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

[alsa-devel] [PATCH 1/2] ASoC: nau8824: move key irq after jd done
by John Hsu 16 Nov '17

16 Nov '17

It is possible to get the fake key press interruption when the codec do jack detection. We think it's proper to move the key interruption configuration after jack detection done. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8824.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 0240759..b7b63ac 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -843,6 +843,11 @@ static void nau8824_jdet_work(struct work_struct *work) event_mask |= SND_JACK_HEADSET; snd_soc_jack_report(nau8824->jack, event, event_mask); + /* Enable short key press and release interruption. */ + regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING, + NAU8824_IRQ_KEY_RELEASE_DIS | + NAU8824_IRQ_KEY_SHORT_PRESS_DIS, 0); + nau8824_sema_release(nau8824); } @@ -850,13 +855,12 @@ static void nau8824_setup_auto_irq(struct nau8824 *nau8824) { struct regmap *regmap = nau8824->regmap; - /* Enable jack ejection, short key press and release interruption. */ + /* Enable jack ejection interruption. */ regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING_1, NAU8824_IRQ_INSERT_EN | NAU8824_IRQ_EJECT_EN, NAU8824_IRQ_EJECT_EN); regmap_update_bits(regmap, NAU8824_REG_INTERRUPT_SETTING, - NAU8824_IRQ_EJECT_DIS | NAU8824_IRQ_KEY_RELEASE_DIS | - NAU8824_IRQ_KEY_SHORT_PRESS_DIS, 0); + NAU8824_IRQ_EJECT_DIS, 0); /* Enable internal VCO needed for interruptions */ nau8824_config_sysclk(nau8824, NAU8824_CLK_INTERNAL, 0); regmap_update_bits(regmap, NAU8824_REG_ENA_CTRL, -- 2.6.4

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[alsa-devel] [PATCH] ASoC: intel: initialize return value properly
by Arnd Bergmann 16 Nov '17

16 Nov '17

When CONFIG_SND_SOC_HDAC_HDMI is disabled, we can run into an uninitialized variable: sound/soc/intel/skylake/skl.c: In function 'skl_resume': sound/soc/intel/skylake/skl.c:326:6: error: 'ret' may be used uninitialized in this function [-Werror=maybe-uninitialized] I have run into this on today's linux-next kernel, but it appears that this is an older problem that was just hard to trigger with randconfig builds as CONFIG_SND_SOC_HDAC_HDMI would in effect be impossible to disable when having SND_SOC_INTEL_SKYLAKE enabled. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/intel/skylake/skl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 31d8634e8aa1..acb0ab470ca6 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -355,6 +355,7 @@ static int skl_resume(struct device *dev) if (ebus->cmd_dma_state) snd_hdac_bus_init_cmd_io(&ebus->bus); + ret = 0; } else { ret = _skl_resume(ebus); -- 2.9.0

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[alsa-devel] [PATCH 0/3] Fixes for AMD Stoney ACP audio
by Alex Deucher 16 Nov '17

16 Nov '17

This patch set is just a couple fixes for the Audio CoProcessor (ACP) on AMD Stoney platforms and a small general fix for the rt5645 codec driver. The entire patch set can also be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=sound-for-next-stoney Thanks! Alex Akshu Agrawal (2): ASoC: AMD: Make the driver name consistent across files ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2 Vijendar Mukunda (1): ASoC: amd: Report accurate hw_ptr during dma sound/soc/amd/Makefile | 4 +-- sound/soc/amd/acp-pcm-dma.c | 77 ++++++++++++++++++++++++++++----------------- sound/soc/amd/acp.h | 10 ++++++ sound/soc/codecs/rt5645.c | 12 +++++++ 4 files changed, 73 insertions(+), 30 deletions(-) -- 2.5.5

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[alsa-devel] [PATCH] ASoC: Intel: kbl: Modify Headset map for Multi-Playback
by Naveen Manohar 16 Nov '17

16 Nov '17

Current topology for Headset resulted in unwanted pop noise, while switching from spk->hs at the start of Headset Playback. Modified topology to fix the same. This modification also requires headset route to be updated. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 6f9a8bc..6dcad0a 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -101,7 +101,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 6072164..271ae3c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -109,7 +109,7 @@ enum { { "ssp0 Tx", NULL, "spk_out" }, { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "hs_out" }, + { "ssp1 Tx", NULL, "codec1_out" }, { "hs_in", NULL, "ssp1 Rx" }, { "ssp1 Rx", NULL, "AIF Capture" }, -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8540: improve FLL performance
by John Hsu 16 Nov '17

