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

[alsa-devel] [PATCH] ASoC: rsnd: core.c: add WARN_ON() on rsnd_channel_normalization()
by Kuninori Morimoto 28 Oct '19

28 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> chan > 8 or chan < 0 shouldn't happen. This patch uses WARN_ON() for such case. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e9242a0..295faad 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -302,7 +302,7 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, int rsnd_channel_normalization(int chan) { - if ((chan > 8) || (chan < 0)) + if (WARN_ON((chan > 8) || (chan < 0))) return 0; /* TDM Extend Mode needs 8ch */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: pci: Fix memory leak in snd_korg1212_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_korg1212_create() the allocated memory for korg1212 is leaked in cases of error. Release korg1212 via snd_korg1212_free() if either of these calls fail: snd_korg1212_downloadDSPCode(), snd_pcm_new(), or snd_ctl_add(). Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/korg1212/korg1212.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 0d81eac0a478..e976e857d915 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2367,8 +2367,10 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, mdelay(CARD_BOOT_DELAY_IN_MS); - if (snd_korg1212_downloadDSPCode(korg1212)) + if (snd_korg1212_downloadDSPCode(korg1212)) { + snd_korg1212_free(korg1212); return -EBUSY; + } K1212_DEBUG_PRINTK("korg1212: dspMemPhy = %08x U[%08x], " "PlayDataPhy = %08x L[%08x]\n" @@ -2383,8 +2385,11 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, korg1212->RoutingTablePhy, LowerWordSwap(korg1212->RoutingTablePhy), korg1212->AdatTimeCodePhy, LowerWordSwap(korg1212->AdatTimeCodePhy)); - if ((err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm)) < 0) + err = snd_pcm_new(korg1212->card, "korg1212", 0, 1, 1, &korg1212->pcm); + if (err < 0) { + snd_korg1212_free(korg1212); return err; + } korg1212->pcm->private_data = korg1212; korg1212->pcm->private_free = snd_korg1212_free_pcm; @@ -2398,8 +2403,10 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci, for (i = 0; i < ARRAY_SIZE(snd_korg1212_controls); i++) { err = snd_ctl_add(korg1212->card, snd_ctl_new1(&snd_korg1212_controls[i], korg1212)); - if (err < 0) + if (err < 0) { + snd_korg1212_free(korg1212); return err; + } } snd_korg1212_proc_init(korg1212); -- 2.17.1

3 7

[alsa-devel] KASAN: slab-out-of-bounds Read in build_audio_procunit
by syzbot 28 Oct '19

