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

[alsa-devel] [PATCH] ALSA: usb-audio: Change the semantics of the enable option
by Tamaki Nishino 30 Nov '17

30 Nov '17

This patch changes the semantics of the enable option for snd-usb-audio in order to allow users to disable a device specified by either or both of the vendor id and the product id. Signed-off-by: Tamaki Nishino <otamachan(a)gmail.com> --- sound/usb/card.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 23d1d23..8018d56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -585,15 +585,24 @@ static int usb_audio_probe(struct usb_interface *intf, * now look for an empty slot and create a new card instance */ for (i = 0; i < SNDRV_CARDS; i++) - if (enable[i] && ! usb_chip[i] && + if (!usb_chip[i] && (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - err = snd_usb_audio_create(intf, dev, i, quirk, - id, &chip); - if (err < 0) + if (enable[i]) { + err = snd_usb_audio_create(intf, dev, i, quirk, + id, &chip); + if (err < 0) + goto __error; + chip->pm_intf = intf; + break; + } else if (vid[i] != -1 || pid[i] != -1) { + dev_info(&dev->dev, + "device (%04x:%04x) is disabled\n", + USB_ID_VENDOR(id), + USB_ID_PRODUCT(id)); + err = -ENOENT; goto __error; - chip->pm_intf = intf; - break; + } } if (!chip) { dev_err(&dev->dev, "no available usb audio device\n"); -- 2.7.4

2 1

[alsa-devel] [RFC PATCH v2 0/7] Fix Intel audio Kconfig issues
by Pierre-Louis Bossart 30 Nov '17

30 Nov '17

The first patch implements what Linus, Takashi and Mark requested: a top-level selector defaulting to 'y' to easily filter all other options and with no impact on code generation. There should be no functionality change and will avoid breaking audio for people using make oldnoconfig. The rest of the patch series does a more in-depth cleanup. It was tested on Baytrail/Cherrytrail/Skylake platforms with no regressions observed and no reports of any compilation issues with 0-day or randconfig. The 4th patch is really the most important one, there were nested configs which made no sense to me. I don't know the history which led to such complicated stuff but simpler is better. The last 3 patches are just clean-ups of the machine driver configs, for some reason there is no consistency in the settings so I tried to apply common sense and use the same rules. At Andy Shevchenko's suggestion, I also replaced the broken dependency on X86_INTEL_LPSS by MFD_INTEL_LPSS for Skylake+ machines, which looks ok but needs to be tested further, hence the RFCv2 status (this change is fine for Chrome, Ubuntu and other distros but better be cautious with a second set of tests with randconfig and on Skylake) Changes since initial RFC: Removed default n Added help text for HASWELL, BAYTRAIL (legacy) and SKYLAKE options Made top level machine driver selection dependent on INTEL_SST_TOPLEVEL. Added help text for PCI and HIFI2 platforms Replaced X86_INTEL_LPSS by MFD_INTEL_LPSS for Skylake+ devices Fixed a couple of indentation issues Pierre-Louis Bossart (7): ASoC: Intel: Fix Kconfig with top-level selector ASoC: Intel: Kconfig: Simplify-clarify ACPI/PCI dependencies ASoC: Intel: document what Kconfig options do ASoC: Intel: Fix nested/unnecessary Kconfig dependencies ASoC: Intel: boards: align Kconfig dependencies for Haswell/Broadwell ASoC: Intel: boards: align Kconfig configurations for HiFi2 ASoC: Intel: boards: align/fix SKL/BXT/KBL Kconfigs sound/soc/intel/Kconfig | 118 +++++++++++++++++++++++-------- sound/soc/intel/boards/Kconfig | 155 ++++++++++++++++++++++------------------- 2 files changed, 172 insertions(+), 101 deletions(-) -- 2.14.1

3 13

[alsa-devel] [PATCH] ASoC: codecs: msm8916-wcd: Fix supported formats
by srinivas.kandagatla＠linaro.org 30 Nov '17

