY_SIZE(nau8825_snd_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825; + int ret, value; + + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->irq = i2c->irq; + nau8825->dev = dev; + + nau8825->jkdet_pullup = device_property_read_bool(dev, + "nuvoton,jkdet-pullup"); + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); + + nau8825_init_regs(nau8825); + + if (nau8825->irq) { + int ret = devm_request_threaded_irq(dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + } + + return 0; +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..4516d64 --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,262 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov(a)chrominium.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +#define NAU8825_SPKR_DWN1R_SHORT (1 << 1) +#define NAU8825_SPKR_DWN1L_SHORT (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) + +/* GPIO12_CTRL (0x1a) */ +/* 0 - pull down, 1 - pull up */ +#define NAU8825_JKDET_PULL_SELECT (1 << 11) + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SINC4 (1 << 4) +#define NAU8825_ADC_SINC_DOWN_MASK 3 +#define NAU8825_ADC_SINC_DOWN_32 0 +#define NAU8825_ADC_SINC_DOWN_64 1 +#define NAU8825_ADC_SINC_DOWN_128 2 +#define NAU8825_ADC_SINC_DOWN_256 3 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_MASK 7 +#define NAU8825_DAC_OVERSAMPLE_64 0 +#define NAU8825_DAC_OVERSAMPLE_256 1 +#define NAU8825_DAC_OVERSAMPLE_128 2 +#define NAU8825_DAC_OVERSAMPLE_32 4 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_MASK (3 << 4) +#define NAU8825_BIAS_VMID_SEL_OPEN (0 << 4) +#define NAU8825_BIAS_VMID_SEL_25KOHM (1 << 4) +#define NAU8825_BIAS_VMID_SEL_125KOHM (2 << 4) +#define NAU8825_BIAS_VMID_SEL_2_5KOHM (3 << 4) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_DIS (1 << 12) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_INTEGR_R (1 << 5) + +/* CHARGE_PUMP_AND_POWER_DOWN_CONTROL (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- 2.5.0

2 1

[alsa-devel] [PATCH v2] topology: Add command line topology tool to build topology binaries
by Liam Girdwood 04 Aug '15

04 Aug '15

Add a command line tool that will parse topology text files and convert to the binary topology data as used by the kernel. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Makefile.am | 3 ++ configure.ac | 5 ++- topology/Makefile.am | 10 +++++ topology/topology.c | 117 +++++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 134 insertions(+), 1 deletion(-) create mode 100644 topology/Makefile.am create mode 100644 topology/topology.c diff --git a/Makefile.am b/Makefile.am index 5bbe588..613f62d 100644 --- a/Makefile.am +++ b/Makefile.am @@ -25,6 +25,9 @@ endif if HAVE_UCM SUBDIRS += alsaucm endif +if HAVE_TOPOLOGY +SUBDIRS += topology +endif EXTRA_DIST= TODO gitcompile AUTOMAKE_OPTIONS=foreign diff --git a/configure.ac b/configure.ac index f09aa54..4c279a9 100644 --- a/configure.ac +++ b/configure.ac @@ -50,6 +50,8 @@ AC_CHECK_HEADERS([alsa/seq.h], [have_seq="yes"], [have_seq="no"], [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/use-case.h], [have_ucm="yes"], [have_ucm="no"], [#include <alsa/asoundlib.h>]) +AC_CHECK_HEADERS([alsa/topology.h], [have_topology="yes"], [have_topology="no"], + [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"], [#include <samplerate.h>]) @@ -58,6 +60,7 @@ AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") AM_CONDITIONAL(HAVE_SEQ, test "$have_seq" = "yes") AM_CONDITIONAL(HAVE_UCM, test "$have_ucm" = "yes") +AM_CONDITIONAL(HAVE_TOPOLOGY, test "$have_topology" = "yes") AM_CONDITIONAL(HAVE_SAMPLERATE, test "$have_samplerate" = "yes") dnl Check for librt @@ -358,7 +361,7 @@ AC_OUTPUT(Makefile alsactl/Makefile alsactl/init/Makefile \ m4/Makefile po/Makefile.in \ alsaconf/alsaconf alsaconf/Makefile \ alsaconf/po/Makefile \ - alsaucm/Makefile \ + alsaucm/Makefile topology/Makefile \ aplay/Makefile include/Makefile iecset/Makefile utils/Makefile \ utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile \ seq/aplaymidi/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ diff --git a/topology/Makefile.am b/topology/Makefile.am new file mode 100644 index 0000000..c370b14 --- /dev/null +++ b/topology/Makefile.am @@ -0,0 +1,10 @@ +bin_PROGRAMS = \ + alsatplg + +alsatplg_SOURCES = topology.c + +AM_CPPFLAGS = \ + -Wall -I$(top_srcdir)/include + +alsatplg_LDADD = -lasound + diff --git a/topology/topology.c b/topology/topology.c new file mode 100644 index 0000000..33c3276 --- /dev/null +++ b/topology/topology.c @@ -0,0 +1,117 @@ +/* + Copyright(c) 2014-2015 Intel Corporation + Copyright(c) 2010-2011 Texas Instruments Incorporated, + All rights reserved. + + This program is free software; you can redistribute it and/or modify + it under the terms of version 2 of the GNU General Public License as + published by the Free Software Foundation. + + This program is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU General Public License + along with this program; if not, write to the Free Software + Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + The full GNU General Public License is included in this distribution + in the file called LICENSE.GPL. +*/ + +#include <stdlib.h> +#include <stdio.h> +#include <stdint.h> +#include <fcntl.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> +#include <sys/stat.h> +#include <dlfcn.h> +#include <getopt.h> +#include <assert.h> + +#include <alsa/asoundlib.h> +#include <alsa/topology.h> +#include "gettext.h" + +static snd_output_t *log; + +static void usage(char *name) +{ + printf( +_("Usage: %s [OPTIONS]...\n" +"\n" +"-h, --help help\n" +"-c, --compile=FILE compile file\n" +"-v, --verbose=LEVEL set verbosity level (0...1)\n" +"-o, --output=FILE set output file\n" +), name); +} + +int main(int argc, char *argv[]) +{ + snd_tplg_t *snd_tplg; + static const char short_options[] = "hc:v:o:"; + static const struct option long_options[] = { + {"help", 0, 0, 'h'}, + {"verbose", 0, 0, 'v'}, + {"compile", 0, 0, 'c'}, + {"output", 0, 0, 'o'}, + {0, 0, 0, 0}, + }; + char *source_file = NULL, *output_file = NULL; + int c, err, verbose = 0, option_index; + +#ifdef ENABLE_NLS + setlocale(LC_ALL, ""); + textdomain(PACKAGE); +#endif + + err = snd_output_stdio_attach(&log, stderr, 0); + assert(err >= 0); + + while ((c = getopt_long(argc, argv, short_options, long_options, &option_index)) != -1) { + switch (c) { + case 'h': + usage(argv[0]); + return 0; + case 'v': + verbose = atoi(optarg); + break; + case 'c': + source_file = optarg; + break; + case 'o': + output_file = optarg; + break; + default: + fprintf(stderr, _("Try `%s --help' for more information.\n"), argv[0]); + return 1; + } + } + + if (source_file == NULL || output_file == NULL) { + usage(argv[0]); + return 1; + } + + snd_tplg = snd_tplg_new(); + if (snd_tplg == NULL) { + fprintf(stderr, _("failed to create new topology context\n")); + return 1; + } + + snd_tplg_verbose(snd_tplg, verbose); + + err = snd_tplg_build_file(snd_tplg, source_file, output_file); + if (err < 0) { + fprintf(stderr, _("failed to compile context %s\n"), source_file); + snd_tplg_free(snd_tplg); + return 1; + } + + snd_tplg_free(snd_tplg); + return 0; +} + -- 2.1.4

2 1

[alsa-devel] [PATCH] topology: Add command line topology tool to build topology binaries
by Liam Girdwood 04 Aug '15

04 Aug '15

Add a command line tool that will parse topology text files and convert to the binary topology data as used by the kernel. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Makefile.am | 3 ++ configure.ac | 5 ++- topology/Makefile.am | 13 ++++++ topology/topology.c | 117 +++++++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 137 insertions(+), 1 deletion(-) create mode 100644 topology/Makefile.am create mode 100644 topology/topology.c diff --git a/Makefile.am b/Makefile.am index 5bbe588..613f62d 100644 --- a/Makefile.am +++ b/Makefile.am @@ -25,6 +25,9 @@ endif if HAVE_UCM SUBDIRS += alsaucm endif +if HAVE_TOPOLOGY +SUBDIRS += topology +endif EXTRA_DIST= TODO gitcompile AUTOMAKE_OPTIONS=foreign diff --git a/configure.ac b/configure.ac index f09aa54..4c279a9 100644 --- a/configure.ac +++ b/configure.ac @@ -50,6 +50,8 @@ AC_CHECK_HEADERS([alsa/seq.h], [have_seq="yes"], [have_seq="no"], [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([alsa/use-case.h], [have_ucm="yes"], [have_ucm="no"], [#include <alsa/asoundlib.h>]) +AC_CHECK_HEADERS([alsa/topology.h], [have_topology="yes"], [have_topology="no"], + [#include <alsa/asoundlib.h>]) AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"], [#include <samplerate.h>]) @@ -58,6 +60,7 @@ AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") AM_CONDITIONAL(HAVE_SEQ, test "$have_seq" = "yes") AM_CONDITIONAL(HAVE_UCM, test "$have_ucm" = "yes") +AM_CONDITIONAL(HAVE_TOPOLOGY, test "$have_topology" = "yes") AM_CONDITIONAL(HAVE_SAMPLERATE, test "$have_samplerate" = "yes") dnl Check for librt @@ -358,7 +361,7 @@ AC_OUTPUT(Makefile alsactl/Makefile alsactl/init/Makefile \ m4/Makefile po/Makefile.in \ alsaconf/alsaconf alsaconf/Makefile \ alsaconf/po/Makefile \ - alsaucm/Makefile \ + alsaucm/Makefile topology/Makefile \ aplay/Makefile include/Makefile iecset/Makefile utils/Makefile \ utils/alsa-utils.spec seq/Makefile seq/aconnect/Makefile \ seq/aplaymidi/Makefile seq/aseqdump/Makefile seq/aseqnet/Makefile \ diff --git a/topology/Makefile.am b/topology/Makefile.am new file mode 100644 index 0000000..f59ce01 --- /dev/null +++ b/topology/Makefile.am @@ -0,0 +1,13 @@ +bin_PROGRAMS = \ + alsatplg + +alsatplg_SOURCES = topology.c + +AM_CPPFLAGS = \ + -Wall -I$(top_srcdir)/include + +alsatplg_LDADD = -lasound + +# local build +AM_CPPFLAGS += -I$(top_srcdir)/../alsa-lib/include +alsatplg_LDADD = -L$(top_srcdir)/../alsa-lib/src/.libs diff --git a/topology/topology.c b/topology/topology.c new file mode 100644 index 0000000..7049bb7 --- /dev/null +++ b/topology/topology.c @@ -0,0 +1,117 @@ +/* + Copyright(c) 2014-2015 Intel Corporation + Copyright(c) 2010-2011 Texas Instruments Incorporated, + All rights reserved. + + This program is free software; you can redistribute it and/or modify + it under the terms of version 2 of the GNU General Public License as + published by the Free Software Foundation. + + This program is distributed in the hope that it will be useful, but + WITHOUT ANY WARRANTY; without even the implied warranty of + MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + General Public License for more details. + + You should have received a copy of the GNU General Public License + along with this program; if not, write to the Free Software + Foundation, Inc., 51 Franklin St - Fifth Floor, Boston, MA 02110-1301 USA. + The full GNU General Public License is included in this distribution + in the file called LICENSE.GPL. +*/ + +#include <stdlib.h> +#include <stdio.h> +#include <stdint.h> +#include <fcntl.h> +#include <unistd.h> +#include <errno.h> +#include <string.h> +#include <sys/stat.h> +#include <dlfcn.h> +#include <getopt.h> +#include <alsa/asoundlib.h> +#include <assert.h> + +#include "topology.h" +#include "gettext.h" + +static snd_output_t *log; + +static void usage(char *name) +{ + printf( +_("Usage: %s [OPTIONS]...\n" +"\n" +"-h, --help help\n" +"-c, --compile=FILE compile file\n" +"-v, --verbose=LEVEL set verbosity level (0...1)\n" +"-o, --output=FILE set output file\n" +), name); +} + +int main(int argc, char *argv[]) +{ + snd_tplg_t *snd_tplg; + static const char short_options[] = "hc:v:o:"; + static const struct option long_options[] = { + {"help", 0, 0, 'h'}, + {"verbose", 0, 0, 'v'}, + {"compile", 0, 0, 'c'}, + {"output", 0, 0, 'o'}, + {0, 0, 0, 0}, + }; + char *source_file = NULL, *output_file = NULL; + int c, err, verbose = 0, option_index; + +#ifdef ENABLE_NLS + setlocale(LC_ALL, ""); + textdomain(PACKAGE); +#endif + + err = snd_output_stdio_attach(&log, stderr, 0); + assert(err >= 0); + + while ((c = getopt_long(argc, argv, short_options, long_options, &option_index)) != -1) { + switch (c) { + case 'h': + usage(argv[0]); + return 0; + case 'v': + verbose = atoi(optarg); + break; + case 'c': + source_file = optarg; + break; + case 'o': + output_file = optarg; + break; + default: + fprintf(stderr, _("Try `%s --help' for more information.\n"), argv[0]); + return 1; + } + } + + if (source_file == NULL || output_file == NULL) { + usage(argv[0]); + return 1; + } + + snd_tplg = snd_tplg_new(); + if (snd_tplg == NULL) { + fprintf(stderr, _("failed to create new topology context\n")); + return 1; + } + + snd_tplg_verbose(snd_tplg, verbose); + + err = snd_tplg_build_file(snd_tplg, source_file, output_file); + if (err < 0) { + fprintf(stderr, _("failed to compile context %s\n"), source_file); + snd_tplg_free(snd_tplg); + return 1; + } + + snd_tplg_free(snd_tplg); + return 0; +} + -- 2.1.4

2 1

[alsa-devel] [PATCH 0/3] ALSA: hdac: stream fixes
by Vinod Koul 04 Aug '15

04 Aug '15

Hi Takashi, I had few off fixes on hdac piled up so time to send them out. First one is a fix for stream assignment for decoupled mode used in SKL driver. Second one is static checker fix reported by Dan Carpenter Last is a potential issue which our team reported so a WA for that. The command DMAs do not finish instantaneously, so we should per spec wait till DMA finishes Jeeja KP (2): ALSA: HDA: Fix stream assignment for host in decoupled mode ALSA: HDA: Dont check return for snd_hdac_chip_readl Vinod Koul (1): ALSA: HDA: wait for RIRB, CORB DMA to finish sound/hda/ext/hdac_ext_controller.c | 6 ------ sound/hda/ext/hdac_ext_stream.c | 2 +- sound/hda/hdac_controller.c | 14 ++++++++++++++ 3 files changed, 15 insertions(+), 7 deletions(-) -- 1.9.1

2 7

[alsa-devel] Question about Component and Card
by Kuninori Morimoto 04 Aug '15

04 Aug '15

Hi Lars Cc Mark I noticed my sound card (which is not upstreamed) can't probe in next branch. I think it is because this patch 1b7c12316982f74a5b8e7704c24cf5524d0723a3 (ASoC: Prevent components from being bound to multiple cards) In my case, my sound device (= CPU) has some DAIs, and these are connected to some codecs. It is shown as below. And I would like to use these as 2 sound cards. ex) aplay -D hw:0 xxx aplay -D hw:1 xxx But, 2nd sound card can't probe because of above patch now. (component->card indicates 1st card) It can be probeed if I used DPCM card (= 1 CPU + 2 Codec), but... But, is my understanding wrong ? codec0: codec@0 { xxx }; codec1: codec@1 { xxx }; sound: sound@0xxxxxx { #sound-dai-cells = <1>; rcar_sound,dai { dai0 { xxxx }; dai1 { xxxx }; }; }; card@0 { compatible = "simple-audio-card"; simple-audio-card,cpu { sound-dai = <&sound 0>; }; simple-audio-card,codec { sound-dai = <&codec0>; }; }; card@1 { compatible = "simple-audio-card"; simple-audio-card,cpu { sound-dai = <&sound 1>; }; simple-audio-card,codec { sound-dai = <&codec1>; }; };

3 5

[alsa-devel] [PATCH v14 0/3] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 03 Aug '15

03 Aug '15

This patch series asks for Russell King's patch http://article.gmane.org/gmane.linux.ports.arm.kernel/411574 "drm/edid: add function to help find SADs" to be applied. v14: (Mark Brown) - protect the CODEC from edid change - change some structure names - remove the static pointer to the HDMI driver functions - get the ELD from a pointer in the device private area - prevent audio from streaming many sources at the same time - build the CODEC DAI's from two different samples - use EXPORT_SYMBOL_GPL for exported functions - simplify exchanges between audio and video by using common data in the device private area v13: - rebase on 4.2.0-rc3 - remove Russell's patches v12: - use Russell King's DRM ELD helper (Mark Brown) v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (3): drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 163 +++++++++++++++++++-- include/sound/tda998x.h | 18 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 141 ++++++++++++++++++ 6 files changed, 370 insertions(+), 11 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.4.6

3 6

[alsa-devel] [TRIVIAL PATCH] ASoC: atmel_ssc_dai: Correct misuse of 0x%<decimal>
by Joe Perches 03 Aug '15

03 Aug '15

Correct misuse of 0x%d in logging message. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/soc/atmel/atmel_ssc_dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 841d059..ba8def5 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -290,7 +290,7 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, int dir, dir_mask; int ret; - pr_debug("atmel_ssc_startup: SSC_SR=0x%u\n", + pr_debug("atmel_ssc_startup: SSC_SR=0x%x\n", ssc_readl(ssc_p->ssc->regs, SR)); /* Enable PMC peripheral clock for this SSC */

2 2

[alsa-devel] [PATCH] ASoC: rt5645: Fix lost pin setting for DMIC1
by Bard Liao 03 Aug '15

03 Aug '15

I2S2_DAC pin can be used for I2S or GPIO. We should set it as GPIO if we use GPIO5 as DMIC1 data pin. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 2 ++ sound/soc/codecs/rt5645.h | 4 ++++ 2 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9d54b34..c34c498 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3356,6 +3356,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, break; case RT5645_DMIC_DATA_GPIO5: + regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, + RT5645_I2S2_DAC_PIN_MASK, RT5645_I2S2_DAC_PIN_GPIO); regmap_update_bits(rt5645->regmap, RT5645_DMIC_CTRL1, RT5645_DMIC_1_DP_MASK, RT5645_DMIC_1_DP_GPIO5); regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 199b22f..dffdda2 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -1693,6 +1693,10 @@ #define RT5645_GP6_PIN_SFT 6 #define RT5645_GP6_PIN_GPIO6 (0x0 << 6) #define RT5645_GP6_PIN_DMIC2_SDA (0x1 << 6) +#define RT5645_I2S2_DAC_PIN_MASK (0x1 << 4) +#define RT5645_I2S2_DAC_PIN_SFT 4 +#define RT5645_I2S2_DAC_PIN_I2S (0x0 << 4) +#define RT5645_I2S2_DAC_PIN_GPIO (0x1 << 4) #define RT5645_GP8_PIN_MASK (0x1 << 3) #define RT5645_GP8_PIN_SFT 3 #define RT5645_GP8_PIN_GPIO8 (0x0 << 3) -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] [beagleboard] Audio cap & Linux version 4.1.3-ti-r6.2
by Micka 03 Aug '15

03 Aug '15

Ok, some news : I find out where it's blocked : I've compiled the alsa-lib, and I've put a lot of printf. In the file pcm.c, in the function : snd_pcm_wait_nocheck This line : err_poll = poll(pfd, npfds, timeout); is called, but I never go out .... my trace show : struct pollfd *pfd: 0xbed64780, npfds: 1, timeout: -1 I don't know if it's ok ? Micka, Le lun. 3 août 2015 à 09:41, Micka <mickamusset(a)gmail.com> a écrit : > From what I see, it looks like that the alsa lib is calling multiple time > SNDRV_PCM_IOCTL_SYNC_PTR > > And it's not calling SNDRV_PCM_IOCTL_WRITEN_FRAMES .... > > Robert, I'm having issue to find out the version of alsa lib, do you have > any idea ? > > Maybe I'm going to compile it > > Le lun. 3 août 2015 à 08:58, Micka <mickamusset(a)gmail.com> a écrit : > >> Hi, >> >> Thx to this site : >> http://www.programdevelop.com/4763397/ >> >> I understand better the flow of ALSA :) >> >> this is my new log : >> >> [ 84.688930] snd_soc_dai_digital_mute 0 0 >> [ 84.689000] snd_pcm_playback_ioctl >> [ 84.689032] snd_pcm_playback_ioctl1 3229892899 >> [ 84.689057] call snd_pcm_common_ioctl1 >> [ 84.689084] snd_pcm_common_ioctl1 3229892899 >> [ 84.689109] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.689624] snd_pcm_playback_ioctl >> [ 84.689664] snd_pcm_playback_ioctl1 3228057875 >> [ 84.689689] call snd_pcm_common_ioctl1 >> [ 84.689715] snd_pcm_common_ioctl1 3228057875 >> [ 84.695219] snd_pcm_playback_ioctl >> [ 84.695269] snd_pcm_playback_ioctl1 3229892899 >> [ 84.695296] call snd_pcm_common_ioctl1 >> [ 84.695322] snd_pcm_common_ioctl1 3229892899 >> [ 84.695347] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.695400] snd_pcm_playback_ioctl >> [ 84.695427] snd_pcm_playback_ioctl1 3229892899 >> [ 84.695451] call snd_pcm_common_ioctl1 >> [ 84.695476] snd_pcm_common_ioctl1 3229892899 >> [ 84.695499] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.696332] snd_pcm_playback_ioctl >> [ 84.696364] snd_pcm_playback_ioctl1 3229892899 >> [ 84.696389] call snd_pcm_common_ioctl1 >> [ 84.696414] snd_pcm_common_ioctl1 3229892899 >> [ 84.696438] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.696474] snd_pcm_playback_ioctl >> [ 84.696501] snd_pcm_playback_ioctl1 3229892899 >> [ 84.696524] call snd_pcm_common_ioctl1 >> [ 84.696548] snd_pcm_common_ioctl1 3229892899 >> [ 84.696572] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.729249] snd_pcm_playback_ioctl >> [ 84.729289] snd_pcm_playback_ioctl1 3229892899 >> [ 84.729303] call snd_pcm_common_ioctl1 >> [ 84.729317] snd_pcm_common_ioctl1 3229892899 >> [ 84.729329] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.729355] snd_pcm_playback_ioctl >> [ 84.729368] snd_pcm_playback_ioctl1 3229892899 >> [ 84.729378] call snd_pcm_common_ioctl1 >> [ 84.729390] snd_pcm_common_ioctl1 3229892899 >> [ 84.729401] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.729787] snd_pcm_playback_ioctl >> [ 84.729801] snd_pcm_playback_ioctl1 3229892899 >> [ 84.729812] call snd_pcm_common_ioctl1 >> [ 84.729824] snd_pcm_common_ioctl1 3229892899 >> [ 84.729835] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.729852] snd_pcm_playback_ioctl >> [ 84.729865] snd_pcm_playback_ioctl1 3229892899 >> [ 84.729875] call snd_pcm_common_ioctl1 >> [ 84.729887] snd_pcm_common_ioctl1 3229892899 >> [ 84.729898] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.731902] snd_pcm_playback_ioctl >> [ 84.731914] snd_pcm_playback_ioctl1 3229892899 >> [ 84.731925] call snd_pcm_common_ioctl1 >> [ 84.731937] snd_pcm_common_ioctl1 3229892899 >> [ 84.731947] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.731962] snd_pcm_playback_ioctl >> [ 84.731974] snd_pcm_playback_ioctl1 3229892899 >> [ 84.731984] call snd_pcm_common_ioctl1 >> [ 84.731996] snd_pcm_common_ioctl1 3229892899 >> [ 84.732007] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.732314] snd_pcm_playback_ioctl >> [ 84.732328] snd_pcm_playback_ioctl1 3229892899 >> [ 84.732338] call snd_pcm_common_ioctl1 >> [ 84.732350] snd_pcm_common_ioctl1 3229892899 >> [ 84.732361] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.732375] snd_pcm_playback_ioctl >> [ 84.732387] snd_pcm_playback_ioctl1 3229892899 >> [ 84.732398] call snd_pcm_common_ioctl1 >> [ 84.732410] snd_pcm_common_ioctl1 3229892899 >> [ 84.732421] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734421] snd_pcm_playback_ioctl >> [ 84.734434] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734444] call snd_pcm_common_ioctl1 >> [ 84.734456] snd_pcm_common_ioctl1 3229892899 >> [ 84.734467] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734481] snd_pcm_playback_ioctl >> [ 84.734493] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734503] call snd_pcm_common_ioctl1 >> [ 84.734515] snd_pcm_common_ioctl1 3229892899 >> [ 84.734526] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734826] snd_pcm_playback_ioctl >> [ 84.734839] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734850] call snd_pcm_common_ioctl1 >> [ 84.734862] snd_pcm_common_ioctl1 3229892899 >> [ 84.734872] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734887] snd_pcm_playback_ioctl >> [ 84.734899] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734909] call snd_pcm_common_ioctl1 >> [ 84.734921] snd_pcm_common_ioctl1 3229892899 >> [ 84.734932] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.734974] snd_pcm_playback_ioctl >> [ 84.734987] snd_pcm_playback_ioctl1 3229892899 >> [ 84.734998] call snd_pcm_common_ioctl1 >> [ 84.735010] snd_pcm_common_ioctl1 3229892899 >> [ 84.735020] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.735034] snd_pcm_playback_ioctl >> [ 84.735046] snd_pcm_playback_ioctl1 16706 >> [ 84.735057] call snd_pcm_common_ioctl1 >> [ 84.735068] snd_pcm_common_ioctl1 16706 >> [ 84.735080] --SNDRV_PCM_IOCTL_START: >> [ 84.735096] snd_pcm_do_start >> [ 84.735108] soc_pcm_trigger >> [ 84.737206] snd_pcm_playback_ioctl >> [ 84.737221] snd_pcm_playback_ioctl1 3229892899 >> [ 84.737232] call snd_pcm_common_ioctl1 >> [ 84.737244] snd_pcm_common_ioctl1 3229892899 >> [ 84.737254] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.737280] snd_pcm_playback_ioctl >> [ 84.737293] snd_pcm_playback_ioctl1 3229892899 >> [ 84.737303] call snd_pcm_common_ioctl1 >> [ 84.737315] snd_pcm_common_ioctl1 3229892899 >> [ 84.737326] SNDRV_PCM_IOCTL_SYNC_PTR: >> [ 84.747620] snd_pcm_playback_ioctl >> [ 84.747647] snd_pcm_playback_ioctl1 3229892899 >> [ 84.747660] call snd_pcm_common_ioctl1 >> [ 84.747673] snd_pcm_common_ioctl1 3229892899 >> [ 84.747685] SNDRV_PCM_IOCTL_SYNC_PTR: >> >> Le ven. 31 juil. 2015 à 14:27, <g4(a)novadsp.com> a écrit : >> >>> Hi Micka >>> >>> > looks like that my config is good ..... But I can't figure out, where >>> i'm blocked .... >>> >>> I am having the same problem with 4.1.0 >>> >>> > Do you know who call the function soc_pcm_prepare ? >>> >>> http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L698 >>> >>> I will try and repeat the same exercise but it will be early next week. >>> >>> >>> -- >>> For more options, visit http://beagleboard.org/discuss >>> --- >>> You received this message because you are subscribed to the Google >>> Groups "BeagleBoard" group. >>> To unsubscribe from this group and stop receiving emails from it, send >>> an email to beagleboard+unsubscribe(a)googlegroups.com. >>> For more options, visit https://groups.google.com/d/optout. >>> >>

