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

[alsa-devel] [PATCH v2 0/6] ASoC: Intel: Skylake: Fix module removal
by Vinod Koul 16 Mar '16

16 Mar '16

Skylake driver were crashing on module removal when we run "alsa reload". The first issues was caused by component framework regression which Russell fixed and in in 4.5. In driver there were still some missing order which needs to be fixed. This series first update the list parsing in core, then ensure proper freeup of code objects, and then ensuring driver does proper cleanup and i915 dependency Changes in v2: - Add acked by Takashi - Remove codec destructor wrapper Vinod Koul (6): ALSA: hda: use list macro for parsing on cleanup ASoC: Intel: Skylake: free codec objects on removal ASoC: Intel: Skylake: Freeup properly on skl_dsp_free ASoC: Intel: Skylake: Unmap the address last ASoC: Intel: Skylake: Call i915 exit last ASoC: Intel: Skylake: remove call to pci_dev_put sound/hda/ext/hdac_ext_stream.c | 5 ++--- sound/soc/intel/skylake/skl-sst-dsp.c | 5 +++++ sound/soc/intel/skylake/skl.c | 17 ++++++++++------- 3 files changed, 17 insertions(+), 10 deletions(-) -- 1.9.1

3 15

[alsa-devel] [PATCH v4] ASoC: Intel: Skylake: remove call to pci_dev_put
by Vinod Koul 16 Mar '16

16 Mar '16

The PCI bus takes pci_dev_get() and pci_dev_put() is also there. So no need for drivers to invoke these. In SKL driver we were calling pci_dev_put() only which is not right, so remove this Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 72971dc55c52..07d9bc1cf3cb 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -724,7 +724,6 @@ static void skl_remove(struct pci_dev *pci) if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); - pci_dev_put(pci); /* codec removal, invoke bus_device_remove */ snd_hdac_ext_bus_device_remove(ebus); -- 1.9.1

1 0

[alsa-devel] [PATCH v3] ASoC: Intel: Skylake: remove call to pci_dev_put
by Vinod Koul 16 Mar '16

16 Mar '16

The PCI bus takes pci_dev_get() and pci_dev_put() is also there. So no need for drivers to invoke these. In SKL driver we were calling pci_dev_put() only which is not right, so remove this Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/skylake/skl.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index b67f3bd0fccd..c337d06793ec 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -732,7 +732,6 @@ static void skl_remove(struct pci_dev *pci) if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); - pci_dev_put(pci); skl_codecs_remove(ebus); skl_platform_unregister(&pci->dev); skl_free_dsp(skl); -- 1.9.1

2 3

[alsa-devel] [patch] ALSA: mixart: silence an uninitialized variable warning
by Dan Carpenter 16 Mar '16

16 Mar '16

We could print the uninitialized value of "stat" in the error message. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index bc81b9f..25c0ddd 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -132,7 +132,7 @@ static int mixart_set_pipe_state(struct mixart_mgr *mgr, } if(start) { - u32 stat; + u32 stat = 0; group_state.pipe_count = 0; /* in case of start same command once again with pipe_count=0 */

3 2

Re: [alsa-devel] [media] sound/usb: Use Media Controller API to share media resources
by Dan Carpenter 16 Mar '16

16 Mar '16

Hello Shuah Khan, This is a semi-automatic email about new static checker warnings. The patch aebb2b89bff0: "[media] sound/usb: Use Media Controller API to share media resources" from Mar 2, 2016, leads to the following Smatch complaint: sound/usb/card.c:615 usb_audio_probe() error: we previously assumed 'quirk' could be null (see line 534) sound/usb/card.c 583 /* 584 * For devices with more than one control interface, we assume the 585 * first contains the audio controls. We might need a more specific 586 * check here in the future. 587 */ 588 if (!chip->ctrl_intf) 589 chip->ctrl_intf = alts; 590 591 chip->txfr_quirk = 0; 592 err = 1; /* continue */ 593 if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { ^^^^^ Old code assumes "quirk" can be NULL. 594 /* need some special handlings */ 595 err = snd_usb_create_quirk(chip, intf, &usb_audio_driver, quirk); 596 if (err < 0) 597 goto __error; 598 } 599 600 if (err > 0) { 601 /* create normal USB audio interfaces */ 602 err = snd_usb_create_streams(chip, ifnum); 603 if (err < 0) 604 goto __error; 605 err = snd_usb_create_mixer(chip, ifnum, ignore_ctl_error); 606 if (err < 0) 607 goto __error; 608 } 609 610 /* we are allowed to call snd_card_register() many times */ 611 err = snd_card_register(chip->card); 612 if (err < 0) 613 goto __error; 614 615 if (quirk->media_device) { ^^^^^^^^^^^^^^^^^^^ Patch adds an unchecked dereference. 616 /* don't want to fail when media_snd_device_create() fails */ 617 media_snd_device_create(chip, intf); regards, dan carpenter

2 1

[alsa-devel] [PATCH 1/2] sound/usb/media: use core routine to initialize media_device
by Mauro Carvalho Chehab 16 Mar '16

16 Mar '16

The media core has now a routine to initialize media_device for USB devices. Use it, instead of doing its own logic, as it warrants that all USB drivers will behave the same. It also warrants that the device will get the same data, no matter if it was initialized initially via snd-usb-audio or via some other driver, like au0828. Signed-off-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- sound/usb/media.c | 18 +++++------------- 1 file changed, 5 insertions(+), 13 deletions(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index 93a50d01490c..44a5de91f748 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -263,19 +263,11 @@ int media_snd_device_create(struct snd_usb_audio *chip, mdev = media_device_get_devres(&usbdev->dev); if (!mdev) return -ENOMEM; - if (!mdev->dev) { - /* register media device */ - mdev->dev = &usbdev->dev; - if (usbdev->product) - strlcpy(mdev->model, usbdev->product, - sizeof(mdev->model)); - if (usbdev->serial) - strlcpy(mdev->serial, usbdev->serial, - sizeof(mdev->serial)); - strcpy(mdev->bus_info, usbdev->devpath); - mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); - media_device_init(mdev); - } + + /* Initialize media device */ + if (!mdev->dev) + media_device_usb_init(mdev, usbdev, NULL); + if (!media_devnode_is_registered(&mdev->devnode)) { ret = media_device_register(mdev); if (ret) { -- 2.5.0

1 0

[alsa-devel] [PATCH] media: add GFP flag to media_*() that could get called in atomic context
by Shuah Khan 16 Mar '16

16 Mar '16

Add GFP flags to media_create_pad_link(), media_create_intf_link(), media_devnode_create(), and media_add_link() that could get called in atomic context to allow callers to pass in the right flags for memory allocation. tree-wide driver changes for media_*() GFP flags change: Change drivers to add gfpflags to interffaces, media_create_pad_link(), media_create_intf_link() and media_devnode_create(). Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> Suggested-by: Mauro Carvalho Chehab <mchehab(a)osg.samsung.com> --- Ran through kbuild-all compile testing. Tested the changes in Win-TV HVR-950Q device drivers/media/dvb-core/dvbdev.c | 26 +++++++----- drivers/media/i2c/s5c73m3/s5c73m3-core.c | 6 ++- drivers/media/i2c/s5k5baf.c | 3 +- drivers/media/i2c/smiapp/smiapp-core.c | 3 +- drivers/media/i2c/tvp5150.c | 3 +- drivers/media/media-entity.c | 30 ++++++++------ drivers/media/platform/exynos4-is/media-dev.c | 19 +++++---- drivers/media/platform/omap3isp/isp.c | 47 ++++++++++++++-------- drivers/media/platform/s3c-camif/camif-core.c | 4 +- drivers/media/platform/vsp1/vsp1_drm.c | 6 +-- drivers/media/platform/vsp1/vsp1_drv.c | 9 +++-- drivers/media/platform/xilinx/xilinx-vipp.c | 4 +- drivers/media/usb/au0828/au0828-core.c | 3 +- drivers/media/usb/uvc/uvc_entity.c | 2 +- drivers/media/v4l2-core/v4l2-dev.c | 5 ++- drivers/media/v4l2-core/v4l2-device.c | 3 +- drivers/media/v4l2-core/v4l2-mc.c | 25 +++++++----- drivers/staging/media/davinci_vpfe/dm365_ipipeif.c | 3 +- drivers/staging/media/davinci_vpfe/dm365_isif.c | 2 +- drivers/staging/media/davinci_vpfe/dm365_resizer.c | 10 +++-- .../staging/media/davinci_vpfe/vpfe_mc_capture.c | 10 ++--- drivers/staging/media/omap4iss/iss.c | 17 +++++--- drivers/staging/media/omap4iss/iss_csi2.c | 6 ++- drivers/staging/media/omap4iss/iss_ipipeif.c | 3 +- drivers/staging/media/omap4iss/iss_resizer.c | 3 +- include/media/media-entity.h | 9 +++-- sound/usb/media.c | 15 ++++--- 27 files changed, 170 insertions(+), 106 deletions(-) diff --git a/drivers/media/dvb-core/dvbdev.c b/drivers/media/dvb-core/dvbdev.c index e1684c5..57f3e1e 100644 --- a/drivers/media/dvb-core/dvbdev.c +++ b/drivers/media/dvb-core/dvbdev.c @@ -399,7 +399,8 @@ static int dvb_register_media_device(struct dvb_device *dvbdev, dvbdev->intf_devnode = media_devnode_create(dvbdev->adapter->mdev, intf_type, 0, - DVB_MAJOR, minor); + DVB_MAJOR, minor, + GFP_KERNEL); if (!dvbdev->intf_devnode) return -ENOMEM; @@ -416,7 +417,7 @@ static int dvb_register_media_device(struct dvb_device *dvbdev, return 0; link = media_create_intf_link(dvbdev->entity, &dvbdev->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (!link) return -ENOMEM; #endif @@ -558,7 +559,8 @@ static int dvb_create_io_intf_links(struct dvb_adapter *adap, if (strncmp(entity->name, name, strlen(name))) continue; link = media_create_intf_link(entity, intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) return -ENOMEM; } @@ -680,7 +682,8 @@ int dvb_create_media_graph(struct dvb_adapter *adap, } if (demux && ca) { ret = media_create_pad_link(demux, 1, ca, - 0, MEDIA_LNK_FL_ENABLED); + 0, MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return -ENOMEM; } @@ -693,7 +696,8 @@ int dvb_create_media_graph(struct dvb_adapter *adap, strlen(DVR_TSOUT))) { ret = media_create_pad_link(demux, ++dvr_pad, - entity, 0, 0); + entity, 0, 0, + GFP_KERNEL); if (ret) return ret; } @@ -701,7 +705,8 @@ int dvb_create_media_graph(struct dvb_adapter *adap, strlen(DEMUX_TSOUT))) { ret = media_create_pad_link(demux, ++demux_pad, - entity, 0, 0); + entity, 0, 0, + GFP_KERNEL); if (ret) return ret; } @@ -713,14 +718,16 @@ int dvb_create_media_graph(struct dvb_adapter *adap, media_device_for_each_intf(intf, mdev) { if (intf->type == MEDIA_INTF_T_DVB_CA && ca) { link = media_create_intf_link(ca, intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) return -ENOMEM; } if (intf->type == MEDIA_INTF_T_DVB_FE && tuner) { link = media_create_intf_link(tuner, intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) return -ENOMEM; } @@ -732,7 +739,8 @@ int dvb_create_media_graph(struct dvb_adapter *adap, */ if (intf->type == MEDIA_INTF_T_DVB_DVR && demux) { link = media_create_intf_link(demux, intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) return -ENOMEM; } diff --git a/drivers/media/i2c/s5c73m3/s5c73m3-core.c b/drivers/media/i2c/s5c73m3/s5c73m3-core.c index 08af58f..dd994f1 100644 --- a/drivers/media/i2c/s5c73m3/s5c73m3-core.c +++ b/drivers/media/i2c/s5c73m3/s5c73m3-core.c @@ -1484,11 +1484,13 @@ static int s5c73m3_oif_registered(struct v4l2_subdev *sd) ret = media_create_pad_link(&state->sensor_sd.entity, S5C73M3_ISP_PAD, &state->oif_sd.entity, OIF_ISP_PAD, - MEDIA_LNK_FL_IMMUTABLE | MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_IMMUTABLE | MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); ret = media_create_pad_link(&state->sensor_sd.entity, S5C73M3_JPEG_PAD, &state->oif_sd.entity, OIF_JPEG_PAD, - MEDIA_LNK_FL_IMMUTABLE | MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_IMMUTABLE | MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); return ret; } diff --git a/drivers/media/i2c/s5k5baf.c b/drivers/media/i2c/s5k5baf.c index db82ed0..3692bec 100644 --- a/drivers/media/i2c/s5k5baf.c +++ b/drivers/media/i2c/s5k5baf.c @@ -1759,7 +1759,8 @@ static int s5k5baf_registered(struct v4l2_subdev *sd) ret = media_create_pad_link(&state->cis_sd.entity, PAD_CIS, &state->sd.entity, PAD_CIS, MEDIA_LNK_FL_IMMUTABLE | - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); return ret; } diff --git a/drivers/media/i2c/smiapp/smiapp-core.c b/drivers/media/i2c/smiapp/smiapp-core.c index a215efe..8c2e9ad 100644 --- a/drivers/media/i2c/smiapp/smiapp-core.c +++ b/drivers/media/i2c/smiapp/smiapp-core.c @@ -2508,7 +2508,8 @@ static int smiapp_register_subdevs(struct smiapp_sensor *sensor) &last->sd.entity, last->sink_pad, MEDIA_LNK_FL_ENABLED | - MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_IMMUTABLE, + GFP_KERNEL); if (rval) { dev_err(&client->dev, "media_create_pad_link failed\n"); diff --git a/drivers/media/i2c/tvp5150.c b/drivers/media/i2c/tvp5150.c index ff18444..fd99683 100644 --- a/drivers/media/i2c/tvp5150.c +++ b/drivers/media/i2c/tvp5150.c @@ -1197,7 +1197,8 @@ static int tvp5150_registered_async(struct v4l2_subdev *sd) return ret; ret = media_create_pad_link(input, 0, &sd->entity, - DEMOD_PAD_IF_INPUT, 0); + DEMOD_PAD_IF_INPUT, 0, + GFP_KERNEL); if (ret < 0) { media_device_unregister_entity(input); return ret; diff --git a/drivers/media/media-entity.c b/drivers/media/media-entity.c index 66a5392..115435b 100644 --- a/drivers/media/media-entity.c +++ b/drivers/media/media-entity.c @@ -566,14 +566,14 @@ EXPORT_SYMBOL_GPL(media_entity_put); * Links management */ -static struct media_link *media_add_link(struct list_head *head) +static struct media_link *media_add_link(struct list_head *head, gfp_t gfpflags) { struct media_link *link; if (in_atomic()) dump_stack(); - link = kzalloc(sizeof(*link), GFP_KERNEL); + link = kzalloc(sizeof(*link), gfpflags); if (link == NULL) return NULL; @@ -615,7 +615,8 @@ static void __media_entity_remove_link(struct media_entity *entity, int media_create_pad_link(struct media_entity *source, u16 source_pad, - struct media_entity *sink, u16 sink_pad, u32 flags) + struct media_entity *sink, u16 sink_pad, u32 flags, + gfp_t gfpflags) { struct media_link *link; struct media_link *backlink; @@ -624,7 +625,7 @@ media_create_pad_link(struct media_entity *source, u16 source_pad, BUG_ON(source_pad >= source->num_pads); BUG_ON(sink_pad >= sink->num_pads); - link = media_add_link(&source->links); + link = media_add_link(&source->links, gfpflags); if (link == NULL) return -ENOMEM; @@ -639,7 +640,7 @@ media_create_pad_link(struct media_entity *source, u16 source_pad, /* Create the backlink. Backlinks are used to help graph traversal and * are not reported to userspace. */ - backlink = media_add_link(&sink->links); + backlink = media_add_link(&sink->links, gfpflags); if (backlink == NULL) { __media_entity_remove_link(source, link); return -ENOMEM; @@ -682,7 +683,7 @@ int media_create_pad_links(const struct media_device *mdev, /* Trivial case: 1:1 relation */ if (source && sink) return media_create_pad_link(source, source_pad, - sink, sink_pad, flags); + sink, sink_pad, flags, GFP_KERNEL); /* Worse case scenario: n:n relation */ if (!source && !sink) { @@ -696,7 +697,7 @@ int media_create_pad_links(const struct media_device *mdev, continue; ret = media_create_pad_link(source, source_pad, sink, sink_pad, - flags); + flags, GFP_KERNEL); if (ret) return ret; flags &= ~(MEDIA_LNK_FL_ENABLED | @@ -718,10 +719,12 @@ int media_create_pad_links(const struct media_device *mdev, if (source) ret = media_create_pad_link(source, source_pad, - entity, sink_pad, flags); + entity, sink_pad, flags, + GFP_KERNEL); else ret = media_create_pad_link(entity, source_pad, - sink, sink_pad, flags); + sink, sink_pad, flags, + GFP_KERNEL); if (ret) return ret; flags &= ~(MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE); @@ -890,14 +893,15 @@ static void media_interface_init(struct media_device *mdev, struct media_intf_devnode *media_devnode_create(struct media_device *mdev, u32 type, u32 flags, - u32 major, u32 minor) + u32 major, u32 minor, + gfp_t gfpflags) { struct media_intf_devnode *devnode; if (in_atomic()) dump_stack(); - devnode = kzalloc(sizeof(*devnode), GFP_KERNEL); + devnode = kzalloc(sizeof(*devnode), gfpflags); if (!devnode) return NULL; @@ -921,11 +925,11 @@ EXPORT_SYMBOL_GPL(media_devnode_remove); struct media_link *media_create_intf_link(struct media_entity *entity, struct media_interface *intf, - u32 flags) + u32 flags, gfp_t gfpflags) { struct media_link *link; - link = media_add_link(&intf->links); + link = media_add_link(&intf->links, gfpflags); if (link == NULL) return NULL; diff --git a/drivers/media/platform/exynos4-is/media-dev.c b/drivers/media/platform/exynos4-is/media-dev.c index feb521f..145d90e 100644 --- a/drivers/media/platform/exynos4-is/media-dev.c +++ b/drivers/media/platform/exynos4-is/media-dev.c @@ -798,7 +798,8 @@ static int __fimc_md_create_fimc_sink_links(struct fimc_md *fmd, sink = &fmd->fimc[i]->vid_cap.subdev.entity; ret = media_create_pad_link(source, pad, sink, - FIMC_SD_PAD_SINK_CAM, flags); + FIMC_SD_PAD_SINK_CAM, flags, + GFP_KERNEL); if (ret) return ret; @@ -818,7 +819,8 @@ static int __fimc_md_create_fimc_sink_links(struct fimc_md *fmd, sink = &fmd->fimc_lite[i]->subdev.entity; ret = media_create_pad_link(source, pad, sink, - FLITE_SD_PAD_SINK, 0); + FLITE_SD_PAD_SINK, 0, + GFP_KERNEL); if (ret) return ret; @@ -850,13 +852,13 @@ static int __fimc_md_create_flite_source_links(struct fimc_md *fmd) sink = &fimc->ve.vdev.entity; /* FIMC-LITE's subdev and video node */ ret = media_create_pad_link(source, FLITE_SD_PAD_SOURCE_DMA, - sink, 0, 0); + sink, 0, 0, GFP_KERNEL); if (ret) break; /* Link from FIMC-LITE to IS-ISP subdev */ sink = &fmd->fimc_is->isp.subdev.entity; ret = media_create_pad_link(source, FLITE_SD_PAD_SOURCE_ISP, - sink, 0, 0); + sink, 0, 0, GFP_KERNEL); if (ret) break; } @@ -880,7 +882,8 @@ static int __fimc_md_create_fimc_is_links(struct fimc_md *fmd) /* Link from FIMC-IS-ISP subdev to FIMC */ sink = &fmd->fimc[i]->vid_cap.subdev.entity; ret = media_create_pad_link(source, FIMC_ISP_SD_PAD_SRC_FIFO, - sink, FIMC_SD_PAD_SINK_FIFO, 0); + sink, FIMC_SD_PAD_SINK_FIFO, 0, + GFP_KERNEL); if (ret) return ret; } @@ -893,7 +896,7 @@ static int __fimc_md_create_fimc_is_links(struct fimc_md *fmd) return 0; return media_create_pad_link(source, FIMC_ISP_SD_PAD_SRC_DMA, - sink, 0, 0); + sink, 0, 0, GFP_KERNEL); } /** @@ -944,7 +947,7 @@ static int fimc_md_create_links(struct fimc_md *fmd) ret = media_create_pad_link(&sensor->entity, pad, &csis->entity, CSIS_PAD_SINK, MEDIA_LNK_FL_IMMUTABLE | - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; @@ -996,7 +999,7 @@ static int fimc_md_create_links(struct fimc_md *fmd) sink = &fmd->fimc[i]->vid_cap.ve.vdev.entity; ret = media_create_pad_link(source, FIMC_SD_PAD_SOURCE, - sink, 0, flags); + sink, 0, flags, GFP_KERNEL); if (ret) break; } diff --git a/drivers/media/platform/omap3isp/isp.c b/drivers/media/platform/omap3isp/isp.c index 5d54e2c..f8d9c84 100644 --- a/drivers/media/platform/omap3isp/isp.c +++ b/drivers/media/platform/omap3isp/isp.c @@ -1669,7 +1669,7 @@ static int isp_link_entity( return -EINVAL; } - return media_create_pad_link(entity, i, input, pad, flags); + return media_create_pad_link(entity, i, input, pad, flags, GFP_KERNEL); } static int isp_register_entities(struct isp_device *isp) @@ -1748,43 +1748,50 @@ static int isp_create_links(struct isp_device *isp) /* Create links between entities and video nodes. */ ret = media_create_pad_link( &isp->isp_csi2a.subdev.entity, CSI2_PAD_SOURCE, - &isp->isp_csi2a.video_out.video.entity, 0, 0); + &isp->isp_csi2a.video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccp2.video_in.video.entity, 0, - &isp->isp_ccp2.subdev.entity, CCP2_PAD_SINK, 0); + &isp->isp_ccp2.subdev.entity, CCP2_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_OF, - &isp->isp_ccdc.video_out.video.entity, 0, 0); + &isp->isp_ccdc.video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_prev.video_in.video.entity, 0, - &isp->isp_prev.subdev.entity, PREV_PAD_SINK, 0); + &isp->isp_prev.subdev.entity, PREV_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_prev.subdev.entity, PREV_PAD_SOURCE, - &isp->isp_prev.video_out.video.entity, 0, 0); + &isp->isp_prev.video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_res.video_in.video.entity, 0, - &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0); + &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_res.subdev.entity, RESZ_PAD_SOURCE, - &isp->isp_res.video_out.video.entity, 0, 0); + &isp->isp_res.video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; @@ -1792,52 +1799,60 @@ static int isp_create_links(struct isp_device *isp) /* Create links between entities. */ ret = media_create_pad_link( &isp->isp_csi2a.subdev.entity, CSI2_PAD_SOURCE, - &isp->isp_ccdc.subdev.entity, CCDC_PAD_SINK, 0); + &isp->isp_ccdc.subdev.entity, CCDC_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccp2.subdev.entity, CCP2_PAD_SOURCE, - &isp->isp_ccdc.subdev.entity, CCDC_PAD_SINK, 0); + &isp->isp_ccdc.subdev.entity, CCDC_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_VP, - &isp->isp_prev.subdev.entity, PREV_PAD_SINK, 0); + &isp->isp_prev.subdev.entity, PREV_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_OF, - &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0); + &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_prev.subdev.entity, PREV_PAD_SOURCE, - &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0); + &isp->isp_res.subdev.entity, RESZ_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_VP, &isp->isp_aewb.subdev.entity, 0, - MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_VP, &isp->isp_af.subdev.entity, 0, - MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &isp->isp_ccdc.subdev.entity, CCDC_PAD_SOURCE_VP, &isp->isp_hist.subdev.entity, 0, - MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_ENABLED | MEDIA_LNK_FL_IMMUTABLE, + GFP_KERNEL); if (ret < 0) return ret; diff --git a/drivers/media/platform/s3c-camif/camif-core.c b/drivers/media/platform/s3c-camif/camif-core.c index 0b44b9a..a49b4c3 100644 --- a/drivers/media/platform/s3c-camif/camif-core.c +++ b/drivers/media/platform/s3c-camif/camif-core.c @@ -266,7 +266,7 @@ static int camif_create_media_links(struct camif_dev *camif) ret = media_create_pad_link(&camif->sensor.sd->entity, 0, &camif->subdev.entity, CAMIF_SD_PAD_SINK, MEDIA_LNK_FL_IMMUTABLE | - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; @@ -274,7 +274,7 @@ static int camif_create_media_links(struct camif_dev *camif) ret = media_create_pad_link(&camif->subdev.entity, i, &camif->vp[i - 1].vdev.entity, 0, MEDIA_LNK_FL_IMMUTABLE | - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); } return ret; diff --git a/drivers/media/platform/vsp1/vsp1_drm.c b/drivers/media/platform/vsp1/vsp1_drm.c index 021fe57..f868dec 100644 --- a/drivers/media/platform/vsp1/vsp1_drm.c +++ b/drivers/media/platform/vsp1/vsp1_drm.c @@ -525,7 +525,7 @@ int vsp1_drm_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&rpf->entity.subdev.entity, RWPF_PAD_SOURCE, &vsp1->bru->entity.subdev.entity, - i, flags); + i, flags, GFP_KERNEL); if (ret < 0) return ret; @@ -536,7 +536,7 @@ int vsp1_drm_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&vsp1->bru->entity.subdev.entity, vsp1->bru->entity.source_pad, &vsp1->wpf[0]->entity.subdev.entity, - RWPF_PAD_SINK, flags); + RWPF_PAD_SINK, flags, GFP_KERNEL); if (ret < 0) return ret; @@ -546,7 +546,7 @@ int vsp1_drm_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&vsp1->wpf[0]->entity.subdev.entity, RWPF_PAD_SOURCE, &vsp1->lif->entity.subdev.entity, - LIF_PAD_SINK, flags); + LIF_PAD_SINK, flags, GFP_KERNEL); if (ret < 0) return ret; diff --git a/drivers/media/platform/vsp1/vsp1_drv.c b/drivers/media/platform/vsp1/vsp1_drv.c index 25750a0..7295587 100644 --- a/drivers/media/platform/vsp1/vsp1_drv.c +++ b/drivers/media/platform/vsp1/vsp1_drv.c @@ -125,7 +125,8 @@ static int vsp1_create_sink_links(struct vsp1_device *vsp1, ret = media_create_pad_link(&source->subdev.entity, source->source_pad, - entity, pad, flags); + entity, pad, flags, + GFP_KERNEL); if (ret < 0) return ret; @@ -157,7 +158,7 @@ static int vsp1_uapi_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&vsp1->wpf[0]->entity.subdev.entity, RWPF_PAD_SOURCE, &vsp1->lif->entity.subdev.entity, - LIF_PAD_SINK, 0); + LIF_PAD_SINK, 0, GFP_KERNEL); if (ret < 0) return ret; } @@ -169,7 +170,7 @@ static int vsp1_uapi_create_links(struct vsp1_device *vsp1) &rpf->entity.subdev.entity, RWPF_PAD_SINK, MEDIA_LNK_FL_ENABLED | - MEDIA_LNK_FL_IMMUTABLE); + MEDIA_LNK_FL_IMMUTABLE, GFP_KERNEL); if (ret < 0) return ret; } @@ -188,7 +189,7 @@ static int vsp1_uapi_create_links(struct vsp1_device *vsp1) ret = media_create_pad_link(&wpf->entity.subdev.entity, RWPF_PAD_SOURCE, &wpf->video->video.entity, 0, - flags); + flags, GFP_KERNEL); if (ret < 0) return ret; } diff --git a/drivers/media/platform/xilinx/xilinx-vipp.c b/drivers/media/platform/xilinx/xilinx-vipp.c index e795a45..b444db4 100644 --- a/drivers/media/platform/xilinx/xilinx-vipp.c +++ b/drivers/media/platform/xilinx/xilinx-vipp.c @@ -158,7 +158,7 @@ static int xvip_graph_build_one(struct xvip_composite_device *xdev, ret = media_create_pad_link(local, local_pad->index, remote, remote_pad->index, - link_flags); + link_flags, GFP_KERNEL); if (ret < 0) { dev_err(xdev->dev, "failed to create %s:%u -> %s:%u link\n", @@ -272,7 +272,7 @@ static int xvip_graph_build_dma(struct xvip_composite_device *xdev) ret = media_create_pad_link(source, source_pad->index, sink, sink_pad->index, - link_flags); + link_flags, GFP_KERNEL); if (ret < 0) { dev_err(xdev->dev, "failed to create %s:%u -> %s:%u link\n", diff --git a/drivers/media/usb/au0828/au0828-core.c b/drivers/media/usb/au0828/au0828-core.c index 552ac58..de9ab11 100644 --- a/drivers/media/usb/au0828/au0828-core.c +++ b/drivers/media/usb/au0828/au0828-core.c @@ -251,7 +251,8 @@ create_link: ret = media_create_pad_link(decoder, DEMOD_PAD_AUDIO_OUT, mixer, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) dev_err(&dev->usbdev->dev, "Mixer Pad Link Create Error: %d\n", ret); diff --git a/drivers/media/usb/uvc/uvc_entity.c b/drivers/media/usb/uvc/uvc_entity.c index ac386bb..d9555dc 100644 --- a/drivers/media/usb/uvc/uvc_entity.c +++ b/drivers/media/usb/uvc/uvc_entity.c @@ -53,7 +53,7 @@ static int uvc_mc_create_links(struct uvc_video_chain *chain, remote_pad = remote->num_pads - 1; ret = media_create_pad_link(source, remote_pad, - sink, i, flags); + sink, i, flags, GFP_KERNEL); if (ret < 0) return ret; } diff --git a/drivers/media/v4l2-core/v4l2-dev.c b/drivers/media/v4l2-core/v4l2-dev.c index d8e5994..b0f53a2 100644 --- a/drivers/media/v4l2-core/v4l2-dev.c +++ b/drivers/media/v4l2-core/v4l2-dev.c @@ -786,7 +786,7 @@ static int video_register_media_controller(struct video_device *vdev, int type) vdev->intf_devnode = media_devnode_create(vdev->v4l2_dev->mdev, intf_type, 0, VIDEO_MAJOR, - vdev->minor); + vdev->minor, GFP_KERNEL); if (!vdev->intf_devnode) { media_device_unregister_entity(&vdev->entity); return -ENOMEM; @@ -797,7 +797,8 @@ static int video_register_media_controller(struct video_device *vdev, int type) link = media_create_intf_link(&vdev->entity, &vdev->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) { media_devnode_remove(vdev->intf_devnode); media_device_unregister_entity(&vdev->entity); diff --git a/drivers/media/v4l2-core/v4l2-device.c b/drivers/media/v4l2-core/v4l2-device.c index 06fa5f1..c7514e7 100644 --- a/drivers/media/v4l2-core/v4l2-device.c +++ b/drivers/media/v4l2-core/v4l2-device.c @@ -266,7 +266,8 @@ int v4l2_device_register_subdev_nodes(struct v4l2_device *v4l2_dev) link = media_create_intf_link(&sd->entity, &vdev->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!link) { err = -ENOMEM; goto clean_up; diff --git a/drivers/media/v4l2-core/v4l2-mc.c b/drivers/media/v4l2-core/v4l2-mc.c index 40ee864..b1652f9 100644 --- a/drivers/media/v4l2-core/v4l2-mc.c +++ b/drivers/media/v4l2-core/v4l2-mc.c @@ -92,7 +92,8 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) continue; ret = media_create_pad_link(entity, 0, io_v4l, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; } @@ -110,18 +111,21 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) ret = media_create_pad_link(tuner, TUNER_PAD_OUTPUT, if_vid, IF_VID_DEC_PAD_IF_INPUT, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; ret = media_create_pad_link(if_vid, IF_VID_DEC_PAD_OUT, decoder, DEMOD_PAD_IF_INPUT, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; } else { ret = media_create_pad_link(tuner, TUNER_PAD_OUTPUT, decoder, DEMOD_PAD_IF_INPUT, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; } @@ -130,7 +134,8 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) ret = media_create_pad_link(tuner, TUNER_PAD_AUD_OUT, if_aud, IF_AUD_DEC_PAD_IF_INPUT, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) return ret; } else { @@ -143,7 +148,7 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) if (io_v4l) { ret = media_create_pad_link(decoder, DEMOD_PAD_VID_OUT, io_v4l, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; } @@ -151,7 +156,7 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) if (io_swradio) { ret = media_create_pad_link(decoder, DEMOD_PAD_VID_OUT, io_swradio, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; } @@ -159,7 +164,7 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) if (io_vbi) { ret = media_create_pad_link(decoder, DEMOD_PAD_VBI_OUT, io_vbi, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, GFP_KERNEL); if (ret) return ret; } @@ -174,13 +179,13 @@ int v4l2_mc_create_media_graph(struct media_device *mdev) ret = media_create_pad_link(entity, 0, tuner, TUNER_PAD_RF_INPUT, - flags); + flags, GFP_KERNEL); break; case MEDIA_ENT_F_CONN_SVIDEO: case MEDIA_ENT_F_CONN_COMPOSITE: ret = media_create_pad_link(entity, 0, decoder, DEMOD_PAD_IF_INPUT, - flags); + flags, GFP_KERNEL); break; default: continue; diff --git a/drivers/staging/media/davinci_vpfe/dm365_ipipeif.c b/drivers/staging/media/davinci_vpfe/dm365_ipipeif.c index 633d645..d1442b9 100644 --- a/drivers/staging/media/davinci_vpfe/dm365_ipipeif.c +++ b/drivers/staging/media/davinci_vpfe/dm365_ipipeif.c @@ -977,7 +977,8 @@ vpfe_ipipeif_register_entities(struct vpfe_ipipeif_device *ipipeif, flags = 0; ret = media_create_pad_link(&ipipeif->video_in.video_dev.entity, 0, - &ipipeif->subdev.entity, 0, flags); + &ipipeif->subdev.entity, 0, flags, + GFP_KERNEL); if (ret < 0) goto fail; diff --git a/drivers/staging/media/davinci_vpfe/dm365_isif.c b/drivers/staging/media/davinci_vpfe/dm365_isif.c index 9905789..2470655 100644 --- a/drivers/staging/media/davinci_vpfe/dm365_isif.c +++ b/drivers/staging/media/davinci_vpfe/dm365_isif.c @@ -1824,7 +1824,7 @@ int vpfe_isif_register_entities(struct vpfe_isif_device *isif, /* connect isif to video node */ ret = media_create_pad_link(&isif->subdev.entity, 1, &isif->video_out.video_dev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_create_link; return 0; diff --git a/drivers/staging/media/davinci_vpfe/dm365_resizer.c b/drivers/staging/media/davinci_vpfe/dm365_resizer.c index a91395c..067f699 100644 --- a/drivers/staging/media/davinci_vpfe/dm365_resizer.c +++ b/drivers/staging/media/davinci_vpfe/dm365_resizer.c @@ -1833,26 +1833,28 @@ int vpfe_resizer_register_entities(struct vpfe_resizer_device *resizer, /* create link between Resizer Crop----> Resizer A*/ ret = media_create_pad_link(&resizer->crop_resizer.subdev.entity, 1, &resizer->resizer_a.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_create_link; /* create link between Resizer Crop----> Resizer B*/ ret = media_create_pad_link(&resizer->crop_resizer.subdev.entity, 2, &resizer->resizer_b.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_create_link; /* create link between Resizer A ----> video out */ ret = media_create_pad_link(&resizer->resizer_a.subdev.entity, 1, - &resizer->resizer_a.video_out.video_dev.entity, 0, flags); + &resizer->resizer_a.video_out.video_dev.entity, 0, flags, + GFP_KERNEL); if (ret < 0) goto out_create_link; /* create link between Resizer B ----> video out */ ret = media_create_pad_link(&resizer->resizer_b.subdev.entity, 1, - &resizer->resizer_b.video_out.video_dev.entity, 0, flags); + &resizer->resizer_b.video_out.video_dev.entity, 0, flags, + GFP_KERNEL); if (ret < 0) goto out_create_link; diff --git a/drivers/staging/media/davinci_vpfe/vpfe_mc_capture.c b/drivers/staging/media/davinci_vpfe/vpfe_mc_capture.c index ec46f36..bee9f5c 100644 --- a/drivers/staging/media/davinci_vpfe/vpfe_mc_capture.c +++ b/drivers/staging/media/davinci_vpfe/vpfe_mc_capture.c @@ -447,32 +447,32 @@ static int vpfe_register_entities(struct vpfe_device *vpfe_dev) if (vpfe_dev->sd[i]->entity.num_pads) { ret = media_create_pad_link(&vpfe_dev->sd[i]->entity, 0, &vpfe_dev->vpfe_isif.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; } ret = media_create_pad_link(&vpfe_dev->vpfe_isif.subdev.entity, 1, &vpfe_dev->vpfe_ipipeif.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; ret = media_create_pad_link(&vpfe_dev->vpfe_ipipeif.subdev.entity, 1, &vpfe_dev->vpfe_ipipe.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; ret = media_create_pad_link(&vpfe_dev->vpfe_ipipe.subdev.entity, 1, &vpfe_dev->vpfe_resizer.crop_resizer.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; ret = media_create_pad_link(&vpfe_dev->vpfe_ipipeif.subdev.entity, 1, &vpfe_dev->vpfe_resizer.crop_resizer.subdev.entity, - 0, flags); + 0, flags, GFP_KERNEL); if (ret < 0) goto out_resizer_register; diff --git a/drivers/staging/media/omap4iss/iss.c b/drivers/staging/media/omap4iss/iss.c index fb80d2b..933c8a7 100644 --- a/drivers/staging/media/omap4iss/iss.c +++ b/drivers/staging/media/omap4iss/iss.c @@ -1065,7 +1065,7 @@ static int iss_register_entities(struct iss_device *iss) } ret = media_create_pad_link(&sensor->entity, 0, input, pad, - flags); + flags, GFP_KERNEL); if (ret < 0) goto done; } @@ -1110,31 +1110,36 @@ static int iss_create_links(struct iss_device *iss) /* Connect the submodules. */ ret = media_create_pad_link( &iss->csi2a.subdev.entity, CSI2_PAD_SOURCE, - &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SINK, 0); + &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &iss->csi2b.subdev.entity, CSI2_PAD_SOURCE, - &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SINK, 0); + &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SOURCE_VP, - &iss->resizer.subdev.entity, RESIZER_PAD_SINK, 0); + &iss->resizer.subdev.entity, RESIZER_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &iss->ipipeif.subdev.entity, IPIPEIF_PAD_SOURCE_VP, - &iss->ipipe.subdev.entity, IPIPE_PAD_SINK, 0); + &iss->ipipe.subdev.entity, IPIPE_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; ret = media_create_pad_link( &iss->ipipe.subdev.entity, IPIPE_PAD_SOURCE_VP, - &iss->resizer.subdev.entity, RESIZER_PAD_SINK, 0); + &iss->resizer.subdev.entity, RESIZER_PAD_SINK, 0, + GFP_KERNEL); if (ret < 0) return ret; diff --git a/drivers/staging/media/omap4iss/iss_csi2.c b/drivers/staging/media/omap4iss/iss_csi2.c index aaca39d..4b40132 100644 --- a/drivers/staging/media/omap4iss/iss_csi2.c +++ b/drivers/staging/media/omap4iss/iss_csi2.c @@ -1352,13 +1352,15 @@ int omap4iss_csi2_create_links(struct iss_device *iss) /* Connect the CSI2a subdev to the video node. */ ret = media_create_pad_link(&csi2a->subdev.entity, CSI2_PAD_SOURCE, - &csi2a->video_out.video.entity, 0, 0); + &csi2a->video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; /* Connect the CSI2b subdev to the video node. */ ret = media_create_pad_link(&csi2b->subdev.entity, CSI2_PAD_SOURCE, - &csi2b->video_out.video.entity, 0, 0); + &csi2b->video_out.video.entity, 0, 0, + GFP_KERNEL); if (ret < 0) return ret; diff --git a/drivers/staging/media/omap4iss/iss_ipipeif.c b/drivers/staging/media/omap4iss/iss_ipipeif.c index 23de833..2f124b0 100644 --- a/drivers/staging/media/omap4iss/iss_ipipeif.c +++ b/drivers/staging/media/omap4iss/iss_ipipeif.c @@ -827,7 +827,8 @@ int omap4iss_ipipeif_create_links(struct iss_device *iss) /* Connect the IPIPEIF subdev to the video node. */ return media_create_pad_link(&ipipeif->subdev.entity, IPIPEIF_PAD_SOURCE_ISIF_SF, - &ipipeif->video_out.video.entity, 0, 0); + &ipipeif->video_out.video.entity, 0, 0, + GFP_KERNEL); } /* diff --git a/drivers/staging/media/omap4iss/iss_resizer.c b/drivers/staging/media/omap4iss/iss_resizer.c index f1d352c..8d5766a 100644 --- a/drivers/staging/media/omap4iss/iss_resizer.c +++ b/drivers/staging/media/omap4iss/iss_resizer.c @@ -869,7 +869,8 @@ int omap4iss_resizer_create_links(struct iss_device *iss) /* Connect the RESIZER subdev to the video node. */ return media_create_pad_link(&resizer->subdev.entity, RESIZER_PAD_SOURCE_MEM, - &resizer->video_out.video.entity, 0, 0); + &resizer->video_out.video.entity, 0, 0, + GFP_KERNEL); } /* diff --git a/include/media/media-entity.h b/include/media/media-entity.h index 6dc9e4e..0665c2d 100644 --- a/include/media/media-entity.h +++ b/include/media/media-entity.h @@ -595,6 +595,7 @@ static inline void media_entity_cleanup(struct media_entity *entity) {}; * @sink: pointer to &media_entity of the sink pad. * @sink_pad: number of the sink pad in the pads array. * @flags: Link flags, as defined in include/uapi/linux/media.h. + * @gfpflags: Get Free Page (GFP) flags, as defined in include/linux/gfp.h * * Valid values for flags: * A %MEDIA_LNK_FL_ENABLED flag indicates that the link is enabled and can be @@ -613,7 +614,7 @@ static inline void media_entity_cleanup(struct media_entity *entity) {}; */ __must_check int media_create_pad_link(struct media_entity *source, u16 source_pad, struct media_entity *sink, - u16 sink_pad, u32 flags); + u16 sink_pad, u32 flags, gfp_t gfpflags); /** * media_create_pad_links() - creates a link between two entities. @@ -883,11 +884,12 @@ void __media_entity_pipeline_stop(struct media_entity *entity); struct media_intf_devnode * __must_check media_devnode_create(struct media_device *mdev, u32 type, u32 flags, - u32 major, u32 minor); + u32 major, u32 minor, gfp_t gfpflags); /** * media_devnode_remove() - removes a device node interface * * @devnode: pointer to &media_intf_devnode to be freed. + * @gfpflags: Get Free Page (GFP) flags, as defined in include/linux/gfp.h * * When a device node interface is removed, all links to it are automatically * removed. @@ -901,6 +903,7 @@ struct media_link * * @entity: pointer to %media_entity * @intf: pointer to %media_interface * @flags: Link flags, as defined in include/uapi/linux/media.h. + * @gfpflags: Get Free Page (GFP) flags, as defined in include/linux/gfp.h * * * Valid values for flags: @@ -921,7 +924,7 @@ struct media_link * */ __must_check media_create_intf_link(struct media_entity *entity, struct media_interface *intf, - u32 flags); + u32 flags, gfp_t gfpflags); /** * __media_remove_intf_link() - remove a single interface link * diff --git a/sound/usb/media.c b/sound/usb/media.c index 93a50d01..f5ac038 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -88,14 +88,16 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, MAJOR(pcm_dev->devt), - MINOR(pcm_dev->devt)); + MINOR(pcm_dev->devt), + GFP_KERNEL); if (!mctl->intf_devnode) { ret = -ENOMEM; goto unregister_entity; } mctl->intf_link = media_create_intf_link(&mctl->media_entity, &mctl->intf_devnode->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!mctl->intf_link) { ret = -ENOMEM; goto devnode_remove; @@ -107,7 +109,8 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, case MEDIA_ENT_F_AUDIO_MIXER: ret = media_create_pad_link(entity, mixer_pad, &mctl->media_entity, 0, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (ret) goto remove_intf_link; break; @@ -180,7 +183,8 @@ int media_snd_mixer_init(struct snd_usb_audio *chip) if (!ctl_intf) { ctl_intf = media_devnode_create(mdev, intf_type, 0, MAJOR(ctl_dev->devt), - MINOR(ctl_dev->devt)); + MINOR(ctl_dev->devt), + GFP_KERNEL); if (!ctl_intf) return -ENOMEM; chip->ctl_intf_media_devnode = ctl_intf; @@ -213,7 +217,8 @@ int media_snd_mixer_init(struct snd_usb_audio *chip) mctl->intf_link = media_create_intf_link(&mctl->media_entity, &ctl_intf->intf, - MEDIA_LNK_FL_ENABLED); + MEDIA_LNK_FL_ENABLED, + GFP_KERNEL); if (!mctl->intf_link) { media_device_unregister_entity(&mctl->media_entity); media_entity_cleanup(&mctl->media_entity); -- 2.5.0

