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

[alsa-devel] [PATCH] ASoC: fsl_esai: Fix divisor calculation failure at lower ratio
by Nicolin Chen 12 Apr '18

12 Apr '18

When the desired ratio is less than 256, the savesub (tolerance) in the calculation would become 0. This will then fail the loop- search immediately without reporting any errors. But if the ratio is smaller enough, there is no need to calculate the tolerance because PM divisor alone is enough to get the ratio. So a simple fix could be just to set PM directly instead of going into the loop-search. Reported-by: Marek Vasut <marex(a)denx.de> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Marek Vasut <marex(a)denx.de> --- sound/soc/fsl/fsl_esai.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 40a7004..da8fd98 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -144,6 +144,13 @@ static int fsl_esai_divisor_cal(struct snd_soc_dai *dai, bool tx, u32 ratio, psr = ratio <= 256 * maxfp ? ESAI_xCCR_xPSR_BYPASS : ESAI_xCCR_xPSR_DIV8; + /* Do not loop-search if PM (1 ~ 256) alone can serve the ratio */ + if (ratio <= 256) { + pm = ratio; + fp = 1; + goto out; + } + /* Set the max fluctuation -- 0.1% of the max devisor */ savesub = (psr ? 1 : 8) * 256 * maxfp / 1000; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix mode setting when changing channel number
by Nicolin Chen 12 Apr '18

12 Apr '18

This is a partial revert (in a cleaner way) of commit ebf08ae3bc90 ("ASoC: fsl_ssi: Keep ssi->i2s_net updated") to fix a regression at test cases when switching between mono and stereo audio. The problem is that ssi->i2s_net is initialized in set_dai_fmt() only, while this set_dai_fmt() is only called during the dai-link probe(). The original patch assumed set_dai_fmt() would be called during every playback instance, so it failed at the overriding use cases. This patch adds the local variable i2s_net back to let regular use cases still follow the mode settings from the set_dai_fmt(). Meanwhile, the original commit of keeping ssi->i2s_net updated was to make set_tdm_slot() clean by checking the ssi->i2s_net directly instead of reading SCR register. However, the change itself is not necessary (or even harmful) because the set_tdm_slot() might fail to check the slot number for Normal-Mode-None-Net settings while mono audio cases still need 2 slots. So this patch can also fix it. And it adds an extra line of comments to declare ssi->i2s_net does not reflect the register value but merely the initial setting from the set_dai_fmt(). Reported-by: Mika Penttilä <mika.penttila(a)nextfour.com> Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Mika Penttilä <mika.penttila(a)nextfour.com> --- sound/soc/fsl/fsl_ssi.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 0823b08..89df2d9 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -217,6 +217,7 @@ struct fsl_ssi_soc_data { * @dai_fmt: DAI configuration this device is currently used with * @streams: Mask of current active streams: BIT(TX) and BIT(RX) * @i2s_net: I2S and Network mode configurations of SCR register + * (this is the initial settings based on the DAI format) * @synchronous: Use synchronous mode - both of TX and RX use STCK and SFCK * @use_dma: DMA is used or FIQ with stream filter * @use_dual_fifo: DMA with support for dual FIFO mode @@ -829,16 +830,23 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } if (!fsl_ssi_is_ac97(ssi)) { + /* + * Keep the ssi->i2s_net intact while having a local variable + * to override settings for special use cases. Otherwise, the + * ssi->i2s_net will lose the settings for regular use cases. + */ + u8 i2s_net = ssi->i2s_net; + /* Normal + Network mode to send 16-bit data in 32-bit frames */ if (fsl_ssi_is_i2s_cbm_cfs(ssi) && sample_size == 16) - ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; + i2s_net = SSI_SCR_I2S_MODE_NORMAL | SSI_SCR_NET; /* Use Normal mode to send mono data at 1st slot of 2 slots */ if (channels == 1) - ssi->i2s_net = SSI_SCR_I2S_MODE_NORMAL; + i2s_net = SSI_SCR_I2S_MODE_NORMAL; regmap_update_bits(regs, REG_SSI_SCR, - SSI_SCR_I2S_NET_MASK, ssi->i2s_net); + SSI_SCR_I2S_NET_MASK, i2s_net); } /* In synchronous mode, the SSI uses STCCR for capture */ -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rsnd: mark PM functions __maybe_unused
by Arnd Bergmann 12 Apr '18

