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

[alsa-devel] [PATCH V3 0/2] ARM: dts: bcm283x: Fix probing of bcm2835-i2s
by Stefan Wahren 17 Jan '18

17 Jan '18

This small series fixes the probing of bcm2835-i2s, which is broken since 517e7a1537a ("ASoC: bcm2835: move to use the clock framework"). Changes in V3: * split up patch into dt-binding and functional part Changes in V2: * add Fixes, Reviewed-by and Tested-by tags Stefan Wahren (2): dt-bindings: bcm283x: Fix register ranges of bcm2835-i2s ARM: dts: bcm283x: Fix probing of bcm2835-i2s Documentation/devicetree/bindings/dma/brcm,bcm2835-dma.txt | 4 ++-- Documentation/devicetree/bindings/sound/brcm,bcm2835-i2s.txt | 9 ++++----- arch/arm/boot/dts/bcm283x.dtsi | 4 ++-- 3 files changed, 8 insertions(+), 9 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 1/2] ASoC: sgtl5000: Do not print error on probe deferral
by Fabio Estevam 17 Jan '18

17 Jan '18

When the MCLK is not yet available when the codec is probed, probe deferral will happen and in this case we should not print an error message. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Correcty mark this patch as 1/2 sound/soc/codecs/sgtl5000.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index f2bb4fe..633cdcf 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1332,10 +1332,13 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, sgtl5000->mclk = devm_clk_get(&client->dev, NULL); if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); - dev_err(&client->dev, "Failed to get mclock: %d\n", ret); /* Defer the probe to see if the clk will be provided later */ if (ret == -ENOENT) ret = -EPROBE_DEFER; + + if (ret != -EPROBE_DEFER) + dev_err(&client->dev, "Failed to get mclock: %d\n", + ret); goto disable_regs; } -- 2.7.4

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[alsa-devel] [PATCH -next] slimbus: qcom: remove redundant dev_err call in qcom_slim_probe()
by Wei Yongjun 17 Jan '18

17 Jan '18

There is a error message within devm_ioremap_resource already, so remove the dev_err call to avoid redundant error message. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- drivers/slimbus/qcom-ctrl.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index ffb46f9..2f671dd 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -529,10 +529,8 @@ static int qcom_slim_probe(struct platform_device *pdev) slim_mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "ctrl"); ctrl->base = devm_ioremap_resource(ctrl->dev, slim_mem); - if (IS_ERR(ctrl->base)) { - dev_err(&pdev->dev, "IOremap failed\n"); + if (IS_ERR(ctrl->base)) return PTR_ERR(ctrl->base); - } sctrl->set_laddr = qcom_set_laddr; sctrl->xfer_msg = qcom_xfer_msg;

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[alsa-devel] [PATCH] ASoC: Intel: remove second duplicated assignment to pointer 'res'
by Colin King 17 Jan '18

17 Jan '18

From: Colin Ian King <colin.king(a)canonical.com> The second assignment to res is identical to the previous assignment so it is redundant and can be removed. Cleans up clang warning: sound/soc/intel/skylake/skl-topology.c:191:25: warning: Value stored to 'res' during its initialization is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/skylake/skl-topology.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 28bc16a8e09a..73af6e19ebbd 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -190,7 +190,6 @@ skl_tplg_free_pipe_mcps(struct skl *skl, struct skl_module_cfg *mconfig) u8 res_idx = mconfig->res_idx; struct skl_module_res *res = &mconfig->module->resources[res_idx]; - res = &mconfig->module->resources[res_idx]; skl->resource.mcps -= res->cps; } -- 2.15.1

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[alsa-devel] Update ALC215 depop optimize
by Kailang 17 Jan '18

17 Jan '18

Hi Takashi, Attach was the update patch for ALC215 depop. This depop method was same as ALC225. BR, Kailang

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[alsa-devel] ALC215/ALC285/ALC289 headset mode support
by Kailang 17 Jan '18

17 Jan '18

Hi Takashi, Attach patch was for ALC215/ALC285/ALC289 headset mode support. Many thanks. BR, Kailang

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[alsa-devel] Silencing plops / click sounds on start / stop of playing audio when snd_hda_intel.power_save=1
by Hans de Goede 17 Jan '18

