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

[alsa-devel] [PATCH] ASoC: Intel: Atom: Remove unneeded linux/miscdevice.h include
by Corentin Labbe 18 Sep '17

18 Sep '17

No file in sound/soc/intel/ use any miscdevice. This patch remove this uncessary include. Signed-off-by: Corentin Labbe <clabbe.montjoie(a)gmail.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 0e928d54305d..4166925a689b 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -23,7 +23,6 @@ #include <linux/interrupt.h> #include <linux/slab.h> #include <linux/io.h> -#include <linux/miscdevice.h> #include <linux/platform_device.h> #include <linux/firmware.h> #include <linux/pm_runtime.h> -- 2.13.5

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[alsa-devel] [PATCH] ALSA: oxygen: Xonar DG(X): make model_xonar_dg const
by Bhumika Goyal 18 Sep '17

18 Sep '17

Make this const as it not modified anywhere. It is only used during a copy operation. Also, add const to the declaration in header. Signed-off-by: Bhumika Goyal <bhumirks(a)gmail.com> --- sound/pci/oxygen/xonar_dg.h | 2 +- sound/pci/oxygen/xonar_dg_mixer.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/oxygen/xonar_dg.h b/sound/pci/oxygen/xonar_dg.h index d461df3..5a07cda 100644 --- a/sound/pci/oxygen/xonar_dg.h +++ b/sound/pci/oxygen/xonar_dg.h @@ -51,6 +51,6 @@ void dump_cs4245_registers(struct oxygen *chip, void dg_resume(struct oxygen *chip); void dg_cleanup(struct oxygen *chip); -extern struct oxygen_model model_xonar_dg; +extern const struct oxygen_model model_xonar_dg; #endif diff --git a/sound/pci/oxygen/xonar_dg_mixer.c b/sound/pci/oxygen/xonar_dg_mixer.c index b885dac..d22fbe8 100644 --- a/sound/pci/oxygen/xonar_dg_mixer.c +++ b/sound/pci/oxygen/xonar_dg_mixer.c @@ -449,7 +449,7 @@ static int dg_mixer_init(struct oxygen *chip) return 0; } -struct oxygen_model model_xonar_dg = { +const struct oxygen_model model_xonar_dg = { .longname = "C-Media Oxygen HD Audio", .chip = "CMI8786", .init = dg_init, -- 1.9.1

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[alsa-devel] [PATCH] ALSA: compress: Remove unused variable
by Guneshwor Singh 18 Sep '17

18 Sep '17

Commit 04c5d5a430fc ("ALSA: compress: Embed struct device") removed the statement that used 'str' but didn't remove the variable itself. So remove it. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/core/compress_offload.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index fec1dfdb14ad..4490a699030b 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -948,14 +948,13 @@ static const struct file_operations snd_compr_file_ops = { static int snd_compress_dev_register(struct snd_device *device) { int ret = -EINVAL; - char str[16]; struct snd_compr *compr; if (snd_BUG_ON(!device || !device->device_data)) return -EBADFD; compr = device->device_data; - pr_debug("reg %s for device %s, direction %d\n", str, compr->name, + pr_debug("reg device %s, direction %d\n", compr->name, compr->direction); /* register compressed device */ ret = snd_register_device(SNDRV_DEVICE_TYPE_COMPRESS, -- 2.14.1

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[alsa-devel] [PATCH] ALSA: hwdep: prevent a harmless shift wrapping bug
by Dan Carpenter 18 Sep '17

18 Sep '17

The "info.index" variable represents a bit in hw->dsp_loaded which is an unsigned int. If it's higher than 31 we hit a shift wrapping bug. This seems harmless, but I wanted to silence the static checker warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index a73baa1242be..8faae3d1455d 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -228,6 +228,8 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw, memset(&info, 0, sizeof(info)); if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; + if (info.index >= 32) + return -EINVAL; /* check whether the dsp was already loaded */ if (hw->dsp_loaded & (1 << info.index)) return -EBUSY;

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[alsa-devel] [PATCH v2] pcm: softvol: add support for S24_LE
by Jörg Krause 18 Sep '17