16 Nov '17

Add these parameters to improve the FLL performance. The comments show as follows: (1)ICTRL_LATCH: FLL DSP speed capability control When FLL running at high frequency with long decimal number, DSP needs to operate at high speed. FLL DSP can optimize between performance and power consumption by ICTRL_LATCH.(111 has highest power consumption.) The default setting can be used to reduce power. (2)CUTOFF500: loop filter cutoff frequency at 500Khz It will give the best FLL performance but highest power consumption to enable the cutoff frequency. FLL Loop Filter enable to reduce FLL output noise, especially,(DCO frequency)/(FLL input reference frequency) is not a integer. (3)GAIN_ERR: FLL gain error correction threshold setting The threshold is comparison between DCO and target frequency. The value 1111 has the most sensitive threshold, that is, 1111 can have the most accurate DCO to target frequency. However, the gain error setting conditionally and inversely depends on FLL input reference clock rate. Higher FLL reference input frequency can only set lower gain error, such as 0000 for input reference from MCLK=12.288Mhz. On the other side, if FLL reference input is from Frame Sync, 48KHz, higher error gain can apply such as 1111. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8540.c | 21 ++++++++++++++------- sound/soc/codecs/nau8540.h | 5 +++++ 2 files changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/nau8540.c b/sound/soc/codecs/nau8540.c index 9565f9a..b08fb7e 100644 --- a/sound/soc/codecs/nau8540.c +++ b/sound/soc/codecs/nau8540.c @@ -615,7 +615,8 @@ static void nau8540_fll_apply(struct regmap *regmap, NAU8540_CLK_SRC_MASK | NAU8540_CLK_MCLK_SRC_MASK, NAU8540_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(regmap, NAU8540_REG_FLL1, - NAU8540_FLL_RATIO_MASK, fll_param->ratio); + NAU8540_FLL_RATIO_MASK | NAU8540_ICTRL_LATCH_MASK, + fll_param->ratio | (0x6 << NAU8540_ICTRL_LATCH_SFT)); /* FLL 16-bit fractional input */ regmap_write(regmap, NAU8540_REG_FLL2, fll_param->fll_frac); /* FLL 10-bit integer input */ @@ -636,13 +637,14 @@ static void nau8540_fll_apply(struct regmap *regmap, NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | NAU8540_FLL_FTR_SW_FILTER); regmap_update_bits(regmap, NAU8540_REG_FLL6, - NAU8540_SDM_EN, NAU8540_SDM_EN); + NAU8540_SDM_EN | NAU8540_CUTOFF500, + NAU8540_SDM_EN | NAU8540_CUTOFF500); } else { regmap_update_bits(regmap, NAU8540_REG_FLL5, NAU8540_FLL_PDB_DAC_EN | NAU8540_FLL_LOOP_FTR_EN | NAU8540_FLL_FTR_SW_MASK, NAU8540_FLL_FTR_SW_ACCU); - regmap_update_bits(regmap, - NAU8540_REG_FLL6, NAU8540_SDM_EN, 0); + regmap_update_bits(regmap, NAU8540_REG_FLL6, + NAU8540_SDM_EN | NAU8540_CUTOFF500, 0); } } @@ -657,17 +659,22 @@ static int nau8540_set_pll(struct snd_soc_codec *codec, int pll_id, int source, switch (pll_id) { case NAU8540_CLK_FLL_MCLK: regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_MCLK); + NAU8540_FLL_CLK_SRC_MASK | NAU8540_GAIN_ERR_MASK, + NAU8540_FLL_CLK_SRC_MCLK | 0); break; case NAU8540_CLK_FLL_BLK: regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_BLK); + NAU8540_FLL_CLK_SRC_MASK | NAU8540_GAIN_ERR_MASK, + NAU8540_FLL_CLK_SRC_BLK | + (0xf << NAU8540_GAIN_ERR_SFT)); break; case NAU8540_CLK_FLL_FS: regmap_update_bits(nau8540->regmap, NAU8540_REG_FLL3, - NAU8540_FLL_CLK_SRC_MASK, NAU8540_FLL_CLK_SRC_FS); + NAU8540_FLL_CLK_SRC_MASK | NAU8540_GAIN_ERR_MASK, + NAU8540_FLL_CLK_SRC_FS | + (0xf << NAU8540_GAIN_ERR_SFT)); break; default: diff --git a/sound/soc/codecs/nau8540.h b/sound/soc/codecs/nau8540.h index dceb04b..732b490 100644 --- a/sound/soc/codecs/nau8540.h +++ b/sound/soc/codecs/nau8540.h @@ -100,9 +100,13 @@ #define NAU8540_CLK_MCLK_SRC_MASK 0xf /* FLL1 (0x04) */ +#define NAU8540_ICTRL_LATCH_SFT 10 +#define NAU8540_ICTRL_LATCH_MASK (0x7 << NAU8540_ICTRL_LATCH_SFT) #define NAU8540_FLL_RATIO_MASK 0x7f /* FLL3 (0x06) */ +#define NAU8540_GAIN_ERR_SFT 12 +#define NAU8540_GAIN_ERR_MASK (0xf << NAU8540_GAIN_ERR_SFT) #define NAU8540_FLL_CLK_SRC_SFT 10 #define NAU8540_FLL_CLK_SRC_MASK (0x3 << NAU8540_FLL_CLK_SRC_SFT) #define NAU8540_FLL_CLK_SRC_MCLK (0 << NAU8540_FLL_CLK_SRC_SFT) @@ -127,6 +131,7 @@ /* FLL6 (0x9) */ #define NAU8540_DCO_EN (0x1 << 15) #define NAU8540_SDM_EN (0x1 << 14) +#define NAU8540_CUTOFF500 (0x1 << 13) /* PCM_CTRL0 (0x10) */ #define NAU8540_I2S_BP_SFT 7 -- 2.6.4