28 Oct '19

Hello, syzbot found the following crash on: HEAD commit: 22be26f7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=103893cf600000 kernel config: https://syzkaller.appspot.com/x/.config?x=387eccb7ac68ec5 dashboard link: https://syzkaller.appspot.com/bug?extid=0620f79a1978b1133fd7 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=16fd6a40e00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1033215f600000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+0620f79a1978b1133fd7(a)syzkaller.appspotmail.com usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 usb 1-1: Product: syz usb 1-1: Manufacturer: syz usb 1-1: SerialNumber: syz ================================================================== BUG: KASAN: slab-out-of-bounds in uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] BUG: KASAN: slab-out-of-bounds in build_audio_procunit+0x1306/0x13f0 sound/usb/mixer.c:2436 Read of size 1 at addr ffff8881d537c7b7 by task kworker/0:2/101 CPU: 0 PID: 101 Comm: kworker/0:2 Not tainted 5.4.0-rc3+ #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 print_address_description.constprop.0+0x36/0x50 mm/kasan/report.c:374 __kasan_report.cold+0x1a/0x33 mm/kasan/report.c:506 kasan_report+0xe/0x20 mm/kasan/common.c:634 uac_processing_unit_iProcessing include/uapi/linux/usb/audio.h:429 [inline] build_audio_procunit+0x1306/0x13f0 sound/usb/mixer.c:2436 parse_audio_processing_unit sound/usb/mixer.c:2468 [inline] parse_audio_unit+0x17e9/0x36f0 sound/usb/mixer.c:2757 snd_usb_mixer_controls+0x715/0xb90 sound/usb/mixer.c:3095 snd_usb_create_mixer+0x2b5/0x1890 sound/usb/mixer.c:3445 usb_audio_probe+0xc76/0x2010 sound/usb/card.c:653 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_set_configuration+0xdf6/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0x99/0x100 drivers/usb/core/driver.c:266 really_probe+0x281/0x6d0 drivers/base/dd.c:548 driver_probe_device+0x104/0x210 drivers/base/dd.c:721 __device_attach_driver+0x1c2/0x220 drivers/base/dd.c:828 bus_for_each_drv+0x162/0x1e0 drivers/base/bus.c:430 __device_attach+0x217/0x360 drivers/base/dd.c:894 bus_probe_device+0x1e4/0x290 drivers/base/bus.c:490 device_add+0xae6/0x16f0 drivers/base/core.c:2201 usb_new_device.cold+0x6a4/0xe79 drivers/usb/core/hub.c:2536 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Allocated by task 101: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] __kasan_kmalloc mm/kasan/common.c:510 [inline] __kasan_kmalloc.constprop.0+0xbf/0xd0 mm/kasan/common.c:483 kmalloc include/linux/slab.h:561 [inline] usb_get_configuration+0x314/0x3050 drivers/usb/core/config.c:857 usb_enumerate_device drivers/usb/core/hub.c:2369 [inline] usb_new_device+0xd3/0x160 drivers/usb/core/hub.c:2505 hub_port_connect drivers/usb/core/hub.c:5183 [inline] hub_port_connect_change drivers/usb/core/hub.c:5323 [inline] port_event drivers/usb/core/hub.c:5469 [inline] hub_event+0x1dd0/0x37e0 drivers/usb/core/hub.c:5551 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 Freed by task 17: save_stack+0x1b/0x80 mm/kasan/common.c:69 set_track mm/kasan/common.c:77 [inline] kasan_set_free_info mm/kasan/common.c:332 [inline] __kasan_slab_free+0x130/0x180 mm/kasan/common.c:471 slab_free_hook mm/slub.c:1424 [inline] slab_free_freelist_hook mm/slub.c:1475 [inline] slab_free mm/slub.c:3018 [inline] kfree+0xe4/0x320 mm/slub.c:3967 urb_destroy drivers/usb/core/urb.c:26 [inline] kref_put include/linux/kref.h:65 [inline] usb_free_urb.part.0+0x7a/0xc0 drivers/usb/core/urb.c:95 usb_free_urb+0x1b/0x30 drivers/usb/core/urb.c:94 usb_start_wait_urb+0x1e5/0x2b0 drivers/usb/core/message.c:79 usb_internal_control_msg drivers/usb/core/message.c:101 [inline] usb_control_msg+0x31c/0x4a0 drivers/usb/core/message.c:152 get_port_status drivers/usb/core/hub.c:569 [inline] hub_ext_port_status+0x125/0x460 drivers/usb/core/hub.c:586 hub_port_status drivers/usb/core/hub.c:608 [inline] hub_activate+0x497/0x1570 drivers/usb/core/hub.c:1111 process_one_work+0x92b/0x1530 kernel/workqueue.c:2269 worker_thread+0x96/0xe20 kernel/workqueue.c:2415 kthread+0x318/0x420 kernel/kthread.c:255 ret_from_fork+0x24/0x30 arch/x86/entry/entry_64.S:352 The buggy address belongs to the object at ffff8881d537c700 which belongs to the cache kmalloc-192 of size 192 The buggy address is located 183 bytes inside of 192-byte region [ffff8881d537c700, ffff8881d537c7c0) The buggy address belongs to the page: page:ffffea000754df00 refcount:1 mapcount:0 mapping:ffff8881da002a00 index:0x0 flags: 0x200000000000200(slab) raw: 0200000000000200 ffffea000754d680 0000000600000006 ffff8881da002a00 raw: 0000000000000000 0000000000100010 00000001ffffffff 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8881d537c680: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ffff8881d537c700: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > ffff8881d537c780: 00 00 00 02 fc fc fc fc fc fc fc fc fc fc fc fc ^ ffff8881d537c800: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8881d537c880: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

3 3

[alsa-devel] [PATCH] ASoC: change 'HDMI/DP, pcm=' to 'HDMI/DP, pcm=' Jack control names
by Jaroslav Kysela 28 Oct '19

28 Oct '19

There is an inconsistency in the names for the HDMI/DP Jack control names between some ASoC drivers and the HDA HDMI driver which introduced this naming in 2011. There might be an impact for the user space (UCM). I will fix the UCM configurations when this patch is applied. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- Documentation/sound/designs/control-names.rst | 5 +++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 2 +- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- 13 files changed, 15 insertions(+), 14 deletions(-) diff --git a/Documentation/sound/designs/control-names.rst b/Documentation/sound/designs/control-names.rst index 7fedd0f33cd9..a5f6e3842df8 100644 --- a/Documentation/sound/designs/control-names.rst +++ b/Documentation/sound/designs/control-names.rst @@ -50,7 +50,7 @@ Internal internal SOURCE ~~~~~~ -=================== ================================================= +=================== =================================================== Master Master Mono Hardware Master @@ -92,7 +92,8 @@ SPDIF output only SPDIF In Digital In HDMI/DP either HDMI or DisplayPort -=================== ================================================= +HDMI/DP,pcm=<num> either HDMI or DisplayPort with the PCM device link +=================== =================================================== Exceptions (deprecated) ----------------------- diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index ac1dea5f9d11..b49028c6fd10 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -618,7 +618,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &broxton_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index adf416a49b48..caaf2f2af287 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -530,7 +530,7 @@ static int bxt_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &broxton_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index bd2d371f2acd..5474f28e0c31 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -548,7 +548,7 @@ static int glk_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &geminilake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb6..367415efed9b 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -548,7 +548,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 829f95fc4179..b30c2148d1f4 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -977,7 +977,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &kabylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 74fe1f3a5479..0965d1f02b2c 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -470,7 +470,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 7cefda341fbf..9f420e978459 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -888,7 +888,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index 58409b6e476e..f717b1d179bd 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -139,7 +139,7 @@ int skl_hda_hdmi_jack_init(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &pcm->hdmi_jack, NULL, 0); diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 3ce8efbeed12..b009e2df25d3 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -607,7 +607,7 @@ static int skylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 1a7ac8bdf543..0a6e650dc698 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -648,7 +648,7 @@ static int skylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 231349a47cc9..56ee4b55dce9 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -489,7 +489,7 @@ static int skylake_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &skylake_hdmi[i], NULL, 0); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4f6e58c3954a..862a7e35cb5c 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -277,7 +277,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); + "HDMI/DP,pcm=%d Jack", pcm->device); err = snd_soc_card_jack_new(card, jack_name, SND_JACK_AVOUT, &sof_hdmi[i], NULL, 0); -- 2.20.1

