January 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 1/3] ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format
by Lars-Peter Clausen 29 Jan '16

29 Jan '16

The DAI link format should be specified for the whole link rather than just one component on the link. So move the format specification for the HDMI audio link from the CPU component to the link itself. Since the sh-mobile-hdmi DAI driver doesn't implement the set_fmt() callback in this case there is no functional difference between only specifying the the format for the CPU side or for the whole link, but the later it will allow us to remove support for just specifying the format for one component. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- No changes since v1. --- arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 6d949f1..75de26c 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -1040,9 +1040,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "sh_mobile_hdmi-hifi", -- 1.8.0

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[alsa-devel] [PATCH] usb: Add native DSD support for PS Audio NuWave DAC
by Jurgen Kramer 29 Jan '16

29 Jan '16

This patch adds native DSD support for the PS Audio NuWave DAC. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 75e9ecb..f437527 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1296,6 +1296,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */ case USB_ID(0x20b1, 0x3023): /* Aune X1S 32BIT/384 DSD DAC */ + case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.5.0

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[alsa-devel] [PATCH] usb: Fix OPPO HA-1 vendor ID
by Jurgen Kramer 29 Jan '16

29 Jan '16

In my patch adding native DSD support for the Oppo HA-1, the wrong vendor ID got through. This patch fixes the vendor ID and aligns the comment. Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- sound/usb/quirks.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 16364d9..75e9ecb 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1287,7 +1287,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ - case USB_ID(0x22d8, 0x0416): /* OPPO HA-1*/ + case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.5.0

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[alsa-devel] [PATCH] amixer: add support for TLV byte control read
by Vinod Koul 29 Jan '16

29 Jan '16

TLV byte control are new type of byte controls added in kernel where controls can have large sizes. For these controls querying with 4096 size fails, so use the queried size to read the control This fixes the crash with current cget/contents on these type of controls Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- amixer/amixer.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index db1849333da3..cfe13592347f 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -588,7 +588,7 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, int level) { int err; - unsigned int item, idx, count, *tlv; + unsigned int item, idx, count, *tlv, tlv_sz; snd_ctl_elem_type_t type; snd_ctl_elem_id_t *id; snd_ctl_elem_info_t *info; @@ -682,13 +682,20 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, __skip_read: if (!snd_ctl_elem_info_is_tlv_readable(info)) goto __skip_tlv; - tlv = malloc(4096); - if ((err = snd_hctl_elem_tlv_read(elem, tlv, 4096)) < 0) { + + if (type == SND_CTL_ELEM_TYPE_BYTES) + tlv_sz = count + 2 * sizeof(unsigned int); + else + tlv_sz = 4096; + + tlv = malloc(tlv_sz); + + if ((err = snd_hctl_elem_tlv_read(elem, tlv, tlv_sz)) < 0) { error("Control %s element TLV read error: %s\n", card, snd_strerror(err)); free(tlv); return err; } - decode_tlv(strlen(space), tlv, 4096); + decode_tlv(strlen(space), tlv, tlv_sz); free(tlv); } __skip_tlv: -- 1.9.1

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[alsa-devel] [PATCH] ALSA: hda - disable dynamic clock gating on Broxton before reset
by libin.yang＠linux.intel.com 29 Jan '16

