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

[alsa-devel] [PATCH] ASoC: dapm: delete dapm_kcontrol_data paths entry before freeing
by Srinivas Kandagatla 04 Jun '18

04 Jun '18

dapm_kcontrol_data is freed as part of dapm_kcontrol_free(), leaving the paths list pointer dangling in the list. This leads to system crash when we try to unload and reload sound card. I hit this bug during ADSP crash/reboot test case on Dragon board DB410c. Below is the kernel BUG with SLAB Poisoning ============================================================================= BUG kmalloc-128 (Tainted: G W ): Poison overwritten ----------------------------------------------------------------------------- Disabling lock debugging due to kernel taint INFO: 0xffff80003cf1c310-0xffff80003cf1c31f. First byte 0x10 instead of 0x6b INFO: Allocated in dapm_kcontrol_data_alloc.isra.37+0x34/0x2a8 age=6929 cpu=0 pid=50 __slab_alloc.isra.24+0x24/0x38 kmem_cache_alloc+0x190/0x1d8 dapm_kcontrol_data_alloc.isra.37+0x34/0x2a8 dapm_create_or_share_kcontrol+0x1d4/0x290 snd_soc_dapm_new_widgets+0x410/0x568 snd_soc_register_card+0xa58/0xcd0 apq8016_sbc_bind+0x31c/0x458 try_to_bring_up_master+0x204/0x2e8 component_add+0x94/0x178 q6pcm_routing_probe+0x38/0x48 platform_drv_probe+0x58/0xb8 driver_probe_device+0x324/0x478 __device_attach_driver+0xa8/0x160 bus_for_each_drv+0x48/0x98 __device_attach+0xc0/0x158 device_initial_probe+0x10/0x18 INFO: Freed in dapm_kcontrol_free+0x40/0x50 age=3135 cpu=1 pid=1792 kfree+0x1bc/0x1d0 dapm_kcontrol_free+0x40/0x50 snd_ctl_free_one+0x20/0x38 snd_ctl_remove+0xf0/0x108 snd_ctl_dev_free+0x3c/0x70 __snd_device_free+0x50/0x88 snd_device_free_all+0x2c/0x50 release_card_device+0x1c/0x78 device_release+0x34/0x98 kobject_put+0x90/0x1f0 put_device+0x14/0x20 snd_card_free+0x54/0x70 snd_soc_unregister_card+0x84/0x138 snd_soc_unregister_component+0xa4/0xd0 q6routing_dai_unbind+0x44/0x78 component_unbind.isra.4+0x28/0x50 INFO: Slab 0xffff7e0000f3c700 objects=25 used=0 fp=0xffff80003cf1fc80 flags=0xfffc00000008100 INFO: Object 0x (ptrval) @offset=768 fp=0x (ptrval) Redzone (ptrval): bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb ................ Redzone (ptrval): bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb ................ Redzone (ptrval): bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb ................ Redzone (ptrval): bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb ................ Redzone (ptrval): bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb ................ Redzone (ptrval): bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb ................ Redzone (ptrval): bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb ................ Redzone (ptrval): bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb bb ................ Object (ptrval): 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b kkkkkkkkkkkkkkkk Object (ptrval): 10 c3 f1 3c 00 80 ff ff 10 c3 f1 3c 00 80 ff ff ...<.......