6 20

[alsa-devel] [PATCH] ALSA: hda: Allow non-Intel device probe gracefully
by Takashi Iwai 28 Oct '19

28 Oct '19

The recent addition of snd_intel_dsp_driver_probe() check caused a spurious kernel warning when the driver is loaded for a non-Intel hardware due to snd_BUG_ON(). Moreover, for such a hardware, we should always return SND_INTEL_DSP_DRIVER_ANY, not check the dsp_driver option at all. This patch fixes these issues for non-Intel devices. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/intel-dsp-config.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 0b2a7201a46a..be1df80ed013 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -307,13 +307,13 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) { const struct config_entry *cfg; - if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) - return dsp_driver; - /* Intel vendor only */ - if (snd_BUG_ON(pci->vendor != 0x8086)) + if (pci->vendor != 0x8086) return SND_INTEL_DSP_DRIVER_ANY; + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + /* * detect DSP by checking class/subclass/prog-id information * class=04 subclass 03 prog-if 00: no DSP, use legacy driver -- 2.16.4

1 0

[alsa-devel] Questions about max98090 codec driver
by Tzung-Bi Shih 28 Oct '19

28 Oct '19

Hi, I am studying an odd issue of max98090 codec on Baytrail-based chromebook. The issue is: when user playback and capture simultaneously, it seems the PLL never get locked if msleep(10) between the SHDN off and on (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…). The playback stream becomes silent and the console keeps printing "PLL unlocked". But, if comment out the msleep(10) between the SHDN off and on, the issue fixed. I am trying to find the reason but facing further more questions and may need your inputs. 1. The commit b8a3ee820f7b ("ASoC: max98090: Add recovery for PLL lock failure") enables ULK interrupts (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…) when PCM stream starting. If max98090 claims its PLL is unlocked, max98090_pll_work() get scheduled to workaround it by SHDN off and on (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…). I feel it is weird to sleep in max98090_pll_work(). Especially, at this line https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.… (it makes less sense to "wait" in another thread). Note that, the threaded IRQF_ONESHOT handler and max98090_pll_work() are in 2 different threads. I guess the original intention is: - disable ULK interrupt in IRQ handler (https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…) - schedule max98090_pll_work() to workaround it - wait 10ms to give PLL chance to lock - enable ULK interrupt again If max98090 claims its PLL is unlocked again, repeat the above by receiving another ULK interrupt. Unfortunately, the odd issue seems not be fixed by my rough implementation of these. 2. According to the datasheet page 164 table 90 (https://datasheets.maximintegrated.com/en/ds/MAX98090.pdf), there are some registers should only be adjusted when SHDN==0. But I fail to find max98090.c tries to set SHDN to 0 and restore it afterwards when writing to these registers. For example, https://elixir.bootlin.com/linux/v5.4-rc2/source/sound/soc/codecs/max98090.…. I am wondering if it would bring any side effects because the datasheet states "Changing these settings during normal operation (SHDN=1) can compromise device stability and performance specifications." 3. By searching some history data, I found a previous version did not have the msleep(10) between the SHDN off and on (https://crrev.com/c/191740, click the file name in the middle of the window to see the diff. Pardon me, I do not find another public repository for this). I am curious if anyone of you still remember why the upstream version contains the msleep(10). I am also curious if anyone of your environment works well with the upstream version max98090.c.

4 5

Re: [alsa-devel] [PATCH V2] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by S.j. Wang 28 Oct '19

28 Oct '19

Hi > > On Fri, Oct 25, 2019 at 03:13:53PM +0800, Shengjiu Wang wrote: > > xrun may happen at the end of stream, the > > trigger->fsl_esai_trigger_stop maybe called in the middle of > > fsl_esai_hw_reset, this may cause esai in wrong state after stop, and > > there may be endless xrun interrupt. > > > > This issue may also happen with trigger->fsl_esai_trigger_start. > > > > So Add spin lock to lock those functions. > > > > Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after > > xrun") > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > Some small comments inline. Once they are addressed, please add: > > Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> > > Thanks > > > --- > > Change in v2 > > -add lock for fsl_esai_trigger_start. > > > > sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ > > 1 file changed, 12 insertions(+) > > > > diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index > > 37b14c48b537..9b28e2af26e4 100644 > > --- a/sound/soc/fsl/fsl_esai.c > > +++ b/sound/soc/fsl/fsl_esai.c > > @@ -33,6 +33,7 @@ > > * @fsysclk: system clock source to derive HCK, SCK and FS > > * @spbaclk: SPBA clock (optional, depending on SoC design) > > * @task: tasklet to handle the reset operation > > + * @lock: spin lock to handle reset and stop behavior > > Should be "between hw_reset() and trigger()" now. > > > * @fifo_depth: depth of tx/rx FIFO > > * @slot_width: width of each DAI slot > > * @slots: number of slots > > @@ -56,6 +57,7 @@ struct fsl_esai { > > struct clk *fsysclk; > > struct clk *spbaclk; > > struct tasklet_struct task; > > + spinlock_t lock; /* Protect reset and stop */ > > We can drop the comments here since you add it to the top. Here is required by the checkpatch.pl, so still is needed. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ALSA: pci: Fix memory leak in snd_ali_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_ali_create() the memory allocated for codec is leaked in case of an error. Release codec if snd_ali_chip_init() fails. Fixes: f9ab2b1c3ab5 ("[ALSA] ali5451 - Code clean up, irq handler fix") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/ali5451/ali5451.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 6e28e381c21a..3ed07a18be4d 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2179,6 +2179,7 @@ static int snd_ali_create(struct snd_card *card, err = snd_ali_chip_init(codec); if (err < 0) { dev_err(card->dev, "ali create: chip init error.\n"); + snd_ali_free(codec); return err; } -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: lx6464es: Fix memory leaks in snd_lx6464es_create
by Navid Emamdoost 28 Oct '19

