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[PATCH] ALSA: hda: Refactor calculating SDnFMT according to specification
by Pawel Harlozinski 02 Sep '20

02 Sep '20

Fix setting SDnFMT based on High Definition Audio Specification Rev. 1.0a page 48. Bits per Sample (BITS): 000 = 8 bits. The data will be packed in memory in 8-bit containers on 16-bit boundaries. 001 = 16 bits. The data will be packed in memory in 16-bit containers on 16-bit boundaries. 010 = 20 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 011 = 24 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 100 = 32 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 101-111 = Reserved Set SDnFMT depending on which format was given, as maxbps only describes container size. Henceforth split cases for formats 20, 24, 32 bits. For format SNDRV_PCM_FORMAT_FLOAT_LE width is equal 32 so as previously it will end up with mask for 32 bits. Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/hda/hdac_device.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 333220f0f8af..2ccf79866f99 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -762,14 +762,13 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_16; break; case 20: + val |= AC_FMT_BITS_20; + break; case 24: + val |= AC_FMT_BITS_24; + break; case 32: - if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) - val |= AC_FMT_BITS_32; - else if (maxbps >= 24) - val |= AC_FMT_BITS_24; - else - val |= AC_FMT_BITS_20; + val |= AC_FMT_BITS_32; break; default: return 0; -- 2.17.1

4 4

[PATCH] Revert "ASoC: Intel: haswell: Power transition refactor"
by Cezary Rojewski 02 Sep '20

