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

[alsa-devel] No Jack sound on Lenovo dock with a T470 connected
by dmummenschanz＠web.de 30 Mar '17

30 Mar '17

Hello, I'm unable to get sound via the jack headphone connection on my lenovo dock when a t470 is connected. Some years ago I've experienced a similar issue with a t440p connected to the same docking station. Back than a patch helped to activate the jack sound: [codec] 0x10ec0292 0x17aa220e 0 [pincfg] 0x16 0x21211010 0x19 0x21a11010 So I've tried to match codec and pincfg to my t470 without success unfortunately: [codec] 0x10ec0298 0x17aa2245 0 [pincfg] 0x17 0x21211010 0x19 0x21a11010 /proc/asound/card0/codec#0: Codec: Realtek ALC298 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0298 Subsystem Id: 0x17aa2245 Revision Id: 0x100103 No Modem Function Group found Default PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=8, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Line Out Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC298 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=0 Amp-Out vals: [0x7f 0x7f] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=0 Amp-Out vals: [0x7f 0x7f] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x40]: 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0xc3 0xc3] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="ALC298 Analog", type="Audio", device=0 Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0x20 0x20] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100511: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x40]: 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x25 Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Dock Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Dock Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 4 0x18 0x19 0x1a 0x1d Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] Connection: 1 0x03 Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Audio Input] wcaps 0x10051b: Stereo Amp-In Amp-In caps: ofs=0x43, nsteps=0x7f, stepsize=0x01, mute=1 Amp-In vals: [0xc3 0xc3] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x24 Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x03 0x03] Pincap 0x00000020: IN Pin Default 0x90a60140: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x13 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40000000: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x14 [Pin Complex] wcaps 0x40050d: Stereo Amp-Out Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010010: OUT EAPD EAPD 0x2: EAPD Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c 0x0d* Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x17 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 3 0x0c* 0x0d 0x06 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x03a11030: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Dock Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00003724: IN Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=03, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x1a [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001373c: IN OUT HP EAPD Detect Vref caps: HIZ 50 GRD 80 100 EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 2 0x0c* 0x0d Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x40648605: [N/A] Modem Line at Ext N/A Conn = RCA, Color = Purple DefAssociation = 0x0, Sequence = 0x5 Pin-ctls: 0x20: IN Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x1e [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400401: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 EPSS Power: setting=D3, actual=D3 Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=150 Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Control: name="Line Out Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x03211020: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 6 0x18 0x19 0x1a 0x1d 0x0b 0x12* Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 6 0x18 0x19 0x1a* 0x1d 0x0b 0x13 Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x12* 0x13 Node 0x25 [Audio Selector] wcaps 0x300101: Stereo Connection: 1 0x1f Any Clues?

1 1

[alsa-devel] [PATCH] ALSA: hda - set intel audio clock to a properly value
by libin.yang＠intel.com 30 Mar '17