<.... Object (ptrval): 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b kkkkkkkkkkkkkkkk Object (ptrval): 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b kkkkkkkkkkkkkkkk Object (ptrval): 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b kkkkkkkkkkkkkkkk Object (ptrval): 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b kkkkkkkkkkkkkkkk Object (ptrval): 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b kkkkkkkkkkkkkkkk Object (ptrval): 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b a5 kkkkkkkkkkkkkkk. Redzone (ptrval): bb bb bb bb bb bb bb bb ........ Padding (ptrval): 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a ZZZZZZZZZZZZZZZZ Padding (ptrval): 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a ZZZZZZZZZZZZZZZZ Padding (ptrval): 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a ZZZZZZZZZZZZZZZZ Padding (ptrval): 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a 5a ZZZZZZZZZZZZZZZZ CPU: 1 PID: 1792 Comm: sh Tainted: G B W 4.17.0-rc7-02229-gb429ee402d16-dirty #202 Hardware name: Qualcomm Technologies, Inc. APQ 8016 SBC (DT) Call trace: dump_backtrace+0x0/0x1b0 show_stack+0x14/0x20 dump_stack+0x9c/0xbc print_trailer+0x124/0x1d8 check_bytes_and_report+0xe8/0x120 check_object+0x24c/0x288 __free_slab+0x9c/0x2f0 discard_slab+0x60/0x88 __slab_free+0x35c/0x3e8 kfree+0x1bc/0x1d0 snd_soc_dapm_free_widget+0xac/0xd0 snd_soc_dapm_free+0x64/0xb8 soc_remove_component+0x50/0x80 soc_remove_dai_links+0x110/0x208 snd_soc_unregister_card+0x9c/0x138 snd_soc_unregister_component+0xa4/0xd0 q6routing_dai_unbind+0x44/0x78 component_unbind.isra.4+0x28/0x50 component_unbind_all+0xc0/0xe8 apq8016_sbc_unbind+0x50/0xa0 take_down_master+0x24/0x48 component_del+0x90/0x130 q6afe_dai_dev_remove+0x40/0x68 platform_drv_remove+0x24/0x50 device_release_driver_internal+0x170/0x208 device_release_driver+0x14/0x20 bus_remove_device+0xcc/0x150 device_del+0x10c/0x310 platform_device_del.part.3+0x24/0x90 platform_device_unregister+0x18/0x30 of_platform_device_destroy+0x94/0x98 q6afe_remove+0x20/0x38 apr_device_remove+0x30/0x70 device_release_driver_internal+0x170/0x208 device_release_driver+0x14/0x20 bus_remove_device+0xcc/0x150 device_del+0x10c/0x310 device_unregister+0x1c/0x70 apr_remove_device+0xc/0x18 device_for_each_child+0x50/0x80 apr_remove+0x18/0x20 rpmsg_dev_remove+0x38/0x68 device_release_driver_internal+0x170/0x208 device_release_driver+0x14/0x20 bus_remove_device+0xcc/0x150 device_del+0x10c/0x310 device_unregister+0x1c/0x70 qcom_smd_remove_device+0xc/0x18 device_for_each_child+0x50/0x80 qcom_smd_unregister_edge+0x3c/0x70 smd_subdev_remove+0x18/0x28 rproc_stop+0x48/0xd8 rproc_shutdown+0x60/0xe8 state_store+0xbc/0xf8 dev_attr_store+0x18/0x28 sysfs_kf_write+0x3c/0x50 kernfs_fop_write+0x118/0x1e0 __vfs_write+0x18/0x110 vfs_write+0xa4/0x1a8 ksys_write+0x48/0xb0 sys_write+0xc/0x18 el0_svc_naked+0x30/0x34 Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 1e9a36389667..36a39ba30226 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -433,6 +433,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); + + list_del(&data->paths); kfree(data->wlist); kfree(data); } -- 2.16.2