5 11

[alsa-devel] [PATCH 0/3] ALSA: usb-audio: Fix NULL dereferences via invalid EP accesses
by Takashi Iwai 16 Mar '16

16 Mar '16

Hi, this is a small patchset to address the possible Oops by accessing the invalid EP that was borked by a malformed firmware. One patch is a clean up while two are fixes. Takashi === Takashi Iwai (3): ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk() ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk() ALSA: usb-audio: Add sanity checks for endpoint accesses sound/usb/clock.c | 2 ++ sound/usb/endpoint.c | 3 +++ sound/usb/mixer_quirks.c | 4 ++++ sound/usb/pcm.c | 2 ++ sound/usb/quirks.c | 22 ++++++++++++++-------- 5 files changed, 25 insertions(+), 8 deletions(-) -- 2.7.3

1 3

[alsa-devel] pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large
by Klement Sekera 16 Mar '16

16 Mar '16

Hi devs, I'm experiencing random hick-ups on my Carbon X1 2nd version with a Xonar U3 USB sound card plugged into the onelink dock. When this happens, the whole desktop freezes, whatever was last playing (second or two) loops and after a while (~30seconds) everything recovers.. I see this in my logs: Mar 16 11:37:07 ksekera-arch rtkit-daemon[1207]: The canary thread is apparently starving. Taking action. Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 4194088 bytes (23776 ms). Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA develop Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_dump(): Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Hardware PCM card 2 'USB Advanced Audio Device' device 0 subdevice 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Its setup is: Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: stream : PLAYBACK Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: access : MMAP_INTERLEAVED Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: format : S16_LE Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: subformat : STD Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: channels : 2 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: rate : 44100 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: exact rate : 44100 (44100/1) Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: msbits : 16 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: buffer_size : 88200 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_size : 44100 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_time : 1000000 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: tstamp_mode : ENABLE Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: tstamp_type : MONOTONIC Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_step : 1 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: avail_min : 87319 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_event : 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: start_threshold : -1 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: stop_threshold : 6206523236469964800 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: silence_threshold: 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: silence_size : 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: boundary : 6206523236469964800 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: appl_ptr : 20607492 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: hw_ptr : 21567814 Mar 16 11:37:07 ksekera-arch rtkit-daemon[1207]: Demoting known real-time threads. Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_delay() returned a value that is exceptionally large: -3674960 bytes (-20833 ms). Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA develop Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: snd_pcm_dump(): Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Hardware PCM card 2 'USB Advanced Audio Device' device 0 subdevice 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: Its setup is: Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: stream : PLAYBACK Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: access : MMAP_INTERLEAVED Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: format : S16_LE Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: subformat : STD Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: channels : 2 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: rate : 44100 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: exact rate : 44100 (44100/1) Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: msbits : 16 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: buffer_size : 88200 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_size : 44100 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_time : 1000000 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: tstamp_mode : ENABLE Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: tstamp_type : MONOTONIC Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_step : 1 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: avail_min : 87319 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: period_event : 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: start_threshold : -1 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: stop_threshold : 6206523236469964800 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: silence_threshold: 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: silence_size : 0 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: boundary : 6206523236469964800 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: appl_ptr : 20653792 Mar 16 11:37:12 ksekera-arch pulseaudio[1203]: E: [alsa-sink-USB Audio] alsa-util.c: hw_ptr : 21572797 Mar 16 11:37:07 ksekera-arch rtkit-daemon[1207]: Successfully demoted thread 1203 of process 1203 (/usr/bin/pulseaudio). Mar 16 11:37:07 ksekera-arch rtkit-daemon[1207]: Demoted 1 threads. I'm on arch linux: ksekera@ksekera-arch ~> cat /proc/asound/version Advanced Linux Sound Architecture Driver Version k4.3.6-1-ck. ksekera@ksekera-arch ~> uname -a Linux ksekera-arch 4.3.6-1-ck #1 SMP PREEMPT Fri Feb 19 20:24:27 EST 2016 x86_64 GNU/Linux Let me know if you need any more info, Regards, Klement

1 0

[alsa-devel] [PATCH] ASoC: dapm: Do not traverse widget hooks to snd-soc-dummy
by Harry Pan 16 Mar '16

16 Mar '16

All components are initially given an empty card when registering platform, and since the commit 6e78108bda78 ("ASoC: core: Don't probe the component which is dummy")', snd-soc-dummy will not be probed so that it remains an empty card assigned. This patch ignores to iterate widget hooks to the 'snd-soc-dummy' component, else it will trigger memory fault because of invalid dereference address of card->widgets. Test by grep -rsI "hello" /sys/devices/platform/skl_nau88l25_ssm4567_i2s/ Conflicts: sound/soc/soc-dapm.c Signed-off-by: Harry Pan <harry.pan(a)intel.com> --- sound/soc/soc-dapm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 581175a..0bc15c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2188,6 +2188,9 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, int count = 0; char *state = "not set"; + if (!strcmp(cmpnt->name, "snd-soc-dummy")) + return 0; + list_for_each_entry(w, &cmpnt->card->widgets, list) { if (w->dapm != dapm) continue; -- 2.1.2

1 0

[alsa-devel] [PATCH 0/6] ASoC: Intel: Skylake: Fix module removal
by Vinod Koul 16 Mar '16

16 Mar '16

Skyalke driver were crashing on module removal when we run "alsa reload". The first issues was caused by component framwork regression which Russell fixed and in in 4.5. In driver there were still some missing order which needs to be fixed. This series first update the list parsing in core, then ensure proper freeup of code objects, and then esnureing driver does proper cleanup and i915 dependency Vinod Koul (6): ALSA: hda: use list macro for parsing on cleanup ASoC: Intel: Skylake: free codec objects on removal ASoC: Intel: Skylake: Freeup properly on skl_dsp_free ASoC: Intel: Skylake: Unmap the address last ASoC: Intel: Skylake: Call i915 exit last ASoC: Intel: Skylake: remove call to pci_dev_put sound/hda/ext/hdac_ext_stream.c | 5 ++--- sound/soc/intel/skylake/skl-sst-dsp.c | 5 +++++ sound/soc/intel/skylake/skl.c | 22 +++++++++++++++------- 3 files changed, 22 insertions(+), 10 deletions(-) Thanks -- 1.9.1

3 14

[alsa-devel] [PATCH] ASoC: rt5616: add I2C dependency
by Arnd Bergmann 16 Mar '16

16 Mar '16

The rt5616 codec registers itself as an i2c driver, but can be enabled even when i2c is turned off, which leads to a build error: codecs/rt5616.c:1419:1: error: data definition has no type or storage class [-Werror] module_i2c_driver(rt5616_i2c_driver); This adds an explicit Kconfig dependency, like the other codec drivers. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 288bc356a881 ("ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616") --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..7ef3a0c16478 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -629,6 +629,7 @@ config SND_SOC_RT5514 config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" + depends on I2C config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" -- 2.7.0

2 1

[alsa-devel] [PATCH v6] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 16 Mar '16

16 Mar '16

This is an initial imx-wm8958 device-tree-only machine driver. The sound card driver support three dai link, AIF1-DAI, AIF2-DAI and AIF3-DAI. We can configure the cpu dai from device tree, if present it will create corresponding sound card subdevice. At least one cpu dai phandle should be set from device tree. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- v5->v6: some header files should be moved to the next patch. So remove it. .../devicetree/bindings/sound/imx-audio-wm8958.txt | 69 +++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 325 +++++++++++++++++++++ 4 files changed, 410 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..7978c62 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,69 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + +Optional properties: + + - cpu-dai1 : The phandle of CPU DAI1 controller, which is for + aif1 of wm8958. At least one cpu dai is required. + + - cpu-dai2 : The phandle of CPU DAI2 controller, which is for + aif2 of wm8958. At least one cpu dai is required. + + - cpu-dai3 : The phandle of CPU DAI3 controller, which is for + aif3 of wm8958. At least one cpu dai is required. + + - dai-link1-name : If present, set the dai link1 name and stream name + to the specific strings. Otherwise, set the dai + link1 name and stream name to "AIF1-DAI". + + - dai-link2-name : If present, set the dai link2 name and stream name + to the specific strings. Otherwise, set the dai + link2 name and stream name to "AIF2-DAI". + + - dai-link3-name : If present, set the dai link3 name and stream name + to the specific strings. Otherwise, set the dai + link3 name and stream name to "AIF3-DAI". + + - fsl,cpu-dai1-master : If present, cpu dai1 works as master, and will + provide bit clock and frame clock. Otherwise, + cpu dai1 works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-ddr3-arm2-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai1 = <&sai1>; + dai-link1-name = "Hifi"; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..5ac200e --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,325 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define AIF1_DAI (0) +#define AIF2_DAI (1) +#define AIF3_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == AIF1_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == AIF1_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == AIF1_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link dai_link[DAI_LINK_NUM]; + struct snd_soc_card card; + unsigned long mclk_freq[WM8958_MCLK_MAX]; + bool is_cpu_dai1_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == AIF1_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == AIF1_DAI) { + if (data->is_cpu_dai1_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (data->is_cpu_dai1_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == AIF1_DAI && data->is_cpu_dai1_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_wm8958_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [AIF1_DAI] = { + .name = "AIF1-DAI", + .stream_name = "AIF1-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [AIF2_DAI] = { + .name = "AIF2-DAI", + .stream_name = "AIF2-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [AIF3_DAI] = { + .name = "AIF3-DAI", + .stream_name = "AIF3-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np[DAI_LINK_NUM], *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev[DAI_LINK_NUM]; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + struct clk *mclk; + char tmp[8], *name; + int num_links = 0; + int ret, i; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + for (i = 0; i < DAI_LINK_NUM; i++) { + sprintf(tmp, "cpu-dai%d", i + 1); + cpu_np[i] = of_parse_phandle(np, tmp, 0); + if (!cpu_np[i]) + continue; + + cpu_pdev[i] = of_find_device_by_node(cpu_np[i]); + if (!cpu_pdev[i]) { + dev_err(&pdev->dev, + "failed to get cpu dai%d platform device\n", i); + ret = -EINVAL; + goto fail; + } + data->dai_link[num_links] = imx_wm8958_dai_link[i]; + data->dai_link[num_links].cpu_dai_name = dev_name(&cpu_pdev[i]->dev); + data->dai_link[num_links].platform_of_node = cpu_np[i]; + + sprintf(tmp, "dai-link%d-name", i + 1); + if(!of_property_read_string(np, tmp, (const char **)&name)) { + data->dai_link[num_links].name = name; + data->dai_link[num_links].stream_name = name; + } + num_links++; + } + + if (!num_links) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + /* cpu dai1 link support cpu dai master and slave mode */ + if (of_property_read_bool(np, "fsl,cpu-dai1-master")) { + data->is_cpu_dai1_master = true; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else { + data->is_cpu_dai1_master = false; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + mclk = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(mclk)) + data->mclk_freq[i] = clk_get_rate(mclk); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = num_links; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np[AIF1_DAI]); + of_node_put(cpu_np[AIF2_DAI]); + of_node_put(cpu_np[AIF3_DAI]); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: cs35l32: prevent unexpected parse result of device tree
by Takashi Sakamoto 16 Mar '16

16 Mar '16

Hi, When compiling recent kernel, I got a warning caused by a codec in ASoC. sound/soc/codecs/cs35l32.c: In function ‘cs35l32_i2c_probe’: sound/soc/codecs/cs35l32.c:278:2: warning: ‘val’ may be used uninitialized in this function [-Wmaybe-uninitialized] switch (val) { ^ sound/soc/codecs/cs35l32.c:272:15: note: ‘val’ was declared here unsigned int val; ^ I'm not an ASoC developer, furthermore, have no knowledge about this Cirrus codec. Although, as a quick glance it may include a logical bug to assign wrong information to structure member as a reult of parsing device tree. This patch is my proposal to fix it. I'm happy to receive your comments. Regards Takashi Sakamoto (1): ASoC: cs35l32: fix unexpected parse result of device tree blob sound/soc/codecs/cs35l32.c | 4 ++++ 1 file changed, 4 insertions(+) -- 2.7.0

2 3

[alsa-devel] [PATCH] ASoC: nau8825: fix interrupt fails and unstable after resume
by John Hsu 16 Mar '16

16 Mar '16

This patch is to fix interrupt fails when resume back. In solution, we add IRQ re-initiation, and reconstruct suspend and resume with set_bias_level function. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 42 ++++++++++++++++++++++++++++++++++++------ 1 file changed, 36 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 504c969..2e2e11a 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1092,6 +1092,36 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static int nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable DDACR needed for interrupts */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + nau8825_restart_jack_detection(regmap); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1121,6 +1151,8 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, "Failed to sync cache: %d\n", ret); return ret; } + if (nau8825->irq) + nau8825_resume_setup(nau8825); } break; @@ -1330,24 +1362,22 @@ static int nau8825_i2c_remove(struct i2c_client *client) #ifdef CONFIG_PM_SLEEP static int nau8825_suspend(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); - disable_irq(client->irq); + disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); return 0; } static int nau8825_resume(struct device *dev) { - struct i2c_client *client = to_i2c_client(dev); struct nau8825 *nau8825 = dev_get_drvdata(dev); regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); + enable_irq(nau8825->irq); return 0; } -- 2.6.4

2 8

[alsa-devel] [PATCH v5] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 15 Mar '16

15 Mar '16

This is an initial imx-wm8958 device-tree-only machine driver. The sound card driver support three dai link, HIFI, VOICE and BT. We can configure the cpu dai from device tree, if present it will create corresponding sound card subdevice. So at least one cpu dai phandle should be set from device tree. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v4->v5: support set dai link name and stream name from device tree. .../devicetree/bindings/sound/imx-audio-wm8958.txt | 69 +++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 328 +++++++++++++++++++++ 4 files changed, 413 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..7978c62 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,69 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + +Optional properties: + + - cpu-dai1 : The phandle of CPU DAI1 controller, which is for + aif1 of wm8958. At least one cpu dai is required. + + - cpu-dai2 : The phandle of CPU DAI2 controller, which is for + aif2 of wm8958. At least one cpu dai is required. + + - cpu-dai3 : The phandle of CPU DAI3 controller, which is for + aif3 of wm8958. At least one cpu dai is required. + + - dai-link1-name : If present, set the dai link1 name and stream name + to the specific strings. Otherwise, set the dai + link1 name and stream name to "AIF1-DAI". + + - dai-link2-name : If present, set the dai link2 name and stream name + to the specific strings. Otherwise, set the dai + link2 name and stream name to "AIF2-DAI". + + - dai-link3-name : If present, set the dai link3 name and stream name + to the specific strings. Otherwise, set the dai + link3 name and stream name to "AIF3-DAI". + + - fsl,cpu-dai1-master : If present, cpu dai1 works as master, and will + provide bit clock and frame clock. Otherwise, + cpu dai1 works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-ddr3-arm2-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai1 = <&sai1>; + dai-link1-name = "Hifi"; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..650b0c8 --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,328 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define AIF1_DAI (0) +#define AIF2_DAI (1) +#define AIF3_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == AIF1_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == AIF1_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == AIF1_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link dai_link[DAI_LINK_NUM]; + struct snd_soc_card card; + unsigned long mclk_freq[WM8958_MCLK_MAX]; + bool is_cpu_dai1_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == AIF1_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == AIF1_DAI) { + if (data->is_cpu_dai1_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (data->is_cpu_dai1_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == AIF1_DAI && data->is_cpu_dai1_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_wm8958_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [AIF1_DAI] = { + .name = "AIF1-DAI", + .stream_name = "AIF1-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [AIF2_DAI] = { + .name = "AIF2-DAI", + .stream_name = "AIF2-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .ops = &imx_wm8958_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [AIF3_DAI] = { + .name = "AIF3-DAI", + .stream_name = "AIF3-DAI", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np[DAI_LINK_NUM], *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev[DAI_LINK_NUM]; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + struct clk *mclk; + char tmp[8], *name; + int num_links = 0; + int ret, i; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + for (i = 0; i < DAI_LINK_NUM; i++) { + sprintf(tmp, "cpu-dai%d", i + 1); + cpu_np[i] = of_parse_phandle(np, tmp, 0); + if (!cpu_np[i]) + continue; + + cpu_pdev[i] = of_find_device_by_node(cpu_np[i]); + if (!cpu_pdev[i]) { + dev_err(&pdev->dev, + "failed to get cpu dai%d platform device\n", i); + ret = -EINVAL; + goto fail; + } + data->dai_link[num_links] = imx_wm8958_dai_link[i]; + data->dai_link[num_links].cpu_dai_name = dev_name(&cpu_pdev[i]->dev); + data->dai_link[num_links].platform_of_node = cpu_np[i]; + + sprintf(tmp, "dai-link%d-name", i + 1); + if(!of_property_read_string(np, tmp, (const char **)&name)) { + data->dai_link[num_links].name = name; + data->dai_link[num_links].stream_name = name; + } + num_links++; + } + + if (!num_links) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + /* cpu dai1 link support cpu dai master and slave mode */ + if (of_property_read_bool(np, "fsl,cpu-dai1-master")) { + data->is_cpu_dai1_master = true; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else { + data->is_cpu_dai1_master = false; + data->dai_link[AIF1_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + mclk = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(mclk)) + data->mclk_freq[i] = clk_get_rate(mclk); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = num_links; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np[AIF1_DAI]); + of_node_put(cpu_np[AIF2_DAI]); + of_node_put(cpu_np[AIF3_DAI]); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix unconditional GPIO toggle via automute
by Takashi Iwai 15 Mar '16

15 Mar '16

Cirrus HD-audio driver may adjust GPIO pins for EAPD dynamically depending on the jack plug state. This works fine for the auto-mute mode where the speaker gets muted upon the HP jack plug. OTOH, when the auto-mute mode is off, this turns off the EAPD unexpectedly depending on the jack state, which results in the silent speaker output. This patch fixes the silent speaker output issue by setting GPIO bits constantly when the auto-mute mode is off. Reported-and-tested-by: moosotc(a)gmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_cirrus.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index c1c855a6c0af..a47e8ae0eb30 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -174,8 +174,12 @@ static void cs_automute(struct hda_codec *codec) snd_hda_gen_update_outputs(codec); if (spec->gpio_eapd_hp || spec->gpio_eapd_speaker) { - spec->gpio_data = spec->gen.hp_jack_present ? - spec->gpio_eapd_hp : spec->gpio_eapd_speaker; + if (spec->gen.automute_speaker) + spec->gpio_data = spec->gen.hp_jack_present ? + spec->gpio_eapd_hp : spec->gpio_eapd_speaker; + else + spec->gpio_data = + spec->gpio_eapd_hp | spec->gpio_eapd_speaker; snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, spec->gpio_data); } -- 2.7.3

1 0

[alsa-devel] No audio output via built-in speaker on a MacMini while headphone jack is plugged in
by moosotc＠gmail.com 15 Mar '16

15 Mar '16

`amixer set Speaker unmute' makes no difference physically unplugging the headphone jack from the socket does, i.e. `amixer set Speaker (un)mute' does the right thing. -- mailto:moosotc@gmail.com

2 8

[alsa-devel] [PATCH] [media] sound/usb: fix NULL dereference in usb_audio_probe()
by Nicolai Stange 15 Mar '16

15 Mar '16

With commit aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources") an access to quirk->media_device without checking for quirk != NULL has been introduced in usb_audio_probe(). With a Plantronics USB headset (device ID 0x047f:0xc010) attached, this results in the following splat at boot time: BUG: unable to handle kernel NULL pointer dereference at 0000000000000014 IP: [<ffffffffa089aa6c>] usb_audio_probe+0x2cc/0x9a0 [snd_usb_audio] Oops: 0000 [#1] SMP [...] CPU: 2 PID: 696 Comm: systemd-udevd Not tainted 4.5.0-next-20160315 #13 Hardware name: Dell Inc. Latitude E6540/0725FP, BIOS A10 06/26/2014 task: ffff88021c88d7c0 ti: ffff88003d5b0000 task.ti: ffff88003d5b0000 RIP: 0010:[<ffffffffa089aa6c>] [<ffffffffa089aa6c>] usb_audio_probe+0x2cc/0x9a0 [snd_usb_audio] [...] Call Trace: [<ffffffff815a8e16>] usb_probe_interface+0x136/0x2d0 [<ffffffff81509edc>] driver_probe_device+0x22c/0x440 [<ffffffff8150a1c1>] __driver_attach+0xd1/0xf0 [<ffffffff8150a0f0>] ? driver_probe_device+0x440/0x440 [<ffffffff815077ec>] bus_for_each_dev+0x6c/0xc0 [<ffffffff815095ce>] driver_attach+0x1e/0x20 [<ffffffff81509013>] bus_add_driver+0x1c3/0x280 [<ffffffff8150ab10>] driver_register+0x60/0xe0 [<ffffffff815a7711>] usb_register_driver+0x81/0x140 [<ffffffffa08c7000>] ? 0xffffffffa08c7000 [<ffffffffa08c701e>] usb_audio_driver_init+0x1e/0x1000 [snd_usb_audio] [<ffffffff81002123>] do_one_initcall+0xb3/0x1f0 [<ffffffff811fb091>] ? __vunmap+0x81/0xd0 [<ffffffff8121b8d2>] ? kmem_cache_alloc_trace+0x182/0x1d0 [<ffffffff811b0267>] ? do_init_module+0x27/0x1d8 [<ffffffff811b029f>] do_init_module+0x5f/0x1d8 [<ffffffff8112ce35>] load_module+0x1fe5/0x27a0 [<ffffffff81129870>] ? __symbol_put+0x60/0x60 [<ffffffff81241690>] ? vfs_read+0x110/0x130 [<ffffffff8112d866>] SYSC_finit_module+0xe6/0x120 [<ffffffff8112d8be>] SyS_finit_module+0xe/0x10 [<ffffffff81003d94>] do_syscall_64+0x64/0x110 [<ffffffff817c0b61>] entry_SYSCALL64_slow_path+0x25/0x25 After encountering this, the system-udevd process seems to be blocked until it is killed when hitting its timeout of 3min. In analogy to the other accesses to members of quirk in usb_audio_probe(), check for quirk != NULL before accessing its ->media_device. Fixes: aebb2b89bff0 ("[media] sound/usb: Use Media Controller API to share media resources") Signed-off-by: Nicolai Stange <nicstange(a)gmail.com> --- Applicable to linux-next-20160315. sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 63244bb..479621e 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -612,7 +612,7 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; - if (quirk->media_device) { + if (quirk && quirk->media_device) { /* don't want to fail when media_snd_device_create() fails */ media_snd_device_create(chip, intf); } -- 2.7.2