17 Jan '18

Hi All, For Fedora 28 we plan to ship the kernel with snd_hda_intel.power_save defaulting to 1: https://fedoraproject.org/wiki/Changes/ImprovedLaptopBatteryLife We've had an update for F27 in updates-testing for a while which made the same change for F27, but we've received several bug reports about plops / click sounds on start / stop of playing audio: https://bugzilla.redhat.com/show_bug.cgi?id=1525104 I know others have probably already tried fixing this, but I'm wondering if there are any ideas how to fix this / things to try? Ideas of my own: 1) Mute the speakers / headphone output in the mixer before activating power-saving. Question how hard would it be to make this happen? 2) Put a delay between various steps involved in turning things off / on. Regards, Hans

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[alsa-devel] [bug report] ASoC: snd_soc_component_driver has snd_compr_ops
by Dan Carpenter 17 Jan '18

17 Jan '18

Hello Kuninori Morimoto, The patch 9e7e3738ab0e: "ASoC: snd_soc_component_driver has snd_compr_ops" from Oct 11, 2017, leads to the following static checker warning: sound/soc/soc-compress.c:109 soc_compr_open() warn: 'cstream->runtime->private_data' double freed sound/soc/soc-compress.c 91 return 0; 92 93 machine_err: 94 for_each_rtdcom(rtd, rtdcom) { 95 component = rtdcom->component; 96 97 /* ignore duplication for now */ 98 if (platform && (component == &platform->component)) 99 continue; 100 101 if (!component->driver->compr_ops || 102 !component->driver->compr_ops->free) 103 continue; 104 105 component->driver->compr_ops->free(cstream); ^^^^^^^^^^^^^^^ This is in a loop so is the really right? We end up freeing cstream over and over? 106 } 107 108 if (platform && platform->driver->compr_ops && platform->driver->compr_ops->free) 109 platform->driver->compr_ops->free(cstream); ^^^^^^^^^^^^^ This second call is what triggers the warning. 110 plat_err: 111 if (cpu_dai->driver->cops && cpu_dai->driver->cops->shutdown) 112 cpu_dai->driver->cops->shutdown(cstream, cpu_dai); 113 out: 114 mutex_unlock(&rtd->pcm_mutex); 115 return ret; 116 } regards, dan carpenter

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[alsa-devel] [PATCH] ALSA: usb-audio: Add a quirk for Nura's first gen headset
by Martin Peres 16 Jan '18

16 Jan '18

The capture interface does not work, and the playback interface actually supports only 48kHz unlike what is advertised (44.1, 32, 22, 16, 8). The only unknown here is if there are other devices that use the same product ID, but given that this ID is currently unknown, I would assume it is specially allocated for the nura headset. Signed-off-by: Martin Peres <martin.peres(a)free.fr> --- sound/usb/quirks-table.h | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 48 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8a59d4782a0f..50252046b01d 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3277,4 +3277,52 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +{ + /* + * Nura's first gen headphones use Cambridge Silicon Radio's vendor + * ID, but it looks like the product ID actually is only for Nura. + * The capture interface does not work at all (even on Windows), + * and only the 48 kHz sample rate works for the playback interface. + */ + USB_DEVICE(0x0a12, 0x1243), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + /* Capture */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_FILL_MAX | + UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x03, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + } + } +}, + #undef USB_DEVICE_VENDOR_SPEC -- 2.15.1

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[alsa-devel] [PATCH] ASoC: au1x: Fix timeout tests in au1xac97c_ac97_read()
by Dan Carpenter 16 Jan '18

16 Jan '18

The loop timeout doesn't work because it's a post op and ends with "tmo" set to -1. I changed it from a post-op to a pre-op and I changed the initial the starting value from 5 to 6 so we still iterate 5 times. I left the other as it was because it's a large number. Fixes: b3c70c9ea62a ("ASoC: Alchemy AC97C/I2SC audio support") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 29a97d52e8ad..66d6c52e7761 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -91,8 +91,8 @@ static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97, do { mutex_lock(&ctx->lock); - tmo = 5; - while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--) + tmo = 6; + while ((RD(ctx, AC97_STATUS) & STAT_CP) && --tmo) udelay(21); /* wait an ac97 frame time */ if (!tmo) { pr_debug("ac97rd timeout #1\n"); @@ -105,7 +105,7 @@ static unsigned short au1xac97c_ac97_read(struct snd_ac97 *ac97, * poll, Forrest, poll... */ tmo = 0x10000; - while ((RD(ctx, AC97_STATUS) & STAT_CP) && tmo--) + while ((RD(ctx, AC97_STATUS) & STAT_CP) && --tmo) asm volatile ("nop"); data = RD(ctx, AC97_CMDRESP);

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