30 Nov '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This codec is configurable for only 16 bit and 32 bit samples, so reflect this in the supported formats also remove 24bit sample from supported list. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/msm8916-wcd-analog.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 5f3c42c4f74a..066ea2f4ce7b 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -267,7 +267,7 @@ #define MSM8916_WCD_ANALOG_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) #define MSM8916_WCD_ANALOG_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) static int btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_BTN_4; diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index a10a724eb448..13354d6304a8 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -194,7 +194,7 @@ SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_48000) #define MSM8916_WCD_DIGITAL_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S32_LE) struct msm8916_wcd_digital_priv { struct clk *ahbclk, *mclk; @@ -645,7 +645,7 @@ static int msm8916_wcd_digital_hw_params(struct snd_pcm_substream *substream, RX_I2S_CTL_RX_I2S_MODE_MASK, RX_I2S_CTL_RX_I2S_MODE_16); break; - case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: snd_soc_update_bits(dai->codec, LPASS_CDC_CLK_TX_I2S_CTL, TX_I2S_CTL_TX_I2S_MODE_MASK, TX_I2S_CTL_TX_I2S_MODE_32); -- 2.15.0

2 1

[alsa-devel] [PATCH] ASoC: nau8825: set clear_irq when imm IRQ happened
by John Hsu 30 Nov '17

30 Nov '17

Although the crosstalk is disabled, it is better to set clear_irq properly when the impedance measurement interrupt happens. It can avoid that the driver clears other IRQs by accident if the active_irq has another IRQ events. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index d3c1a02..603cd72 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1733,10 +1733,9 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) nau8825->xtalk_event_mask = event_mask; } } else if (active_irq & NAU8825_IMPEDANCE_MEAS_IRQ) { - if (nau8825->xtalk_enable) { + if (nau8825->xtalk_enable) schedule_work(&nau8825->xtalk_work); - clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; - } + clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == NAU8825_JACK_INSERTION_DETECTED) { /* One more step to check GPIO status directly. Thus, the -- 2.6.4

3 3

[alsa-devel] [PATCH v4 1/2] ALSA: pcm: add SNDRV_PCM_FORMAT_{S, U}20
by Maciej S. Szmigiero 30 Nov '17