18 Sep '17

Tested with the Wolfson WM8524 DAC on a i.MX6UL board and the following ALSA configuration file using the pcm test utility from alsa-lib: """ $ cat /etc/asound.conf pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "hw:0" } control { name "Master" card 0 } } ctl.!default { type hw card 0 } ctl.softvol { type hw card 0 } $ pcm -D softvol -o S24_LE -c 2 -r 48000 """ The data in the Synchronous Audio Interface (SAI) of the i.MX6UL is aligned the following way: """ 31 30 29 28 | 27 26 25 24 | 23 22 21 20 | .. | 3 2 1 0 ## ## ## ## ## ## ## ## [ DATA[23:0] ] """ Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- v2: * add CONVERT_AREA_S24_LE algorithm instead of using the same one as S32_LE --- src/pcm/pcm_softvol.c | 56 ++++++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 53 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 1fe5784d..68a35b56 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -251,6 +251,44 @@ static inline short MULTI_DIV_short(short a, unsigned int b, int swap) } \ } \ } while (0) + +#define CONVERT_AREA_S24_LE() do { \ + unsigned int ch, fr; \ + int *src, *dst; \ + int tmp; \ + for (ch = 0; ch < channels; ch++) { \ + src_area = &src_areas[ch]; \ + dst_area = &dst_areas[ch]; \ + src = snd_pcm_channel_area_addr(src_area, src_offset); \ + dst = snd_pcm_channel_area_addr(dst_area, dst_offset); \ + src_step = snd_pcm_channel_area_step(src_area) \ + / sizeof(int); \ + dst_step = snd_pcm_channel_area_step(dst_area) \ + / sizeof(int); \ + GET_VOL_SCALE; \ + fr = frames; \ + if (! vol_scale) { \ + while (fr--) { \ + *dst = 0; \ + dst += dst_step; \ + } \ + } else if (vol_scale == 0xffff) { \ + while (fr--) { \ + *dst = *src; \ + src += dst_step; \ + dst += src_step; \ + } \ + } else { \ + while (fr--) { \ + tmp = *src << 8; \ + tmp = (signed int) tmp >> 8; \ + *dst = MULTI_DIV_24(tmp, vol_scale); \ + src += dst_step; \ + dst += src_step; \ + } \ + } \ + } \ +} while (0) #define GET_VOL_SCALE \ switch (ch) { \ @@ -315,6 +353,10 @@ static void softvol_convert_stereo_vol(snd_pcm_softvol_t *svol, CONVERT_AREA(int, !snd_pcm_format_cpu_endian(svol->sformat)); break; + case SND_PCM_FORMAT_S24_LE: + /* 24bit samples */ + CONVERT_AREA_S24_LE(); + break; case SND_PCM_FORMAT_S24_3LE: CONVERT_AREA_S24_3LE(); break; @@ -366,6 +408,10 @@ static void softvol_convert_mono_vol(snd_pcm_softvol_t *svol, CONVERT_AREA(int, !snd_pcm_format_cpu_endian(svol->sformat)); break; + case SND_PCM_FORMAT_S24_LE: + /* 24bit samples */ + CONVERT_AREA_S24_LE(); + break; case SND_PCM_FORMAT_S24_3LE: CONVERT_AREA_S24_3LE(); break; @@ -422,6 +468,7 @@ static int snd_pcm_softvol_hw_refine_cprepare(snd_pcm_t *pcm, { (1ULL << SND_PCM_FORMAT_S16_LE) | (1ULL << SND_PCM_FORMAT_S16_BE) | + (1ULL << SND_PCM_FORMAT_S24_LE) | (1ULL << SND_PCM_FORMAT_S32_LE) | (1ULL << SND_PCM_FORMAT_S32_BE), (1ULL << (SND_PCM_FORMAT_S24_3LE - 32)) @@ -577,10 +624,11 @@ static int snd_pcm_softvol_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * param if (slave->format != SND_PCM_FORMAT_S16_LE && slave->format != SND_PCM_FORMAT_S16_BE && slave->format != SND_PCM_FORMAT_S24_3LE && + slave->format != SND_PCM_FORMAT_S24_LE && slave->format != SND_PCM_FORMAT_S32_LE && slave->format != SND_PCM_FORMAT_S32_BE) { - SNDERR("softvol supports only S16_LE, S16_BE, S24_3LE, S32_LE " - " or S32_BE"); + SNDERR("softvol supports only S16_LE, S16_BE, S24_LE, S24_3LE, " + "S32_LE or S32_BE"); return -EINVAL; } svol->sformat = slave->format; @@ -863,6 +911,7 @@ int snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, sformat != SND_PCM_FORMAT_S16_LE && sformat != SND_PCM_FORMAT_S16_BE && sformat != SND_PCM_FORMAT_S24_3LE && + sformat != SND_PCM_FORMAT_S24_LE && sformat != SND_PCM_FORMAT_S32_LE && sformat != SND_PCM_FORMAT_S32_BE) return -EINVAL; @@ -1082,9 +1131,10 @@ int _snd_pcm_softvol_open(snd_pcm_t **pcmp, const char *name, sformat != SND_PCM_FORMAT_S16_LE && sformat != SND_PCM_FORMAT_S16_BE && sformat != SND_PCM_FORMAT_S24_3LE && + sformat != SND_PCM_FORMAT_S24_LE && sformat != SND_PCM_FORMAT_S32_LE && sformat != SND_PCM_FORMAT_S32_BE) { - SNDERR("only S16_LE, S16_BE, S24_3LE, S32_LE or S32_BE format is supported"); + SNDERR("only S16_LE, S16_BE, S24_LE, S24_3LE, S32_LE or S32_BE format is supported"); snd_config_delete(sconf); return -EINVAL; } -- 2.14.1

