November 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: Intel: kbl: Modify map for Headset Playback to fix pop-noise
by Naveen Manohar 03 Nov '17

03 Nov '17

Patch fixes wrong path in commit 0b06122fc8d0 ("ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref") which resulted in pop noise. Current topology for Headset results in unwanted pop noise, while switching from spk->hs at the start of Headset Playback. Hence re-introduced mixin-mixout dsp module in topology for headset playback pipe to fix the regression. And the corresponding modification for headset route is updated here. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> --- v1->v2: updated commit message to suit the change more appropriately. sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 6f9a8bc..6dcad0a 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -101,7 +101,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 6072164..271ae3c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -109,7 +109,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5514: The ACPI also should use the function rt5514_parse_dp()
by Oder Chiou 03 Nov '17

03 Nov '17

The patch fixed that the ACPI cannot access the device property from the function rt5514_parse_dp(). Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index d7956ab..2a5b5d7 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -1143,7 +1143,7 @@ static const struct acpi_device_id rt5514_acpi_match[] = { MODULE_DEVICE_TABLE(acpi, rt5514_acpi_match); #endif -static int rt5514_parse_dt(struct rt5514_priv *rt5514, struct device *dev) +static int rt5514_parse_dp(struct rt5514_priv *rt5514, struct device *dev) { device_property_read_u32(dev, "realtek,dmic-init-delay-ms", &rt5514->pdata.dmic_init_delay); @@ -1183,8 +1183,8 @@ static int rt5514_i2c_probe(struct i2c_client *i2c, if (pdata) rt5514->pdata = *pdata; - else if (i2c->dev.of_node) - rt5514_parse_dt(rt5514, &i2c->dev); + else + rt5514_parse_dp(rt5514, &i2c->dev); rt5514->i2c_regmap = devm_regmap_init_i2c(i2c, &rt5514_i2c_regmap); if (IS_ERR(rt5514->i2c_regmap)) { -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: Intel: kbl: Modify Headset map for Multi-Playback
by Vinod Koul 03 Nov '17

03 Nov '17

On Fri, Nov 03, 2017 at 02:18:34PM +0530, Naveen M wrote: > On Fri, Nov 03, 2017 at 01:51:00PM +0530, Vinod Koul wrote: > > On Fri, Nov 03, 2017 at 08:58:17AM +0530, Naveen Manohar wrote: > > > Current topology for Headset resulted in unwanted pop noise, > > > while switching from spk->hs at the start of Headset Playback. > > > > what is the reason for the pop ? > > > There is an artifact observed at the start of the stream, which is not > seen earlier with mixin-mixout dsp module, through which data was > passing-through. > > > > Modified topology to fix the same. > > > > how does that fix it, sorry am quite unclear about that part > > > Only for headset playback pipe, we re-introduced the mixin-mixout which > was there earlier. This does not fix the problem but removes regression > (pop-noise in headset playback) introduced by earlier > commit-ID:0b06122fc8d0949a0aa7b817d98e031104802611 Then please mention that in the log. IIUC you should say it fixes the wrong path in commit 0b06122fc8d0 ("ASoC: Intel: kbl: Add map for new DAIs for Multi-Playback & Echo Ref") which resulted in pop noise. > > > > This modification also requires headset route to be updated. > > > > that seems to be the only change here, so not sure I follow the changes dont > > for this > > > As the topology was modified to earlier path, earlier route change was > reverted for headset playback. > > > > > > > Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> > > > Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> > > > --- > > > sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- > > > sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- > > > 2 files changed, 2 insertions(+), 2 deletions(-) > > > > > > diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c > > > index 6f9a8bc..6dcad0a 100644 > > > --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c > > > +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c > > > @@ -101,7 +101,7 @@ enum { > > > { "ssp0 Tx", NULL, "spk_out" }, > > > > > > { "AIF Playback", NULL, "ssp1 Tx" }, > > > - { "ssp1 Tx", NULL, "hs_out" }, > > > + { "ssp1 Tx", NULL, "codec1_out" }, > > > > > > { "hs_in", NULL, "ssp1 Rx" }, > > > { "ssp1 Rx", NULL, "AIF Capture" }, > > > diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c > > > index 6072164..271ae3c 100644 > > > --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c > > > +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c > > > @@ -109,7 +109,7 @@ enum { > > > { "ssp0 Tx", NULL, "spk_out" }, > > > > > > { "AIF Playback", NULL, "ssp1 Tx" }, > > > - { "ssp1 Tx", NULL, "hs_out" }, > > > + { "ssp1 Tx", NULL, "codec1_out" }, > > > > > > { "hs_in", NULL, "ssp1 Rx" }, > > > { "ssp1 Rx", NULL, "AIF Capture" }, > > > -- > > > 1.9.1 > > > > > > > -- > > ~Vinod > > -- -- ~Vinod