28 Oct '19

In the implementation of snd_lx6464es_create() it there are memory leaks when error happens. Go to error path if any of these calls fail: lx_init_dsp(), lx_pcm_create(), lx_proc_create(), snd_ctl_add(). Fixes: 02bec4904508 ("ALSA: lx6464es - driver for the digigram lx6464es interface") Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> --- sound/pci/lx6464es/lx6464es.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index fe10714380f2..7c6e8f4ef826 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -1020,25 +1020,26 @@ static int snd_lx6464es_create(struct snd_card *card, err = lx_init_dsp(chip); if (err < 0) { dev_err(card->dev, "error during DSP initialization\n"); - return err; + goto cleanup; } err = lx_pcm_create(chip); if (err < 0) - return err; + goto cleanup; err = lx_proc_create(card, chip); if (err < 0) - return err; + goto cleanup; err = snd_ctl_add(card, snd_ctl_new1(&lx_control_playback_switch, chip)); if (err < 0) - return err; + goto cleanup; *rchip = chip; return 0; +cleanup: device_new_failed: free_irq(pci->irq, chip); -- 2.17.1

3 2

[alsa-devel] [PATCH -next] ASoC: SOF: select SND_INTEL_DSP_CONFIG in SND_SOC_SOF_PCI
by Mao Wenan 28 Oct '19

28 Oct '19

When SND_SOC_SOF_PCI=y, and SND_INTEL_DSP_CONFIG=m, below errors can be seen: sound/soc/sof/sof-pci-dev.o: In function `sof_pci_probe': sof-pci-dev.c:(.text+0xb9): undefined reference to `snd_intel_dsp_driver_probe' After commit 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code"), sof_pci_probe() will call snd_intel_dsp_driver_probe(), so it should select SND_INTEL_DSP_CONFIG in Kconfig SND_SOC_SOF_PCI. Fixes: 82d9d54a6c0e ("ALSA: hda: add Intel DSP configuration / probe code") Signed-off-by: Mao Wenan <maowenan(a)huawei.com> --- sound/soc/sof/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 56a3ab6..a9b2be2 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -16,6 +16,7 @@ config SND_SOC_SOF_PCI select SND_SOC_ACPI if ACPI select SND_SOC_SOF_OPTIONS select SND_SOC_SOF_INTEL_PCI if SND_SOC_SOF_INTEL_TOPLEVEL + select SND_INTEL_DSP_CONFIG help This adds support for PCI enumeration. This option is required to enable Intel Skylake+ devices -- 2.7.4

2 1

[alsa-devel] [PATCH] Fix JACK plugin buffer sizes
by John McKay 27 Oct '19

27 Oct '19

I was having a conversation on the Archlinux forums[1] about using the alsa-plugins JACK plugin. While one of the bugs that we found has already been fixed in the git alsa-lib, there is another bug relating to buffer sizes. Currently snd_pcm_hw_params_set_buffer_size_near() doesn't really work when you are using this plugin unless you happen to request a buffer size that almost exactly matches a whole multiple of periods that the JACK is asking for. After some debugging, I discovered the reason is that some plugins, notably the 'plug' plugin that most everybody is going to use to change frequency/format directly attempt to manipulate the buffer size. This causes the code in pcm_ioplug.c to fail because it cannot find a number of periods/period time that can fit the buffer. The included patch stops this by specifying the list of acceptable buffer sizes as well, so that when other plugins try to change the buffer it always picks one that will work. It does have some problems in the JACK plugin now offers a number of periods/sizes that don't exist but in practice asking for huge buffers just seems to give you a more sane value. [1] https://bbs.archlinux.org/viewtopic.php?id=250116 --- jack/pcm_jack.c 2019-05-10 06:57:24.000000000 +0000 +++ jack/pcm_jack.c 2019-10-27 03:43:36.053734268 +0000 @@ -428,6 +428,7 @@ unsigned int format = SND_PCM_FORMAT_FLOAT; unsigned int rate = jack_get_sample_rate(jack->client); unsigned int psize_list[MAX_PERIODS_MULTIPLE]; + unsigned int bsize_list[MAX_PERIODS_MULTIPLE]; unsigned int nframes = jack_get_buffer_size(jack->client); unsigned int jack_buffer_bytes = (snd_pcm_format_size(format, nframes) * jack->num_ports); @@ -441,6 +442,9 @@ for (i = 1; i <= ARRAY_SIZE(psize_list); i++) psize_list[i-1] = jack_buffer_bytes * i; + for (i = 1; i <= ARRAY_SIZE(bsize_list); i++) + bsize_list[i-1] = jack_buffer_bytes * (i + 1); + jack->sample_bits = snd_pcm_format_physical_width(format); if ((err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_ACCESS, ARRAY_SIZE(access_list), access_list)) < 0 || @@ -452,6 +456,8 @@ rate, rate)) < 0 || (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_PERIOD_BYTES, ARRAY_SIZE(psize_list), psize_list)) < 0 || + (err = snd_pcm_ioplug_set_param_list(&jack->io, SND_PCM_IOPLUG_HW_BUFFER_BYTES, + ARRAY_SIZE(bsize_list), bsize_list)) < 0 || (err = snd_pcm_ioplug_set_param_minmax(&jack->io, SND_PCM_IOPLUG_HW_PERIODS, 2, 64)) < 0) return err;