02 Sep '20

This reverts commit 8ec7d6043263ecf250b9b7c0dd8ade899487538a. While addressing existing power-cycle limitations for sound/soc/intel/haswell solution, change brings regression for standard audio userspace flows e.g.: when using PulseAudio. Occasional sound-card initialization fail is still better than pernament audio distortions, so revert the change. Fixes: 8ec7d6043263 ("ASoC: Intel: haswell: Power transition refactor") Reported-by: Christian Bundy <christianbundy(a)fraction.io> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/haswell/sst-haswell-dsp.c | 185 ++++++++++------------ 1 file changed, 81 insertions(+), 104 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c index de80e19454c1..88c3f63bded9 100644 --- a/sound/soc/intel/haswell/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -243,92 +243,45 @@ static irqreturn_t hsw_irq(int irq, void *context) return ret; } -#define CSR_DEFAULT_VALUE 0x8480040E -#define ISC_DEFAULT_VALUE 0x0 -#define ISD_DEFAULT_VALUE 0x0 -#define IMC_DEFAULT_VALUE 0x7FFF0003 -#define IMD_DEFAULT_VALUE 0x7FFF0003 -#define IPCC_DEFAULT_VALUE 0x0 -#define IPCD_DEFAULT_VALUE 0x0 -#define CLKCTL_DEFAULT_VALUE 0x7FF -#define CSR2_DEFAULT_VALUE 0x0 -#define LTR_CTRL_DEFAULT_VALUE 0x0 -#define HMD_CTRL_DEFAULT_VALUE 0x0 - -static void hsw_set_shim_defaults(struct sst_dsp *sst) -{ - sst_dsp_shim_write_unlocked(sst, SST_CSR, CSR_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_ISRX, ISC_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_ISRD, ISD_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IMRX, IMC_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IMRD, IMD_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IPCX, IPCC_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IPCD, IPCD_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_CLKCTL, CLKCTL_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_CSR2, CSR2_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_LTRC, LTR_CTRL_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_HMDC, HMD_CTRL_DEFAULT_VALUE); -} - -/* all clock-gating minus DCLCGE and DTCGE */ -#define SST_VDRTCL2_CG_OTHER 0xB7D - static void hsw_set_dsp_D3(struct sst_dsp *sst) { + u32 val; u32 reg; - /* disable clock core gating */ + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* stall, reset and set 24MHz XOSC */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST, - SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST); - - /* DRAM power gating all */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg |= SST_VDRTCL0_ISRAMPGE_MASK | - SST_VDRTCL0_DSRAMPGE_MASK; - reg &= ~(SST_VDRTCL0_D3SRAMPGD); - reg |= SST_VDRTCL0_D3PGD; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - udelay(50); + /* enable power gating and switch off DRAM & IRAM blocks */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + val |= SST_VDRTCL0_DSRAMPGE_MASK | + SST_VDRTCL0_ISRAMPGE_MASK; + val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD); + writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); - /* PLL shutdown enable */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_APLLSE_MASK; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* switch off audio PLL */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_APLLSE_MASK; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); - /* disable MCLK */ + /* disable MCLK(clkctl.smos = 0) */ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK, 0); - - /* switch clock gating */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_CG_OTHER; - reg &= ~(SST_VDRTCL2_DTCGE); - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* enable DTCGE separatelly */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DTCGE; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + SST_CLKCTL_MASK, 0); - /* set shim defaults */ - hsw_set_shim_defaults(sst); - - /* set D3 */ - reg = readl(sst->addr.pci_cfg + SST_PMCS); - reg |= SST_PMCS_PS_MASK; - writel(reg, sst->addr.pci_cfg + SST_PMCS); + /* Set D3 state, delay 50 us */ + val = readl(sst->addr.pci_cfg + SST_PMCS); + val |= SST_PMCS_PS_MASK; + writel(val, sst->addr.pci_cfg + SST_PMCS); udelay(50); - /* enable clock core gating */ + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE; + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + udelay(50); + } static void hsw_reset(struct sst_dsp *sst) @@ -346,62 +299,75 @@ static void hsw_reset(struct sst_dsp *sst) SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); } -/* recommended CSR state for power-up */ -#define SST_CSR_D0_MASK (0x18A09C0C | SST_CSR_DCS_MASK) - static int hsw_set_dsp_D0(struct sst_dsp *sst) { - u32 reg; + int tries = 10; + u32 reg, fw_dump_bit; - /* disable clock core gating */ + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* switch clock gating */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_CG_OTHER; - reg &= ~(SST_VDRTCL2_DTCGE); - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* Disable D3PG (VDRTCTL0.D3PGD = 1) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - /* set D0 */ + /* Set D0 state */ reg = readl(sst->addr.pci_cfg + SST_PMCS); - reg &= ~(SST_PMCS_PS_MASK); + reg &= ~SST_PMCS_PS_MASK; writel(reg, sst->addr.pci_cfg + SST_PMCS); - /* DRAM power gating none */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg &= ~(SST_VDRTCL0_ISRAMPGE_MASK | - SST_VDRTCL0_DSRAMPGE_MASK); - reg |= SST_VDRTCL0_D3SRAMPGD; - reg |= SST_VDRTCL0_D3PGD; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - mdelay(10); + /* check that ADSP shim is enabled */ + while (tries--) { + reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK; + if (reg == 0) + goto finish; + + msleep(1); + } + + return -ENODEV; - /* set shim defaults */ - hsw_set_shim_defaults(sst); +finish: + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); + + /* stall DSP core, set clk to 192/96Mhz */ + sst_dsp_shim_update_bits_unlocked(sst, + SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK, + SST_CSR_STALL | SST_CSR_DCS(4)); - /* restore MCLK */ + /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK, SST_CLKCTL_MASK); + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); - /* PLL shutdown disable */ + /* Stall and reset core, set CSR */ + hsw_reset(sst); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_APLLSE_MASK); + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_D0_MASK, SST_CSR_SBCS0 | SST_CSR_SBCS1 | - SST_CSR_STALL | SST_CSR_DCS(4)); udelay(50); - /* enable clock core gating */ + /* switch on audio PLL */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE; + reg &= ~SST_VDRTCL2_APLLSE_MASK; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* clear reset */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_RST, 0); + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; + /* for D0, always enable the block(DSRAM[0]) used for FW dump */ + fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; + writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); + /* disable DMA finish function for SSP0 & SSP1 */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, @@ -418,6 +384,12 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY | SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0); + /* clear IPC registers */ + sst_dsp_shim_write(sst, SST_IPCX, 0x0); + sst_dsp_shim_write(sst, SST_IPCD, 0x0); + sst_dsp_shim_write(sst, 0x80, 0x6); + sst_dsp_shim_write(sst, 0xe0, 0x300a); + return 0; } @@ -443,6 +415,11 @@ static void hsw_sleep(struct sst_dsp *sst) { dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n"); + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL, + SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS); + hsw_set_dsp_D3(sst); dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n"); } -- 2.17.1