29 Jan '16

From: Libin Yang <libin.yang(a)linux.intel.com> On Broxton, to make sure the reset controller works properly, MISCBDCGE bit (bit 6) in CGCTL (0x48) of PCI configuration space need be cleared before reset and set back to 1 after reset. Otherwise, it may prevent the CORB/RIRB logic from being reset. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index aeaa9a5..12c5442 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -90,6 +90,8 @@ enum { #define NVIDIA_HDA_ENABLE_COHBIT 0x01 /* Defines for Intel SCH HDA snoop control */ +#define INTEL_HDA_CGCTL 0x48 +#define INTEL_HDA_CGCTL_MISCBDCGE (0x1 << 6) #define INTEL_SCH_HDA_DEVC 0x78 #define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11) @@ -536,10 +538,21 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) { struct hdac_bus *bus = azx_bus(chip); struct pci_dev *pci = chip->pci; + u32 val; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, true); + if (IS_BROXTON(pci)) { + pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val); + val = val & (~(INTEL_HDA_CGCTL_MISCBDCGE)); + pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); + } azx_init_chip(chip, full_reset); + if (IS_BROXTON(pci)) { + pci_read_config_dword(pci, INTEL_HDA_CGCTL, &val); + val = val | INTEL_HDA_CGCTL_MISCBDCGE; + pci_write_config_dword(pci, INTEL_HDA_CGCTL, val); + } if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) snd_hdac_set_codec_wakeup(bus, false); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt5616: Add support sample rate to 192KHz
by Xing Zheng 29 Jan '16

29 Jan '16

Reference the TRM, the ALC5616 support one 24bit/8KHz ~ 192KHz I2S/PCM Interface for stereo DAC and stereo ADC. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- sound/soc/codecs/rt5616.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 1c10d8e..3220e5c 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1218,7 +1218,7 @@ static int rt5616_resume(struct snd_soc_codec *codec) #define rt5616_resume NULL #endif -#define RT5616_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5616_STEREO_RATES SNDRV_PCM_RATE_8000_192000 #define RT5616_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -- 1.7.9.5

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[alsa-devel] [PATCH 1/6] ASoC: wm5110: Fix up snd_soc_register_platform error path
by Charles Keepax 29 Jan '16

29 Jan '16

Whilst there is nothing wrong with the error path here it looks a little bit odd, this patches makes the code a little more idiomatic. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 6088d30..83760b7 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2366,7 +2366,7 @@ static int wm5110_probe(struct platform_device *pdev) ret = snd_soc_register_platform(&pdev->dev, &wm5110_compr_platform); if (ret < 0) { dev_err(&pdev->dev, "Failed to register platform: %d\n", ret); - goto error; + return ret; } ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110, @@ -2376,7 +2376,6 @@ static int wm5110_probe(struct platform_device *pdev) snd_soc_unregister_platform(&pdev->dev); } -error: return ret; } -- 2.1.4

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[alsa-devel] [PATCH] ASoC: cs47l24: Add voice control compressed stream
by Richard Fitzgerald 29 Jan '16