12 Apr '18

The suspend/resume callbacks are now optional, leading to a warning when they are unused: sound/soc/sh/rcar/core.c:1548:12: error: 'rsnd_resume' defined but not used [-Werror=unused-function] static int rsnd_resume(struct device *dev) ^~~~~~~~~~~ sound/soc/sh/rcar/core.c:1539:12: error: 'rsnd_suspend' defined but not used [-Werror=unused-function] static int rsnd_suspend(struct device *dev) This marks the as __maybe_unused to avoid the warning. Fixes: f8a9a29c4fe9 ("ASoC: rsnd: set pm_ops in hibernate-compatible way") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sh/rcar/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6a76688a8ba9..94f081b93258 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1536,7 +1536,7 @@ static int rsnd_remove(struct platform_device *pdev) return ret; } -static int rsnd_suspend(struct device *dev) +static int __maybe_unused rsnd_suspend(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); @@ -1545,7 +1545,7 @@ static int rsnd_suspend(struct device *dev) return 0; } -static int rsnd_resume(struct device *dev) +static int __maybe_unused rsnd_resume(struct device *dev) { struct rsnd_priv *priv = dev_get_drvdata(dev); -- 2.9.0

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[alsa-devel] [PATCH] ASoC: Intel: atom: fix ACPI/PCI Kconfig
by Pierre-Louis Bossart 12 Apr '18

12 Apr '18

The split between ACPI and PCI platforms generated issues with randconfig: with SND_SST_ATOM_HIFI2_PLATFORM_PCI=y and SND_SST_ATOM_HIFI2_PLATFORM=m, we get this module link failure: ERROR: "sst_context_init" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_context_cleanup" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_alloc_drv_context" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "intel_sst_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! ERROR: "sst_configure_runtime_pm" [sound/soc/intel/atom/sst/snd-intel-sst-acpi.ko] undefined! To keep things simple, let's expose two configs for SND_SST_ATOM_HIFI2_PLATFORM_PCI and SND_SST_ATOM_HIFI2_PLATFORM_ACPI, which select a common SND_SST_ATOM_HIFI2_PLATFORM option. To avoid breaking existing solutions with the semantics change, SND_SST_ATOM_HIFI2_PLATFORM_ACPI uses "default ACPI" so that "make oldnoconfig" and "make olddefconfig" still work as expected. Also remove mentions of Medfield while we are at it since it was removed recently. Reported-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 4772c16ede52 ("ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 22 +++++++++++++--------- 1 file changed, 13 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index f2c9e8c5970a..d7ac77b0197b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -72,24 +72,28 @@ config SND_SOC_INTEL_BAYTRAIL for Baytrail Chromebooks but this option is now deprecated and is not recommended, use SND_SST_ATOM_HIFI2_PLATFORM instead. +config SND_SST_ATOM_HIFI2_PLATFORM + tristate + select SND_SOC_COMPRESS + config SND_SST_ATOM_HIFI2_PLATFORM_PCI - tristate "PCI HiFi2 (Medfield, Merrifield) Platforms" + tristate "PCI HiFi2 (Merrifield) Platforms" depends on X86 && PCI select SND_SST_IPC_PCI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM help - If you have a Intel Medfield or Merrifield/Edison platform, then + If you have a Intel Merrifield/Edison platform, then enable this option by saying Y or m. Distros will typically not - enable this option: Medfield devices are not available to - developers and while Merrifield/Edison can run a mainline kernel with - limited functionality it will require a firmware file which - is not in the standard firmware tree + enable this option: while Merrifield/Edison can run a mainline + kernel with limited functionality it will require a firmware file + which is not in the standard firmware tree -config SND_SST_ATOM_HIFI2_PLATFORM +config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" + default ACPI depends on X86 && ACPI select SND_SST_IPC_ACPI - select SND_SOC_COMPRESS + select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH select IOSF_MBI help -- 2.14.1

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[alsa-devel] [Xen-devel][PATCH v3 0/5] sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 12 Apr '18

12 Apr '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Hello, all! In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - bump protocol version to 2 - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events - add XENSND_OP_HW_PARAM_QUERY request to read/update stream configuration space: request passes desired intervals/formats for the stream parameters and the response returns allowed intervals and formats mask that can be used. Changes since v2: 1. Konrad's r-b tag for version patch 2. MAJOR: changed req/resp/evt packet sizes from 32 to 64 octets 3. Reworked XENSND_OP_HW_PARAM_QUERY so it now sends all parameters at once, allowing to check all the configuration space. 4. Minor documentation cleanup (added missed "reserved" fields) Changes since v1: 1. Changed protocol version definition from string to integer, so it can easily be used in comparisons. Konrad, I have removed your r-b tag for the reason of this change. 2. In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameter given: request passes desired parameter interval (mask) and the response to this request returns min/max interval (mask) for the parameter to be used. Parameters supported by this request/response: - format mask - sample rate interval - number of channels interval - buffer size, interval, frames - period size, interval, frames Oleksandr Andrushchenko (5): sndif: Introduce protocol version sndif: Fix missed "reserved" fields in comments sndif: Make requests and responses 64 octets long sndif: Add explicit back and front synchronization sndif: Add explicit back and front parameter negotiation xen/include/public/io/sndif.h | 322 +++++++++++++++++++++++++++++++++++++++--- 1 file changed, 306 insertions(+), 16 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: dmic: Fix clock parenting
by Peter Ujfalusi 12 Apr '18