4 4

[PATCH v2 00/10] ALSA: convert tasklets to use new
by Allen Pais 02 Sep '20

02 Sep '20

Commit 12cc923f1ccc ("tasklet: Introduce new initialization API")' introduced a new tasklet initialization API. This series converts all the sound drivers to use the new tasklet_setup() API The series is based on 5.9-rc3 (f75aef392f86) v2: Fixed subject lines. Allen Pais (10): ALSA: core: convert tasklets to use new tasklet_setup() API ALSA: firewire: convert tasklets to use new tasklet_setup() API ALSA: pci/asihpi: convert tasklets to use new tasklet_setup() API ALSA: riptide: convert tasklets to use new tasklet_setup() API ALSA: hdsp: convert tasklets to use new tasklet_setup() API ASoc: fsl_esai: convert tasklets to use new tasklet_setup() API ASoC: siu: convert tasklets to use new tasklet_setup() API ASoC: txx9: convert tasklets to use new tasklet_setup() API ALSA: usb-audio: convert tasklets to use new tasklet_setup() API ALSA: ua101: convert tasklets to use new tasklet_setup() API sound/core/timer.c | 7 +++---- sound/firewire/amdtp-stream.c | 8 ++++---- sound/pci/asihpi/asihpi.c | 9 ++++----- sound/pci/riptide/riptide.c | 6 +++--- sound/pci/rme9652/hdsp.c | 6 +++--- sound/pci/rme9652/hdspm.c | 7 +++---- sound/soc/fsl/fsl_esai.c | 7 +++---- sound/soc/sh/siu_pcm.c | 10 ++++------ sound/soc/txx9/txx9aclc.c | 7 +++---- sound/usb/midi.c | 7 +++---- sound/usb/misc/ua101.c | 7 +++---- 11 files changed, 36 insertions(+), 45 deletions(-) -- 2.25.1

4 16

[bug report] 'ASoC: Intel: haswell: Power transition refactor' and PulseAudio
by Christian Bundy 01 Sep '20

01 Sep '20

After upgrading to Linux 5.8 I discovered an audio issue on my device that was introduced in 8ec7d6043263ecf250b9b7c0dd8ade899487538a [0]. I used 'git bisect' to identify the commit that introduced the bug and have confirmed that reverting the commit resolves the problem Reproduction: 1. Play any audio via PulseAudio. 2. Observe that the audio output is fuzzy and choppy. I can use programs like mpv to play audio without PulseAudio, and the audio is fine, but as soon as I open a process that uses PulseAudio it will ruin the audio output for all processes (including mpv) until I reboot. I'm using a 2015 Chromebook Pixel ("Samus") and have confirmed this problem with a friend who has the same device. Is there anything I can do to help debug this instead of sending a patch to revert the commit? Relevant lspci output: 00:03.0 Audio device: Intel Corporation Broadwell-U Audio Controller (rev 09) Subsystem: Intel Corporation Broadwell-U Audio Controller Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 50 Region 0: Memory at e1218000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit- Address: fee00378 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0 ExtTag- RBE- FLReset+ DevCtl: CorrErr- NonFatalErr- FatalErr- UnsupReq- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- FLReset- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- NonFatalErr- FatalErr- UnsupReq- AuxPwr- TransPend- Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel [0]: https://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git/commit/?h=…

