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

[alsa-devel] [PATCH v2] ASoC: topology: Fix bugs of freeing soc topology.
by Liam Girdwood 26 Mar '18

26 Mar '18

From: Yan Wang <yan.wang(a)linux.intel.com> In snd_soc_tplg_component_remove(), it should compare index and not dobj->index with SND_SOC_TPLG_INDEX_ALL for removing all topology objects. Signed-off-by: Yan Wang <yan.wang(a)linux.intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Notes: Sign-off added since V1. sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index b363f2409f72..e5ee7a5a2b02 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2563,7 +2563,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) /* match index */ if (dobj->index != index && - dobj->index != SND_SOC_TPLG_INDEX_ALL) + index != SND_SOC_TPLG_INDEX_ALL) continue; switch (dobj->type) { -- 2.14.1

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[alsa-devel] [PATCH] ALSA: pcm: Use dma_bytes as size parameter in dma_mmap_coherent()
by Stefan Roese 26 Mar '18

26 Mar '18

When trying to use the driver (e.g. aplay *.wav), the 4MiB DMA buffer will get mmapp'ed in 16KiB chunks. But this fails with the 2nd 16KiB area, as the page offset is outside of the VMA range (size), which is currently used as size parameter in snd_pcm_lib_default_mmap(). By using the DMA buffer size (dma_bytes) instead, the complete DMA buffer can be mmapp'ed and the issue is fixed. This issue was detected on an ARM platform (TI AM57xx) using the RME HDSP MADI PCIe soundcard. Signed-off-by: Stefan Roese <sr(a)denx.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 77ba50ddcf9e..d18b3982548b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3422,7 +3422,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area, substream->runtime->dma_area, substream->runtime->dma_addr, - area->vm_end - area->vm_start); + substream->runtime->dma_bytes); #endif /* CONFIG_X86 */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; -- 2.16.3

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[alsa-devel] [PATCH 1/2] ASoC: mediatek: remove redundant MODULE_ALIAS entries
by sean.wang＠mediatek.com 26 Mar '18

26 Mar '18

From: Sean Wang <sean.wang(a)mediatek.com> MODULE_ALIAS exports information to allow the module to be auto-loaded at boot for the drivers registered using legacy platform registration. However, currently the driver is always used by DT-only platform, MODULE_ALIAS is redundant and should be removed properly. Signed-off-by: Sean Wang <sean.wang(a)mediatek.com> --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 1 - sound/soc/mediatek/mt2701/mt2701-wm8960.c | 1 - sound/soc/mediatek/mt8173/mt8173-max98090.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650.c | 1 - 6 files changed, 6 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 70f61d5..f664b9d 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -409,4 +409,3 @@ module_platform_driver(mt2701_cs42448_machine); MODULE_DESCRIPTION("MT2701 CS42448 ALSA SoC machine driver"); MODULE_AUTHOR("Ir Lian <ir.lian(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("mt2701 cs42448 soc card"); diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index a08ce23..a137057 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -172,5 +172,4 @@ module_platform_driver(mt2701_wm8960_machine); MODULE_DESCRIPTION("MT2701 WM8960 ALSA SoC machine driver"); MODULE_AUTHOR("Ryder Lee <ryder.lee(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("mt2701 wm8960 soc card"); diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index e0c2b23..6e58af6 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -203,5 +203,4 @@ module_platform_driver(mt8173_max98090_driver); MODULE_DESCRIPTION("MT8173 MAX98090 ALSA SoC machine driver"); MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:mt8173-max98090"); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 5a9a548..addbbb3 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -251,5 +251,4 @@ module_platform_driver(mt8173_rt5650_rt5514_driver); MODULE_DESCRIPTION("MT8173 RT5650 and RT5514 SoC machine driver"); MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:mtk-rt5650-rt5514"); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index b724808..f46e4c8 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -308,5 +308,4 @@ module_platform_driver(mt8173_rt5650_rt5676_driver); MODULE_DESCRIPTION("MT8173 RT5650 and RT5676 SoC machine driver"); MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:mtk-rt5650-rt5676"); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 679fc8b..e2e42e6 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -335,5 +335,4 @@ module_platform_driver(mt8173_rt5650_driver); MODULE_DESCRIPTION("MT8173 RT5650 SoC machine driver"); MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:mtk-rt5650"); -- 2.7.4

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[alsa-devel] Roland SD50 issues
by Keith A. Milner 24 Mar '18

