August 2015 - Alsa-devel - mailman.alsa-project.org

[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

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[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

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[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)>] > > > > >

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[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

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[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

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[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

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[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

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