4 7

[PATCH] soundwire: fix error handling
by trix＠redhat.com 01 Sep '20

01 Sep '20

From: Tom Rix <trix(a)redhat.com> clang static analysis flags this problem stream.c:844:9: warning: Use of memory after it is freed kfree(bus->defer_msg.msg->buf); ^~~~~~~~~~~~~~~~~~~~~~~ This happens in an error handler cleaning up memory allocated for elements in a list. list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; kfree(bus->defer_msg.msg->buf); kfree(bus->defer_msg.msg); } And is triggered when the call to sdw_bank_switch() fails. There are a two problems. First, when sdw_bank_switch() fails, though it frees memory it does not clear bus's reference 'defer_msg.msg' to that memory. The second problem is the freeing msg->buf. In some cases msg will be NULL so this will dereference a null pointer. Need to check before freeing. Fixes: 99b8a5d608a6 ("soundwire: Add bank switch routine") Signed-off-by: Tom Rix <trix(a)redhat.com> --- drivers/soundwire/stream.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 37290a799023..6e36deb505b1 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -717,6 +717,7 @@ static int sdw_bank_switch(struct sdw_bus *bus, int m_rt_count) kfree(wbuf); error_1: kfree(wr_msg); + bus->defer_msg.msg = NULL; return ret; } @@ -840,9 +841,10 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) error: list_for_each_entry(m_rt, &stream->master_list, stream_node) { bus = m_rt->bus; - - kfree(bus->defer_msg.msg->buf); - kfree(bus->defer_msg.msg); + if (bus->defer_msg.msg) { + kfree(bus->defer_msg.msg->buf); + kfree(bus->defer_msg.msg); + } } msg_unlock: -- 2.18.1

5 8

[PATCH v4] ASoC: tlv320adcx140: Fix accessing uninitialized adcx140->dev
by Camel Guo 01 Sep '20

01 Sep '20

From: Camel Guo <camelg(a)axis.com> In adcx140_i2c_probe, adcx140->dev is accessed before its initialization. This commit fixes this bug. Fixes: 689c7655b50c ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Acked-by: Dan Murphy <dmurphy(a)ti.com> Signed-off-by: Camel Guo <camel.guo(a)axis.com> --- v4: - Add Acked-By from Dan v3: - Add Fixes tag v2: - Fix typo sound/soc/codecs/tlv320adcx140.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 5cd50d841177..7ae6ec374be3 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -980,6 +980,8 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, if (!adcx140) return -ENOMEM; + adcx140->dev = &i2c->dev; + adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(adcx140->gpio_reset)) @@ -1007,7 +1009,7 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, ret); return ret; } - adcx140->dev = &i2c->dev; + i2c_set_clientdata(i2c, adcx140); return devm_snd_soc_register_component(&i2c->dev, -- 2.20.1

2 1

[PATCH] ASoC: rt5682: Prefer async probe
by Douglas Anderson 01 Sep '20

01 Sep '20

The probe of rt5682 is pretty slow. A quick measurement shows that it takes ~650 ms on at least one board. There's no reason to block all other drivers waiting for this probe to finish. Set the flag to allow other drivers to probe while we're probing. Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- NOTE: I haven't done any analysis of the driver to see _why_ it's so slow, only that I have measured it to be slow. Someone could certainly take the time to profile / optimize it, but in any case it still won't hurt to be async. This is a very safe flag to turn on since: 1. It's not like our probe order was defined by anything anyway. When we probe is at the whim of when our i2c controller probes and that can be any time. 2. If some other driver needs us then they have to handle the fact that we might not have probed yet anyway. 3. There may be other drivers probing at the same time as us anyway because _they_ used async probe. While I won't say that it's impossible to tickle a bug by turning on async probe, I would assert that in almost all cases the bug was already there and needed to be fixed anyway. sound/soc/codecs/rt5682-i2c.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 85aba311bdc8..6b4e0eb30c89 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -294,6 +294,7 @@ static struct i2c_driver rt5682_i2c_driver = { .name = "rt5682", .of_match_table = rt5682_of_match, .acpi_match_table = rt5682_acpi_match, + .probe_type = PROBE_PREFER_ASYNCHRONOUS, }, .probe = rt5682_i2c_probe, .shutdown = rt5682_i2c_shutdown, -- 2.28.0.402.g5ffc5be6b7-goog