24 Mar '18

In my occasional, but ongoing, quest to try to debug some of the non- functioning audio on newer Roland/Boss devices, I've managed to get hold of a Roland SD-50 to do some comparisons. The lsusb -v and the aplay -L output for the SD-50 are attached. This device works with jack if I use a sample rate of 44100, buffer size 256, 3 periods/buffer. I can record and playback with no obvious issues. However, I can't seem to get this device working directly with ALSA. If I try playing back via audacity, the meter bars move, but there's no audio output. If I try aplay as follows: aplay -Dplughw:CARD=SD50 /usr/share/sounds/alsa/Front_Center.wav It pauses for a second as if it's playing, but I get no audio output. If I use the -v flag for aplay I get the following: Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Plug PCM: Rate conversion PCM (44100, sformat=S16_LE) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 Slave: Route conversion PCM (sformat=S24_3LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22051 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 16539 stop_threshold : 22051 silence_threshold: 0 silence_size : 0 boundary : 6206804711446675456 Slave: Hardware PCM card 2 'SD-50' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S24_3LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 24 buffer_size : 22051 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 16539 stop_threshold : 22051 silence_threshold: 0 silence_size : 0 boundary : 6206804711446675456 appl_ptr : 0 hw_ptr : 0 I'm trying to work out why I'm unable to get any audio output with ALSA directly, whilst it happily works with Jack. Any pointers are gratefully received. Cheers, Keith

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[alsa-devel] [PATCH 0/2] ALSA: usb-audio: UAC2 jack detection patchset
by Andrew Chant 24 Mar '18

24 Mar '18

These patches implement UAC2 jack detection. This allows USB-to-3.5mm adapters and similar USB devices to report jack state via alsa mixers, using the reporting provided by the UAC2 connector control. The first patch fixes up the function used to query UAC controls. The long lines in the table are unfortunate but keep the table readable. The long lines elsewhere in the patch were in line with existing long lines due to current code structure. Andrew Chant (2): ALSA: usb-audio: fix uac control query argument ALSA: usb-audio: UAC2 jack detection include/linux/usb/audio-v2.h | 4 +- sound/usb/clock.c | 5 +- sound/usb/mixer.c | 168 +++++++++++++++++++++++++++++------ 3 files changed, 146 insertions(+), 31 deletions(-) -- 2.17.0.rc0.231.g781580f067-goog

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[alsa-devel] [PATCH v3 0/2] UAC2 jack detection and clock validity
by Andrew Chant 24 Mar '18

24 Mar '18

These patches implement UAC2 jack detection. This allows USB-to-3.5mm adapters and similar USB devices to report jack state via alsa control, using the reporting provided by the UAC2 connector control. The second patch updates the clock validity control so it doesn't show up in mixer UIs. I declined to mark the controls as VOLATILE since clients should always be notified of changes, since UAC devices are expected to generate interrupt messages when a control value changes. v3 changes: - move channel count correction to first patch. - fix incorrect comment formatting and spacing. v2 changes: - use SNDRV_CTL_ELEM_IFACE_CARD instead of _MIXER - correct channel count to 1, cmask to 0. - add validity patch. Andrew Chant (2): ALSA: usb-audio: UAC2 jack detection ALSA: usb-audio: update clock valid control sound/usb/mixer.c | 117 ++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 109 insertions(+), 8 deletions(-) -- 2.17.0.rc0.231.g781580f067-goog

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[alsa-devel] [PATCH v2 0/2] UAC2 jack detection and clock validity
by Andrew Chant 24 Mar '18

24 Mar '18

These patches implement UAC2 jack detection. This allows USB-to-3.5mm adapters and similar USB devices to report jack state via alsa control, using the reporting provided by the UAC2 connector control. The second patch updates the clock validity control so it doesn't show up in mixer UIs. I declined to mark the controls as VOLATILE since clients should always be notified of changes, since UAC devices are expected to generate interrupt messages when a control value changes. v2 changes: - use SNDRV_CTL_ELEM_IFACE_CARD instead of _MIXER - correct channel count to 1, cmask to 0. - add validity patch. Andrew Chant (2): ALSA: usb-audio: UAC2 jack detection ALSA: usb-audio: update clock valid control sound/usb/mixer.c | 117 ++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 109 insertions(+), 8 deletions(-) -- 2.17.0.rc0.231.g781580f067-goog

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[alsa-devel] [PATCH 3/3] ASoC: TSCS42xx: Add CCF support to get sysclk
by Steven Eckhoff 24 Mar '18