30 Nov '17

This format is similar to existing SNDRV_PCM_FORMAT_{S,U}20_3 that keep 20-bit PCM samples in 3 bytes, however i.MX6 platform SSI FIFO does not allow 3-byte accesses (including DMA) so a 4-byte (more conventional) format is needed for it. Signed-off-by: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: Drop "_4" suffix from these formats since they aren't non-standard ones, use empty format slots starting from format number 25 for them, add information that they are LSB justified formats. Changes from v2: Adapt a comment in sound/core/pcm_misc.c so it still refers to the same sample formats as before. Changes from v3: Make the comment style in include/uapi/sound/asound.h match the style of comments at other sample formats. Corresponding alsa-lib changes will be posted as soon as this patch is merged on the kernel side, to keep alsa-lib and kernel synchronized. include/sound/pcm.h | 8 ++++++++ include/sound/soc-dai.h | 2 ++ include/uapi/sound/asound.h | 9 +++++++++ sound/core/pcm_misc.c | 19 ++++++++++++++++++- 4 files changed, 37 insertions(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 24febf9e177c..e054c583d3b3 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -169,6 +169,10 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_IMA_ADPCM _SNDRV_PCM_FMTBIT(IMA_ADPCM) #define SNDRV_PCM_FMTBIT_MPEG _SNDRV_PCM_FMTBIT(MPEG) #define SNDRV_PCM_FMTBIT_GSM _SNDRV_PCM_FMTBIT(GSM) +#define SNDRV_PCM_FMTBIT_S20_LE _SNDRV_PCM_FMTBIT(S20_LE) +#define SNDRV_PCM_FMTBIT_U20_LE _SNDRV_PCM_FMTBIT(U20_LE) +#define SNDRV_PCM_FMTBIT_S20_BE _SNDRV_PCM_FMTBIT(S20_BE) +#define SNDRV_PCM_FMTBIT_U20_BE _SNDRV_PCM_FMTBIT(U20_BE) #define SNDRV_PCM_FMTBIT_SPECIAL _SNDRV_PCM_FMTBIT(SPECIAL) #define SNDRV_PCM_FMTBIT_S24_3LE _SNDRV_PCM_FMTBIT(S24_3LE) #define SNDRV_PCM_FMTBIT_U24_3LE _SNDRV_PCM_FMTBIT(U24_3LE) @@ -202,6 +206,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_LE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_LE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_LE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FMTBIT_S16 SNDRV_PCM_FMTBIT_S16_BE @@ -213,6 +219,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_FMTBIT_FLOAT SNDRV_PCM_FMTBIT_FLOAT_BE #define SNDRV_PCM_FMTBIT_FLOAT64 SNDRV_PCM_FMTBIT_FLOAT64_BE #define SNDRV_PCM_FMTBIT_IEC958_SUBFRAME SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FMTBIT_S20 SNDRV_PCM_FMTBIT_S20_BE +#define SNDRV_PCM_FMTBIT_U20 SNDRV_PCM_FMTBIT_U20_BE #endif struct snd_pcm_file { diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 58acd00cae19..d970879944fc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -102,6 +102,8 @@ struct snd_compr_stream; SNDRV_PCM_FMTBIT_S16_BE |\ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S20_LE |\ + SNDRV_PCM_FMTBIT_S20_BE |\ SNDRV_PCM_FMTBIT_S24_3LE |\ SNDRV_PCM_FMTBIT_S24_3BE |\ SNDRV_PCM_FMTBIT_S32_LE |\ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index c227ccba60ae..07d61583fd02 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -214,6 +214,11 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_IMA_ADPCM ((__force snd_pcm_format_t) 22) #define SNDRV_PCM_FORMAT_MPEG ((__force snd_pcm_format_t) 23) #define SNDRV_PCM_FORMAT_GSM ((__force snd_pcm_format_t) 24) +#define SNDRV_PCM_FORMAT_S20_LE ((__force snd_pcm_format_t) 25) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_S20_BE ((__force snd_pcm_format_t) 26) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_U20_LE ((__force snd_pcm_format_t) 27) /* in four bytes, LSB justified */ +#define SNDRV_PCM_FORMAT_U20_BE ((__force snd_pcm_format_t) 28) /* in four bytes, LSB justified */ +/* gap in the numbering for a future standard linear format */ #define SNDRV_PCM_FORMAT_SPECIAL ((__force snd_pcm_format_t) 31) #define SNDRV_PCM_FORMAT_S24_3LE ((__force snd_pcm_format_t) 32) /* in three bytes */ #define SNDRV_PCM_FORMAT_S24_3BE ((__force snd_pcm_format_t) 33) /* in three bytes */ @@ -248,6 +253,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_LE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_LE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_LE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_LE #endif #ifdef SNDRV_BIG_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_BE @@ -259,6 +266,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_FLOAT SNDRV_PCM_FORMAT_FLOAT_BE #define SNDRV_PCM_FORMAT_FLOAT64 SNDRV_PCM_FORMAT_FLOAT64_BE #define SNDRV_PCM_FORMAT_IEC958_SUBFRAME SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE +#define SNDRV_PCM_FORMAT_S20 SNDRV_PCM_FORMAT_S20_BE +#define SNDRV_PCM_FORMAT_U20 SNDRV_PCM_FORMAT_U20_BE #endif typedef int __bitwise snd_pcm_subformat_t; diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 9be81025372f..c4eb561d2008 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -163,13 +163,30 @@ static struct pcm_format_data pcm_formats[(INT)SNDRV_PCM_FORMAT_LAST+1] = { .width = 32, .phys = 32, .le = 0, .signd = 0, .silence = { 0x69, 0x69, 0x69, 0x69 }, }, - /* FIXME: the following three formats are not defined properly yet */ + /* FIXME: the following two formats are not defined properly yet */ [SNDRV_PCM_FORMAT_MPEG] = { .le = -1, .signd = -1, }, [SNDRV_PCM_FORMAT_GSM] = { .le = -1, .signd = -1, }, + [SNDRV_PCM_FORMAT_S20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_S20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 1, + .silence = {}, + }, + [SNDRV_PCM_FORMAT_U20_LE] = { + .width = 20, .phys = 32, .le = 1, .signd = 0, + .silence = { 0x00, 0x00, 0x08, 0x00 }, + }, + [SNDRV_PCM_FORMAT_U20_BE] = { + .width = 20, .phys = 32, .le = 0, .signd = 0, + .silence = { 0x00, 0x08, 0x00, 0x00 }, + }, + /* FIXME: the following format is not defined properly yet */ [SNDRV_PCM_FORMAT_SPECIAL] = { .le = -1, .signd = -1, },