12 Apr '18

From: Tero Kristo <t-kristo(a)ti.com> In 4.16 the clock hierarchy got changed by a5c82a09d876 ARM: dts: omap4: add clkctrl nodes The fck of dmic is no longer a mux clock, it's parent is. Signed-off-by: Tero Kristo <t-kristo(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: stable(a)vger.kernel.org # 4.16+ --- Mark, can you apply this for 4.17 as DMIC got broken already in 4.16. Regards, Peter sound/soc/omap/omap-dmic.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index c7101d851213..51dd7c65096b 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -281,7 +281,7 @@ static int omap_dmic_dai_trigger(struct snd_pcm_substream *substream, static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, unsigned int freq) { - struct clk *parent_clk; + struct clk *parent_clk, *mux; char *parent_clk_name; int ret = 0; @@ -329,14 +329,21 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, return -ENODEV; } + mux = clk_get_parent(dmic->fclk); + if (IS_ERR(mux)) { + dev_err(dmic->dev, "can't get fck mux parent\n"); + clk_put(parent_clk); + return -ENODEV; + } + mutex_lock(&dmic->mutex); if (dmic->active) { /* disable clock while reparenting */ pm_runtime_put_sync(dmic->dev); - ret = clk_set_parent(dmic->fclk, parent_clk); + ret = clk_set_parent(mux, parent_clk); pm_runtime_get_sync(dmic->dev); } else { - ret = clk_set_parent(dmic->fclk, parent_clk); + ret = clk_set_parent(mux, parent_clk); } mutex_unlock(&dmic->mutex); @@ -349,6 +356,7 @@ static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id, dmic->fclk_freq = freq; err_busy: + clk_put(mux); clk_put(parent_clk); return ret; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ALSA: line6: Use correct endpoint type for midi output
by Fabián Inostroza 12 Apr '18

12 Apr '18

Hi, Sending MIDI messages to a PODxt through the USB connection shows "usb_submit_urb failed" in dmesg and the message is not received by the POD. The error is caused because in the funcion send_midi_async() in midi.c there is a call to usb_sndbulkpipe() for endpoint 3 OUT, but the PODxt USB descriptor shows that this endpoint it's an interrupt endpoint. Patch tested with PODxt only. Signed-off-by: Fabián Inostroza <fabianinostroza(a)udec.cl> --- sound/usb/line6/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/midi.c b/sound/usb/line6/midi.c index 6d7cde56..e2cf55c5 100644 --- a/sound/usb/line6/midi.c +++ b/sound/usb/line6/midi.c @@ -125,7 +125,7 @@ static int send_midi_async(struct usb_line6 *line6, unsigned char *data, } usb_fill_int_urb(urb, line6->usbdev, - usb_sndbulkpipe(line6->usbdev, + usb_sndintpipe(line6->usbdev, line6->properties->ep_ctrl_w), transfer_buffer, length, midi_sent, line6, line6->interval); -- 2.16.3

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[alsa-devel] [PATCH 1/2] ASoC: AMD: Support headset button on Stoney DA7219
by Akshu Agrawal 12 Apr '18

12 Apr '18

Adds headset button support. TEST=Tested Volume UP/Down functionality Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index b205c78..d9491e1 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -33,6 +33,7 @@ #include <linux/gpio.h> #include <linux/module.h> #include <linux/i2c.h> +#include <linux/input.h> #include <linux/acpi.h> #include "../codecs/da7219.h" @@ -51,6 +52,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack *jack; dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); @@ -80,6 +82,12 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } + jack = &cz_jack; + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + da7219_aad_jack_det(component, &cz_jack); return 0; -- 1.9.1

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[alsa-devel] BUG: unable to handle kernel paging request in snd_pcm_format_set_silence
by syzbot 11 Apr '18