1 0

[alsa-devel] Alsa issue faced in port audio
by Srinivasan S 03 Aug '15

03 Aug '15

Dear ALSA Community, Am running port-audio(pjproject-2.4 & 2.3) on linux kernel version 3.14 & alsa-lib verison 1.0.27, but some how portaudio ALSA handles under/overrun in blocking read/write: it is never reported back to the application: handleXrun tries to recover, usually this works and leaves the stream in STATE_PREPARED. The xrun flag is set but neither the portaudio ALSA read nor the write methods look at this when they are in the read/write while loop. In case of playback this causes bad "framesAvail" values. The read funtion enters an endless loop. In port audio in writeframes for each time for read samples playback/capture poll will happen in this poll(snd_poll_descriptor_revents) we will get read/write event from device in the success case, when Xrun occurs the poll always returns 0 or NULL, which indicates A value of 0 indicates that the call timed out and no file descriptors were ready.But we traced the source code file descriptor is available & not corrupted Could you please let us know how this can be fixed in ALSA Kindly do the needful as early as possible Srinivasan S

1 1

[alsa-devel] EAPD enable for HP ALC221
by Kailang 03 Aug '15

03 Aug '15

Hi Takashi, HP ALC221 platform needs to enable EAPD for speaker output. Attach patch was enable the EAPD for speaker. Thanks. BR, Kailang

2 6

[alsa-devel] [PATCH] ALSA: hda - remove no physical connection pins from pin_quirk table
by Hui Wang 03 Aug '15

03 Aug '15

To cleanup the pin_quirk table: - rewrite the pin_config_match(), comparing all pins on the machine with the corresponding pins in the quirk table. - remove all 0x4xxxxxxx pin configurations from pin_quirk table - after removing the 0x4xxxxxxx pin configurations, some pin tables are exactly same, so removing the redudant pin tables. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/hda_auto_parser.c | 27 +++- sound/pci/hda/patch_realtek.c | 299 ++++------------------------------------ 2 files changed, 48 insertions(+), 278 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 03b7399..7f57a14 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -887,11 +887,32 @@ EXPORT_SYMBOL_GPL(snd_hda_apply_fixup); static bool pin_config_match(struct hda_codec *codec, const struct hda_pintbl *pins) { - for (; pins->nid; pins++) { - u32 def_conf = snd_hda_codec_get_pincfg(codec, pins->nid); - if (pins->val != def_conf) + int i; + + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + hda_nid_t nid = pin->nid; + u32 cfg = pin->cfg; + const struct hda_pintbl *t_pins; + int found; + + t_pins = pins; + found = 0; + for (; t_pins->nid; t_pins++) { + if (t_pins->nid == nid) { + found = 1; + if (t_pins->val == cfg) + break; + else if ((cfg & 0xf0000000) == 0x40000000 && (t_pins->val & 0xf0000000) == 0x40000000) + break; + else + return false; + } + } + if (!found && (cfg & 0xf0000000) != 0x40000000) return false; } + return true; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 18ae17e..d9a8b9f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5386,399 +5386,192 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {} }; -#define ALC255_STANDARD_PINS \ - {0x18, 0x411111f0}, \ - {0x19, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ - {0x1e, 0x411111f0} - #define ALC256_STANDARD_PINS \ {0x12, 0x90a60140}, \ {0x14, 0x90170110}, \ - {0x19, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ {0x21, 0x02211020} #define ALC282_STANDARD_PINS \ - {0x14, 0x90170110}, \ - {0x18, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ - {0x1e, 0x411111f0} - -#define ALC288_STANDARD_PINS \ - {0x17, 0x411111f0}, \ - {0x18, 0x411111f0}, \ - {0x19, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1e, 0x411111f0} + {0x14, 0x90170110} #define ALC290_STANDARD_PINS \ - {0x12, 0x99a30130}, \ - {0x13, 0x40000000}, \ - {0x16, 0x411111f0}, \ - {0x17, 0x411111f0}, \ - {0x19, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ - {0x1e, 0x411111f0} + {0x12, 0x99a30130} #define ALC292_STANDARD_PINS \ {0x14, 0x90170110}, \ - {0x15, 0x0221401f}, \ - {0x1a, 0x411111f0}, \ - {0x1b, 0x411111f0}, \ - {0x1d, 0x40700001} - -#define ALC298_STANDARD_PINS \ - {0x18, 0x411111f0}, \ - {0x19, 0x411111f0}, \ - {0x1a, 0x411111f0}, \ - {0x1e, 0x411111f0}, \ - {0x1f, 0x411111f0} + {0x15, 0x0221401f} static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, - {0x12, 0x40300000}, {0x14, 0x90170110}, - {0x17, 0x411111f0}, - {0x1d, 0x40538029}, {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60140}, {0x14, 0x90170110}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60160}, {0x14, 0x90170120}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x40000000}, {0x14, 0x90170130}, - {0x17, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, {0x1b, 0x01014020}, - {0x1d, 0x4054c029}, - {0x1e, 0x411111f0}, {0x21, 0x0221103f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x40000000}, {0x14, 0x90170150}, - {0x17, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, {0x1b, 0x02011020}, - {0x1d, 0x4054c029}, - {0x1e, 0x411111f0}, {0x21, 0x0221105f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - {0x12, 0x40000000}, {0x14, 0x90170110}, - {0x17, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, {0x1b, 0x01014020}, - {0x1d, 0x4054c029}, - {0x1e, 0x411111f0}, {0x21, 0x0221101f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, {0x17, 0x90170140}, - {0x18, 0x40000000}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41163b05}, - {0x1e, 0x411111f0}, {0x21, 0x0321102f}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60160}, {0x14, 0x90170130}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211040}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60160}, {0x14, 0x90170140}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211050}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60170}, {0x14, 0x90170120}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60170}, {0x14, 0x90170130}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211040}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60170}, {0x14, 0x90170140}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211050}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell Inspiron 5548", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60180}, {0x14, 0x90170130}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211040}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC255_STANDARD_PINS, {0x12, 0x90a60160}, {0x14, 0x90170120}, - {0x17, 0x40000000}, - {0x1d, 0x40700001}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC256_STANDARD_PINS, - {0x13, 0x40000000}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC256_STANDARD_PINS, - {0x13, 0x411111f0}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}), - SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC256_STANDARD_PINS, - {0x13, 0x411111f0}, - {0x1d, 0x4077992d}, - {0x1e, 0x411111ff}), + ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, - {0x13, 0x40000000}, {0x14, 0x90170110}, {0x15, 0x0421101f}, - {0x16, 0x411111f0}, - {0x17, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x04a11020}, - {0x1b, 0x411111f0}, - {0x1d, 0x40748605}, - {0x1e, 0x411111f0}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED, {0x12, 0x90a60140}, - {0x13, 0x40000000}, {0x14, 0x90170110}, {0x15, 0x0421101f}, - {0x16, 0x411111f0}, - {0x17, 0x411111f0}, {0x18, 0x02811030}, - {0x19, 0x411111f0}, {0x1a, 0x04a1103f}, - {0x1b, 0x02011020}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}), + {0x1b, 0x02011020}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP 15 Touchsmart", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC282_STANDARD_PINS, {0x12, 0x99a30130}, - {0x17, 0x40000000}, {0x19, 0x03a11020}, - {0x1d, 0x40f41905}, {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC282_STANDARD_PINS, {0x12, 0x99a30130}, - {0x17, 0x40020008}, {0x19, 0x03a11020}, - {0x1d, 0x40e00001}, {0x21, 0x03211040}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC282_STANDARD_PINS, {0x12, 0x99a30130}, - {0x17, 0x40000000}, {0x19, 0x03a11030}, - {0x1d, 0x40e00001}, {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC282_STANDARD_PINS, {0x12, 0x99a30130}, - {0x17, 0x40000000}, - {0x19, 0x03a11030}, - {0x1d, 0x40f00001}, - {0x21, 0x03211020}), - SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, - ALC282_STANDARD_PINS, - {0x12, 0x99a30130}, - {0x17, 0x40000000}, {0x19, 0x04a11020}, - {0x1d, 0x40f00001}, {0x21, 0x0421101f}), - SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, - ALC282_STANDARD_PINS, - {0x12, 0x99a30130}, - {0x17, 0x40000000}, - {0x19, 0x03a11030}, - {0x1d, 0x40f00001}, - {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0282, 0x103c, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED, ALC282_STANDARD_PINS, {0x12, 0x90a60140}, - {0x17, 0x40000000}, {0x19, 0x04a11030}, - {0x1d, 0x40f00001}, {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC282_STANDARD_PINS, {0x12, 0x90a60130}, - {0x17, 0x40020008}, - {0x19, 0x411111f0}, - {0x1d, 0x40e00001}, {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60160}, {0x14, 0x90170120}, - {0x17, 0x40000000}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x40700001}, - {0x1e, 0x411111f0}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0283, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC282_STANDARD_PINS, {0x12, 0x90a60130}, - {0x17, 0x40020008}, {0x19, 0x03a11020}, - {0x1d, 0x40e00001}, {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0288, 0x1028, "Dell", ALC288_FIXUP_DELL_XPS_13_GPIO6, - ALC288_STANDARD_PINS, {0x12, 0x90a60120}, - {0x13, 0x40000000}, {0x14, 0x90170110}, - {0x1d, 0x4076832d}, {0x21, 0x0321101f}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, - {0x14, 0x411111f0}, {0x15, 0x04211040}, {0x18, 0x90170112}, - {0x1a, 0x04a11020}, - {0x1d, 0x4075812d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, - {0x14, 0x411111f0}, {0x15, 0x04211040}, {0x18, 0x90170110}, - {0x1a, 0x04a11020}, - {0x1d, 0x4075812d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, - {0x14, 0x411111f0}, {0x15, 0x0421101f}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11020}, - {0x1d, 0x4075812d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, - {0x14, 0x411111f0}, {0x15, 0x04211020}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11040}, - {0x1d, 0x4076a12d}), + {0x1a, 0x04a11040}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x14, 0x90170110}, {0x15, 0x04211020}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11040}, - {0x1d, 0x4076a12d}), + {0x1a, 0x04a11040}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x14, 0x90170110}, {0x15, 0x04211020}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11020}, - {0x1d, 0x4076a12d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0290, 0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1, ALC290_STANDARD_PINS, {0x14, 0x90170110}, {0x15, 0x0421101f}, - {0x18, 0x411111f0}, - {0x1a, 0x04a11020}, - {0x1d, 0x4075812d}), + {0x1a, 0x04a11020}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, - {0x13, 0x411111f0}, {0x16, 0x01014020}, - {0x18, 0x411111f0}, - {0x19, 0x01a19030}, - {0x1e, 0x411111f0}), + {0x19, 0x01a19030}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, {0x12, 0x90a60140}, - {0x13, 0x411111f0}, {0x16, 0x01014020}, {0x18, 0x02a19031}, - {0x19, 0x01a1903e}, - {0x1e, 0x411111f0}), + {0x19, 0x01a1903e}), SND_HDA_PIN_QUIRK(0x10ec0292, 0x1028, "Dell", ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, - {0x12, 0x90a60140}, - {0x13, 0x411111f0}, - {0x16, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1e, 0x411111f0}), + {0x12, 0x90a60140}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, - {0x12, 0x40000000}, {0x13, 0x90a60140}, {0x16, 0x21014020}, - {0x18, 0x411111f0}, - {0x19, 0x21a19030}, - {0x1e, 0x411111f0}), + {0x19, 0x21a19030}), SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_STANDARD_PINS, - {0x12, 0x40000000}, - {0x13, 0x90a60140}, - {0x16, 0x411111f0}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1e, 0x411111f0}), - SND_HDA_PIN_QUIRK(0x10ec0293, 0x1028, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC292_STANDARD_PINS, - {0x12, 0x40000000}, - {0x13, 0x90a60140}, - {0x16, 0x21014020}, - {0x18, 0x411111f0}, - {0x19, 0x21a19030}, - {0x1e, 0x411111ff}), + {0x13, 0x90a60140}), SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, - ALC298_STANDARD_PINS, {0x12, 0x90a60130}, - {0x13, 0x40000000}, - {0x14, 0x411111f0}, {0x17, 0x90170140}, - {0x1d, 0x4068a36d}, {0x21, 0x03211020}), {} }; @@ -6672,77 +6465,33 @@ static const struct hda_model_fixup alc662_fixup_models[] = { static const struct snd_hda_pin_quirk alc662_pin_fixup_tbl[] = { SND_HDA_PIN_QUIRK(0x10ec0662, 0x1028, "Dell", ALC662_FIXUP_DELL_MIC_NO_PRESENCE, - {0x12, 0x4004c000}, {0x14, 0x01014010}, - {0x15, 0x411111f0}, - {0x16, 0x411111f0}, {0x18, 0x01a19020}, - {0x19, 0x411111f0}, {0x1a, 0x0181302f}, - {0x1b, 0x0221401f}, - {0x1c, 0x411111f0}, - {0x1d, 0x4054c601}, - {0x1e, 0x411111f0}), + {0x1b, 0x0221401f}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, {0x12, 0x99a30130}, {0x14, 0x90170110}, {0x15, 0x0321101f}, - {0x16, 0x03011020}, - {0x18, 0x40000008}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41000001}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x16, 0x03011020}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, {0x12, 0x99a30140}, {0x14, 0x90170110}, {0x15, 0x0321101f}, - {0x16, 0x03011020}, - {0x18, 0x40000008}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41000001}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x16, 0x03011020}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, {0x12, 0x99a30150}, {0x14, 0x90170110}, {0x15, 0x0321101f}, - {0x16, 0x03011020}, - {0x18, 0x40000008}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41000001}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x16, 0x03011020}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell", ALC668_FIXUP_AUTO_MUTE, - {0x12, 0x411111f0}, {0x14, 0x90170110}, {0x15, 0x0321101f}, - {0x16, 0x03011020}, - {0x18, 0x40000008}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x41000001}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x16, 0x03011020}), SND_HDA_PIN_QUIRK(0x10ec0668, 0x1028, "Dell XPS 15", ALC668_FIXUP_AUTO_MUTE, {0x12, 0x90a60130}, {0x14, 0x90170110}, - {0x15, 0x0321101f}, - {0x16, 0x40000000}, - {0x18, 0x411111f0}, - {0x19, 0x411111f0}, - {0x1a, 0x411111f0}, - {0x1b, 0x411111f0}, - {0x1d, 0x40d6832d}, - {0x1e, 0x411111f0}, - {0x1f, 0x411111f0}), + {0x15, 0x0321101f}), {} }; -- 1.9.1

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Re: [alsa-devel] [PATCH] Fix logical-not-parentheses warning
by Takashi Iwai 03 Aug '15

03 Aug '15

On Sun, 02 Aug 2015 11:08:57 +0200, Tomer Barletz wrote: > > This fixes the following warning, that is seen with gcc 5.1: > warning: logical not is only applied to the left hand side of comparison [-Wlogical-not-parentheses]. > > Signed-off-by: Tomer Barletz <barletz(a)gmail.com> Applied, thanks. Takashi > --- > sound/pci/oxygen/oxygen_mixer.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c > index 6492bca..4ca1266 100644 > --- a/sound/pci/oxygen/oxygen_mixer.c > +++ b/sound/pci/oxygen/oxygen_mixer.c > @@ -88,7 +88,7 @@ static int dac_mute_put(struct snd_kcontrol *ctl, > int changed; > > mutex_lock(&chip->mutex); > - changed = !value->value.integer.value[0] != chip->dac_mute; > + changed = (!value->value.integer.value[0]) != chip->dac_mute; > if (changed) { > chip->dac_mute = !value->value.integer.value[0]; > chip->model.update_dac_mute(chip); > -- > 2.4.3 > >

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[alsa-devel] Beaglebone black : Audio cap & Linux version 4.1.3-ti-r6.2
by Micka 03 Aug '15

03 Aug '15

HI everyone, I'm trying to test the sound output with the driver tlv320aic3104. I'm stuck somewhere and I can't figure it out ! I Put a lot of printk and I don't know where to put other printk. Do you know who call the function soc_pcm_prepare ? 693 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L693> */**694 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L694> * * Called by ALSA when the PCM substream is prepared, can set format, sample*695 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L695> * * rate, etc. This function is non atomic and can be called multiple times,*696 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L696> * * it can refer to the runtime info.*697 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L697> * */*698 <http://lxr.free-electrons.com/source/sound/soc/soc-pcm.c#L698> static int soc_pcm_prepare <http://lxr.free-electrons.com/ident?i=soc_pcm_prepare>(struct snd_pcm_substream <http://lxr.free-electrons.com/ident?i=snd_pcm_substream> *substream) Looks like that it's ALSA, but I can't find where ? I tried to find a line in the kernel calling the function: rtd->ops <http://lxr.free-electrons.com/ident?i=ops>.prepare(.... ) can't find it .... my trace : [ 550.269286] ASoC: tlv320aic3x-hifi <-> 48038000.mcasp info: [ 550.269363] ASoC: rate mask 0x7fe [ 550.269398] ASoC: min ch 2 max ch 2 [ 550.269431] ASoC: min rate 8000 max rate 96000 [ 550.275809] tlv aic3x_hw_params [ 550.275862] snd_soc_read: snd soc component read: ret 0 reg 9 val 64 [ 550.275934] snd_soc_component_write reg 9 val 64 [ 550.275987] snd_soc_component_write reg 101 val 0 [ 550.276030] snd_soc_component_write reg 7 val 10 [ 550.276063] snd_soc_component_write reg 2 val 0 [ 550.289260] snd_soc_component_write reg 11 val 1 [ 550.289339] snd_soc_component_write reg 4 val 32 [ 550.289375] snd_soc_component_write reg 5 val 30 [ 550.289407] snd_soc_component_write reg 6 val 0 [ 550.289952] soc_pcm_prepare [ 550.290003] snd_soc_component_write reg 10 val 0 [ 550.290043] soc_dapm_stream_event 1 [ 550.290107] soc_dapm_stream_event: dapm_power_widgets:start dapm_power_widgets 1 [ 550.290150] trace_snd_soc_dapm_start [ 550.290176] dapm reset [ 550.290226] trace_snd_soc_dapm_widget_power [ 550.290257] trace_snd_soc_dapm_widget_power [ 550.290284] trace_snd_soc_dapm_widget_power [ 550.290311] trace_snd_soc_dapm_widget_power [ 550.290339] trace_snd_soc_dapm_widget_power [ 550.290370] trace_snd_soc_dapm_widget_power [ 550.290395] trace_snd_soc_dapm_widget_power [ 550.290423] trace_snd_soc_dapm_widget_power [ 550.290448] trace_snd_soc_dapm_widget_power [ 550.290477] trace_snd_soc_dapm_widget_power [ 550.290503] trace_snd_soc_dapm_widget_power [ 550.290531] trace_snd_soc_dapm_widget_power [ 550.290557] trace_snd_soc_dapm_widget_power [ 550.290582] trace_snd_soc_dapm_widget_power [ 550.290608] trace_snd_soc_dapm_widget_power [ 550.290634] trace_snd_soc_dapm_widget_power [ 550.290659] trace_snd_soc_dapm_widget_power [ 550.290686] trace_snd_soc_dapm_widget_power [ 550.290712] trace_snd_soc_dapm_widget_power [ 550.290738] trace_snd_soc_dapm_widget_power [ 550.290765] trace_snd_soc_dapm_widget_power [ 550.290791] trace_snd_soc_dapm_widget_power [ 550.290817] trace_snd_soc_dapm_widget_power [ 550.290842] trace_snd_soc_dapm_widget_power [ 550.290879] dap set bias to off [ 550.290910] dapm_pre_sequence_async off [ 550.290937] snd soc dapm set bias level 1 [ 550.290962] trace_snd_soc_bias_level_start 1 [ 550.290988] trace_snd_soc_bias_level_done 1 [ 550.291011] dapm_pre_sequence_async onsnd soc dapm set bias level 2 [ 550.291047] trace_snd_soc_bias_level_start 2 [ 550.291073] trace_snd_soc_bias_level_done 2 [ 550.291318] dapm_pre_sequence_async off [ 550.291354] snd soc dapm set bias level 1 [ 550.291381] trace_snd_soc_bias_level_start 1 [ 550.291405] soc set bias level [ 550.291432] tlv set bias level 1 [ 550.291456] tlv standby [ 550.291480] tlv set power [ 550.300504] snd_soc_read: snd soc component read: ret 0 reg 5 val 30 [ 550.300583] snd_soc_read: snd soc component read: ret 0 reg 6 val 0 [ 550.300631] snd_soc_component_write reg 5 val 30 [ 550.301103] snd_soc_component_write reg 6 val 0 [ 550.301577] trace_snd_soc_bias_level_done 1 [ 550.301611] dapm_pre_sequence_async onsnd soc dapm set bias level 2 [ 550.301650] trace_snd_soc_bias_level_start 2 [ 550.301675] soc set bias level [ 550.301701] tlv set bias level 2 [ 550.301724] tlv prepare [ 550.301757] trace_snd_soc_bias_level_done 2 [ 550.301864] dapm event WILL_PMU [ 550.301897] dapm event WILL_PMU [ 550.301923] dapm event WILL_PMU [ 550.301948] dapm event WILL_PMU [ 550.301972] dapm event WILL_PMU [ 550.301997] dapm event WILL_PMU [ 550.302021] dapm event WILL_PMU [ 550.302045] dapm event WILL_PMU [ 550.302069] dapm event WILL_PMU [ 550.302094] dapm event WILL_PMU [ 550.302119] dapm event WILL_PMU [ 550.302143] dapm event WILL_PMU [ 550.302168] dapm event WILL_PMU [ 550.302192] dapm event WILL_PMU [ 550.302216] dapm event WILL_PMU [ 550.302240] dapm event WILL_PMU [ 550.302265] dapm event WILL_PMU [ 550.302289] dapm event WILL_PMU [ 550.302313] dapm event WILL_PMU [ 550.302337] dapm event WILL_PMU [ 550.302362] dapm event WILL_PMU [ 550.302386] dapm event WILL_PMU [ 550.302410] dapm event WILL_PMU [ 550.302435] dapm event WILL_PMU [ 550.302460] dapm sq run 1 [ 550.302489] dapm sq run 1 [ 550.302523] dapm event PRE_PMU [ 550.302550] dapm event PRE_PMD [ 550.302575] dapm event PRE_PMU [ 550.302599] dapm event PRE_PMD [ 550.302623] dapm event PRE_PMU [ 550.302647] dapm event PRE_PMD [ 550.302671] dapm event PRE_PMU [ 550.302696] dapm event PRE_PMD [ 550.302720] dapm event PRE_PMU [ 550.302744] dapm event PRE_PMD [ 550.302769] dapm event PRE_PMU [ 550.302793] dapm event PRE_PMD [ 550.302817] dapm event POST_PMU [ 550.302842] dapm event POST_PMD [ 550.302867] dapm event POST_PMU [ 550.302891] dapm event POST_PMD [ 550.302915] dapm event POST_PMU [ 550.302939] dapm event POST_PMD [ 550.302964] dapm event POST_PMU [ 550.302988] dapm event POST_PMD [ 550.303012] dapm event POST_PMU [ 550.303036] dapm event POST_PMD [ 550.303060] dapm event POST_PMU [ 550.303084] dapm event POST_PMD [ 550.303110] dapm event PRE_PMU [ 550.303134] dapm event PRE_PMD [ 550.303159] dapm event POST_PMU [ 550.303183] dapm event POST_PMD [ 550.303209] dapm event PRE_PMU [ 550.303234] dapm event PRE_PMD [ 550.303258] dapm event PRE_PMU [ 550.303282] dapm event PRE_PMD [ 550.303306] dapm event PRE_PMU [ 550.303331] dapm event PRE_PMD [ 550.303355] dapm event PRE_PMU [ 550.303379] dapm event PRE_PMD [ 550.303403] dapm event POST_PMU [ 550.303427] dapm event POST_PMD [ 550.303451] dapm event POST_PMU [ 550.303475] dapm event POST_PMD [ 550.303499] dapm event POST_PMU [ 550.303524] dapm event POST_PMD [ 550.303548] dapm event POST_PMU [ 550.303572] dapm event POST_PMD [ 550.303598] dapm event PRE_PMU [ 550.303622] dapm event PRE_PMD [ 550.303647] dapm event PRE_PMU [ 550.303671] dapm event PRE_PMD [ 550.304162] dapm event POST_PMU [ 550.304199] dapm event POST_PMD [ 550.304226] dapm event POST_PMU [ 550.304250] dapm event POST_PMD [ 550.304278] dapm event PRE_PMU [ 550.304303] dapm event PRE_PMD [ 550.304328] dapm event PRE_PMU [ 550.304352] dapm event PRE_PMD [ 550.304377] dapm event PRE_PMU [ 550.304401] dapm event PRE_PMD [ 550.304426] dapm event PRE_PMU [ 550.304450] dapm event PRE_PMD [ 550.304474] dapm event POST_PMU [ 550.304499] dapm event POST_PMD [ 550.304523] dapm event POST_PMU [ 550.304547] dapm event POST_PMD [ 550.304572] dapm event POST_PMU [ 550.304596] dapm event POST_PMD [ 550.304620] dapm event POST_PMU [ 550.304644] dapm event POST_PMD [ 550.304670] dapm event PRE_PMU [ 550.304695] dapm event PRE_PMD [ 550.305156] dapm event POST_PMU [ 550.305191] dapm event POST_PMD [ 550.305219] dapm event PRE_PMU [ 550.305245] dapm event PRE_PMD [ 550.305696] dapm event POST_PMU [ 550.305730] dapm event POST_PMD [ 550.305757] dapm event PRE_PMU [ 550.305782] dapm event PRE_PMD [ 550.306275] dapm event POST_PMU [ 550.306312] dapm event POST_PMD [ 550.306341] dapm event PRE_PMU [ 550.306366] dapm event PRE_PMD [ 550.306849] dapm event POST_PMU [ 550.306885] dapm event POST_PMD [ 550.306914] dapm event PRE_PMU [ 550.306939] dapm event PRE_PMD [ 550.317926] dapm event POST_PMU [ 550.317958] dapm event POST_PMD [ 550.317977] dapm event PRE_PMU [ 550.317991] dapm event PRE_PMD [ 550.318401] dapm event POST_PMU [ 550.318422] dapm event POST_PMD [ 550.318439] dapm event PRE_PMU [ 550.318454] dapm event PRE_PMD [ 550.318467] dapm event POST_PMU [ 550.318480] dapm event POST_PMD [ 550.318496] dapm_power_widgets run all the bias changes in parrallel [ 550.321470] dapm_post_sequence_async prepare 1 [ 550.321498] snd soc dapm set bias level 3 [ 550.321513] trace_snd_soc_bias_level_start 3 [ 550.321527] soc set bias level [ 550.321543] tlv set bias level 3 [ 550.321555] tlv bias on [ 550.321570] trace_snd_soc_bias_level_done 3 [ 550.321649] dapm_power_widgets dapm_power_widgets [ 550.321666] DAPM sequencing finished [ 550.321680] trace_snd_soc_dapm_done [ 550.321695] pcm prepare call snd_soc_dai_digital_mute [ 550.321711] snd_soc_dai_digital_mute 0 0 [ 550.321725] tlv aic3x_mute [ 550.321739] snd_soc_read: snd soc component read: ret 0 reg 43 val 168 [ 550.321772] snd_soc_read: snd soc component read: ret 0 reg 44 val 168 [ 550.321798] snd_soc_component_write reg 43 val 40 [ 550.322224] snd_soc_component_write reg 44 val 40 [ 550.322630] pcm prepare call snd_soc_dai_digital_mute for cpu_dai [ 550.322651] snd_soc_dai_digital_mute 0 0 > >