1 0

[alsa-devel] [PATCH] ALSA: bebob: Fix prototype of helper function to return negative value
by Takashi Sakamoto 26 Oct '19

26 Oct '19

A helper function of ALSA bebob driver returns negative value in a function which has a prototype to return unsigned value. This commit fixes it by changing the prototype. Fixes: eb7b3a056cd8 ("ALSA: bebob: Add commands and connections/streams management" Cc: <stable(a)vger.kernel.org> # v3.16+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_stream.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 7ac0d9f495c4..f7f0db5aa811 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -252,8 +252,7 @@ int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob, return err; } -static unsigned int -map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) +static int map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s) { unsigned int sec, sections, ch, channels; unsigned int pcm, midi, location; -- 2.20.1

2 1

[alsa-devel] [PATCH v2 00/12] Add Samus Hotwording for RT5677
by Curtis Malainey 26 Oct '19

26 Oct '19

This patch series adds the hotwording implementation used in the Pixelbook on the RT5677 driver. Known Issues: There is a known issue where the system will fail to detect a hotword if suspended while the stream is open. This is due to the fact that the haswell-dsp suspends its I2S MCLK before the RT5677 suspends which causes the writes and reads to become corrupted as a result. Any recommendations to correct this behaviour would be appreciated. Changes since v1: * Addressed Mark's and Kuninori's comments * Adjusted DAI Links * Squashed related patches * Removed all kcontrol writes * Addressed race condition with dapm * Added firmware boot check * Moved all clocking to driver API * Moved DAPM routes that spanned drivers to machine driver Cc: Dylan Reid <dgreid(a)chromium.org> Cc: Ben Zhang <benzh(a)chromium.org> Ben Zhang (8): ASoC: rt5677: Add a PCM device for streaming hotword via SPI ASoC: rt5677: Load firmware via SPI using delayed work ASoC: bdw-rt5677: Add a DAI link for rt5677 SPI PCM device ASoC: rt5677: Enable jack detect while DSP is running ASoC: rt5677: Add DAPM audio path for hotword stream ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile ASoC: rt5677: Stop and restart DSP over suspend/resume ASoC: rt5677: Disable irq at suspend Curtis Malainey (4): ASoC: rt5677: Allow VAD to be shut on/off at all times ASoC: bdw-rt5677: Turn on MCLK1 for DSP via DAPM ASoC: rt5677: Set ADC clock to use PLL and enable ASRC ASoC: rt5677: Wait for DSP to boot before loading firmware sound/soc/codecs/rt5677-spi.c | 397 ++++++++++++++++++++++++ sound/soc/codecs/rt5677-spi.h | 1 + sound/soc/codecs/rt5677.c | 447 +++++++++++++++++++++++----- sound/soc/codecs/rt5677.h | 11 +- sound/soc/intel/boards/bdw-rt5677.c | 51 ++++ 5 files changed, 834 insertions(+), 73 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

4 19

[alsa-devel] [PATCH V2] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 25 Oct '19

25 Oct '19

The output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitaion in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Change in v2 - remove p2p/m2m word - use use_ideal_rate sound/soc/fsl/fsl_asrc.c | 37 +++++++++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..89cf333154c7 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -259,8 +259,11 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * It configures those ASRC registers according to a configuration instance * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. + * + * Note: + * use_ideal_rate = true is need by some case which need higher performance. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool use_ideal_rate) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +271,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,8 +355,10 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; - if (div[IN] == 0) { + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; + if (div[IN] == 0 || (!ideal && (div[IN] > 1024 || rem[IN] != 0))) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); return -EINVAL; @@ -360,18 +366,29 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) clk = asrc_priv->asrck_clk[clk_index[OUT]]; - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; + /* + * Output rate should be align with the out samplerate. If set too + * high output rate, there will be lots of Overload. + * But some case need higher performance, then we can use + * IDEAL_RATIO_RATE specifically for such case. + */ + clk_rate = clk_get_rate(clk); + if (ideal && use_ideal_rate) + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); else - div[OUT] = clk_get_rate(clk) / outrate; + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; - if (div[OUT] == 0) { + if (div[OUT] == 0 || (!ideal && (div[OUT] > 1024 || rem[OUT] != 0))) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", outrate, clk_index[OUT]); return -EINVAL; } + /* Divider range is [1, 1024] */ + div[IN] = min_t(u32, 1024, div[IN]); + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +577,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, false); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH V2] ASoC: fsl_esai: Add spin lock to protect reset, stop and start
by Shengjiu Wang 25 Oct '19