4 4

Re: [alsa-devel] ASoC: TLV320AIC3x: Adding additional functionality for 3106 with [Patch] for discuss
by Peter Ujfalusi 15 Mar '16

15 Mar '16

On 03/14/2016 04:57 PM, Timur Karaldin wrote: > Hi, Peter! > What's about placing capture controls in the Playback section of Alsamixer > view? Yeah, it is annoying. They can be fixed by: "Line1L Switch" -> "Line1L Capture Switch" and so on. On the other hand we had these unfortunate control names I think since the beginning and it is kind of an ABI for existing products. I don't have issue changing them, but existing setups relying on the control names might break... > Does resetting of some registers during start/stop of recording > connect with it? Not sure what you mean. -- Péter > > 14.03.2016 17:47, Peter Ujfalusi пишет: >> Hi Timur, >> >> On 03/14/16 13:06, Timur Karaldin wrote: >>> Hi, colleagues! >>> >>> I'm newbie with ALSA devel. I need to get extra functionality from codec >>> through ALSA mixer. >>> Task: 1. making possibility to set-up gain for each input (XXX_2_LADCCTRL and >>> XXX_2_RADCCTRL registers) >> This part is a bit tricky as in the registers: >> value 0..8 is gain from 0dB to -12dB in steps of -1.5dB and value 0xf means >> basically mute/disconnect. >> >>> 2. making possibility to set-up debounce for button and headset detection ( >>> HEADSET_DETECT_CTRL_A register) >>> 3. making possibility to enable high-power AC-coupled output >>> (HEADSET_DETECT_CTRL_A register) >>> 4. making possibility to enable stereo pseudo differential output >>> (HEADSET_DETECT_CTRL_B register) >>> 5. enable headset detection (HEADSET_DETECT_CTRL_A) >>> 6. getting headset detect status (HEADSET_DETECT_CTRL_A) >>> >>> Well, my patch works in generally, but I don't like it very much, because: >>> 1. All controls for input amplifiers' gain are shown in Alsamixer-Playback >>> section. But not only mine, capture controls from base codec's version are >>> shown in playback section too, for example (Right or Left) PGA Mixer Line1(L >>> or R)/2(L or R)/MIC3(L or R), which is definitely capture settings. Should I >>> worried about it? because, as I understand, it could not be setted correctly >>> during changing playback/capture mode for power-saving reason. >>> 2. "Right PGA Mixer Line1R" control (and similar) using the same bits in >>> LINE1R_2_RADCCTRL register as my "Line1R RADC Gain" control (and similar) and >>> put there b"1111" when user switch disconnect it from ADC. So my control is >>> not shown when user disconnect it from ADC. >>> 3. When I start/stop arecord some of my options were reset (debounces, headset >>> enable, high-power AC-coupled, pseudo differential output). When I set up it >>> in shell and then start arecord, it's reset again. Is it the reason because of >>> displaying controls in playback mixer section? >>> >>> I will be very pleased for any advise, thank you! >>> >>> >>> here is my patch for tlv320aic3x.c >>> --- >>> sound/soc/codecs/tlv320aic3x.c | 55 >>> +++++++++++++++++++++++++++++++++++++++--- >>> 1 file changed, 52 insertions(+), 3 deletions(-) >>> >>> diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c >>> index d7349bc..27d031e 100644 >>> --- a/sound/soc/codecs/tlv320aic3x.c >>> +++ b/sound/soc/codecs/tlv320aic3x.c >>> @@ -122,11 +122,24 @@ static const struct reg_default aic3x_reg[] = { >>> { 108, 0x00 }, { 109, 0x00 }, >>> }; >>> >>> +static bool aic3106_volatile(struct device *dev, unsigned int reg) >>> +{ >>> + switch(reg) >>> + { >>> + case AIC3X_HEADSET_DETECT_CTRL_A: >>> + case AIC3X_HEADSET_DETECT_CTRL_B: >>> + return true; >>> + default: >>> + return false; >>> + } >>> +} >>> + >>> static const struct regmap_config aic3x_regmap = { >>> .reg_bits = 8, >>> .val_bits = 8, >>> >>> .max_register = DAC_ICC_ADJ, >>> + .volatile_reg = aic3106_volatile, >>> .reg_defaults = aic3x_reg, >>> .num_reg_defaults = ARRAY_SIZE(aic3x_reg), >>> .cache_type = REGCACHE_RBTREE, >>> @@ -196,7 +209,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, >>> struct snd_kcontrol *kcontrol, int event) >>> { >>> struct snd_soc_codec *codec = w->codec; >>> - struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); >>> + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); >> You have couple of these nop changes in the patch... >> >>> switch (event) { >>> case SND_SOC_DAPM_POST_PMU: >>> @@ -270,6 +283,26 @@ static const struct soc_enum aic3x_agc_decay_enum[] = { >>> SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), >>> }; >>> >>> +static const char *aic3x_adc_gain[] = { "0dB", "-1.5dB", "-3dB", "-4.5dB", >>> "-6dB", "-7.5dB", "-9dB", "-10.5dB", "-12dB" }; >>> +static const struct soc_enum aic3x_adc_gain_enum[] = { >>> + SOC_ENUM_SINGLE(MIC3LR_2_LADC_CTRL, 4, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(MIC3LR_2_RADC_CTRL, 0, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 3, 9, aic3x_adc_gain), >>> + SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 3, 9, aic3x_adc_gain), >>> +}; >> So the issue is that we have the DAPM switches controlling exactly the same >> registers. I believe if you would set the gain in a way that bit0 is 1, then >> DAPM will think that the path is disconnected. >> Also if you would set the gain and then mute and unmute the path you would >> have lost the gain you wanted to have.. >> >> The only way I can think of implementing these mixers is to have two sets of >> custom callbacks. one set is to set/get the gain and the other is to set/get >> the mute/disconnect on these. >> When the path is disconnected you should not write the gain change to the >> chip, but cache it and if the path is unmuted, you write the cached gain. When >> you mute the path you should take the set gain first, cache it, then >> disconnect the path. >> >> For these gains you should have DECLARE_TLV_DB_SCALE() and use >> SOC_SINGLE_TLV(). Make sure that the control name matches with the >> corresponding DAPM widget's name so ALSA can match them correctly. >> >>> +static const char *aic3x_headset_debounce[] = { "16ms", "32ms", "64ms", >>> "128ms", "256ms", "512ms" }; >>> +static const struct soc_enum aic3x_headset_debounce_enum[] = { >>> + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 2, 6, >>> aic3x_headset_debounce), >>> +}; >>> +static const char *aic3x_button_debounce[] = { "0ms", "8ms", "16ms", >>> "32ms" }; >>> +static const struct soc_enum aic3x_button_debounce_enum[] = { >>> + SOC_ENUM_SINGLE(AIC3X_HEADSET_DETECT_CTRL_A, 0, 4, >>> aic3x_button_debounce), >>> +}; >>> + >>> + >>> + >>> /* >>> * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps >>> */ >>> @@ -399,6 +432,19 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] >>> = { >>> SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), >>> >>> SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), >>> + /* Additional controls */ >>> + SOC_ENUM("Mic3L LADC Gain", aic3x_adc_gain_enum[0]), >>> + SOC_ENUM("Mic3R RADC Gain", aic3x_adc_gain_enum[1]), >>> + SOC_ENUM("Line1L LADC Gain", aic3x_adc_gain_enum[2]), >>> + SOC_ENUM("Line1R RADC Gain", aic3x_adc_gain_enum[3]), >>> + SOC_ENUM("Line2L LADC Gain", aic3x_adc_gain_enum[4]), >>> + SOC_ENUM("Line2R RADC Gain", aic3x_adc_gain_enum[5]), >> Please avoid using arrays as the driver no longer use these anymore. >> >>> + SOC_ENUM("Headset Debounce", aic3x_headset_debounce_enum), >>> + SOC_ENUM("Button Debounce", aic3x_button_debounce_enum), >>> + SOC_SINGLE("Headset Detected", AIC3X_HEADSET_DETECT_CTRL_A, >>> AIC3X_HEADSET_DETECT_SHIFT, AIC3X_HEADSET_DETECT_MASK, 0), >>> + SOC_SINGLE("Headset Detect Enable", AIC3X_HEADSET_DETECT_CTRL_A, 7, 1, >>> 0), >>> + SOC_SINGLE("High power output Ac-coupled", AIC3X_HEADSET_DETECT_CTRL_B, >>> 7, 1, 0), >>> + SOC_SINGLE("Stereo pseudodifferential output", >>> AIC3X_HEADSET_DETECT_CTRL_B, 3, 1, 0), >>> }; >>> >>> static const struct snd_kcontrol_new aic3x_mono_controls[] = { >>> @@ -614,7 +660,7 @@ static const struct snd_soc_dapm_widget >>> aic3x_dapm_widgets[] = { >>> SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32", >>> AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), >>> >>> - /* Mic Bias */ >>> + /* Mic Bias */ >> nop change >> >>> SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, >>> mic_bias_event, >>> SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), >>> @@ -692,7 +738,7 @@ static const struct snd_soc_dapm_route intercon[] = { >>> {"Left Line2L Mux", "single-ended", "LINE2L"}, >>> {"Left Line2L Mux", "differential", "LINE2L"}, >>> >>> - {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, >>> + {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, >> nop change >> >>> {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, >>> {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, >>> {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, >>> @@ -814,6 +860,7 @@ static const struct snd_soc_dapm_route intercon[] = { >>> {"Right HPCOM Mux", "external feedback", "Right HPCOM Mixer"}, >>> {"Right HP Com", NULL, "Right HPCOM Mux"}, >>> {"HPRCOM", NULL, "Right HP Com"}, >>> + >> why? >> >>> }; >>> >>> static const struct snd_soc_dapm_route intercon_mono[] = { >>> @@ -918,7 +965,9 @@ static int aic3x_hw_params(struct snd_pcm_substream >>> *substream, >>> data = (LDAC2LCH | RDAC2RCH); >>> data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000; >>> if (params_rate(params) >= 64000) >>> + { >>> data |= DUAL_RATE_MODE; >>> + } >> why? >> >>> snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data); >>> >>> /* codec sample rate select */ >>> -- >>> and tlv320aic3x.h >>> --- >>> sound/soc/codecs/tlv320aic3x.h | 6 ++++++ >>> 1 file changed, 6 insertions(+) >>> >>> diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h >>> index e521ac3..8ff42d2 100644 >>> --- a/sound/soc/codecs/tlv320aic3x.h >>> +++ b/sound/soc/codecs/tlv320aic3x.h >>> @@ -271,6 +271,12 @@ enum { >>> AIC3X_BUTTON_DEBOUNCE_32MS = 3 >>> }; >>> >>> +typedef struct { >>> + struct platform_device* pdev; >>> + struct proc_dir_entry *proc_value; >>> + struct snd_soc_codec *codec; >>> +} aic3106_detect_t; >>> + >>> #define AIC3X_HEADSET_DETECT_ENABLED 0x80 >>> #define AIC3X_HEADSET_DETECT_SHIFT 5 >>> #define AIC3X_HEADSET_DETECT_MASK 3 >>> -- >> >> When submitting please separate the patches by functionality. It is easier to >> review the changes. >> > >

2 2

[alsa-devel] [PATCH v3] ASoC: cs35l32: avoid uninitialized variable access
by Arnd Bergmann 15 Mar '16

15 Mar '16

gcc warns about the possibilty of accessing a property read from devicetree in cs35l32_i2c_probe() when it has not been initialized because CONFIG_OF is disabled: sound/soc/codecs/cs35l32.c: In function 'cs35l32_i2c_probe': sound/soc/codecs/cs35l32.c:278:2: warning: 'val' may be used uninitialized in this function [-Wmaybe-uninitialized] The code is actually correct because it checks the dev->of_node variable first and we know this is NULL here when CONFIG_OF is disabled, but Russell King noticed that it's broken when we probe the device using DT, and the properties are absent. The code already has some checking for incorrect values, and I keep that checking unchanged here, but add an additional check for an error returned by the property accessor functions that now gets handled the same way as incorrect data in the properties. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v3: Restore a line that was accidentally removed, as pointed out by Brian Austin. v2: fix bug in case of CONFIG_OF and missing properties v1: only address warning for the !CONFIG_OF case sound/soc/codecs/cs35l32.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 44c30fe3e315..287d13740be4 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -274,7 +274,9 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) pdata->sdout_share = val; - of_property_read_u32(np, "cirrus,boost-manager", &val); + if (of_property_read_u32(np, "cirrus,boost-manager", &val)) + val = -1u; + switch (val) { case CS35L32_BOOST_MGR_AUTO: case CS35L32_BOOST_MGR_AUTO_AUDIO: @@ -282,13 +284,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BOOST_MGR_FIXED: pdata->boost_mng = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,boost-manager DT value %d\n", val); pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; } - of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + if (of_property_read_u32(np, "cirrus,sdout-datacfg", &val)) + val = -1u; switch (val) { case CS35L32_DATA_CFG_LR_VP: case CS35L32_DATA_CFG_LR_STAT: @@ -296,13 +300,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_DATA_CFG_LR_VPSTAT: pdata->sdout_datacfg = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,sdout-datacfg DT value %d\n", val); pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; } - of_property_read_u32(np, "cirrus,battery-threshold", &val); + if (of_property_read_u32(np, "cirrus,battery-threshold", &val)) + val = -1u; switch (val) { case CS35L32_BATT_THRESH_3_1V: case CS35L32_BATT_THRESH_3_2V: @@ -310,13 +316,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_THRESH_3_4V: pdata->batt_thresh = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-threshold DT value %d\n", val); pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; } - of_property_read_u32(np, "cirrus,battery-recovery", &val); + if (of_property_read_u32(np, "cirrus,battery-recovery", &val)) + val = -1u; switch (val) { case CS35L32_BATT_RECOV_3_1V: case CS35L32_BATT_RECOV_3_2V: @@ -326,6 +334,7 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_RECOV_3_6V: pdata->batt_recov = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-recovery DT value %d\n", val); -- 2.7.0

3 2

[alsa-devel] [PATCH] ASoC: nau8825: improve FLL function for better performance
by John Hsu 15 Mar '16

15 Mar '16

In FLL calculation, increase VCO/DCO frequency for better performance. Besides, have different register configuration according to fraction or not when apply FLL parameters. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 47 +++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 13 ++++++++----- 2 files changed, 42 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 1c87299..9023dbd 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -31,7 +31,7 @@ #include "nau8825.h" #define NAU_FREF_MAX 13500000 -#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 struct nau8825_fll { @@ -957,8 +957,8 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, struct nau8825_fll *fll_param) { - u64 fvco; - unsigned int fref, i; + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. @@ -983,18 +983,23 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fll_param->ratio = fll_ratio[i].val; /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. - * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * FDCO must be within the 90MHz - 124MHz or the FFL cannot be * guaranteed across the full range of operation. * FDCO = freq_out * 2 * mclk_src_scaling */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { fvco = 256 * fs * 2 * mclk_src_scaling[i].param; - if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX) - break; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } } - if (i == ARRAY_SIZE(mclk_src_scaling)) + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) return -EINVAL; - fll_param->mclk_src = mclk_src_scaling[i].val; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional * input based on FDCO, FREF and FLL ratio. @@ -1009,7 +1014,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, struct nau8825_fll *fll_param) { regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, NAU8825_FLL_RATIO_MASK, fll_param->ratio); /* FLL 16-bit fractional input */ @@ -1022,10 +1028,25 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, - NAU8825_FLL_FILTER_SW_MASK, 0x0000); - /* FLL sigma delta modulator enable */ - regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + if (fll_param->fll_frac) { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_FILTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); + } else { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, NAU8825_FLL_FTR_SW_ACCU); + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_SDM_EN, 0); + } } /* freq_out must be 256*Fs in order to achieve the best performance */ diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 8ceb5f3..01bd386 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -118,15 +118,18 @@ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) /* FLL5 (0x08) */ -#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) +#define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8825_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8825_FLL_CLK_SW_MASK (0x1 << 13) +#define NAU8825_FLL_CLK_SW_N2 (0x1 << 13) +#define NAU8825_FLL_CLK_SW_REF (0x0 << 13) +#define NAU8825_FLL_FTR_SW_MASK (0x1 << 12) +#define NAU8825_FLL_FTR_SW_ACCU (0x1 << 12) +#define NAU8825_FLL_FTR_SW_FILTER (0x0 << 12) /* FLL6 (0x9) */ -#define NAU8825_DCO_EN_MASK (0x1 << 15) #define NAU8825_DCO_EN (0x1 << 15) -#define NAU8825_DCO_DIS (0x0 << 15) -#define NAU8825_SDM_EN_MASK (0x1 << 14) #define NAU8825_SDM_EN (0x1 << 14) -#define NAU8825_SDM_DIS (0x0 << 14) /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) -- 2.6.4

2 1

[alsa-devel] [PATCH 0/3] *** new features for alsabat ***
by han.lu＠intel.com 15 Mar '16

15 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Add 3 features for alsabat: 1. add default device name for playback and capture 2. add standalone mode, to bypass data analysis 3. add bash test script to cover basic alsabat features Lu, Han (3): alsabat: add default device name for playback and capture alsabat: add standalone mode alsabat: add bash test script bat/Makefile.am | 13 +++--- bat/alsa.c | 12 ------ bat/alsabat-test.sh | 112 ++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/alsabat.1 | 9 +++++ bat/bat.c | 18 ++++++++- bat/common.h | 3 ++ configure.ac | 5 ++- 7 files changed, 153 insertions(+), 19 deletions(-) create mode 100755 bat/alsabat-test.sh -- 2.5.0

2 4

[alsa-devel] [patch] ALSA: mixart: silence unitialized variable warnings
by Dan Carpenter 15 Mar '16

15 Mar '16

We print can print the uninitialized memory on error. Which is an info leak, I suppose but it's basically harmless. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 24a1955..58fd79eb 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -726,7 +726,7 @@ int mixart_update_playback_stream_level(struct snd_mixart* chip, int is_aes, int int volume[2]; struct mixart_msg request; struct mixart_set_out_stream_level_req set_level; - u32 status; + u32 status = 0; struct mixart_pipe *pipe; memset(&set_level, 0, sizeof(set_level)); @@ -778,7 +778,7 @@ int mixart_update_capture_stream_level(struct snd_mixart* chip, int is_aes) struct mixart_pipe *pipe; struct mixart_msg request; struct mixart_set_in_audio_level_req set_level; - u32 status; + u32 status = 0; if(is_aes) { idx = 1;

2 1

[alsa-devel] Max9768 driver in kernel 4.1
by Micka 15 Mar '16

15 Mar '16

Hi, I'm trying to use the max9768, I have defined it correctly in the dts file : max9768: max9768@49 { compatible = "max,max9768"; reg = <0x49>; }; I can see that in the driver, i go in the function max9768_i2c_probe <http://lxr.free-electrons.com/ident?v=4.1;i=max9768_i2c_probe> without problem, and I don't see errors when executing this line : err <http://lxr.free-electrons.com/ident?v=4.1;i=err> = snd_soc_register_codec <http://lxr.free-electrons.com/ident?v=4.1;i=snd_soc_register_codec>(&client <http://lxr.free-electrons.com/ident?v=4.1;i=client>->dev <http://lxr.free-electrons.com/ident?v=4.1;i=dev>, &max9768_codec_driver <http://lxr.free-electrons.com/ident?v=4.1;i=max9768_codec_driver>, NULL <http://lxr.free-electrons.com/ident?v=4.1;i=NULL>, 0); I have put some log : [ 9.213797] ASoC: dai register 2-0049 #0 [ 9.213806] ASoC: Registered codec 'max9768.2-0049' my problem is that i don't know how to configure the volume ? Any idea ? where is it ? I don't see where is it in the amixer controls list : root@beaglebone:~# amixer controls numid=1,iface=MIXER,name='PCM Playback Volume' numid=15,iface=MIXER,name='Line DAC Playback Volume' numid=14,iface=MIXER,name='Line PGA Bypass Volume' numid=20,iface=MIXER,name='Line Playback Switch' numid=33,iface=MIXER,name='ADC HPF Cut-off' numid=23,iface=MIXER,name='AGC Switch' numid=30,iface=MIXER,name='De-emphasis Switch' numid=17,iface=MIXER,name='HP DAC Playback Volume' numid=16,iface=MIXER,name='HP PGA Bypass Volume' numid=21,iface=MIXER,name='HP Playback Switch' numid=19,iface=MIXER,name='HPCOM DAC Playback Volume' numid=18,iface=MIXER,name='HPCOM PGA Bypass Volume' numid=22,iface=MIXER,name='HPCOM Playback Switch' numid=26,iface=MIXER,name='Left AGC Attack time' numid=28,iface=MIXER,name='Left AGC Decay time' numid=24,iface=MIXER,name='Left AGC Target level' numid=75,iface=MIXER,name='Left DAC Mux' numid=49,iface=MIXER,name='Left HP Mixer DACL1 Switch' numid=7,iface=MIXER,name='Left HP Mixer DACR1 Playback Volume' numid=51,iface=MIXER,name='Left HP Mixer DACR1 Switch' numid=48,iface=MIXER,name='Left HP Mixer PGAL Bypass Switch' numid=50,iface=MIXER,name='Left HP Mixer PGAR Bypass Switch' numid=6,iface=MIXER,name='Left HP Mixer PGAR Bypass Volume' numid=41,iface=MIXER,name='Left HPCOM Mixer DACL1 Switch' numid=11,iface=MIXER,name='Left HPCOM Mixer DACR1 Playback Volume' numid=43,iface=MIXER,name='Left HPCOM Mixer DACR1 Switch' numid=40,iface=MIXER,name='Left HPCOM Mixer PGAL Bypass Switch' numid=42,iface=MIXER,name='Left HPCOM Mixer PGAR Bypass Switch' numid=10,iface=MIXER,name='Left HPCOM Mixer PGAR Bypass Volume' numid=74,iface=MIXER,name='Left HPCOM Mux' numid=57,iface=MIXER,name='Left Line Mixer DACL1 Switch' numid=3,iface=MIXER,name='Left Line Mixer DACR1 Playback Volume' numid=59,iface=MIXER,name='Left Line Mixer DACR1 Switch' numid=56,iface=MIXER,name='Left Line Mixer PGAL Bypass Switch' numid=58,iface=MIXER,name='Left Line Mixer PGAR Bypass Switch' numid=2,iface=MIXER,name='Left Line Mixer PGAR Bypass Volume' numid=71,iface=MIXER,name='Left Line1L Mux' numid=70,iface=MIXER,name='Left Line1R Mux' numid=64,iface=MIXER,name='Left PGA Mixer Line1L Switch' numid=65,iface=MIXER,name='Left PGA Mixer Line1R Switch' numid=66,iface=MIXER,name='Left PGA Mixer Mic2L Switch' numid=67,iface=MIXER,name='Left PGA Mixer Mic2R Switch' numid=34,iface=MIXER,name='Output Driver Power-On time' numid=35,iface=MIXER,name='Output Driver Ramp-up step' numid=32,iface=MIXER,name='PGA Capture Switch' numid=31,iface=MIXER,name='PGA Capture Volume' numid=27,iface=MIXER,name='Right AGC Attack time' numid=29,iface=MIXER,name='Right AGC Decay time' numid=25,iface=MIXER,name='Right AGC Target level' numid=73,iface=MIXER,name='Right DAC Mux' numid=9,iface=MIXER,name='Right HP Mixer DACL1 Playback Volume' numid=45,iface=MIXER,name='Right HP Mixer DACL1 Switch' numid=47,iface=MIXER,name='Right HP Mixer DACR1 Switch' numid=44,iface=MIXER,name='Right HP Mixer PGAL Bypass Switch' numid=8,iface=MIXER,name='Right HP Mixer PGAL Bypass Volume' numid=46,iface=MIXER,name='Right HP Mixer PGAR Bypass Switch' numid=13,iface=MIXER,name='Right HPCOM Mixer DACL1 Playback Volume' numid=37,iface=MIXER,name='Right HPCOM Mixer DACL1 Switch' numid=39,iface=MIXER,name='Right HPCOM Mixer DACR1 Switch' numid=36,iface=MIXER,name='Right HPCOM Mixer PGAL Bypass Switch' numid=12,iface=MIXER,name='Right HPCOM Mixer PGAL Bypass Volume' numid=38,iface=MIXER,name='Right HPCOM Mixer PGAR Bypass Switch' numid=72,iface=MIXER,name='Right HPCOM Mux' numid=5,iface=MIXER,name='Right Line Mixer DACL1 Playback Volume' numid=53,iface=MIXER,name='Right Line Mixer DACL1 Switch' numid=55,iface=MIXER,name='Right Line Mixer DACR1 Switch' numid=52,iface=MIXER,name='Right Line Mixer PGAL Bypass Switch' numid=4,iface=MIXER,name='Right Line Mixer PGAL Bypass Volume' numid=54,iface=MIXER,name='Right Line Mixer PGAR Bypass Switch' numid=69,iface=MIXER,name='Right Line1L Mux' numid=68,iface=MIXER,name='Right Line1R Mux' numid=61,iface=MIXER,name='Right PGA Mixer Line1L Switch' numid=60,iface=MIXER,name='Right PGA Mixer Line1R Switch' numid=62,iface=MIXER,name='Right PGA Mixer Mic2L Switch' numid=63,iface=MIXER,name='Right PGA Mixer Mic2R Switch' Thank you for your work and help !

1 0

[alsa-devel] [PATCH 00/10] *** alsabat features and fixes ***
by han.lu＠intel.com 15 Mar '16

15 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> add alsabat features and some fixes. 1. add default device name 2. add standalone mode 3. add tinyalsa support 4. add bash script for basic features test on alsa and tinyalsa 5. refactor alsa thread process to make clearer structure, and allow more common functions shared by alsa and tinyalsa 6. some tiny fixes on code, text and comments Lu, Han (10): alsabat: add default device name for playback and capture alsabat: add standalone mode alsabat: add tinyalsa support alsabat: clean the thread loopback of alsa capture alsabat: refactor wav file process alsabat: use common data generator function alsabat: add interrupt handler for shutdown alsabat: fix an incorrect print alsabat: use variable for thread return value alsabat: add bash test script bat/Makefile.am | 14 +- bat/alsa.c | 215 +++++++++-------------------- bat/alsa.h | 3 + bat/alsabat.1 | 12 ++ bat/bat.c | 80 ++++++++++- bat/common.c | 89 ++++++++++++ bat/common.h | 14 ++ bat/testbat.sh | 160 +++++++++++++++++++++ bat/tinyalsa.c | 422 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/tinyalsa.h | 23 +++ configure.ac | 9 +- 11 files changed, 883 insertions(+), 158 deletions(-) create mode 100644 bat/testbat.sh create mode 100644 bat/tinyalsa.c create mode 100644 bat/tinyalsa.h -- 2.5.0

3 24

[alsa-devel] [PATCH v5 0/9] ASoC: Add mediatek HDMI codec support
by Philipp Zabel 15 Mar '16

15 Mar '16

Hi, these patches add an interface for Jyri's generic hdmi-codec driver [1] to the mediatek hdmi bridge driver. Since there haven't been many comments after the last submission, the patches are mostly unchanged. A new patch adds HDMI dai-links to the recently merged mt8173-rt5650 machine driver and I have updated the HDMI notification patch (and renamed patch 3 to mention the mt8173-rt5650-rt5676 machine driver). Changes since v4: - Update hdmi notifier to latest version - Added HDMI dai-links to new mt8173-rt5650 machine driver The mediatek drm patches apply on top of https://patchwork.kernel.org/patch/8337071/ ("drm/mediatek: Add HDMI support"), and need this patch: [2] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") The asoc patches depend on these two patches: https://patchwork.kernel.org/patch/7215121/ ("ALSA: pcm: add IEC958 channel status helper for hw_params") [1] https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver PC Liao (1): ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver Philipp Zabel (7): drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver ASoC: mediatek: address dai link array entries by enum video: rmk's HDMI notification prototype drm/mediatek: hdmi: issue notifications ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: hdmi-codec: Add ELD control drm/mediatek: hdmi: use helper function for N and CTS calculation .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +- drivers/gpu/drm/mediatek/Kconfig | 1 + drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_drm_hdmi_drv.c | 148 +++++++++++++++++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 89 +++++----- drivers/gpu/drm/mediatek/mtk_hdmi.h | 10 +- drivers/gpu/drm/mediatek/mtk_hdmi_hw.c | 179 ++++----------------- drivers/gpu/drm/mediatek/mtk_hdmi_hw.h | 4 +- drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 +++++++ include/linux/hdmi-notifier.h | 44 +++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/hdmi-codec.c | 123 +++++++++++++- sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 66 +++++++- sound/soc/mediatek/mt8173-rt5650.c | 47 ++++++ 18 files changed, 587 insertions(+), 217 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.7.0

4 13

Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Mark Brown 14 Mar '16

14 Mar '16

On Fri, Mar 11, 2016 at 02:27:03AM +0530, Jeeja KP wrote: > On Wed, Mar 02, 2016 at 11:02:15AM +0900, Mark Brown wrote: > > On Tue, Feb 16, 2016 at 09:47:47PM +0530, Jeeja KP wrote: > > Like I keep saying if you're thinking about this in terms of DPCM you're > > solving the wrong problem - we already support supplying parameters for > > device<->device links, we already have systems that need to configure > > things for device<->device links so we can already see that if we're > > doing something that only works for things that are on SoC and can use > > DPCM then we're not solving the problem. > No this is not exactly about DPCM. It is about the device<->device links that > assume you can have only codecs as hostless and thus change the direction > for the link. Since we are rendering from SoC we don't need direction change. You're routing them via DSP though? > > This is clearly confused, if we can't figure out what the two devices we > > are connecting are based on just looking at the devices then that's > > really not a good sign that our interfaces are sensible and easy to work > > with. > Right now dai_link assumes device<->device links based on params so we > cannot specify params for anything where we dont want direction change and would > like to treat as normal direction dai_link. So we propose this to move to > codec_link flag and use params only for creating controls based on params > provided. What I'm telling you is that you shouldn't be writing hacks like this to bodge things but coming up with scalable and maintainable solutions. I already discussed this with Vinod several times, we've discussed the idea that if you're having trouble with the asymmetries in the definitions of DAI links you should make the DAI links symmetric so things like this just work. We need clear, comprehensible abstractions which work robustly not some mess of special case handling.