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[alsa-devel] [PATCH] [PATCH v3] ASoC: Intel: kbl: Move codec sysclk config to codec_init function
by mac.chiang＠intel.com 04 Jun '18

04 Jun '18

From: Mac Chiang <mac.chiang(a)intel.com> On APL, commit fd0f237572ad ("ASoC: Intel: bxt: Move codec sysclk config to codec_init function") fixed an issue related to jack detection. The MCLK for DA7219 does not change in this platform, but is currently being configured everytime as part of the platform_clock event handler for DAPM. The upshot of this is that we have unnecessary calls to this function, and it also means that if a stream hasn't yet been started, DA7219 driver does not have the correct MCLK rates programmed and so the HP detection feature does not operate as expected. The same fix is needed on KBL. This patch rectifies this issue by moving the sysclk call to codec_init function so it's only called once at initialisation. Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Notes(changelog): v2: - correct the MCLK to 24.576M v3: - described the comments clearly --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index c332a32..ad3029d 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -66,14 +66,6 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, - SND_SOC_CLOCK_IN); - if (ret) { - dev_err(card->dev, "can't set codec sysclk configuration\n"); - return ret; - } - if (SND_SOC_DAPM_EVENT_OFF(event)) { ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); @@ -170,9 +162,18 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_jack *jack; int ret; + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + /* * Headset buttons map to the google Reference headset. * These can be configured by userspace. -- 2.7.4

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[alsa-devel] [PATCH v8] ASoC: pxa: switch to new ac97 bus support
by Robert Jarzmik 03 Jun '18

03 Jun '18

Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v7: added a NULL pointer check v7 was in https://patchwork.kernel.org/patch/9951919/ --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..f1f25704fe52 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -36,7 +36,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484ab3c2ad67..1a0b55beb282 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -20,13 +20,12 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S tristate diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1b41c0f2a8fb..91ead9cbbb9d 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -219,6 +212,9 @@ static const struct snd_soc_component_driver pxa_ac97_component = { static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -231,10 +227,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codecs_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -245,8 +245,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.11.0

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[alsa-devel] Remove clkdiv and pll setters from pxa dais
by Daniel Mack 02 Jun '18

02 Jun '18

Hi, As discussed in the simple-card thread the other day, the .set_clkdiv and .set_pll callbacks for DAIs are considered legacy. In order to support PXA platforms with DT, we need get rid of those for the DAIs of PXA hardware. On the good side, pxa-ssp is the only problematic one of the PXA dais, the ac97, i2s don't have any special clocking knbos, and the mmp-sspa should be straight forward to fix. I looked into pxa-ssp and it seems the conversion to calculate the audio clock dividers automatically seems doable. There are currently 4 platforms that use this cpu dai in mainline that also use snd_soc_dai_set_pll() or snd_soc_dai_set_clkdiv(): * Zylonite * Brownstone * Magician * Raumfeld I can test modifications on the latter one, but for the others, I need to understand what the machine drivers are doing, and there are some bits I don't grok. For instance, code for Zylonite does this: /* Add 1 to the width for the leading clock cycle */ pll_out = rate * (width + 1) * 8; The commit which introduced these lines dates back to 2009 and was done by you, Mark. Can you remember what the reason for this was? I've never seen sample frames with 17 bits :) This is a setup that we can't generically describe through .hw_params() or .dai_fmt() in the cpu dai, correct? In the magician code, you'll find this: case 22050: acps = 5622000; switch (width) { case 16: /* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; } break; case 44100: acps = 5622000; switch (width) { case 16: /* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_2; break; default: /* 32 */ /* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */ acds = PXA_SSP_CLK_AUDIO_DIV_1; } break; So for both 22050 and 44100, the base frequency and all dividers are the same, which can't be right. I assume these rates have never been used. I'll ignore this and implement the table in the datasheet which should do the right thing. Philipp? What we need, however, is a way to describe whether the dai is mclk master or slave. Would we add a DT propery for that? Thanks, Daniel

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[alsa-devel] [PATCH] DMA: OMAP: fix OMAP1510 incorrect residue_granularity
by Janusz Krzysztofik 02 Jun '18