25 Oct '19

xrun may happen at the end of stream, the trigger->fsl_esai_trigger_stop maybe called in the middle of fsl_esai_hw_reset, this may cause esai in wrong state after stop, and there may be endless xrun interrupt. This issue may also happen with trigger->fsl_esai_trigger_start. So Add spin lock to lock those functions. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Change in v2 -add lock for fsl_esai_trigger_start. sound/soc/fsl/fsl_esai.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 37b14c48b537..9b28e2af26e4 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -33,6 +33,7 @@ * @fsysclk: system clock source to derive HCK, SCK and FS * @spbaclk: SPBA clock (optional, depending on SoC design) * @task: tasklet to handle the reset operation + * @lock: spin lock to handle reset and stop behavior * @fifo_depth: depth of tx/rx FIFO * @slot_width: width of each DAI slot * @slots: number of slots @@ -56,6 +57,7 @@ struct fsl_esai { struct clk *fsysclk; struct clk *spbaclk; struct tasklet_struct task; + spinlock_t lock; /* Protect reset and stop */ u32 fifo_depth; u32 slot_width; u32 slots; @@ -676,8 +678,10 @@ static void fsl_esai_hw_reset(unsigned long arg) { struct fsl_esai *esai_priv = (struct fsl_esai *)arg; bool tx = true, rx = false, enabled[2]; + unsigned long lock_flags; u32 tfcr, rfcr; + spin_lock_irqsave(&esai_priv->lock, lock_flags); /* Save the registers */ regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); @@ -715,6 +719,8 @@ static void fsl_esai_hw_reset(unsigned long arg) fsl_esai_trigger_start(esai_priv, tx); if (enabled[rx]) fsl_esai_trigger_start(esai_priv, rx); + + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); } static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -722,6 +728,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + unsigned long lock_flags; esai_priv->channels[tx] = substream->runtime->channels; @@ -729,12 +736,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_start(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&esai_priv->lock, lock_flags); fsl_esai_trigger_stop(esai_priv, tx); + spin_unlock_irqrestore(&esai_priv->lock, lock_flags); break; default: return -EINVAL; @@ -1002,6 +1013,7 @@ static int fsl_esai_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, esai_priv); + spin_lock_init(&esai_priv->lock); ret = fsl_esai_hw_init(esai_priv); if (ret) return ret; -- 2.21.0

2 1

[alsa-devel] [PATCH v2] ASoC: ti: omap-mcpdm: Add support for pdmclk clock handling
by Peter Ujfalusi 25 Oct '19

25 Oct '19

McPDM module receives it's functional clock from external source. This clock is the pdmclk provided by the twl6040 audio IC. If the clock is not available all register accesses to McPDM fails and the module is not operational. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Hi, this has been lurking in my next-wip branch for some time... It might not needed anymore as I have not heard about issues with McPDM for a while. It was dropped back then because some core parts were not picked in time and this caused some issues, but can not find the exact reason Regards, Peter sound/soc/ti/omap-mcpdm.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index b8c8290265c7..2c536947b69f 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -17,6 +17,7 @@ #include <linux/err.h> #include <linux/io.h> #include <linux/irq.h> +#include <linux/clk.h> #include <linux/slab.h> #include <linux/pm_runtime.h> #include <linux/of_device.h> @@ -42,6 +43,7 @@ struct omap_mcpdm { int irq; struct pm_qos_request pm_qos_req; int latency[2]; + struct clk *pdmclk; struct mutex mutex; @@ -416,6 +418,10 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int ret; + ret = clk_prepare_enable(mcpdm->pdmclk); + if (ret) + return ret; + pm_runtime_enable(mcpdm->dev); /* Disable lines while request is ongoing */ @@ -454,6 +460,7 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) if (pm_qos_request_active(&mcpdm->pm_qos_req)) pm_qos_remove_request(&mcpdm->pm_qos_req); + clk_disable_unprepare(mcpdm->pdmclk); return 0; } @@ -473,12 +480,19 @@ static int omap_mcpdm_suspend(struct snd_soc_dai *dai) mcpdm->pm_active_count++; } + clk_disable_unprepare(mcpdm->pdmclk); + return 0; } static int omap_mcpdm_resume(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = clk_prepare_enable(mcpdm->pdmclk); + if (ret) + return ret; if (mcpdm->pm_active_count) { while (mcpdm->pm_active_count--) @@ -573,6 +587,15 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; + mcpdm->pdmclk = devm_clk_get(&pdev->dev, "pdmclk"); + if (IS_ERR(mcpdm->pdmclk)) { + if (PTR_ERR(mcpdm->pdmclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + dev_warn(&pdev->dev, "Error getting pdmclk (%ld)!\n", + PTR_ERR(mcpdm->pdmclk)); + mcpdm->pdmclk = NULL; + } + ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, &omap_mcpdm_dai, 1); -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 2