2 1

[alsa-devel] [PATCH] topology: fix debug output to print correct "max" value.
by Liam Girdwood 14 Mar '16

14 Mar '16

Debug log is printing num_regs instead of max in the max section. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- src/topology/ctl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/topology/ctl.c b/src/topology/ctl.c index 1c073f7..c250227 100644 --- a/src/topology/ctl.c +++ b/src/topology/ctl.c @@ -350,7 +350,7 @@ int tplg_parse_control_bytes(snd_tplg_t *tplg, return -EINVAL; be->max = atoi(val); - tplg_dbg("\t%s: %d\n", id, be->num_regs); + tplg_dbg("\t%s: %d\n", id, be->max); continue; } -- 2.5.0

2 9

[alsa-devel] [GIT PULL] ASoC updates for v4.6
by Mark Brown 14 Mar '16

14 Mar '16

The following changes since commit f6cede5b49e822ebc41a099fe41ab4989f64e2cb: Linux 4.5-rc7 (2016-03-06 14:48:03 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.6 for you to fetch changes up to d4a6360f19c1c551afcba42be98df04651fab31b: Merge remote-tracking branches 'asoc/topic/sunxi', 'asoc/topic/topology' and 'asoc/topic/wm8974' into asoc-next (2016-03-13 15:17:56 +0700) ---------------------------------------------------------------- ASoC: Updates for v4.6 The main thing in terms of the core this time around has been some additional framework work for dynamic topologies (though we *still* don't appear to have a stable ABI for the topology code, it's probably worth considering if this will ever happen...). Otherwise the work has almost all been in the drivers: - HDMI support for Sky Lake, along with other fixes and enhancements for the Intel drivers. - Lots of improvements to the Renesas drivers. - Capture support for Qualcomm drivers. - Support for TI DaVinci DRA7xxx devices. - New machine drivers for Freescale systems with Cirrus CODECs, Mediatek systems with RT5650 CODECs. - New CPU drivers for Allwinner S/PDIF controllers - New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514. ---------------------------------------------------------------- Alan Cox (1): ASoC: Intel: Skylake: remove bogus comparison of an array with NULL Andreas Irestål (4): ASoC: adau17x1: Cache writes when core clock is disabled ASoC: adau17x1: Correct typos in file headers ASoC: adau17x1: Add adau17x1 bindings document ASoC: adau17x1: Add basic DT support for adau17x1 Andrzej Hajda (1): ASoC: max9867: fix type of variable containing error codes Antonio Ospite (1): ASoC: rk3036: fix missing dependency on REGMAP_MMIO Arnd Bergmann (3): ASoC: s3c24xx: use const snd_soc_component_driver pointer ASoC: pxa: remove unused variable ASoC: rockchip: use __maybe_unused to hide st_irq_syscfg_resume Axel Lin (7): ASoC: rt5659: Staticise rt5659_i2c_shutdown ASoC: max98926: Fix blr_clk_ratio calculation ASoC: max98926: Constify max98926_reg and max98926_regmap ASoC: max9867: Add CONFIG_SND_SOC_MAX9867 to build ASoC: max9867: Make max9867_i2c_id NULL terminated ASoC: max9867: Constify max9867_reg and max9867_regmap ASoC: max9867: Use params_width instead of open-coded Bard Liao (4): ASoC: rt298: enable IRQ for jack detection ASoC: rt298: Don't enable IRQ in i2c_probe ASoC: rt5640: Set PLL src according to source ASoC: rt5640: add supplys for dac power Caesar Wang (4): ASoC: rt5616: add mclk property for rt5616 document ASoC: rt5616: trivial: fix the typo ASoC: rt5616: add the mclk for the codec driver ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616 Carlo Caione (1): ASoC: cht_bsw_rt5645: Enable jack detection Charles Keepax (14): ASoC: wm5110: Fix up snd_soc_register_platform error path ASoC: wm_adsp: Use more generic naming for regions and caps definitions ASoC: wm_adsp: Add debug audio trace firmware ASoC: wm_adsp: Move setting of DSP speed into CODEC specific code ASoC: wm5110: Add support for audio trace firmware ASoC: wm5102: Add support for the audio trace firmware ASoC: wm_adsp: Add missing const from region definitions ASoC: wm_adsp: Return sample rate in wm_adsp_compr_pointer ASoC: arizona: Add support for SNDRV_PCM_RATE_KNOT ASoC: wm_adsp: Avoid narrow race condition on compr pointer ASoC: wm_adsp: Explicitly set the control access flags in all cases ASoC: wm_adsp: Fixup some odd line wrapping ASoC: wm_adsp: wm_coeff_{read|write}_control should use passed length ASoC: wm_adsp: Use correct local length in error message Damien.Horsley (1): ASoC: pcm3168a: Call clk_set_rate in pcm3168a_set_dai_sysclk Dan Carpenter (1): ASoC: max9867: silence and array overflow warning Dharageswari.R (3): ASoC: Intel: Skylake: Get node id using helper ASoC: Intel: Skylake: Add skl_tplg_be_get_cpr_module() helper ASoC: Intel: Skylake: Add MCLK configuration Fabio Estevam (1): ASoC: hdac_hdmi: Remove 'edev' NULL check Felipe F. Tonello (1): ASoC: fsl-asoc-card: add cs4271 and cs4272 support Florian Vaussard (2): ASoC: ads117x: Add bindings documentation for TI ADS117x ADC ASoC: ads117x: Add device tree compatible string Jacob Siverskog (3): ASoC: pcm179x: Split into core and SPI parts ASoC: pcm179x: Add I2C interface driver ASoC: pcm179x: Support continuous rates Jayachandran B (1): ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core Jeeja KP (16): ASoC: Intel: Skylake: Allow module parameter set after bind ASoC: Intel: Skylake: Add shutdown callback ASoC: Intel: Skylake: Fill BE blob with default params ASoC: hdac_hdmi: Add jack reporting ASoC: hdac_hdmi: Add PM support ASoC: Intel: boards: Enable HDMI and DP on skl_rt286 machine ASoC: Intel: boards: Enable HDMI and DP on nau88l2 machine ASoC: Intel: boards: Remove WoV sink in skl_rt286 ASoC: Intel: boards: Remove WoV sink in nau88125_ssm4567 machine ASoC: Intel: boards: Remove WoV sink in nau88125_max98357a machine ASoC: Intel: Skylake: correct direction in blob query ASoC: Intel: Skylake: Add IVI feedback path in nau88125_ssm4567 machine ASoC: Intel: Skylake: remove IVI codec loop in skl_nau88l25_ssm4567 machine ASoC: Intel: add dmabuffer to common sst_dsp ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops ASoC: hdac_hdmi: Add broxton device ID Jianqun Xu (1): ASoC: rockchip: add bindings for rk3399 i2s John Hsu (1): ASoC: nau8825: fix issue that pop noise when start playback John Lin (3): ASoC: rt5645: improve headphone pop when system resumes from S3 ASoC: rt5645: extend delay time for headphone pop noise ASoC: rt5616: add missing mute control for HPVOL Koro Chen (3): ASoC: mediatek: Add machine driver for ALC5650 codec ASoC: mediatek: Add I03/I04 widgets and corresponding routes ASoC: mediatek: Add machine driver for rt5650 rt5514 codec Kuninori Morimoto (59): ASoC: rsnd: remove unsed *parent ASoC: rsnd: use rsnd_mod_init() for ADG ASoC: rsnd: add debug message for rsnd_mod_call() ASoC: rsnd: don't update status if rsnd_mod_call() doesn't match ASoC: rsnd: try to connect connected mod is not error ASoC: rsnd: always call probe/remove for MUX ASoC: rsnd: don't overwrite io on rsnd_cmd_init() ASoC: rsnd: select each SRC correctly for CMD data path ASoC: rsnd: each mod has status again for CTU/MUX support ASoC: rsnd: attach Audio-DMAC-periperi correctly ASoC: rsnd: ctu: add rsnd_mix_activation() ASoC: rsnd: ctu: add rsnd_ctu_halt() ASoC: rsnd: ctu: settings matches to datasheet ASoC: rsnd: tidyup Playback/Capture sequence ASoC: rsnd: disable SRC.out/in in same time ASoC: rsnd: don't auto-recover when under/over run error ASoC: rsnd: add .irq callback ASoC: soc-pcm: copy dpcm->hw_params and call be_hw_params_fixup every time ASoC: rsnd: rsnd_ssi_is_multi_slave() macro uses rsnd_ssi_multi_slaves() ASoC: rsnd: rename RSND_DVC_CHANNELS to RSND_MAX_CHANNELS ASoC: rsnd: setup SRC_ROUTE_MODE0 before SRC_SRCIR ASoC: rsnd: indicates status failed SSI ASoC: rsnd: SSI function parameter uses "mod" instead of "ssi" ASoC: rsnd: fixup forever loop bug on SSI ASoC: rsnd: add missing .irq callback for DMA ASoC: rsnd: tidyup SSI init/start sequence ASoC: rsnd: Parent SSI attach is not needed if not clock master ASoC: rsnd: move rsnd_ssi_irq() position ASoC: rsnd: judge multi SSI in runtime ASoC: rsnd: judge work SSI in runtime regmap: add regmap_update_bits_base() regmap: merge regmap_update_bits() into macro regmap: merge regmap_update_bits_async() into macro regmap: merge regmap_update_bits_check() into macro regmap: merge regmap_update_bits_check_async() into macro regmap: add regmap_field_update_bits_base() regmap: merge regmap_field_write() into macro regmap: merge regmap_field_update_bits() into macro regmap: add regmap_fields_update_bits_base() regmap: merge regmap_fields_write() into macro regmap: merge regmap_fields_update_bits() into macro regmap: add regmap_field_force_xxx() macros regmap: add regmap_fields_force_xxx() macros ASoC: rsnd: rsnd_write() / rsnd_bset() uses regmap _force_ function ASoC: rsnd: don't call update callback if it was NULL ASoC: rsrc-card: add convert channels support ASoC: rsnd: tidyup rsnd_ssiu_init_gen2() ASoC: rsnd: add CTU basic support ASoC: rsnd: add rsnd_runtime_channel_xxx() ASoC: rsnd: add CTU support ASoC: rsrc-card: tidyup convert-channels explain ASoC: rsnd: don't enable SRC Sync mode when Capture with CMD ASoC: rsnd: SRC sync mode is not related to clock master mode ASoC: rsnd: SRC function parameter uses "mod" instead of "src" ASoC: rsnd: rename rsnd_enable_sync_convert() to rsnd_src_sync_is_enabled() ASoC: rsnd: add rsnd_src_get_in/out_rate() ASoC: rsnd: SRC TIMSEL support for Capture ASoC: rsnd: CMD TIMSEL support for Capture ASoC: rsnd: add simplified module explanation Lars-Peter Clausen (1): ASoC: ssm4567: Reset device before regcache_sync() Lukas Wunner (1): ASoC: Intel: Use acpi_dev_present() Maciej S. Szmigiero (1): ASoC: fsl_ssi: remove explicit register defaults Mans Rullgard (2): ASoC: wm8974: configure pll and mclk divider automatically ASoC: mxs-saif: check BUSY bit in hw_params() only if not mclk_in_use Marcus Cooper (2): ASoC: sunxi: add sun4i SPDIF transceiver ASoC: sunxi: Add support for the SPDIF block Mark Brown (31): regmap: Add explict native endian flag to DT bindings MIPS: dt: Explicitly specify native endian behaviour for syscon regmap: mmio: Convert to regmap_bus and fix accessor usage regmap: Return an error if a caller attempts to do an unsupported raw read regmap: cache: Fall back to register by register read for cache defaults Merge branch 'fix/mmio' of git://git.kernel.org/.../broonie/regmap into regmap-mmio Merge branch 'fix/intel' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/drm' of git://git.kernel.org/.../broonie/sound into asoc-hdac Merge tag 'asoc-fix-v4.5-rc4' into asoc-qcom Merge branch 'topic/mmio' of git://git.kernel.org/.../broonie/regmap into asoc-fsl-ssl Merge tag 'v4.5-rc5' into asoc-mtk Merge branch 'topic/hdac' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge branch 'topic/update-bits' of git://git.kernel.org/.../broonie/regmap into asoc-rcar Merge tag 'asoc-fix-v4.5-rc6' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/core-pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm179x' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm3168a' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ab8500', 'asoc/topic/adau17x1', 'asoc/topic/ads117x', 'asoc/topic/adsp' and 'asoc/topic/arizona' into asoc-next Merge remote-tracking branches 'asoc/topic/atmel', 'asoc/topic/bcm2835' and 'asoc/topic/cs42xx8' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/fsl-card' and 'asoc/topic/fsl-mpc5200' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-sai', 'asoc/topic/fsl-ssl', 'asoc/topic/hdac' and 'asoc/topic/max9867' into asoc-next Merge remote-tracking branches 'asoc/topic/max98926', 'asoc/topic/mtk', 'asoc/topic/mxs-saif', 'asoc/topic/nau8825' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa' and 'asoc/topic/qcom' into asoc-next Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/rockchip', 'asoc/topic/rt298', 'asoc/topic/rt5514' and 'asoc/topic/rt5616' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5640', 'asoc/topic/rt5659', 'asoc/topic/samsung' and 'asoc/topic/ssm4567' into asoc-next Merge remote-tracking branches 'asoc/topic/sunxi', 'asoc/topic/topology' and 'asoc/topic/wm8974' into asoc-next Martin Sperl (1): ASoC: bcm2835: move to use the clock framework Mengdong Lin (4): ALSA: pcm: Add snd_pcm_rate_range_to_bits() ASoC: topology: Add FE DAIs dynamically ASoC: topology: Add FE DAI links dynamically ASoC: topology: Fix setting of stream rates, rate_min and rate_max Nicolin Chen (1): ASoC: fsl-asoc-card: Don't add DAPM routes for ASRC if it doesn't exist Oder Chiou (1): ASoC: rt5514: add rt5514 codec driver PC Liao (2): ASoC: mediatek: Enable 33bit memory address to support 4GB DRAM ASoC: mediatek: remove soft reset and add second I2S clock Peter Ujfalusi (4): ASoC: davinci-mcasp: Discourage use of fck_parent for clock reparenting ASoC: davinci-mcasp: dai format runtime reconfiguration ASoC: davinci: Kconfig: Update the edma-pcm section's dependency and help ASoC: omap-hdmi-audio: Support for DRA7xx family Philipp Zabel (1): ASoC: mediatek: address dai link array entries by enum Pierre-Louis Bossart (5): ASoC: Intel: common: filter ACPI devices with _STA return value ASoC: Intel: common: add translation from HID to codec-name ASoC: Intel: bytcr_rt5640: change quirk position ASoC: Intel: bytcr_rt5640: use HID translation util ASoC: Intel: Atom: add support for CHT w/ RT5640 Rajan Vaja (1): ASoC: dapm: Remove unneeded variable ret returning 0 Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep codec power active during enumeration. Richard Fitzgerald (2): ASoC: cs47l24: Add voice control compressed stream ASoC: arizona: Make logging of FLL calculations clearer Sathya Prakash M R (1): ASoC: Intel: boards: Add IV feedback route for nau88l25_ssm4567 machine Sathyanarayana Nujella (1): ASoC: Intel: boards: Enable HDMI and DP on SKL nau88l25_max98357 machine Senthilnathan Veppur (1): ASoC: Intel: Bxtn: Add Broxton PCI ID Songjun Wu (1): ASoC: atmel_ssc_dai: distinguish the different SSC Srinivas Kandagatla (14): ASoC: qcom: use snd_dma_alloc/free* apis ASoC: qcom: add wrdma register details to lpass_variant ASoC: qcom: rename rdmactl_audif_start to dmactrl_audif_start ASoC: qcom: pass direction to dma allocation ASoC: qcom: ipq806x: add error in dma allocation. ASoC: qcom: rename rdma_ch_bit_map to dma_ch_bit_map ASoC: qcom: ipq806x: add wrdma related register offsets ASoC: qcom: add mic related i2s control register defines ASoC: qcom: add wrdma register definitions ASoC: qcom: add generic bit masks for RDMA and WRDMA ASoC: qcom: apq8016: set the correct max register for regmap ASoC: qcom: add mic support ASoC: qcom: apq8016-sbc: add mic support ASoC: qcom: apq8016: add wrdma support Stephen Boyd (1): ASoC: qcom: Don't specify LE device endianness Subhransu S. Prusty (15): ASoC: hdac_hdmi: Add hotplug notification and read ELD ASoC: hdac_hdmi: Apply constraints based on ELD drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi: create dais based on number of cvts ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Enable playback on all enumerated ports ASoC: hdac_hdmi: Add infoframe support for dp audio ASoC: hdac_hdmi: Fix possible memory leak in hw_params ASoC: hdac_hdmi: Don't fail in dai startup to make userland happy ASoC: hdac_hdmi: Fix to reconfigure registers in runtime resume ASoC: Intel: Skylake: Fix possible memory corruption in codec dai dma params ASoC: Intel: Skylake: Add HDMI FE and BE cpu dais ASoC: hdac_hdmi: Fix to wait for D3 before powering off codec ASoC: hdac_hdmi: Fix infoframe programming Sudip Mukherjee (6): ASoC: topology: fix build warning ASoC: fix memory leak ASoC: intel: remove unused variable ASoC: pcm3168a: remove unused variable ASoC: rt5640: remove unused variable ASoC: qcom: fix build error Sugar Zhang (3): ASoC: rockchip: i2s: add default values for registers ASoC: rt5640: add master clock handling for rt5640 ASoC: rockchip: add bindings for spdif controller Vinod Koul (5): ASoC: Intel: Skylake: Update the delay check ASoC: Intel: Skylake: Generate topology name for NHLT table header ASoC: Intel: Skylake: Add i915 enabling in skl probe ASoC: Intel: Skylake: Move module id defines ASoC: Intel: Skylake: Fix whitepsace issues Wolfram Sang (2): ASoC: sh: rcar: core: don't open code of_device_get_match_data() ASoC: sh: rcar: rsrc-card: don't open code of_device_get_match_data() Xing Zheng (1): ASoC: rt5616: Add support sample rate to 192KHz Xiubo Li (3): ASoC: ab8500: remove the useless 'break' after 'return' ASoC: fsl_sai: Use usleep_range() instead of msleep() ASoC: fsl-mpc5200: Use usleep_range() instead of msleep() Zidan Wang (1): ASoC: cs42xx8: fix the noise in the right dac channel with mono playback anish kumar (3): ASoC: Add max98926 codec driver ASoC: Add max9867 codec driver ASoC: max9867: Changed the if-else case to switch case .../devicetree/bindings/regmap/regmap.txt | 11 +- .../devicetree/bindings/sound/adi,adau17x1.txt | 24 + .../devicetree/bindings/sound/fsl-asoc-card.txt | 9 + .../devicetree/bindings/sound/max9867.txt | 17 + .../devicetree/bindings/sound/max98926.txt | 32 + .../bindings/sound/mt8173-rt5650-rt5514.txt | 15 + .../devicetree/bindings/sound/mt8173-rt5650.txt | 15 + .../devicetree/bindings/sound/pcm179x.txt | 11 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 340 ++++++ .../bindings/sound/renesas,rsrc-card.txt | 1 + .../devicetree/bindings/sound/rockchip-i2s.txt | 1 + .../devicetree/bindings/sound/rockchip-spdif.txt | 8 +- Documentation/devicetree/bindings/sound/rt5514.txt | 25 + Documentation/devicetree/bindings/sound/rt5616.txt | 6 + Documentation/devicetree/bindings/sound/rt5640.txt | 3 + .../bindings/sound/sunxi,sun4i-spdif.txt | 39 + .../devicetree/bindings/sound/ti,ads117x.txt | 11 + arch/mips/boot/dts/brcm/bcm6328.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm6368.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7125.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7346.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7358.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7360.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7362.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7420.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7425.dtsi | 2 +- arch/mips/boot/dts/brcm/bcm7435.dtsi | 2 +- drivers/base/regmap/regcache.c | 41 +- drivers/base/regmap/regmap-mmio.c | 259 +++-- drivers/base/regmap/regmap.c | 211 +--- drivers/misc/atmel-ssc.c | 1 + include/drm/drm_edid.h | 12 + include/linux/platform_data/adau17x1.h | 2 +- include/linux/regmap.h | 85 +- include/sound/pcm.h | 2 + include/sound/soc-topology.h | 21 +- include/sound/soc.h | 2 +- include/trace/events/asoc.h | 8 +- sound/core/pcm_misc.c | 30 + sound/soc/atmel/atmel_ssc_dai.c | 27 +- sound/soc/bcm/bcm2835-i2s.c | 284 +---- sound/soc/codecs/Kconfig | 38 +- sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ab8500-codec.c | 9 +- sound/soc/codecs/adau1761-i2c.c | 14 +- sound/soc/codecs/adau1761-spi.c | 14 +- sound/soc/codecs/adau1761.c | 10 +- sound/soc/codecs/adau1781-i2c.c | 10 + sound/soc/codecs/adau1781-spi.c | 10 + sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau17x1.h | 6 +- sound/soc/codecs/ads117x.c | 12 + sound/soc/codecs/arizona.c | 71 +- sound/soc/codecs/arizona.h | 4 +- sound/soc/codecs/cs42l51.c | 8 +- sound/soc/codecs/cs42xx8.c | 10 +- sound/soc/codecs/cs47l24.c | 123 +- sound/soc/codecs/da732x.c | 8 +- sound/soc/codecs/hdac_hdmi.c | 1219 ++++++++++++++++++-- sound/soc/codecs/hdac_hdmi.h | 6 + sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 4 +- sound/soc/codecs/max9867.c | 546 +++++++++ sound/soc/codecs/max9867.h | 83 ++ sound/soc/codecs/max98926.c | 606 ++++++++++ sound/soc/codecs/max98926.h | 848 ++++++++++++++ sound/soc/codecs/nau8825.c | 169 ++- sound/soc/codecs/nau8825.h | 16 +- sound/soc/codecs/pcm179x-i2c.c | 73 ++ sound/soc/codecs/pcm179x-spi.c | 72 ++ sound/soc/codecs/pcm179x.c | 56 +- sound/soc/codecs/pcm179x.h | 9 +- sound/soc/codecs/pcm3168a.c | 8 +- sound/soc/codecs/rt298.c | 7 +- sound/soc/codecs/rt298.h | 8 + sound/soc/codecs/rt5514.c | 982 ++++++++++++++++ sound/soc/codecs/rt5514.h | 252 ++++ sound/soc/codecs/rt5616.c | 414 ++++--- sound/soc/codecs/rt5640.c | 91 +- sound/soc/codecs/rt5640.h | 2 + sound/soc/codecs/rt5645.c | 15 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/ssm4567.c | 5 + sound/soc/codecs/tlv320dac33.c | 9 +- sound/soc/codecs/wl1273.c | 13 +- sound/soc/codecs/wm5102.c | 93 +- sound/soc/codecs/wm5110.c | 75 +- sound/soc/codecs/wm8753.c | 6 +- sound/soc/codecs/wm8904.c | 4 +- sound/soc/codecs/wm8958-dsp2.c | 8 +- sound/soc/codecs/wm8974.c | 93 ++ sound/soc/codecs/wm8983.c | 14 +- sound/soc/codecs/wm8985.c | 14 +- sound/soc/codecs/wm8994.c | 4 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/codecs/wm9081.c | 8 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_adsp.c | 139 ++- sound/soc/codecs/wm_adsp.h | 8 +- sound/soc/davinci/Kconfig | 3 +- sound/soc/davinci/davinci-mcasp.c | 12 + sound/soc/fsl/Kconfig | 4 +- sound/soc/fsl/fsl-asoc-card.c | 26 +- sound/soc/fsl/fsl_sai.c | 3 +- sound/soc/fsl/mpc5200_psc_ac97.c | 3 +- sound/soc/intel/Kconfig | 4 + sound/soc/intel/atom/sst/sst_acpi.c | 4 + sound/soc/intel/atom/sst/sst_ipc.c | 2 - sound/soc/intel/boards/bytcr_rt5640.c | 50 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 17 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 34 +- sound/soc/intel/boards/mfld_machine.c | 16 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 122 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 130 ++- sound/soc/intel/boards/skl_rt286.c | 113 +- sound/soc/intel/common/sst-acpi.h | 3 + sound/soc/intel/common/sst-dsp-priv.h | 1 + sound/soc/intel/common/sst-match-acpi.c | 45 +- sound/soc/intel/skylake/skl-messages.c | 124 +- sound/soc/intel/skylake/skl-nhlt.c | 34 + sound/soc/intel/skylake/skl-pcm.c | 106 +- sound/soc/intel/skylake/skl-sst-dsp.c | 14 +- sound/soc/intel/skylake/skl-sst-dsp.h | 7 +- sound/soc/intel/skylake/skl-sst.c | 3 - sound/soc/intel/skylake/skl-topology.c | 210 +++- sound/soc/intel/skylake/skl-topology.h | 27 + sound/soc/intel/skylake/skl-tplg-interface.h | 3 +- sound/soc/intel/skylake/skl.c | 88 ++ sound/soc/intel/skylake/skl.h | 15 +- sound/soc/mediatek/Kconfig | 22 +- sound/soc/mediatek/Makefile | 2 + sound/soc/mediatek/mt8173-rt5650-rt5514.c | 258 +++++ sound/soc/mediatek/mt8173-rt5650-rt5676.c | 18 +- sound/soc/mediatek/mt8173-rt5650.c | 236 ++++ sound/soc/mediatek/mtk-afe-common.h | 1 + sound/soc/mediatek/mtk-afe-pcm.c | 47 +- sound/soc/mxs/mxs-saif.c | 2 +- sound/soc/omap/n810.c | 18 +- sound/soc/omap/omap-hdmi-audio.c | 1 + sound/soc/omap/rx51.c | 18 +- sound/soc/pxa/brownstone.c | 2 - sound/soc/pxa/corgi.c | 12 +- sound/soc/pxa/magician.c | 6 +- sound/soc/pxa/poodle.c | 12 +- sound/soc/pxa/spitz.c | 12 +- sound/soc/pxa/tosa.c | 12 +- sound/soc/qcom/Kconfig | 7 +- sound/soc/qcom/apq8016_sbc.c | 10 +- sound/soc/qcom/lpass-apq8016.c | 31 +- sound/soc/qcom/lpass-cpu.c | 147 ++- sound/soc/qcom/lpass-ipq806x.c | 11 +- sound/soc/qcom/lpass-lpaif-reg.h | 116 +- sound/soc/qcom/lpass-platform.c | 244 ++-- sound/soc/qcom/lpass.h | 10 +- sound/soc/rockchip/rockchip_i2s.c | 13 + sound/soc/rockchip/rockchip_spdif.c | 17 +- sound/soc/samsung/i2s.c | 21 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c-i2s-v2.h | 2 +- sound/soc/sh/rcar/adg.c | 218 ++-- sound/soc/sh/rcar/cmd.c | 18 +- sound/soc/sh/rcar/core.c | 160 ++- sound/soc/sh/rcar/ctu.c | 281 ++++- sound/soc/sh/rcar/dma.c | 56 +- sound/soc/sh/rcar/dvc.c | 33 +- sound/soc/sh/rcar/gen.c | 56 +- sound/soc/sh/rcar/mix.c | 4 +- sound/soc/sh/rcar/rsnd.h | 115 +- sound/soc/sh/rcar/rsrc-card.c | 27 +- sound/soc/sh/rcar/src.c | 171 ++- sound/soc/sh/rcar/ssi.c | 334 +++--- sound/soc/sh/rcar/ssiu.c | 20 +- sound/soc/soc-core.c | 8 +- sound/soc/soc-dapm.c | 12 +- sound/soc/soc-pcm.c | 24 +- sound/soc/soc-topology.c | 240 ++-- sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun4i-spdif.c | 550 +++++++++ 181 files changed, 10507 insertions(+), 2285 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau17x1.txt create mode 100644 Documentation/devicetree/bindings/sound/max9867.txt create mode 100644 Documentation/devicetree/bindings/sound/max98926.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650.txt create mode 100644 Documentation/devicetree/bindings/sound/rt5514.txt create mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,ads117x.txt create mode 100644 sound/soc/codecs/hdac_hdmi.h create mode 100755 sound/soc/codecs/max9867.c create mode 100755 sound/soc/codecs/max9867.h create mode 100644 sound/soc/codecs/max98926.c create mode 100644 sound/soc/codecs/max98926.h create mode 100644 sound/soc/codecs/pcm179x-i2c.c create mode 100644 sound/soc/codecs/pcm179x-spi.c create mode 100644 sound/soc/codecs/rt5514.c create mode 100644 sound/soc/codecs/rt5514.h create mode 100644 sound/soc/mediatek/mt8173-rt5650-rt5514.c create mode 100644 sound/soc/mediatek/mt8173-rt5650.c create mode 100644 sound/soc/sunxi/sun4i-spdif.c

2 1

Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Mark Brown 14 Mar '16

14 Mar '16

On Mon, Mar 14, 2016 at 11:27:42PM +0530, Jeeja KP wrote: > We discussed this and yes we are looking at ways to solve this longer > term with DAPM but today DAPM doesn't help us in managing DMAs for our > controller. I believe Vinod did discuss this limitation during ELC. > We are starting this discussion internally and will try to solve > this part. We discussed using CODEC-CODEC links for this. If you want to add DMA operations to them then that's a simple matter of programming (which IIRC we did discuss). > Now for us to mange the internal sinks and source how should we go about > modelling them in DAPM and provide hw_params for the configuration? In the same way as anyone not doing it on a SoC does; at the minute that means working out any settings that need configuration by looking at what happens when the signal goes out onto an audio interface. Ideally we'd have a mechanism for propagating digital configuration along audio paths.

1 0

[alsa-devel] [PATCH - tinycompress - brace issue 1/1] tinycompress: fix compilation error: missing brace issue
by francois.gaffie＠intel.com 14 Mar '16

14 Mar '16

From: François Gaffie <francois.gaffie(a)intel.com> Signed-off-by: François Gaffie <francois.gaffie(a)intel.com> diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 68626a4..d35801b 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -273,4 +273,7 @@ int is_compress_ready(struct compress *compress); /* Returns a human readable reason for the last error */ const char *compress_get_error(struct compress *compress); +#if defined(__cplusplus) +} +#endif #endif diff --git a/include/tinycompress/tinymp3.h b/include/tinycompress/tinymp3.h index a709c39..7015320 100644 --- a/include/tinycompress/tinymp3.h +++ b/include/tinycompress/tinymp3.h @@ -102,5 +102,7 @@ enum mp3_stereo_mode { DUAL = 0x02, MONO = 0x03 }; - +#if defined(__cplusplus) +} +#endif #endif -- 2.5.0 --------------------------------------------------------------------- Intel Corporation SAS (French simplified joint stock company) Registered headquarters: "Les Montalets"- 2, rue de Paris, 92196 Meudon Cedex, France Registration Number: 302 456 199 R.C.S. NANTERRE Capital: 4,572,000 Euros This e-mail and any attachments may contain confidential material for the sole use of the intended recipient(s). Any review or distribution by others is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies.