1 1

[alsa-devel] [PATCH] ALSA: hda - Fix a wrong busy check in alt PCM open
by Takashi Iwai 03 Aug '15

03 Aug '15

Currently, the alt PCM open callback returns -EBUSY when an independent HP is turned off, supposing that it conflicts with the main PCM. However, obviously, this check is wrong when the independent HP itself isn't enabled but the alt PCM was explicitly created via alc_dac_nid by a codec driver. Reported-and-tested-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac0db1679f09..b077bb644434 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5175,7 +5175,7 @@ static int alt_playback_pcm_open(struct hda_pcm_stream *hinfo, int err = 0; mutex_lock(&spec->pcm_mutex); - if (!spec->indep_hp_enabled) + if (spec->indep_hp && !spec->indep_hp_enabled) err = -EBUSY; else spec->active_streams |= 1 << STREAM_INDEP_HP; -- 2.4.5

2 3

[alsa-devel] [PATCH] ALSA: hda - Add pin quirk for the headset mic jack detection on Dell laptop
by woodrow.shen＠canonical.com 03 Aug '15

03 Aug '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> The new Dell laptop with codec 256 can't detect headset mic when headset was inserted on the machine. From alsa-info, we check init_pin_configs and need to define the new register value for pin 0x1d & 0x1e because the original macro ALC256_STANDARD_PINS can't match pin definition. Also, the macro ALC256_STANDARD_PINS is simplified by removing them. This makes headset mic works on laptop. Codec: Realtek ALC256 Vendor Id: 0x10ec0256 Subsystem Id: 0x102806f2 BugLink: https://bugs.launchpad.net/bugs/1478497 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d94c0e3..4ae877c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5398,8 +5398,6 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x19, 0x411111f0}, \ {0x1a, 0x411111f0}, \ {0x1b, 0x411111f0}, \ - {0x1d, 0x40700001}, \ - {0x1e, 0x411111f0}, \ {0x21, 0x02211020} #define ALC282_STANDARD_PINS \ @@ -5556,10 +5554,19 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS, - {0x13, 0x40000000}), + {0x13, 0x40000000}, + {0x1d, 0x40700001}, + {0x1e, 0x411111f0}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS, - {0x13, 0x411111f0}), + {0x13, 0x411111f0}, + {0x1d, 0x40700001}, + {0x1e, 0x411111f0}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC256_STANDARD_PINS, + {0x13, 0x411111f0}, + {0x1d, 0x4077992d}, + {0x1e, 0x411111ff}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x13, 0x40000000}, -- 2.1.4

3 6

[alsa-devel] [PATCH 3/6] ASoC: fsl_ssi: enable AC'97 asymmetric rates
by Maciej S. Szmigiero 02 Aug '15

02 Aug '15

AC'97 bus can support asymmetric playback/capture rates so enable them in this case in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index a83b900..7f4f0b9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1377,7 +1377,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Are the RX and the TX clocks locked? */ if (!of_find_property(np, "fsl,ssi-asynchronous", NULL)) { - ssi_private->cpu_dai_drv.symmetric_rates = 1; + if (!fsl_ssi_is_ac97(ssi_private)) + ssi_private->cpu_dai_drv.symmetric_rates = 1; + ssi_private->cpu_dai_drv.symmetric_channels = 1; ssi_private->cpu_dai_drv.symmetric_samplebits = 1; }

2 4

[alsa-devel] [PATCH 5/6] ASoC: fsl_ssi: instantiate AC'97 CODEC
by Maciej S. Szmigiero 02 Aug '15

02 Aug '15

Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 21 +++++++++++++++++++++ 1 files changed, 21 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 154bcf6..8e5ff5e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1460,6 +1460,27 @@ done: _fsl_ssi_set_dai_fmt(&pdev->dev, ssi_private, ssi_private->dai_fmt); + if (fsl_ssi_is_ac97(ssi_private)) { + u32 ssi_idx; + + ret = of_property_read_u32(np, "cell-index", &ssi_idx); + if (ret) { + dev_err(&pdev->dev, "cannot get SSI index property\n"); + goto error_sound_card; + } + + ssi_private->pdev = + platform_device_register_data(NULL, + "ac97-codec", ssi_idx, NULL, 0); + if (IS_ERR(ssi_private->pdev)) { + ret = PTR_ERR(ssi_private->pdev); + dev_err(&pdev->dev, + "failed to register AC97 codec platform: %d\n", + ret); + goto error_sound_card; + } + } + return 0; error_sound_card:

2 3

Re: [alsa-devel] [PATCH] Fix logical-not-parentheses warning
by Joe Perches 02 Aug '15

02 Aug '15

On Sun, 2015-08-02 at 02:08 -0700, Tomer Barletz wrote: > This fixes the following warning, that is seen with gcc 5.1: > warning: logical not is only applied to the left hand side of > comparison [-Wlogical-not-parentheses]. [] > diff --git a/sound/pci/oxygen/oxygen_mixer.c [] > @@ -88,7 +88,7 @@ static int dac_mute_put(struct snd_kcontrol *ctl, > int changed; > > mutex_lock(&chip->mutex); > - changed = !value->value.integer.value[0] != chip->dac_mute; > + changed = (!value->value.integer.value[0]) != chip->dac_mute; It seems this is because dac_mute's value is inverted 0 means true, 1 is false. I think it'd be simpler if dac_mute was converted to a standard bool (or perhaps renamed to not_muted) and this test was rewritten. Using !!value->value.integer.value[0] would also be more readable and intelligible.

1 0

Re: [alsa-devel] [PATCH] Fix logical-not-parentheses warning
by Takashi Iwai 02 Aug '15

02 Aug '15

On Sun, 02 Aug 2015 07:43:22 +0200, Tomer Barletz wrote: > > Signed-off-by: Tomer Barletz <barletz(a)gmail.com> This is rather a gcc5 problem that doesn't recognize the expression correctly. Though, this change make things worse, so I'm willing to take it. But please give the warning message itself in changelog to show exactly what is the issue. The subject doesn't enough to understand why this change is needed. thanks, Takashi > --- > sound/pci/oxygen/oxygen_mixer.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c > index 6492bca..4ca1266 100644 > --- a/sound/pci/oxygen/oxygen_mixer.c > +++ b/sound/pci/oxygen/oxygen_mixer.c > @@ -88,7 +88,7 @@ static int dac_mute_put(struct snd_kcontrol *ctl, > int changed; > > mutex_lock(&chip->mutex); > - changed = !value->value.integer.value[0] != chip->dac_mute; > + changed = (!value->value.integer.value[0]) != chip->dac_mute; > if (changed) { > chip->dac_mute = !value->value.integer.value[0]; > chip->model.update_dac_mute(chip); > -- > 2.4.3 > >

1 0

Re: [alsa-devel] [pulseaudio-discuss] Strange hardware: XFX nForce 780i SLI MCP
by Raymond Yau 02 Aug '15

02 Aug '15

> > Yesterday I got an interesting email related to dcaenc support. > > The person has an "XFX nForce 780i SLI" MCP board: http://www.nvidia.com/object/product_nforce_780i_sli_us.html , http://www.evga.com/support/manuals/files/132-CK-NF78.pdf > > As you see on page 18 of the manual, this board has no HDMI output, but there is an optical SPDIF. Strangely enough, this SPDIF is his "hdmi:CARD=NVidia,DEV=0" device. > > Given the recent discussions related to dynamically creating PCM devices for HDMI, I think you will be interested in this alsa-info output. > > Is there anything here that would need special treatment by ALSA or PulseAudio? Or, is this just a misnumbered/misnamed codec? > This can also be regarded as bug of hda_auto_parser.c which does not determine type of digital pin correctly Those internal spdif connector which can connect to an optional spdif headser or spdif input header of the graphic card for hdmi audio output http://www.intel.com/support/motherboards/desktop/sb/CS-032871.htm But the pincap of these kind of digital pin usually does not support DETECT nor HDMI

1 0

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: Add DSP management routines
by Vinod Koul 01 Aug '15

01 Aug '15

This series adds NHLT table support in the driver. This also adds support for dsp init, modules configuration and messaging support Hardik T Shah (1): ASoC: Intel: Skylake: Add helpers for SRC and converter modules Jeeja KP (5): ASoC: Intel: Skylake: Add NHLT support to get BE config ASoC: Intel: Skylake: Add dsp and ipc init helpers ASoC: Intel: Skylake: Add functions for DSP module configuration ASoC: Intel: Skylake: Add DSP module init and binding routines ASoC: Intel: Skylake: Add pipe management helpers sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-messages.c | 828 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 141 +++++ sound/soc/intel/skylake/skl-nhlt.h | 116 ++++ sound/soc/intel/skylake/skl-topology.h | 276 +++++++++ sound/soc/intel/skylake/skl-tplg-interface.h | 88 +++ sound/soc/intel/skylake/skl.h | 13 + 7 files changed, 1463 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/skylake/skl-messages.c create mode 100644 sound/soc/intel/skylake/skl-nhlt.c create mode 100644 sound/soc/intel/skylake/skl-nhlt.h create mode 100644 sound/soc/intel/skylake/skl-topology.h create mode 100644 sound/soc/intel/skylake/skl-tplg-interface.h -- 1.9.1

2 18

[alsa-devel] [PATCH 00/27] Export I2C and OF module aliases in missing drivers
by Javier Martinez Canillas 01 Aug '15

01 Aug '15

Hello, Short version: This series add the missing MODULE_DEVICE_TABLE() for OF and I2C tables to export that information so modules have the correct aliases built-in and autoloading works correctly. Longer version: Currently it's mandatory for I2C drivers to have an I2C device ID table regardless if the device was registered using platform data or OF. This is because the I2C core needs an I2C device ID table for two reasons: 1) Match the I2C client with a I2C device ID so a struct i2c_device_id is passed to the I2C driver probe() function. 2) Export the module aliases from the I2C device ID table so userspace can auto-load the correct module. This is because i2c_device_uevent always reports a MODALIAS of the form i2c:<client->name>. Lee Jones posted a patch series [0] to solve 1) by allowing the I2C drivers to have a probe() function that does not get a i2c_device_id. The problem is that his series didn't take into account 2) so if that was merged and the I2C ID table is removed from all the drivers that don't needed it, module auto-loading will break for those. But even now there are many I2C drivers were module auto-loading is not working because of the fact that the I2C core always reports the MODALIAS as i2c:<client->name> and many developers didn't expect this. I've identified I2C drivers with 3 types of different issues: a) Those that have an i2c_table but are not exported. The match works and the correct i2c_device_id is passed on probe but since the ID table is not exported, module auto-load won't work. b) Those that have a of_table but are not exported. This is currently not an issue since even when the of_table is used to match the dev with the driver, an OF modalias is not reported by the I2C core. But if the I2C core is changed to report the MODALIAS of the form of:N*T*C as it's made by other subsystems, then module auto-load will break for these drivers. c) Those that don't have a of_table but should since are OF drivers with DT bindings doc for them. Since the I2C core does not report a OF modalias and since i2c_device_match() fallbacks to match the device part of the compatible string with the I2C device ID table, many OF drivers don't have an of_table to match. After all having a I2C device ID table is mandatory so it works without a of_table. So, in order to not make mandatory to have a I2C device ID table, at least a) and b) needs to be addressed, this series does that. c) should be fixed too since it seems wrong that a driver with a DT binding document, does not have a OF table and export it to modules. Also stripping the vendor part from the compatible string to match with the I2C devices ID table and reporting only the device name to user-space doesn't seem to be correct. I've identified at least two drivers that have the same name on their I2C device ID table so the manufacturer prefix is important. But I've not tried to fix c) yet since that is not so easy to automate due drivers not having all the information (i.e: the device name can match a documented compatible string device part but without the vendor prefix is hard to tell). I split the changes so the patches in this series are independent and can be picked individually by subsystem maintainers. Patch #27 changes the logic of i2c_device_uevent() to report an OF modalias if the device was registered using OF. But this patch is included in the series only as an RFC for illustration purposes since changing that without fixing c) will break module auto-loading for the drivers of devices registered with OF but that don't have a of_match_table. Although arguably, those drivers were relying on the assumption that a MODALIAS=i2c:<foo> would always be reported even for the OF case which is not the true on other subsystems. [0]: https://lkml.org/lkml/2014/8/28/283 Best regards, Javier Javier Martinez Canillas (27): mfd: stw481x: Export I2C module alias information spi: xcomm: Export I2C module alias information iio: Export I2C module alias information in missing drivers [media] Export I2C module alias information in missing drivers macintosh: therm_windtunnel: Export I2C module alias information misc: eeprom: Export I2C module alias information in missing drivers Input: Export I2C module alias information in missing drivers power: Export I2C module alias information in missing drivers i2c: core: Export I2C module alias information in dummy driver backlight: tosa: Export I2C module alias information [media] staging: media: lirc: Export I2C module alias information usb: phy: isp1301: Export I2C module alias information ALSA: ppc: keywest: Export I2C module alias information hwmon: (nct7904) Export I2C module alias information regulator: fan53555: Export I2C module alias information mfd: Export OF module alias information in missing drivers iio: Export OF module alias information in missing drivers hwmon: (g762) Export OF module alias information extcon: Export OF module alias information in missing drivers ASoC: Export OF module alias information in missing codec drivers rtc: Export OF module alias information in missing drivers macintosh: therm_windtunnel: Export OF module alias information leds: Export OF module alias information in missing drivers [media] smiapp: Export OF module alias information Input: touchscreen - Export OF module alias information regulator: isl9305: Export OF module alias information i2c: (RFC, don't apply) report OF style modalias when probing using DT drivers/extcon/extcon-rt8973a.c | 1 + drivers/extcon/extcon-sm5502.c | 1 + drivers/hwmon/g762.c | 1 + drivers/hwmon/nct7904.c | 1 + drivers/i2c/i2c-core.c | 9 +++++++++ drivers/iio/accel/mma8452.c | 1 + drivers/iio/accel/stk8312.c | 1 + drivers/iio/accel/stk8ba50.c | 1 + drivers/iio/light/cm32181.c | 1 + drivers/iio/light/cm3232.c | 1 + drivers/iio/light/cm36651.c | 1 + drivers/iio/light/gp2ap020a00f.c | 1 + drivers/iio/light/stk3310.c | 1 + drivers/input/misc/gp2ap002a00f.c | 1 + drivers/input/touchscreen/egalax_ts.c | 1 + drivers/input/touchscreen/goodix.c | 1 + drivers/input/touchscreen/mms114.c | 1 + drivers/leds/leds-pca963x.c | 1 + drivers/leds/leds-tca6507.c | 1 + drivers/macintosh/therm_windtunnel.c | 2 ++ drivers/media/i2c/ir-kbd-i2c.c | 1 + drivers/media/i2c/s5k6a3.c | 1 + drivers/media/i2c/smiapp/smiapp-core.c | 1 + drivers/mfd/rt5033.c | 1 + drivers/mfd/stw481x.c | 1 + drivers/mfd/tps65217.c | 1 + drivers/mfd/tps65218.c | 1 + drivers/misc/eeprom/eeprom.c | 1 + drivers/misc/eeprom/max6875.c | 1 + drivers/power/bq24190_charger.c | 1 + drivers/power/rt5033_battery.c | 2 +- drivers/regulator/fan53555.c | 1 + drivers/regulator/isl9305.c | 1 + drivers/rtc/rtc-ab-b5ze-s3.c | 1 + drivers/rtc/rtc-isl12022.c | 1 + drivers/rtc/rtc-isl12057.c | 1 + drivers/spi/spi-xcomm.c | 1 + drivers/staging/media/lirc/lirc_zilog.c | 1 + drivers/usb/phy/phy-isp1301.c | 1 + drivers/video/backlight/tosa_bl.c | 1 + sound/ppc/keywest.c | 1 + sound/soc/codecs/da9055.c | 1 + sound/soc/codecs/wm8510.c | 1 + sound/soc/codecs/wm8523.c | 1 + sound/soc/codecs/wm8580.c | 1 + 45 files changed, 54 insertions(+), 1 deletion(-) -- 2.4.3

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Re: [alsa-devel] Abouut adv7511 driver
by Lars-Peter Clausen 01 Aug '15

01 Aug '15

Cc: alsa-ml On 07/31/2015 07:38 AM, Kuninori Morimoto wrote: > > Hi Lars > Cc Mark > > I noticed Analog Devices has adv7511 support on > http://wiki.analog.com/resources/tools-software/linux-drivers/drm/adv7511 > > And I tried this driver on my board, and it works. > > if my understanding was correct, latest linux kernel doesn't have > adv7511 "sound" support yet. > Thus, I would like to have it on linux kernel. > According to this driver/file, it is under GPL-2, so it is no problem > from license point of view. > > According to git log Signd-off-by, Lars/Mark are listed on this adv7511 sound driver. > but I wonder why it is not on upstream ? Are there some reason ?? > If you don't mind, I would like to send it to ML. > This is OK ? The reason why the driver is not upstream at the moment is that the way it implements things is definitely not the right way. But there is still a lot of discussion on what actually is the right way to implement audio support for HDMI devices. Once that has been sorted out the driver will hopefully go upstream. - Lars -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

4 3

[alsa-devel] AudioFire4 quirk
by Johannes Oertel 01 Aug '15

01 Aug '15

Hi, when I try to use the snd-fireworks driver with my AudioFire4 (see the output of /proc/asound/cardX/firewire/firmware below) on my machine (kernel 4.1), it doesn't work. I am experiencing exactly the same problems as Harry with his AudioFire2 [1]; dmesg was reporting discontinuities. A few days ago, I saw Takashi Sakamoto's most recent patch for making the AudioFire2 work in this case [2]. I adapted his patch (see below) to apply to my situation with the AudioFire4 and now it is working flawlessly! However, according to the README in Takashi's GitHub repository [3], he is testing the driver with an AudioFire4 as well, so I asked him why I need this patch and he does not. He told me he is using firmware version 5.5 while I'm on 5.7.3. He said he is going to test his device with different firmware versions soon and write a patch, if needed. Finally, he asked me to post this whole issue to this list as well and that's what I have just done. :) Best regards, Johannes [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-August/080379.html [2] http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=9c68… [3] https://github.com/takaswie/snd-firewire-improve /proc/asound/cardX/firewire/firmware: guid_hi: 0x148604 guid_lo: 0x5BA8D7BF type: 0xAF4 version: 0x0 vendor_name: Echo Digital Audio model_name: AudioFire4 dsp_version: 0x0 arm_version: 0x5070300 fpga_version: 0x3000200 flags: 0xE1 max_sample_rate: 0x17700 min_sample_rate: 0x7D00 supported_clock: 0x9 phys out: 0x6 phys in: 0x6 phys in grps: 0x2 phys in grp[0]: type 0x0, count 0x4 phys in grp[1]: type 0x1, count 0x2 phys out grps: 0x2 phys out grps[0]: type 0x0, count 0x4 phys out grps[1]: type 0x1, count 0x2 amdtp rx pcm channels 1x: 0x6 amdtp tx pcm channels 1x: 0x6 amdtp rx pcm channels 2x: 0x6 amdtp tx pcm channels 2x: 0x6 amdtp rx pcm channels 4x: 0x6 amdtp tx pcm channels 4x: 0x6 midi out ports: 0x1 midi in ports: 0x1 mixer playback channels: 0x6 mixer capture channels: 0x6 The patch: --- sound/firewire/fireworks/fireworks.c | 2 ++ sound/firewire/fireworks/fireworks.h | 1 + sound/firewire/fireworks/fireworks_stream.c | 4 ++-- 3 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index c670db4..544bdbf 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -250,6 +250,8 @@ efw_probe(struct fw_unit *unit, goto error; if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2) efw->is_af2 = true; + if (entry->model_id == MODEL_ECHO_AUDIOFIRE_4) + efw->is_af4 = true; if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) efw->is_af9 = true; diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 9493122..9a7dcbb 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -74,6 +74,7 @@ struct snd_efw { /* for quirks */ bool is_af2; + bool is_af4; bool is_af9; u32 firmware_version; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index a0762dd..182b96f 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -172,8 +172,8 @@ int snd_efw_stream_init_duplex(struct snd_efw *efw) efw->tx_stream.flags |= CIP_DBC_IS_END_EVENT; /* Fireworks reset dbc at bus reset. */ efw->tx_stream.flags |= CIP_SKIP_DBC_ZERO_CHECK; - /* AudioFire2 starts packets with non-zero dbc. */ - if (efw->is_af2) + /* AudioFire2 and AudioFire4 start packets with non-zero dbc. */ + if (efw->is_af2 || efw->is_af4) efw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; /* AudioFire9 always reports wrong dbs. */ if (efw->is_af9) -- 2.4.6

2 1

[alsa-devel] Laptop internal speaker volume is apparently controlled by the headphone control (Intel HDA Panther Point)
by Dang Sananikone 01 Aug '15

01 Aug '15

Hi, I've just noticed that the internal speaker volume on my laptop is now controlled by the headphone control in alsamixer. I think this may be a recent change although I can't be sure. I'm using Arch Linux with kernel 4.1.3-1. I've uploaded the alsa info here: http://www.alsa-project.org/db/?f=f04e506dad863823b18f8f9e9d0261fdacd72695 Let me know if I can help. Kind Regards, Dang.

