January 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: soc-core: fix init platform memory handling
by Curtis Malainey 11 Jan '19

11 Jan '19

snd_soc_init_platform initializes pointers to snd_soc_dai_link which is statically allocated and it does this by devm_kzalloc. In the event of an EPROBE_DEFER the memory will be freed and the pointers are left dangling. snd_soc_init_platform sees the dangling pointers and assumes they are pointing to initialized memory and does not reallocate them on the second probe attempt which results in a use after free bug since devm has freed the memory from the first probe attempt. Since the intention for snd_soc_dai_link->platform is that it can be set statically by the machine driver we need to respect the pointer in the event we did not set it but still catch dangling pointers. The solution is to add a flag to track whether the pointer was dynamically allocated or not. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 11 ++++++----- 2 files changed, 12 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8ec1de856ee7e..e665f111b0d27 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -985,6 +985,12 @@ struct snd_soc_dai_link { /* Do not create a PCM for this DAI link (Backend link) */ unsigned int ignore:1; + /* + * This driver uses legacy platform naming. Set by the core, machine + * drivers should not modify this value. + */ + unsigned int legacy_platform:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0934b36645b3e..cdcc417c94ca1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1034,17 +1034,18 @@ static int snd_soc_init_platform(struct snd_soc_card *card, * this function should be removed in the future */ /* convert Legacy platform link */ - if (!platform) { + if (!platform || dai_link->legacy_platform) { platform = devm_kzalloc(card->dev, sizeof(struct snd_soc_dai_link_component), GFP_KERNEL); if (!platform) return -ENOMEM; - dai_link->platform = platform; - platform->name = dai_link->platform_name; - platform->of_node = dai_link->platform_of_node; - platform->dai_name = NULL; + dai_link->platform = platform; + dai_link->legacy_platform = 1; + platform->name = dai_link->platform_name; + platform->of_node = dai_link->platform_of_node; + platform->dai_name = NULL; } /* if there's no platform we match on the empty platform */ -- 2.20.1.97.g81188d93c3-goog

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[alsa-devel] [PATCH] ASoC: Intel: sst: Delay machine device creation until after initialization
by Stephan Gerhold 10 Jan '19