1 0

[alsa-devel] [PATCH] ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type
by Subhransu S. Prusty 14 Mar '16

14 Mar '16

nvhdmi_chmap_cea_alloc_validate_get_type calls itself recursively using chmap ops causing the double fault. Fixed by adding the default validate_get_type handling inside nvdia validate_get_type handler. Link: https://bugzilla.kernel.org/show_bug.cgi?id=114311 Fixes: 67b90cb ("ALSA: hda - Create common chmap object") Reported-by: Andreas Reis <andreas.reis(a)gmail.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Tested-by: Andreas Reis <andreas.reis(a)gmail.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f59e87b..a386a65 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2827,8 +2827,12 @@ static int nvhdmi_chmap_cea_alloc_validate_get_type(struct hdac_chmap *chmap, if (cap->ca_index == 0x00 && channels == 2) return SNDRV_CTL_TLVT_CHMAP_FIXED; - return chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, channels); + /* If the speaker allocation matches the channel count, it is OK. */ + if (cap->channels != channels) + return -1; + + /* all channels are remappable freely */ + return SNDRV_CTL_TLVT_CHMAP_VAR; } static int nvhdmi_chmap_validate(struct hdac_chmap *chmap, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41.
by Vittorio Gambaletta (VittGam) 14 Mar '16

14 Mar '16

The clock measurement on the AC'97 audio card found in the IBM ThinkPad X41 will often fail, so add a quirk entry to fix it. Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=441087 Cc: <stable(a)vger.kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Vittorio Gambaletta <linuxbugs(a)vittgam.net> --- When the clock measurement doesn't fail, the `pos` variable after line 2861 will be around 48367, but when it does fail it will be somewhere between 45000 and 47500. So the bus clock gets set between 48500 and 52000 (!), resulting in faster audio for rates with final clocks below 48000, and distorted audio for rates with final clocks above 48000. The problem is always reproducible on kernel 4.4.5, but very rarely on 3.13. This patch fixes the problem for my computer, and will also speed up module loading a bit as said on Bugzilla, but I've got some questions regarding other hardware... Do we want to protect against overclock for all hardware, by never setting the measured bus clock higher than 48000? Do we actually risk burning or otherwise permanently ruining something here, or not? Does hardware clocked above 48000 exist? Also, is line 2874 actually correct? As `pos` gets higher, the bus clock gets lower, and vice versa. Note that in this function the bus clock is always 48000 before it gets updated here; so the code first computes 48000 * 48000, and then it divides the result by `pos`. It doesn't seem right to me... By the way, the history of that line goes beyond Git (so before kernel 2.6.12-rc2). --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2879,6 +2879,7 @@ static struct snd_pci_quirk intel8x0_clock_list[] = { SND_PCI_QUIRK(0x0e11, 0x008a, "AD1885", 41000), + SND_PCI_QUIRK(0x1014, 0x0581, "AD1981B", 48000), SND_PCI_QUIRK(0x1028, 0x00be, "AD1885", 44100), SND_PCI_QUIRK(0x1028, 0x0177, "AD1980", 48000), SND_PCI_QUIRK(0x1028, 0x01ad, "AD1981B", 48000),

2 1

[alsa-devel] [PATCH] ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda
by Aaron Plattner 14 Mar '16

14 Mar '16

Vendor ID 0x10de0082 is used by a yet-to-be-named GPU chip. This chip also has the 2-ch audio swapping bug, so patch_nvhdmi is appropriate here. Signed-off-by: Aaron Plattner <aplattner(a)nvidia.com> --- Another day, another new chip with the same stupid bug. The hardware folks still claim they're going to fix it someday. sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bcbc4ee10130..651a9ee494db 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3659,6 +3659,7 @@ HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP", patch_nvhdmi), +HDA_CODEC_ENTRY(0x10de0082, "GPU 82 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de0083, "GPU 83 HDMI/DP", patch_nvhdmi), HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch), HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), -- 2.7.3

2 1

[alsa-devel] [PATCH] ASoC: davinci-mcasp: dai format runtime reconfiguration
by Peter Ujfalusi 13 Mar '16

13 Mar '16

In case when the dai format is set via the dai_link the format configuration happens once when the links are probed. If the McASP lose context after this, the information will be lost and McASP will not going to work correctly. To overcome this issue, we save the fmt and set it within hw_params as well. Reported-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1bd86c8e7bc8..b0d1b9f93fa1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -78,6 +78,7 @@ struct davinci_mcasp { u32 fifo_base; struct device *dev; struct snd_pcm_substream *substreams[2]; + unsigned int dai_fmt; /* McASP specific data */ int tdm_slots; @@ -399,6 +400,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, bool fs_pol_rising; bool inv_fs = false; + if (!fmt) + return 0; + pm_runtime_get_sync(mcasp->dev); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: @@ -530,6 +534,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL); mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL); } + + mcasp->dai_fmt = fmt; out: pm_runtime_put(mcasp->dev); return ret; @@ -1048,6 +1054,10 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int period_size = params_period_size(params); int ret; + ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt); + if (ret) + return ret; + /* * If mcasp is BCLK master, and a BCLK divider was not provided by * the machine driver, we need to calculate the ratio. -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: davinci: Kconfig: Update the edma-pcm section's dependency and help
by Peter Ujfalusi 13 Mar '16

13 Mar '16

Instead of depending on individual SoCs make the edma-pcm depend on the eDMA dmaengine driver (TI_EDMA). Update the help text and add DRA7xx family since they have eDMA integrated as well along with sDMA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index e0f6581494bf..a834558fd01e 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -5,7 +5,7 @@ config SND_DAVINCI_SOC config SND_EDMA_SOC tristate "SoC Audio for Texas Instruments chips using eDMA" - depends on SOC_AM33XX || SOC_AM43XX || ARCH_DAVINCI + depends on TI_EDMA select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M here if you want audio support for TI SoC which uses eDMA. @@ -13,6 +13,7 @@ config SND_EDMA_SOC - daVinci devices - AM335x - AM437x/AM438x + - DRA7xx family config SND_DAVINCI_SOC_I2S tristate -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: qcom: fix build error
by Sudip Mukherjee 13 Mar '16

13 Mar '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> While building m32r allmodconfig the build failed with: ERROR: "bad_dma_ops" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! ERROR: "dma_common_mmap" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! To satisfy the dependency CONFIG_SND_SOC_LPASS_PLATFORM should depend on HAS_DMA. Some other configs also needs the dependency on HAS_DMA as they are directly or indirectly selecting SND_SOC_LPASS_PLATFORM. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- m32r build log at: https://travis-ci.org/sudipm-mukherjee/parport/jobs/115245240 sound/soc/qcom/Kconfig | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 3cc252e..8ec9a07 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -11,21 +11,24 @@ config SND_SOC_LPASS_CPU config SND_SOC_LPASS_PLATFORM tristate + depends on HAS_DMA select REGMAP_MMIO config SND_SOC_LPASS_IPQ806X tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_LPASS_APQ8016 tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_STORM tristate "ASoC I2S support for Storm boards" - depends on SND_SOC_QCOM + depends on SND_SOC_QCOM && HAS_DMA select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help @@ -34,7 +37,7 @@ config SND_SOC_STORM config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" - depends on SND_SOC_QCOM + depends on SND_SOC_QCOM && HAS_DMA select SND_SOC_LPASS_APQ8016 help Support for Qualcomm Technologies LPASS audio block in -- 1.9.1

2 1

[alsa-devel] [PATCH v4] ASoC: nau8825: fix issue that pop noise when start playback
by Ben Zhang 13 Mar '16

13 Mar '16

From: John Hsu <KCHSU0(a)nuvoton.com> Reduce pop noise in power up and down sequence when playback. The DAPM widgets graph is reconstructed to ensure the register write sequence at playback matches exactly to the v5 clickless sequence provided by Nuvoton. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 169 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/nau8825.h | 16 ++++- 2 files changed, 141 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c1b87c5..1c87299 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -84,6 +84,7 @@ static const struct nau8825_fll_attr fll_pre_scalar[] = { static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_IIC_ADDR_SET, 0x0 }, { NAU8825_REG_CLK_DIVIDER, 0x0050 }, { NAU8825_REG_FLL1, 0x0 }, { NAU8825_REG_FLL2, 0x3126 }, @@ -158,8 +159,7 @@ static const struct reg_default nau8825_reg_defaults[] = { static bool nau8825_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { - case NAU8825_REG_ENA_CTRL: - case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_ENA_CTRL ... NAU8825_REG_FLL_VCO_RSV: case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: @@ -184,8 +184,7 @@ static bool nau8825_readable_reg(struct device *dev, unsigned int reg) static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) { switch (reg) { - case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: - case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_RESET ... NAU8825_REG_FLL_VCO_RSV: case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: case NAU8825_REG_INTERRUPT_MASK: case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: @@ -227,10 +226,42 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) static int nau8825_pump_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + switch (event) { case SND_SOC_DAPM_POST_PMU: /* Prevent startup click by letting charge pump to ramp up */ msleep(10); + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, NAU8825_JAMNODCLOW); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8825_output_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Disables the TESTDAC to let DAC signal pass through. */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, NAU8825_BIAS_TESTDAC_EN); break; default: return -EINVAL; @@ -316,10 +347,10 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, NAU8825_SAR_ADC_EN_SFT, 0), - SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), - SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), - SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACL Clock", 3, NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR Clock", 3, NAU8825_REG_RDAC, 9, 0, NULL, 0), SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR_SFT, 0), @@ -330,29 +361,48 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), - SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, - 0), + SND_SOC_DAPM_PGA_S("HP amp L", 0, + NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP amp R", 0, + NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, - nau8825_pump_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("Charge Pump", 1, NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_pump_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA("Output Driver R Stage 1", + SND_SOC_DAPM_PGA_S("Output Driver R Stage 1", 4, NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), - SND_SOC_DAPM_PGA("Output Driver L Stage 1", + SND_SOC_DAPM_PGA_S("Output Driver L Stage 1", 4, NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), - SND_SOC_DAPM_PGA("Output Driver R Stage 2", + SND_SOC_DAPM_PGA_S("Output Driver R Stage 2", 5, NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), - SND_SOC_DAPM_PGA("Output Driver L Stage 2", + SND_SOC_DAPM_PGA_S("Output Driver L Stage 2", 5, NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1, + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 6, NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1, + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 6, NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), - SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0), - SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("Output DACL", 7, + NAU8825_REG_CHARGE_PUMP, 8, 1, nau8825_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_S("Output DACR", 7, + NAU8825_REG_CHARGE_PUMP, 9, 1, nau8825_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* HPOL/R are ungrounded by disabling 16 Ohm pull-downs on playback */ + SND_SOC_DAPM_PGA_S("HPOL Pulldown", 8, + NAU8825_REG_HSD_CTRL, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("HPOR Pulldown", 8, + NAU8825_REG_HSD_CTRL, 1, 1, NULL, 0), + + /* High current HPOL/R boost driver */ + SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, + NAU8825_REG_BOOST, 9, 1, NULL, 0), + + /* Class G operation control*/ + SND_SOC_DAPM_PGA_S("Class G", 10, + NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), @@ -375,24 +425,27 @@ static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { {"DACR Mux", "DACR", "DDACR"}, {"HP amp L", NULL, "DACL Mux"}, {"HP amp R", NULL, "DACR Mux"}, - {"HP amp L", NULL, "HP amp power"}, - {"HP amp R", NULL, "HP amp power"}, - {"ADACL", NULL, "HP amp L"}, - {"ADACR", NULL, "HP amp R"}, - {"ADACL", NULL, "ADACL Clock"}, - {"ADACR", NULL, "ADACR Clock"}, - {"Output Driver L Stage 1", NULL, "ADACL"}, - {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Charge Pump", NULL, "HP amp L"}, + {"Charge Pump", NULL, "HP amp R"}, + {"ADACL", NULL, "Charge Pump"}, + {"ADACR", NULL, "Charge Pump"}, + {"ADACL Clock", NULL, "ADACL"}, + {"ADACR Clock", NULL, "ADACR"}, + {"Output Driver L Stage 1", NULL, "ADACL Clock"}, + {"Output Driver R Stage 1", NULL, "ADACR Clock"}, {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, {"Output DACL", NULL, "Output Driver L Stage 3"}, {"Output DACR", NULL, "Output Driver R Stage 3"}, - {"HPOL", NULL, "Output DACL"}, - {"HPOR", NULL, "Output DACR"}, - {"HPOL", NULL, "Charge Pump"}, - {"HPOR", NULL, "Charge Pump"}, + {"HPOL Pulldown", NULL, "Output DACL"}, + {"HPOR Pulldown", NULL, "Output DACR"}, + {"HP Boost Driver", NULL, "HPOL Pulldown"}, + {"HP Boost Driver", NULL, "HPOR Pulldown"}, + {"Class G", NULL, "HP Boost Driver"}, + {"HPOL", NULL, "Class G"}, + {"HPOR", NULL, "Class G"}, }; static int nau8825_hw_params(struct snd_pcm_substream *substream, @@ -659,11 +712,10 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) break; } - if (type & SND_JACK_HEADPHONE) { - /* Unground HPL/R */ - regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); - } - + /* Leaving HPOL/R grounded after jack insert by default. They will be + * ungrounded as part of the widget power up sequence at the beginning + * of playback to reduce pop. + */ return type; } @@ -768,6 +820,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) { struct regmap *regmap = nau8825->regmap; + /* Latch IIC LSB value */ + regmap_write(regmap, NAU8825_REG_IIC_ADDR_SET, 0x0001); /* Enable Bias/Vmid */ regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); @@ -780,10 +834,10 @@ static void nau8825_init_regs(struct nau8825 *nau8825) nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT); /* Disable Boost Driver, Automatic Short circuit protection enable */ regmap_update_bits(regmap, NAU8825_REG_BOOST, - NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | - NAU8825_SHORT_SHUTDOWN_EN, - NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | - NAU8825_SHORT_SHUTDOWN_EN); + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_DIS | + NAU8825_HP_BOOST_G_DIS | NAU8825_SHORT_SHUTDOWN_EN, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_DIS | + NAU8825_HP_BOOST_G_DIS | NAU8825_SHORT_SHUTDOWN_EN); regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, NAU8825_JKDET_OUTPUT_EN, @@ -822,6 +876,35 @@ static void nau8825_init_regs(struct nau8825 *nau8825) NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128); regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128); + /* Disable DACR/L power */ + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL); + /* Enable TESTDAC. This sets the analog DAC inputs to a '0' input + * signal to avoid any glitches due to power up transients in both + * the analog and digital DAC circuit. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDAC_EN, NAU8825_BIAS_TESTDAC_EN); + /* CICCLP off */ + regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, + NAU8825_DAC_CLIP_OFF, NAU8825_DAC_CLIP_OFF); + + /* Class AB bias current to 2x, DAC Capacitor enable MSB/LSB */ + regmap_update_bits(regmap, NAU8825_REG_ANALOG_CONTROL_2, + NAU8825_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8825_DAC_CAPACITOR_MSB | NAU8825_DAC_CAPACITOR_LSB, + NAU8825_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8825_DAC_CAPACITOR_MSB | NAU8825_DAC_CAPACITOR_LSB); + /* Class G timer 64ms */ + regmap_update_bits(regmap, NAU8825_REG_CLASSG_CTRL, + NAU8825_CLASSG_TIMER_MASK, + 0x20 << NAU8825_CLASSG_TIMER_SFT); + /* DAC clock delay 2ns, VREF */ + regmap_update_bits(regmap, NAU8825_REG_RDAC, + NAU8825_RDAC_CLK_DELAY_MASK | NAU8825_RDAC_VREF_MASK, + (0x2 << NAU8825_RDAC_CLK_DELAY_SFT) | + (0x3 << NAU8825_RDAC_VREF_SFT)); } static const struct regmap_config nau8825_regmap_config = { diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb..8ceb5f3 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -14,6 +14,7 @@ #define NAU8825_REG_RESET 0x00 #define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_IIC_ADDR_SET 0x02 #define NAU8825_REG_CLK_DIVIDER 0x03 #define NAU8825_REG_FLL1 0x04 #define NAU8825_REG_FLL2 0x05 @@ -129,7 +130,7 @@ /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) -/* 0 - short to GND, 1 - open */ +/* 0 - open, 1 - short to GND */ #define NAU8825_SPKR_DWN1R (1 << 1) #define NAU8825_SPKR_DWN1L (1 << 0) @@ -251,12 +252,18 @@ /* DACR_CTRL (0x34) */ #define NAU8825_DACR_CH_SEL_SFT 9 +/* CLASSG_CTRL (0x50) */ +#define NAU8825_CLASSG_TIMER_SFT 8 +#define NAU8825_CLASSG_TIMER_MASK (0x3f << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_EN (1 << 0) + /* I2C_DEVICE_ID (0x58) */ #define NAU8825_GPIO2JD1 (1 << 7) #define NAU8825_SOFTWARE_ID_MASK 0x3 #define NAU8825_SOFTWARE_ID_NAU8825 0x0 /* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_TESTDAC_EN (0x3 << 8) #define NAU8825_BIAS_VMID (1 << 6) #define NAU8825_BIAS_VMID_SEL_SFT 4 #define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) @@ -274,6 +281,12 @@ #define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) #define NAU8825_POWERUP_ADCL (1 << 6) +/* RDAC (0x73) */ +#define NAU8825_RDAC_CLK_DELAY_SFT 4 +#define NAU8825_RDAC_CLK_DELAY_MASK (0x7 << NAU8825_RDAC_CLK_DELAY_SFT) +#define NAU8825_RDAC_VREF_SFT 2 +#define NAU8825_RDAC_VREF_MASK (0x3 << NAU8825_RDAC_VREF_SFT) + /* MIC_BIAS (0x74) */ #define NAU8825_MICBIAS_JKSLV (1 << 14) #define NAU8825_MICBIAS_JKR2 (1 << 12) @@ -284,6 +297,7 @@ /* BOOST (0x76) */ #define NAU8825_PRECHARGE_DIS (1 << 13) #define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_DIS (1 << 9) #define NAU8825_HP_BOOST_G_DIS (1 << 8) #define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) -- 2.7.0.rc3.207.g0ac5344

2 1

[alsa-devel] [PATCH] ASoC: omap-hdmi-audio: Support for DRA7xx family
by Peter Ujfalusi 13 Mar '16

13 Mar '16

DRA7xx family is compatible with the OMAP5 HDMI. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-hdmi-audio.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/omap/omap-hdmi-audio.c b/sound/soc/omap/omap-hdmi-audio.c index f83cc2bc0fc4..64425d352962 100644 --- a/sound/soc/omap/omap-hdmi-audio.c +++ b/sound/soc/omap/omap-hdmi-audio.c @@ -345,6 +345,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev) dai_drv = &omap4_hdmi_dai; break; case OMAPDSS_VER_OMAP5: + case OMAPDSS_VER_DRA7xx: dai_drv = &omap5_hdmi_dai; break; default: -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: rsnd: add simplified module explanation
by Kuninori Morimoto 13 Mar '16

13 Mar '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas sound driver user needs to read its datasheet when create DT. But it is difficult to understand, because it has many modules (SRC/CTU/MIX/DVC/SSIU/SSI/AudioDMAC/AudioDMACperiperi), and many features (Asynchronous/Synchronous mode on SRC, CTU matrix, DVC volume settings feature, Multi-SSI/TDM-SSI, etc). This patch adds simplified explanation to help setting/understanding. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 340 +++++++++++++++++++++ sound/soc/sh/rcar/dvc.c | 23 ++ 2 files changed, 363 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 8ee0fa9..c7b29df 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -1,6 +1,337 @@ Renesas R-Car sound +============================================= +* Modules +============================================= + +Renesas R-Car sound is constructed from below modules +(for Gen2 or later) + + SCU : Sampling Rate Converter Unit + - SRC : Sampling Rate Converter + - CMD + - CTU : Channel Transfer Unit + - MIX : Mixer + - DVC : Digital Volume and Mute Function + SSIU : Serial Sound Interface Unit + SSI : Serial Sound Interface + +See detail of each module's channels, connection, limitation on datasheet + +============================================= +* Multi channel +============================================= + +Multi channel is supported by Multi-SSI, or TDM-SSI. + + Multi-SSI : 6ch case, you can use stereo x 3 SSI + TDM-SSI : 6ch case, you can use TDM + +============================================= +* Enable/Disable each modules +============================================= + +See datasheet to check SRC/CTU/MIX/DVC connect-limitation. +DT controls enabling/disabling module. +${LINUX}/arch/arm/boot/dts/r8a7790-lager.dts can be good example. +This is example of + +Playback: [MEM] -> [SRC2] -> [DVC0] -> [SSIU0/SSI0] -> [codec] +Capture: [MEM] <- [DVC1] <- [SRC3] <- [SSIU1/SSI1] <- [codec] + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&ssi0 &src2 &dvc0>; + capture = <&ssi1 &src3 &dvc1>; + }; + }; + }; + +You can use below. +${LINUX}/arch/arm/boot/dts/r8a7790.dts can be good example. + + &src0 &ctu00 &mix0 &dvc0 &ssi0 + &src1 &ctu01 &mix1 &dvc1 &ssi1 + &src2 &ctu02 &ssi2 + &src3 &ctu03 &ssi3 + &src4 &ssi4 + &src5 &ctu10 &ssi5 + &src6 &ctu11 &ssi6 + &src7 &ctu12 &ssi7 + &src8 &ctu13 &ssi8 + &src9 &ssi9 + +============================================= +* SRC (Sampling Rate Converter) +============================================= + + [xx]Hz [yy]Hz + ------> [SRC] ------> + +SRC can convert [xx]Hz to [yy]Hz. Then, it has below 2 modes + + Asynchronous mode: input data / output data are based on different clocks. + you can use this mode on Playback / Capture + Synchronous mode: input data / output data are based on same clocks. + This mode will be used if system doesn't have its input clock, + for example digital TV case. + you can use this mode on Playback + +------------------ +** Asynchronous mode +------------------ + +You need to use "renesas,rsrc-card" sound card for it. +example) + + sound { + compatible = "renesas,rsrc-card"; + ... + /* + * SRC Asynchronous mode setting + * Playback: + * All input data will be converted to 48kHz + * Capture: + * Inputed 48kHz data will be converted to + * system specified Hz + */ + convert-rate = <48000>; + ... + cpu { + sound-dai = <&rcar_sound>; + }; + codec { + ... + }; + }; + +------------------ +** Synchronous mode +------------------ + + > amixer set "SRC Out Rate" on + > aplay xxxx.wav + > amixer set "SRC Out Rate" 48000 + > amixer set "SRC Out Rate" 44100 + +============================================= +* CTU (Channel Transfer Unit) +============================================= + + [xx]ch [yy]ch + ------> [CTU] --------> + +CTU can convert [xx]ch to [yy]ch, or exchange outputed channel. +CTU conversion needs matrix settings. +For more detail information, see below + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - SCU Operation + - CMD Block + - Functional Blocks in CMD + + Renesas R-Car datasheet + - Sampling Rate Converter Unit (SCU) + - Register Description + - CTUn Scale Value exx Register (CTUn_SVxxR) + + ${LINUX}/sound/soc/sh/rcar/ctu.c + - comment of header + +You need to use "renesas,rsrc-card" sound card for it. +example) + + sound { + compatible = "renesas,rsrc-card"; + ... + /* + * CTU setting + * All input data will be converted to 2ch + * as output data + */ + convert-channels = <2>; + ... + cpu { + sound-dai = <&rcar_sound>; + }; + codec { + ... + }; + }; + +Ex) Exchange output channel + Input -> Output + 1ch -> 0ch + 0ch -> 1ch + + example of using matrix + output 0ch = (input 0ch x 0) + (input 1ch x 1) + output 1ch = (input 0ch x 1) + (input 1ch x 0) + + amixer set "CTU Reset" on + amixer set "CTU Pass" 9,10 + amixer set "CTU SV0" 0,4194304 + amixer set "CTU SV1" 4194304,0 + + example of changing connection + amixer set "CTU Reset" on + amixer set "CTU Pass" 2,1 + +============================================= +* MIX (Mixer) +============================================= + +MIX merges 2 sounds path. You can see 2 sound interface on system, +and these sounds will be merged by MIX. + + aplay -D plughw:0,0 xxxx.wav & + aplay -D plughw:0,1 yyyy.wav + +You need to use "renesas,rsrc-card" sound card for it. +Ex) + [MEM] -> [SRC1] -> [CTU02] -+-> [MIX0] -> [DVC0] -> [SSI0] + | + [MEM] -> [SRC2] -> [CTU03] -+ + + sound { + compatible = "renesas,rsrc-card"; + ... + cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + codec { + ... + }; + }; + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; + }; + dai1 { + playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; + }; + }; + }; + +============================================= +* DVC (Digital Volume and Mute Function) +============================================= + +DVC controls Playback/Capture volume. + +Playback Volume + amixer set "DVC Out" 100% + +Capture Volume + amixer set "DVC In" 100% + +Playback Mute + amixer set "DVC Out Mute" on + +Capture Mute + amixer set "DVC In Mute" on + +Volume Ramp + amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" + amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" + amixer set "DVC Out Ramp" on + aplay xxx.wav & + amixer set "DVC Out" 80% // Volume Down + amixer set "DVC Out" 100% // Volume Up + +============================================= +* SSIU (Serial Sound Interface Unit) +============================================= + +There is no DT settings for SSIU, because SSIU will be automatically +selected via SSI. +SSIU can avoid some under/over run error, because it has some buffer. +But you can't use it if SSI was PIO mode. +In DMA mode, you can select not to use SSIU by using "no-busif" on DT. + + &ssi0 { + no-busif; + }; + +============================================= +* SSI (Serial Sound Interface) +============================================= + +** PIO mode + +You can use PIO mode which is for connection check by using. +Note: The system will drop non-SSI modules in PIO mode +even though if DT is selecting other modules. + + &ssi0 { + pio-transfer + }; + +** DMA mode without SSIU + +You can use DMA without SSIU. +Note: under/over run, or noise are likely to occur + + &ssi0 { + no-busif; + }; + +** PIN sharing + +Each SSI can share WS pin. It is based on platform. +This is example if SSI1 want to share WS pin with SSI0 + + &ssi1 { + shared-pin; + }; + +** Multi-SSI + +You can use Multi-SSI. +This is example of SSI0/SSI1/SSI2 (= for 6ch) + + &rcar_sound { + ... + rcar_sound,dai { + dai0 { + playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; + }; + }; + }; + +** TDM-SSI + +You can use TDM with SSI. +This is example of TDM 6ch. +Driver can automatically switches TDM <-> stereo mode in this case. + + rsnd_tdm: sound { + compatible = "simple-audio-card"; + ... + simple-audio-card,cpu { + /* system can use TDM 6ch */ + dai-tdm-slot-num = <6>; + sound-dai = <&rcar_sound>; + }; + simple-audio-card,codec { + ... + }; + }; + + +============================================= Required properties: +============================================= + - compatible : "renesas,rcar_sound-<soctype>", fallbacks "renesas,rcar_sound-gen1" if generation1, and "renesas,rcar_sound-gen2" if generation2 @@ -64,7 +395,10 @@ DAI subnode properties: - playback : list of playback modules - capture : list of capture modules + +============================================= Example: +============================================= rcar_sound: sound@ec500000 { #sound-dai-cells = <1>; @@ -250,7 +584,9 @@ rcar_sound: sound@ec500000 { }; }; +============================================= Example: simple sound card +============================================= rsnd_ak4643: sound { compatible = "simple-audio-card"; @@ -290,7 +626,9 @@ Example: simple sound card shared-pin; }; +============================================= Example: simple sound card for TDM +============================================= rsnd_tdm: sound { compatible = "simple-audio-card"; @@ -309,7 +647,9 @@ Example: simple sound card for TDM }; }; +============================================= Example: simple sound card for Multi channel +============================================= &rcar_sound { pinctrl-0 = <&sound_pins &sound_clk_pins>; diff --git a/sound/soc/sh/rcar/dvc.c b/sound/soc/sh/rcar/dvc.c index 93b11e1..02d971f 100644 --- a/sound/soc/sh/rcar/dvc.c +++ b/sound/soc/sh/rcar/dvc.c @@ -8,6 +8,29 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ + +/* + * Playback Volume + * amixer set "DVC Out" 100% + * + * Capture Volume + * amixer set "DVC In" 100% + * + * Playback Mute + * amixer set "DVC Out Mute" on + * + * Capture Mute + * amixer set "DVC In Mute" on + * + * Volume Ramp + * amixer set "DVC Out Ramp Up Rate" "0.125 dB/64 steps" + * amixer set "DVC Out Ramp Down Rate" "0.125 dB/512 steps" + * amixer set "DVC Out Ramp" on + * aplay xxx.wav & + * amixer set "DVC Out" 80% // Volume Down + * amixer set "DVC Out" 100% // Volume Up + */ + #include "rsnd.h" #define RSND_DVC_NAME_SIZE 16 -- 1.9.1

2 1

[alsa-devel] [PATCH v4 0/6] add IEC958 channel status control helpers
by Arnaud Pouliquen 12 Mar '16

12 Mar '16

V4 - Link pcm control, defined in dai, to PCM device: Discard solution based on "pcm_new" ops for DAI initialisations replace it by an helper function that registers pcm control and links it to pcm device. - Define helpers function to handle 'IEC958 Playback Default' control . add snd_pcm_iec958_ctl_new helper for compatibility with snd_soc_add_dai_pcm_controls implementation . add substream parameter for snd_pcm_create_iec958_ctl function . fixes based on Takashi remarks. V3: - Define helpers function to handle 'IEC958 Playback Default' control - add "pcm_new" ops for DAI initialisations that need pcm runtime context. This patchset is needed to be able to associate control to the PCM device. The iec control is linked to the DAI but also to the PCM device. Furthermore, some platforms like sti support both HDMI and SPDIF outputs in parallel. Logical way to differentiate them is to link them to pcm device index. This patchset is extracted from following RFC http://permalink.gmane.org/gmane.linux.alsa.devel/149876 (sti: add audio interface to the hdmi driver) Notice that "ASoC: hdmi-codec: add IEC control" patch depends on integration of the hdmi-codec driver Patches update: - ALSA: pcm: add IEC958 channel status control helper - make mutex usage mandatory. - Set control index to pcm device id ( needed to be supported by iecset) - ASoC: core: add code to complete dai init after pcm creation - fix dai param of pcm_new ops for cpu dai. - ASoC: sti: use iec channel status control helper new patch: implementation of the helper for sti cpu-dai V2: - patch: ALSA: pcm: add IEC958 channel status control helper - Return 1 instead of 0 in snd_pcm_iec958_put - Add .access field in control structure - I have kept condition on mutex for flexibility (but could be cleaned to force user to use a mutex) V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Arnaud Pouliquen (5): ASoC: core: add snd_soc_add_dai_pcm_controls helper ASoC: sti: use snd_soc_add_dai_pcm_controls helper ALSA: pcm: add IEC958 channel status control helper ASoC: core: allow private data for snd_soc_add_dai_pcm_controls ASoC: sti: use iec channel status control helper include/sound/hdmi-codec.h | 1 + include/sound/pcm_iec958.h | 21 +++++ include/sound/soc-dai.h | 1 + include/sound/soc.h | 3 + sound/core/pcm_iec958.c | 125 ++++++++++++++++++++++++++++++ sound/soc/codecs/hdmi-codec.c | 54 ++++++++++++++++++++++++++++++++----------- sound/soc/soc-core.c | 168 +++++++++++++++++++++++++++++++++++----- sound/soc/sti/Kconfig | 1 + sound/soc/sti/sti_uniperif.c | 57 +++++++------- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 80 +++++-------------- 11 files changed, 386 insertions(+), 126 deletions(-) -- 1.9.1

4 11

[alsa-devel] [PATCH 1/1] aplay/arecord: handle parsing errors of parameter values
by nebelbank＠posteo.de 11 Mar '16

11 Mar '16

From: erwin <nebelbank(a)posteo.de> Signed-off-by: erwin <nebelbank(a)posteo.de> --- It took me a while to find out why the signal from "-d hw:1,0" sounded so unexpected: was from another pair of microphones... diff --git a/aplay/aplay.c b/aplay/aplay.c index 7eacee3..7acaa83 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -427,6 +427,22 @@ enum { OPT_FATAL_ERRORS, }; +static long parse_long(const char *str, int *err) +{ + long val; + char *endptr; + + errno = 0; + val = strtol(str, &endptr, 0); + + if (errno != 0 || *endptr != '\0') + *err = -1; + else + *err = 0; + + return val; +} + int main(int argc, char *argv[]) { int option_index; @@ -558,7 +574,11 @@ int main(int argc, char *argv[]) } break; case 'c': - rhwparams.channels = strtol(optarg, NULL, 0); + rhwparams.channels = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid channels argument '%s'"), optarg); + return 1; + } if (rhwparams.channels < 1 || rhwparams.channels > 256) { error(_("value %i for channels is invalid"), rhwparams.channels); return 1; @@ -585,7 +605,11 @@ int main(int argc, char *argv[]) } break; case 'r': - tmp = strtol(optarg, NULL, 0); + tmp = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid rate argument '%s'"), optarg); + return 1; + } if (tmp < 300) tmp *= 1000; rhwparams.rate = tmp; @@ -595,32 +619,64 @@ int main(int argc, char *argv[]) } break; case 'd': - timelimit = strtol(optarg, NULL, 0); + timelimit = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid duration argument '%s'"), optarg); + return 1; + } break; case 'N': nonblock = 1; open_mode |= SND_PCM_NONBLOCK; break; case 'F': - period_time = strtol(optarg, NULL, 0); + period_time = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid period time argument '%s'"), optarg); + return 1; + } break; case 'B': - buffer_time = strtol(optarg, NULL, 0); + buffer_time = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid buffer time argument '%s'"), optarg); + return 1; + } break; case OPT_PERIOD_SIZE: - period_frames = strtol(optarg, NULL, 0); + period_frames = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid period size argument '%s'"), optarg); + return 1; + } break; case OPT_BUFFER_SIZE: - buffer_frames = strtol(optarg, NULL, 0); + buffer_frames = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid buffer size argument '%s'"), optarg); + return 1; + } break; case 'A': - avail_min = strtol(optarg, NULL, 0); + avail_min = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid min available space argument '%s'"), optarg); + return 1; + } break; case 'R': - start_delay = strtol(optarg, NULL, 0); + start_delay = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid start delay argument '%s'"), optarg); + return 1; + } break; case 'T': - stop_delay = strtol(optarg, NULL, 0); + stop_delay = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid stop delay argument '%s'"), optarg); + return 1; + } break; case 'v': verbose++; @@ -671,7 +727,11 @@ int main(int argc, char *argv[]) test_position = 1; break; case OPT_TEST_COEF: - test_coef = strtol(optarg, NULL, 0); + test_coef = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid test coef argument '%s'"), optarg); + return 1; + } if (test_coef < 1) test_coef = 1; break; @@ -679,7 +739,11 @@ int main(int argc, char *argv[]) test_nowait = 1; break; case OPT_MAX_FILE_TIME: - max_file_time = strtol(optarg, NULL, 0); + max_file_time = parse_long(optarg, &err); + if (err < 0) { + error(_("invalid max file time argument '%s'"), optarg); + return 1; + } break; case OPT_PROCESS_ID_FILE: pidfile_name = optarg; -- 2.5.0

2 3

[alsa-devel] [PATCH] ASoC: wm_adsp: Use correct local length in error message
by Charles Keepax 11 Mar '16

11 Mar '16

Fixes: 44029e9e1290 ("ASoC: wm_adsp: wm_coeff_{read|write}_control should use passed length") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3ee1c72..3c15481 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -807,7 +807,7 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, ret = regmap_raw_read(dsp->regmap, reg, scratch, len); if (ret) { adsp_err(dsp, "Failed to read %zu bytes from %x: %d\n", - ctl->len, reg, ret); + len, reg, ret); kfree(scratch); return ret; } -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: hda - fix the mic mute button and led problem for a Lenovo AIO
by Hui Wang 11 Mar '16

11 Mar '16

This Lenovo ThinkCentre AIO also uses Line2 as mic mute button and uses GPIO2 to control the mic mute led, so applying this quirk can make both the button and led work. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1555912 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 93d2156..4f5ca0b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5556,6 +5556,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), + SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: dice: use structure to represent register parameters instead of array with basic type element
by Takashi Sakamoto 11 Mar '16