4 3

[PATCH 1/2] ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811
by Sylwester Nawrocki 01 Sep '20

01 Sep '20

The WM8994_MICBIAS register is not available in the WM1811 CODEC so skip initialization of that register for that device. This suppresses an error during boot: "wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec" Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm_hubs.c | 3 +++ sound/soc/codecs/wm_hubs.h | 1 + 3 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 038be66..b3ba053 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4193,11 +4193,13 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994->hubs.dcs_readback_mode = 2; break; } + wm8994->hubs.micd_scthr = true; break; case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.micd_scthr = true; switch (control->revision) { case 0: diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 891effe..0c88184 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1223,6 +1223,9 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + if (!hubs->micd_scthr) + return 0; + snd_soc_component_update_bits(component, WM8993_MICBIAS, WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | WM8993_MICB1_LVL | WM8993_MICB2_LVL, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 4b8e5f0..988b29e 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -27,6 +27,7 @@ struct wm_hubs_data { int hp_startup_mode; int series_startup; int no_series_update; + bool micd_scthr; bool no_cache_dac_hp_direct; struct list_head dcs_cache; -- 2.7.4

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[PATCH 00/11] soundwire: intel: add multi-link support
by Bard Liao 01 Sep '20

01 Sep '20

This series enables multi-link support for Intel platforms. Bard Liao (1): soundwire: intel: Only call sdw stream APIs for the first cpu_dai Pierre-Louis Bossart (10): soundwire: intel: disable shim wake on suspend soundwire: intel: ignore software command retries soundwire: intel: add multi-link support soundwire: intel: add missing support for all clock stop modes soundwire: bus: update multi-link definition with hw sync details soundwire: intel: add multi-link hw_synchronization information soundwire: stream: enable hw_sync as needed by hardware soundwire: intel: add dynamic debug trace for clock-stop invalid configs soundwire: intel: pass link_mask information to each master soundwire: intel: don't manage link power individually drivers/soundwire/intel.c | 309 ++++++++++++++++++++++++++++----- drivers/soundwire/intel.h | 2 + drivers/soundwire/intel_init.c | 1 + drivers/soundwire/stream.c | 15 +- include/linux/soundwire/sdw.h | 6 + 5 files changed, 282 insertions(+), 51 deletions(-) -- 2.17.1

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[PATCH] ALSA: pcm: oss: Remove superfluous WARN_ON() for mulaw sanity check
by Takashi Iwai 01 Sep '20

01 Sep '20

The PCM OSS mulaw plugin has a check of the format of the counter part whether it's a linear format. The check is with snd_BUG_ON() that emits WARN_ON() when the debug config is set, and it confuses syzkaller as if it were a serious issue. Let's drop snd_BUG_ON() for avoiding that. While we're at it, correct the error code to a more suitable, EINVAL. Reported-by: syzbot+23b22dc2e0b81cbfcc95(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/mulaw.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 3788906421a7..fe27034f2846 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, snd_BUG(); return -EINVAL; } - if (snd_BUG_ON(!snd_pcm_format_linear(format->format))) - return -ENXIO; + if (!snd_pcm_format_linear(format->format)) + return -EINVAL; err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion", src_format, dst_format, -- 2.16.4

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