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[alsa-devel] [PATCH] ASoC: Intel: fix Kconfig dependencies
by Arnd Bergmann 02 Nov '17

02 Nov '17

I ran into multiple problems during randconfig builds of the recently changed Kconfig logic for Intel ASoC drivers: - Building without DMADEVICES doesn't work in general - With that dependency added, we can relax the 'depends on X86' again and allow compile-testing, except for SND_SST_ATOM_HIFI2_PLATFORM, which depends on X86 for asm/platform_sst_audio.h - Skylake requires SND_SOC_INTEL_SST_ACPI, so we have to depend on ACPI in turn - Haswell needs SND_DMA_SGBUF for snd_sgbuf_aligned_pages() With the new set of dependencies, I no longer get any build failures. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/Kconfig | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a59dda39007e..e18118209b75 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -31,13 +31,14 @@ config SND_SOC_ACPI_INTEL_MATCH config SND_SOC_INTEL_SST_TOPLEVEL tristate "Intel ASoC SST drivers" - depends on X86 + depends on X86 || COMPILE_TEST + depends on DMADEVICES select SND_SOC_INTEL_MACH select SND_SOC_INTEL_COMMON config SND_SOC_INTEL_HASWELL tristate "Intel ASoC SST driver for Haswell/Broadwell" - depends on SND_SOC_INTEL_SST_TOPLEVEL + depends on SND_SOC_INTEL_SST_TOPLEVEL && SND_DMA_SGBUF select SND_SOC_INTEL_SST select SND_SOC_INTEL_SST_FIRMWARE @@ -49,12 +50,12 @@ config SND_SOC_INTEL_BAYTRAIL config SND_SST_ATOM_HIFI2_PLATFORM tristate "Intel ASoC SST driver for HiFi2 platforms (*field, *trail)" - depends on SND_SOC_INTEL_SST_TOPLEVEL + depends on SND_SOC_INTEL_SST_TOPLEVEL && X86 select SND_SOC_COMPRESS config SND_SOC_INTEL_SKYLAKE tristate "Intel ASoC SST driver for SKL/BXT/KBL/GLK/CNL" - depends on SND_SOC_INTEL_SST_TOPLEVEL + depends on SND_SOC_INTEL_SST_TOPLEVEL && PCI && ACPI select SND_HDA_EXT_CORE select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY -- 2.9.0

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[alsa-devel] [INTERNAL][PATCH 0/7] ASoC: stm32: Add SAI synchronization support
by Olivier Moysan 02 Nov '17

02 Nov '17

This patch-set adds support of synchronization features for SAI interface. It also adds minor fixes and improvements. Olivier Moysan (7): ASoC: stm32: Add synchronization to SAI bindings ASoC: stm32: sai: Move static settings to DAI init ASoC: stm32: sai: Fix DMA burst size ASoC: stm32: sai: fix stop management in isr ASoC: stm32: sai: Remove spurious IRQs on stop ASoC: stm32: sai: Fix get reset controller ASoC: stm32: sai: Add synchronization support .../devicetree/bindings/sound/st,stm32-sai.txt | 14 +- sound/soc/stm/stm32_sai.c | 162 ++++++++++++++++++++- sound/soc/stm/stm32_sai.h | 22 ++- sound/soc/stm/stm32_sai_sub.c | 156 +++++++++++++++++--- 4 files changed, 318 insertions(+), 36 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: wm8741: Use snd_soc_update_bits rather than hard coding
by Charles Keepax 02 Nov '17