[alsa-devel] sof-skl_hda: move hdmi as separate use case
by GitHub pull_request - opened 25 Oct '19

25 Oct '19

alsa-project/alsa-ucm-conf pull request #6 was opened from juimonen: Move hdmi definitions as separate use case and conf file. This will enable Pulseaudio to create separate profile for hdmi and make routing much more fluent than with single profile. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/6 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/6.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] sof-skl_hda: fix headphone jack name
by GitHub pull_request - opened 25 Oct '19

25 Oct '19

alsa-project/alsa-ucm-conf pull request #5 was opened from juimonen: Change "Headphone Jack" to "Headphone Mic Jack". Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/5 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/5.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [ANNOUNCE] 2019 Linux Audio miniconference
by Mark Brown 25 Oct '19

25 Oct '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue) and will be free of charge to attend, generously sponsored by Intel. Thus far for the agenda we have: - Use case management enhancements (Curtis) - DSP framework integration (Liam?) - Soundwire status (Pierre?) - Componentisation status/plans (Me) As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! If you're planning to attend please fill out the form here: https://docs.google.com/forms/d/e/1FAIpQLSd6FV-tWYZ1fHlCmyzQhG98OtYH9W9GX-1… This event will be covered by the same code of conduct as ELC-E. Thanks again to Intel for supporting this event. Thanks, Mark

8 19

[alsa-devel] [PATCH v5 1/2] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 25 Oct '19

25 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems. The examples are taken from the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Tested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/intel-dsp-config.h | 34 +++++ sound/hda/Kconfig | 10 +- sound/hda/Makefile | 5 +- sound/hda/intel-dsp-config.c | 249 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +- sound/pci/hda/hda_intel.c | 49 ++---- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 +-- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 + 11 files changed, 322 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..601e617918b8 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,6 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +snd-intel-dspcfg-$(CONFIG_SND_INTEL_NHLT) += intel-nhlt.o +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..d9f6d9e872b4 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,249 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <linux/bits.h> +#include <linux/dmi.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/intel-dsp-config.h> +#include <sound/intel-nhlt.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); + +#define FLAG_SST BIT(0) +#define FLAG_SOF BIT(1) +#define FLAG_SOF_ONLY_IF_DMIC BIT(16) + +struct config_entry { + u32 flags; + u16 device; + const struct dmi_system_id *dmi_table; +}; + +/* + * configuration table + * - the order of similar PCI ID entries is important! + * - the first successful match will win + */ +static const struct config_entry config_table[] = { +/* Cometlake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) + { + /* prefer SST */ + .flags = FLAG_SST, + .device = 0x02c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + { + .flags = FLAG_SOF, + .device = 0x02c8, + }, +#endif +/* Cometlake-H */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) + { + .flags = FLAG_SST, + .device = 0x06c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + { + .flags = FLAG_SOF, + .device = 0x06c8, + }, +#endif +/* Merrifield */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + { + .flags = FLAG_SOF, + .device = 0x119a, + }, +#endif +/* Broxton-T */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x1a98, + }, +#endif +/* Geminilake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { + .flags = FLAG_SOF, + .device = 0x3198, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) + { + .flags = FLAG_SST, + .device = 0x3198, + }, +#endif +/* Icelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x34c8, + }, +#endif +/* Elkhart Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b55, + }, +#endif +/* Appololake (Broxton-P) */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF, + .device = 0x5a98, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Up Squared", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + { + .flags = FLAG_SST, + .device = 0x5a98, + }, +#endif +/* Cannonlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9dc8, + }, +#endif +/* Sunrise Point-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d70, + }, +#endif +/* Kabylake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d71, + }, +#endif +/* Tigerlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa0c8, + }, +#endif +/* Coffelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa348, + }, +#endif +}; + +static const struct config_entry *snd_intel_dsp_find_config + (struct pci_dev *pci, const struct config_entry *table, u32 len) +{ + u16 device; + + device = pci->device; + for (; len > 0; len--, table++) { + if (table->device != device) + continue; + if (table->dmi_table && !dmi_check_system(table->dmi_table)) + continue; + return table; + } + return NULL; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + const struct config_entry *cfg; + + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* find the configuration for the specific device */ + cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table)); + if (!cfg) + return SND_INTEL_DSP_DRIVER_ANY; + + if (cfg->flags & FLAG_SOF) { + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + } else { + return SND_INTEL_DSP_DRIVER_SOF; + } + } + + if (cfg->flags & FLAG_SST) + return SND_INTEL_DSP_DRIVER_SST; + + return SND_INTEL_DSP_DRIVER_LEGACY; +} +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..a916963822e0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool dsp_driver = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,9 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(dsp_driver, bool, 0444); +MODULE_PARM_DESC(dsp_driver, "Allow DSP driver selection (bypass this driver) " + "(0=off, 1=on) (default=1)"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2021,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2038,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2061,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

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[alsa-devel] [PATCH -next] ALSA: intel_hdmi: Remove dev_err() on platform_get_irq() failure
by YueHaibing 25 Oct '19