2 1

[alsa-devel] Using regmap_multi_reg_write() with regcache_sync()
by Sudheer 31 Jul '15

31 Jul '15

Hi Mark, We are trying to measure cold start output latency for audio playback usecase on our platform and observing that the latency is high. As a part of optimization, we have profiled regcache_sync() API when the codec is out of reset and observed that regcache_sync() takes little long time for synching all of the codec registers (around 100 registers). But, the bus connecting the codec has the capability to send multiple register writes at once. It seems regmap_multi_reg_write() can be used for triggering multiple register writes at once. So, we would like to know, if it is possible to use regmap_multi_reg_write() or a similar function with regcache_sync that intern calls bus driver's multi_reg_write API to sync to the HW, that will help in reducing the latency. Your comments/suggestions are highly appreciated. Thanks & Regards, Sudheer Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH v3 00/13] alsa-lib topology support
by Liam Girdwood 31 Jul '15

31 Jul '15

This series adds support for parsing topology data text files and converting to the binary topology format accepted by the kernel. Changes since V2 :- o Added documentation in doxygen format describing API and text file format. o use SNDERR macro for error reporting. o Added text copy helper. o Make all parsed keywords lowercase o Assignment for some structure copies instead of memcpy(). o Fixed the private data parser to handle byte, short and word data. Changes since V1 :- o Terminate all strncpy() strings with NULL. o Use snd_pcm_format_value() instead of local lookup. o Set a maximum size for private data files. o Set capture/playback stream direction for new PCMs o Only set TLV access flags for TLV CTLs. Liam Girdwood (13): topology: uapi: Add UAPI headers for topology ABI topology: Add topology core parser. topology: Add text section parser. topology: Add PCM parser. topology: Add operations parser topology: Add private data parser topology: Add DAPM object parser topology: Add CTL parser topology: Add Channel map parser. topology: Add binary file builder. topology: autotools: Add build support for topology core topology: doxygen: Add doxygen support for topology core. conf: topology: Add topology file for broadwell audio DSP configure.ac | 11 +- doc/doxygen.cfg.in | 7 +- doc/index.doxygen | 1 + include/Makefile.am | 4 + include/sound/Makefile.am | 3 +- include/sound/asoc.h | 388 ++++++++++++++++++ include/sound/tlv.h | 23 ++ include/topology.h | 497 ++++++++++++++++++++++ src/Makefile.am | 7 + src/conf/Makefile.am | 2 +- src/conf/topology/Makefile.am | 1 + src/conf/topology/broadwell/Makefile.am | 4 + src/conf/topology/broadwell/broadwell.conf | 375 +++++++++++++++++ src/topology/Makefile.am | 19 + src/topology/builder.c | 327 +++++++++++++++ src/topology/channel.c | 122 ++++++ src/topology/ctl.c | 613 +++++++++++++++++++++++++++ src/topology/dapm.c | 562 +++++++++++++++++++++++++ src/topology/data.c | 396 ++++++++++++++++++ src/topology/elem.c | 187 +++++++++ src/topology/ops.c | 84 ++++ src/topology/parser.c | 359 ++++++++++++++++ src/topology/pcm.c | 639 +++++++++++++++++++++++++++++ src/topology/text.c | 88 ++++ src/topology/tplg_local.h | 234 +++++++++++ 25 files changed, 4948 insertions(+), 5 deletions(-) create mode 100644 include/sound/asoc.h create mode 100644 include/sound/tlv.h create mode 100644 include/topology.h create mode 100644 src/conf/topology/Makefile.am create mode 100644 src/conf/topology/broadwell/Makefile.am create mode 100644 src/conf/topology/broadwell/broadwell.conf create mode 100644 src/topology/Makefile.am create mode 100644 src/topology/builder.c create mode 100644 src/topology/channel.c create mode 100644 src/topology/ctl.c create mode 100644 src/topology/dapm.c create mode 100644 src/topology/data.c create mode 100644 src/topology/elem.c create mode 100644 src/topology/ops.c create mode 100644 src/topology/parser.c create mode 100644 src/topology/pcm.c create mode 100644 src/topology/text.c create mode 100644 src/topology/tplg_local.h -- 2.1.4

2 16

[alsa-devel] [PATCH v2 0/22] On-demand device probing
by Tomeu Vizoso 31 Jul '15

31 Jul '15

Hello, I have a problem with the panel on my Tegra Chromebook taking longer than expected to be ready during boot (Stéphane Marchesin reported what is basically the same issue in [0]), and have looked into ordered probing as a better way of solving this than moving nodes around in the DT or playing with initcall levels and linking order. While reading the thread [1] that Alexander Holler started with his series to make probing order deterministic, it occurred to me that it should be possible to achieve the same by probing devices as they are referenced by other devices. This basically reuses the information that is already implicit in the probe() implementations, saving us from refactoring existing drivers or adding information to DTBs. During review of v1 of this series Linus Walleij suggested that it should be the device driver core to make sure that dependencies are ready before probing a device. I gave this idea a try [2] but Mark Brown pointed out to the logic duplication between the resource acquisition and dependency discovery code paths (though I think it's fairly minor). To address that code duplication I experimented with Arnd's devm_probe [3] concept of having drivers declare their dependencies instead of acquiring them during probe, and while it worked [4], I don't think we end up winning anything when compared to just probing devices on-demand from resource getters. One remaining objection is to the "sprinkling" of calls to fwnode_ensure_device() in the resource getters of each subsystem, but I think it's the right thing to do given that the storage of resources is currently subsystem-specific. We could avoid the above by moving resource storage into the core, but I don't think there's a compelling case for that. I have tested this on boards with Tegra, iMX.6, Exynos and OMAP SoCs, and these patches were enough to eliminate all the deferred probes (except one in PandaBoard because omap_dma_system doesn't have a firmware node as of yet). With this series I get the kernel to output to the panel in 0.5s, instead of 2.8s. Regards, Tomeu [0] http://lists.freedesktop.org/archives/dri-devel/2014-August/066527.html [1] https://lkml.org/lkml/2014/5/12/452 [2] https://lkml.org/lkml/2015/6/17/305 [3] http://article.gmane.org/gmane.linux.ports.arm.kernel/277689 [4] https://lkml.org/lkml/2015/7/21/441 Changes in v2: - Move delay to platform.c - Use set_primary_fwnode() - Use of_node_full_name() - Move the logic for finding a platform device from its firmware node to of/platform.c as it's not needed for ACPI nodes. - Add acpi_dev_get_device() - Add fwnode_ensure_device() so the mechanism for probing devices on demand is independent of the firmware format. - Acquire regulator device lock before returning from regulator_dev_lookup() Tomeu Vizoso (22): platform: delay device-driver matches until late_initcall of/platform: Set fwnode field for new devices device property: add fwnode_get_name() of/platform: add of_platform_device_find() ACPI: add acpi_dev_get_device() device property: add fwnode_ensure_device() gpio: Probe GPIO drivers on demand gpio: Probe pinctrl devices on demand regulator: core: Reduce critical area in _regulator_get regulator: core: Probe regulators on demand drm: Probe panels on demand drm/tegra: Probe dpaux devices on demand i2c: core: Probe i2c master devices on demand pwm: Probe PWM chip devices on demand backlight: Probe backlight devices on demand usb: phy: Probe phy devices on demand clk: Probe clk providers on demand pinctrl: Probe pinctrl devices on demand phy: core: Probe phy providers on demand dma: of: Probe DMA controllers on demand power-supply: Probe power supplies on demand ASoC: core: Probe components on demand drivers/base/platform.c | 28 +++++++++++ drivers/base/property.c | 73 +++++++++++++++++++++++++++ drivers/clk/clk.c | 3 ++ drivers/dma/of-dma.c | 2 + drivers/gpio/gpiolib-of.c | 4 ++ drivers/gpu/drm/drm_panel.c | 2 + drivers/gpu/drm/tegra/dpaux.c | 2 + drivers/i2c/i2c-core.c | 2 + drivers/of/platform.c | 61 +++++++++++++++++++++++ drivers/phy/phy-core.c | 2 + drivers/pinctrl/devicetree.c | 1 + drivers/power/power_supply_core.c | 2 + drivers/pwm/core.c | 2 + drivers/regulator/core.c | 99 +++++++++++++++++++++---------------- drivers/usb/phy/phy.c | 2 + drivers/video/backlight/backlight.c | 2 + include/linux/acpi.h | 10 ++++ include/linux/of_platform.h | 1 + include/linux/property.h | 4 ++ sound/soc/soc-core.c | 6 ++- 20 files changed, 264 insertions(+), 44 deletions(-) -- 2.4.3

3 4

[alsa-devel] [PATCH] ASoC: cs4349: Fix up setting PWR_DWN bit
by Axel Lin 30 Jul '15

30 Jul '15

The PWR_DWN is Bit 7, so current code does not set the PWR_DWN bit. Fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4349.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 852be85..0d010c2 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -326,7 +326,7 @@ static int cs4349_runtime_suspend(struct device *dev) struct cs4349_private *cs4349 = dev_get_drvdata(dev); int ret; - ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, 1); + ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, PWR_DWN); if (ret < 0) return ret; -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: cs4349: Remove unneeded NULL test for cs4349->reset_gpio
by Axel Lin 30 Jul '15

30 Jul '15

It's safe to call gpiod_set_value_cansleep() with NULL desc. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4349.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index a7538ae..852be85 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -300,8 +300,7 @@ static int cs4349_i2c_probe(struct i2c_client *client, if (IS_ERR(cs4349->reset_gpio)) return PTR_ERR(cs4349->reset_gpio); - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); i2c_set_clientdata(client, cs4349); @@ -316,8 +315,7 @@ static int cs4349_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); /* Hold down reset */ - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); return 0; } @@ -335,8 +333,7 @@ static int cs4349_runtime_suspend(struct device *dev) regcache_cache_only(cs4349->regmap, true); /* Hold down reset */ - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 0); + gpiod_set_value_cansleep(cs4349->reset_gpio, 0); return 0; } @@ -350,8 +347,7 @@ static int cs4349_runtime_resume(struct device *dev) if (ret < 0) return ret; - if (cs4349->reset_gpio) - gpiod_set_value_cansleep(cs4349->reset_gpio, 1); + gpiod_set_value_cansleep(cs4349->reset_gpio, 1); regcache_cache_only(cs4349->regmap, false); regcache_sync(cs4349->regmap); -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: cs4349: Fix setting digital interface format
by Axel Lin 30 Jul '15

30 Jul '15

Mode Control - Register 02h Digital Interface Format (DIF[2:0]) Bits 6-4 DIF2 DIF1 DIF0 Description 0 0 0 Left-Justified, up to 24-bit data 0 0 1 I²S, up to 24-bit data 0 1 0 Right-Justified, 16-bit data 0 1 1 Right-Justified, 24-bit data 1 0 0 TDM slot 0 1 0 1 TDM slot 1 1 1 0 TDM slot 2 1 1 1 TDM slot 3 The DIF_MASK is 0x70, so current code does not correctly set the DIFx setting. Fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- I assume the path "ASoC: cs4349: Don't use rtd->codec" will be applied first, so this patch is on top of that patch. sound/soc/codecs/cs4349.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index 7f5c3c70..989289d 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -98,17 +98,16 @@ static int cs4349_pcm_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct cs4349_private *cs4349 = snd_soc_codec_get_drvdata(codec); - int mode, fmt, ret; + int fmt, ret; - mode = snd_soc_read(codec, CS4349_MODE); cs4349->rate = params_rate(params); switch (cs4349->mode) { case SND_SOC_DAIFMT_I2S: - mode |= MODE_FORMAT(DIF_I2S); + fmt = DIF_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - mode |= MODE_FORMAT(DIF_LEFT_JST); + fmt = DIF_LEFT_JST; break; case SND_SOC_DAIFMT_RIGHT_J: switch (params_width(params)) { @@ -121,13 +120,13 @@ static int cs4349_pcm_hw_params(struct snd_pcm_substream *substream, default: return -EINVAL; } - mode |= MODE_FORMAT(fmt); break; default: return -EINVAL; } - ret = snd_soc_write(codec, CS4349_MODE, mode); + ret = snd_soc_update_bits(codec, CS4349_MODE, DIF_MASK, + MODE_FORMAT(fmt)); if (ret < 0) return ret; -- 2.1.0

3 2

[alsa-devel] [RFC][PATCH] ALSA: control: ensure returning from blocked system-call after closing
by Takashi Sakamoto 30 Jul '15

30 Jul '15

Some operations with blocking flag for ALSA control character devices don't return to userspace when closing a file descriptor corresponding to the character device. I think this is a bug of ALSA core. While, I find no resources to assist this aspect. In a manual of close(2)/read(2), there'no description about it. On SUSv2, there's descriptions for pipe or FIFO for this behaviour, while nothing for character devices. On Linux kernel documentation, I cannot find this kind of explaination. Thus, I just stand on my experiences to use these operations for files on filesystems, sockets and virtual character devices. It may merely be due to my misunderstanding. Well, additionally, in unsubscribed state, read operation for the character device returns with EBADFD. On the other hand, in subscribed state and once the operations are going to be blocked, they don't return to userspace even if any unsubscribe ioctl(2) operations are executed. This losts a consistency about the actual effects of unsubscribe operation. It's my pleasure to discuss about these bugs (and my patch candidate to fix them). You can see these bugs with this sample program. = Sample program = #include <stdio.h> #include <stdlib.h> #include <errno.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <unistd.h> #include <sys/ioctl.h> #include <sound/asound.h> #include <pthread.h> static void *thread_run(void *arg) { int fd = *(int *)arg; char buf[512] = {}; while (read(fd, buf, sizeof(buf)) >= 0) ; return (void *)NULL; } int main(int argc, char *argv[]) { pthread_t thread = pthread_self(); int fd; int subscribe; fd = open("/dev/snd/controlC0", O_RDONLY); if (fd < 0) { printf("%s\n", strerror(errno)); return EXIT_FAILURE; } subscribe = 1; if (ioctl(fd, SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS, &subscribe) < 0) { printf("%s\n", strerror(errno)); close(fd); return EXIT_FAILURE; } if (pthread_create(&thread, NULL, thread_run, (void *)&fd) < 0) { printf("%s\n", strerror(errno)); close(fd); return EXIT_FAILURE; } sleep(1); if (argc == 1) { close(fd); } else { subscribe = 0; if (ioctl(fd, SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS, &subscribe) < 0) { printf("%s\n", strerror(errno)); close(fd); return EXIT_FAILURE; } } /* Here, I expect blocking state of read(2) should be released. But... */ pthread_join(thread, NULL); return EXIT_SUCCESS; } Takashi Sakamoto (1): ALSA: control: add .flush operation to release blocking operation sound/core/control.c | 38 +++++++++++++++++++++++++++++++------- 1 file changed, 31 insertions(+), 7 deletions(-) -- 2.1.4

3 4

[alsa-devel] [PATCH 1/2] ASoC: cs4349: Fix suspend/resume
by Lars-Peter Clausen 30 Jul '15

30 Jul '15

dev_get_drvdata() will not return the snd_soc_runtime to which this CODEC is attached, so the current code will result in undefined behavior. To fix this just use regmap_update_bits(cs4349->regmap, ...) directly instead of snd_soc_update_bits(rtd->codec, ...). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/cs4349.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index a8df8a74..db18799 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -328,10 +328,9 @@ static int cs4349_i2c_remove(struct i2c_client *client) static int cs4349_runtime_suspend(struct device *dev) { struct cs4349_private *cs4349 = dev_get_drvdata(dev); - struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; - ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 1); + ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, 1); if (ret < 0) return ret; @@ -347,10 +346,9 @@ static int cs4349_runtime_suspend(struct device *dev) static int cs4349_runtime_resume(struct device *dev) { struct cs4349_private *cs4349 = dev_get_drvdata(dev); - struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; - ret = snd_soc_update_bits(rtd->codec, CS4349_MISC, PWR_DWN, 0); + ret = regmap_update_bits(cs4349->regmap, CS4349_MISC, PWR_DWN, 0); if (ret < 0) return ret; -- 2.1.4

3 4

[alsa-devel] [PATCH 1/2] ASoC: cs4349: Set .writeable_reg for cs4349_regmap
by Axel Lin 30 Jul '15

30 Jul '15

The first valid register index is 1 rather than 0, and the CS4349_CHIPID is readonly. So set .writeable_reg to avoid writing to these registers. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs4349.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4349.c b/sound/soc/codecs/cs4349.c index f4fccc6..eb55889 100644 --- a/sound/soc/codecs/cs4349.c +++ b/sound/soc/codecs/cs4349.c @@ -54,14 +54,17 @@ struct cs4349_private { static bool cs4349_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4349_CHIPID: - case CS4349_MODE: - case CS4349_VMI: - case CS4349_MUTE: - case CS4349_VOLA: - case CS4349_VOLB: - case CS4349_RMPFLT: - case CS4349_MISC: + case CS4349_CHIPID ... CS4349_MISC: + return true; + default: + return false; + } +} + +static bool cs4349_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS4349_MODE ... CS4349_MISC: return true; default: return false; @@ -272,6 +275,7 @@ static const struct regmap_config cs4349_regmap = { .reg_defaults = cs4349_reg_defaults, .num_reg_defaults = ARRAY_SIZE(cs4349_reg_defaults), .readable_reg = cs4349_readable_register, + .writeable_reg = cs4349_writeable_register, .cache_type = REGCACHE_RBTREE, }; -- 2.1.0

3 4

[alsa-devel] [PATCH] ALSA: hda - Fix MacBook Pro 5,2 quirk
by Takashi Iwai 30 Jul '15

30 Jul '15

MacBook Pro 5,2 with ALC889 codec had already a fixup entry, but this seems not working correctly, a fix for pin NID 0x15 is needed in addition. It's equivalent with the fixup for MacBook Air 1,1, so use this instead. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102131 Reported-and-tested-by: Jeffery Miller <jefferym(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 18ae17ebb356..c456c04e0928 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2222,7 +2222,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x4300, "iMac 9,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), - SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_MBA11_VREF), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), -- 2.4.6

1 0

[alsa-devel] [PATCH 1/6] ASoC: fsl_ssi: enable IPG clock during AC'97 reg access
by Maciej S. Szmigiero 30 Jul '15

30 Jul '15

IPG clock have to be enabled during AC'97 CODEC register access in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 19 +++++++++++++++++++ 1 files changed, 19 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 484ff20..8185edc 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1127,10 +1127,17 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct regmap *regs = fsl_ac97_data->regs; unsigned int lreg; unsigned int lval; + int ret; if (reg > 0x7f) return; + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 write clk_prepare_enable failed: %d\n", + ret); + return; + } lreg = reg << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1141,6 +1148,8 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, regmap_update_bits(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_RDWR_MASK, CCSR_SSI_SACNT_WR); udelay(100); + + clk_disable_unprepare(fsl_ac97_data->clk); } static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, @@ -1151,6 +1160,14 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short val = -1; u32 reg_val; unsigned int lreg; + int ret; + + ret = clk_prepare_enable(fsl_ac97_data->clk); + if (ret) { + pr_err("ac97 read clk_prepare_enable failed: %d\n", + ret); + return -1; + } lreg = (reg & 0x7f) << 12; regmap_write(regs, CCSR_SSI_SACADD, lreg); @@ -1162,6 +1179,8 @@ static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, regmap_read(regs, CCSR_SSI_SACDAT, &reg_val); val = (reg_val >> 4) & 0xffff; + clk_disable_unprepare(fsl_ac97_data->clk); + return val; }

2 2

[alsa-devel] [PATCH 4/6] ASoC: fsl_ssi: add AC'97 ops setting check and cleanup
by Maciej S. Szmigiero 30 Jul '15

30 Jul '15

Check whether setting AC'97 ops succeeded and clean them on removal so the fsl_ssi driver can be reloaded. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 7f4f0b9..154bcf6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1340,7 +1340,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) fsl_ac97_data = ssi_private; - snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); + if (ret) { + dev_err(&pdev->dev, "could not set AC'97 ops\n"); + return ret; + } } else { /* Initialize this copy of the CPU DAI driver structure */ memcpy(&ssi_private->cpu_dai_drv, &fsl_ssi_dai_template, @@ -1480,6 +1484,9 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->soc->imx) fsl_ssi_imx_clean(pdev, ssi_private); + if (fsl_ssi_is_ac97(ssi_private)) + snd_soc_set_ac97_ops(NULL); + return 0; }

1 0

[alsa-devel] [PATCH 2/6] ASoC: fsl_ssi: AC'97 DAI driver needs probe method too
by Maciej S. Szmigiero 30 Jul '15

30 Jul '15

AC'97 DAI driver struct need the same probe method as I2S one to setup DMA params in fsl_ssi driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8185edc..a83b900 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1101,6 +1101,7 @@ static const struct snd_soc_component_driver fsl_ssi_component = { static struct snd_soc_dai_driver fsl_ssi_ac97_dai = { .bus_control = true, + .probe = fsl_ssi_dai_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 2,

1 0

[alsa-devel] [RFC PATCH 0/4] better support for bursty DMA usages
by Pierre-Louis Bossart 30 Jul '15

30 Jul '15

Set of patches to fix issues with hw_ptr fuzziness [1] and increased buffering w/ DSPs 1. disable rewinds to allow for new HDaudio SPIB DMA functionality (fetch up to the application pointer, rewinds not supported) 2. report max in-flight bytes to avoid problems with stale data (late wake-ups, rewinds) [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093646.html TODO: 1. fixes and alsa-lib updates (compile-tested only for now) 2. get feedback 3. if supported, set DMA buffering based on negotiation between driver and app (capabilities vs. latency needs) Pierre-Louis Bossart (4): ALSA: core: let low-level driver or userspace disable rewinds ALSA: core: add .notify callback for pcm ops ALSA: core: add report of max dma burst ALSA: hda: add default value for max_dma_burst include/sound/pcm.h | 5 +++++ include/uapi/sound/asound.h | 6 ++++-- sound/core/pcm_lib.c | 19 +++++++++++++++++++ sound/core/pcm_native.c | 33 ++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.c | 1 + 5 files changed, 61 insertions(+), 3 deletions(-) -- 1.9.1

6 33

Re: [alsa-devel] [ALC668]: Asus N751JK - Incorrect default pin assignment for external base speaker and external microphone not working
by Raymond Yau 30 Jul '15

30 Jul '15

> > "... after you replaced speakers.conf and lineout.conf by subwoofer.conf ..." > > I must have missed something again. I was under the impression that changing the configuration files was a workaround when the patch would be insufficient. Later today or tomorrow, I will reread this part and apply changes to the configuration files and report back. Ports: analog-output-speaker: Speakers (priority: 10000) analog-output-headphones: Headphones (priority: 9000, not available) Active Port: analog-output-speaker When you select model=dell-headset-multi, the driver create "Headphone Mic Jack" kctl with type 1 which is SND_JACK_HEADPHONE type 2 is SND_JACK_MICROPHONE type 3 is SND_JACK_HEADSET but hda jack does not use this type https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/include/s… JACK created Headphone Mic, type 1 CTRL: add: Headphone Mic Jack:0 JACK report Headphone Mic, status 0 JACK created Headset Mic Phantom, type 2 CTRL: add: Headset Mic Phantom Jack:0 JACK report Headset Mic Phantom, status 2 CTL Notify: Headset Mic Phantom Jack:0, mask=1 JACK created Internal Mic Phantom, type 2 CTRL: add: Internal Mic Phantom Jack:0 JACK report Internal Mic Phantom, status 2 CTL Notify: Internal Mic Phantom Jack:0, mask=1 JACK created Speaker Front Phantom, type 4 CTRL: add: Speaker Front Phantom Jack:0 JACK report Speaker Front Phantom, status 4 CTL Notify: Speaker Front Phantom Jack:0, mask=1 JACK created External Subwoofer, type 4 CTRL: add: External Subwoofer Jack:0 The problem of hda jack kctls are those retaskable I/O jack controls 1) multi io jacks in those 3 stack desktop 2) headphone/mic jack in asus netbook 3) dell-headset-multi combo jack the role of the jack in 1) change from input to output when "Channel mode" change from "2ch" to "4ch" or "6ch" the role of the jack in 2) and 3) change from Output to Input when "Capture Source" change from Internal Mic/Headset Mic to Headphone Mic In your case, Headphone Mic Jack kctl return the presence of HP jack except when the combo jack is set to support conventional mic (i.e. when the driver change pin ctl of hp pin node 0x15 to zero and set pin ctl of mic pin to IN), The driver already unmute the internal speaker when you change capture source to Headphone Mic The only point is whether pulseaudio still regard this Headphone Mic Jack as the availability of the headphone port when the capture source is Headphone Mic http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/?id=ca4942e89cf462…

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix race between PM ops and HDA init/probe
by U. Artie Eoff 29 Jul '15

29 Jul '15

PM ops could be triggered before HDA is done initializing and cause PM to set HDA controller to D3Hot. This can result in "CORB reset timeout#2, CORBRP = 65535" and "no codecs initialized". Additionally, PM ops can be triggered before azx_probe_continue finishes (async probe). This can result in a NULL deref kernel crash. To fix this, avoid PM ops if !chip->running. Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> --- sound/pci/hda/hda_intel.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735bdcb04ce8..c38c68f57938 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -867,7 +867,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) + if (chip->disabled || hda->init_failed || !chip->running) return 0; bus = azx_bus(chip); @@ -902,7 +902,7 @@ static int azx_resume(struct device *dev) chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) + if (chip->disabled || hda->init_failed || !chip->running) return 0; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL @@ -1027,7 +1027,7 @@ static int azx_runtime_idle(struct device *dev) return 0; if (!power_save_controller || !azx_has_pm_runtime(chip) || - azx_bus(chip)->codec_powered) + azx_bus(chip)->codec_powered || !chip->running) return -EBUSY; return 0; -- 2.1.0

3 3

[alsa-devel] [PATCH v2] ASoC: rt5677: Return error if devm_gpiod_get_optional return ERR_PTR
by Axel Lin 29 Jul '15

29 Jul '15

If devm_gpiod_get_optional() return ERR_PTR, it means something wrong so request gpio fails. We had better return error in such case. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rt5677.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 8d70911..83e25a3 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5133,14 +5133,14 @@ static int rt5677_i2c_probe(struct i2c_client *i2c, if (IS_ERR(rt5677->pow_ldo2)) { ret = PTR_ERR(rt5677->pow_ldo2); dev_err(&i2c->dev, "Failed to request POW_LDO2: %d\n", ret); - rt5677->pow_ldo2 = 0; + return ret; } rt5677->reset_pin = devm_gpiod_get_optional(&i2c->dev, "realtek,reset", GPIOD_OUT_HIGH); if (IS_ERR(rt5677->reset_pin)) { ret = PTR_ERR(rt5677->reset_pin); dev_err(&i2c->dev, "Failed to request RESET: %d\n", ret); - rt5677->reset_pin = 0; + return ret; } if (rt5677->pow_ldo2 || rt5677->reset_pin) { -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: tas2552: Remove unneeded NULL test for tas2552->enable_gpio
by Axel Lin 29 Jul '15

29 Jul '15

It's safe to call gpiod_set_value() with NULL desc. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/tas2552.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 2f6a65a..e3a0bca 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -493,8 +493,7 @@ static int tas2552_runtime_suspend(struct device *dev) regcache_cache_only(tas2552->regmap, true); regcache_mark_dirty(tas2552->regmap); - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 0); + gpiod_set_value(tas2552->enable_gpio, 0); return 0; } @@ -503,8 +502,7 @@ static int tas2552_runtime_resume(struct device *dev) { struct tas2552_data *tas2552 = dev_get_drvdata(dev); - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 1); + gpiod_set_value(tas2552->enable_gpio, 1); tas2552_sw_shutdown(tas2552, 0); @@ -585,8 +583,7 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec) return ret; } - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 1); + gpiod_set_value(tas2552->enable_gpio, 1); ret = pm_runtime_get_sync(codec->dev); if (ret < 0) { @@ -610,8 +607,7 @@ static int tas2552_codec_probe(struct snd_soc_codec *codec) return 0; probe_fail: - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 0); + gpiod_set_value(tas2552->enable_gpio, 0); regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), tas2552->supplies); @@ -624,8 +620,7 @@ static int tas2552_codec_remove(struct snd_soc_codec *codec) pm_runtime_put(codec->dev); - if (tas2552->enable_gpio) - gpiod_set_value(tas2552->enable_gpio, 0); + gpiod_set_value(tas2552->enable_gpio, 0); return 0; }; -- 2.1.0

3 2

[alsa-devel] [RFT][PATCH 1/2] ASoC: max98088: Get rid of max98088_access table
by Axel Lin 29 Jul '15