11 Mar '16

In dice interface, two blocks of register are accessible via IEEE 1394 asynchronous transaction to represent the number of supported isochronous streams and the number of quadlets for stream information. Current ALSA dice driver uses array with 'unsigned int' element for temporary cache of these information. But using structure is preferable for begin easily comprehensible. This commit applies a local structure for this aim. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 110 +++++++++++++++++++------------------- 1 file changed, 54 insertions(+), 56 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 2077f18..845d5e5 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -12,6 +12,11 @@ #define CALLBACK_TIMEOUT 200 #define NOTIFICATION_TIMEOUT_MS (2 * MSEC_PER_SEC) +struct reg_params { + unsigned int count; + unsigned int size; +}; + const unsigned int snd_dice_rates[SND_DICE_RATES_COUNT] = { /* mode 0 */ [0] = 32000, @@ -65,7 +70,9 @@ static int ensure_phase_lock(struct snd_dice *dice) return 0; } -static int get_register_params(struct snd_dice *dice, unsigned int params[4]) +static int get_register_params(struct snd_dice *dice, + struct reg_params *tx_params, + struct reg_params *rx_params) { __be32 reg[2]; int err; @@ -73,14 +80,16 @@ static int get_register_params(struct snd_dice *dice, unsigned int params[4]) err = snd_dice_transaction_read_tx(dice, TX_NUMBER, reg, sizeof(reg)); if (err < 0) return err; - params[0] = min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS); - params[1] = be32_to_cpu(reg[1]) * 4; + tx_params->count = + min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS); + tx_params->size = be32_to_cpu(reg[1]) * 4; err = snd_dice_transaction_read_rx(dice, RX_NUMBER, reg, sizeof(reg)); if (err < 0) return err; - params[2] = min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS); - params[3] = be32_to_cpu(reg[1]) * 4; + rx_params->count = + min_t(unsigned int, be32_to_cpu(reg[0]), MAX_STREAMS); + rx_params->size = be32_to_cpu(reg[1]) * 4; return 0; } @@ -105,21 +114,21 @@ static void release_resources(struct snd_dice *dice) } static void stop_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, - unsigned int count, unsigned int size) + struct reg_params *params) { __be32 reg; unsigned int i; - for (i = 0; i < count; i++) { + for (i = 0; i < params->count; i++) { reg = cpu_to_be32((u32)-1); if (dir == AMDTP_IN_STREAM) { snd_dice_transaction_write_tx(dice, - size * i + TX_ISOCHRONOUS, - &reg, sizeof(reg)); + params->size * i + TX_ISOCHRONOUS, + &reg, sizeof(reg)); } else { snd_dice_transaction_write_rx(dice, - size * i + RX_ISOCHRONOUS, - &reg, sizeof(reg)); + params->size * i + RX_ISOCHRONOUS, + &reg, sizeof(reg)); } } } @@ -180,8 +189,7 @@ static int keep_resources(struct snd_dice *dice, } static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, - unsigned int rate, unsigned int count, - unsigned int size) + unsigned int rate, struct reg_params *params) { __be32 reg[2]; unsigned int i, pcm_chs, midi_ports; @@ -197,15 +205,15 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, resources = dice->rx_resources; } - for (i = 0; i < count; i++) { + for (i = 0; i < params->count; i++) { if (dir == AMDTP_IN_STREAM) { err = snd_dice_transaction_read_tx(dice, - size * i + TX_NUMBER_AUDIO, - reg, sizeof(reg)); + params->size * i + TX_NUMBER_AUDIO, + reg, sizeof(reg)); } else { err = snd_dice_transaction_read_rx(dice, - size * i + RX_NUMBER_AUDIO, - reg, sizeof(reg)); + params->size * i + RX_NUMBER_AUDIO, + reg, sizeof(reg)); } if (err < 0) return err; @@ -219,12 +227,12 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir, reg[0] = cpu_to_be32(resources[i].channel); if (dir == AMDTP_IN_STREAM) { err = snd_dice_transaction_write_tx(dice, - size * i + TX_ISOCHRONOUS, - reg, sizeof(reg[0])); + params->size * i + TX_ISOCHRONOUS, + reg, sizeof(reg[0])); } else { err = snd_dice_transaction_write_rx(dice, - size * i + RX_ISOCHRONOUS, - reg, sizeof(reg[0])); + params->size * i + RX_ISOCHRONOUS, + reg, sizeof(reg[0])); } if (err < 0) return err; @@ -247,14 +255,14 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) { unsigned int curr_rate; unsigned int i; - unsigned int reg_params[4]; + struct reg_params tx_params, rx_params; bool need_to_start; int err; if (dice->substreams_counter == 0) return -EIO; - err = get_register_params(dice, reg_params); + err = get_register_params(dice, &tx_params, &rx_params); if (err < 0) return err; @@ -271,12 +279,12 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) /* Judge to need to restart streams. */ for (i = 0; i < MAX_STREAMS; i++) { - if (i < reg_params[0]) { + if (i < tx_params.count) { if (amdtp_streaming_error(&dice->tx_stream[i]) || !amdtp_stream_running(&dice->tx_stream[i])) break; } - if (i < reg_params[2]) { + if (i < rx_params.count) { if (amdtp_streaming_error(&dice->rx_stream[i]) || !amdtp_stream_running(&dice->rx_stream[i])) break; @@ -287,10 +295,8 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) if (need_to_start) { /* Stop transmission. */ snd_dice_transaction_clear_enable(dice); - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], - reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], - reg_params[3]); + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); release_resources(dice); err = ensure_phase_lock(dice); @@ -301,12 +307,10 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) } /* Start both streams. */ - err = start_streams(dice, AMDTP_IN_STREAM, rate, reg_params[0], - reg_params[1]); + err = start_streams(dice, AMDTP_IN_STREAM, rate, &tx_params); if (err < 0) goto error; - err = start_streams(dice, AMDTP_OUT_STREAM, rate, reg_params[2], - reg_params[3]); + err = start_streams(dice, AMDTP_OUT_STREAM, rate, &rx_params); if (err < 0) goto error; @@ -318,10 +322,10 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) } for (i = 0; i < MAX_STREAMS; i++) { - if ((i < reg_params[0] && + if ((i < tx_params.count && !amdtp_stream_wait_callback(&dice->tx_stream[i], CALLBACK_TIMEOUT)) || - (i < reg_params[2] && + (i < rx_params.count && !amdtp_stream_wait_callback(&dice->rx_stream[i], CALLBACK_TIMEOUT))) { err = -ETIMEDOUT; @@ -333,8 +337,8 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) return err; error: snd_dice_transaction_clear_enable(dice); - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], reg_params[3]); + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); release_resources(dice); return err; } @@ -346,18 +350,16 @@ error: */ void snd_dice_stream_stop_duplex(struct snd_dice *dice) { - unsigned int reg_params[4]; + struct reg_params tx_params, rx_params; if (dice->substreams_counter > 0) return; snd_dice_transaction_clear_enable(dice); - if (get_register_params(dice, reg_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], - reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], - reg_params[3]); + if (get_register_params(dice, &tx_params, &rx_params) == 0) { + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); } release_resources(dice); @@ -444,15 +446,13 @@ end: void snd_dice_stream_destroy_duplex(struct snd_dice *dice) { - unsigned int reg_params[4]; + struct reg_params tx_params, rx_params; snd_dice_transaction_clear_enable(dice); - if (get_register_params(dice, reg_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], - reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], - reg_params[3]); + if (get_register_params(dice, &tx_params, &rx_params) == 0) { + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); } release_resources(dice); @@ -462,7 +462,7 @@ void snd_dice_stream_destroy_duplex(struct snd_dice *dice) void snd_dice_stream_update_duplex(struct snd_dice *dice) { - unsigned int reg_params[4]; + struct reg_params tx_params, rx_params; /* * On a bus reset, the DICE firmware disables streaming and then goes @@ -474,11 +474,9 @@ void snd_dice_stream_update_duplex(struct snd_dice *dice) */ dice->global_enabled = false; - if (get_register_params(dice, reg_params) == 0) { - stop_streams(dice, AMDTP_IN_STREAM, reg_params[0], - reg_params[1]); - stop_streams(dice, AMDTP_OUT_STREAM, reg_params[2], - reg_params[3]); + if (get_register_params(dice, &tx_params, &rx_params) == 0) { + stop_streams(dice, AMDTP_IN_STREAM, &tx_params); + stop_streams(dice, AMDTP_OUT_STREAM, &rx_params); } } -- 2.7.0

2 2

Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Mark Brown 10 Mar '16

10 Mar '16

On Tue, Feb 16, 2016 at 09:47:47PM +0530, Jeeja KP wrote: > We have no mechanism to configure hw_params for DAIs which are hostless. > Like a loop or DAIs connected to internal sinks/sources. Please propose patches, don't send enormous design documents. They're big, take a long time to read and don't really get us anywhere unless they're extremely clear. > Approach: > 1. Allow params to be defined for other BE dai link also. this > currently applicable only for codec_codec link. Like I keep saying if you're thinking about this in terms of DPCM you're solving the wrong problem - we already support supplying parameters for device<->device links, we already have systems that need to configure things for device<->device links so we can already see that if we're doing something that only works for things that are on SoC and can use DPCM then we're not solving the problem. DPCM works pretty well for small, simple DSPs that are mostly in line in the middle of a standard DAI but it does have limitations which you're now running into. Your systems don't fit well with what DPCM supports naturally. > struct snd_soc_dai_link { > const struct snd_soc_pcm_stream *params; > unsigned int num_params; > > + /* flag to create codec codec based on the flag*/ > + unsigned int codec_link:1; > + This is clearly confused, if we can't figure out what the two devices we are connecting are based on just looking at the devices then that's really not a good sign that our interfaces are sensible and easy to work with.

2 1

[alsa-devel] [PATCH] ALSA: pcm: Avoid "BUG:" string for warnings again
by Takashi Iwai 10 Mar '16

10 Mar '16

The commit [d507941beb1e: ALSA: pcm: Correct PCM BUG error message] made the warning prefix back to "BUG:" due to its previous wrong prefix. But a kernel message containing "BUG:" seems taken as an Oops message wrongly by some brain-dead daemons, and it annoys users in the end. Instead of teaching daemons, change the string again to a more reasonable one. Fixes: 507941beb1e ('ALSA: pcm: Correct PCM BUG error message') Cc: <stable(a)vger.kernel.org> # v3.19+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6b5a811e01a5..3a9b66c6e09c 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -322,7 +322,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, char name[16]; snd_pcm_debug_name(substream, name, sizeof(name)); pcm_err(substream->pcm, - "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n", + "invalid position: %s, pos = %ld, buffer size = %ld, period size = %ld\n", name, pos, runtime->buffer_size, runtime->period_size); } -- 2.7.2

1 0

[alsa-devel] [PATCH - pcm 1/1] pcm: fix 'unable to create IPC shm instance' caused by fork from a thread
by IceBsi 10 Mar '16

10 Mar '16

From: Qing Cai <bsiice(a)msn.com> As stated in manpage SHMCTL(2), shm_nattch is "No. of current attaches" (i.e., number of processes attached to the shared memeory). If an application uses alsa-lib and invokes fork() from a thread of the application, there may be the following execution sequence: 1. execute the following statement: pcm_direct.c:110: dmix->shmptr = shmat(dmix->shmid, 0, 0) (shm_nattch becomes 1) 2. invoke fork() in some thread. (shm_nattch becomes 2) 3. execute the following statement: pcm_direct.c:122: if (buf.shm_nattch == 1) 4. execute the following statement: pcm_direct.c:131: if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) (As stated in manpage SHMGET(2), "When a new shared memory segment is created, its contents are initialized to zero values", so dmix->shmptr->magic is 0) 5. execute the following statements: pcm_direct.c:132: snd_pcm_direct_shm_discard(dmix) pcm_direct.c:133: return -EINVAL The above execution sequence will cause the following error: unable to create IPC shm instance This error causes multimedia application has no sound. This error rarely occurs, probability is about 1%. Signed-off-by: Qing Cai <bsiice(a)msn.com> Signed-off-by: Qing Cai <caiqing(a)neusoft.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index fd3877c..14de734 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -91,13 +91,19 @@ int snd_pcm_direct_semaphore_create_or_connect(snd_pcm_direct_t *dmix) int snd_pcm_direct_shm_create_or_connect(snd_pcm_direct_t *dmix) { struct shmid_ds buf; - int tmpid, err; + int tmpid, err, first_instance = 0; retryget: dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), - IPC_CREAT | dmix->ipc_perm); + dmix->ipc_perm); + if (dmix->shmid < 0) { + if (errno == ENOENT) + if ((dmix->shmid = shmget(dmix->ipc_key, sizeof(snd_pcm_direct_share_t), + IPC_CREAT | IPC_EXCL | dmix->ipc_perm)) != -1) + first_instance = 1; + } err = -errno; - if (dmix->shmid < 0){ + if (dmix->shmid < 0) { if (errno == EINVAL) if ((tmpid = shmget(dmix->ipc_key, 0, dmix->ipc_perm)) != -1) if (!shmctl(tmpid, IPC_STAT, &buf)) @@ -119,7 +125,7 @@ retryget: snd_pcm_direct_shm_discard(dmix); return err; } - if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ + if (first_instance) { /* we're the first user, clear the segment */ memset(dmix->shmptr, 0, sizeof(snd_pcm_direct_share_t)); if (dmix->ipc_gid >= 0) { buf.shm_perm.gid = dmix->ipc_gid; -- 2.7.2

3 3

[alsa-devel] [PATCH] ALSA: hda - Add a sanity check of pin / port mapping on i915 HDMI/DP
by Takashi Iwai 10 Mar '16

10 Mar '16

There is an implicit rule to map between pin NID and port number on Intel HDMI/DP codec: the mapping is fixed only for NID 0x05, 0x06 and 0x07. For avoiding the possible memory corruption, add a sanity check for the NID value and splat WARN_ON() for invalid accesses. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdac_i915.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index f6854dbd7d8d..fb96aead8257 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -126,6 +126,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_get_display_clk); */ static int pin2port(hda_nid_t pin_nid) { + if (WARN_ON(pin_nid < 5 || pin_nid > 7)) + return -1; return pin_nid - 4; } @@ -144,10 +146,14 @@ static int pin2port(hda_nid_t pin_nid) int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate) { struct i915_audio_component *acomp = bus->audio_component; + int port; if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) return -ENODEV; - return acomp->ops->sync_audio_rate(acomp->dev, pin2port(nid), rate); + port = pin2port(nid); + if (port < 0) + return -EINVAL; + return acomp->ops->sync_audio_rate(acomp->dev, port, rate); } EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); @@ -175,11 +181,15 @@ int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes) { struct i915_audio_component *acomp = bus->audio_component; + int port; if (!acomp || !acomp->ops || !acomp->ops->get_eld) return -ENODEV; - return acomp->ops->get_eld(acomp->dev, pin2port(nid), audio_enabled, + port = pin2port(nid); + if (port < 0) + return -EINVAL; + return acomp->ops->get_eld(acomp->dev, port, audio_enabled, buffer, max_bytes); } EXPORT_SYMBOL_GPL(snd_hdac_acomp_get_eld); -- 2.7.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Don't handle ELD notify from invalid port
by Takashi Iwai 10 Mar '16

10 Mar '16

The current Intel HDMI codec driver supports only three fixed ports from port B to port D. However, i915 driver may assign a DP on other ports, e.g. port A, when no eDP is used. This incompatibility is caught later at pin_nid_to_pin_index() and results in a warning message like "HDMI: pin nid 4 not registered" at each time. This patch filters out such invalid events beforehand, so that the kernel won't be too grumbling. Reported-by: Stefan Assmann <sassmann(a)kpanic.de> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_hdmi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bcbc4ee10130..b90e4f3967de 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2432,6 +2432,10 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) struct hda_codec *codec = audio_ptr; int pin_nid = port + 0x04; + /* we assume only from port-B to port-D */ + if (port < 1 || port > 3) + return; + /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ -- 2.7.2

1 0

Re: [alsa-devel] [PATCH v6 5/6] drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding
by Jyri Sarha 10 Mar '16

10 Mar '16

On 03/09/16 15:52, kbuild test robot wrote: > Hi Jyri, > > [auto build test WARNING on drm/drm-next] > [also build test WARNING on v4.5-rc7 next-20160309] > [if your patch is applied to the wrong git tree, please drop us a note to help improving the system] > > url: https://github.com/0day-ci/linux/commits/Jyri-Sarha/Implement-generic-ASoC-… > base: git://people.freedesktop.org/~airlied/linux.git drm-next > config: x86_64-rhel (attached as .config) > reproduce: > # save the attached .config to linux build tree > make ARCH=x86_64 > > All warnings (new ones prefixed by >>): > > drivers/gpu/drm/i2c/tda998x_drv.c: In function 'tda998x_get_audio_ports': >>> drivers/gpu/drm/i2c/tda998x_drv.c:1345:7: warning: 'size' is used uninitialized in this function [-Wuninitialized] > size /= sizeof(u32); > ^ > > vim +/size +1345 drivers/gpu/drm/i2c/tda998x_drv.c > > 1329 if (IS_ERR(priv->audio_pdev)) > 1330 return PTR_ERR(priv->audio_pdev); > 1331 > 1332 return 0; > 1333 } > 1334 > 1335 /* I2C driver functions */ > 1336 > 1337 static int tda998x_get_audio_ports(struct tda998x_priv *priv, > 1338 struct device_node *np) > 1339 { > 1340 const u32 *port_data; > 1341 u32 size; > 1342 int i; > 1343 > 1344 port_data = of_get_property(np, "audio-ports", &size); >> 1345 size /= sizeof(u32); Oops, that is a real bug. I should test if port_data is not null before assuming that the size has been set (/me thinks that he had that line there at some point but then it vanished). Actually initializing size to 0 would be enough, but I think I'll do the both just to be more explicit. And I'll start using PTR_ERR_OR_ZERO() while I am at it. > 1346 > 1347 if (size > 2 * ARRAY_SIZE(priv->audio_port) || size % 2 != 0) { > 1348 dev_err(&priv->hdmi->dev, > 1349 "Bad number of elements in audio-ports dt-property\n"); > 1350 return -EINVAL; > 1351 } > 1352 > 1353 size /= 2; > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation >

1 0

[alsa-devel] [PATCH] ALSA: hda - Apply reboot D3 fix for CX20724 codec, too
by Takashi Iwai 10 Mar '16

10 Mar '16

Just like CX20722, CX7024 codec also requires the power down at reboot in order to reduce the noise at reboot/shutdown. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=113511 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 6122b8ca872f..56fefbd85782 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -204,8 +204,13 @@ static void cx_auto_reboot_notify(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - if (codec->core.vendor_id != 0x14f150f2) + switch (codec->core.vendor_id) { + case 0x14f150f2: /* CX20722 */ + case 0x14f150f4: /* CX20724 */ + break; + default: return; + } /* Turn the CX20722 codec into D3 to avoid spurious noises from the internal speaker during (and after) reboot */ -- 2.7.2

1 0

[alsa-devel] [PATCH] tinycompress: INSTALL: Update the instructions
by Vinod Koul 10 Mar '16

10 Mar '16

Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- INSTALL | 33 +++++++++++++++++++++++++++++++-- 1 file changed, 31 insertions(+), 2 deletions(-) diff --git a/INSTALL b/INSTALL index 2099840756e6..5ac7670b0a7c 100644 --- a/INSTALL +++ b/INSTALL @@ -1,5 +1,34 @@ -Installation Instructions -************************* + + Tinycompress Installation Instructions + ************************************** + +Install from tarball +==================== + +For installation you can use these commands: + + ./configure + make install + + +Compilation from Git +==================== + +gitcompile script allows to build and then install + + ./gitcompile + make install + + +Compilation for Android +======================= + +Android makefile Android.mk is inculded in this project and should be +packaged into Android build system + + +Configure help +============== Copyright (C) 1994-1996, 1999-2002, 2004-2013 Free Software Foundation, Inc. -- 1.9.1

1 0

[alsa-devel] [PATCH 0/6] Support for CherryTrail devices
by Pierre-Louis Bossart 10 Mar '16

10 Mar '16

Fix dependency on DW_DMAC_CORE Fix BIOS issues with better ACPI filtering for active devices Change codec name based on filtered HID Re-use bytcr-rt5640 driver Tested on Tronsmart Ara x5 (CHT, headset only) and Asus T100TA (BYT) to make sure nothing was broken Thanks to Mika Westerberg for his support on ACPI Pierre-Louis Bossart (6): ASoC: Intel: allow for DW_DMAC_CORE as module ASoC: Intel: common: filter ACPI devices with _STA return value ASoC: Intel: common: add translation from HID to codec-name ASoC: Intel: bytcr_rt5640: change quirk position ASoC: Intel: bytcr_rt5640: use HID translation util ASoC: Intel: Atom: add support for CHT w/ RT5640 sound/soc/intel/Kconfig | 4 +-- sound/soc/intel/atom/sst/sst_acpi.c | 4 +++ sound/soc/intel/boards/bytcr_rt5640.c | 50 +++++++++++++++++++++---------- sound/soc/intel/common/sst-acpi.h | 3 ++ sound/soc/intel/common/sst-match-acpi.c | 53 +++++++++++++++++++++++++++++++-- 5 files changed, 94 insertions(+), 20 deletions(-) -- 1.9.1

3 18

[alsa-devel] [PATCH] ASoC: nau8825: extend FLL function
by John Hsu 10 Mar '16

10 Mar '16

Extend FLL clock source from MCLK/BCLK/FS. Modify FLL calculation and parameter setting. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 128 ++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 21 ++++++-- 2 files changed, 108 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index c1b87c5..504c969 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -31,7 +31,7 @@ #include "nau8825.h" #define NAU_FREF_MAX 13500000 -#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 struct nau8825_fll { @@ -874,8 +874,8 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, struct nau8825_fll *fll_param) { - u64 fvco; - unsigned int fref, i; + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by dividing * freq_in by 1, 2, 4, or 8 using FLL pre-scalar. @@ -900,18 +900,23 @@ static int nau8825_calc_fll_param(unsigned int fll_in, unsigned int fs, fll_param->ratio = fll_ratio[i].val; /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. - * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * FDCO must be within the 90MHz - 124MHz or the FFL cannot be * guaranteed across the full range of operation. * FDCO = freq_out * 2 * mclk_src_scaling */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { fvco = 256 * fs * 2 * mclk_src_scaling[i].param; - if (NAU_FVCO_MIN < fvco && fvco < NAU_FVCO_MAX) - break; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } } - if (i == ARRAY_SIZE(mclk_src_scaling)) + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) return -EINVAL; - fll_param->mclk_src = mclk_src_scaling[i].val; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; /* Calculate the FLL 10-bit integer input and the FLL 16-bit fractional * input based on FDCO, FREF and FLL ratio. @@ -926,7 +931,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, struct nau8825_fll *fll_param) { regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, NAU8825_FLL_RATIO_MASK, fll_param->ratio); /* FLL 16-bit fractional input */ @@ -939,10 +945,25 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, - NAU8825_FLL_FILTER_SW_MASK, 0x0000); - /* FLL sigma delta modulator enable */ - regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + NAU8825_FLL_CLK_SW_MASK, NAU8825_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + if (fll_param->fll_frac) { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_FILTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); + } else { + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FTR_SW_MASK, NAU8825_FLL_FTR_SW_ACCU); + regmap_update_bits(nau8825->regmap, + NAU8825_REG_FLL6, NAU8825_SDM_EN, 0); + } } /* freq_out must be 256*Fs in order to achieve the best performance */ @@ -970,6 +991,36 @@ static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, return 0; } +static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) +{ + int ret = 0; + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + return 0; + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + return ret; + } + } + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + return ret; + } + } + + return 0; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -981,29 +1032,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); - - /* We selected MCLK source but the clock itself managed externally */ - if (!nau8825->mclk) - break; - - if (!nau8825->mclk_freq) { - ret = clk_prepare_enable(nau8825->mclk); - if (ret) { - dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); - return ret; - } - } - - if (nau8825->mclk_freq != freq) { - nau8825->mclk_freq = freq; - - freq = clk_round_rate(nau8825->mclk, freq); - ret = clk_set_rate(nau8825->mclk, freq); - if (ret) { - dev_err(nau8825->dev, "Unable to set mclk rate\n"); - return ret; - } - } + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; break; case NAU8825_CLK_INTERNAL: @@ -1011,7 +1042,32 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_DCO_EN); regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + case NAU8825_CLK_FLL_MCLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_MCLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_BLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_BLK); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + break; + case NAU8825_CLK_FLL_FS: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_FS); if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb..8237693 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -112,20 +112,28 @@ /* FLL3 (0x06) */ #define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) +#define NAU8825_FLL_CLK_SRC_SFT 10 +#define NAU8825_FLL_CLK_SRC_MASK (0x3 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_MCLK (0 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_BLK (0x2 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) /* FLL5 (0x08) */ -#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) +#define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8825_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8825_FLL_CLK_SW_MASK (0x1 << 13) +#define NAU8825_FLL_CLK_SW_N2 (0x1 << 13) +#define NAU8825_FLL_CLK_SW_REF (0x0 << 13) +#define NAU8825_FLL_FTR_SW_MASK (0x1 << 12) +#define NAU8825_FLL_FTR_SW_ACCU (0x1 << 12) +#define NAU8825_FLL_FTR_SW_FILTER (0x0 << 12) /* FLL6 (0x9) */ -#define NAU8825_DCO_EN_MASK (0x1 << 15) #define NAU8825_DCO_EN (0x1 << 15) -#define NAU8825_DCO_DIS (0x0 << 15) -#define NAU8825_SDM_EN_MASK (0x1 << 14) #define NAU8825_SDM_EN (0x1 << 14) -#define NAU8825_SDM_DIS (0x0 << 14) /* HSD_CTRL (0xc) */ #define NAU8825_HSD_AUTO_MODE (1 << 6) @@ -306,6 +314,9 @@ enum { NAU8825_CLK_MCLK = 0, NAU8825_CLK_INTERNAL, + NAU8825_CLK_FLL_MCLK, + NAU8825_CLK_FLL_BLK, + NAU8825_CLK_FLL_FS, }; struct nau8825 { -- 2.6.4

2 5

[alsa-devel] [PATCH v6 0/6] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 10 Mar '16

10 Mar '16

There is currently two other patch series[1][2] that depend on the three first (ALSA-) patches of this series. It would be nice to get at least those three applied sooner rather than later, so please tell me if I can do anything to get those in. BTW, I applied the relevant patches from the two series to on top of this one. Obviously the two series conflicted with each other a bit and also the IEC958 helper patch from [2] conflicted with the IEC958 patch from my series (which is weird since Arnaud's series depends on my series...). After resolving the conflicts everything compiled fine, but for some reason neither the 'ELD' or IEC958 Playback Default' appeared on my BBB. I'll try to find time to debug this further at some point. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105509.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105502.html Changes since v5 - Rebased on top of the latest drm-next branch - Allow 32 bit samplewidth in snd_pcm_create_iec958_consumer() and snd_pcm_create_iec958_consumer_hw_params() - Propose new simpler DT binding for tda998x audio - Squash tda998x audio DT binding together with hdmi-codec integration Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 Jyri Sarha (6): ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC hdmi-codec and add audio DT binding ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 18 + arch/arm/boot/dts/am335x-boneblack.dts | 71 +++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 271 ++++++++++++-- include/drm/i2c/tda998x.h | 24 +- include/dt-bindings/display/tda998x.h | 7 + include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 53 ++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 411 +++++++++++++++++++++ 12 files changed, 898 insertions(+), 72 deletions(-) create mode 100644 include/dt-bindings/display/tda998x.h create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

5 16

[alsa-devel] [PATCH] ASoC: hdac_hdmi: Fix infoframe programming
by Subhransu S. Prusty 10 Mar '16

10 Mar '16

Audio infoframe used incorrect buffer, so fix it. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 7851dd2..4f7bd1f 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -303,7 +303,6 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, if (ret < 0) return ret; - dip = (u8 *)&frame; break; case DRM_ELD_CONN_TYPE_DP: @@ -332,9 +331,9 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, /* Fill infoframe. Index auto-incremented */ hdac_hdmi_set_dip_index(hdac, pin_nid, 0x0, 0x0); if (conn_type == DRM_ELD_CONN_TYPE_HDMI) { - for (i = 0; i < sizeof(frame); i++) + for (i = 0; i < sizeof(buffer); i++) snd_hdac_codec_write(&hdac->hdac, pin_nid, 0, - AC_VERB_SET_HDMI_DIP_DATA, dip[i]); + AC_VERB_SET_HDMI_DIP_DATA, buffer[i]); } else { for (i = 0; i < sizeof(dp_ai); i++) snd_hdac_codec_write(&hdac->hdac, pin_nid, 0, -- 1.9.1

2 1

[alsa-devel] [PATCH 0/4] ALSA: dice: enable to handle several streams
by Takashi Sakamoto 09 Mar '16

09 Mar '16

Hi, This patchset updates my previous RFT, and goes for merging. [alsa-devel] [RFT][PATCH 0/4] ALSA: dice: enabled to handle several streams http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105387.html For the previous one, Stefan Richter worked as a tester with his devices (Focusrite Saffire Pro 24 and 40), while current patchset includes changes related to userspace interface. Therefore, I don't add Tested-by. In previous one, ALSA dice driver uses subdevice for userspace to access any PCM substreams. On the other hand, the design of PCM subdevice leaves a bit cumbersome for userspace to handle several subdevices simultaneously in one process. These PCM substreams on a batch of AMDTP streams and it's better for ALSA dice driver to enable one process to handle them with simpler way. Based on this concept, this patchset adds PCM character devices, instead of PCM subdevice. Thus, userspace applications can utilize PCM substreams via individual PCM character devices. In configuration space of alsa-lib, via 'hw:0,0' and 'hw:0,1' if available, like: $ arecord -l ... card 1: Pro26000f5a [Pro26-000f5a], device 0: DICE [Pro26-000f5a] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Pro26000f5a [Pro26-000f5a], device 1: DICE [Pro26-000f5a] Subdevices: 1/1 Subdevice #0: subdevice #0 For the other parts, there's no code changes. Therefore, Stefan's result still has meanings. Changes: * Add additional PCM character device instead of PCM subdevice * Correct string entries to force-pcm * Improve comments Takashi Sakamoto (4): ALSA: dice: have two sets of isochronous resources/streams ALSA: dice: handle whole available isochronous streams ALSA: dice: handle several PCM substreams when any isochronous streams are available ALSA: dice: force to add two pcm devices for listed models sound/firewire/dice/dice-midi.c | 8 +- sound/firewire/dice/dice-pcm.c | 147 +++++++++----- sound/firewire/dice/dice-stream.c | 398 ++++++++++++++++++++++++-------------- sound/firewire/dice/dice.c | 41 ++++ sound/firewire/dice/dice.h | 33 +++- 5 files changed, 424 insertions(+), 203 deletions(-) -- 2.7.0

3 8

[alsa-devel] Scheduled alsa-lib release?
by Takashi Iwai 09 Mar '16

09 Mar '16

Hi, in the previous meeting, there have been requests about the regular "predictable" release of alsa-lib. We thought of aligning with kernel release cycle, judging from the nature of alsa-lib. alsa-lib 1.1.0 was released in the beginning of November, which was just after 4.3 kernel release. Now 4.5 kernel release will be likely in the next week or one week after that. Jaroslav, how about releasing alsa-lib 1.1.1 around that? There are a few pending patches, but they should be available soon. thanks, Takashi

4 6

[alsa-devel] alsa-plugins: pcm_a52.c uses recently removed avcodec_{alloc, free}_frame
by Daniel Kirchner 09 Mar '16

09 Mar '16

pcm_a52.c uses avcodec_alloc_frame and avcodec_free_frame that have been replaced by av_frame_alloc and av_frame_free in recent versions of libavcodec. The following patch should fix this while remaining backwards compatible. --- a/a52/pcm_a52.c.orig 2016-03-04 18:21:12.606805770 +0100 +++ b/a52/pcm_a52.c 2016-03-04 18:34:22.653240501 +0100 @@ -62,6 +62,11 @@ #define AV_CODEC_ID_AC3 CODEC_ID_AC3 #endif +#if LIBAVCODEC_VERSION_INT < 0x371c01 +#define av_frame_alloc avcodec_alloc_frame +#define av_frame_free avcodec_free_frame +#endif + struct a52_ctx { snd_pcm_ioplug_t io; snd_pcm_t *slave; @@ -513,7 +518,7 @@ rec->inbuf = NULL; } #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) - avcodec_free_frame(&rec->frame); + av_frame_free(&rec->frame); #else av_freep(&rec->frame); #endif @@ -557,7 +562,7 @@ { struct a52_ctx *rec = io->private_data; #ifdef USE_AVCODEC_FRAME - rec->frame = avcodec_alloc_frame(); + rec->frame = av_frame_alloc(); if (!rec->frame) return -ENOMEM; if (av_samples_alloc(rec->frame->data, rec->frame->linesize,

2 1

[alsa-devel] [PATCH] Show sequencer sound card numer/PID via aconnect
by Martin Koegler 09 Mar '16