10 Jan '19

Right now, the machine devices are created early, before the SST context is initialized. This means that SST might not be fully initialized if sst_acpi_probe() fails later on (e.g. after sst_platform_get_resources() if the IRQ does not exist). However, at least sst-mfld-platform assumes that sst_register_dsp() was called to set the (global) "sst" device pointer, which happens only later in sst_acpi_probe() when sst_context_init() is called. This may cause a NULL pointer dereference later when the ALSA device is first opened: BUG: unable to handle kernel NULL pointer dereference at 0000000000000008 PGD 0 P4D 0 Oops: 0000 [#1] PREEMPT SMP PTI CPU: 0 PID: 790 Comm: pulseaudio Not tainted 4.20.0-rc6-mainline-00161-g6531e115b7ab #1 Hardware name: ASUSTeK COMPUTER INC. ME176C/ME176C, BIOS 5.6.5 09/16/2015 RIP: 0010:sst_handle_vb_timer+0x61/0x1b0 [snd_soc_sst_atom_hifi2_platform] Code: 44 24 04 e9 84 00 00 00 31 c9 c7 04 24 ff ff ff ff 66 89 4c 24 06 84 db 0f 84 90 00 00 00 48 8b 05 c4 23 01 00 be 01 00 00 00 <48> 8b 78 08 48 8b 40 10 48 8b 40 48 e8 2e 5e d7 f0 89 c3 85 c0 78 RSP: 0018:ffff9d968099fa30 EFLAGS: 00010202 RAX: 0000000000000000 RBX: 0000000000000001 RCX: 0000000000000000 RDX: 0000000080000001 RSI: 0000000000000001 RDI: 00000000ffffffff RBP: ffff968d33384618 R08: 0000000000000001 R09: 00000000000002e3 R10: ffff968d333a0800 R11: 0000000000000000 R12: ffff968d34bc7c00 R13: ffff968d333a3eb0 R14: 0000000000000001 R15: ffff968d333a08c0 FS: 00007f63a7e7b200(0000) GS:ffff968d37600000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 0000000000000008 CR3: 00000000297b8000 CR4: 00000000001006f0 Call Trace: sst_enable_ssp+0x24/0x40 [snd_soc_sst_atom_hifi2_platform] soc_pcm_open+0xeb/0x960 [snd_soc_core] ? __debugfs_create_file+0xcd/0x120 dpcm_be_dai_startup+0x183/0x3c0 [snd_soc_core] dpcm_fe_dai_open+0x10c/0xab0 [snd_soc_core] snd_pcm_open_substream+0x7f/0x140 [snd_pcm] snd_pcm_open+0xe6/0x220 [snd_pcm] ? wake_up_q+0x70/0x70 snd_pcm_playback_open+0x3d/0x70 [snd_pcm] chrdev_open+0xa3/0x1b0 ? cdev_put.part.0+0x20/0x20 do_dentry_open+0x12f/0x350 path_openat+0x2d1/0x14e0 ? inotify_handle_event+0x17b/0x1e0 do_filp_open+0x93/0x100 ? snd_card_file_remove+0x14b/0x170 [snd] ? __check_object_size+0x102/0x189 ? _raw_spin_unlock+0x12/0x30 do_sys_open+0x186/0x210 do_syscall_64+0x55/0x160 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x7f63a992f4c2 Code: 25 00 00 41 00 3d 00 00 41 00 74 4c 48 8d 05 85 70 0d 00 8b 00 85 c0 75 6d 89 f2 b8 01 01 00 00 48 89 fe bf 9c ff ff ff 0f 05 <48> 3d 00 f0 ff ff 0f 87 a2 00 00 00 48 8b 4c 24 28 64 48 33 0c 25 RSP: 002b:00007ffe71196b70 EFLAGS: 00000246 ORIG_RAX: 0000000000000101 RAX: ffffffffffffffda RBX: 0000000000080802 RCX: 00007f63a992f4c2 RDX: 0000000000080802 RSI: 00007ffe71196d20 RDI: 00000000ffffff9c RBP: 00000000ffffffff R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00007ffe71196c00 R13: 0000000000000004 R14: 00007ffe71196d20 R15: 0000558466055a80 CR2: 0000000000000008 ---[ end trace 34534a02650ee26c ]--- This can be avoided if the machine device creation is delayed in sst_acpi_probe() until after sst_context_init(), when sst_register_dsp() is guaranteed to have already been called. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- An other option to fix this would be to add proper NULL checks in the probe method of sst-mfld-platform and/or sst_enable_ssp(). Maybe this should be done additionally, but at least in my opinion there is not much point in registering the machine devices if they end up being broken anyway. sound/soc/intel/atom/sst/sst_acpi.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index ac542535b9d5..493d32923815 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -345,6 +345,18 @@ static int sst_acpi_probe(struct platform_device *pdev) mach->mach_params.acpi_ipc_irq_index = pdata->res_info->acpi_ipc_irq_index; + /* Fill sst platform data */ + ctx->pdata = pdata; + strcpy(ctx->firmware_name, mach->fw_filename); + + ret = sst_platform_get_resources(ctx); + if (ret) + return ret; + + ret = sst_context_init(ctx); + if (ret < 0) + return ret; + plat_dev = platform_device_register_data(dev, pdata->platform, -1, NULL, 0); if (IS_ERR(plat_dev)) { @@ -365,18 +377,6 @@ static int sst_acpi_probe(struct platform_device *pdev) return PTR_ERR(mdev); } - /* Fill sst platform data */ - ctx->pdata = pdata; - strcpy(ctx->firmware_name, mach->fw_filename); - - ret = sst_platform_get_resources(ctx); - if (ret) - return ret; - - ret = sst_context_init(ctx); - if (ret < 0) - return ret; - sst_configure_runtime_pm(ctx); platform_set_drvdata(pdev, ctx); return ret; -- 2.20.0

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[alsa-devel] Pioneer DDJ-SX3 usb audio quirks
by ard 10 Jan '19

10 Jan '19

Hi Guys, I am trying to get my pioneer to work. Everything works except for the PCM, which seems to be nothing more than an isosynchronous stream of 12 channels S32_LE output and 10 channels of S32_LE input. (Determined by usbpcap data patterns and mixxx channel selection). I tried to make a quirk out of it, but I got puzzled by the ifnum. The PCM channels are both endpoints of a single interface, so I wonder how I should write that down. This is the lsusb -v: Bus 001 Device 005: ID 2b73:0023 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x2b73 idProduct 0x0023 bcdDevice 1.02 iManufacturer 1 Pioneer DJ Corporation iProduct 2 PIONEER DJ DDJ-SX3 iSerial 3 201706 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 137 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 1024 bytes bInterval 1 bRefresh 0 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 9 bInCollection 1 baInterfaceNr( 0) 2 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 0 MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 65 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 3 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 3 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 2 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 2 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x5f70 (Bus Powered) Debug Mode And here is my preliminary quirk (yes, I need to remove the octo capture ;-) ) commit 55a009f7e77acf6d75a297e0a01839fdeb580e65 Author: Ard van Breemen <ard(a)kwaak.net> Date: Wed Jan 9 15:36:46 2019 +0100 try 1 SX3 diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 15cbe2565703..8493dbe536bd 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3324,5 +3324,60 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * PIONEER DJ DDJ-SX3 + * PCM is 12 channels out, 10 channels in @ 44.1 fixed + * interface 0, vendor class alt setting 1 for endpoints 5 and 0x86 + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0023), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "", */ + /* .product_name = "OCTO-CAPTURE", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x05, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = & (const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 10, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x86, + .ep_attr = 0x05, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC ---- Since this is a copy paste of the roland octo capture with fixes for the numbers I know, I am still wondering about this: Should I use .ifnum=1 for every snd_usb_audio_quirk? And in eacht audioformat should I use .iface = 0? I also wonder what the .altset_idx means. Regards, Ard van Breemen -- .signature not found