02 Nov '17

Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Only build tested but Sergej should be able to test on actually hardware for us. Thanks, Charles sound/soc/codecs/wm8741.c | 35 +++++++++++++++++++++-------------- 1 file changed, 21 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index b8c1940f2243..3c96af057a3e 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -196,7 +196,7 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8741_priv *wm8741 = snd_soc_codec_get_drvdata(codec); - u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1FC; + unsigned int iface; int i; /* The set of sample rates that can be supported depends on the @@ -223,15 +223,16 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_width(params)) { case 16: + iface = 0x0; break; case 20: - iface |= 0x0001; + iface = 0x1; break; case 24: - iface |= 0x0002; + iface = 0x2; break; case 32: - iface |= 0x0003; + iface = 0x3; break; default: dev_dbg(codec->dev, "wm8741_hw_params: Unsupported bit size param = %d", @@ -242,7 +243,9 @@ static int wm8741_hw_params(struct snd_pcm_substream *substream, dev_dbg(codec->dev, "wm8741_hw_params: bit size param = %d, rate param = %d", params_width(params), params_rate(params)); - snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); + snd_soc_update_bits(codec, WM8741_FORMAT_CONTROL, WM8741_IWL_MASK, + iface); + return 0; } @@ -295,7 +298,7 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - u16 iface = snd_soc_read(codec, WM8741_FORMAT_CONTROL) & 0x1C3; + unsigned int iface; /* check master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -308,18 +311,19 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - iface |= 0x0008; + iface = 0x08; break; case SND_SOC_DAIFMT_RIGHT_J: + iface = 0x00; break; case SND_SOC_DAIFMT_LEFT_J: - iface |= 0x0004; + iface = 0x04; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x000C; + iface = 0x0C; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x001C; + iface = 0x1C; break; default: return -EINVAL; @@ -330,13 +334,13 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_NB_NF: break; case SND_SOC_DAIFMT_IB_IF: - iface |= 0x0010; + iface |= 0x10; break; case SND_SOC_DAIFMT_IB_NF: - iface |= 0x0020; + iface |= 0x20; break; case SND_SOC_DAIFMT_NB_IF: - iface |= 0x0030; + iface |= 0x30; break; default: return -EINVAL; @@ -347,7 +351,10 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai, fmt & SND_SOC_DAIFMT_FORMAT_MASK, ((fmt & SND_SOC_DAIFMT_INV_MASK))); - snd_soc_write(codec, WM8741_FORMAT_CONTROL, iface); + snd_soc_update_bits(codec, WM8741_FORMAT_CONTROL, + WM8741_BCP_MASK | WM8741_LRP_MASK | WM8741_FMT_MASK, + iface); + return 0; } -- 2.11.0

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[alsa-devel] [PATCH] ASoC: intel: include linux/module.h as needed
by Arnd Bergmann 02 Nov '17

02 Nov '17

The MODULE_DESCRIPTION() macro is only available when including the linux/module.h header. Apparently this is included indirectly from sst-firmware.c in some configurations, but not in others: sound/soc/intel/common/sst-firmware.c:1278:20: error: expected declaration specifiers or '...' before string constant MODULE_DESCRIPTION("Intel SST Firmware Loader"); ^~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/soc/intel/common/sst-firmware.c:1279:16: error: expected declaration specifiers or '...' before string constant This adds the missing include line. Fixes: a395bdd6b24b ("ASoC: intel: Fix sst-dsp dependency on dw stuff") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/common/sst-firmware.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/common/sst-firmware.c b/sound/soc/intel/common/sst-firmware.c index 79a9fdf94d38..657afc02f1c4 100644 --- a/sound/soc/intel/common/sst-firmware.c +++ b/sound/soc/intel/common/sst-firmware.c @@ -19,6 +19,7 @@ #include <linux/sched.h> #include <linux/firmware.h> #include <linux/export.h> +#include <linux/module.h> #include <linux/platform_device.h> #include <linux/dma-mapping.h> #include <linux/dmaengine.h> -- 2.9.0