09 Mar '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This patch adds support for displaying the sound card number/PID to aconnect. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- configure.ac | 9 +++++++++ seq/aconnect/aconnect.c | 17 ++++++++++++++++- 2 files changed, 25 insertions(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index bdb133c..9e48190 100644 --- a/configure.ac +++ b/configure.ac @@ -41,6 +41,15 @@ AC_CHECK_HEADERS([alsa/topology.h], [have_topology="yes"], [have_topology="no"], AC_CHECK_HEADERS([samplerate.h], [have_samplerate="yes"], [have_samplerate="no"], [#include <samplerate.h>]) +AC_CHECK_LIB([asound], [snd_seq_client_info_get_card], [HAVE_SEQ_CLIENT_INFO_GET_CARD="yes"]) +if test "$HAVE_SEQ_CLIENT_INFO_GET_CARD" = "yes" ; then + AC_DEFINE([HAVE_SEQ_CLIENT_INFO_GET_CARD], 1, [alsa-lib supports snd_seq_client_info_get_card]) +fi +AC_CHECK_LIB([asound], [snd_seq_client_info_get_pid], [HAVE_SEQ_CLIENT_INFO_GET_PID="yes"]) +if test "$HAVE_SEQ_CLIENT_INFO_GET_PID" = "yes" ; then + AC_DEFINE([HAVE_SEQ_CLIENT_INFO_GET_PID], 1, [alsa-lib supports snd_seq_client_info_get_pid]) +fi + AM_CONDITIONAL(HAVE_PCM, test "$have_pcm" = "yes") AM_CONDITIONAL(HAVE_MIXER, test "$have_mixer" = "yes") AM_CONDITIONAL(HAVE_RAWMIDI, test "$have_rawmidi" = "yes") diff --git a/seq/aconnect/aconnect.c b/seq/aconnect/aconnect.c index 8d6cebb..85a7770 100644 --- a/seq/aconnect/aconnect.c +++ b/seq/aconnect/aconnect.c @@ -166,11 +166,26 @@ static void print_port(snd_seq_t *seq, snd_seq_client_info_t *cinfo, snd_seq_port_info_t *pinfo, int count) { if (! count) { - printf(_("client %d: '%s' [type=%s]\n"), + int card = -1, pid = -1; + + printf(_("client %d: '%s' [type=%s"), snd_seq_client_info_get_client(cinfo), snd_seq_client_info_get_name(cinfo), (snd_seq_client_info_get_type(cinfo) == SND_SEQ_USER_CLIENT ? _("user") : _("kernel"))); + +#ifdef HAVE_SEQ_CLIENT_INFO_GET_CARD + card = snd_seq_client_info_get_card(cinfo); +#endif + if (card != -1) + printf(",card=%d", card); + +#ifdef HAVE_SEQ_CLIENT_INFO_GET_PID + pid = snd_seq_client_info_get_pid(cinfo); +#endif + if (pid != -1) + printf(",pid=%d", pid); + printf("]\n"); } printf(" %3d '%-16s'\n", snd_seq_port_info_get_port(pinfo), -- 2.1.4

2 1

[alsa-devel] alsa-plugins: pcm_a52.c uses recently removed avcodec_{alloc, free}_frame
by Daniel Kirchner 09 Mar '16

09 Mar '16

pcm_a52.c uses avcodec_alloc_frame and avcodec_free_frame that have been replaced by av_frame_alloc and av_frame_free in recent versions of libavcodec. The following patch should fix this while remaining backwards compatible. --- a/a52/pcm_a52.c.orig 2016-03-04 18:21:12.606805770 +0100 +++ b/a52/pcm_a52.c 2016-03-04 18:34:22.653240501 +0100 @@ -62,6 +62,11 @@ #define AV_CODEC_ID_AC3 CODEC_ID_AC3 #endif +#if LIBAVCODEC_VERSION_INT < 0x371c01 +#define av_frame_alloc avcodec_alloc_frame +#define av_frame_free avcodec_free_frame +#endif + struct a52_ctx { snd_pcm_ioplug_t io; snd_pcm_t *slave; @@ -513,7 +518,7 @@ rec->inbuf = NULL; } #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0) - avcodec_free_frame(&rec->frame); + av_frame_free(&rec->frame); #else av_freep(&rec->frame); #endif @@ -557,7 +562,7 @@ { struct a52_ctx *rec = io->private_data; #ifdef USE_AVCODEC_FRAME - rec->frame = avcodec_alloc_frame(); + rec->frame = av_frame_alloc(); if (!rec->frame) return -ENOMEM; if (av_samples_alloc(rec->frame->data, rec->frame->linesize,

1 0

[alsa-devel] [PATCH] ASoC: rt5640: remove unused variable
by Sudip Mukherjee 09 Mar '16

09 Mar '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> We are getting build warning about: sound/soc/codecs/rt5640.c:1892:11: warning: unused variable 'dai_sel' The use of the variable was removed but the variable itself was not removed. Fixes: c467fc0e010b ("ASoC: rt5640: Set PLL src according to source") Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index af9b5f1..e8b5ba0 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1889,7 +1889,7 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); struct rl6231_pll_code pll_code; - int ret, dai_sel; + int ret; if (source == rt5640->pll_src && freq_in == rt5640->pll_in && freq_out == rt5640->pll_out) -- 1.9.1

2 1

[alsa-devel] [PATCH v4] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 08 Mar '16

08 Mar '16

This is an initial imx-wm8958 device-tree-only machine driver. The sound card driver support three dai link, HIFI, VOICE and BT. We can configure the cpu dai from device tree, if present it will create corresponding sound card subdevice. So at least one cpu dai phandle should be set from device tree. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v3->v4: remove the dummy cpu dai, support set cpu dai1/2/3 from device tree. When cpu-dai1/2/3 present it will create croresponding sound card subdevice. At least one cpu dai should be configured. .../devicetree/bindings/sound/imx-audio-wm8958.txt | 56 ++++ arch/arm/configs/imx_v6_v7_defconfig | 1 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 327 +++++++++++++++++++++ 5 files changed, 400 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..c013287 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,56 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + +Optional properties: + + - cpu-dai1 : The phandle of CPU DAI1 controller. At least one + cpu dai is required. + + - cpu-dai2 : The phandle of CPU DAI2 controller. At least one + cpu dai is required. + + - cpu-dai3 : The phandle of CPU DAI3 controller. At least one + cpu dai is required. + + - fsl,cpu-dai1-master : If present, cpu dai1 works as master, and will + provide bit clock and frame clock. Otherwise, + cpu dai1 works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai1 = <&sai1>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; +}; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..7ac5856 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -251,6 +251,7 @@ CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_PHYCORE_AC97=y CONFIG_SND_SOC_EUKREA_TLV320=y +CONFIG_SND_SOC_IMX_WM8958=y CONFIG_SND_SOC_IMX_WM8962=y CONFIG_SND_SOC_IMX_SGTL5000=y CONFIG_SND_SOC_IMX_SPDIF=y diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..0e8a9b2 --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,327 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link dai_link[DAI_LINK_NUM]; + int num_links; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_cpu_dai1_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + if (data->is_cpu_dai1_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (data->is_cpu_dai1_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == HIFI_DAI && data->is_cpu_dai1_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .ops = &imx_voice_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np[DAI_LINK_NUM], *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev[DAI_LINK_NUM]; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + int ret, i; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + for (i = 0; i < DAI_LINK_NUM; i++) { + sprintf(tmp, "cpu-dai%d", i + 1); + cpu_np[i] = of_parse_phandle(np, tmp, 0); + if (cpu_np[i]) { + cpu_pdev[i] = of_find_device_by_node(cpu_np[i]); + if (!cpu_pdev[i]) { + dev_err(&pdev->dev, + "failed to get cpu dai%d platform device\n", i); + ret = -EINVAL; + goto fail; + } + + data->dai_link[data->num_links] = imx_wm8958_dai_link[i]; + data->dai_link[data->num_links].cpu_dai_name = dev_name(&cpu_pdev[i]->dev); + data->dai_link[data->num_links++].platform_of_node = cpu_np[i]; + } + } + + if (!data->num_links) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + /* Hifi dai link support codec master and slave mode */ + if (of_property_read_bool(pdev->dev.of_node, "fsl,cpu-dai1-master")) { + data->is_cpu_dai1_master = true; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } else { + data->is_cpu_dai1_master = false; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(data->mclk[i])) + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = data->num_links; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np[HIFI_DAI]); + of_node_put(cpu_np[VOICE_DAI]); + of_node_put(cpu_np[BT_DAI]); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

3 3

[alsa-devel] [PATCH RFC v5 0/8] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 08 Mar '16

08 Mar '16

There is very little change to this series since the RFC v4, but because of the recent interest on the hdmi-codec patch I decided to send another version of the series. Changes since RFC v4, - Rebased on top of the latest drm-next branch - Split the hdmi-codec abort functionality into a separate patch for better visibility of what it is all about - This does not affect the tda998x patches as the abort functionality is not used - Drop S18_3* formats from I2S_FORMATS and add a comment about formats not supported by HDMI There was also some comments about the DT ports binding for tda998x by Jean-Francois Moine. However I did not have time to come up with alternative approach. All in all my tda998x patches should be considered more as a proof of concept for the hdmi-codec part, rather than a serious attempt to get those patches in. However, all comments to those patch are more than welcome as they will help me to come up with something that could finally get merged. Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 The binding for tda998x is taken from Jean Francois' patch series[1] on the same subject. The implementation of the of-node parsing has some minor changes from my self. Here is what I think at least could or should still be done, but non of that stuff does not sounds critical right now. Missing from tda998x driver side - hdmi_codec_ops audio_startup() implementation for audio abort support - multi channel audio support (I would need specs and preferably some HW to test for this). Missing from ASoC side generic implementation: - channel_allocation handling is completely left for the video side driver, see if ASoC side could help in any way - snd_soc_jack functionality to handle hdmi cable plug/unplug events [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095596.html Best regards, Jyri Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports Jyri Sarha (7): ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add audio abort() callback for video side to use drm/i2c: tda998x: Remove include/sound/tda998x.h and fix graph parsing drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/display/bridge/tda998x.txt | 51 +++ arch/arm/boot/dts/am335x-boneblack.dts | 90 ++++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 300 ++++++++++++--- include/drm/i2c/tda998x.h | 24 +- include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 52 ++- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdmi-codec.c | 411 +++++++++++++++++++++ 11 files changed, 964 insertions(+), 79 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c -- 1.9.1

7 27

Re: [alsa-devel] Testing SAI with mainline
by Fabio Estevam 08 Mar '16

08 Mar '16

On Tue, Mar 8, 2016 at 12:03 AM, Zidan Wang <zidan.wang(a)nxp.com> wrote: > IMX_DMATYPE_SAI is 24. Thanks for the clarification. I have submitted the patches adding SAI support for mx6sx. Now I am trying to get SAI to work on mx7d-sdb. Please see the consolidated changes below. After running audio playback the system hangs. Any ideas? Thanks diff --git a/arch/arm/boot/dts/imx7d-sdb.dts b/arch/arm/boot/dts/imx7d-sdb.dts index b267f79..17320a1 100644 --- a/arch/arm/boot/dts/imx7d-sdb.dts +++ b/arch/arm/boot/dts/imx7d-sdb.dts @@ -95,6 +95,30 @@ regulator-max-microvolt = <1800000>; }; }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "mx7d-sdb-sound-card"; + simple-audio-card,format = "i2s"; + simple-audio-card,bitclock-master = <&dailink_master>; + simple-audio-card,frame-master = <&dailink_master>; + + simple-audio-card,cpu { + sound-dai = <&sai1>; + }; + + dailink_master: simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clks IMX7D_AUDIO_MCLK_ROOT_CLK>; + }; + }; +}; + +&clks { + assigned-clocks = <&clks IMX7D_AUDIO_MCLK_ROOT_SRC>, + <&clks IMX7D_AUDIO_MCLK_ROOT_CLK>; + assigned-clock-parents = <&clks IMX7D_PLL_AUDIO_POST_DIV>; + assigned-clock-rates = <0>, <12288000>; }; &adc1 { @@ -264,14 +288,23 @@ status = "okay"; codec: wm8960@1a { + #sound-dai-cells = <0>; compatible = "wlf,wm8960"; reg = <0x1a>; - clocks = <&clks IMX7D_AUDIO_MCLK_ROOT_CLK>; - clock-names = "mclk"; wlf,shared-lrclk; }; }; +&sai1 { + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai1>; + assigned-clocks = <&clks IMX7D_SAI1_ROOT_SRC>, + <&clks IMX7D_SAI1_ROOT_CLK>; + assigned-clock-parents = <&clks IMX7D_PLL_AUDIO_POST_DIV>; + assigned-clock-rates = <0>, <36864000>; + status = "okay"; +}; + &uart1 { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_uart1>; @@ -390,6 +423,16 @@ >; }; + pinctrl_sai1: sai1grp { + fsl,pins = < + MX7D_PAD_SAI1_MCLK__SAI1_MCLK 0x1f + MX7D_PAD_ENET1_RX_CLK__SAI1_TX_BCLK 0x1f + MX7D_PAD_ENET1_CRS__SAI1_TX_SYNC 0x1f + MX7D_PAD_ENET1_COL__SAI1_TX_DATA0 0x30 + MX7D_PAD_ENET1_TX_CLK__SAI1_RX_DATA0 0x1f + >; + }; + pinctrl_uart1: uart1grp { fsl,pins = < MX7D_PAD_UART1_TX_DATA__UART1_DCE_TX 0x79 diff --git a/arch/arm/boot/dts/imx7d.dtsi b/arch/arm/boot/dts/imx7d.dtsi index b5a50e0..f21b010 100644 --- a/arch/arm/boot/dts/imx7d.dtsi +++ b/arch/arm/boot/dts/imx7d.dtsi @@ -693,6 +693,20 @@ status = "disabled"; }; + sai1: sai@308a0000 { + #sound-dai-cells = <0>; + compatible = "fsl,imx7d-sai"; + reg = <0x308a0000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX7D_SAI1_IPG_CLK>, + <&clks IMX7D_SAI1_ROOT_CLK>, + <&clks IMX7D_CLK_DUMMY>, <&clks IMX7D_CLK_DUMMY>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "rx", "tx"; + dmas = <&sdma 8 24 0>, <&sdma 9 24 0>; + status = "disabled"; + }; + i2c1: i2c@30a20000 { #address-cells = <1>; #size-cells = <0>; @@ -884,6 +898,17 @@ status = "disabled"; }; + sdma: sdma@30bd0000 { + compatible = "fsl,imx7d-sdma"; + reg = <0x30bd0000 0x10000>; + interrupts = <GIC_SPI 2 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clks IMX7D_SDMA_CORE_CLK>, + <&clks IMX7D_AHB_CHANNEL_ROOT_CLK>; + clock-names = "ipg", "ahb"; + #dma-cells = <3>; + fsl,sdma-ram-script-name = "imx/sdma/sdma-imx7d.bin"; + }; + fec1: ethernet@30be0000 { compatible = "fsl,imx7d-fec", "fsl,imx6sx-fec"; reg = <0x30be0000 0x10000>; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..6e69bf55 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -258,6 +258,7 @@ CONFIG_SND_SOC_IMX_MC13783=y CONFIG_SND_SOC_FSL_ASOC_CARD=y CONFIG_SND_SOC_CS42XX8_I2C=y CONFIG_SND_SOC_TLV320AIC3X=y +CONFIG_SND_SOC_WM8960=y CONFIG_SND_SIMPLE_CARD=y CONFIG_USB=y CONFIG_USB_EHCI_HCD=y diff --git a/drivers/clk/imx/clk-imx7d.c b/drivers/clk/imx/clk-imx7d.c index fbb6a8c..c2f0c85 100644 --- a/drivers/clk/imx/clk-imx7d.c +++ b/drivers/clk/imx/clk-imx7d.c @@ -22,6 +22,10 @@ #include "clk.h" +static u32 share_count_sai1; +static u32 share_count_sai2; +static u32 share_count_sai3; + static struct clk *clks[IMX7D_CLK_END]; static const char *arm_a7_sel[] = { "osc", "pll_arm_main_clk", "pll_enet_500m_clk", "pll_dram_main_clk", @@ -777,9 +781,12 @@ static void __init imx7d_clocks_init(struct device_node *ccm_node) clks[IMX7D_MIPI_DSI_ROOT_CLK] = imx_clk_gate2("mipi_dsi_root_clk", "mipi_dsi_post_div", base + 0x4650, 0); clks[IMX7D_MIPI_CSI_ROOT_CLK] = imx_clk_gate2("mipi_csi_root_clk", "mipi_csi_post_div", base + 0x4640, 0); clks[IMX7D_MIPI_DPHY_ROOT_CLK] = imx_clk_gate2("mipi_dphy_root_clk", "mipi_dphy_post_div", base + 0x4660, 0); - clks[IMX7D_SAI1_ROOT_CLK] = imx_clk_gate2("sai1_root_clk", "sai1_post_div", base + 0x48c0, 0); - clks[IMX7D_SAI2_ROOT_CLK] = imx_clk_gate2("sai2_root_clk", "sai2_post_div", base + 0x48d0, 0); - clks[IMX7D_SAI3_ROOT_CLK] = imx_clk_gate2("sai3_root_clk", "sai3_post_div", base + 0x48e0, 0); + clks[IMX7D_SAI1_ROOT_CLK] = imx_clk_gate2_shared("sai1_root_clk", "sai1_post_div", base + 0x48c0, 0, &share_count_sai1); + clks[IMX7D_SAI1_IPG_CLK] = imx_clk_gate2_shared("sai1_ipg_clk", "ipg_root_clk", base + 0x48c0, 0, &share_count_sai1); + clks[IMX7D_SAI2_ROOT_CLK] = imx_clk_gate2_shared("sai2_root_clk", "sai2_post_div", base + 0x48d0, 0, &share_count_sai2); + clks[IMX7D_SAI2_IPG_CLK] = imx_clk_gate2_shared("sai2_ipg_clk", "ipg_root_clk", base + 0x48d0, 0, &share_count_sai2); + clks[IMX7D_SAI3_ROOT_CLK] = imx_clk_gate2_shared("sai3_root_clk", "sai3_post_div", base + 0x48e0, 0, &share_count_sai3); + clks[IMX7D_SAI3_IPG_CLK] = imx_clk_gate2_shared("sai3_ipg_clk", "ipg_root_clk", base + 0x48e0, 0, &share_count_sai3); clks[IMX7D_SPDIF_ROOT_CLK] = imx_clk_gate2("spdif_root_clk", "spdif_post_div", base + 0x44d0, 0); clks[IMX7D_ENET1_REF_ROOT_CLK] = imx_clk_gate2("enet1_ref_root_clk", "enet1_ref_post_div", base + 0x44e0, 0); clks[IMX7D_ENET1_TIME_ROOT_CLK] = imx_clk_gate2("enet1_time_root_clk", "enet1_time_post_div", base + 0x44f0, 0); diff --git a/drivers/dma/imx-sdma.c b/drivers/dma/imx-sdma.c index 0f6fd42..620849f 100644 --- a/drivers/dma/imx-sdma.c +++ b/drivers/dma/imx-sdma.c @@ -477,6 +477,24 @@ static struct sdma_driver_data sdma_imx6q = { .script_addrs = &sdma_script_imx6q, }; +static struct sdma_script_start_addrs sdma_script_imx7d = { + .ap_2_ap_addr = 644, + .uart_2_mcu_addr = 819, + .mcu_2_app_addr = 749, + .uartsh_2_mcu_addr = 1034, + .mcu_2_shp_addr = 962, + .app_2_mcu_addr = 685, + .shp_2_mcu_addr = 893, + .spdif_2_mcu_addr = 1102, + .mcu_2_spdif_addr = 1136, +}; + +static struct sdma_driver_data sdma_imx7d = { + .chnenbl0 = SDMA_CHNENBL0_IMX35, + .num_events = 48, + .script_addrs = &sdma_script_imx7d, +}; + static const struct platform_device_id sdma_devtypes[] = { { .name = "imx25-sdma", @@ -497,6 +515,9 @@ static const struct platform_device_id sdma_devtypes[] = { .name = "imx6q-sdma", .driver_data = (unsigned long)&sdma_imx6q, }, { + .name = "imx7d-sdma", + .driver_data = (unsigned long)&sdma_imx7d, + }, { /* sentinel */ } }; @@ -509,6 +530,7 @@ static const struct of_device_id sdma_dt_ids[] = { { .compatible = "fsl,imx35-sdma", .data = &sdma_imx35, }, { .compatible = "fsl,imx31-sdma", .data = &sdma_imx31, }, { .compatible = "fsl,imx25-sdma", .data = &sdma_imx25, }, + { .compatible = "fsl,imx7d-sdma", .data = &sdma_imx7d, }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, sdma_dt_ids); @@ -1384,6 +1406,7 @@ static void sdma_issue_pending(struct dma_chan *chan) #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V1 34 #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V2 38 #define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V3 41 +#define SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V4 42 static void sdma_add_scripts(struct sdma_engine *sdma, const struct sdma_script_start_addrs *addr) @@ -1433,6 +1456,9 @@ static void sdma_load_firmware(const struct firmware *fw, void *context) case 3: sdma->script_number = SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V3; break; + case 4: + sdma->script_number = SDMA_SCRIPT_ADDRS_ARRAY_SIZE_V4; + break; default: dev_err(sdma->dev, "unknown firmware version\n"); goto err_firmware; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a..62d041c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -909,7 +909,8 @@ config SND_SOC_WM8955 tristate config SND_SOC_WM8960 - tristate +tristate "Wolfson Microelectronics WM8960 CODEC" + depends on I2C config SND_SOC_WM8961 tristate diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..b60497c 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -797,7 +797,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx7d-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -898,6 +899,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx7d-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids);

1 0

[alsa-devel] [PATCH v2 2/3] Provide sequencer sound card number / PID via alsa-lib
by Martin Koegler 08 Mar '16

08 Mar '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This changes adds support for exposing the card number for kernel clients as well as the PID for user client. It supports kernels with and without the required support. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- include/seq.h | 2 ++ include/sound/asequencer.h | 6 ++++-- src/seq/seq.c | 26 ++++++++++++++++++++++++++ src/seq/seq_hw.c | 12 +++++++++++- 4 files changed, 43 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index 9576822..d05940e 100644 --- a/include/seq.h +++ b/include/seq.h @@ -143,6 +143,8 @@ snd_seq_client_type_t snd_seq_client_info_get_type(const snd_seq_client_info_t * const char *snd_seq_client_info_get_name(snd_seq_client_info_t *info); int snd_seq_client_info_get_broadcast_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..3ac70fd 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -24,7 +24,7 @@ /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +356,9 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card; /* RO: card number[kernel] */ + int pid; /* RO: pid[user] */ + char reserved[56]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..4405e68 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,6 +1522,32 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** + * \brief Get the sound card number. + * \param info client_info container + * \return card number or -1 if value is not available. + * + * Only available for SND_SEQ_KERNEL_CLIENT clients. + */ +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) +{ + assert(info); + return info->card; +} + +/** + * \brief Get the owning PID. + * \param info client_info container + * \return pid or -1 if value is not available. + * + * Only available for SND_SEQ_USER_CLIENT clients. + */ +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info) +{ + assert(info); + return info->pid; +} + +/** * \brief (DEPRECATED) Get the event filter bitmap of a client_info container * \param info client_info container * \return NULL if no event filter, or pointer to event filter bitmap diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..24350b6 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int micro_version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,10 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -368,6 +373,10 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + if (hw->micro_version < SNDRV_PROTOCOL_MICRO(SNDRV_SEQ_VERSION_MAX)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -480,6 +489,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->micro_version = SNDRV_PROTOCOL_MICRO(ver); if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { -- 2.1.4

3 8

[alsa-devel] [PATCH 0/9] ASoC: wm*: Fix enum ctl accesses
by Takashi Iwai 08 Mar '16

08 Mar '16

Hi, here is a patchset for correcting the wrong access types to the enum ctl values in various wm* codec drivers. I didn't put Cc to stable. If they might be used for 64 bit architectures, it's a real problem. In that case, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (9): ASoC: wm8753: Fix enum ctl accesses in a wrong type ASoC: wm8904: Fix enum ctl accesses in a wrong type ASoC: wm8958: Fix enum ctl accesses in a wrong type ASoC: wm8983: Fix enum ctl accesses in a wrong type ASoC: wm8985: Fix enum ctl accesses in a wrong type ASoC: wm8994: Fix enum ctl accesses in a wrong type ASoC: wm8996: Fix enum ctl accesses in a wrong type ASoC: wm9081: Fix enum ctl accesses in a wrong type ASoC: wm_adsp: Fix enum ctl accesses in a wrong type sound/soc/codecs/wm8753.c | 6 +++--- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8958-dsp2.c | 8 ++++---- sound/soc/codecs/wm8983.c | 14 +++++++------- sound/soc/codecs/wm8985.c | 14 +++++++------- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 8 ++++---- sound/soc/codecs/wm_adsp.c | 8 ++++---- 9 files changed, 34 insertions(+), 34 deletions(-) -- 2.7.2

3 19

[alsa-devel] [RFC PATCH 0/4] MST DP audio support
by libin.yang＠linux.intel.com 08 Mar '16

08 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> The patchset are tmp patches. The API between audio and gfx is still in discussion. Libin Yang (4): ALSA: hda - codec add DP MST support for connection list ALSA: hda - add DP mst verb support ALSA - hda: add devid support in acom ALSA - hda: add DP MST support drivers/gpu/drm/i915/intel_audio.c | 2 +- include/drm/i915_component.h | 2 +- include/sound/hda_i915.h | 6 +- sound/hda/hdac_i915.c | 7 +- sound/pci/hda/hda_codec.c | 217 ++++++++++++++++++++++++++++++++++--- sound/pci/hda/hda_codec.h | 8 ++ sound/pci/hda/patch_hdmi.c | 172 ++++++++++++++++++++--------- 7 files changed, 340 insertions(+), 74 deletions(-) -- 1.9.1

3 12

Re: [alsa-devel] Testing SAI with mainline
by Fabio Estevam 08 Mar '16

08 Mar '16

Hi Zidan, On Mon, Mar 7, 2016 at 11:49 PM, Zidan Wang <zidan.wang(a)nxp.com> wrote: > You should change the sdma peripheral types of SAI to 24. Thanks! That did the trick. I have also tried with 25, which is what NXP BSP uses, but it did not work. Care to explain why 24 is needed here? I will prepare some patches and submit them tomorrow so that we have SAI working on this board. Thanks for your help

1 0

[alsa-devel] [PATCH v5 22/22] sound/usb: Use Media Controller API to share media resources
by Shuah Khan 07 Mar '16

07 Mar '16

Change ALSA driver to use Media Controller API to share media resources with DVB and V4L2 drivers on a AU0828 media device. Media Controller specific initialization is done after sound card is registered. ALSA creates Media interface and entity function graph nodes for Control, Mixer, PCM Playback, and PCM Capture devices. snd_usb_hw_params() will call Media Controller enable source handler interface to request the media resource. If resource request is granted, it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is returned. Media specific cleanup is done in usb_audio_disconnect(). Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> --- Changes since v4: 1. Renamed media_* interfaces in sound/usb to media_snd* to avoid potential conflicts with the MC interfaces. This addresses Laurent Pinchart's comment. 2. Added Acked-by from Takashi Iwai sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 +++ sound/usb/card.h | 3 + sound/usb/media.c | 318 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 72 +++++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 ++++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 11 files changed, 448 insertions(+), 5 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index a452ad7..d14bf41 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,6 +15,7 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE + select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -22,6 +23,9 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. +config SND_USB_AUDIO_USE_MEDIA_CONTROLLER + bool + config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 2d2d122..8dca3c4 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,6 +15,8 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o +snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o + snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 1f09d95..258cf70 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,6 +66,7 @@ #include "format.h" #include "power.h" #include "stream.h" +#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -561,6 +562,11 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; + if (quirk->media_device) { + /* don't want to fail when media_snd_device_create() fails */ + media_snd_device_create(chip, intf); + } + usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -617,6 +623,14 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } + /* + * Nice to check quirk && quirk->media_device + * need some special handlings. Doesn't look like + * we have access to quirk here + * Acceses mixer_list + */ + media_snd_device_delete(chip); + /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 71778ca..34a0898 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,6 +105,8 @@ struct snd_usb_endpoint { struct list_head list; }; +struct media_ctl; + struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -156,6 +158,7 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ + struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c new file mode 100644 index 0000000..6c0dfd8 --- /dev/null +++ b/sound/usb/media.c @@ -0,0 +1,318 @@ +/* + * media.c - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ + +#include <linux/init.h> +#include <linux/list.h> +#include <linux/mutex.h> +#include <linux/slab.h> +#include <linux/usb.h> + +#include <sound/pcm.h> +#include <sound/core.h> + +#include "usbaudio.h" +#include "card.h" +#include "mixer.h" +#include "media.h" + +static int media_snd_enable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->enable_source) + return mctl->media_dev->enable_source(&mctl->media_entity, + &mctl->media_pipe); + return 0; +} + +static void media_snd_disable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->disable_source) + mctl->media_dev->disable_source(&mctl->media_entity); +} + +int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream) +{ + struct media_device *mdev; + struct media_ctl *mctl; + struct device *pcm_dev = &pcm->streams[stream].dev; + u32 intf_type; + int ret = 0; + u16 mixer_pad; + struct media_entity *entity; + + mdev = subs->stream->chip->media_dev; + if (!mdev) + return -ENODEV; + + if (subs->media_ctl) + return 0; + + /* allocate media_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; + mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; + mixer_pad = 1; + } else { + intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; + mctl->media_pad.flags = MEDIA_PAD_FL_SINK; + mixer_pad = 2; + } + mctl->media_entity.name = pcm->name; + media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) + goto err1; + + mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, + MAJOR(pcm_dev->devt), + MINOR(pcm_dev->devt)); + if (!mctl->intf_devnode) { + ret = -ENOMEM; + goto err2; + } + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &mctl->intf_devnode->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + ret = -ENOMEM; + goto err3; + } + + /* create link between mixer and audio */ + media_device_for_each_entity(entity, mdev) { + switch (entity->function) { + case MEDIA_ENT_F_AUDIO_MIXER: + ret = media_create_pad_link(entity, mixer_pad, + &mctl->media_entity, 0, + MEDIA_LNK_FL_ENABLED); + if (ret) + goto err4; + break; + } + } + + subs->media_ctl = mctl; + return 0; + +err4: + media_remove_intf_link(mctl->intf_link); +err3: + media_devnode_remove(mctl->intf_devnode); +err2: + media_device_unregister_entity(&mctl->media_entity); +err1: + kfree(mctl); + return ret; +} + +void media_snd_stream_delete(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl && mctl->media_dev) { + struct media_device *mdev; + + mdev = subs->stream->chip->media_dev; + if (mdev && media_devnode_is_registered(&mdev->devnode)) { + media_devnode_remove(mctl->intf_devnode); + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + subs->media_ctl = NULL; + } +} + +int media_snd_start_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + return media_snd_enable_source(mctl); + return 0; +} + +void media_snd_stop_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + media_snd_disable_source(mctl); +} + +int media_snd_mixer_init(struct snd_usb_audio *chip) +{ + struct device *ctl_dev = &chip->card->ctl_dev; + struct media_intf_devnode *ctl_intf; + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + struct media_mixer_ctl *mctl; + u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; + int ret; + + if (!mdev) + return -ENODEV; + + ctl_intf = chip->ctl_intf_media_devnode; + if (!ctl_intf) { + ctl_intf = media_devnode_create(mdev, intf_type, 0, + MAJOR(ctl_dev->devt), + MINOR(ctl_dev->devt)); + if (!ctl_intf) + return -ENOMEM; + chip->ctl_intf_media_devnode = ctl_intf; + } + + list_for_each_entry(mixer, &chip->mixer_list, list) { + + if (mixer->media_mixer_ctl) + continue; + + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; + mctl->media_entity.name = chip->card->mixername; + mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; + mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; + mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; + media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, + mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) { + kfree(mctl); + return ret; + } + + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &ctl_intf->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + kfree(mctl); + return -ENOMEM; + } + mctl->intf_devnode = ctl_intf; + mixer->media_mixer_ctl = mctl; + } + return 0; +} + +static void media_snd_mixer_delete(struct snd_usb_audio *chip) +{ + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + + if (!mdev) + return; + + list_for_each_entry(mixer, &chip->mixer_list, list) { + struct media_mixer_ctl *mctl; + + mctl = mixer->media_mixer_ctl; + if (!mixer->media_mixer_ctl) + continue; + + if (media_devnode_is_registered(&mdev->devnode)) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + mixer->media_mixer_ctl = NULL; + } + if (media_devnode_is_registered(&mdev->devnode)) + media_devnode_remove(chip->ctl_intf_media_devnode); + chip->ctl_intf_media_devnode = NULL; +} + +int media_snd_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) +{ + struct media_device *mdev; + struct usb_device *usbdev = interface_to_usbdev(iface); + int ret; + + mdev = media_device_get_devres(&usbdev->dev); + if (!mdev) + return -ENOMEM; + if (!mdev->dev) { + /* register media device */ + mdev->dev = &usbdev->dev; + if (usbdev->product) + strlcpy(mdev->model, usbdev->product, + sizeof(mdev->model)); + if (usbdev->serial) + strlcpy(mdev->serial, usbdev->serial, + sizeof(mdev->serial)); + strcpy(mdev->bus_info, usbdev->devpath); + mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); + media_device_init(mdev); + } + if (!media_devnode_is_registered(&mdev->devnode)) { + ret = media_device_register(mdev); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't register media device. Error: %d\n", + ret); + return ret; + } + } + + /* save media device - avoid lookups */ + chip->media_dev = mdev; + + /* Create media entities for mixer and control dev */ + ret = media_snd_mixer_init(chip); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't create media mixer entities. Error: %d\n", + ret); + + /* clear saved media_dev */ + chip->media_dev = NULL; + + return ret; + } + return 0; +} + +void media_snd_device_delete(struct snd_usb_audio *chip) +{ + struct media_device *mdev = chip->media_dev; + + media_snd_mixer_delete(chip); + + if (mdev) { + if (media_devnode_is_registered(&mdev->devnode)) + media_device_unregister(mdev); + chip->media_dev = NULL; + } +} diff --git a/sound/usb/media.h b/sound/usb/media.h new file mode 100644 index 0000000..1dcdcdc --- /dev/null +++ b/sound/usb/media.h @@ -0,0 +1,72 @@ +/* + * media.h - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ +#ifndef __MEDIA_H + +#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER + +#include <media/media-device.h> +#include <media/media-entity.h> +#include <sound/asound.h> + +struct media_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad; + struct media_pipeline media_pipe; +}; + +/* + * One source pad each for SNDRV_PCM_STREAM_CAPTURE and + * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link + * to AUDIO Source +*/ +#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) + +struct media_mixer_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; + struct media_pipeline media_pipe; +}; + +int media_snd_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface); +void media_snd_device_delete(struct snd_usb_audio *chip); +int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream); +void media_snd_stream_delete(struct snd_usb_substream *subs); +int media_snd_start_pipeline(struct snd_usb_substream *subs); +void media_snd_stop_pipeline(struct snd_usb_substream *subs); +#else +static inline int media_snd_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) + { return 0; } +static inline void media_snd_device_delete(struct snd_usb_audio *chip) { } +static inline int media_snd_stream_init(struct snd_usb_substream *subs, + struct snd_pcm *pcm, int stream) + { return 0; } +static inline void media_snd_stream_delete(struct snd_usb_substream *subs) { } +static inline int media_snd_start_pipeline(struct snd_usb_substream *subs) + { return 0; } +static inline void media_snd_stop_pipeline(struct snd_usb_substream *subs) { } +#endif +#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 3417ef3..f378944 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,6 +3,8 @@ #include <sound/info.h> +struct media_mixer_ctl; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -22,6 +24,7 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; + struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9245f52..b0370d5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,6 +35,7 @@ #include "pcm.h" #include "clock.h" #include "power.h" +#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -715,10 +716,14 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; + ret = media_snd_start_pipeline(subs); + if (ret) + return ret; + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - return ret; + goto err_ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -732,22 +737,27 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - return -EINVAL; + ret = -EINVAL; + goto err_ret; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; + +err_ret: + media_snd_stop_pipeline(subs); + return ret; } /* @@ -759,6 +769,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; + media_snd_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1219,6 +1230,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; + int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1232,7 +1244,12 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - return setup_hw_info(runtime, subs); + ret = setup_hw_info(runtime, subs); + if (ret == 0) + ret = media_snd_stream_init(subs, as->pcm, direction); + if (ret) + snd_usb_autosuspend(subs->stream->chip); + return ret; } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1241,6 +1258,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); + media_snd_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c60a776..9d087b1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,6 +2886,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + .media_device = 1, \ } \ } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index c4dc577..51258a1 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,6 +36,7 @@ #include "format.h" #include "clock.h" #include "stream.h" +#include "media.h" /* * free a substream @@ -52,6 +53,7 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); + media_snd_stream_delete(subs); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b665d85..a161c7c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,6 +30,9 @@ * */ +struct media_device; +struct media_intf_devnode; + struct snd_usb_audio { int index; struct usb_device *dev; @@ -60,6 +63,8 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ + struct media_device *media_dev; + struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -110,6 +115,7 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; + bool media_device; const void *data; }; -- 2.5.0