29 Jul '15

The max98088_access table is used for look up readable/writable/volatile attributes of registers. The readable/writable/volatile registers are mostly in continuous ranges, so we can replace the max98088_access table entirely by using case range. Below is a summary of the readable/writeable/volatile registers: readable registers: 0x00 ~ 0xC9, 0xFF writeable registers: 0x03 ~ 0xC9, 0xFF volatile registers: 0x00 ~ 0x03, 0xFF Note, 0x00 should be read-only according to the datasheet. This patch reworks the implement for .readable and .volatile and also add implementation for .writable callback. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98088.c | 305 ++++---------------------------------------- 1 file changed, 25 insertions(+), 280 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 99c2daa..2c2df17 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -258,292 +258,36 @@ static const struct reg_default max98088_reg[] = { { 0xc9, 0x00 }, /* C9 DAI2 biquad */ }; -static struct { - int readable; - int writable; - int vol; -} max98088_access[M98088_REG_CNT] = { - { 0xFF, 0xFF, 1 }, /* 00 IRQ status */ - { 0xFF, 0x00, 1 }, /* 01 MIC status */ - { 0xFF, 0x00, 1 }, /* 02 jack status */ - { 0x1F, 0x1F, 1 }, /* 03 battery voltage */ - { 0xFF, 0xFF, 0 }, /* 04 */ - { 0xFF, 0xFF, 0 }, /* 05 */ - { 0xFF, 0xFF, 0 }, /* 06 */ - { 0xFF, 0xFF, 0 }, /* 07 */ - { 0xFF, 0xFF, 0 }, /* 08 */ - { 0xFF, 0xFF, 0 }, /* 09 */ - { 0xFF, 0xFF, 0 }, /* 0A */ - { 0xFF, 0xFF, 0 }, /* 0B */ - { 0xFF, 0xFF, 0 }, /* 0C */ - { 0xFF, 0xFF, 0 }, /* 0D */ - { 0xFF, 0xFF, 0 }, /* 0E */ - { 0xFF, 0xFF, 0 }, /* 0F interrupt enable */ - - { 0xFF, 0xFF, 0 }, /* 10 master clock */ - { 0xFF, 0xFF, 0 }, /* 11 DAI1 clock mode */ - { 0xFF, 0xFF, 0 }, /* 12 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 13 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 14 DAI1 format */ - { 0xFF, 0xFF, 0 }, /* 15 DAI1 clock */ - { 0xFF, 0xFF, 0 }, /* 16 DAI1 config */ - { 0xFF, 0xFF, 0 }, /* 17 DAI1 TDM */ - { 0xFF, 0xFF, 0 }, /* 18 DAI1 filters */ - { 0xFF, 0xFF, 0 }, /* 19 DAI2 clock mode */ - { 0xFF, 0xFF, 0 }, /* 1A DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1B DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1C DAI2 format */ - { 0xFF, 0xFF, 0 }, /* 1D DAI2 clock */ - { 0xFF, 0xFF, 0 }, /* 1E DAI2 config */ - { 0xFF, 0xFF, 0 }, /* 1F DAI2 TDM */ - - { 0xFF, 0xFF, 0 }, /* 20 DAI2 filters */ - { 0xFF, 0xFF, 0 }, /* 21 data config */ - { 0xFF, 0xFF, 0 }, /* 22 DAC mixer */ - { 0xFF, 0xFF, 0 }, /* 23 left ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 24 right ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 25 left HP mixer */ - { 0xFF, 0xFF, 0 }, /* 26 right HP mixer */ - { 0xFF, 0xFF, 0 }, /* 27 HP control */ - { 0xFF, 0xFF, 0 }, /* 28 left REC mixer */ - { 0xFF, 0xFF, 0 }, /* 29 right REC mixer */ - { 0xFF, 0xFF, 0 }, /* 2A REC control */ - { 0xFF, 0xFF, 0 }, /* 2B left SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2C right SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2D SPK control */ - { 0xFF, 0xFF, 0 }, /* 2E sidetone */ - { 0xFF, 0xFF, 0 }, /* 2F DAI1 playback level */ - - { 0xFF, 0xFF, 0 }, /* 30 DAI1 playback level */ - { 0xFF, 0xFF, 0 }, /* 31 DAI2 playback level */ - { 0xFF, 0xFF, 0 }, /* 32 DAI2 playbakc level */ - { 0xFF, 0xFF, 0 }, /* 33 left ADC level */ - { 0xFF, 0xFF, 0 }, /* 34 right ADC level */ - { 0xFF, 0xFF, 0 }, /* 35 MIC1 level */ - { 0xFF, 0xFF, 0 }, /* 36 MIC2 level */ - { 0xFF, 0xFF, 0 }, /* 37 INA level */ - { 0xFF, 0xFF, 0 }, /* 38 INB level */ - { 0xFF, 0xFF, 0 }, /* 39 left HP volume */ - { 0xFF, 0xFF, 0 }, /* 3A right HP volume */ - { 0xFF, 0xFF, 0 }, /* 3B left REC volume */ - { 0xFF, 0xFF, 0 }, /* 3C right REC volume */ - { 0xFF, 0xFF, 0 }, /* 3D left SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3E right SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3F MIC config */ - - { 0xFF, 0xFF, 0 }, /* 40 MIC threshold */ - { 0xFF, 0xFF, 0 }, /* 41 excursion limiter filter */ - { 0xFF, 0xFF, 0 }, /* 42 excursion limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 43 ALC */ - { 0xFF, 0xFF, 0 }, /* 44 power limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 45 power limiter config */ - { 0xFF, 0xFF, 0 }, /* 46 distortion limiter config */ - { 0xFF, 0xFF, 0 }, /* 47 audio input */ - { 0xFF, 0xFF, 0 }, /* 48 microphone */ - { 0xFF, 0xFF, 0 }, /* 49 level control */ - { 0xFF, 0xFF, 0 }, /* 4A bypass switches */ - { 0xFF, 0xFF, 0 }, /* 4B jack detect */ - { 0xFF, 0xFF, 0 }, /* 4C input enable */ - { 0xFF, 0xFF, 0 }, /* 4D output enable */ - { 0xFF, 0xFF, 0 }, /* 4E bias control */ - { 0xFF, 0xFF, 0 }, /* 4F DAC power */ - - { 0xFF, 0xFF, 0 }, /* 50 DAC power */ - { 0xFF, 0xFF, 0 }, /* 51 system */ - { 0xFF, 0xFF, 0 }, /* 52 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 53 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 54 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 55 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 56 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 57 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 58 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 59 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5A DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5B DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5C DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5D DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5E DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5F DAI1 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 60 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 61 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 62 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 63 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 64 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 65 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 66 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 67 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 68 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 69 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6A DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6B DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6C DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6D DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6E DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6F DAI1 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* 70 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 71 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 72 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 73 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 74 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 75 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 76 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 77 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 78 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 79 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 7A DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7B DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7C DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7D DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7E DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7F DAI1 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* 80 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 81 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 82 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 83 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 84 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 85 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 86 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 87 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 88 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 89 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8A DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8B DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8C DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8D DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8E DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 8F DAI2 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 90 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 91 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 92 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 93 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 94 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 95 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 96 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 97 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 98 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 99 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9A DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9B DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9C DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9D DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9E DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9F DAI2 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* A0 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A1 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A2 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A3 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A4 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A5 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A6 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A7 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A8 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A9 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AA DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AB DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AC DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AD DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AE DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AF DAI2 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* B0 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B1 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B2 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B3 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B4 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B5 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B6 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B7 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B8 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B9 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BA DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BB DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BC DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BD DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BE DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BF DAI1 biquad */ - - { 0xFF, 0xFF, 0 }, /* C0 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C1 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C2 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C3 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C4 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C5 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C6 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C7 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C8 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C9 DAI2 biquad */ - { 0x00, 0x00, 0 }, /* CA */ - { 0x00, 0x00, 0 }, /* CB */ - { 0x00, 0x00, 0 }, /* CC */ - { 0x00, 0x00, 0 }, /* CD */ - { 0x00, 0x00, 0 }, /* CE */ - { 0x00, 0x00, 0 }, /* CF */ - - { 0x00, 0x00, 0 }, /* D0 */ - { 0x00, 0x00, 0 }, /* D1 */ - { 0x00, 0x00, 0 }, /* D2 */ - { 0x00, 0x00, 0 }, /* D3 */ - { 0x00, 0x00, 0 }, /* D4 */ - { 0x00, 0x00, 0 }, /* D5 */ - { 0x00, 0x00, 0 }, /* D6 */ - { 0x00, 0x00, 0 }, /* D7 */ - { 0x00, 0x00, 0 }, /* D8 */ - { 0x00, 0x00, 0 }, /* D9 */ - { 0x00, 0x00, 0 }, /* DA */ - { 0x00, 0x00, 0 }, /* DB */ - { 0x00, 0x00, 0 }, /* DC */ - { 0x00, 0x00, 0 }, /* DD */ - { 0x00, 0x00, 0 }, /* DE */ - { 0x00, 0x00, 0 }, /* DF */ - - { 0x00, 0x00, 0 }, /* E0 */ - { 0x00, 0x00, 0 }, /* E1 */ - { 0x00, 0x00, 0 }, /* E2 */ - { 0x00, 0x00, 0 }, /* E3 */ - { 0x00, 0x00, 0 }, /* E4 */ - { 0x00, 0x00, 0 }, /* E5 */ - { 0x00, 0x00, 0 }, /* E6 */ - { 0x00, 0x00, 0 }, /* E7 */ - { 0x00, 0x00, 0 }, /* E8 */ - { 0x00, 0x00, 0 }, /* E9 */ - { 0x00, 0x00, 0 }, /* EA */ - { 0x00, 0x00, 0 }, /* EB */ - { 0x00, 0x00, 0 }, /* EC */ - { 0x00, 0x00, 0 }, /* ED */ - { 0x00, 0x00, 0 }, /* EE */ - { 0x00, 0x00, 0 }, /* EF */ - - { 0x00, 0x00, 0 }, /* F0 */ - { 0x00, 0x00, 0 }, /* F1 */ - { 0x00, 0x00, 0 }, /* F2 */ - { 0x00, 0x00, 0 }, /* F3 */ - { 0x00, 0x00, 0 }, /* F4 */ - { 0x00, 0x00, 0 }, /* F5 */ - { 0x00, 0x00, 0 }, /* F6 */ - { 0x00, 0x00, 0 }, /* F7 */ - { 0x00, 0x00, 0 }, /* F8 */ - { 0x00, 0x00, 0 }, /* F9 */ - { 0x00, 0x00, 0 }, /* FA */ - { 0x00, 0x00, 0 }, /* FB */ - { 0x00, 0x00, 0 }, /* FC */ - { 0x00, 0x00, 0 }, /* FD */ - { 0x00, 0x00, 0 }, /* FE */ - { 0xFF, 0x00, 1 }, /* FF */ -}; - static bool max98088_readable_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].readable; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... 0xC9: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } +} + +static bool max98088_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98088_REG_03_BATTERY_VOLTAGE ... 0xC9: + return true; + default: + return false; + } } static bool max98088_volatile_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].vol; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... M98088_REG_03_BATTERY_VOLTAGE: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } } static const struct regmap_config max98088_regmap = { @@ -551,6 +295,7 @@ static const struct regmap_config max98088_regmap = { .val_bits = 8, .readable_reg = max98088_readable_register, + .writeable_reg = max98088_writeable_register, .volatile_reg = max98088_volatile_register, .max_register = 0xff, -- 2.1.0

2 3

[alsa-devel] [PATCH] ASoC: wm8983: Get rid of wm8983_access_masks table
by Axel Lin 29 Jul '15

29 Jul '15

The max8983_access table is used for look up readable/writable attributes of registers. The writable registers are mostly in continuous ranges, so we can replace max8983_access table by using case range. The read fields are all 0, so just drop implement of .readable callback. Also set .max_register setting. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/wm8983.c | 80 +++++++++-------------------------------------- 1 file changed, 14 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index f6861cc..f3193fb 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -84,66 +84,6 @@ static const struct reg_default wm8983_defaults[] = { { 0x3D, 0x0000 }, /* R61 - BIAS CTRL */ }; -static const struct wm8983_reg_access { - u16 read; /* Mask of readable bits */ - u16 write; /* Mask of writable bits */ -} wm8983_access_masks[WM8983_MAX_REGISTER + 1] = { - [0x00] = { 0x0000, 0x01FF }, /* R0 - Software Reset */ - [0x01] = { 0x0000, 0x01FF }, /* R1 - Power management 1 */ - [0x02] = { 0x0000, 0x01FF }, /* R2 - Power management 2 */ - [0x03] = { 0x0000, 0x01EF }, /* R3 - Power management 3 */ - [0x04] = { 0x0000, 0x01FF }, /* R4 - Audio Interface */ - [0x05] = { 0x0000, 0x003F }, /* R5 - Companding control */ - [0x06] = { 0x0000, 0x01FD }, /* R6 - Clock Gen control */ - [0x07] = { 0x0000, 0x000F }, /* R7 - Additional control */ - [0x08] = { 0x0000, 0x003F }, /* R8 - GPIO Control */ - [0x09] = { 0x0000, 0x0070 }, /* R9 - Jack Detect Control 1 */ - [0x0A] = { 0x0000, 0x004F }, /* R10 - DAC Control */ - [0x0B] = { 0x0000, 0x01FF }, /* R11 - Left DAC digital Vol */ - [0x0C] = { 0x0000, 0x01FF }, /* R12 - Right DAC digital vol */ - [0x0D] = { 0x0000, 0x00FF }, /* R13 - Jack Detect Control 2 */ - [0x0E] = { 0x0000, 0x01FB }, /* R14 - ADC Control */ - [0x0F] = { 0x0000, 0x01FF }, /* R15 - Left ADC Digital Vol */ - [0x10] = { 0x0000, 0x01FF }, /* R16 - Right ADC Digital Vol */ - [0x12] = { 0x0000, 0x017F }, /* R18 - EQ1 - low shelf */ - [0x13] = { 0x0000, 0x017F }, /* R19 - EQ2 - peak 1 */ - [0x14] = { 0x0000, 0x017F }, /* R20 - EQ3 - peak 2 */ - [0x15] = { 0x0000, 0x017F }, /* R21 - EQ4 - peak 3 */ - [0x16] = { 0x0000, 0x007F }, /* R22 - EQ5 - high shelf */ - [0x18] = { 0x0000, 0x01FF }, /* R24 - DAC Limiter 1 */ - [0x19] = { 0x0000, 0x007F }, /* R25 - DAC Limiter 2 */ - [0x1B] = { 0x0000, 0x01FF }, /* R27 - Notch Filter 1 */ - [0x1C] = { 0x0000, 0x017F }, /* R28 - Notch Filter 2 */ - [0x1D] = { 0x0000, 0x017F }, /* R29 - Notch Filter 3 */ - [0x1E] = { 0x0000, 0x017F }, /* R30 - Notch Filter 4 */ - [0x20] = { 0x0000, 0x01BF }, /* R32 - ALC control 1 */ - [0x21] = { 0x0000, 0x00FF }, /* R33 - ALC control 2 */ - [0x22] = { 0x0000, 0x01FF }, /* R34 - ALC control 3 */ - [0x23] = { 0x0000, 0x000F }, /* R35 - Noise Gate */ - [0x24] = { 0x0000, 0x001F }, /* R36 - PLL N */ - [0x25] = { 0x0000, 0x003F }, /* R37 - PLL K 1 */ - [0x26] = { 0x0000, 0x01FF }, /* R38 - PLL K 2 */ - [0x27] = { 0x0000, 0x01FF }, /* R39 - PLL K 3 */ - [0x29] = { 0x0000, 0x000F }, /* R41 - 3D control */ - [0x2A] = { 0x0000, 0x01E7 }, /* R42 - OUT4 to ADC */ - [0x2B] = { 0x0000, 0x01BF }, /* R43 - Beep control */ - [0x2C] = { 0x0000, 0x0177 }, /* R44 - Input ctrl */ - [0x2D] = { 0x0000, 0x01FF }, /* R45 - Left INP PGA gain ctrl */ - [0x2E] = { 0x0000, 0x01FF }, /* R46 - Right INP PGA gain ctrl */ - [0x2F] = { 0x0000, 0x0177 }, /* R47 - Left ADC BOOST ctrl */ - [0x30] = { 0x0000, 0x0177 }, /* R48 - Right ADC BOOST ctrl */ - [0x31] = { 0x0000, 0x007F }, /* R49 - Output ctrl */ - [0x32] = { 0x0000, 0x01FF }, /* R50 - Left mixer ctrl */ - [0x33] = { 0x0000, 0x01FF }, /* R51 - Right mixer ctrl */ - [0x34] = { 0x0000, 0x01FF }, /* R52 - LOUT1 (HP) volume ctrl */ - [0x35] = { 0x0000, 0x01FF }, /* R53 - ROUT1 (HP) volume ctrl */ - [0x36] = { 0x0000, 0x01FF }, /* R54 - LOUT2 (SPK) volume ctrl */ - [0x37] = { 0x0000, 0x01FF }, /* R55 - ROUT2 (SPK) volume ctrl */ - [0x38] = { 0x0000, 0x004F }, /* R56 - OUT3 mixer ctrl */ - [0x39] = { 0x0000, 0x00FF }, /* R57 - OUT4 (MONO) mix ctrl */ - [0x3D] = { 0x0000, 0x0100 } /* R61 - BIAS CTRL */ -}; - /* vol/gain update regs */ static const int vol_update_regs[] = { WM8983_LEFT_DAC_DIGITAL_VOL, @@ -605,12 +545,19 @@ static int eqmode_put(struct snd_kcontrol *kcontrol, return 0; } -static bool wm8983_readable(struct device *dev, unsigned int reg) +static bool wm8983_writeable(struct device *dev, unsigned int reg) { - if (reg > WM8983_MAX_REGISTER) - return 0; - - return wm8983_access_masks[reg].read != 0; + switch (reg) { + case WM8983_SOFTWARE_RESET ... WM8983_RIGHT_ADC_DIGITAL_VOL: + case WM8983_EQ1_LOW_SHELF ... WM8983_DAC_LIMITER_2: + case WM8983_NOTCH_FILTER_1 ... WM8983_NOTCH_FILTER_4: + case WM8983_ALC_CONTROL_1 ... WM8983_PLL_K_3: + case WM8983_3D_CONTROL ... WM8983_OUT4_MONO_MIX_CTRL: + case WM8983_BIAS_CTRL: + return true; + default: + return false; + } } static int wm8983_dac_mute(struct snd_soc_dai *dai, int mute) @@ -1048,8 +995,9 @@ static const struct regmap_config wm8983_regmap = { .reg_defaults = wm8983_defaults, .num_reg_defaults = ARRAY_SIZE(wm8983_defaults), .cache_type = REGCACHE_RBTREE, + .max_register = WM8983_MAX_REGISTER, - .readable_reg = wm8983_readable, + .writeable_reg = wm8983_writeable, }; #if defined(CONFIG_SPI_MASTER) -- 2.1.0

2 1

[alsa-devel] [PATCH 1/2] ASoC: Intel: haswell: fix initialize 'NULL device *' issue
by Jie Yang 29 Jul '15

29 Jul '15

Add initialization for sst_hsw.dev at init stage, which fix the 'NULL device *' warning issues. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index f95f271..f6efa9d 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -2119,6 +2119,8 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) if (hsw == NULL) return -ENOMEM; + hsw->dev = dev; + ipc = &hsw->ipc; ipc->dev = dev; ipc->ops.tx_msg = hsw_tx_msg; -- 1.9.1

2 3

[alsa-devel] [PATCH 0/6] ASoC: Fix sparse warnings
by Lars-Peter Clausen 29 Jul '15

29 Jul '15

Hi, This series fixes a couple of sparse warnings that have appeared recently in the ASoC tree. Non of them are critical mostly just missing annotations or storage qualifiers. It still makes sense to fix those though since it reduces the noise that has to be filtered through when checking new code for new (potentially critical) warnings. - Lars Lars-Peter Clausen (6): ASoC: wm8804: Drop dublicate const ASoC: lpass-ipq806x: Staticise local symbols ASoC: rcar ctu: Staticise local symbols ASoC: uniperf: Add missing __iomem annotation ASoC: uniperf: Staticise local symbols ASoC: xtfpga-i2s: Add missing __rcu annotation sound/soc/codecs/wm8804.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/sh/rcar/ctu.c | 2 +- sound/soc/sti/uniperif.h | 2 +- sound/soc/sti/uniperif_player.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) -- 2.1.4

7 18

[alsa-devel] [PATCH] ASoC: max98090: Simplify max98090_readable_register implementation
by Axel Lin 29 Jul '15

29 Jul '15

The readable registers are in consecutive ranges: 0x01 ~ 0x03, 0x0D ~ 0xD1, 0xFF So simplify the implementation by specifying a range of consecutive values in a single case label. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98090.c | 71 ++------------------------------------------- 1 file changed, 2 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 91c9f9f..315c71f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -267,75 +267,8 @@ static bool max98090_volatile_register(struct device *dev, unsigned int reg) static bool max98090_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case M98090_REG_DEVICE_STATUS: - case M98090_REG_JACK_STATUS: - case M98090_REG_INTERRUPT_S: - case M98090_REG_RESERVED: - case M98090_REG_LINE_INPUT_CONFIG: - case M98090_REG_LINE_INPUT_LEVEL: - case M98090_REG_INPUT_MODE: - case M98090_REG_MIC1_INPUT_LEVEL: - case M98090_REG_MIC2_INPUT_LEVEL: - case M98090_REG_MIC_BIAS_VOLTAGE: - case M98090_REG_DIGITAL_MIC_ENABLE: - case M98090_REG_DIGITAL_MIC_CONFIG: - case M98090_REG_LEFT_ADC_MIXER: - case M98090_REG_RIGHT_ADC_MIXER: - case M98090_REG_LEFT_ADC_LEVEL: - case M98090_REG_RIGHT_ADC_LEVEL: - case M98090_REG_ADC_BIQUAD_LEVEL: - case M98090_REG_ADC_SIDETONE: - case M98090_REG_SYSTEM_CLOCK: - case M98090_REG_CLOCK_MODE: - case M98090_REG_CLOCK_RATIO_NI_MSB: - case M98090_REG_CLOCK_RATIO_NI_LSB: - case M98090_REG_CLOCK_RATIO_MI_MSB: - case M98090_REG_CLOCK_RATIO_MI_LSB: - case M98090_REG_MASTER_MODE: - case M98090_REG_INTERFACE_FORMAT: - case M98090_REG_TDM_CONTROL: - case M98090_REG_TDM_FORMAT: - case M98090_REG_IO_CONFIGURATION: - case M98090_REG_FILTER_CONFIG: - case M98090_REG_DAI_PLAYBACK_LEVEL: - case M98090_REG_DAI_PLAYBACK_LEVEL_EQ: - case M98090_REG_LEFT_HP_MIXER: - case M98090_REG_RIGHT_HP_MIXER: - case M98090_REG_HP_CONTROL: - case M98090_REG_LEFT_HP_VOLUME: - case M98090_REG_RIGHT_HP_VOLUME: - case M98090_REG_LEFT_SPK_MIXER: - case M98090_REG_RIGHT_SPK_MIXER: - case M98090_REG_SPK_CONTROL: - case M98090_REG_LEFT_SPK_VOLUME: - case M98090_REG_RIGHT_SPK_VOLUME: - case M98090_REG_DRC_TIMING: - case M98090_REG_DRC_COMPRESSOR: - case M98090_REG_DRC_EXPANDER: - case M98090_REG_DRC_GAIN: - case M98090_REG_RCV_LOUTL_MIXER: - case M98090_REG_RCV_LOUTL_CONTROL: - case M98090_REG_RCV_LOUTL_VOLUME: - case M98090_REG_LOUTR_MIXER: - case M98090_REG_LOUTR_CONTROL: - case M98090_REG_LOUTR_VOLUME: - case M98090_REG_JACK_DETECT: - case M98090_REG_INPUT_ENABLE: - case M98090_REG_OUTPUT_ENABLE: - case M98090_REG_LEVEL_CONTROL: - case M98090_REG_DSP_FILTER_ENABLE: - case M98090_REG_BIAS_CONTROL: - case M98090_REG_DAC_CONTROL: - case M98090_REG_ADC_CONTROL: - case M98090_REG_DEVICE_SHUTDOWN: - case M98090_REG_EQUALIZER_BASE ... M98090_REG_EQUALIZER_BASE + 0x68: - case M98090_REG_RECORD_BIQUAD_BASE ... M98090_REG_RECORD_BIQUAD_BASE + 0x0E: - case M98090_REG_DMIC3_VOLUME: - case M98090_REG_DMIC4_VOLUME: - case M98090_REG_DMIC34_BQ_PREATTEN: - case M98090_REG_RECORD_TDM_SLOT: - case M98090_REG_SAMPLE_RATE: - case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + case M98090_REG_DEVICE_STATUS ... M98090_REG_INTERRUPT_S: + case M98090_REG_LINE_INPUT_CONFIG ... 0xD1: case M98090_REG_REVISION_ID: return true; default: -- 2.1.0

3 3

[alsa-devel] [PATCH] sound/usb: add dB range mapping for some devices.
by Yao-Wen Mao 29 Jul '15

29 Jul '15

Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2. Signed-off-by: Yao-Wen Mao <yaowen(a)google.com> --- sound/usb/mixer_maps.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index e5000da..f0c9e49 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -339,6 +339,19 @@ static struct usbmix_name_map gamecom780_map[] = { static const struct usbmix_name_map scms_usb3318_map[] = { { 10, NULL }, { 0 } + +/* Bose companion 5, the dB conversion factor is 16 instead of 256 */ +static struct usbmix_dB_map bose_companion5_dB = {-5006, -6}; +static struct usbmix_name_map bose_companion5_map[] = { + { 3, NULL, .dB = &bose_companion5_dB }, + { 0 } /* terminator */ +}; + +/* Dragonfly DAC 1.2, the dB conversion factor is 1 instead of 256 */ +static struct usbmix_dB_map dragonfly_1_2_dB = {0, 5000}; +static struct usbmix_name_map dragonfly_1_2_map[] = { + { 7, NULL, .dB = &dragonfly_1_2_dB }, + { 0 } /* terminator */ }; /* @@ -451,6 +464,16 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x25c4, 0x0003), .map = scms_usb3318_map, }, + { + /* Bose Companion 5 */ + .id = USB_ID(0x05a7, 0x1020), + .map = bose_companion5_map, + }, + { + /* Dragonfly DAC 1.2 */ + .id = USB_ID(0x21b4, 0x0081), + .map = dragonfly_1_2_map, + }, { 0 } /* terminator */ }; -- 2.5.0.rc2.392.g76e840b

2 2

Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel driver
by Paulo Roberto de Oliveira Castro 29 Jul '15