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[alsa-devel] [PATCH] ASoC: stm32: sai: attempt to fix uninitialized variable
by Arnd Bergmann 02 Nov '17

02 Nov '17

After commit 61fb4ff70377 ("ASoC: stm32: sai: Move static settings to DAI init"), we get a valid warning about the use of an uninitialized variable: sound/soc/stm/stm32_sai_sub.c: In function 'stm32_sai_hw_params': sound/soc/stm/stm32_sai_sub.c:485:7: error: 'cr1' may be used uninitialized in this function [-Werror=maybe-uninitialized] I could not immediately figure out what the code should do instead, this is the hack I applied locally to shut up the warning. It may or may not be correct, please take a look. Fixes: 61fb4ff70377 ("ASoC: stm32: sai: Move static settings to DAI init") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/stm/stm32_sai_sub.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 150ad546d8b9..d8a51fce368a 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -480,6 +480,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai, /* Mode, data format and channel config */ cr1_mask = SAI_XCR1_DS_MASK; + cr1 = 0; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: cr1 |= SAI_XCR1_DS_SET(SAI_DATASIZE_8); -- 2.9.0

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[alsa-devel] [PATCH] ASoC: fix build warning in soc-core.c
by Randy Dunlap 02 Nov '17

02 Nov '17

From: Randy Dunlap <rdunlap(a)infradead.org> Fix kernel-doc build error. A symbol that ends with an underscore character ('_') has special meaning in reST (reStructuredText), so add a '*' to prevent this error and to indicate that there are several of these values to choose from. ../sound/soc/soc-core.c:2799: ERROR: Unknown target name: "snd_soc_daifmt". Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- lnx-414-rc7.orig/sound/soc/soc-core.c +++ lnx-414-rc7/sound/soc/soc-core.c @@ -2796,7 +2796,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_r /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI - * @fmt: SND_SOC_DAIFMT_ format value. + * @fmt: SND_SOC_DAIFMT_* format value. * * Configures the DAI hardware format and clocking. */

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[alsa-devel] Edirol M-16DX broken after fw upgrade
by András Murányi 02 Nov '17

02 Nov '17

Hi List, The Edirol M-16DX has been working out of the box with ALSA (using firmware v1.21), capable of recording 18 channels, until upgraded to firmware v2.0. symptom: When connected via USB, the device's lcd displays "USB ON LINE" which never disappears. Sound still goes through the device but the interface (knobs, buttons) don't react any more. On the computer, the sound device gets listed, but it cannot be opened. alsa-info: http://www.alsa-project.org/db/?f=d094a5abda1b562c6ba572e4dd9425aa30c8a42d dmesg: [18641.439112] usb 2-4: new high-speed USB device number 5 using ehci-pci [18641.554781] usb 2-4: New USB device found, idVendor=0582, idProduct=00c4 [18641.554794] usb 2-4: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [18641.554802] usb 2-4: Product: M-16DX [18641.554807] usb 2-4: Manufacturer: EDIROL [18647.744008] usb 2-4: 1:1: usb_set_interface failed (-110) [18652.743929] usb 2-4: 1:1: usb_set_interface failed (-110) [18657.744786] usb 2-4: 1:1: usb_set_interface failed (-110) [18662.745466] usb 2-4: 1:1: usb_set_interface failed (-110) [18667.745220] usb 2-4: 1:1: usb_set_interface failed (-110) [18672.746071] usb 2-4: 1:1: usb_set_interface failed (-110) [18677.747007] usb 2-4: 1:1: usb_set_interface failed (-110) [18682.747709] usb 2-4: 1:1: usb_set_interface failed (-110) [18687.748416] usb 2-4: 1:1: usb_set_interface failed (-110) [18692.748261] usb 2-4: 1:1: usb_set_interface failed (-110) [18697.749174] usb 2-4: 1:1: usb_set_interface failed (-110) [18702.749893] usb 2-4: 1:1: usb_set_interface failed (-110) I'll be more than happy to provide more info and to test possible solutions. This is a wonderful device and it was working like a charm until the firmware change. Please, if possible, cc your reply to my personal address too. Thanks, András

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