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[alsa-devel] [RESENT PATCH v9 1/2] ASoC: rt5514: Add devicetree bindings for rt5514-spi
by Jeffy Chen 18 Sep '17

18 Sep '17

Add devicetree bindings documentation file for rt5514 spi dsp codec. Also update rt5514 i2c dt-binding's compatible to distinguish it from rt5514 spi. Signed-off-by: Jeffy Chen <jeffy.chen(a)rock-chips.com> --- Changes in v9: Address comments from Brian. .../devicetree/bindings/sound/rt5514-spi.txt | 29 ++++++++++++++++++++++ Documentation/devicetree/bindings/sound/rt5514.txt | 4 +-- 2 files changed, 31 insertions(+), 2 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5514-spi.txt diff --git a/Documentation/devicetree/bindings/sound/rt5514-spi.txt b/Documentation/devicetree/bindings/sound/rt5514-spi.txt new file mode 100644 index 000000000000..7911d1adbd29 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5514-spi.txt @@ -0,0 +1,29 @@ +RT5514 SPI audio CODEC + +This device supports SPI only. + +Required properties: + +- compatible : "realtek,rt5514" or "realtek,rt5514", "realtek,rt5514-spi". + +- reg : The SPI address of the device. + +Optional properties: + +- interrupt-parent: Should be the phandle for the interrupt controller +- interrupts: The interrupt number to the cpu. The interrupt specifier format + depends on the interrupt controller. + +Example: + +codec: rt5514-spi@0 { + compatible = "realtek,rt5514", "realtek,rt5514-spi"; + reg = <0>; + + pinctrl-names = "default"; + pinctrl-0 = <&mic_int>; + + interrupt-parent = <&gpio1>; + interrupts = <13 IRQ_TYPE_LEVEL_HIGH>; + wakeup-source; +}; diff --git a/Documentation/devicetree/bindings/sound/rt5514.txt b/Documentation/devicetree/bindings/sound/rt5514.txt index 929ca6756b02..841a1159e234 100644 --- a/Documentation/devicetree/bindings/sound/rt5514.txt +++ b/Documentation/devicetree/bindings/sound/rt5514.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "realtek,rt5514". +- compatible : "realtek,rt5514" or "realtek,rt5514", "realtek,rt5514-i2c". - reg : The I2C address of the device. @@ -28,6 +28,6 @@ Pins on the device (for linking into audio routes) for RT5514: Example: codec: rt5514@57 { - compatible = "realtek,rt5514"; + compatible = "realtek,rt5514", "realtek,rt5514-i2c"; reg = <0x57>; }; -- 2.11.0

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[alsa-devel] Problems with Chromebook Pixel 2015 audio
by Tom Rini 18 Sep '17

18 Sep '17

Hey folks, Entirely unrelated to the regression on this board that we dealt with the other week, I have (and have had for a bit), a problem with audio on this laptop. After a while of use, I'll have a problem like this: Sep 12 20:59:30 oliver kernel: haswell-pcm-audio haswell-pcm-audio: FW loaded, mailbox readback FW info: type 01, - version: 00.00, build 77, source commit id: 876ac6906f31a43b6772b23c7c983ce9dcb18a19 (everything is fine, audio works, is in use) Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: error: audio DSP boot timeout IPCD 0x0 IPCX 0x0 Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: ipc: --message timeout-- ipcx 0x83000000 isr 0x00000000 ipcd 0x00000000 imrx 0x7fff0000 Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: error: stream commit failed Sep 12 21:24:04 oliver kernel: System PCM: error: failed to commit stream -110 Sep 12 21:24:04 oliver kernel: haswell-pcm-audio haswell-pcm-audio: ASoC: haswell-pcm-audio hw params failed: -110 Sep 12 21:24:04 oliver kernel: System PCM: ASoC: hw_params FE failed -110 With those last 4 messages repeating constantly (~3 sets per second, and the above are from v4.13, but I've had similar for as long as the driver has been mainline I think) and it only stops when I reboot. Rebooting may or may not clear the problem, last night I had to power off to get audio to come back up. I'm not quite sure where to begin debugging this issue. Does anyone have any suggestions? Thanks! -- Tom

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Re: [alsa-devel] [PATCH v2] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Vinod Koul 18 Sep '17