30 Mar '17

From: Libin Yang <libin.yang(a)intel.com> On some Intel platforms, the audio clock may not be set correctly with initial setting. This will cause the audio playback/capture rates wrong. This patch checks the audio clock setting and will set it to a properly value if it is not set correct. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=188411 Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/sound/hda_register.h | 12 +++-- sound/hda/ext/hdac_ext_controller.c | 6 +-- sound/pci/hda/hda_intel.c | 91 +++++++++++++++++++++++++++++++++++++ 3 files changed, 103 insertions(+), 6 deletions(-) diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 0013063..7ea16cb 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -227,6 +227,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_PPLCLLPU 0xC /* registers for Multiple Links Capability Structure */ +/* Multiple Links Capability */ +#define AZX_REG_ML_CAP_BASE 0xc00 #define AZX_ML_CAP_ID 0x2 #define AZX_REG_ML_MLCH 0x00 #define AZX_REG_ML_MLCD 0x04 @@ -243,9 +245,13 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 -#define AZX_MLCTL_SPA (1<<16) -#define AZX_MLCTL_CPA 23 - +#define AZX_REG_ML_LCAPx(x) (AZX_REG_ML_CAP_BASE + (0x40 + 0x40 * x)) +#define AZX_REG_ML_LCTLx(x) (AZX_REG_ML_CAP_BASE + (0x44 + 0x40 * x)) +#define ML_LCTL_SCF_MASK 0xF +#define AZX_MLCTL_SPA (0x1 << 16) +#define AZX_MLCTL_CPA (0x1 << 23) +#define AZX_MLCTL_SPA_SHIFT 16 +#define AZX_MLCTL_CPA_SHIFT 23 /* registers for DMA Resume Capability Structure */ #define AZX_DRSM_CAP_ID 0x5 diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 2614691..84f3b81 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -171,7 +171,7 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) { int timeout; u32 val; - int mask = (1 << AZX_MLCTL_CPA); + int mask = (1 << AZX_MLCTL_CPA_SHIFT); udelay(3); timeout = 150; @@ -179,10 +179,10 @@ static int check_hdac_link_power_active(struct hdac_ext_link *link, bool enable) do { val = readl(link->ml_addr + AZX_REG_ML_LCTL); if (enable) { - if (((val & mask) >> AZX_MLCTL_CPA)) + if (((val & mask) >> AZX_MLCTL_CPA_SHIFT)) return 0; } else { - if (!((val & mask) >> AZX_MLCTL_CPA)) + if (!((val & mask) >> AZX_MLCTL_CPA_SHIFT)) return 0; } udelay(3); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8256a8..017f64f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -539,6 +539,94 @@ static void bxt_reduce_dma_latency(struct azx *chip) azx_writel(chip, SKL_EM4L, val); } +/* + * ML_LCAP bits: + * bit 0: 6 MHz Supported + * bit 1: 12 MHz Supported + * bit 2: 24 MHz Supported + * bit 3: 48 MHz Supported + * bit 4: 96 MHz Supported + * bit 5: 192 MHz Supported + */ +static int intel_get_lctl_scf(struct azx *chip) +{ + u32 val; + + val = azx_readl(chip, ML_LCAPx(0)); + + if (val & (1 << 2)) + return 2; + else if (val & (1 << 3)) + return 3; + else if (val & (1 << 1)) + return 1; + else if (val & (1 << 4)) + return 4; + else if (val & (1 << 5)) + return 5; + + dev_warn(chip->card->dev, "set audio clock frequency to 6MHz"); + return 0; +} + +static void intel_init_lctl(struct azx *chip) +{ + u32 val; + int timeout; + + /* 0. check lctl register value is correct or not */ + /* the codecs are sharing the first link setting by default */ + val = azx_readl(chip, ML_LCTLx(0)); + /* if SCF is already set, let's use it */ + if ((val & ML_LCTL_SCF_MASK) != 0) + return; + + /* + * Before operatiing on SPA, CPA must match SPA. + * Any deviation may result in undefined behavior. + */ + if (((val & AZX_MLCTL_SPA) >> AZX_MLCTL_SPA_SHIFT) ^ + ((val & AZX_MLCTL_CPA) >> AZX_MLCTL_CPA_SHIFT)) + return; + + /* 1. turn link down: set SPA to 0 and wait CPA to 0 */ + val = azx_readl(chip, ML_LCTLx(0)); + val &= ~AZX_MLCTL_SPA; + azx_writel(chip, ML_LCTLx(0), val); + /* wait for CPA */ + timeout = 50; + while (timeout) { + if ((azx_readl(chip, ML_LCTLx(0)) & AZX_MLCTL_CPA) == 0) + break; + timeout--; + udelay(10); + } + if (!timeout) + goto SET_SPA; + /* need add 100ns delay for hardware ready */ + udelay(100); + + /* 2. update SCF to select a properly audio clock*/ + val &= ~ML_LCTL_SCF_MASK; + val |= intel_get_lctl_scf(chip); + azx_writel(chip, ML_LCTLx(0), val); + +SET_SPA: + /* 4. turn link up: set SPA to 1 and wait CPA to 1 */ + val = azx_readl(chip, ML_LCTLx(0)); + val |= AZX_MLCTL_SPA; + azx_writel(chip, ML_LCTLx(0), val); + timeout = 50; + while (timeout) { + if ((azx_readl(chip, ML_LCTLx(0)) & AZX_MLCTL_CPA) != 0) + break; + timeout--; + udelay(10); + } + /* need add 100ns delay for hardware ready */ + udelay(100); +} + static void hda_intel_init_chip(struct azx *chip, bool full_reset) { struct hdac_bus *bus = azx_bus(chip); @@ -564,6 +652,9 @@ static void hda_intel_init_chip(struct azx *chip, bool full_reset) /* reduce dma latency to avoid noise */ if (IS_BXT(pci)) bxt_reduce_dma_latency(chip); + + if (IS_SKL_PLUS(pci)) + intel_init_lctl(chip); } /* calculate runtime delay from LPIB */ -- 2.7.4