02 Jun '18

Commit 0198d7bb8a0c ("ASoC: omap-mcbsp: Convert to use the sdma-pcm instead of omap-pcm") resulted in broken audio playback on OMAP1510 (discovered on Amstrad Delta). When running on OMAP1510, omap-pcm used to obtain DMA offset from snd_dmaengine_pcm_pointer_no_residue() based on DMA interrupt triggered software calculations instead of snd_dmaengine_pcm_pointer() which depended on residue value calculated from omap_dma_get_src_pos(). Similar code path is still available in now used sound/soc/soc-generic-dmaengine-pcm.c but it is not triggered. It was verified already before that omap_get_dma_src_pos() from arch/arm/plat-omap/dma.c didn't work correctly for OMAP1510 - see commit 1bdd7419910c ("ASoC: OMAP: fix OMAP1510 broken PCM pointer callback") for details. Apparently the same applies to its successor, omap_dma_get_src_pos() from drivers/dma/omap-dma.c. On the other hand, snd_dmaengine_pcm_pointer_no_residue() is described as depreciated and discouraged for use in new drivers because of its unreliable accuracy. However, it seems the only working option for OPAM1510 now, as long as a software calculated residue is not implemented as OMAP1510 fallback in omap-dma. Using snd_dmaengine_pcm_pointer_no_residue() code path instead of snd_dmaengine_pcm_pointer() in sound/soc/soc-generic-dmaengine-pcm.c can be triggered in two ways: - by passing pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE from sound/soc/omap/sdma-pcm.c, - by passing dma_caps.residue_granularity = DMA_RESIDUE_GRANULARITY_DESCRIPTOR from drivers/dma/omap-dma.c. Let's do the latter. Created and tested against next-20180531 tag from linux-next tree. Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- drivers/dma/omap-dma.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/drivers/dma/omap-dma.c b/drivers/dma/omap-dma.c index d21c19822feb..56399bd45179 100644 --- a/drivers/dma/omap-dma.c +++ b/drivers/dma/omap-dma.c @@ -1485,7 +1485,11 @@ static int omap_dma_probe(struct platform_device *pdev) od->ddev.src_addr_widths = OMAP_DMA_BUSWIDTHS; od->ddev.dst_addr_widths = OMAP_DMA_BUSWIDTHS; od->ddev.directions = BIT(DMA_DEV_TO_MEM) | BIT(DMA_MEM_TO_DEV); - od->ddev.residue_granularity = DMA_RESIDUE_GRANULARITY_BURST; + if (__dma_omap15xx(od->plat->dma_attr)) + od->ddev.residue_granularity = + DMA_RESIDUE_GRANULARITY_DESCRIPTOR; + else + od->ddev.residue_granularity = DMA_RESIDUE_GRANULARITY_BURST; od->ddev.max_burst = SZ_16M - 1; /* CCEN: 24bit unsigned */ od->ddev.dev = &pdev->dev; INIT_LIST_HEAD(&od->ddev.channels); -- 2.16.1

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[alsa-devel] [PATCH] ALSA: pci/hda: Remove unused, broken, header file
by Ben Hutchings 01 Jun '18

01 Jun '18

sound/pci/hda/local.h seems to be an earlier version of sound/hda/local.h; it was added at the same time but doesn't seem to have ever been used (within the git history). Most of its macros depend on a hdac_read_parm() function which is not defined anywhere. Signed-off-by: Ben Hutchings <ben.hutchings(a)codethink.co.uk> --- sound/pci/hda/local.h | 40 ---------------------------------------- 1 file changed, 40 deletions(-) delete mode 100644 sound/pci/hda/local.h diff --git a/sound/pci/hda/local.h b/sound/pci/hda/local.h deleted file mode 100644 index 3b8b7d78f9e0..000000000000 --- a/sound/pci/hda/local.h +++ /dev/null @@ -1,40 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0 */ -/* - */ - -#ifndef __HDAC_LOCAL_H -#define __HDAC_LOCAL_H - -int hdac_read_parm(struct hdac_device *codec, hda_nid_t nid, int parm); - -#define get_wcaps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) -/* get the widget type from widget capability bits */ -static inline int get_wcaps_type(unsigned int wcaps) -{ - if (!wcaps) - return -1; /* invalid type */ - return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; -} - -#define get_pin_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_PIN_CAP) - -static inline -unsigned int get_pin_cfg(struct hdac_device *codec, hda_nid_t nid) -{ - unsigned int val; - - if (snd_hdac_read(codec, nid, AC_VERB_GET_CONFIG_DEFAULT, 0, &val)) - return -1; - return val; -} - -#define get_amp_caps(codec, nid, dir) \ - hdac_read_parm(codec, nid, (dir) == HDA_OUTPUT ? \ - AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP) - -#define get_power_caps(codec, nid) \ - hdac_read_parm(codec, nid, AC_PAR_POWER_STATE) - -#endif /* __HDAC_LOCAL_H */ -- Ben Hutchings, Software Developer Codethink Ltd https://www.codethink.co.uk/ Dale House, 35 Dale Street Manchester, M1 2HF, United Kingdom