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[alsa-devel] [PATCH] ASoC: nau8824: change FVCO maximum threshold
by John Hsu 16 Nov '17

16 Nov '17

Change the maximum of FDCO which remains between 90MHz-100MHz. FDCO must be within the 90MHz-100MHz or the FFL cannot be guaranteed across the full range of operation. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 8a9a993..088e0ce 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -43,7 +43,7 @@ static bool nau8824_is_jack_inserted(struct nau8824 *nau8824); /* the parameter threshold of FLL */ #define NAU_FREF_MAX 13500000 -#define NAU_FVCO_MAX 124000000 +#define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 /* scaling for mclk from sysclk_src output */ -- 2.6.4

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[alsa-devel] [PATCH 0/2] ALSA-miXart: Fine-tuning for seven function implementations
by SF Markus Elfring 16 Nov '17

16 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 16 Nov 2017 16:27:32 +0100 Two update suggestions were taken into account from static source code analysis. Markus Elfring (2): Adjust 23 function calls together with a variable assignment Use common error handling code in snd_mixart_probe() sound/pci/mixart/mixart.c | 88 ++++++++++++++++++++++------------------- sound/pci/mixart/mixart_hwdep.c | 17 +++++--- sound/pci/mixart/mixart_mixer.c | 44 +++++++++++++++------ 3 files changed, 93 insertions(+), 56 deletions(-) -- 2.15.0

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[alsa-devel] [PATCH 0/2] ASoC: Intel: kbl: update dmic fixup params
by Naveen Manohar 16 Nov '17

16 Nov '17

Update dmic fixup params to always expose QUAD channel capture device, enabling User-space to control channel count. Assign min and max channel only based on input hw_params. And removes choosing the channel count based on pdata. Naveen Manohar (2): ASoC: Intel: kbl: Remove option of choosing CH count based on pdata ASoC: Intel: kbl: update dmic fixup params sound/soc/intel/boards/kbl_rt5663_max98927.c | 44 +++++++++------------------- 1 file changed, 14 insertions(+), 30 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rcar: revert IOMMU support so far
by Kuninori Morimoto 16 Nov '17