29 Jan '16

This patch adds firmware compressed stream capture support and DAI hookups for voice control firmware (based on the wm5110 implementation.) Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs47l24.c | 98 +++++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 96 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index dc5ae7f..1992077 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -779,6 +779,9 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "Voice Control DSP", NULL, "DSP3" }, + { "Voice Control DSP", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, { "IN1R PGA", NULL, "IN1R" }, @@ -973,12 +976,68 @@ static struct snd_soc_dai_driver cs47l24_dai[] = { .symmetric_rates = 1, .symmetric_samplebits = 1, }, + { + .name = "cs47l24-cpu-voicectrl", + .capture = { + .stream_name = "Voice Control CPU", + .channels_min = 1, + .channels_max = 1, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .compress_new = snd_soc_new_compress, + }, + { + .name = "cs47l24-dsp-voicectrl", + .capture = { + .stream_name = "Voice Control DSP", + .channels_min = 1, + .channels_max = 1, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + }, }; +static int cs47l24_open(struct snd_compr_stream *stream) +{ + struct snd_soc_pcm_runtime *rtd = stream->private_data; + struct cs47l24_priv *priv = snd_soc_codec_get_drvdata(rtd->codec); + struct arizona *arizona = priv->core.arizona; + int n_adsp; + + if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) { + n_adsp = 2; + } else { + dev_err(arizona->dev, + "No suitable compressed stream for DAI '%s'\n", + rtd->codec_dai->name); + return -EINVAL; + } + + return wm_adsp_compr_open(&priv->core.adsp[n_adsp], stream); +} + +static irqreturn_t cs47l24_adsp2_irq(int irq, void *data) +{ + struct cs47l24_priv *priv = data; + struct arizona *arizona = priv->core.arizona; + int ret; + + ret = wm_adsp_compr_handle_irq(&priv->core.adsp[2]); + if (ret == -ENODEV) { + dev_err(arizona->dev, "Spurious compressed data IRQ\n"); + return IRQ_NONE; + } + + return IRQ_HANDLED; +} + static int cs47l24_codec_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct cs47l24_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->core.arizona; int ret; priv->core.arizona->dapm = dapm; @@ -987,6 +1046,14 @@ static int cs47l24_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); + ret = arizona_request_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, + "ADSP2 Compressed IRQ", cs47l24_adsp2_irq, + priv); + if (ret != 0) { + dev_err(codec->dev, "Failed to request DSP IRQ: %d\n", ret); + return ret; + } + ret = wm_adsp2_codec_probe(&priv->core.adsp[1], codec); if (ret) goto err_adsp2_codec_probe; @@ -1014,13 +1081,14 @@ err_adsp2_codec_probe: static int cs47l24_codec_remove(struct snd_soc_codec *codec) { struct cs47l24_priv *priv = snd_soc_codec_get_drvdata(codec); - + struct arizona *arizona = priv->core.arizona; wm_adsp2_codec_remove(&priv->core.adsp[1], codec); wm_adsp2_codec_remove(&priv->core.adsp[2], codec); priv->core.arizona->dapm = NULL; + arizona_free_irq(arizona, ARIZONA_IRQ_DSP_IRQ1, priv); return 0; } @@ -1057,6 +1125,19 @@ static struct snd_soc_codec_driver soc_codec_dev_cs47l24 = { .num_dapm_routes = ARRAY_SIZE(cs47l24_dapm_routes), }; +static struct snd_compr_ops cs47l24_compr_ops = { + .open = cs47l24_open, + .free = wm_adsp_compr_free, + .set_params = wm_adsp_compr_set_params, + .get_caps = wm_adsp_compr_get_caps, + .trigger = wm_adsp_compr_trigger, + .pointer = wm_adsp_compr_pointer, + .copy = wm_adsp_compr_copy, +}; + +static struct snd_soc_platform_driver cs47l24_compr_platform = { + .compr_ops = &cs47l24_compr_ops, +}; static int cs47l24_probe(struct platform_device *pdev) { struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); @@ -1120,12 +1201,25 @@ static int cs47l24_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24, + ret = snd_soc_register_platform(&pdev->dev, &cs47l24_compr_platform); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register platform: %d\n", ret); + return ret; + } + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24, cs47l24_dai, ARRAY_SIZE(cs47l24_dai)); + + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); + snd_soc_unregister_platform(&pdev->dev); + } + + return ret; } static int cs47l24_remove(struct platform_device *pdev) { + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: fsl: select SND_SOC_FSL_SAI or SND_SOC_FSL_SSI depending on SoC type
by Lothar Waßmann 29 Jan '16

29 Jan '16

i.MX6UL does not provide an SSI interface like the other i.MX6 SoCs, but only an SAI interface. Select the appropriate interface(s) depending on the enabled SoC types. Signed-off-by: Lothar Waßmann <LW(a)KARO-electronics.de> --- sound/soc/fsl/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 14dfdee..c128823 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -258,7 +258,8 @@ config SND_SOC_IMX_SGTL5000 select SND_SOC_SGTL5000 select SND_SOC_IMX_PCM_DMA select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI + select SND_SOC_FSL_SAI if SOC_IMX6UL + select SND_SOC_FSL_SSI if SOC_IMX6Q || SOC_IMX6SL || SOC_IMX6SX help Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. -- 2.1.4

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[alsa-devel] ESi Juli@ XTe ALSA support?
by Erick Omar Delgado 29 Jan '16

29 Jan '16

Dear ALSA developers, I would like to know if the ESi Juli@ XTe (http://www.esi-audio.com/pro ducts/juliaxte/) is supported on ALSA for music/audio production? Thank you for your time. Regards, Erick Omar Delgado

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