18 Sep '17

On Sun, Sep 10, 2017 at 11:03:20AM +0530, Naveen M wrote: > rt5663 needs mclk/sclk early to synchronize its internal clocks. > Clocks are enabled in hw_params and disabled in platform_clk_control > as the codec requires mclk/bclk need to turned ON before set_sysclk > call for a successful clock synchronization. ah there was a v2, sorry missed that. > > Depends-on: patch series ("ASoC: Intel: Skylake: Add a clk driver to > enable ssp clks early") > ASoC: Intel: kbl: Enable mclk and ssp sclk early > ASoC: Intel: Skylake: Modify skl_dsp_set_dma_control API arguments > ASoC: Intel: Skylake: Parse nhlt to populate clock information > ASoC: Intel: Skylake: Prepare DMA control IPC to enable/disable clock > ASoC: Intel: Skylake: Register clock device and ops > ASoC: Intel: Skylake: Add ssp clock driver this is good info but is useless for applying > > Signed-off-by: Naveen M <naveen.m(a)intel.com> > Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> > > V2: Add Depends-on you have removed the dashes after s-o-b, so when I apply this patch the depends on, V2 also get applied. It is go d practice to apply the .patch file yourself after hand editing and seeing if the output is something you really want. Can you fix this along with comments on V1. -- ~Vinod

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Re: [alsa-devel] [PATCH] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Vinod Koul 18 Sep '17

18 Sep '17

On Sun, Sep 10, 2017 at 08:32:59AM +0530, Naveen M wrote: > rt5663 needs mclk/sclk early to synchronize its internal clocks. > Clocks are enabled in hw_params and disabled in platform_clk_control > as the codec requires mclk/bclk need to turned ON before set_sysclk > call for a successful clock synchronization. This would be dependent on the series Subhransu posted right? If so please do mention that > +static int kabylake_enable_ssp_clks(struct snd_soc_card *card) > +{ > + we don't need the empty line here > + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); > + int ret; > + > + /* Enable MCLK */ > + ret = clk_set_rate(priv->mclk, 24000000); > + if (ret < 0) { > + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", ret); > + return ret; > + } > + > + ret = clk_prepare_enable(priv->mclk); > + if (ret < 0) { > + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); > + return ret; > + } > + > + /* Enable SCLK */ > + ret = clk_set_rate(priv->sclk, 3072000); why do we need both mclk and sclk for this codec? One should be enough for codec to sync > + if (ret < 0) { > + dev_err(card->dev, "Can't set rate for sclk, err: %d\n", ret); > + clk_disable_unprepare(priv->mclk); > + return ret; > + } > + > + ret = clk_prepare_enable(priv->sclk); > + if (ret < 0) { > + dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); > + clk_disable_unprepare(priv->mclk); > + } > + > + return 0; this looks wrong to me, shouldn't this be return ret; > @@ -647,6 +710,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) > { > struct kbl_codec_private *ctx; > struct skl_machine_pdata *pdata; > + int ret = 0; > > ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); > if (!ctx) > @@ -662,6 +726,34 @@ static int kabylake_audio_probe(struct platform_device *pdev) > dmic_constraints = pdata->dmic_num == 2 ? > &constraints_dmic_2ch : &constraints_dmic_channels; > > + ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); > + if (IS_ERR(ctx->mclk)) { > + ret = PTR_ERR(ctx->mclk); > + if (ret == -ENOENT) { > + dev_info(&pdev->dev, > + "Failed to get ssp1_mclk, defer probe\n"); can you check this, I think driver core emits warnings as well, so do we need it here? > + return -EPROBE_DEFER; > + } > + > + dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", > + ret); > + return ret; > + } > + > + ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); > + if (IS_ERR(ctx->sclk)) { > + ret = PTR_ERR(ctx->sclk); > + if (ret == -ENOENT) { > + dev_info(&pdev->dev, > + "Failed to get ssp1_sclk, defer probe\n"); shouldn't the mclk ref be given back? -- ~Vinod

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[alsa-devel] [PATCH] ASoC: Intel: Headset button support in kabylake machine driver
by Naveen M 18 Sep '17

18 Sep '17

Patch adds headset button support for kabylake machine driver (kbl_rt5663_max98927). Signed-off-by: Naveen M <naveen.m(a)intel.com> diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 7f76074..e45b6fe 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -17,6 +17,7 @@ * GNU General Public License for more details. */ +#include <linux/input.h> #include <linux/module.h> #include <linux/platform_device.h> #include <sound/core.h> @@ -208,6 +209,7 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) int ret; struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_jack *jack; /* * Headset buttons map to the google Reference headset. @@ -221,6 +223,13 @@ static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); return ret; } + + jack = &ctx->kabylake_headset; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + rt5663_set_jack_detect(codec, &ctx->kabylake_headset); return ret; } -- 1.9.1

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