16 Nov '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 4821d914fe74 ("ASoC: rsnd: use dma_sync_single_for_xxx() for IOMMU") had supported IOMMU, but it breaks normal sound "recorde" and both PulseAudio's "playback/recorde". The sound will be noisy. That commit was using dma_sync_single_for_xxx(), and driver should make sure memory is protected during CPU or Device are using it. But if driver returns current "residue" data size correctly on pointer function, player/recorder will access to protected memory. IOMMU feature should be supported, but I don't know how to handle it without memory cache problem at this point. Thus, this patch simply revert it to avoid current noisy sound. Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Tested-by: Ryo Kodama <ryo.kodama.vz(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 4 +-- sound/soc/sh/rcar/dma.c | 86 ++++-------------------------------------------- 2 files changed, 8 insertions(+), 82 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 8cd900d..ebdfa34 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1329,8 +1329,8 @@ static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) return snd_pcm_lib_preallocate_pages_for_all( rtd->pcm, - SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + SNDRV_DMA_TYPE_DEV, + rtd->card->snd_card->dev, PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index fd557ab..4d750bdf 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -26,10 +26,7 @@ struct rsnd_dmaen { struct dma_chan *chan; dma_cookie_t cookie; - dma_addr_t dma_buf; unsigned int dma_len; - unsigned int dma_period; - unsigned int dma_cnt; }; struct rsnd_dmapp { @@ -71,38 +68,10 @@ struct rsnd_dma_ctrl { /* * Audio DMAC */ -#define rsnd_dmaen_sync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 1) -#define rsnd_dmaen_unsync(dmaen, io, i) __rsnd_dmaen_sync(dmaen, io, i, 0) -static void __rsnd_dmaen_sync(struct rsnd_dmaen *dmaen, struct rsnd_dai_stream *io, - int i, int sync) -{ - struct device *dev = dmaen->chan->device->dev; - enum dma_data_direction dir; - int is_play = rsnd_io_is_play(io); - dma_addr_t buf; - int len, max; - size_t period; - - len = dmaen->dma_len; - period = dmaen->dma_period; - max = len / period; - i = i % max; - buf = dmaen->dma_buf + (period * i); - - dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - - if (sync) - dma_sync_single_for_device(dev, buf, period, dir); - else - dma_sync_single_for_cpu(dev, buf, period, dir); -} - static void __rsnd_dmaen_complete(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); bool elapsed = false; unsigned long flags; @@ -115,22 +84,9 @@ static void __rsnd_dmaen_complete(struct rsnd_mod *mod, */ spin_lock_irqsave(&priv->lock, flags); - if (rsnd_io_is_working(io)) { - rsnd_dmaen_unsync(dmaen, io, dmaen->dma_cnt); - - /* - * Next period is already started. - * Let's sync Next Next period - * see - * rsnd_dmaen_start() - */ - rsnd_dmaen_sync(dmaen, io, dmaen->dma_cnt + 2); - + if (rsnd_io_is_working(io)) elapsed = true; - dmaen->dma_cnt++; - } - spin_unlock_irqrestore(&priv->lock, flags); if (elapsed) @@ -165,14 +121,8 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - if (dmaen->chan) { - int is_play = rsnd_io_is_play(io); - + if (dmaen->chan) dmaengine_terminate_all(dmaen->chan); - dma_unmap_single(dmaen->chan->device->dev, - dmaen->dma_buf, dmaen->dma_len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - } return 0; } @@ -237,11 +187,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct device *dev = rsnd_priv_to_dev(priv); struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; - dma_addr_t buf; - size_t len; - size_t period; int is_play = rsnd_io_is_play(io); - int i; int ret; cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; @@ -258,19 +204,10 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - len = snd_pcm_lib_buffer_bytes(substream); - period = snd_pcm_lib_period_bytes(substream); - buf = dma_map_single(dmaen->chan->device->dev, - substream->runtime->dma_area, - len, - is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE); - if (dma_mapping_error(dmaen->chan->device->dev, buf)) { - dev_err(dev, "dma map failed\n"); - return -EIO; - } - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - buf, len, period, + substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, DMA_PREP_INTERRUPT | DMA_CTRL_ACK); @@ -282,18 +219,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, desc->callback = rsnd_dmaen_complete; desc->callback_param = rsnd_mod_get(dma); - dmaen->dma_buf = buf; - dmaen->dma_len = len; - dmaen->dma_period = period; - dmaen->dma_cnt = 0; - - /* - * synchronize this and next period - * see - * __rsnd_dmaen_complete() - */ - for (i = 0; i < 2; i++) - rsnd_dmaen_sync(dmaen, io, i); + dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); dmaen->cookie = dmaengine_submit(desc); if (dmaen->cookie < 0) { -- 1.9.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix out-of-bound error
by Jaejoong Kim 16 Nov '17