3 5

[alsa-devel] [PATCH] ALSA: usb-audio: Fix the enable parameter behavior
by Tamaki Nishino 30 Nov '17

30 Nov '17

The enable parameter doesn't seem to work as expected. Even when the enable parameter is set to false in /etc/modprobe.d/alsa-base.conf like "options snd-usb-audio index=-2 vid=0x1415 pid=0x1d27 enable=0", the specified device is not disabled. This patch fixes the enable parameter behavior to disable the device correctly. Signed-off-by: Tamaki Nishino <otamachan(a)gmail.com> --- sound/usb/card.c | 21 +++++++++++++++------ 1 file changed, 15 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 23d1d23..8018d56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -585,15 +585,24 @@ static int usb_audio_probe(struct usb_interface *intf, * now look for an empty slot and create a new card instance */ for (i = 0; i < SNDRV_CARDS; i++) - if (enable[i] && ! usb_chip[i] && + if (!usb_chip[i] && (vid[i] == -1 || vid[i] == USB_ID_VENDOR(id)) && (pid[i] == -1 || pid[i] == USB_ID_PRODUCT(id))) { - err = snd_usb_audio_create(intf, dev, i, quirk, - id, &chip); - if (err < 0) + if (enable[i]) { + err = snd_usb_audio_create(intf, dev, i, quirk, + id, &chip); + if (err < 0) + goto __error; + chip->pm_intf = intf; + break; + } else if (vid[i] != -1 || pid[i] != -1) { + dev_info(&dev->dev, + "device (%04x:%04x) is disabled\n", + USB_ID_VENDOR(id), + USB_ID_PRODUCT(id)); + err = -ENOENT; goto __error; - chip->pm_intf = intf; - break; + } } if (!chip) { dev_err(&dev->dev, "no available usb audio device\n"); -- 2.7.4

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[alsa-devel] WARNING in initialize_timer
by syzbot 30 Nov '17