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[alsa-devel] [PATCH] ASoC: soc-core: Hold client_mutex around soc_init_dai_link()
by Rohit kumar 10 Jan '19

10 Jan '19

soc_init_dai_link() calls soc_find_component() which needs to be within client_mutex lock. Add client_mutex lock around soc_init_dai_link() in snd_soc_register_card() to avoid lockdep warning. Fixes: 8780cf1142a5 ("ASoC: soc-core: defer card probe until all component is added to list") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Rohit kumar <rohitkr(a)codeaurora.org> Signed-off-by: Ajit Pandey <ajitp(a)codeaurora.org> --- sound/soc/soc-core.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index eec92f1..0934b36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1027,6 +1027,7 @@ static int snd_soc_init_platform(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_dai_link_component *platform = dai_link->platform; + /* * FIXME * @@ -2754,15 +2755,18 @@ int snd_soc_register_card(struct snd_soc_card *card) if (!card->name || !card->dev) return -EINVAL; + mutex_lock(&client_mutex); for_each_card_prelinks(card, i, link) { ret = soc_init_dai_link(card, link); if (ret) { dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); + mutex_unlock(&client_mutex); return ret; } } + mutex_unlock(&client_mutex); dev_set_drvdata(card->dev, card); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[alsa-devel] [PATCH -next] ASoC: Intel: bytcht_es8316: Set correct platform drvdata in snd_byt_cht_es8316_mc_probe()
by Wei Yongjun 10 Jan '19

10 Jan '19

The snd_byt_cht_es8316_mc_remove() use the platform drvdata as a type of 'struct byt_cht_es8316_private', but snd_byt_cht_es8316_mc_probe() set it to 'struct snd_soc_card', fix to pass correct platform drvdata to platform_set_drvdata(). Fixes: 0d3e91da0750 ("ASoC: Intel: bytcht_es8316: Add external speaker mux support") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/intel/boards/bytcht_es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index cdf2061..42111d8 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -538,7 +538,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) dev_err(dev, "snd_soc_register_card failed: %d\n", ret); return ret; } - platform_set_drvdata(pdev, &byt_cht_es8316_card); + platform_set_drvdata(pdev, priv); return 0; }

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[alsa-devel] only 1.00 dB of playback volume change
by rm.riches＠jacob21819.net 10 Jan '19

10 Jan '19

(I sent this first to alsa-user and was instructed to send it to alsa-devel, as well.) With a new OS installation on a pre-existing machine, some of the USB audio devices now have only 1.00 dB of playback volume adjustment, which of course is not enough to be terribly useful. The affected USB devices show up as "Audio Advantage MicroII" in the output of "aplay -L". From LSB, they show up as these: Bus 001 Device 004: ID 0d8c:0103 C-Media Electronics, Inc. CM102-A+/102S+ Audio Controller Bus 001 Device 003: ID 0d8c:0103 C-Media Electronics, Inc. CM102-A+/102S+ Audio Controller I just installed Devuan Ascii on the machine. It's running kernel 4.9.0-8-amd64 and alsa-utils version 1.1.3-1. Doing "amixer -c 1 contents" or "amixer -c 2 contents" shows this for playback volume: numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=0,0 | dBminmax-min=-1.00dB,max=0.00dB and numid=3,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=255,step=0 : values=77,77 | dBminmax-min=-1.00dB,max=0.00dB It looks like I'm running kernel 4.9.0-8-amd64 and alsa-utils version 1.1.3-1. Using alsamixer to run the controls from end to end, I cannot perceive any difference in volume level. In contrast, the other device I use (either HDA Intel or HDA NVidia) has around a 60dB range according to amixer, and running the control in alsamixer from end to end has a huge range of gain. Previously, the exact same hardware was running Debian 7/Wheezy and then Slackware 14.2. With both those operating systems, the USB sound devices had a very adequate volume control range, certainly a whole lot more than 1dB. Being as the USB sound devices are the ones my wife uses, I'd be extremely grateful if somebody could point me a to way (if one exists) to a usefully greater range of volume control. Any suggestions? Assuming this qualifies as a bug, what channel should I use to report it so it can be queued to be fixed? Thanks, Robert