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[alsa-devel] [PATCH v2] ASoC: simple-card: set cpu dai clk in hw_params
by Daniel Mack 01 Jun '18

01 Jun '18

The simple-card driver currently accepts a clock node in the cpu dai sub-node and only uses it as an alternative to the 'system-clock-frequency' property to get the current frequency. This patch adds another use of the passed clock node. If mclk-fs is specified, the clocks in cpu and codec dai sub-nodes will be set to the calculated rate (stream rate * mclk_fs) in hw_params. This allows platforms to pass tuneable clocks as phandle that will automatically be set to the right rates. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- This is the only patch that's left from the series of three that I posted earlier. v1 → v2: * Apply the setting to both the codec and the cpu dai * Check for errors from clk_set_rate() and avoid unnecessary rate changes. .../devicetree/bindings/sound/simple-card.txt | 5 +++++ sound/soc/generic/simple-card.c | 21 +++++++++++++++++++++ 2 files changed, 26 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt index 17c13e74667d..a4c72d09cd45 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ b/Documentation/devicetree/bindings/sound/simple-card.txt @@ -86,6 +86,11 @@ Optional CPU/CODEC subnodes properties: in dai startup() and disabled with clk_disable_unprepare() in dai shutdown(). + If a clock is specified and a + multiplication factor is given with + mclk-fs, the clock will be set to the + calculated mclk frequency when the + stream starts. - system-clock-direction-out : specifies clock direction as 'out' on initialization. It is useful for some aCPUs with fixed clocks. diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6959a74a6f49..4a516c428b3d 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -135,6 +135,18 @@ static void asoc_simple_card_shutdown(struct snd_pcm_substream *substream) asoc_simple_card_clk_disable(&dai_props->codec_dai); } +static int asoc_simple_set_clk_rate(struct asoc_simple_dai *simple_dai, + unsigned long rate) +{ + if (!simple_dai->clk) + return 0; + + if (clk_get_rate(simple_dai->clk) == rate) + return 0; + + return clk_set_rate(simple_dai->clk, rate); +} + static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -154,6 +166,15 @@ static int asoc_simple_card_hw_params(struct snd_pcm_substream *substream, if (mclk_fs) { mclk = params_rate(params) * mclk_fs; + + ret = asoc_simple_set_clk_rate(&dai_props->codec_dai, mclk); + if (ret < 0) + return ret; + + ret = asoc_simple_set_clk_rate(&dai_props->cpu_dai, mclk); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (ret && ret != -ENOTSUPP) -- 2.14.3

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[alsa-devel] [PATCH] ASoC: Intel: kbl: Move codec sysclk config to codec_init function
by mac.chiang＠intel.com 01 Jun '18

01 Jun '18

From: Mac Chiang <mac.chiang(a)intel.com> unexpected error: PLL input clock 0 below valid range backport from broonie's tree commit fd0f237572ad ("ASoC: Intel: bxt: Move codec sysclk config to codec_init function") The MCLK for DA7219 does not change in this platform, but is currently being configured everytime as part of the platform_clock event handler for DAPM. The upshot of this is that we have unnecessary calls to this function, and it also means that if a stream hasn't yet been started, DA7219 driver does not have the correct MCLK rates programmed and so the HP detection feature does not operate as expected. This patch rectifies this issue by moving the sysclk call to codec_init function so it's only called once at initialisation. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> Acked-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Singed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 60e739f..88fbc6a 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -65,14 +65,6 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, - SND_SOC_CLOCK_IN); - if (ret) { - dev_err(card->dev, "can't set codec sysclk configuration\n"); - return ret; - } - if (SND_SOC_DAPM_EVENT_OFF(event)) { ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); @@ -169,9 +161,18 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_jack *jack; int ret; + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 19200000, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + /* * Headset buttons map to the google Reference headset. * These can be configured by userspace. -- 2.7.4