30 Nov '17

Hello, syzkaller hit the following crash on 43570f0383d6d5879ae585e6c3cf027ba321546f git://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/master compiler: gcc (GCC) 7.1.1 20170620 .config is attached Raw console output is attached. Unfortunately, I don't have any reproducer for this bug yet. binder: 4519:4521 ioctl 40046205 2 returned -22 binder: 4519:4521 ioctl c0106403 20008ff0 returned -22 WARNING: CPU: 0 PID: 4528 at sound/core/seq/seq_timer.c:358 initialize_timer+0x22d/0x290 sound/core/seq/seq_timer.c:358 Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 4528 Comm: syz-executor7 Not tainted 4.15.0-rc1+ #199 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x257 lib/dump_stack.c:53 panic+0x1e4/0x41c kernel/panic.c:183 __warn+0x1dc/0x200 kernel/panic.c:547 report_bug+0x211/0x2d0 lib/bug.c:184 fixup_bug.part.11+0x37/0x80 arch/x86/kernel/traps.c:177 fixup_bug arch/x86/kernel/traps.c:246 [inline] do_error_trap+0x2d7/0x3e0 arch/x86/kernel/traps.c:295 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:314 invalid_op+0x18/0x20 arch/x86/entry/entry_64.S:930 RIP: 0010:initialize_timer+0x22d/0x290 sound/core/seq/seq_timer.c:358 RSP: 0018:ffff8801c7dbf880 EFLAGS: 00010012 RAX: 0000000000010000 RBX: ffff8801cdb75800 RCX: ffffffff840e956d RDX: 00000000000006b6 RSI: ffffc90001f35000 RDI: ffff8801cdb75850 RBP: ffff8801c7dbf8a8 R08: ffffffff86445678 R09: 1ffff10038fb7eb9 R10: ffff8801c97be2c0 R11: 0000000000000001 R12: 0000000000000000 R13: 0000000000000286 R14: ffff8801cdb75850 R15: ffff8801d92ecb40 seq_timer_start sound/core/seq/seq_timer.c:391 [inline] snd_seq_timer_start+0x147/0x2b0 sound/core/seq/seq_timer.c:405 snd_seq_queue_process_event sound/core/seq/seq_queue.c:698 [inline] snd_seq_control_queue+0x2b9/0x640 sound/core/seq/seq_queue.c:759 event_input_timer+0x25/0x30 sound/core/seq/seq_system.c:118 snd_seq_deliver_single_event.constprop.11+0x2fb/0x940 sound/core/seq/seq_clientmgr.c:621 snd_seq_deliver_event+0x176/0x840 sound/core/seq/seq_clientmgr.c:822 snd_seq_client_enqueue_event+0x2b6/0x420 sound/core/seq/seq_clientmgr.c:940 snd_seq_write+0x34d/0x720 sound/core/seq/seq_clientmgr.c:1069 __vfs_write+0xef/0x970 fs/read_write.c:480 vfs_write+0x18f/0x510 fs/read_write.c:544 SYSC_write fs/read_write.c:589 [inline] SyS_write+0xef/0x220 fs/read_write.c:581 entry_SYSCALL_64_fastpath+0x1f/0x96 RIP: 0033:0x4529d9 RSP: 002b:00007fb9be36ec58 EFLAGS: 00000212 ORIG_RAX: 0000000000000001 RAX: ffffffffffffffda RBX: 0000000000758248 RCX: 00000000004529d9 RDX: 0000000000000030 RSI: 0000000020e6ffd0 RDI: 000000000000001c RBP: 00000000000005a2 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000212 R12: 00000000006f57d0 R13: 00000000ffffffff R14: 00007fb9be36f6d4 R15: 000000000000002a Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- 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. Please credit me with: Reported-by: syzbot <syzkaller(a)googlegroups.com> syzbot will keep track of this bug report. Once a fix for this bug is committed, please reply to this email with: #syz fix: exact-commit-title 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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Re: [alsa-devel] about the dual codecs on Lenovo p520
by Hui Wang 30 Nov '17

30 Nov '17

Hello Takashi, On the Lenovo P520, there are two analog codecs, the codec#0 is alc662, and lineout, linein and mic-in are on the rear panel and are connected to codec#0, the codec#1 is alc235, it connects an internal speaker, front-panel headset jack (headphone+mic). After booting up, only the lineout on codec#0 works, if i manually change value for some amixer controls (pulseaudio can't handle new controls like "Front-Panel"), the linein and mic-in on codec#0 also can work. But all ports on codec#1 can't work no matter how I change the value of the amixer controls. And you mentioned UCM profile, where can i find it and use it? the alsa-info.txt for this machine is at here： http://pastebin.ubuntu.com/26078174/ Thanks, Hui. On 2017年11月30日 15:45, Kailang wrote: > For the UCM frofile was provided by Takashi. > UCM profile was include in ALSA-LIB. > You could ask Takashi. > > >> -----Original Message----- >> From: Hui Wang [mailto:hui.wang@canonical.com] >> Sent: Thursday, November 30, 2017 3:30 PM >> To: Kailang >> Subject: about the dual codecs on Lenovo p520 >> >> Hi Kailang, >> >> >> We have a Lenovo P520, but the audio on this machine does not >> work, this machine has two codecs alc662 (codec#0) for >> rear-panel and alc235 >> (codec#1) for front-panel. Basically both the output and >> input can't work (only lineout on rear-panel works), it is >> because the pulseaudio can't set "Front-Panel ...". >> >> And I found you sent a patch to support dual codecs on P520, >> and you mentioned UCM in the commit header. Where can I find >> that UCM, or how can I make other ports ( like speaker on >> codec#1, headset port on codec#1, line-in and mic-in on codec#0) work? >> >> commit ca169cc2f9e1f8ed9c867b197a49d6dd05e5436d >> Author: Kailang Yang <kailang(a)realtek.com> >> Date: Tue Apr 25 16:17:40 2017 +0800 >> >> ALSA: hda/realtek - Add Dual Codecs support for Lenovo P520/420 >> >> Lenovo P520/420 build with two codecs. >> ALC233 for front panel. >> ALC662 for rear panel. >> This patch will rename capture name for slove conflicts. >> And create a card longname for UCM profile. >> >> Signed-off-by: Kailang Yang <kailang(a)realtek.com> >> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> >> >> >> Thanks, >> >> Hui. >> >> >> >> ------Please consider the environment before printing this e-mail. >>