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[alsa-devel] [PATCH v2 0/2] ALSA: HDAudio: enable dynamic selection between legacy and Skylake drivers
by Pierre-Louis Bossart 09 Jan '19

09 Jan '19

Use detection logic based on the PCI class/subclass/prog-if in the HDaudio legacy probe, and add a module parameter to bypass or force the automatic detection should it ever fail (or in case of missing firmware or topology files) This patchset was tested on 1. an SKL HP device with the DSP disabled 2. a WHL Acer Swift3 laptop 3. APL NUC6 4. KBL NUC7 5 reboot tests were done for each device with the DSP presence detected reliably. The "pci_binding" parameter was also tested on the WHL device. The dynamic selection adds about 5ms worst case. [ 2.813741] snd_hda_intel 0000:00:1f.3: The DSP is enabled on this platform, aborting probe [ 2.850588] snd_soc_skl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 The SOF driver will use exactly the same mechanism, and add a mutual exclusion with the Skylake/SST driver to avoid having 3 drivers for the same PCI ID. Two is complicated enough. Changes since v1 (Feedback from Takashi) a) added explanations and renamed Kconfigs to SND_HDA_INTEL_DSP_DETECTION_XYZ b) less verbose dev_info c) fixed pci_binding parameter Pierre-Louis Bossart (2): ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection include/sound/hdaudio.h | 6 ++++ sound/pci/hda/Kconfig | 62 ++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 54 +++++++++++++++++++++++++---- sound/soc/intel/Kconfig | 6 ++++ sound/soc/intel/skylake/skl.c | 48 ++++++++++++++++++-------- 6 files changed, 155 insertions(+), 23 deletions(-) base-commit: ba26f635bf7253b42b08135152b18a2c8082abb7 -- 2.17.1

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[alsa-devel] ASoC: Intel: sst: Missing IRQ at index 5 on BYT-T device
by Stephan Gerhold 09 Jan '19

09 Jan '19

Hi, I have an Intel Bay Trail (BYT-T) tablet that was originally shipped with Android. With the right quirks, bytcr-rt5640 is working fine, but there is a problem in sst_acpi.c that is preventing it from working with a mainline kernel: Even though this is a BYT-T device, there is no IRQ at index 5 in the ACPI DSDT table. This means that SST will fail to probe, and actually leads to a NULL pointer dereference later when the ALSA device is first opened. (I have submitted a possible solution for this as "[PATCH] ASoC: Intel: sst: Delay machine device creation until after initialization") The correct IRQ is actually located on index 0, just like it is already being used for BYT-CR devices. So if I force is_byt_cr() to return TRUE, everything works as expected. Here is the relevant part from the ACPI DSDT table: Name (_ADR, Zero) // _ADR: Address Name (_HID, "80860F28" /* Intel SST Audio DSP */) // _HID: Hardware ID Name (_CID, "80860F28" /* Intel SST Audio DSP */) // _CID: Compatible ID Name (_DDN, "Intel(R) Low Power Audio Controller - 80860F28") // _DDN: DOS Device Name Name (_SUB, "80867270") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Name (RBUF, ResourceTemplate () { Memory32Fixed (ReadWrite, 0x12345678, // Address Base 0x00200000, // Address Length _Y08) Memory32Fixed (ReadWrite, 0xFE830000, // Address Base 0x00001000, // Address Length _Y09) Memory32Fixed (ReadWrite, 0x55AA55AA, // Address Base 0x00200000, // Address Length _Y0A) Interrupt (ResourceConsumer, Level, ActiveLow, Exclusive, ,, ) { 0x0000001D, } }) Unlike many of the other DSDT dumps I've looked at, there is only one interrupt listed. Full ACPI DSDT table is at [1]. Since there is no IRQ at index 5, platform_get_irq will return -ENXIO. Couldn't we fall back to index 0 in this case? I would say that if the seemingly "correct" IRQ at index 5 does not even exist, we still have a better chance of picking the right one if we try the one at index 0. Or we could check the number of interrupts that are actually available. The other option would be some kind of DMI-based quirk, but personally I would prefer to avoid that.. (In my opinion, there is way too much device specific code with the quirks etc already...) Or do you have any other suggestions? Thanks, Stephan [1]: https://github.com/me176c-dev/me176c-acpi/blob/f48c78c11b0819b899f988407b6e…