11 Apr '18

Hello, syzbot hit the following crash on upstream commit f2d285669aae656dfeafa0bf25e86bbbc5d22329 (Tue Apr 3 17:45:39 2018 +0000) Merge tag 'pm-4.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/rafael/linux-pm syzbot dashboard link: https://syzkaller.appspot.com/bug?extid=f973e52978d95f818f69 So far this crash happened 2 times on upstream. syzkaller reproducer: https://syzkaller.appspot.com/x/repro.syz?id=5368241429938176 Raw console output: https://syzkaller.appspot.com/x/log.txt?id=6066534392791040 Kernel config: https://syzkaller.appspot.com/x/.config?id=686016073509112605 compiler: gcc (GCC) 7.1.1 20170620 user-space arch: i386 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+f973e52978d95f818f69(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 BUG: unable to handle kernel paging request at ffffc90004cc7000 PGD 1dad42067 P4D 1dad42067 PUD 1dad43067 PMD 1c5cb3067 PTE 0 Oops: 0002 [#1] SMP KASAN Dumping ftrace buffer: (ftrace buffer empty) Modules linked in: CPU: 0 PID: 5516 Comm: syz-executor3 Not tainted 4.16.0+ #288 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:memset_erms+0x9/0x10 arch/x86/lib/memset_64.S:65 RSP: 0018:ffff8801d7fc7368 EFLAGS: 00010246 RAX: fffff520009b9600 RBX: ffffc90004cc7000 RCX: 0000000000104500 RDX: 0000000000104500 RSI: 0000000000000000 RDI: ffffc90004cc7000 RBP: ffff8801d7fc7388 R08: 0000000000000000 R09: ffffc90004cc7000 R10: 00000000000208a0 R11: fffff520009b969f R12: 0000000000104500 R13: 0000000000000000 R14: ffffffff88c9185c R15: 0000000000000010 FS: 0000000000000000(0000) GS:ffff8801db000000(0063) knlGS:00000000f7f5fb40 CS: 0010 DS: 002b ES: 002b CR0: 0000000080050033 CR2: ffffc90004cc7000 CR3: 00000001d765c000 CR4: 00000000001406f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: memset include/linux/string.h:330 [inline] snd_pcm_format_set_silence+0x13f/0x4b0 sound/core/pcm_misc.c:431 loopback_prepare+0x92b/0xbf0 sound/drivers/aloop.c:347 snd_pcm_do_prepare+0x58/0x90 sound/core/pcm_native.c:1640 snd_pcm_action_single+0x70/0x110 sound/core/pcm_native.c:1084 snd_pcm_action_nonatomic+0x95/0xa0 sound/core/pcm_native.c:1151 snd_pcm_prepare+0x115/0x1a0 sound/core/pcm_native.c:1687 snd_pcm_kernel_ioctl+0x97/0x240 sound/core/pcm_native.c:2998 snd_pcm_oss_prepare+0x49/0x150 sound/core/oss/pcm_oss.c:1104 snd_pcm_oss_make_ready+0xf9/0x130 sound/core/oss/pcm_oss.c:1132 snd_pcm_oss_set_trigger+0x384/0x930 sound/core/oss/pcm_oss.c:1959 snd_pcm_oss_poll+0x7aa/0xd60 sound/core/oss/pcm_oss.c:2727 do_pollfd fs/select.c:832 [inline] do_poll fs/select.c:883 [inline] do_sys_poll+0x719/0x1050 fs/select.c:977 SYSC_poll fs/select.c:1035 [inline] SyS_poll+0x10d/0x450 fs/select.c:1023 do_syscall_32_irqs_on arch/x86/entry/common.c:330 [inline] do_fast_syscall_32+0x3ec/0xf9f arch/x86/entry/common.c:392 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7f63c99 RSP: 002b:00000000f7f5f0ac EFLAGS: 00000282 ORIG_RAX: 00000000000000a8 RAX: ffffffffffffffda RBX: 0000000020000100 RCX: 0000000000000001 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Code: 48 c1 e9 03 40 0f b6 f6 48 b8 01 01 01 01 01 01 01 01 48 0f af c6 f3 48 ab 89 d1 f3 aa 4c 89 c8 c3 90 49 89 f9 40 88 f0 48 89 d1 <f3> aa 4c 89 c8 c3 90 49 89 fa 40 0f b6 ce 48 b8 01 01 01 01 01 RIP: memset_erms+0x9/0x10 arch/x86/lib/memset_64.S:65 RSP: ffff8801d7fc7368 CR2: ffffc90004cc7000 ---[ end trace 95a476f9da224f32 ]--- --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

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[alsa-devel] [PATCH v2 0/5] ASoC: Intel: geminilake: Adds da7219+max98537 machine driver support
by Naveen Manohar 11 Apr '18

11 Apr '18

Patch adds required changes in bxt machine & corresponding driver entry in machine to use MAX98357A codec as speaker on SSP1 and DA7219 codec as headset on SSP2 for GLK board. Naveen Manohar (5): ASoC: Intel: broxton: reduce machine name for bxt_da7219_max98357a ASoC: Intel: Headset button support in broxton machine driver ASoC: Intel: Add Geminilake Dialog Maxim machine driver support ASoC: Intel: glk: Add DAI links for Multi-Playback ASoC: Intel: Add Geminilake Dialog+Maxim machine driver entry sound/soc/intel/boards/bxt_da7219_max98357a.c | 152 ++++++++++++++++++++++---- sound/soc/intel/skylake/skl.c | 9 +- 2 files changed, 141 insertions(+), 20 deletions(-) -- v1->v2: Unified dai-link usage between bxt & glk mach, corrected jack CTRL 2.7.4

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