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[alsa-devel] [PATCH] ALSA: oxygen - Fix snd_oxygen module not loading for some (new?) Xonar DG SI cards.
by Eugene Ganeev 30 Mar '17

30 Mar '17

My Xonar DG SI card is showing up in lspci but no module is loaded for it. The patch just adds a new value with card's PCI ID to oxygen_ids array. Signed-off-by: Eugene Ganeev <easymodofrf(a)gmail.com> --- sound/pci/oxygen/oxygen.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index ada6c256378e..99ba0354d4cc 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -110,6 +110,7 @@ static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, /* Asus Xonar DG */ { OXYGEN_PCI_SUBID(0x1043, 0x8467), .driver_data = MODEL_XONAR_DG }, + { OXYGEN_PCI_SUBID(0x1043, 0x855e), .driver_data = MODEL_XONAR_DG }, /* Asus Xonar DGX */ { OXYGEN_PCI_SUBID(0x1043, 0x8521), .driver_data = MODEL_XONAR_DGX }, /* PCI 2.0 HD Audio */ -- 2.11.1

2 3

[alsa-devel] [PATCH 0/3] clkdev: add devm_get_clk_from_child()
by Kuninori Morimoto 30 Mar '17

30 Mar '17

Hi Stephen This is v5 of "clkdev: add devm_of_clk_get()", but new series. I hope my understanding was correct with your idea. Kuninori Morimoto (3): 1) clkdev: add devm_get_clk_from_child() 2) ASoC: simple-card: use devm_get_clk_from_child() 3) ASoC: simple-card-utils: enable clocks/clock-names/clock-ranges .../devicetree/bindings/sound/simple-card.txt | 32 +++++++++++++++ drivers/clk/clk-devres.c | 21 ++++++++++ include/linux/clk.h | 29 ++++++++++++-- include/sound/simple_card_utils.h | 11 +++--- sound/soc/generic/simple-card-utils.c | 45 ++++++++++++++++++++-- sound/soc/generic/simple-card.c | 4 +- sound/soc/generic/simple-scu-card.c | 4 +- 7 files changed, 129 insertions(+), 17 deletions(-) -- 1.9.1

4 20

[alsa-devel] [PATCH] sound: oxygen: simply setting of the shortname for Xonar DG cards
by Clemens Ladisch 29 Mar '17

29 Mar '17

We don't need to manually set the card name; with an entry in the names[] array, this happens automatically. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- sound/pci/oxygen/oxygen.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 74afb6b75976..e36ed8af55ad 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -767,6 +767,8 @@ static int get_oxygen_model(struct oxygen *chip, [MODEL_FANTASIA] = "TempoTec HiFier Fantasia", [MODEL_SERENADE] = "TempoTec HiFier Serenade", [MODEL_HG2PCI] = "CMI8787-HG2PCI", + [MODEL_XONAR_DG] = "Xonar DG", + [MODEL_XONAR_DGX] = "Xonar DGX", }; chip->model = model_generic; @@ -829,12 +831,8 @@ static int get_oxygen_model(struct oxygen *chip, chip->model.dac_channels_mixer = 2; break; case MODEL_XONAR_DG: - chip->model = model_xonar_dg; - chip->model.shortname = "Xonar DG"; - break; case MODEL_XONAR_DGX: chip->model = model_xonar_dg; - chip->model.shortname = "Xonar DGX"; break; } if (id->driver_data == MODEL_MERIDIAN ||