29 Jul '15

I'm not an expert on applying patches to the kernel (never did that before), therefore, I don't know if I did everything right. What I did was: downloaded 4.2-rc4 from kernel.org[1], added the line by hand (couldn't apply the patch using patch), copied the config file from /boot/ (the config fedora 22 uses for its kernel) to .config on the root of the kernel directory, runned "make oldconfig" (typing ENTER for every question, is that right?), "make", "make modules_install", "make install". Rebooted, and chose the new kernel I just installed. The sound worked like a charm, the speakers worked as expected, and the volume was as loud as on Windows. Is the patch going to be applied on master? Any chance that fix reaches my Fedora 22 (with backport to older kernel versions)? Thanks for the help, Paulo [1] https://www.kernel.org/pub/linux/kernel/v4.x/testing/linux-4.2-rc4.tar.xz 2015-07-28 10:49 GMT-03:00 Paulo Roberto de Oliveira Castro < p.oliveira.castro(a)gmail.com>: > > > ---------- Forwarded message --------- > From: Takashi Iwai <tiwai(a)suse.de> > Date: Tue, Jul 28, 2015, 03:35 > Subject: Re: [alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel > driver > To: Paulo Roberto de Oliveira Castro <p.oliveira.castro(a)gmail.com> > Cc: <alsa-devel(a)alsa-project.org> > > > On Tue, 28 Jul 2015 05:35:34 +0200, > Paulo Roberto de Oliveira Castro wrote: > > > > Hi everyone, > > > > After installing Fedora 22 on my Dell Vostro 5480, the speakers don't > work, > > and the sound only comes out of the subwoofer. Here is the output of > > alsa-info.sh. After talking to "debianuser" on IRC, we found this > > workaround: > > > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_PIN_WIDGET_CONTROL 0x40 # enable, > > set it to OUT > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_CONNECT_SEL 0 # switch to 0x0c > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0xa000 # unmute left > > sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0x9000 # unmute > right > > > > We suspect that it's a kernel driver bug. This makes the speakers work > > alongside the headphones (both at the same time) but I need help making > it > > working when headphones are not plugged in. > > Does the patch below work? Try with 4.2-rc kernel at best. > > > Takashi > > --- > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 4ae877c3b6a1..18ae17ebb356 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -5185,6 +5185,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] > = { > SND_PCI_QUIRK(0x1028, 0x064a, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x064b, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", > ALC288_FIXUP_DELL_XPS_13), > + SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", > ALC290_FIXUP_SUBWOOFER_HSJACK), > SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", > ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x06da, "Dell", > ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), >

2 1

[alsa-devel] [PATCH] ASoC: dapm: fix module unload/reload failed issue by reset kcontrol's widget list
by Jie Yang 29 Jul '15

29 Jul '15

We need reset the related kcontrol's widget list during the freeing of a widget, otherwise, the widget list may be outdated and reference to it may introduce errors(they usually occurs during driver modules unloading/reloading). Here adding a func dapm_update_kcontrols_of_freeing_widget and call it at dapm_update_kcontrols_of_freeing_widget, to fix those issues. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/soc-dapm.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..1ad8144 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2305,6 +2305,35 @@ static void dapm_free_path(struct snd_soc_dapm_path *path) kfree(path); } +/* update all dapm kcontrols that related to a widget which being freed*/ +static void dapm_update_kcontrols_of_freeing_widget(struct snd_soc_dapm_widget * w) +{ + struct dapm_kcontrol_data *data = NULL; + int i, j, n; + + if (w && w->kcontrols) { + for (i = 0; i < w->num_kcontrols; i++) { + if (w->kcontrols[i] == NULL) + continue; + data = snd_kcontrol_chip(w->kcontrols[i]); + if (data == NULL) + continue; + for (j = 0, n = 0; j < data->wlist->num_widgets; j++) { + if (data->wlist->widgets[j] == w) { + data->wlist->widgets[j] = NULL; + n++; + } + } + + if (n) { + data->wlist->num_widgets -= n; + data->widget = NULL; + } + } + } + +} + /* free all dapm widgets and resources */ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) { @@ -2326,6 +2355,8 @@ static void dapm_free_widgets(struct snd_soc_dapm_context *dapm) list_for_each_entry_safe(p, next_p, &w->sinks, list_source) dapm_free_path(p); + dapm_update_kcontrols_of_freeing_widget(w); + kfree(w->kcontrols); kfree(w->name); kfree(w); -- 1.9.1

4 4

[alsa-devel] [PATCH 1/2] ALSA: hda - skip runtime PM if async probe is not finished
by U. Artie Eoff 28 Jul '15

28 Jul '15

Crash can occur if runtime PM is triggered before the async probe finishes via the azx_firmware_cb when CONFIG_SND_HDA_PATCH_LOADER is defined. Don't execute PM ops if the async probe has not completed yet. The probe is finished when chip->running is true. Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 735bdcb04ce8..b729b25a6ad6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1023,7 +1023,7 @@ static int azx_runtime_idle(struct device *dev) chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); - if (chip->disabled || hda->init_failed) + if (chip->disabled || hda->init_failed || !chip->running) return 0; if (!power_save_controller || !azx_has_pm_runtime(chip) || -- 1.9.3

4 9

[alsa-devel] [PATCH v13 0/3] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 28 Jul '15

28 Jul '15

This patch series asks for Russell King's patch http://article.gmane.org/gmane.linux.ports.arm.kernel/411574 drm/edid: add function to help find SADs to be applayed. v13: - rebase on 4.2.0-rc3 - remove Russell's patches v12: - use Russell King's DRM ELD helper (Mark Brown) v11: - reduce the patch series to adding the tda998x codec only v10: - add the generic dt-card - define the audio ports from a DT graph of ports (Russell King) - reuse HDMI constants (Andrew Jackson - Jyri Sarha) - alloc rate_constraints in codec (Jyri Sarha) - fix bad number of channels (Jyri Sarha) - correct codec generation from config (Russell King - Jyri Sarha) - no module init/exit (Russell King) v9: - back to a TDA998x specific CODEC - more comments - change magic values to constants v8: - change some comments about the patches v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (3): drm/i2c: tda998x: Add support of a DT graph of ports drm/i2c: tda998x: Change drvdata for audio extension ASoC: tda998x: add a codec to the HDMI transmitter .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++++++ drivers/gpu/drm/i2c/tda998x_drv.c | 178 +++++++++++++++++++-- include/sound/tda998x.h | 23 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda998x.c | 132 +++++++++++++++ 6 files changed, 381 insertions(+), 11 deletions(-) create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/tda998x.c -- 2.1.4

4 12

[alsa-devel] rockchip rk3288 registration
by Michael Trimarchi 28 Jul '15

28 Jul '15

Hi I have created a simple sound card to test registration on i2s but I have some problems that after a while in SND_SOC_DAIFMT_CBM_CFM the device stop to register and receive a input/output error. It means that I can not receive anymore from the device arecord -c 2 -r 96000 -f S32_LE -D hw:0 > record.pcm Recording WAVE 'stdin' : Signed 32 bit Little Endian, Rate 96000 Hz, Stereo arecord: pcm_read:2031: read error: Input/output error The result size is a bit random. I have no problem on playback until now. Any idea? Michael

1 1

[alsa-devel] Running PJSIP for voip calls in alsa
by Srinivasan S 28 Jul '15

28 Jul '15

Dear Clemens, Could you please let me know whether the below issue is in PJSIP or alsa or driver or in the ti sdk8 BSP's (rootfs) 1. We are running PJSIP for establishing VOIP calls on ti sdk 8 BSP's (linux kernel 3.14), we are seeing the below error messages in the PJSIP, Could you please let me know what could be issue ? 08:32:42.082 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.083 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.086 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.088 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.088 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.091 Master/sound Underflow, buf_cnt=0, will generate 1 frame 08:32:42.112 Master/sound Underflow, buf_cnt=0, will generate 1 frame In failure call pjsip reports Master/sound Underflow continuously, At this point the control stays at pjsip and application will wait for it and all functionality in idle state at application and after this, no other feature will work until restart the application. The above problem observed with new rootfs (ti sdk 8) but when we tried with old rootfs (ti sdk 7) we are able to make calls with no problem. Few calls are long duration calls(33 and 55 minutes). With ti sdk 8 kernel, I tried replacing even the alsa libraries from ti sdk 7 rootfs to ti sdk 8 rootfs, but still the issue persists, /usr/lib/libasound.la /usr/lib/libasound.so /usr/lib/alsa-lib/smixer/smixer-ac97.la /usr/lib/alsa-lib/smixer/smixer-ac97.so /usr/lib/alsa-lib/smixer/smixer-hda.la /usr/lib/alsa-lib/smixer/smixer-hda.so /usr/lib/alsa-lib/smixer/smixer-sbase.la /usr/lib/alsa-lib/smixer/smixer-sbase.so Firstly could you please let me know why this issue is seen in ti sdk 8 BSP's (ie., in rootfs), & what could be the problem in ti sdk 8 rootfs ?? Apart from the above is there any other ALSA related configuration files, needs to copied from the ti sdk 7 rootfs to ti sdk 8 rootfs, inorder to resolve this issue If possible, could you please validate this PJSIP version 2.3 on ti sdk 8 BSP's on your EVK kit & let me know whether this issue is reproducible Awaiting for your replies, Many Thanks in advance

3 11

[alsa-devel] [PATCH] sound/usb: add dB range mapping quirks.
by Yao-Wen Mao 28 Jul '15

28 Jul '15

Some usb audio devices don't follow the specification of the volume range. Add dB_mapping_quirks to fix some devices' dB ranges so that the function snd_mixer_selem_set_playback_dB in alsa-lib can set the correct value for those devices. Signed-off-by: Yao-Wen Mao <yaowen(a)google.com> --- sound/usb/mixer.c | 30 ++++++++++++++++++++++++++++++ 1 file changed, 30 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6b3acba..b6b27dc 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -933,6 +933,32 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } } +/* dB mapping quirks */ +static void dB_mapping_quirks(struct usb_mixer_elem_info *cval, + struct snd_kcontrol *kctl) +{ + switch (cval->mixer->chip->usb_id) { + case USB_ID(0x05a7, 0x1020): /* Bose Companion 5 */ + if (strcmp(kctl->id.name, "PCM Playback Volume") == 0) { + cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 16; + cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 16; + } + break; + case USB_ID(0x21b4, 0x0081): /* Dragonfly DAC 1.2 */ + if (strcmp(kctl->id.name, "PCM Playback Volume") == 0) { + cval->dBmin = convert_signed_value(cval, cval->min) * 100; + cval->dBmax = convert_signed_value(cval, cval->max) * 100; + } + break; + case USB_ID(0x046d, 0x0a29): /* Logitech H800 */ + if (strcmp(kctl->id.name, "PCM Playback Volume") == 0) { + cval->dBmin = convert_signed_value(cval, cval->min); + cval->dBmax = convert_signed_value(cval, cval->max); + } + break; + } +} + /* * retrieve the minimum and maximum values for the specified control */ @@ -1027,6 +1053,10 @@ static int get_min_max_with_quirks(struct usb_mixer_elem_info *cval, */ cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 256; cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 256; + + if (kctl) + dB_mapping_quirks(cval, kctl); + if (cval->dBmin > cval->dBmax) { /* something is wrong; assume it's either from/to 0dB */ if (cval->dBmin < 0) -- 2.5.0.rc2.392.g76e840b

2 5

[alsa-devel] Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_jack.so
by Srinivasan S 28 Jul '15

28 Jul '15

Dear Clemens, Am trying to cross compile, could you pls help me out for cross_compiling Am facing some configuration issue as shown below srinivasan@tata-HP-Elite-7100-Microtower-PC:~/alsa-plugins-1.0.29$ ./configure --host=arm-linux CC=/opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc --prefix=/home/srinivasan/alsa-plugins-1.0.29/ checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for arm-linux-strip... no checking for strip... strip checking for a thread-safe mkdir -p... /bin/mkdir -p checking for gawk... gawk checking whether make sets $(MAKE)... yes checking whether to enable maintainer-specific portions of Makefiles... yes checking for arm-linux-gcc... /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc checking whether the C compiler works... yes checking for C compiler default output file name... a.out checking for suffix of executables... checking whether we are cross compiling... yes checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc accepts -g... yes checking for /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc option to accept ISO C89... none needed checking for style of include used by make... GNU checking dependency style of /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc... gcc3 checking build system type... x86_64-unknown-linux-gnu checking host system type... arm-unknown-linux-gnu checking how to print strings... printf checking for a sed that does not truncate output... /bin/sed checking for grep that handles long lines and -e... /bin/grep checking for egrep... /bin/grep -E checking for fgrep... /bin/grep -F checking for ld used by /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc... /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/arm-linux-gnueabihf/bin/ld checking if the linker (/opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/arm-linux-gnueabihf/bin/ld) is GNU ld... yes checking for BSD- or MS-compatible name lister (nm)... no checking for arm-linux-dumpbin... no checking for arm-linux-link... no checking for dumpbin... no checking for link... link -dump configure: WARNING: using cross tools not prefixed with host triplet checking the name lister (nm) interface... BSD nm checking whether ln -s works... yes checking the maximum length of command line arguments... 1572864 checking whether the shell understands some XSI constructs... yes checking whether the shell understands "+="... yes checking how to convert x86_64-unknown-linux-gnu file names to arm-unknown-linux-gnu format... func_convert_file_noop checking how to convert x86_64-unknown-linux-gnu file names to toolchain format... func_convert_file_noop checking for /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/arm-linux-gnueabihf/bin/ld option to reload object files... -r checking for arm-linux-objdump... no checking for objdump... objdump checking how to recognize dependent libraries... pass_all checking for arm-linux-dlltool... no checking for dlltool... no checking how to associate runtime and link libraries... printf %s\n checking for arm-linux-ar... no checking for ar... ar checking for archiver @FILE support... @ checking for arm-linux-strip... strip checking for arm-linux-ranlib... no checking for ranlib... ranlib checking command to parse nm output from /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc object... ok checking for sysroot... no checking for arm-linux-mt... no checking for mt... mt checking if mt is a manifest tool... no checking how to run the C preprocessor... /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc -E checking for ANSI C header files... yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for dlfcn.h... yes checking for objdir... .libs checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc supports -fno-rtti -fno-exceptions... no checking for /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc option to produce PIC... -fPIC -DPIC checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc PIC flag -fPIC -DPIC works... yes checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc static flag -static works... yes checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc supports -c -o file.o... yes checking if /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc supports -c -o file.o... (cached) yes checking whether the /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin/arm-linux-gnueabihf-gcc linker (/opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/arm-linux-gnueabihf/bin/ld) supports shared libraries... yes checking whether -lc should be explicitly linked in... no checking dynamic linker characteristics... GNU/Linux ld.so checking how to hardcode library paths into programs... immediate checking whether stripping libraries is possible... yes checking if libtool supports shared libraries... yes checking whether to build shared libraries... yes checking whether to build static libraries... no checking for ANSI C header files... (cached) yes ./configure: line 11534: CC_NOUNDEFINED: command not found checking for arm-linux-pkg-config... no checking for pkg-config... /opt/ti-sdk-am335x-evm-07.00.00.00/linux-devkit/sysroots/i686-arago-linux/usr/bin//pkg-config checking pkg-config is at least version 0.9.0... yes checking for ALSA... no configure: error: Package requirements (alsa >= 1.0.11) were not met: No package 'alsa' found Consider adjusting the PKG_CONFIG_PATH environment variable if you installed software in a non-standard prefix. Alternatively, you may set the environment variables ALSA_CFLAGS and ALSA_LIBS to avoid the need to call pkg-config. See the pkg-config man page for more details. srinivasan@tata-HP-Elite-7100-Microtower-PC:~/alsa-plugins-1.0.29$ Thanks Srini

2 6

[alsa-devel] pcm prepare
by Andy Ng 28 Jul '15

28 Jul '15

Hi, I am quite surprised that by sending hardware params with alsa-lib the prepare function is called automatically in the driver. In my case, I would like to open and configure pcmNp and pcmNc and then call prepare for pcmNp and pcmPc channels. It is quite imperative to execute specific configuration steps after both channels are opened and configured. I was wondering if there is any ioctl interface that can be hooked in alsa-lib to pass extra configuration info before start the streams. Any ideas? Andy

2 1

[alsa-devel] Bug on Dell Vostro 5480 (Intel Wildcat) kernel driver
by Paulo Roberto de Oliveira Castro 28 Jul '15

28 Jul '15

Hi everyone, After installing Fedora 22 on my Dell Vostro 5480, the speakers don't work, and the sound only comes out of the subwoofer. Here is the output of alsa-info.sh. After talking to "debianuser" on IRC, we found this workaround: sudo hda-verb /dev/snd/hwC1D0 0x14 SET_PIN_WIDGET_CONTROL 0x40 # enable, set it to OUT sudo hda-verb /dev/snd/hwC1D0 0x14 SET_CONNECT_SEL 0 # switch to 0x0c sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0xa000 # unmute left sudo hda-verb /dev/snd/hwC1D0 0x14 SET_AMP_GAIN_MUTE 0x9000 # unmute right We suspect that it's a kernel driver bug. This makes the speakers work alongside the headphones (both at the same time) but I need help making it working when headphones are not plugged in. Thanks, Paulo

2 2

[alsa-devel] [PATCH v2] ALSA: hda - Fix race condition between HDA driver and runtime PM
by U. Artie Eoff 28 Jul '15

28 Jul '15

Don't execute runtime suspend if HDA is not finished initializing. Otherwise, the following errors can occur in hda: snd_hda_intel 0000:00:1b.0: CORB reset timeout#2, CORBRP = 65535 snd_hda_intel 0000:00:1b.0: no codecs initialized Debugged and root-cause found by zhuo-hao.lee(a)intel.com v2: remove unnecessary extra check for chip init (libin) Signed-off-by: U. Artie Eoff <ullysses.a.eoff(a)intel.com> --- sound/pci/hda/hda_intel.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b729b25a6ad6..15d4e28fc64e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1017,17 +1017,19 @@ static int azx_runtime_idle(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; struct hda_intel *hda; + struct hdac_bus *bus; if (!card) return 0; chip = card->private_data; hda = container_of(chip, struct hda_intel, chip); + bus = azx_bus(chip); if (chip->disabled || hda->init_failed || !chip->running) return 0; if (!power_save_controller || !azx_has_pm_runtime(chip) || - azx_bus(chip)->codec_powered) + bus->codec_powered || !bus->chip_init) return -EBUSY; return 0; -- 1.9.3

3 5

[alsa-devel] [PATCH v2 1/2] ASoC: max98088: Get rid of max98088_access table
by Axel Lin 28 Jul '15

28 Jul '15

The max98088_access table is used for look up readable/writable/volatile attributes of registers. The readable/writable/volatile registers are mostly in continuous ranges, so we can replace the max98088_access table entirely by using case range. Below is a summary of the readable/writeable/volatile registers: readable registers: 0x00 ~ 0xC9, 0xFF writeable registers: 0x03 ~ 0xC9 volatile registers: 0x00 ~ 0x03, 0xFF Note, 0x00 should be read-only according to the datasheet. This patch reworks the implement for .readable and .volatile and also add implementation for .writable callback. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- v2: no code change. Just update commit log, 0xff register is not writeable. sound/soc/codecs/max98088.c | 305 ++++---------------------------------------- 1 file changed, 25 insertions(+), 280 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 99c2daa..2c2df17 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -258,292 +258,36 @@ static const struct reg_default max98088_reg[] = { { 0xc9, 0x00 }, /* C9 DAI2 biquad */ }; -static struct { - int readable; - int writable; - int vol; -} max98088_access[M98088_REG_CNT] = { - { 0xFF, 0xFF, 1 }, /* 00 IRQ status */ - { 0xFF, 0x00, 1 }, /* 01 MIC status */ - { 0xFF, 0x00, 1 }, /* 02 jack status */ - { 0x1F, 0x1F, 1 }, /* 03 battery voltage */ - { 0xFF, 0xFF, 0 }, /* 04 */ - { 0xFF, 0xFF, 0 }, /* 05 */ - { 0xFF, 0xFF, 0 }, /* 06 */ - { 0xFF, 0xFF, 0 }, /* 07 */ - { 0xFF, 0xFF, 0 }, /* 08 */ - { 0xFF, 0xFF, 0 }, /* 09 */ - { 0xFF, 0xFF, 0 }, /* 0A */ - { 0xFF, 0xFF, 0 }, /* 0B */ - { 0xFF, 0xFF, 0 }, /* 0C */ - { 0xFF, 0xFF, 0 }, /* 0D */ - { 0xFF, 0xFF, 0 }, /* 0E */ - { 0xFF, 0xFF, 0 }, /* 0F interrupt enable */ - - { 0xFF, 0xFF, 0 }, /* 10 master clock */ - { 0xFF, 0xFF, 0 }, /* 11 DAI1 clock mode */ - { 0xFF, 0xFF, 0 }, /* 12 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 13 DAI1 clock control */ - { 0xFF, 0xFF, 0 }, /* 14 DAI1 format */ - { 0xFF, 0xFF, 0 }, /* 15 DAI1 clock */ - { 0xFF, 0xFF, 0 }, /* 16 DAI1 config */ - { 0xFF, 0xFF, 0 }, /* 17 DAI1 TDM */ - { 0xFF, 0xFF, 0 }, /* 18 DAI1 filters */ - { 0xFF, 0xFF, 0 }, /* 19 DAI2 clock mode */ - { 0xFF, 0xFF, 0 }, /* 1A DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1B DAI2 clock control */ - { 0xFF, 0xFF, 0 }, /* 1C DAI2 format */ - { 0xFF, 0xFF, 0 }, /* 1D DAI2 clock */ - { 0xFF, 0xFF, 0 }, /* 1E DAI2 config */ - { 0xFF, 0xFF, 0 }, /* 1F DAI2 TDM */ - - { 0xFF, 0xFF, 0 }, /* 20 DAI2 filters */ - { 0xFF, 0xFF, 0 }, /* 21 data config */ - { 0xFF, 0xFF, 0 }, /* 22 DAC mixer */ - { 0xFF, 0xFF, 0 }, /* 23 left ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 24 right ADC mixer */ - { 0xFF, 0xFF, 0 }, /* 25 left HP mixer */ - { 0xFF, 0xFF, 0 }, /* 26 right HP mixer */ - { 0xFF, 0xFF, 0 }, /* 27 HP control */ - { 0xFF, 0xFF, 0 }, /* 28 left REC mixer */ - { 0xFF, 0xFF, 0 }, /* 29 right REC mixer */ - { 0xFF, 0xFF, 0 }, /* 2A REC control */ - { 0xFF, 0xFF, 0 }, /* 2B left SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2C right SPK mixer */ - { 0xFF, 0xFF, 0 }, /* 2D SPK control */ - { 0xFF, 0xFF, 0 }, /* 2E sidetone */ - { 0xFF, 0xFF, 0 }, /* 2F DAI1 playback level */ - - { 0xFF, 0xFF, 0 }, /* 30 DAI1 playback level */ - { 0xFF, 0xFF, 0 }, /* 31 DAI2 playback level */ - { 0xFF, 0xFF, 0 }, /* 32 DAI2 playbakc level */ - { 0xFF, 0xFF, 0 }, /* 33 left ADC level */ - { 0xFF, 0xFF, 0 }, /* 34 right ADC level */ - { 0xFF, 0xFF, 0 }, /* 35 MIC1 level */ - { 0xFF, 0xFF, 0 }, /* 36 MIC2 level */ - { 0xFF, 0xFF, 0 }, /* 37 INA level */ - { 0xFF, 0xFF, 0 }, /* 38 INB level */ - { 0xFF, 0xFF, 0 }, /* 39 left HP volume */ - { 0xFF, 0xFF, 0 }, /* 3A right HP volume */ - { 0xFF, 0xFF, 0 }, /* 3B left REC volume */ - { 0xFF, 0xFF, 0 }, /* 3C right REC volume */ - { 0xFF, 0xFF, 0 }, /* 3D left SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3E right SPK volume */ - { 0xFF, 0xFF, 0 }, /* 3F MIC config */ - - { 0xFF, 0xFF, 0 }, /* 40 MIC threshold */ - { 0xFF, 0xFF, 0 }, /* 41 excursion limiter filter */ - { 0xFF, 0xFF, 0 }, /* 42 excursion limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 43 ALC */ - { 0xFF, 0xFF, 0 }, /* 44 power limiter threshold */ - { 0xFF, 0xFF, 0 }, /* 45 power limiter config */ - { 0xFF, 0xFF, 0 }, /* 46 distortion limiter config */ - { 0xFF, 0xFF, 0 }, /* 47 audio input */ - { 0xFF, 0xFF, 0 }, /* 48 microphone */ - { 0xFF, 0xFF, 0 }, /* 49 level control */ - { 0xFF, 0xFF, 0 }, /* 4A bypass switches */ - { 0xFF, 0xFF, 0 }, /* 4B jack detect */ - { 0xFF, 0xFF, 0 }, /* 4C input enable */ - { 0xFF, 0xFF, 0 }, /* 4D output enable */ - { 0xFF, 0xFF, 0 }, /* 4E bias control */ - { 0xFF, 0xFF, 0 }, /* 4F DAC power */ - - { 0xFF, 0xFF, 0 }, /* 50 DAC power */ - { 0xFF, 0xFF, 0 }, /* 51 system */ - { 0xFF, 0xFF, 0 }, /* 52 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 53 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 54 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 55 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 56 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 57 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 58 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 59 DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5A DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5B DAI1 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 5C DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5D DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5E DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 5F DAI1 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 60 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 61 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 62 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 63 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 64 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 65 DAI1 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 66 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 67 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 68 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 69 DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6A DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6B DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6C DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6D DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6E DAI1 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 6F DAI1 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* 70 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 71 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 72 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 73 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 74 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 75 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 76 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 77 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 78 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 79 DAI1 EQ4 */ - { 0xFF, 0xFF, 0 }, /* 7A DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7B DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7C DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7D DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7E DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 7F DAI1 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* 80 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 81 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 82 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 83 DAI1 EQ5 */ - { 0xFF, 0xFF, 0 }, /* 84 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 85 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 86 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 87 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 88 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 89 DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8A DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8B DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8C DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8D DAI2 EQ1 */ - { 0xFF, 0xFF, 0 }, /* 8E DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 8F DAI2 EQ2 */ - - { 0xFF, 0xFF, 0 }, /* 90 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 91 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 92 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 93 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 94 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 95 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 96 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 97 DAI2 EQ2 */ - { 0xFF, 0xFF, 0 }, /* 98 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 99 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9A DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9B DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9C DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9D DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9E DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* 9F DAI2 EQ3 */ - - { 0xFF, 0xFF, 0 }, /* A0 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A1 DAI2 EQ3 */ - { 0xFF, 0xFF, 0 }, /* A2 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A3 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A4 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A5 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A6 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A7 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A8 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* A9 DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AA DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AB DAI2 EQ4 */ - { 0xFF, 0xFF, 0 }, /* AC DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AD DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AE DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* AF DAI2 EQ5 */ - - { 0xFF, 0xFF, 0 }, /* B0 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B1 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B2 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B3 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B4 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B5 DAI2 EQ5 */ - { 0xFF, 0xFF, 0 }, /* B6 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B7 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B8 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* B9 DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BA DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BB DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BC DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BD DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BE DAI1 biquad */ - { 0xFF, 0xFF, 0 }, /* BF DAI1 biquad */ - - { 0xFF, 0xFF, 0 }, /* C0 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C1 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C2 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C3 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C4 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C5 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C6 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C7 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C8 DAI2 biquad */ - { 0xFF, 0xFF, 0 }, /* C9 DAI2 biquad */ - { 0x00, 0x00, 0 }, /* CA */ - { 0x00, 0x00, 0 }, /* CB */ - { 0x00, 0x00, 0 }, /* CC */ - { 0x00, 0x00, 0 }, /* CD */ - { 0x00, 0x00, 0 }, /* CE */ - { 0x00, 0x00, 0 }, /* CF */ - - { 0x00, 0x00, 0 }, /* D0 */ - { 0x00, 0x00, 0 }, /* D1 */ - { 0x00, 0x00, 0 }, /* D2 */ - { 0x00, 0x00, 0 }, /* D3 */ - { 0x00, 0x00, 0 }, /* D4 */ - { 0x00, 0x00, 0 }, /* D5 */ - { 0x00, 0x00, 0 }, /* D6 */ - { 0x00, 0x00, 0 }, /* D7 */ - { 0x00, 0x00, 0 }, /* D8 */ - { 0x00, 0x00, 0 }, /* D9 */ - { 0x00, 0x00, 0 }, /* DA */ - { 0x00, 0x00, 0 }, /* DB */ - { 0x00, 0x00, 0 }, /* DC */ - { 0x00, 0x00, 0 }, /* DD */ - { 0x00, 0x00, 0 }, /* DE */ - { 0x00, 0x00, 0 }, /* DF */ - - { 0x00, 0x00, 0 }, /* E0 */ - { 0x00, 0x00, 0 }, /* E1 */ - { 0x00, 0x00, 0 }, /* E2 */ - { 0x00, 0x00, 0 }, /* E3 */ - { 0x00, 0x00, 0 }, /* E4 */ - { 0x00, 0x00, 0 }, /* E5 */ - { 0x00, 0x00, 0 }, /* E6 */ - { 0x00, 0x00, 0 }, /* E7 */ - { 0x00, 0x00, 0 }, /* E8 */ - { 0x00, 0x00, 0 }, /* E9 */ - { 0x00, 0x00, 0 }, /* EA */ - { 0x00, 0x00, 0 }, /* EB */ - { 0x00, 0x00, 0 }, /* EC */ - { 0x00, 0x00, 0 }, /* ED */ - { 0x00, 0x00, 0 }, /* EE */ - { 0x00, 0x00, 0 }, /* EF */ - - { 0x00, 0x00, 0 }, /* F0 */ - { 0x00, 0x00, 0 }, /* F1 */ - { 0x00, 0x00, 0 }, /* F2 */ - { 0x00, 0x00, 0 }, /* F3 */ - { 0x00, 0x00, 0 }, /* F4 */ - { 0x00, 0x00, 0 }, /* F5 */ - { 0x00, 0x00, 0 }, /* F6 */ - { 0x00, 0x00, 0 }, /* F7 */ - { 0x00, 0x00, 0 }, /* F8 */ - { 0x00, 0x00, 0 }, /* F9 */ - { 0x00, 0x00, 0 }, /* FA */ - { 0x00, 0x00, 0 }, /* FB */ - { 0x00, 0x00, 0 }, /* FC */ - { 0x00, 0x00, 0 }, /* FD */ - { 0x00, 0x00, 0 }, /* FE */ - { 0xFF, 0x00, 1 }, /* FF */ -}; - static bool max98088_readable_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].readable; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... 0xC9: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } +} + +static bool max98088_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case M98088_REG_03_BATTERY_VOLTAGE ... 0xC9: + return true; + default: + return false; + } } static bool max98088_volatile_register(struct device *dev, unsigned int reg) { - return max98088_access[reg].vol; + switch (reg) { + case M98088_REG_00_IRQ_STATUS ... M98088_REG_03_BATTERY_VOLTAGE: + case M98088_REG_FF_REV_ID: + return true; + default: + return false; + } } static const struct regmap_config max98088_regmap = { @@ -551,6 +295,7 @@ static const struct regmap_config max98088_regmap = { .val_bits = 8, .readable_reg = max98088_readable_register, + .writeable_reg = max98088_writeable_register, .volatile_reg = max98088_volatile_register, .max_register = 0xff, -- 2.1.0