25 Oct '19

platform_get_irq() will call dev_err() itself on failure, so there is no need for the driver to also do this. This is detected by coccinelle. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/x86/intel_hdmi_audio.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 5fd4e32..cd389d2 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1708,10 +1708,8 @@ static int hdmi_lpe_audio_probe(struct platform_device *pdev) /* get resources */ irq = platform_get_irq(pdev, 0); - if (irq < 0) { - dev_err(&pdev->dev, "Could not get irq resource: %d\n", irq); + if (irq < 0) return irq; - } res_mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res_mmio) { -- 2.7.4

2 1

[alsa-devel] [PATCH][next] ASoC: rt5677: Add missing null check for failed allocation of rt5677_dsp
by Colin King 25 Oct '19

25 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> The allocation of rt5677_dsp can potentially fail and return null, so add a null check and return -ENOMEM on a memory allocation failure. Addresses-Coverity: ("Dereference null return") Fixes: a0e0d135427c ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5677-spi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 36c02d200cfc..3a17643fcd9f 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -376,6 +376,8 @@ static int rt5677_spi_pcm_probe(struct snd_soc_component *component) rt5677_dsp = devm_kzalloc(component->dev, sizeof(*rt5677_dsp), GFP_KERNEL); + if (!rt5677_dsp) + return -ENOMEM; rt5677_dsp->dev = &g_spi->dev; mutex_init(&rt5677_dsp->dma_lock); INIT_DELAYED_WORK(&rt5677_dsp->copy_work, rt5677_spi_copy_work); -- 2.20.1

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[alsa-devel] [PATCH v2] ASoC: soc-dpcm: tidyup for_each_dpcm_xx() macro
by Kuninori Morimoto 25 Oct '19

25 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> for_each_dpcm_xx() macro is using "dpcm" as parameter (1), but, it is also struct member (2). #define for_each_dpcm_fe(be, stream, dpcm) \ list_for_each_entry(dpcm, &(be)->dpcm[stream]...) ^^^^(1) ^^^^(2) Thus, it will be compile error if user not used "dpcm" as parameter for_each_dpcm_fe(be, stream, dp) ^^ This patch fixup it. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - based on mark/for-5.5 branch include/sound/soc-dpcm.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index e55aeb0..b654ebf 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -103,15 +103,15 @@ struct snd_soc_dpcm_runtime { int trigger_pending; /* trigger cmd + 1 if pending, 0 if not */ }; -#define for_each_dpcm_fe(be, stream, dpcm) \ - list_for_each_entry(dpcm, &(be)->dpcm[stream].fe_clients, list_fe) - -#define for_each_dpcm_be(fe, stream, dpcm) \ - list_for_each_entry(dpcm, &(fe)->dpcm[stream].be_clients, list_be) -#define for_each_dpcm_be_safe(fe, stream, dpcm, _dpcm) \ - list_for_each_entry_safe(dpcm, _dpcm, &(fe)->dpcm[stream].be_clients, list_be) -#define for_each_dpcm_be_rollback(fe, stream, dpcm) \ - list_for_each_entry_continue_reverse(dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_fe(be, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(be)->dpcm[stream].fe_clients, list_fe) + +#define for_each_dpcm_be(fe, stream, _dpcm) \ + list_for_each_entry(_dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_safe(fe, stream, _dpcm, __dpcm) \ + list_for_each_entry_safe(_dpcm, __dpcm, &(fe)->dpcm[stream].be_clients, list_be) +#define for_each_dpcm_be_rollback(fe, stream, _dpcm) \ + list_for_each_entry_continue_reverse(_dpcm, &(fe)->dpcm[stream].be_clients, list_be) /* can this BE stop and free */ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt5677-spi: fix sparse warnings
by Curtis Malainey 25 Oct '19

25 Oct '19

Fix bugs reported by kbuild test robot Fixes: a0e0d135427c ("ASoC: rt5677: Add a PCM device for streaming hotword via SPI") Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 36c02d200cfc..958871b342ea 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -145,7 +145,7 @@ static int rt5677_spi_hw_free( snd_soc_component_get_drvdata(component); mutex_lock(&rt5677_dsp->dma_lock); - rt5677_dsp->substream = 0; + rt5677_dsp->substream = NULL; mutex_unlock(&rt5677_dsp->dma_lock); return snd_pcm_lib_free_vmalloc_buffer(substream); @@ -362,7 +362,7 @@ static void rt5677_spi_copy_work(struct work_struct *work) mutex_unlock(&rt5677_dsp->dma_lock); } -struct page *rt5677_spi_pcm_page( +static struct page *rt5677_spi_pcm_page( struct snd_soc_component *component, struct snd_pcm_substream *substream, unsigned long offset) -- 2.24.0.rc0.303.g954a862665-goog

2 1

[alsa-devel] [PATCH trivial] ASoC: pxa: poodle: Spelling s/enpoints/endpoints/, s/connetion/connection/
by Geert Uytterhoeven 25 Oct '19

25 Oct '19

Fix misspelling of "endpoints" and "connection". Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/pxa/poodle.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 48d5c2252b10081d..59ef04d0467a622d 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -56,7 +56,7 @@ static void poodle_ext_control(struct snd_soc_dapm_context *dapm) snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); } - /* set the enpoints to their new connetion states */ + /* set the endpoints to their new connection states */ if (poodle_spk_func == POODLE_SPK_ON) snd_soc_dapm_enable_pin(dapm, "Ext Spk"); else -- 2.17.1

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