2 1

[alsa-devel] dmix, snd_pcm_drain(), SND_PCM_NONBLOCK - bug?
by Max Kellermann 29 Mar '17

29 Mar '17

Hi, I have received a bug report for the Music Player Daemon which complains about a freezing MPD process, after I switched MPD's ALSA output to non-blocking mode: https://bugs.musicpd.org/view.php?id=4662 This happens when snd_pcm_drain() is used on a dmix PCM which is in SND_PCM_NONBLOCK mode. snd_pcm_drain() will be called repeatedly, because ALSA's file descriptor is always ready. According to my research, the bug is in __snd_pcm_dmix_drain(). This function always returns -EAGAIN when SND_PCM_NONBLOCK is set: if (pcm->mode & SND_PCM_NONBLOCK) return -EAGAIN; This comes very early in the function, before any other relevant checks. It can never finish that way, and calling it again at any time will never do anything. Thus, snd_pcm_drain() appears to be broken in non-blocking mode. Or am I misunderstanding how snd_pcm_drain() is supposed to be used in non-blocking mode? Max

3 4

[alsa-devel] [PATCH] ASoC: WM8962: Let codec driver enable/disable its MCLK
by Daniel Baluta 29 Mar '17

29 Mar '17

From: Nicolin Chen <b42378(a)freescale.com> WM8962 needs its MCLK when powerup in wm8962_resume(). Thus it's better to control the MCLK in codec driver. Thus remove the clock enable in machine dirver accordingly. While at it, get rid of imx_wm8962_remove function since it is now empty. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/imx-wm8962.c | 40 ++++++++-------------------------------- 1 file changed, 8 insertions(+), 32 deletions(-) diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 1b60958..3d894d9 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -33,7 +33,6 @@ struct imx_wm8962_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - struct clk *codec_clk; unsigned int clk_frequency; }; @@ -163,6 +162,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) struct imx_priv *priv = &card_priv; struct i2c_client *codec_dev; struct imx_wm8962_data *data; + struct clk *codec_clk; int int_port, ext_port; int ret; @@ -231,19 +231,14 @@ static int imx_wm8962_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); - if (IS_ERR(data->codec_clk)) { - ret = PTR_ERR(data->codec_clk); + codec_clk = devm_clk_get(&codec_dev->dev, NULL); + if (IS_ERR(codec_clk)) { + ret = PTR_ERR(codec_clk); dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); goto fail; } - data->clk_frequency = clk_get_rate(data->codec_clk); - ret = clk_prepare_enable(data->codec_clk); - if (ret) { - dev_err(&codec_dev->dev, "failed to enable codec clk: %d\n", ret); - goto fail; - } + data->clk_frequency = clk_get_rate(codec_clk); data->dai.name = "HiFi"; data->dai.stream_name = "HiFi"; @@ -258,10 +253,10 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.dev = &pdev->dev; ret = snd_soc_of_parse_card_name(&data->card, "model"); if (ret) - goto clk_fail; + goto fail; ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); if (ret) - goto clk_fail; + goto fail; data->card.num_links = 1; data->card.owner = THIS_MODULE; data->card.dai_link = &data->dai; @@ -277,16 +272,9 @@ static int imx_wm8962_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto clk_fail; + goto fail; } - of_node_put(ssi_np); - of_node_put(codec_np); - - return 0; - -clk_fail: - clk_disable_unprepare(data->codec_clk); fail: of_node_put(ssi_np); of_node_put(codec_np); @@ -294,17 +282,6 @@ static int imx_wm8962_probe(struct platform_device *pdev) return ret; } -static int imx_wm8962_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - - if (!IS_ERR(data->codec_clk)) - clk_disable_unprepare(data->codec_clk); - - return 0; -} - static const struct of_device_id imx_wm8962_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8962", }, { /* sentinel */ } @@ -318,7 +295,6 @@ static struct platform_driver imx_wm8962_driver = { .of_match_table = imx_wm8962_dt_ids, }, .probe = imx_wm8962_probe, - .remove = imx_wm8962_remove, }; module_platform_driver(imx_wm8962_driver); -- 2.7.4