1 1

[alsa-devel] [PATCH v3] ASoC: Add support for NAU8825 codec to ASoC
by Chih-Chiang Chang 28 Jul '15

28 Jul '15

The NAU88L25 is an ultra-low power high performance audio codec designed for smartphone, tablet PC, and other portable devices by Nuvoton, now add linux driver support for it. Signed-off-by: Chih-Chiang Chang <ccchang12(a)nuvoton.com> --- v3->v2: - fix the wrong definition of reg_default - fix the flow of set_sys_clk() and nau8825_set_bias_level() - remove unnecessary code in nau8825_volatile_register() and nau8825_i2c_probe() - add trigger function for ADC/DAC control - add some register definitions - fix some coding style issues v2->v1: - fixes according to Lars-Peter Clausen's review comments - removes unused platform data file - corrects the naming of DAPM input widget - fixes some wrong coding of SOC widgets and other codes - adds definition and remark for config FLL clock - moves the code of reset hardware registers from codec_probe() to i2c_probe() - removes unused codes sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 724 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 399 +++++++++++++++++++++++++ 4 files changed, 1130 insertions(+) create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index efaafce..1edb781 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -76,6 +76,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -470,6 +471,10 @@ config SND_SOC_MAX98925 config SND_SOC_MAX9850 tristate +config SND_SOC_NAU8825 + tristate "Nuvoton NAU8825 CODEC" + depends on I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index cf160d9..db0a4ec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -69,6 +69,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -256,6 +257,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..d3b306a --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,724 @@ +/* + * nau8825.c + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/regmap.h> +#include <linux/acpi.h> +#include <asm/div64.h> +#include <sound/jack.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/delay.h> +#include "nau8825.h" + +static const DECLARE_TLV_DB_SCALE(out_hp_vol_tlv, -5400, 100, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -10350, 50, 0); + +static const struct snd_kcontrol_new nau8825_snd_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_ADC_DGAIN_CTRL, + NAU8825_ADC_DGAIN_SFT, + NAU8825_ADC_VOL_RSCL_RANGE, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_HSVOL_CTRL, + NAU8825_L_HSVOL_SFT, NAU8825_R_HSVOL_SFT, + NAU8825_VOL_RSCL_RANGE, 1, out_hp_vol_tlv), +}; + +static const struct snd_kcontrol_new nau8825_hpo_mix[] = { + SOC_DAPM_SINGLE("HP L Switch", NAU8825_HSVOL_CTRL, + NAU8825_L_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("HP R Switch", NAU8825_HSVOL_CTRL, + NAU8825_R_MUTE_SFT, 1, 1), +}; + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + /* Input */ + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_SUPPLY("MIC BIAS", NAU8825_BOOST, NAU8825_G_BIAS_SFT, 0, + NULL, 0), + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, NAU8825_DAC_CTRL, + NAU8825_DAC_L_SFT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, NAU8825_DAC_CTRL, + NAU8825_DAC_R_SFT, 0), + /* SPO/HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, nau8825_hpo_mix, + ARRAY_SIZE(nau8825_hpo_mix)), + SND_SOC_DAPM_SUPPLY("HP amp", NAU8825_CLASSG_CTRL, + NAU8825_CLASSG_EN_SHIFT, 0, NULL, 0), + /* Output */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"AIF1TX", NULL, "MIC"}, + {"DAC L1", NULL, "AIF1RX"}, + {"DAC R1", NULL, "AIF1RX"}, + {"HPO MIX", "HP L Switch", "DAC L1"}, + {"HPO MIX", "HP R Switch", "DAC R1"}, + {"HPOL", NULL, "HPO MIX"}, + {"HPOR", NULL, "HPO MIX"}, + {"HPOL", NULL, "HP amp"}, + {"HPOR", NULL, "HP amp"}, +}; + +static int nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_I2S_PCM_CTRL_1, + NAU8825_I2S_DL_MASK, val_len); + return 0; +} + +static int nau8825_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + if (mute) + regmap_update_bits(nau8825->regmap, NAU8825_DAC_MUTE_CTRL, + NAU8825_SOFT_MUTE_MASK, NAU8825_SOFT_MUTE_EN); + else + regmap_update_bits(nau8825->regmap, NAU8825_DAC_MUTE_CTRL, + NAU8825_SOFT_MUTE_MASK, NAU8825_SOFT_MUTE_DIS); + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + dev_dbg(codec->dev, " %s ###nau8825_set_dai_fmt %x\n", __func__, fmt); + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= NAU8825_I2S_BP_INV; + break; + default: + dev_alert(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + reg_val |= NAU8825_I2S_DF_RIGHT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= NAU8825_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= NAU8825_I2S_DF_PCM_B; + reg_val |= NAU8825_I2S_PCMB_EN; + break; + default: + dev_alert(codec->dev, "Invalid DAI I2S/PCM format\n"); + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_I2S_PCM_CTRL_1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK + | NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + reg_val); + return 0; +} + +static void config_fll_clk_12m(struct snd_soc_codec *codec) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0x0003); + regmap_update_bits(nau8825->regmap, NAU8825_FLL_1, + NAU8825_FLL_RATIO_MASK, 0x0001); + /* FLL 16-bit fractional input */ + regmap_write(nau8825->regmap, NAU8825_FLL_2, 0xc49b); + /* FLL 10-bit integer input */ + regmap_update_bits(nau8825->regmap, NAU8825_FLL_3, + NAU8825_FLL_INTEGER_MASK, 0x0020); + /* FLL pre-scaler */ + regmap_update_bits(nau8825->regmap, NAU8825_FLL_4, + NAU8825_FLL_REF_DIV_MASK, 0x0800); + /* select divied VCO input */ + regmap_update_bits(nau8825->regmap, NAU8825_FLL_5, + NAU8825_FLL_FILTER_SW_MASK, 0x0000); + /* FLL sigma delta modulator enable */ + regmap_update_bits(nau8825->regmap, NAU8825_FLL_6, + NAU8825_SDM_EN_MASK, 0x4000); +} + +static void set_sys_clk(struct snd_soc_codec *codec, int sys_clk) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + pr_debug("%s :: sys_clk=%x\n", __func__, sys_clk); + switch (sys_clk) { + case NAU8825_INTERNALCLOCK: + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_SYSCLK_EN_MASK, NAU8825_SYSCLK_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_FLL_6, + NAU8825_DCO_EN_MASK, NAU8825_DCO_EN); + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_SYSCLK_EN_MASK, NAU8825_SYSCLK_EN); + break; + case NAU8825_MCLK: + default: + regmap_update_bits(nau8825->regmap, NAU8825_FLL_6, + NAU8825_DCO_EN_MASK, NAU8825_DCO_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_I2S_PCM_CTRL_2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + /* FLL clock source from MCLK */ + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_SYSCLK_EN_MASK, NAU8825_SYSCLK_DIS); + mdelay(2); + regmap_update_bits(nau8825->regmap, NAU8825_CLK_DIVIDER, + NAU8825_SYSCLK_EN_MASK, NAU8825_SYSCLK_EN); + break; + } +} + +static int nau8825_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (clk_id) { + case NAU8825_MCLK: + config_fll_clk_12m(codec); + set_sys_clk(codec, clk_id); + break; + case NAU8825_INTERNALCLOCK: + set_sys_clk(codec, clk_id); + break; + default: + dev_err(codec->dev, "Wrong clock src\n"); + return -EINVAL; + } + return 0; +} + +static int nau8825_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + /* All power is driven by DAPM system*/ + dev_dbg(codec->dev, "###nau8825_set_bias_level BIAS_ON\n"); + regmap_update_bits(nau8825->regmap, NAU8825_BIAS_ADJ, + NAU8825_VMIDSEL_MASK, NAU8825_VMIDSEL_125KOHM); + regmap_update_bits(nau8825->regmap, NAU8825_BIAS_ADJ, + NAU8825_VMID_MASK, NAU8825_VMID_EN); + regmap_update_bits(nau8825->regmap, NAU8825_BOOST, + NAU8825_G_BIAS_MASK, NAU8825_G_BIAS_EN); + break; + case SND_SOC_BIAS_OFF: + dev_dbg(codec->dev, "###nau8825_set_bias_level OFF\n"); + set_sys_clk(codec, NAU8825_INTERNALCLOCK); + regmap_update_bits(nau8825->regmap, NAU8825_BIAS_ADJ, + NAU8825_VMID_MASK, NAU8825_VMID_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_BOOST, + NAU8825_G_BIAS_MASK, NAU8825_G_BIAS_DIS); + break; + default: + break; + } + codec->dapm.bias_level = level; + dev_dbg(codec->dev, "## nau8825_set_bias_level %d\n", level); + return 0; +} + + +static const struct reg_default nau8825_reg[] = { + {0x000, 0x0000}, + {0x001, 0x00ff}, + {0x003, 0x0050}, + {0x004, 0x0000}, + {0x005, 0x3126}, + {0x006, 0x0008}, + {0x007, 0x0010}, + {0x008, 0x0000}, + {0x009, 0x6000}, + {0x00a, 0xf13c}, + {0x00c, 0x000c}, + {0x00d, 0x0000}, + {0x00f, 0x0800}, + {0x010, 0x0000}, + {0x011, 0x0000}, + {0x012, 0x0010}, + {0x013, 0x0015}, + {0x014, 0x0110}, + {0x015, 0x0000}, + {0x016, 0x0000}, + {0x017, 0x0000}, + {0x018, 0x0000}, + {0x019, 0x0000}, + {0x01a, 0x0000}, + {0x01b, 0x0000}, + {0x01c, 0x000b}, + {0x01d, 0x8010}, + {0x01e, 0x0000}, + {0x01f, 0x0000}, + {0x020, 0x0000}, + {0x021, 0x0000}, + {0x022, 0x0000}, + {0x023, 0x0000}, + {0x024, 0x0000}, + {0x025, 0x0000}, + {0x026, 0x0000}, + {0x027, 0x0000}, + {0x028, 0x0000}, + {0x029, 0x0000}, + {0x02a, 0x0000}, + {0x02b, 0x0010}, + {0x02c, 0x0001}, + {0x02d, 0x0000}, + {0x02f, 0x0000}, + {0x030, 0x01cf}, + {0x031, 0x0000}, + {0x032, 0x0000}, + {0x033, 0x00cf}, + {0x034, 0x0000}, + {0x038, 0x1486}, + {0x039, 0x0f12}, + {0x03a, 0x25ff}, + {0x03b, 0x3457}, + {0x045, 0x1486}, + {0x046, 0x0f12}, + {0x047, 0x25f9}, + {0x048, 0x3457}, + {0x04c, 0x0000}, + {0x04d, 0x0000}, + {0x050, 0x0000}, + {0x051, 0x0000}, + {0x055, 0x0000}, + {0x058, 0x0000}, + {0x059, 0x0000}, + {0x066, 0x0000}, + {0x068, 0x0000}, + {0x069, 0x0000}, + {0x06a, 0x0020}, + {0x071, 0x0011}, + {0x072, 0x0020}, + {0x073, 0x0008}, + {0x074, 0x0006}, + {0x076, 0x0000}, + {0x077, 0x0000}, + {0x07f, 0x0000}, + {0x080, 0x0300}, + {0x081, 0x0000}, + {0x082, 0x0000}, +}; + +static int nau8825_codec_remove(struct snd_soc_codec *codec) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + regmap_write(nau8825->regmap, NAU8825_RESET, 0x00); + regmap_write(nau8825->regmap, NAU8825_RESET, 0x00); + return 0; +} + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + /* bias current settings */ + regmap_write(nau8825->regmap, 0x0072, 0x0260); + /* enable bias */ + nau8825_set_bias_level(codec, SND_SOC_BIAS_ON); + mdelay(10); + /* DAC digital default gain 0 dB */ + regmap_write(nau8825->regmap, 0x0033, 0x00cf); + regmap_write(nau8825->regmap, 0x0034, 0x02cf); + /* DAC driver default gain -29 dB */ + regmap_write(nau8825->regmap, 0x0032, 0x075d); + /* ADC digital default gain 8 dB */ + regmap_write(nau8825->regmap, 0x0030, 0x00d2); + /* ehance I2C SDA driver strength */ + regmap_write(nau8825->regmap, 0x0080, 0x0800); + /* enable ADC/DAC clocks */ + regmap_write(nau8825->regmap, 0x0001, 0x07fd); + /* headphone output */ + regmap_write(nau8825->regmap, 0x00C, 0x000c); + return 0; +} + +static const struct snd_soc_codec_driver soc_codec_driver_nau8825 = { + .probe = nau8825_codec_probe, + .remove = nau8825_codec_remove, + .suspend_bias_off = true, + .set_bias_level = nau8825_set_bias_level, + .controls = nau8825_snd_controls, + .num_controls = ARRAY_SIZE(nau8825_snd_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static bool nau8825_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_RESET: + case NAU8825_ENA_CTRL: + case NAU8825_CLK_EN: + case NAU8825_CLK_DIVIDER: + case NAU8825_FLL_1: + case NAU8825_FLL_2: + case NAU8825_FLL_3: + case NAU8825_FLL_4: + case NAU8825_FLL_5: + case NAU8825_FLL_6: + case NAU8825_HEADSET_CTRL: + case NAU8825_JACK_DET_CTRL: + case NAU8825_IRQ_MASK: + case NAU8825_IRQ_STATUS: + case NAU8825_IRQ_CLEAR: + case NAU8825_IRQ_CTRL: + case NAU8825_SAR_ADC: + case NAU8825_VDET_COEFFICIENT: + case NAU8825_VDET_THRESHOLD_1: + case NAU8825_VDET_THRESHOLD_2: + case NAU8825_GPIO12_CTRL: + case NAU8825_GPIO34_CTRL: + case NAU8825_I2S_PCM_CTRL_1: + case NAU8825_I2S_PCM_CTRL_2: + case NAU8825_ADC_RATE: + case NAU8825_DAC_CTRL1: + case NAU8825_DAC_CTRL2: + case NAU8825_IMM_MODE_CTRL: + case NAU8825_IMM_RMS_VALUE: + case NAU8825_ADC_DGAIN_CTRL: + case NAU8825_DAC_MUTE_CTRL: + case NAU8825_HSVOL_CTRL: + case NAU8825_DACL_CTRL: + case NAU8825_DACR_CTRL: + case NAU8825_SAR_ADC_OUTPUT: + case NAU8825_ANALOG_CTRL_2: + case NAU8825_ANALOG_ADC_1: + case NAU8825_ANALOG_ADC_2: + case NAU8825_DAC_CTRL: + case NAU8825_MIC_BIAS: + case NAU8825_BOOST: + case NAU8825_CLASSG_CTRL: + case NAU8825_I2C_DEVICE_ID: + case NAU8825_BIAS_ADJ: + case NAU8825_POWER_UP_CTRL: + case NAU8825_CHARGE_BUMP_CTRL: + case NAU8825_CHARGE_BUMP_RD: + case NAU8825_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_RESET: + case NAU8825_CLK_DIVIDER: + case NAU8825_FLL_1: + case NAU8825_FLL_2: + case NAU8825_FLL_3: + case NAU8825_FLL_4: + case NAU8825_FLL_5: + case NAU8825_FLL_6: + case NAU8825_ANALOG_CTRL_2: + case NAU8825_ANALOG_ADC_1: + case NAU8825_ANALOG_ADC_2: + case NAU8825_DAC_CTRL: + case NAU8825_MIC_BIAS: + case NAU8825_BOOST: + case NAU8825_CLASSG_CTRL: + case NAU8825_I2C_DEVICE_ID: + case NAU8825_BIAS_ADJ: + case NAU8825_POWER_UP_CTRL: + case NAU8825_CHARGE_BUMP_CTRL: + case NAU8825_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static const struct regmap_config nau8825_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = NAU8825_MAX_REGISTER, + .volatile_reg = nau8825_volatile_register, + .readable_reg = nau8825_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg), +}; + +static int ena_adc(struct nau8825_priv *nau8825) +{ + /* adc & clock settings */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_ADC_CLK_MASK, NAU8825_ADC_CLK_EN); + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_ADC_MASK, NAU8825_ADC_EN); + /* adc PGA settings */ + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + NAU8825_FEPGA_GAIN_MASK, 0x1b00); + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + NAU8825_FEPGA_MASK, NAU8825_FEPGA_EN); + /* mic bias output level (1.1V) */ + regmap_update_bits(nau8825->regmap, NAU8825_MIC_BIAS, + NAU8825_MIC_BIAS_LVL_MSK, 0x04); + /* enable mic bias */ + regmap_update_bits(nau8825->regmap, NAU8825_MIC_BIAS, + NAU8825_MIC_POWERUP_MSK, NAU8825_MIC_POWERUP_EN); + mdelay(10); + return 0; +} + +static int dis_adc(struct nau8825_priv *nau8825) +{ + /* disable PGA */ + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + NAU8825_FEPGA_MASK, NAU8825_FEPGA_DIS); + /* adc & clock settings */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_ADC_MASK, NAU8825_ADC_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_ADC_CLK_MASK, NAU8825_ADC_CLK_DIS); + return 0; +} + +static int ena_dac(struct nau8825_priv *nau8825) +{ + /* charge bump settings */ + regmap_update_bits(nau8825->regmap, NAU8825_CHARGE_BUMP_CTRL, + 0x07ff, 0x720); + /* enable DAC clock */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_DAC_CLK_MASK, NAU8825_DAC_CLK_EN); + /* enable DAC channel */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_DAC_MASK, NAU8825_DAC_EN); + /* classG parameter settings */ + regmap_update_bits(nau8825->regmap, NAU8825_CLASSG_CTRL, + ~NAU8825_CLASSG_EN_MASK, 0x2007); + /* enable DAC */ + regmap_write(nau8825->regmap, NAU8825_DAC_CTRL, 0x332c); + /* output driver settings */ + regmap_write(nau8825->regmap, NAU8825_POWER_UP_CTRL, 0x073c); + /* power up main driver */ + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + NAU8825_PUP_MAIN_MASK, NAU8825_PUP_MAIN_ENA); + mdelay(200); + /* enable charge bump */ + regmap_update_bits(nau8825->regmap, NAU8825_CHARGE_BUMP_CTRL, + NAU8825_PD_DAC_MASK, NAU8825_PD_DAC_DIS); + return 0; +} + +static int dis_dac(struct nau8825_priv *nau8825) +{ + /* disable charge bump */ + regmap_update_bits(nau8825->regmap, NAU8825_CHARGE_BUMP_CTRL, + NAU8825_PD_DAC_MASK, NAU8825_PD_DAC_ENA); + /* disable classG */ + regmap_update_bits(nau8825->regmap, NAU8825_CLASSG_CTRL, + NAU8825_CLASSG_PATH_MASK, NAU8825_CLASSG_PATH_DIS); + /* disable DAC */ + regmap_update_bits(nau8825->regmap, NAU8825_DAC_CTRL, + NAU8825_DAC_LR_MASK, NAU8825_DAC_LR_DIS); + regmap_update_bits(nau8825->regmap, NAU8825_DAC_CTRL, + NAU8825_DAC_CLK_LR_MASK, NAU8825_DAC_CLK_LR_DIS); + /* disable output driver */ + regmap_update_bits(nau8825->regmap, NAU8825_POWER_UP_CTRL, + 0x3f, 0x00); + /* disable DAC channel */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_L_DAC_MASK, NAU8825_L_DAC_DIS); + /* disable DAC clock */ + regmap_update_bits(nau8825->regmap, NAU8825_ENA_CTRL, + NAU8825_DAC_CLK_MASK, NAU8825_DAC_CLK_DIS); + return 0; +} + +static int nau8825_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8825_priv *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + config_fll_clk_12m(codec); + set_sys_clk(codec, NAU8825_MCLK); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ena_dac(nau8825); + else + ena_adc(nau8825); + break; + case SNDRV_PCM_TRIGGER_RESUME: + break; + case SNDRV_PCM_TRIGGER_STOP: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dis_dac(nau8825); + else + dis_adc(nau8825); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + break; + default: + break; + } + return 0; +} + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .trigger = nau8825_trigger, + .hw_params = nau8825_hw_params, + .set_sysclk = nau8825_dai_set_sysclk, + .set_fmt = nau8825_set_dai_fmt, + .digital_mute = nau8825_dac_mute, +}; + +static struct snd_soc_dai_driver nau8825_dai_driver[] = { + { + .name = "nau8825-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, + } +}; + + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct nau8825_priv *nau8825; + int ret; + + nau8825 = devm_kzalloc(&i2c->dev, sizeof(*nau8825), + GFP_KERNEL); + if (nau8825 == NULL) + return -ENOMEM; + nau8825->i2c = i2c; + i2c_set_clientdata(i2c, nau8825); + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap); + if (IS_ERR(nau8825->regmap)) { + ret = PTR_ERR(nau8825->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + /* software reset */ + regmap_write(nau8825->regmap, NAU8825_RESET, 0x00); + regmap_write(nau8825->regmap, NAU8825_RESET, 0x00); + + /* register sound card */ + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_nau8825, + nau8825_dai_driver, + ARRAY_SIZE(nau8825_dai_driver)); + return ret; +} + +static int nau8825_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_id[] = { + {"nau8825", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_id); + +static const struct acpi_device_id nau8825_acpi_match[] = { + { "10508825", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8825_acpi_match); + +static struct i2c_driver nau8825_i2c_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(nau8825_acpi_match), + }, + .probe = nau8825_i2c_probe, + .remove = (nau8825_i2c_remove), + .id_table = nau8825_i2c_id, +}; +module_i2c_driver(nau8825_i2c_driver); + +MODULE_DESCRIPTION("ASoC NAU8825 codec driver"); +MODULE_AUTHOR("Nuvoton"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..3050e7c --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,399 @@ +/* + * nau8825.h + * + * Copyright 2015 Nuvoton Technology Corp. + * Author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _NAU8825_H +#define _NAU8825_H + +#define NAU8825_RESET 0x00 +#define NAU8825_ENA_CTRL 0x01 +#define NAU8825_CLK_EN 0x02 +#define NAU8825_CLK_DIVIDER 0x03 +#define NAU8825_FLL_1 0x04 +#define NAU8825_FLL_2 0x05 +#define NAU8825_FLL_3 0x06 +#define NAU8825_FLL_4 0x07 +#define NAU8825_FLL_5 0x08 +#define NAU8825_FLL_6 0x09 +#define NAU8825_HEADSET_CTRL 0x0C +#define NAU8825_JACK_DET_CTRL 0x0D +#define NAU8825_IRQ_MASK 0x0F +#define NAU8825_IRQ_STATUS 0x10 +#define NAU8825_IRQ_CLEAR 0x11 +#define NAU8825_BTN_STATUS 0x11 +#define NAU8825_IRQ_CTRL 0x12 +#define NAU8825_SAR_ADC 0x13 +#define NAU8825_VDET_COEFFICIENT 0x14 +#define NAU8825_VDET_THRESHOLD_1 0x15 +#define NAU8825_VDET_THRESHOLD_2 0x16 +#define NAU8825_GPIO34_CTRL 0x19 +#define NAU8825_GPIO12_CTRL 0x1A +#define NAU8825_TDM_CTRL 0x1B +#define NAU8825_I2S_PCM_CTRL_1 0x1C +#define NAU8825_I2S_PCM_CTRL_2 0x1D +#define NAU8825_BIQ_CTRL 0x20 +#define NAU8825_BIQ_COF1 0x21 +#define NAU8825_BIQ_COF2 0x22 +#define NAU8825_BIQ_COF3 0x23 +#define NAU8825_BIQ_COF4 0x24 +#define NAU8825_BIQ_COF5 0x25 +#define NAU8825_BIQ_COF6 0x26 +#define NAU8825_BIQ_COF7 0x27 +#define NAU8825_BIQ_COF8 0x28 +#define NAU8825_BIQ_COF9 0x29 +#define NAU8825_BIQ_COF10 0x2A +#define NAU8825_ADC_RATE 0x2B +#define NAU8825_DAC_CTRL1 0x2C +#define NAU8825_DAC_CTRL2 0x2D +#define NAU8825_DAC_DGAIN_CTRL 0x2F +#define NAU8825_ADC_DGAIN_CTRL 0x30 +#define NAU8825_DAC_MUTE_CTRL 0x31 +#define NAU8825_HSVOL_CTRL 0x32 +#define NAU8825_DACL_CTRL 0x33 +#define NAU8825_DACR_CTRL 0x34 +#define NAU8825_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_ADC_DRC_SLOPES 0x3A +#define NAU8825_ADC_DRC_ATKDCY 0x3B +#define NAU8825_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_DAC_DRC_SLOPES 0x47 +#define NAU8825_DAC_DRC_ATKDCY 0x48 +#define NAU8825_IMM_MODE_CTRL 0x4C +#define NAU8825_IMM_RMS_VALUE 0x4D +#define NAU8825_CLASSG_CTRL 0x50 +#define NAU8825_I2C_DEVICE_ID 0x58 +#define NAU8825_SAR_ADC_OUTPUT 0x59 +#define NAU8825_BIAS_ADJ 0x66 +#define NAU8825_ANALOG_CTRL_2 0x6A +#define NAU8825_ANALOG_ADC_1 0x71 +#define NAU8825_ANALOG_ADC_2 0x72 +#define NAU8825_DAC_CTRL 0x73 +#define NAU8825_MIC_BIAS 0x74 +#define NAU8825_BOOST 0x76 +#define NAU8825_POWER_UP_CTRL 0x7F +#define NAU8825_CHARGE_BUMP_CTRL 0x80 +#define NAU8825_CHARGE_BUMP_RD 0x81 +#define NAU8825_GENERAL_STATUS 0x82 +#define NAU8825_MAX_REGISTER 0xFF + +/* reg. NAU8825_ENA_CTRL (0x01) */ +#define NAU8825_R_DAC_MASK (0x1 << 10) +#define NAU8825_R_DAC_EN (0x1 << 10) +#define NAU8825_R_DAC_DIS (0x0 << 10) +#define NAU8825_L_DAC_MASK (0x1 << 9) +#define NAU8825_L_DAC_EN (0x1 << 9) +#define NAU8825_L_DAC_DIS (0x0 << 9) +#define NAU8825_DAC_MASK (0x3 << 9) +#define NAU8825_DAC_EN (0x3 << 9) +#define NAU8825_DAC_DIS (0x0 << 9) + +#define NAU8825_R_DAC_MASK (0x1 << 10) +#define NAU8825_R_DAC_EN (0x1 << 10) +#define NAU8825_R_DAC_DIS (0x0 << 10) +#define NAU8825_L_DAC_MASK (0x1 << 9) +#define NAU8825_L_DAC_EN (0x1 << 9) +#define NAU8825_L_DAC_DIS (0x0 << 9) +#define NAU8825_ADC_MASK (0x1 << 8) +#define NAU8825_ADC_EN (0x1 << 8) +#define NAU8825_ADC_DIS (0x0 << 8) +#define NAU8825_ADC_CLK_MASK (0x1 << 7) +#define NAU8825_ADC_CLK_EN (0x1 << 7) +#define NAU8825_ADC_CLK_DIS (0x0 << 7) +#define NAU8825_DAC_CLK_MASK (0x1 << 6) +#define NAU8825_DAC_CLK_EN (0x1 << 6) +#define NAU8825_DAC_CLK_DIS (0x0 << 6) +#define NAU8825_IMM_CLK_MASK (0x1 << 5) +#define NAU8825_IMM_CLK_EN (0x1 << 5) +#define NAU8825_IMM_CLK_DIS (0x0 << 5) +#define NAU8825_I2S_CLK_MASK (0x1 << 4) +#define NAU8825_I2S_CLK_EN (0x1 << 4) +#define NAU8825_I2S_CLK_DIS (0x0 << 4) +#define NAU8825_BIST_CLK_MASK (0x1 << 3) +#define NAU8825_BIST_CLK_EN (0x1 << 3) +#define NAU8825_BIST_CLK_DIS (0x0 << 3) +#define NAU8825_OTP_CLK_MASK (0x1 << 2) +#define NAU8825_OTP_CLK_EN (0x1 << 2) +#define NAU8825_OTP_CLK_DIS (0x0 << 2) +#define NAU8825_SAR_CLK_MASK (0x1 << 1) +#define NAU8825_SAR_CLK_EN (0x1 << 1) +#define NAU8825_SAR_CLK_DIS (0x0 << 1) +#define NAU8825_DRC_CLK_MASK (0x1 << 0) +#define NAU8825_DRC_CLK_EN (0x1 << 0) +#define NAU8825_DRC_CLK_DIS (0x0 << 0) + +/* reg. NAU8825_CLK_DIVIDER (0x03) */ +#define NAU8825_SYSCLK_EN_MASK (0x1 << 15) +#define NAU8825_SYSCLK_EN (0x1 << 15) +#define NAU8825_SYSCLK_DIS (0x0 << 15) +#define NAU8825_CLK_MCLK_SRC_MASK (0xF << 0) + +/* reg. NAU8825_FLL_1 (0x04) */ +#define NAU8825_FLL_RATIO_MASK (0x7F << 0) + +/* reg. NAU8825_FLL_3 (0x06) */ +#define NAU8825_FLL_INTEGER_MASK (0x3FF << 0) + +/* reg. NAU8825_FLL_4 (0x07) */ +#define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) + +/* reg. NAU8825_FLL_5 (0x08) */ +#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) + +/* reg. NAU8825_FLL_6 (0x09) */ +#define NAU8825_DCO_EN_MASK (0x1 << 15) +#define NAU8825_DCO_EN (0x1 << 15) +#define NAU8825_DCO_DIS (0x0 << 15) +#define NAU8825_SDM_EN_MASK (0x1 << 14) + +/* reg. NAU8825_HEADSET_CTRL (0x0c) */ +#define NAU8825_RESET_HSD_MASK (0x1 << 15) +#define NAU8825_RESET_HSD_EN (0x1 << 15) +#define NAU8825_RESET_HSD_DIS (0x0 << 15) +#define NAU8825_AUTO_DETECT_MASK (0x1 << 6) +#define NAU8825_AUTO_DETECT_EN (0x1 << 6) +#define NAU8825_AUTO_DETECT_DIS (0x0 << 6) +#define NAU8825_MANUAL_START_MASK (0x1 << 4) +#define NAU8825_MANUAL_START_EN (0x1 << 4) +#define NAU8825_MANUAL_START_DIS (0x0 << 4) +#define NAU8825_ENGND_MASK (0x3 << 2) + +/* reg. NAU8825_JACK_DET_CTRL (0x0d) */ +#define NAU8825_DB_BP_MODE_MASK (0x1 << 8) +#define NAU8825_DB_BP_MODE_EN (0x1 << 8) +#define NAU8825_DB_BP_MODE_DIS (0x0 << 8) + +/* reg. NAU8825_IRQ_MASK (0x0f) */ +#define NAU8825_IRQ_QE_MASK (0x1 << 11) +#define NAU8825_IRQ_QE_EN (0x1 << 11) +#define NAU8825_IRQ_QE_DIS (0x0 << 11) +#define NAU8825_IRQ_EJECT_MASK (0x1 << 2) +#define NAU8825_IRQ_EJECT_EN (0x1 << 2) +#define NAU8825_IRQ_EJECT_DIS (0x0 << 2) + + +/* reg. NAU8825_IRQ_STATUS (0x10) */ +#define NAU8825_HSD_COMPLETE_INT (0x1 << 10) +#define NAU8825_EMRG_INT (0x1 << 9) +#define NAU8825_RMS_INT (0x1 << 8) +#define NAU8825_BTN_RELEASE_INT (0x1 << 7) +#define NAU8825_LONG_BTN_INT (0x1 << 6) +#define NAU8825_SHORT_BTN_INT (0x1 << 5) +#define NAU8825_BTN_INT (0x7 << 5) +#define NAU8825_MIC_DET_INT (0x1 << 4) +#define NAU8825_JACK_EJCT_INT (0x3 << 2) +#define NAU8825_JACK_DET_INT (0x3 << 0) +#define NAU8825_JACK_OUT (0x1 << 2) +#define NAU8825_JACK_IN (0x1 << 0) +#define NAU8825_JACK_INT (0xF << 0) + +/* reg. NAU8825_BTN_STATUS (0x11) */ +#define NAU8825_SHORT_BTN_MASK (0xFF << 8) +#define NAU8825_LONG_BTN_MASK (0xFF << 0) +#define NAU8825_BTN_0 (0x01 << 0) +#define NAU8825_BTN_1 (0x01 << 1) +#define NAU8825_BTN_2 (0x01 << 2) +#define NAU8825_BTN_3 (0x01 << 3) + +/* reg. NAU8825_IRQ_CTRL (0x12) */ +#define NAU8825_HEADSET_INT_MASK (0x1 << 10) +#define NAU8825_HEADSET_INT_EN (0x0 << 10) +#define NAU8825_HEADSET_INT_DIS (0x1 << 10) +#define NAU8825_RMS_INT_MASK (0x1 << 8) +#define NAU8825_RMS_INT_EN (0x0 << 8) +#define NAU8825_RMS_INT_DIS (0x1 << 8) +#define NAU8825_KEYREL_INT_MASK (0x1 << 7) +#define NAU8825_KEYREL_INT_EN (0x0 << 7) +#define NAU8825_KEYREL_INT_DIS (0x1 << 7) +#define NAU8825_LONGKEY_INT_MASK (0x1 << 6) +#define NAU8825_LONGKEY_INT_EN (0x0 << 6) +#define NAU8825_LONGKEY_INT_DIS (0x1 << 6) +#define NAU8825_SHORTKEY_INT_MASK (0x1 << 5) +#define NAU8825_SHORTKEY_INT_EN (0x0 << 5) +#define NAU8825_SHORTKEY_INT_DIS (0x1 << 5) +#define NAU8825_EJECT_INT_MASK (0x1 << 2) +#define NAU8825_EJECT_INT_EN (0x0 << 2) +#define NAU8825_EJECT_INT_DIS (0x1 << 2) +#define NAU8825_JACK_IN_INT_MASK (0x1 << 0) +#define NAU8825_JACK_IN_INT_EN (0x0 << 0) +#define NAU8825_JACK_IN_INT_DIS (0x1 << 0) + +/* reg. NAU8825_SAR_ADC (0x13) */ +#define NAU8825_SAR_EN_MASK (0x1 << 12) +#define NAU8825_SAR_EN (0x1 << 12) +#define NAU8825_SAR_DIS (0x0 << 12) +#define NAU8825_JKSLV_MASK (0x1 << 11) +#define NAU8825_JKSLV_EN (0x1 << 11) +#define NAU8825_JKSLV_DIS (0x0 << 11) + +/* reg. NAU8825_I2S_PCM_CTRL_1 (0x1C) */ +#define NAU8825_I2S_TRI_STATE_MASK (0x1 << 15) +#define NAU8825_I2S_TRI_STATE_EN (0x1 << 15) +#define NAU8825_I2S_TRI_STATE_DIS (0x0 << 15) +#define NAU8825_I2S_BP_MASK (0x1 << 7) +#define NAU8825_I2S_BP_INV (0x1 << 7) +#define NAU8825_I2S_PCMB_MASK (0x1 << 6) +#define NAU8825_I2S_PCMB_EN (0x1 << 6) +#define NAU8825_I2S_DL_MASK (0x3 << 2) +#define NAU8825_I2S_DF_MASK 0x3 +#define NAU8825_I2S_DF_RIGHT 0x0 +#define NAU8825_I2S_DF_LEFT 0x1 +#define NAU8825_I2S_DF_I2S 0x2 +#define NAU8825_I2S_DF_PCM_A 0x3 +#define NAU8825_I2S_DF_PCM_B 0x3 + +/* reg. NAU8825_I2S_PCM_CTRL_2 (0x1D) */ +#define NAU8825_I2S_MS_MASK (0x1 << 3) +#define NAU8825_I2S_MS_MASTER (0x1 << 3) +#define NAU8825_I2S_MS_SLAVE (0x0 << 3) + +/* reg. NAU8825_BIQ_CTRL (0x20) */ +#define NAU8825_BIQ_WRT_MASK (0x1 << 4) +#define NAU8825_BIQ_WRT_EN (0x1 << 4) +#define NAU8825_BIQ_WRT_DIS (0x0 << 4) + +/* reg. NAU8825_ADC_DGAIN_CTRL (0x30) */ +#define NAU8825_ADC_DGAIN_SFT 0 + +/* reg. NAU8825_DAC_MUTE_CTRL (0x31) */ +#define NAU8825_SOFT_MUTE_MASK (0x1 << 9) +#define NAU8825_SOFT_MUTE_EN (0x1 << 9) +#define NAU8825_SOFT_MUTE_DIS (0x0 << 9) + +/* reg. NAU8825_HSVOL_CTRL (0x32) */ +#define NAU8825_R_MUTE (0x1 << 15) +#define NAU8825_R_MUTE_SFT 15 +#define NAU8825_L_MUTE (0x1 << 14) +#define NAU8825_L_MUTE_SFT 14 +#define NAU8825_L_HSVOL_SFT 6 +#define NAU8825_R_HSVOL_SFT 0 + +/* reg. NAU8825_IMM_MODE_CTRL (0x4C) */ +#define NAU8825_IMM_MODE_EN_MASK (0x1 << 3) +#define NAU8825_IMM_MODE_EN (0x1 << 3) +#define NAU8825_IMM_MODE_DIS (0x0 << 3) + +/* reg. NAU8825_CLASSG_CTRL (0x50) */ +#define NAU8825_CLASSG_PATH_MASK (0x3 << 1) +#define NAU8825_CLASSG_PATH_EN (0x3 << 1) +#define NAU8825_CLASSG_PATH_DIS (0x3 << 0) +#define NAU8825_CLASSG_LEFT_SHIFT 1 +#define NAU8825_CLASSG_EN_SHIFT 0 +#define NAU8825_CLASSG_EN_MASK (0x1 << 0) +#define NAU8825_CLASSG_EN_SHIFT 0 + +/*#define NAU8825_I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (0x1 << 7) + +/* reg. NAU8825_BIAS_ADJ (0x66) */ +#define NAU8825_VMID_MASK (0x1 << 6) +#define NAU8825_VMID_EN (0x1 << 6) +#define NAU8825_VMID_DIS (0x0 << 6) +#define NAU8825_VMIDSEL_MASK (0x3 << 4) +#define NAU8825_VMIDSEL_125KOHM (0x2 << 4) + +/* reg. NAU8825_ANALOG_CTRL_2 (0x6A) */ +#define NAU8825_VMID_MASK (0x1 << 6) +#define NAU8825_VMID_EN (0x1 << 6) +#define NAU8825_VMID_DIS (0x0 << 6) + +/* reg. NAU8825_DAC_CTRL (0x73) */ +#define NAU8825_DAC_R_SFT 13 +#define NAU8825_DAC_L_SFT 12 +#define NAU8825_DAC_CLK_R_SFT 9 +#define NAU8825_DAC_CLK_L_SFT 8 +#define NAU8825_DAC_LR_MASK (0x3 << 12) +#define NAU8825_DAC_LR_EN (0x3 << 12) +#define NAU8825_DAC_LR_DIS (0x0 << 12) +#define NAU8825_DAC_CLK_LR_MASK (0x3 << 8) +#define NAU8825_DAC_CLK_LR_EN (0x3 << 8) +#define NAU8825_DAC_CLK_LR_DIS (0x0 << 8) +#define NAU8825_DAC_CLK_DELAY_MASK (0x7 << 4) +#define NAU8825_DAC_REF_VOLT_MASK (0x3 << 2) + +/* reg. NAU8825_MIC_BIAS (0x74) */ +#define NAU8825_INT2KB_MSK (0x1 << 14) +#define NAU8825_INT2KB_EN (0x1 << 14) +#define NAU8825_INT2KB_DIS (0x0 << 14) +#define NAU8825_INT2KA_MSK (0x1 << 12) +#define NAU8825_INT2KA_EN (0x1 << 12) +#define NAU8825_INT2KA_DIS (0x0 << 12) +#define NAU8825_MIC_POWERUP_MSK (0x1 << 8) +#define NAU8825_MIC_POWERUP_EN (0x1 << 8) +#define NAU8825_MIC_POWERUP_DIS (0x0 << 8) +#define NAU8825_MIC_BIAS_LVL_MSK (0x7 << 0) + +/* reg. NAU8825_BOOST (0x76) */ +#define NAU8825_G_BIAS_MASK (0x1 << 12) +#define NAU8825_G_BIAS_SFT 12 +#define NAU8825_G_BIAS_EN (0x1 << 12) +#define NAU8825_G_BIAS_DIS (0x0 << 12) +#define NAU8825_BOOST_DRV_MASK (0x1 << 9) +#define NAU8825_BOOST_DRV_EN (0x0 << 9) +#define NAU8825_BOOST_DRV_DIS (0x1 << 9) + +/* reg. NAU8825_POWER_UP_CTRL (0x7F) */ +#define NAU8825_FEPGA_MASK (0x1 << 14) +#define NAU8825_FEPGA_EN (0x1 << 14) +#define NAU8825_FEPGA_DIS (0x0 << 14) +#define NAU8825_FEPGA_GAIN_MASK (0x3F << 8) +#define NAU8825_OUTPUT_DRIVER_MASK (0x3F << 0) +#define NAU8825_OUTPUT_DRIVER_EN (0x3F << 0) +#define NAU8825_OUTPUT_DRIVER_DIS (0x0 << 0) +#define NAU8825_PUP_INTEG_MASK (0x3 << 4) +#define NAU8825_PUP_INTEG_ENA (0x3 << 4) +#define NAU8825_PUP_INTEG_DIS (0x0 << 4) +#define NAU8825_PUP_OUT_MASK (0x3 << 2) +#define NAU8825_PUP_OUT_ENA (0x3 << 2) +#define NAU8825_PUP_OUT_DIS (0x0 << 2) +#define NAU8825_PUP_MAIN_MASK (0x3 << 0) +#define NAU8825_PUP_MAIN_ENA (0x3 << 0) +#define NAU8825_PUP_MAIN_DIS (0x0 << 0) + + +/* reg. NAU8825_CHARGE_BUMP_CTRL (0x80) */ +#define NAU8825_PD_DAC_MASK (0x3 << 8) +#define NAU8825_PD_DAC_ENA (0x3 << 8) +#define NAU8825_PD_DAC_DIS (0x0 << 8) +#define NAU8825_CB_CLK_MASK (0x1 << 7) +#define NAU8825_CB_CLK_EN (0x1 << 7) +#define NAU8825_CB_CLK_DIS (0x0 << 7) +#define NAU8825_CB_MASK (0x1 << 5) +#define NAU8825_CB_EN (0x1 << 5) +#define NAU8825_CB_DIS (0x0 << 5) + +/* reg. NAU8825_GENERAL_STATUS (0x82) */ +#define NAU8825_OUT12_MASK (0x3 << 10) +#define NAU8825_OUT2EN_OUT1EN (0x3 << 10) +#define NAU8825_OUT2EN_OUT1DIS (0x2 << 10) +#define NAU8825_OUT2DIS_OUT1EN (0x1 << 10) +#define NAU8825_OUT2DIS_OUT1DIS (0x0 << 10) + +/* Volume Rescale */ +#define NAU8825_VOL_RSCL_RANGE 0x36 +#define NAU8825_ADC_VOL_RSCL_RANGE 0xFF + +/* Data format */ +#define NAU8825_I2S_DL_16 (0x0 << 2) +#define NAU8825_I2S_DL_20 (0x1 << 2) +#define NAU8825_I2S_DL_24 (0x2 << 2) +#define NAU8825_I2S_DL_32 (0x3 << 2) + +enum { + NAU8825_INTERNALCLOCK = 0, + NAU8825_MCLK, +}; + +struct nau8825_priv { + struct regmap *regmap; + struct i2c_client *i2c; +}; +#endif /* _NAU8825_H */ + -- 1.9.3

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[alsa-devel] [PATCH 0/7] ASoC: Cleanup AC'97 reset handling
by Lars-Peter Clausen 27 Jul '15

27 Jul '15

AC'97 devices require that they are properly reset before they can be used. Currently ASoC drivers do this by hand each on its own. This lead to many similar but slightly different implementations. E.g. Some supporting warm reset, some don't. And even though AC'97 do have a device ID which uniquely identifies a device most drivers only use the content of register 0 for device verification. This patch series introduces a new generic helper function in the ALSA AC'97 core that can be used to reset a device and verify its ID if required. And then converts drivers to use this new helper function instead of custom code. snd_soc_new_ac97_codec() is extended to be able to perform a reset between the allocation and registration of the AC'97 device, this will allow more drivers to use it. - Lars Lars-Peter Clausen (7): ALSA: ac97: Add helper function to reset the AC97 device ASoC: ac97: Add support for resetting device before registration ASoC: ad1980: Use core AC'97 reset helper ASoC: stac9766: Use core reset helper ASoC: wm9705: Use core AC'97 reset helper ASoC: wm9712: Use core AC'97 reset helper ASoC: wm9713: Use core AC'97 reset helper include/sound/ac97_codec.h | 2 ++ include/sound/soc.h | 3 ++- sound/ac97_bus.c | 62 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1980.c | 36 +++++++++----------------- sound/soc/codecs/stac9766.c | 57 +++++------------------------------------ sound/soc/codecs/wm9705.c | 40 ++++++----------------------- sound/soc/codecs/wm9712.c | 45 ++++++-------------------------- sound/soc/codecs/wm9713.c | 48 ++++++----------------------------- sound/soc/codecs/wm9713.h | 2 -- sound/soc/soc-ac97.c | 30 ++++++++++++++++++---- 10 files changed, 134 insertions(+), 191 deletions(-) -- 2.1.4

4 20

[alsa-devel] [ALC668]: Asus N751JK - Incorrect default pin assignment for external base speaker and external microphone not working
by Arthur Borsboom 27 Jul '15

27 Jul '15

Hi, While being in the process of making my audio work on my new notebook, I noticed that the audio hardware is partially working out-of-the-Linux-kernel-box v4.0.7 Arch. The following two things seem not to work. -------------------------------- 1. External base speaker -------------------------------- The Asus N751JK notebook (ALC668) has a small external 'base' speaker (jack 2,5mm 4 rings), which I only can get to work with the help of hdajackretask with the following setting: * Override unconnected pin 0x1a and connect it to Internal Speaker (LFE). In the sound kernel tree I have noticed PCI ID fixes which are more or less equal to this situation. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/tree/sound/pci… This SND_PCI_QUIRK definition looks a lot like it: SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A), The Vendor/PCI ID is different for this laptop and I guess a new line should be added something like this: SND_PCI_QUIRK(0x1043, 0x17bd, "Asus N751JK", ALC662_FIXUP_BASS_1A), -------------------------------- 2. External microphone -------------------------------- The Asus N751JK notebook (ALC668) has a 3,5mm 4-ring combo jack for headphones and microphone. The headphone is detected correct and auto-mute works good (jack detection). The microphone is not detected and does not work. I was unable to get it to work with hdajackretask or any of the Asus model settings, such as: * options snd_hda_intel model=,asus-mode4 I was able to get the microphone to work with the following setting: * options snd_hda_intel model=,dell-headset-multi This setting creates two new items in PulseAudio making a total of three: * Internal microphone (works out of the box) * Headset microphone (works indeed with a headset) * Microphone (don't know if this works, since I don't have a single microphone, but it does not work with the headset) Unfortunately the auto-input-device-switch to the Headset microphone does not seem to work. When I plugin a headset, the headphones get detected and the Internal speakers are muted. The Input device does not automatically switch from Internal microphone to Headset microphone. I have to do this manually, after which it works. I don't know if the microphone has a jack detection. If it does not, maybe it is possible to use the headphones jack detection? I have added the alsa-info without any snd_intel / alsa modifications here: http://www.alsa-project.org/db/?f=0b5135bd12f28bb5cb182f83bfa21bc50022faa6 I don't mind to help to test, for example module options, patches and/or compile kernels. Who is able and willing to help to fix this issue? With kind regards, Arthur Borsboom. -- Arthur Borsboom Email: arthurborsboom(a)gmail.com

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