16 Nov '17

The snd_usb_copy_string_desc() retrieves the usb string corresponding to the index number thought the usb_string(). And for NULL-terminated, insert '0' by using the return value of usb_string() as the index of buffer to hold the string. The problem is that the usb_string() also returns the length of the string read(>= 0), but it can also return a negative value, the error or status value of usb_control_msg(). If iClockSource is '0' as shown below, usb_string() will return -EINVAL. This will result in '0' being inserted into buf[-22], and the following KASAN out-of-bound error message will be output. AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 10 (CLOCK_SOURCE) bClockID 1 bmAttributes 0x07 Internal programmable Clock (synced to SOF) bmControls 0x07 Clock Frequency Control (read/write) Clock Validity Control (read-only) bAssocTerminal 0 iClockSource 0 To fix out-of-bound error, insert 0 only if the return value of usb_string() is greater than 0. ================================================================== BUG: KASAN: stack-out-of-bounds in parse_audio_unit+0x1327/0x1960 [snd_usb_audio] Write of size 1 at addr ffff88007e66735a by task systemd-udevd/18376 CPU: 0 PID: 18376 Comm: systemd-udevd Not tainted 4.13.0+ #3 Hardware name: LG Electronics 15N540-RFLGL/White Tip Mountain, BIOS 15N5 Call Trace: dump_stack+0x63/0x8d print_address_description+0x70/0x290 ? parse_audio_unit+0x1327/0x1960 [snd_usb_audio] kasan_report+0x265/0x350 __asan_store1+0x4a/0x50 parse_audio_unit+0x1327/0x1960 [snd_usb_audio] ? save_stack+0xb5/0xd0 ? save_stack_trace+0x1b/0x20 ? save_stack+0x46/0xd0 ? kasan_kmalloc+0xad/0xe0 ? kmem_cache_alloc_trace+0xff/0x230 ? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio] ? usb_audio_probe+0x4de/0xf40 [snd_usb_audio] ? usb_probe_interface+0x1f5/0x440 ? driver_probe_device+0x3ed/0x660 ? build_feature_ctl+0xb10/0xb10 [snd_usb_audio] ? save_stack_trace+0x1b/0x20 ? init_object+0x69/0xa0 ? snd_usb_find_csint_desc+0xa8/0xf0 [snd_usb_audio] snd_usb_mixer_controls+0x1dc/0x370 [snd_usb_audio] ? build_audio_procunit+0x890/0x890 [snd_usb_audio] ? snd_usb_create_mixer+0xb0/0x4b0 [snd_usb_audio] ? kmem_cache_alloc_trace+0xff/0x230 ? usb_ifnum_to_if+0xbd/0xf0 snd_usb_create_mixer+0x25b/0x4b0 [snd_usb_audio] ? snd_usb_create_stream+0x255/0x2c0 [snd_usb_audio] usb_audio_probe+0x4de/0xf40 [snd_usb_audio] ? snd_usb_autosuspend.part.7+0x30/0x30 [snd_usb_audio] ? __pm_runtime_idle+0x90/0x90 ? kernfs_activate+0xa6/0xc0 ? usb_match_one_id_intf+0xdc/0x130 ? __pm_runtime_set_status+0x2d4/0x450 usb_probe_interface+0x1f5/0x440 Cc: <stable(a)vger.kernel.org> Signed-off-by: Jaejoong Kim <climbbb.kim(a)gmail.com> --- The AudioControl Interface Descriptor in commit message is from lsusb output with real usb audio DAC. The usb audio product causing the OOB are as follows: http://www.lg.com/uk/lg-friends/lg-AFD-1200 It only prints OOB error and usb audio works well. :) sound/usb/mixer.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index e630813..5a83c2c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -204,7 +204,8 @@ static int snd_usb_copy_string_desc(struct mixer_build *state, int index, char *buf, int maxlen) { int len = usb_string(state->chip->dev, index, buf, maxlen - 1); - buf[len] = 0; + if (len > 0) + buf[len] = 0; return len; } -- 2.7.4

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