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[alsa-devel] [PATCH v3 0/2] imx-wm8962: Let codec driver enable/disable its MCLK
by Daniel Baluta 29 Mar '17

29 Mar '17

This is a small cleanup patch series which removes MLCK enable/disable from imx-wm8962 machine driver and lets the codec handle that. While reviewing this, Nicolin noticed that codec_clk cleanup is not correct so the second patch fixes this. Changes since v2: * added 2/2 patch to fix codec_clk cleanup * fixed prefix subject Daniel Baluta (2): ASoC: imx-wm8962: Let codec driver enable/disable its MCLK ASoC: imx-wm8962: Fix codec_clk cleanup sound/soc/fsl/imx-wm8962.c | 41 +++++++++-------------------------------- 1 file changed, 9 insertions(+), 32 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH v2 00/11] ASoC: Intel: Skylake: More driver updates
by jeeja.kp＠intel.com 29 Mar '17

29 Mar '17

From: Jeeja KP <jeeja.kp(a)intel.com> This patch series provides updates on the following: o Fix LLCH register read. o Add 16-bit constraint to FE bxt_rt298 machine. o Fix DMA position reporting for capture stream. o Use the sig_bits to define dai bps caps. o Update sig_bits based on converter capability. o Rearrangement of code to cleanup SKL SST library. o Add support for deferred DSP module bind. Changes in v2: - Addressed Pierre's comments for "Fix DMA position reporting for capture stream" patch. B, Jayachandran (1): ALSA: hda: Fix LLCH register read G Kranthi (1): ASoC: Intel: Skylake: Add 16-bit constraint to FE bxt_rt298 machine Hardik T Shah (1): ASoC: Intel: Skylake: Fix DMA position reporting for capture stream Jeeja KP (5): ASoC: Intel: Skylake: Use the sig_bits to define dai bps capability ASoC: hdac_hdmi: Update sig_bits based on converter capability ASoC: Intel: Skylake: Rearrangement of code to cleanup SKL SST library ASoC: Intel: Skylake: Fix module state after unbind and delete ASoC: Intel: Skylake: Add support for deferred DSP module bind Vinod Koul (3): ASoC: Intel: Skylake: Don't unload module when in use ASoC: Intel: Skylake: Remove redundant vmixer handler ASoC: Intel: Skylake: remove hard coded ACPI path sound/hda/hdac_controller.c | 2 +- sound/soc/codecs/hdac_hdmi.c | 4 +- sound/soc/intel/boards/bxt_rt298.c | 3 + sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-nhlt.c | 7 +- sound/soc/intel/skylake/skl-pcm.c | 95 ++++++++++++++++--- sound/soc/intel/skylake/skl-sst-dsp.h | 24 +++-- sound/soc/intel/skylake/skl-sst-ipc.h | 8 ++ sound/soc/intel/skylake/skl-sst-utils.c | 60 ++---------- sound/soc/intel/skylake/skl-sst.c | 5 + sound/soc/intel/skylake/skl-topology.c | 159 +++++++++++++++++++++++++------- sound/soc/intel/skylake/skl-topology.h | 17 ++-- sound/soc/intel/skylake/skl.h | 1 + 13 files changed, 263 insertions(+), 124 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: cs35l35: Clear reset_gpio on the error path in probe
by Charles Keepax 29 Mar '17

29 Mar '17

The error path in probe attempts to put the device back into reset. Should we fail to get the reset_gpio (such as a probe defer) we will leave the error value in there, which the gpiod_set_value_cansleep on the error path will attempt to deference. Fix this issue by clearing reset_gpio before we head into the error path. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs35l35.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index 1d9f332..9688274 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -1412,10 +1412,10 @@ static int cs35l35_i2c_probe(struct i2c_client *i2c_client, GPIOD_OUT_LOW); if (IS_ERR(cs35l35->reset_gpio)) { ret = PTR_ERR(cs35l35->reset_gpio); + cs35l35->reset_gpio = NULL; if (ret == -EBUSY) { dev_info(dev, "Reset line busy, assuming shared reset\n"); - cs35l35->reset_gpio = NULL; } else { dev_err(dev, "Failed to get reset GPIO: %d\n", ret); goto err; -- 2.1.4

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