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Re: [alsa-devel] [PATCH RESEND 1/2] ASoC: Intel: kbl: Remove option of choosing CH count based on pdata
by Mark Brown 30 Nov '17

30 Nov '17

On Wed, Nov 15, 2017 at 03:37:42PM +0530, Naveen Manohar wrote: > Remove option of choosing channel count based on pdata for > kbl_rt5663_max98927 machine driver. > As User-space expects 4ch DMIC data and uses channel-map in ucm to > figure-out the desired channel to use, Hence removing driver change > to allow choosing channels. To repeat my previous question which I don't seem to see a response to: | Which userspace and are you sure there are no boards that only have two | of the mics?

3 2

[alsa-devel] [PATCH] ASoC: amd: added error checks in dma driver
by Vijendar Mukunda 30 Nov '17

30 Nov '17

added additional error checks in acp dma driver Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 39 ++++++++++++++++++++++++++++++++++----- 1 file changed, 34 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index fb09578..71ab5b5 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -848,6 +848,9 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; + buffersize = frames_to_bytes(runtime, runtime->buffer_size); bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); @@ -873,6 +876,8 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; + if (!rtd) + return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, PLAYBACK_START_DMA_DESCR_CH12, @@ -1082,7 +1087,11 @@ static int acp_audio_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, audio_drv_data); /* Initialize the ACP */ - acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + status = acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); + if (status) { + dev_err(&pdev->dev, "ACP Init failed\n"); + return status; + } status = snd_soc_register_platform(&pdev->dev, &acp_asoc_platform); if (status != 0) { @@ -1099,9 +1108,14 @@ static int acp_audio_probe(struct platform_device *pdev) static int acp_audio_remove(struct platform_device *pdev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(&pdev->dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) { + dev_err(&pdev->dev, "ACP Deinit failed\n"); + return status; + } snd_soc_unregister_platform(&pdev->dev); pm_runtime_disable(&pdev->dev); @@ -1111,9 +1125,14 @@ static int acp_audio_remove(struct platform_device *pdev) static int acp_pcm_resume(struct device *dev) { u16 bank; + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed\n"); + return status; + } if (adata->play_stream && adata->play_stream->runtime) { /* For Stoney, Memory gating is disabled,i.e SRAM Banks @@ -1145,18 +1164,28 @@ static int acp_pcm_resume(struct device *dev) static int acp_pcm_runtime_suspend(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_deinit(adata->acp_mmio); + status = acp_deinit(adata->acp_mmio); + if (status) { + dev_err(dev, "ACP Deinit failed\n"); + return status; + } acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } static int acp_pcm_runtime_resume(struct device *dev) { + int status; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio, adata->asic_type); + status = acp_init(adata->acp_mmio, adata->asic_type); + if (status) { + dev_err(dev, "ACP Init failed\n"); + return status; + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } -- 2.7.4

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