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

[alsa-devel] [PATCH] ASoC: rockchip: i2s: Adjust devm usage
by Vaishali Thakkar 19 Aug '15

19 Aug '15

Remove use of snd_soc_unregister_component in remove function as devm_snd_soc_register_component in probe function automatically handles it. Also, convert call of snd_dmaengine_pcm_register to managed resource function devm_snd_dmaengine_pcm_register and remove usage of snd_dmaengine_pcm_unregister in probe and remove functions. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/rockchip/rockchip_i2s.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index acb5be5..b936102 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -483,16 +483,14 @@ static int rockchip_i2s_probe(struct platform_device *pdev) goto err_suspend; } - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - goto err_pcm_register; + return ret; } return 0; -err_pcm_register: - snd_dmaengine_pcm_unregister(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) i2s_runtime_suspend(&pdev->dev); @@ -512,8 +510,6 @@ static int rockchip_i2s_remove(struct platform_device *pdev) clk_disable_unprepare(i2s->mclk); clk_disable_unprepare(i2s->hclk); - snd_dmaengine_pcm_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: arizona: Fix gain settings of FLL in free-run mode
by Charles Keepax 19 Aug '15

19 Aug '15

From: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> The wrong register was used to set the gain of ref loop, when changing the FLL output on an active FLL. This patch corrects the offset of the gain register. Signed-off-by: Nikesh Oswal <Nikesh.Oswal(a)wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e759d39..42b2021 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2058,7 +2058,7 @@ static int arizona_enable_fll(struct arizona_fll *fll) if (already_enabled) { /* Facilitate smooth refclk across the transition */ - regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x7, + regmap_update_bits_async(fll->arizona->regmap, fll->base + 0x9, ARIZONA_FLL1_GAIN_MASK, 0); regmap_update_bits_async(fll->arizona->regmap, fll->base + 1, ARIZONA_FLL1_FREERUN, -- 1.7.2.5

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[alsa-devel] [PATCH v2] ALSA: usb-audio: Recurse before saving terminal properties
by Julian Scheel 19 Aug '15

19 Aug '15

The input terminal parser recurses into the referenced clock entity to verify it is existant and thus the terminal descriptor is valid. The actual property values of the term instance which is initially parsed must not be overriden by the recursion. For this to work the term properties have to be assigned after recursing into the referenced clock entity descriptors. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- Changes in v2: - Store term->id after recursion as well, as it gets overriden in the recursion - Add comments explaining that the recursion is only for validation of the descriptor validity --- sound/usb/mixer.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81055d3..c50790c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -731,15 +731,21 @@ static int check_input_term(struct mixer_build *state, int id, term->name = d->iTerminal; } else { /* UAC_VERSION_2 */ struct uac2_input_terminal_descriptor *d = p1; - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le32_to_cpu(d->bmChannelConfig); - term->name = d->iTerminal; - /* call recursively to get the clock selectors */ + /* call recursively to verify that the + * referenced clock entity is valid */ err = check_input_term(state, d->bCSourceID, term); if (err < 0) return err; + + /* save input term properties after recursion, + * to ensure they are not overriden by the + * recursion calls */ + term->id = id; + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le32_to_cpu(d->bmChannelConfig); + term->name = d->iTerminal; } return 0; case UAC_FEATURE_UNIT: { -- 2.5.0

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[alsa-devel] [PATCH] ASoc: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 19 Aug '15

19 Aug '15

(Referring to this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html) This patch adds the capability to use multiple codecs on the same DAI link where one codec is used for playback and another one is used for capture. As noted in the thread, there might be more places where the new snd_soc_dai_stream_valid() function is needed, however, these two were sufficient for our use case (SSM2518 playback and ICS-43432 capture). Signed-off-by: Ricard Wanderlof <ricardw(a)xis.com> --- sound/soc/soc-pcm.c | 36 ++++++++++++++++++++++++++++++++++++ 1 file changed, 36 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6b0136e..df1f261 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -35,6 +35,32 @@ #define DPCM_MAX_BE_USERS 8 /** + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * @dai: The DAI to check for stream support + * @stream: The stream type (playback / capture) to check for + * + * Checks if the DAI in question supports the indicated stream type. + * Needed to skip over codecs when iterating over + * snd_soc_pcm_runtime->codec_dais in order to skip codecs that are playback- + * or capture-only, which would otherwise inhibit capture or playback, + * respectively, to the other codec in a multiple codec / common I2S bus + * environment. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return !!codec_stream->rates; +} + +/** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated * @stream: Direction of the PCM stream @@ -371,6 +397,12 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* Skip codecs which don't support the current stream type. */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +859,10 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* Skip codecs which don't support the current stream type. */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; -- 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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[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

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[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

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[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

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[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

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[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

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[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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