24 Mar '18

The TSCS42xx relies on set_sysclk to get a unique clock id and rate, which prevents it from being used with the simple-card. Remove set_sysclk callback Add CCF support to get clock id and rate Add clocks and clock-names to device tree binding Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- .../devicetree/bindings/sound/tscs42xx.txt | 6 ++ sound/soc/codecs/tscs42xx.c | 98 ++++++++++++++-------- sound/soc/codecs/tscs42xx.h | 2 +- 3 files changed, 69 insertions(+), 37 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt index 2ac2f0996697..7eea32e9d078 100644 --- a/Documentation/devicetree/bindings/sound/tscs42xx.txt +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -8,9 +8,15 @@ Required Properties: - reg : <0x71> for analog mic <0x69> for digital mic + - clock-names: Must one of the following "mclk1", "xtal", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + Example: wookie: codec@69 { compatible = "tempo,tscs42A2"; reg = <0x69>; + clock-names = "xtal"; + clocks = <&audio_xtal>; }; diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d41852a37ac6..081af0e05d33 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/delay.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/tlv.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -40,6 +41,9 @@ struct tscs42xx { struct mutex pll_lock; struct regmap *regmap; + + struct clk *sysclk; + int sysclk_src_id; }; struct coeff_ram_ctl { @@ -1251,13 +1255,46 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return 0; } -static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) +static const struct snd_soc_dai_ops tscs42xx_dai_ops = { + .hw_params = tscs42xx_hw_params, + .mute_stream = tscs42xx_mute_stream, + .set_fmt = tscs42xx_set_dai_fmt, + .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, +}; + +static int part_is_valid(struct tscs42xx *tscs42xx) { - struct snd_soc_component *component = codec_dai->component; + int val; int ret; + unsigned int reg; - switch (clk_id) { + ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); + if (ret < 0) + return ret; + + val = reg << 8; + ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); + if (ret < 0) + return ret; + + val |= reg; + + switch (val) { + case 0x4A74: + case 0x4A73: + return true; + default: + return false; + }; +} + +static int set_sysclk(struct snd_soc_component *component) +{ + struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); + unsigned long freq; + int ret; + + switch (tscs42xx->sysclk_src_id) { case TSCS42XX_PLL_SRC_XTAL: case TSCS42XX_PLL_SRC_MCLK1: ret = snd_soc_component_write(component, R_PLLREFSEL, @@ -1285,6 +1322,7 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } + freq = clk_get_rate(tscs42xx->sysclk); ret = set_pll_ctl_from_input_freq(component, freq); if (ret < 0) { dev_err(component->dev, @@ -1295,41 +1333,13 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static const struct snd_soc_dai_ops tscs42xx_dai_ops = { - .hw_params = tscs42xx_hw_params, - .mute_stream = tscs42xx_mute_stream, - .set_fmt = tscs42xx_set_dai_fmt, - .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, - .set_sysclk = tscs42xx_set_dai_sysclk, -}; - -static int part_is_valid(struct tscs42xx *tscs42xx) +static int tscs42xx_probe(struct snd_soc_component *component) { - int val; - int ret; - unsigned int reg; - - ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); - if (ret < 0) - return ret; - - val = reg << 8; - ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); - if (ret < 0) - return ret; - - val |= reg; - - switch (val) { - case 0x4A74: - case 0x4A73: - return true; - default: - return false; - }; + return set_sysclk(component); } static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { + .probe = tscs42xx_probe, .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1387,11 +1397,15 @@ static const struct reg_sequence tscs42xx_patch[] = { { R_AIC2, RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED }, }; +static char const * const src_names[TSCS42XX_PLL_SRC_CNT] = { + "xtal", "mclk1", "mclk2"}; + static int tscs42xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct tscs42xx *tscs42xx; - int ret = 0; + int src; + int ret; tscs42xx = devm_kzalloc(&i2c->dev, sizeof(*tscs42xx), GFP_KERNEL); if (!tscs42xx) { @@ -1402,6 +1416,18 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, tscs42xx); + for (src = TSCS42XX_PLL_SRC_XTAL; src < TSCS42XX_PLL_SRC_CNT; src++) { + tscs42xx->sysclk = devm_clk_get(&i2c->dev, src_names[src]); + if (!IS_ERR(tscs42xx->sysclk)) { + break; + } else if (PTR_ERR(tscs42xx->sysclk) != -ENOENT) { + ret = PTR_ERR(tscs42xx->sysclk); + dev_err(&i2c->dev, "Failed to get sysclk (%d)\n", ret); + return ret; + } + } + tscs42xx->sysclk_src_id = src; + tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); if (IS_ERR(tscs42xx->regmap)) { ret = PTR_ERR(tscs42xx->regmap); diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index d4a30bcbf64b..814c8f3c4a68 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -7,10 +7,10 @@ #define __WOOKIE_H__ enum { - TSCS42XX_PLL_SRC_NONE, TSCS42XX_PLL_SRC_XTAL, TSCS42XX_PLL_SRC_MCLK1, TSCS42XX_PLL_SRC_MCLK2, + TSCS42XX_PLL_SRC_CNT, }; #define R_HPVOLL 0x0 -- 2.15.1