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[alsa-devel] [PATCH v3 0/2] slimbus: Add QCOM SLIMBus NGD driver
by Srinivas Kandagatla 01 Jun '18

01 Jun '18

This patchset adds support to basic version of Qualcomm NGD SLIMBus controller driver found SoCs from B family. This controller is light-weight SLIMBus controller driver responsible for communicating with slave HW directly over the bus using messaging interface, and communicating with master component residing on ADSP for bandwidth and data-channel management. Some parts fot the work is derived from Qualcomm downstream kernels and intial work from Karthikeyan Ramasubramanian <kramasub(a)codeaurora.org> and Sagar Dharia <sdharia(a)codeaurora.org> Tested this patchset on DB820c with WCD9335 codec and on msm8974 with wcd9320 codec. I have pushed my working branch to [1] incase someone want to try. This patch has dependency on https://lkml.org/lkml/2018/5/17/251 Changes sinc v1 (https://lkml.org/lkml/2018/5/25/296): - Added Tested-by and Reviewed-by Tags - Added Credits to Qualcomm engineers in commit log - Fixed a spelling mistake for prepare. Thanks, srini [1] https://git.linaro.org/people/srinivas.kandagatla/linux.git/log/?h=slimbus-… Srinivas Kandagatla (2): slimbus: ngd: dt-bindings: Add slim ngd dt bindings slimbus: ngd: Add qcom SLIMBus NGD driver .../bindings/slimbus/slim-ngd-qcom-ctrl.txt | 84 ++ drivers/slimbus/Kconfig | 11 + drivers/slimbus/Makefile | 3 + drivers/slimbus/qcom-ngd-ctrl.c | 1342 ++++++++++++++++++++ drivers/slimbus/slimbus.h | 8 + 5 files changed, 1448 insertions(+) create mode 100644 Documentation/devicetree/bindings/slimbus/slim-ngd-qcom-ctrl.txt create mode 100644 drivers/slimbus/qcom-ngd-ctrl.c -- 2.16.2

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[alsa-devel] [PATCH] ALSA: hda - Handle kzalloc() failure in snd_hda_attach_pcm_stream()
by Bo Chen 01 Jun '18

01 Jun '18

When 'kzalloc()' fails in 'snd_hda_attach_pcm_stream()', a new pcm instance is created without setting its operators via 'snd_pcm_set_ops()'. Following operations on the new pcm instance can trigger kernel null pointer dereferences and cause kernel oops. This bug was found with my work on building a gray-box fault-injection tool for linux-kernel-module binaries. A kernel null pointer dereference was confirmed from line 'substream->ops->open()' in function 'snd_pcm_open_substream()' in file 'sound/core/pcm_native.c'. This patch fixes the bug by calling 'snd_device_free()' in the error handling path of 'kzalloc()', which removes the new pcm instance from the snd card before returns with an error code. Signed-off-by: Bo Chen <chenbo(a)pdx.edu> --- sound/pci/hda/hda_controller.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index d1eb14842340..a12e594d4e3b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -748,8 +748,10 @@ int snd_hda_attach_pcm_stream(struct hda_bus *_bus, struct hda_codec *codec, return err; strlcpy(pcm->name, cpcm->name, sizeof(pcm->name)); apcm = kzalloc(sizeof(*apcm), GFP_KERNEL); - if (apcm == NULL) + if (apcm == NULL) { + snd_device_free(chip->card, pcm); return -ENOMEM; + } apcm->chip = chip; apcm->pcm = pcm; apcm->codec = codec; -- 2.17.0

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