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[alsa-devel] Applied "ASoC: tlv320aic32x4: SND_SOC_DAPM_MICBIAS is deprecated" to the asoc tree
by Mark Brown 09 Jan '19

09 Jan '19

The patch ASoC: tlv320aic32x4: SND_SOC_DAPM_MICBIAS is deprecated has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 04d979d7a7bac2f645cd827ea37e5ffa5b4e1f97 Mon Sep 17 00:00:00 2001 From: b-ak <anur.bhargav(a)gmail.com> Date: Wed, 9 Jan 2019 22:41:21 +0530 Subject: [PATCH] ASoC: tlv320aic32x4: SND_SOC_DAPM_MICBIAS is deprecated SND_SOC_DAPM_MICBIAS is deprecated, replace it with SND_SOC_DAPM_SUPPLY. MICBIAS voltage wasn't supplied to the microphone with the older SND_SOC_DAPM_MICBIAS widget, hence the microphone wouldn't work. This patch fixes the problem. Signed-off-by: b-ak <anur.bhargav(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 30 +++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic32x4.h | 1 + 2 files changed, 30 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index e2b5a11b16d1..521663d8b585 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -79,6 +79,32 @@ struct aic32x4_priv { struct device *dev; }; +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Change Mic Bias Registor */ + snd_soc_component_update_bits(component, AIC32X4_MICBIAS, + AIC32x4_MICBIAS_MASK, + AIC32X4_MICBIAS_LDOIN | + AIC32X4_MICBIAS_2075V); + printk(KERN_DEBUG "%s: Mic Bias will be turned ON\n", __func__); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_component_update_bits(component, AIC32X4_MICBIAS, + AIC32x4_MICBIAS_MASK, 0); + printk(KERN_DEBUG "%s: Mic Bias will be turned OFF\n", + __func__); + break; + } + + return 0; +} + + static int aic32x4_get_mfp1_gpio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -450,7 +476,9 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, in3r_to_lmixer_controls), - SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", AIC32X4_MICBIAS, 6, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("HPL"), SND_SOC_DAPM_OUTPUT("HPR"), diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h index e9df49edbf19..c2d74025bf4b 100644 --- a/sound/soc/codecs/tlv320aic32x4.h +++ b/sound/soc/codecs/tlv320aic32x4.h @@ -195,6 +195,7 @@ int aic32x4_remove(struct device *dev); /* AIC32X4_MICBIAS */ #define AIC32X4_MICBIAS_LDOIN BIT(3) #define AIC32X4_MICBIAS_2075V 0x60 +#define AIC32x4_MICBIAS_MASK GENMASK(6, 3) /* AIC32X4_LMICPGANIN */ #define AIC32X4_LMICPGANIN_IN2R_10K 0x10 -- 2.20.1

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[alsa-devel] [PATCH] ASoC: Intel: Boards: move the codec PLL configuration to _init
by mac.chiang＠intel.com 09 Jan '19

09 Jan '19

From: Mac Chiang <mac.chiang(a)intel.com> move the codec PLL to rt5682_codec_init, because codec only need to config the clock source/PLL once. As the result, remove the platform_clock_controls since no need to control clock anymore. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- sound/soc/intel/boards/glk_rt5682_max98357a.c | 45 +++++---------------------- 1 file changed, 7 insertions(+), 38 deletions(-) diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index c74c4f1..8f83b18 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -55,39 +55,6 @@ enum { GLK_DPCM_AUDIO_HDMI3_PB, }; -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret = 0; - - codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_OFF(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, 0); - if (ret) - dev_err(card->dev, "failed to stop sysclk: %d\n", ret); - } else if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, - GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); - if (ret < 0) { - dev_err(card->dev, "can't set codec pll: %d\n", ret); - return ret; - } - } - - if (ret) - dev_err(card->dev, "failed to start internal clk: %d\n", ret); - - return ret; -} - static const struct snd_kcontrol_new geminilake_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone Jack"), SOC_DAPM_PIN_SWITCH("Headset Mic"), @@ -102,14 +69,10 @@ static const struct snd_soc_dapm_widget geminilake_widgets[] = { SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route geminilake_map[] = { /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -117,7 +80,6 @@ static const struct snd_soc_dapm_route geminilake_map[] = { { "Spk", NULL, "Speaker" }, /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, /* digital mics */ @@ -177,6 +139,13 @@ static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_jack *jack; int ret; + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + /* Configure sysclk for codec */ ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); -- 2.7.4

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