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[alsa-devel] [PATCH 1/3] ALSA: usb-audio: Add native DSD support for Luxman DA-06
by Nobutaka Okabe 23 Mar '18

23 Mar '18

Add native DSD support quirk for Luxman DA-06 DAC, by adding the PID/VID 1852:5065. Rename "is_marantz_denon_dac()" function to "is_itf_usb_dsd_2alts_dac()" to cover broader device family sharing the same USB audio implementation(*). For the same reason, rename "is_teac_dsd_dac()" function to "is_itf_usb_dsd_3alts_dac()". (*) These devices have the same USB controller "ITF-USB DSD", supplied by INTERFACE Co., Ltd. "ITF-USB DSD" USB controller has two patterns, Pattern 1. (2 altsets version) - Altset 0: for control - Altset 1: for stream (S32) - Altset 2: for stream (S32, DSD_U32) Pattern 2. (3 altsets version) - Altset 0: for control - Altset 1: for stream (S16) - Altset 2: for stream (S32) - Altset 3: for stream (S32, DSD_U32) "is_itf_usb_dsd_2alts_dac()" returns true, if the DAC has "Pattern 1" USB controller, and "is_itf_usb_dsd_3alts_dac()" returns true, if "Pattern2". Signed-off-by: Nobutaka Okabe <nob77413(a)gmail.com> --- sound/usb/quirks.c | 29 ++++++++++++++++------------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 794224e1d6df..006da37ad0d9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1149,24 +1149,27 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) return false; } -/* Marantz/Denon USB DACs need a vendor cmd to switch +/* ITF-USB DSD based DACs need a vendor cmd to switch * between PCM and native DSD mode + * (2 altsets version) */ -static bool is_marantz_denon_dac(unsigned int id) +static bool is_itf_usb_dsd_2alts_dac(unsigned int id) { switch (id) { case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + case USB_ID(0x1852, 0x5065): /* Luxman DA-06 */ return true; } return false; } -/* TEAC UD-501/UD-503/NT-503 USB DACs need a vendor cmd to switch - * between PCM/DOP and native DSD mode +/* ITF-USB DSD based DACs need a vendor cmd to switch + * between PCM and native DSD mode + * (3 altsets version) */ -static bool is_teac_dsd_dac(unsigned int id) +static bool is_itf_usb_dsd_3alts_dac(unsigned int id) { switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ @@ -1183,7 +1186,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct usb_device *dev = subs->dev; int err; - if (is_marantz_denon_dac(subs->stream->chip->usb_id)) { + if (is_itf_usb_dsd_2alts_dac(subs->stream->chip->usb_id)) { /* First switch to alt set 0, otherwise the mode switch cmd * will not be accepted by the DAC */ @@ -1204,7 +1207,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); - } else if (is_teac_dsd_dac(subs->stream->chip->usb_id)) { + } else if (is_itf_usb_dsd_3alts_dac(subs->stream->chip->usb_id)) { /* Vendor mode switch cmd is required. */ switch (fmt->altsetting) { case 3: /* DSD mode (DSD_U32) requested */ @@ -1300,10 +1303,10 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); - /* Marantz/Denon devices with USB DAC functionality need a delay + /* ITF-USB DSD based DACs functionality need a delay * after each class compliant request */ - if (is_marantz_denon_dac(chip->usb_id) + if (is_itf_usb_dsd_2alts_dac(chip->usb_id) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); @@ -1390,14 +1393,14 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; } - /* Denon/Marantz devices with USB DAC functionality */ - if (is_marantz_denon_dac(chip->usb_id)) { + /* ITF-USB DSD based DACs (2 altsets version) */ + if (is_itf_usb_dsd_2alts_dac(chip->usb_id)) { if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } - /* TEAC devices with USB DAC functionality */ - if (is_teac_dsd_dac(chip->usb_id)) { + /* ITF-USB DSD based DACs (3 altsets version) */ + if (is_itf_usb_dsd_3alts_dac(chip->usb_id)) { if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } -- 2.11.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Add native DSD support for TEAC UD-301
by Nobutaka Okabe 23 Mar '18

23 Mar '18

Add native DSD support quirk for TEAC UD-301 DAC, by adding the PID/VID 0644:804a. Signed-off-by: Nobutaka Okabe <nob77413(a)gmail.com> --- Annotation: This patch should be applied before the "Luxman DA-06" patch set. Previously, I sent this patch to alsa-dev ML, but the status is "pending". So, I send again. sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ea8f3de92fa4..794224e1d6df 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1171,6 +1171,7 @@ static bool is_teac_dsd_dac(unsigned int id) switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ case USB_ID(0x0644, 0x8044): /* Esoteric D-05X */ + case USB_ID(0x0644, 0x804a): /* TEAC UD-301 */ return true; } return false; -- 2.11.1

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