3 8

[alsa-devel] Testing SAI with mainline
by Fabio Estevam 07 Mar '16

07 Mar '16

Hi, I am trying to get SAI to work on a mx6sx sdb board. Here are the changes I made: http://pastebin.com/aq32B24v I can see the driver getting probed: [ 2.198276] fsl-asoc-card sound: wm8962 <-> 21d4000.sai mapping ok ,but when attempting a playback it fails: $ aplay clarinet.wav [ 110.843619] fsl-sai 21d4000.sai: isr: Transmit underrun detected [ 110.849841] fsl-sai 21d4000.sai: isr: Transmit underrun detected [ 110.855995] fsl-sai 21d4000.sai: isr: Transmit underrun detected (repeats forever). Any ideas as to how to get SAI working on this platform? Zidan, How have you tested SAI when you usptream the driver? Thanks

2 2

[alsa-devel] [PATCH] ALSA: hda - Fix unexpected resume through regmap code path
by Takashi Iwai 07 Mar '16

07 Mar '16

HD-audio driver has a mechanism to trigger the runtime resume automatically at accessing the verbs. This auto-resume, however, causes the mutex deadlock when invoked from the regmap handler since the regmap keeps the mutex while auto-resuming. For avoiding that, there is some tricky check in the HDA regmap handler to return -EAGAIN error to back-off when the codec is powered down. Then the caller of regmap r/w will retry after properly turning on the codec power. This works in most cases, but there seems a slight race between the codec power check and the actual on-demand auto-resume trigger. This resulted in the lockdep splat, eventually leading to a real deadlock. This patch tries to address the race window by getting the runtime PM refcount at the check time using pm_runtime_get_if_in_use(). With this call, we can keep the power on only when the codec has been already turned on, and back off if not. For keeping the code consistency, the code touching the runtime PM is stored in hdac_device.c although it's used only locally in hdac_regmap.c. Reported-by: Jiri Slaby <jslaby(a)suse.cz> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 2 ++ sound/hda/hdac_device.c | 16 ++++++++++++ sound/hda/hdac_regmap.c | 69 ++++++++++++++++++++++++++++++++----------------- 3 files changed, 64 insertions(+), 23 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index c21c38ce7450..93e63c56f48f 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -168,11 +168,13 @@ int snd_hdac_power_up(struct hdac_device *codec); int snd_hdac_power_down(struct hdac_device *codec); int snd_hdac_power_up_pm(struct hdac_device *codec); int snd_hdac_power_down_pm(struct hdac_device *codec); +int snd_hdac_keep_power_up(struct hdac_device *codec); #else static inline int snd_hdac_power_up(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_up_pm(struct hdac_device *codec) { return 0; } static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; } +static inline int snd_hdac_keep_power_up(struct hdac_device *codec) { return 0; } #endif /* diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index e361024eabb6..d1a4d6973330 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -611,6 +611,22 @@ int snd_hdac_power_up_pm(struct hdac_device *codec) } EXPORT_SYMBOL_GPL(snd_hdac_power_up_pm); +/* like snd_hdac_power_up_pm(), but only increment the pm count when + * already powered up. Returns -1 if not powered up, 1 if incremented + * or 0 if unchanged. Only used in hdac_regmap.c + */ +int snd_hdac_keep_power_up(struct hdac_device *codec) +{ + if (!atomic_inc_not_zero(&codec->in_pm)) { + int ret = pm_runtime_get_if_in_use(&codec->dev); + if (!ret) + return -1; + if (ret < 0) + return 0; + } + return 1; +} + /** * snd_hdac_power_down_pm - power down the codec * @codec: the codec object diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index eb8f7c30cb09..bdbcd6b75ff6 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -21,13 +21,16 @@ #include <sound/hdaudio.h> #include <sound/hda_regmap.h> -#ifdef CONFIG_PM -#define codec_is_running(codec) \ - (atomic_read(&(codec)->in_pm) || \ - !pm_runtime_suspended(&(codec)->dev)) -#else -#define codec_is_running(codec) true -#endif +static int codec_pm_lock(struct hdac_device *codec) +{ + return snd_hdac_keep_power_up(codec); +} + +static void codec_pm_unlock(struct hdac_device *codec, int lock) +{ + if (lock == 1) + snd_hdac_power_down_pm(codec); +} #define get_verb(reg) (((reg) >> 8) & 0xfff) @@ -238,20 +241,28 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) struct hdac_device *codec = context; int verb = get_verb(reg); int err; + int pm_lock = 0; - if (!codec_is_running(codec) && verb != AC_VERB_GET_POWER_STATE) - return -EAGAIN; + if (verb != AC_VERB_GET_POWER_STATE) { + pm_lock = codec_pm_lock(codec); + if (pm_lock < 0) + return -EAGAIN; + } reg |= (codec->addr << 28); - if (is_stereo_amp_verb(reg)) - return hda_reg_read_stereo_amp(codec, reg, val); - if (verb == AC_VERB_GET_PROC_COEF) - return hda_reg_read_coef(codec, reg, val); + if (is_stereo_amp_verb(reg)) { + err = hda_reg_read_stereo_amp(codec, reg, val); + goto out; + } + if (verb == AC_VERB_GET_PROC_COEF) { + err = hda_reg_read_coef(codec, reg, val); + goto out; + } if ((verb & 0x700) == AC_VERB_SET_AMP_GAIN_MUTE) reg &= ~AC_AMP_FAKE_MUTE; err = snd_hdac_exec_verb(codec, reg, 0, val); if (err < 0) - return err; + goto out; /* special handling for asymmetric reads */ if (verb == AC_VERB_GET_POWER_STATE) { if (*val & AC_PWRST_ERROR) @@ -259,7 +270,9 @@ static int hda_reg_read(void *context, unsigned int reg, unsigned int *val) else /* take only the actual state */ *val = (*val >> 4) & 0x0f; } - return 0; + out: + codec_pm_unlock(codec, pm_lock); + return err; } static int hda_reg_write(void *context, unsigned int reg, unsigned int val) @@ -267,6 +280,7 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) struct hdac_device *codec = context; unsigned int verb; int i, bytes, err; + int pm_lock = 0; if (codec->caps_overwriting) return 0; @@ -275,14 +289,21 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) reg |= (codec->addr << 28); verb = get_verb(reg); - if (!codec_is_running(codec) && verb != AC_VERB_SET_POWER_STATE) - return codec->lazy_cache ? 0 : -EAGAIN; + if (verb != AC_VERB_SET_POWER_STATE) { + pm_lock = codec_pm_lock(codec); + if (pm_lock < 0) + return codec->lazy_cache ? 0 : -EAGAIN; + } - if (is_stereo_amp_verb(reg)) - return hda_reg_write_stereo_amp(codec, reg, val); + if (is_stereo_amp_verb(reg)) { + err = hda_reg_write_stereo_amp(codec, reg, val); + goto out; + } - if (verb == AC_VERB_SET_PROC_COEF) - return hda_reg_write_coef(codec, reg, val); + if (verb == AC_VERB_SET_PROC_COEF) { + err = hda_reg_write_coef(codec, reg, val); + goto out; + } switch (verb & 0xf00) { case AC_VERB_SET_AMP_GAIN_MUTE: @@ -319,10 +340,12 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) reg |= (verb + i) << 8 | ((val >> (8 * i)) & 0xff); err = snd_hdac_exec_verb(codec, reg, 0, NULL); if (err < 0) - return err; + goto out; } - return 0; + out: + codec_pm_unlock(codec, pm_lock); + return err; } static const struct regmap_config hda_regmap_cfg = { -- 2.7.2

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[alsa-devel] [PATCH v3 0/7] ALSA: hda: Move chmap helpers to core
by Subhransu S. Prusty 07 Mar '16

07 Mar '16

The HDMI channel map helper APIs and control registration can be reused by ASoC skylake HDMI drivers. A common chmap object is created, channel map control callback handlers, helper APIs and data structures are moved to the core. Driver object store an instance of this chmap object to register chmap controls and program channel allocation for HDMI infoframe. Driver registers ops for specific operations. changes in v3: - Namespace managed as per hda frmwrk requirement - Address review comments in v2 Subhransu S. Prusty (7): ALSA: hda - Create common chmap object ALSA: hda - Register chmap obj as priv data instead of codec ALSA: hda - Use hdac name space for CEA spk alloc structure ALSA: hda - Add hdmi chmap verb programming ops to chmap object ALSA: hda - chmap helper args modified to use generic hdac objs. ALSA: hda - Move chmap support helpers/ops to core ALSA: hda - Use snd_hdac namespace prefix for chmap exported APIs include/sound/hda_chmap.h | 76 +++++ sound/hda/Makefile | 2 +- sound/hda/hdmi_chmap.c | 791 +++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_eld.c | 31 +- sound/pci/hda/patch_hdmi.c | 824 +++++---------------------------------------- 5 files changed, 950 insertions(+), 774 deletions(-) create mode 100644 include/sound/hda_chmap.h create mode 100644 sound/hda/hdmi_chmap.c -- 1.9.1

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Re: [alsa-devel] [PATCH 4/4] ALSA: dice: force to add two pcm devices for listed models
by Takashi Sakamoto 07 Mar '16

07 Mar '16

On Mar 7 2016 22:43, Stefan Richter wrote: > On Mar 07 Takashi Sakamoto wrote: >> Thanks for your report. Totally: >> * The patched snd-dice still works fine with Saffire Pro 24. >> * The pached snd-dice successfully manages several isochronous streams >> without code bugs (i.e. kernel NULL pointer dereference or >> continuous locking) with Saffire Pro 24/40. >> * The pached snd-dice still has mute issue with Saffire Pro 40. > > I concur. While I was only able to perform just a few tests yet, there > doesn't appear to be any regression, and this patchset adds support for > the previously inaccessible channels at the second transmitter and > receiver. > >> Anyway, the result cannot block applying this patchset for 4.6. I'll >> post it in this night for merging. > > Thanks. I posted. Thanks for your test ;) >> Well, about the mute issue, I have no idea to fix it at all. At least, >> for my test units, the patched snd-dice works fine without such issues. >> Therefore, it's nearly model-dependent issue and I have no way to do for >> it. FYI, I tested with: >> * OHCI 1394 host controllers >> * VIA VT6315 >> * TI XIO2213 >> * Dice II units >> * TC Electronic ImpactTwin >> * TC Electronic Konnekt24D >> * TC Electronic Konnekt8 >> * Dice Jr. units >> * TC Electronic DesktopKonnect6 >> * Dice Mini units >> * Focusrite Saffire Pro 26 > > FWIW, Saffire PRO 24 and 40 are based on Dice Junior according to FFADO's > device database. If desired I could open up my 24 and 40 to check. If you don't mind it, please. (Because once opening your unit, usually, you cannot get repair service from sellers.) And I wrote wrong for Dice Jr./Mini units. Correctly: * Dice Jr. units * TC Electronic DesktopKonnect6 * Focusrite Saffire Pro 26 * I have no units with Dice Mini (Oh...). Anyway, all of my units with Dice Jr. works fine. So we can judge that Pro 40 has its own quirk which we don't know. > Furthermore FYI, the FFADO userspace streaming driver has got the following > two known issues with the PRO 40: > http://subversion.ffado.org/ticket/326 > playback at 96 kHz is choppy (but fine at 44.1, 48, 88.2 kHz; > reproduced by multiple users, 100% reproducible by myself too) > http://subversion.ffado.org/ticket/372 > MIDI playback problem (perhaps dependent on firmware revision; > cannot reproduce this myself due to lack of MIDI equipment) Hm. With a lack of something like packet dump or detailed case-investigation, I can do nothing for the unit. I have no fashion to lost my time for meaningless talks, sorry. For our information, I put your result of hinawa-dice-common-cui from Saffire Pro 40. (meaningless lines are omitted.) $ ./hinawa-dice-common-cui 00130e040140dd28 current-status owner address: ffc1000100000000 global enabled: False clock status: locked: 1 nominal: 48000 clock source detection: locked: slip: measured rate: 48000 dice version: 1.0.12.0 latest notification: 0x00000010 clock source: Internal sampling rate: 48000 nickname: Pro40-00dd28 Transmitted streams: Stream 0: iso-channel: 4294967295 speed: 2 pcm: 10 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: IP 1 1: IP 2 2: IP 3 3: IP 4 4: IP 5 5: IP 6 6: IP 7 7: IP 8 8: SPDIF L 9: SPDIF R Stream 1: iso-channel: 4294967295 speed: 2 pcm: 10 midi: 0 iec60958: caps: 0 enable: 0 formation: 0: ADAT 1 1: ADAT 2 2: ADAT 3 3: ADAT 4 4: ADAT 5 5: ADAT 6 6: ADAT 7 7: ADAT 8 8: Loop 1 9: Loop 2 Received streams: Stream 0: iso-channel: 4294967295 start: 0 pcm: 12 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: Mon 1 1: Mon 2 2: Line 3 3: Line 4 4: Line 5 5: Line 6 6: Line 7 7: Line 8 8: Line 9 9: Line 10 10: SPDIF L 11: SPDIF R Stream 1: iso-channel: 4294967295 start: 0 pcm: 8 midi: 0 iec60958: caps: 0 enable: 0 formation: 0: ADAT 1 1: ADAT 2 2: ADAT 3 3: ADAT 4 4: ADAT 5 5: ADAT 6 6: ADAT 7 7: ADAT 8 Synchronization infomation: clock-source: Internal locked: 1 sampling-rate: 48000 ADAT: data-bits: 0 no-data: 1 $ ./hinawa-dice-common-cui 00130e040140dd28 sampling-rate set 96000 $ ./hinawa-dice-common-cui 00130e040140dd28 current-status owner address: ffc1000100000000 global enabled: False clock status: locked: 1 nominal: 96000 clock source detection: locked: slip: measured rate: 96000 dice version: 1.0.12.0 latest notification: 0x00000010 clock source: Internal sampling rate: 96000 nickname: Pro40-00dd28 Transmitted streams: Stream 0: iso-channel: 4294967295 speed: 2 pcm: 16 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: IP 1 1: IP 2 2: IP 3 3: IP 4 4: IP 5 5: IP 6 6: IP 7 7: IP 8 8: SPDIF L 9: SPDIF R 10: ADAT 1 11: ADAT 2 12: ADAT 3 13: ADAT 4 14: Loop 1 15: Loop 2 Received streams: Stream 0: iso-channel: 4294967295 start: 0 pcm: 16 midi: 1 iec60958: caps: 0 enable: 0 formation: 0: Mon 1 1: Mon 2 2: Line 3 3: Line 4 4: Line 5 5: Line 6 6: Line 7 7: Line 8 8: Line 9 9: Line 10 10: SPDIF L 11: SPDIF R 12: ADAT 1 13: ADAT 2 14: ADAT 3 15: ADAT 4 Synchronization infomation: clock-source: Internal locked: 1 sampling-rate: 96000 ADAT: data-bits: 0 no-data: 1 Regards Takashi Sakamoto

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[alsa-devel] [PATCH] ASoC: cs35l32: avoid uninitialized variable access
by Arnd Bergmann 07 Mar '16

07 Mar '16

gcc warns about the possibilty of accessing a property read from devicetree in cs35l32_i2c_probe() when it has not been initialized because CONFIG_OF is disabled: sound/soc/codecs/cs35l32.c: In function 'cs35l32_i2c_probe': sound/soc/codecs/cs35l32.c:278:2: warning: 'val' may be used uninitialized in this function [-Wmaybe-uninitialized] The code is actually correct because it checks the dev->of_node variable first and we know this is NULL here when CONFIG_OF is disabled, but Russell King noticed that it's broken when we probe the device using DT, and the properties are absent. The code already has some checking for incorrect values, and I keep that checking unchanged here, but add an additional check for an error returned by the property accessor functions that now gets handled the same way as incorrect data in the properties. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cs35l32.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 44c30fe3e315..c490dc74121b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -274,21 +274,24 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, if (of_property_read_u32(np, "cirrus,sdout-share", &val) >= 0) pdata->sdout_share = val; - of_property_read_u32(np, "cirrus,boost-manager", &val); + if (of_property_read_u32(np, "cirrus,boost-manager", &val)) + val = -1u; + switch (val) { case CS35L32_BOOST_MGR_AUTO: case CS35L32_BOOST_MGR_AUTO_AUDIO: case CS35L32_BOOST_MGR_BYPASS: case CS35L32_BOOST_MGR_FIXED: - pdata->boost_mng = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,boost-manager DT value %d\n", val); pdata->boost_mng = CS35L32_BOOST_MGR_BYPASS; } - of_property_read_u32(np, "cirrus,sdout-datacfg", &val); + if (of_property_read_u32(np, "cirrus,sdout-datacfg", &val)) + val = -1u; switch (val) { case CS35L32_DATA_CFG_LR_VP: case CS35L32_DATA_CFG_LR_STAT: @@ -296,13 +299,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_DATA_CFG_LR_VPSTAT: pdata->sdout_datacfg = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,sdout-datacfg DT value %d\n", val); pdata->sdout_datacfg = CS35L32_DATA_CFG_LR; } - of_property_read_u32(np, "cirrus,battery-threshold", &val); + if (of_property_read_u32(np, "cirrus,battery-threshold", &val)) + val = -1u; switch (val) { case CS35L32_BATT_THRESH_3_1V: case CS35L32_BATT_THRESH_3_2V: @@ -310,13 +315,15 @@ static int cs35l32_handle_of_data(struct i2c_client *i2c_client, case CS35L32_BATT_THRESH_3_4V: pdata->batt_thresh = val; break; + case -1u: default: dev_err(&i2c_client->dev, "Wrong cirrus,battery-threshold DT value %d\n", val); pdata->batt_thresh = CS35L32_BATT_THRESH_3_3V; } - of_property_read_u32(np, "cirrus,battery-recovery", &val); + if (of_property_read_u32(np, "cirrus,battery-recovery", &val)) + val = -1u; switch (val) { case CS35L32_BATT_RECOV_3_1V: case CS35L32_BATT_RECOV_3_2V: -- 2.7.0

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[alsa-devel] [GIT PULL] ASoC fixes for v4.5
by Mark Brown 07 Mar '16

07 Mar '16

The following changes since commit fc77dbd34c5c99bce46d40a2491937c3bcbd10af: Linux 4.5-rc6 (2016-02-28 08:41:20 -0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.5-rc6 for you to fetch changes up to 02db17ff376893f65f45974b0897e70e6695034e: Merge remote-tracking branches 'asoc/fix/wm8985', 'asoc/fix/wm8994', 'asoc/fix/wm8996', 'asoc/fix/wm9081' and 'asoc/fix/wm9713' into asoc-linus (2016-03-05 21:27:13 +0900) ---------------------------------------------------------------- ASoC: Fixes for v4.5 This is far too big a set of fixes for this late in the release cycle but the overwhelming bulk is essentially the same simple fix from Takashi for a cut'n'pasted 64 bit cleanliness issue in the userspace interface where drivers were accessing things using the wrong element in a union which worked OK on 32 bit platforms as the correct element happened to be aligned the same way but with 64 bit platforms ABIs are different and the two members of the union are laid out in different places. They aren't all tagged to stable since some of these chips have vanishingly little chance of being used in 64 bit systems. The other changes are: - A fix for Qualcomm devices to work on big endian systems. The original change is actually correct but triggered a bug in regmap which is too invasive to fix for this cycle and can be worked around by just letting regmap pick the default. - A fix for the Samsung I2S driver locking which wasn't using IRQ safe spinlocks when it needed to. - A fix for the new Intel Sky Lake driver forgetting that C pointer arithmetic takes the type of the pointer into consideration. - A revert of a change to the FSL SSI driver that broke some systems. - A fix for the cleanup path of the wm9713 driver. - A fix for some incorrect register definitions in the ADAU17x1 driver that caused misclocking in some configurations. - A fix for the tracepoints for jack detection to avoid using an internal field of the core jack structure which is no longer present in all configurations. - A fix for another of the new Intel drivers which tried to write to a string literal. ---------------------------------------------------------------- Alan (1): ASoC: Intel: Skylake: fix pointer scaling Andreas Irestål (1): ASoC: adau17x1: Fix incorrect BCLK ratio definitions Arnd Bergmann (1): ASoC: trace: fix printing jack name Carlo Caione (1): ASoC: cht_bsw_rt5645: Fix writing to string literal Charles Keepax (1): ASoC: samsung: Use IRQ safe spin lock calls Fabio Estevam (1): ASoC: fsl_ssi: Go back to explicit register defaults Mark Brown (6): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/ab8500', 'asoc/fix/adau17x1', 'asoc/fix/adsp', 'asoc/fix/cs42l51', 'asoc/fix/da732x' and 'asoc/fix/fsl-ssi' into asoc-linus Merge remote-tracking branches 'asoc/fix/jack', 'asoc/fix/max98088', 'asoc/fix/max98095', 'asoc/fix/omap', 'asoc/fix/pxa' and 'asoc/fix/qcom-be' into asoc-linus Merge remote-tracking branches 'asoc/fix/samsung', 'asoc/fix/tlv320dac33', 'asoc/fix/wl1273', 'asoc/fix/wm8753', 'asoc/fix/wm8904', 'asoc/fix/wm8958' and 'asoc/fix/wm8983' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8985', 'asoc/fix/wm8994', 'asoc/fix/wm8996', 'asoc/fix/wm9081' and 'asoc/fix/wm9713' into asoc-linus Robert Jarzmik (1): ASoC: wm9713: fix regmap free path Stephen Boyd (1): ASoC: qcom: Don't specify LE device endianness Takashi Iwai (25): ASoC: dapm: Fix ctl value accesses in a wrong type ASoC: pxa: corgi: Fix enum ctl accesses in a wrong type ASoC: pxa: magician: Fix enum ctl accesses in a wrong type ASoC: pxa: poodle: Fix enum ctl accesses in a wrong type ASoC: pxa: spitz: Fix enum ctl accesses in a wrong type ASoC: pxa: tosa: Fix enum ctl accesses in a wrong type ASoC: omap: n810: Fix enum ctl accesses in a wrong type ASoC: omap: rx51: Fix enum ctl accesses in a wrong type ASoC: intel: mfld: Fix enum ctl accesses in a wrong type ASoC: cs42l51: Fix enum ctl accesses in a wrong type ASoC: da732x: Fix enum ctl accesses in a wrong type ASoC: ab8500: Fix enum ctl accesses in a wrong type ASoC: max98088: Fix enum ctl accesses in a wrong type ASoC: max98095: Fix enum ctl accesses in a wrong type ASoC: tlv320dac33: Fix enum ctl accesses in a wrong type ASoC: wl1273: Fix enum ctl accesses in a wrong type ASoC: wm8753: Fix enum ctl accesses in a wrong type ASoC: wm8904: Fix enum ctl accesses in a wrong type ASoC: wm8958: Fix enum ctl accesses in a wrong type ASoC: wm8983: Fix enum ctl accesses in a wrong type ASoC: wm8985: Fix enum ctl accesses in a wrong type ASoC: wm8994: Fix enum ctl accesses in a wrong type ASoC: wm8996: Fix enum ctl accesses in a wrong type ASoC: wm9081: Fix enum ctl accesses in a wrong type ASoC: wm_adsp: Fix enum ctl accesses in a wrong type include/trace/events/asoc.h | 8 +++---- sound/soc/codecs/ab8500-codec.c | 8 +++---- sound/soc/codecs/adau17x1.h | 6 ++--- sound/soc/codecs/cs42l51.c | 8 +++---- sound/soc/codecs/da732x.c | 8 +++---- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/codecs/tlv320dac33.c | 9 ++++--- sound/soc/codecs/wl1273.c | 13 +++++----- sound/soc/codecs/wm8753.c | 6 ++--- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8958-dsp2.c | 8 +++---- sound/soc/codecs/wm8983.c | 14 +++++------ sound/soc/codecs/wm8985.c | 14 +++++------ sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 8 +++---- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wm_adsp.c | 8 +++---- sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++----------------- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 ++++- sound/soc/intel/boards/mfld_machine.c | 16 ++++++------- sound/soc/intel/skylake/skl-topology.c | 2 +- sound/soc/omap/n810.c | 18 +++++++------- sound/soc/omap/rx51.c | 18 +++++++------- sound/soc/pxa/corgi.c | 12 +++++----- sound/soc/pxa/magician.c | 6 ++--- sound/soc/pxa/poodle.c | 12 +++++----- sound/soc/pxa/spitz.c | 12 +++++----- sound/soc/pxa/tosa.c | 12 +++++----- sound/soc/qcom/lpass-cpu.c | 1 - sound/soc/samsung/i2s.c | 21 ++++++++++------- sound/soc/soc-dapm.c | 8 +++---- 33 files changed, 162 insertions(+), 160 deletions(-)

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Re: [alsa-devel] ERROR: "isa_io_base" [sound/pci/vx222/snd-vx222.ko] undefined!
by Tomi Valkeinen 07 Mar '16

07 Mar '16

On 06/03/16 14:38, kbuild test robot wrote: > Hi Nicolai, > > FYI, the error/warning still remains. > > tree: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git master > head: 67944024c1cdd897e49a09b0d6af3ea38d1388ca > commit: 48fd8ecf29e3c1199432e173858f5ca4bc529738 framebuffer: disable vgacon on microblaze arch > date: 7 months ago > config: microblaze-allmodconfig (attached as .config) > reproduce: > wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross > chmod +x ~/bin/make.cross > git checkout 48fd8ecf29e3c1199432e173858f5ca4bc529738 > # save the attached .config to linux build tree > make.cross ARCH=microblaze > > All errors (new ones prefixed by >>): > >>> ERROR: "isa_io_base" [sound/pci/vx222/snd-vx222.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/trident/snd-trident.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-via82xx.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-via82xx-modem.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-sonicvibes.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-maestro3.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-fm801.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-es1968.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-es1938.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-ens1371.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-ens1370.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-cmipci.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-azt3328.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/snd-als300.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/riptide/snd-riptide.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/pcxhr/snd-pcxhr.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/oxygen/snd-oxygen-lib.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/lx6464es/snd-lx6464es.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/ice1712/snd-ice1724.ko] undefined! >>> ERROR: "isa_io_base" [sound/pci/ice1712/snd-ice1712.ko] undefined! I didn't look at this closely, but sounds like disabling vgacon on microblaze made sound drivers fail because they or microblaze does not enable something in the kernel config. Tomi

3 3

[alsa-devel] [PATCH] ASoC: rt5616: add missing mute control for HPVOL
by John Lin 07 Mar '16

07 Mar '16

Add missing kcontrol for HPVOL mute control. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5616.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3704d07..f527b5b 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -309,6 +309,8 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { /* Headphone Output Volume */ SOC_DOUBLE("HP Playback Switch", RT5616_HP_VOL, RT5616_L_MUTE_SFT, RT5616_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("HPVOL Playback Switch", RT5616_HP_VOL, + RT5616_VOL_L_SFT, RT5616_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5616_HP_VOL, RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rsrc-card: tidyup convert-channels explain
by Kuninori Morimoto 07 Mar '16

07 Mar '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt index 5abebf7..255ece3 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt @@ -30,7 +30,7 @@ Optional subnode properties: - frame-inversion : bool property. Add this if the dai-link uses frame clock inversion. - convert-rate : platform specified sampling rate convert -- convert-channels : platform specified channel size convert +- convert-channels : platform specified converted channel size (2 - 8 ch) - audio-prefix : see audio-routing - audio-routing : A list of the connections between audio components. Each entry is a pair of strings, the first being the connection's sink, -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5640: Set PLL src according to source
by Bard Liao 07 Mar '16

07 Mar '16

rt5640_set_dai_pll set pll source according to given source and dai id. However, the pll source should be set according to given source only. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6cd84fb..863c190 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1902,21 +1902,12 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_MCLK); break; case RT5640_PLL1_S_BCLK1: + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); + break; case RT5640_PLL1_S_BCLK2: - dai_sel = get_sdp_info(codec, dai->id); - if (dai_sel < 0) { - dev_err(codec->dev, - "Failed to get sdp info: %d\n", dai_sel); - return -EINVAL; - } - if (dai_sel & RT5640_U_IF1) { - snd_soc_update_bits(codec, RT5640_GLB_CLK, - RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK1); - } - if (dai_sel & RT5640_U_IF2) { - snd_soc_update_bits(codec, RT5640_GLB_CLK, - RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); - } + snd_soc_update_bits(codec, RT5640_GLB_CLK, + RT5640_PLL1_SRC_MASK, RT5640_PLL1_SRC_BCLK2); break; default: dev_err(codec->dev, "Unknown PLL source %d\n", source); -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH 0/7] ASoC: rsnd: update SRC activates conditions
by Kuninori Morimoto 07 Mar '16

07 Mar '16

Hi Mark Current Renesas R-Car sound SRC had cared only Playback, because we had heard that it can't use under Capture. But it was telephone-game. accurately, SRC has Sync/Async mode, and it can't use Sync mode when Capture. And, SRC activates conditions is related to CMD, not DVC. These patches update these. Kuninori Morimoto (7): ASoC: rsnd: don't enable SRC Sync mode when Capture with CMD ASoC: rsnd: SRC sync mode is not related to clock master mode ASoC: rsnd: SRC function parameter uses "mod" instead of "src" ASoC: rsnd: rename rsnd_enable_sync_convert() to rsnd_src_sync_is_enabled() ASoC: rsnd: add rsnd_src_get_in/out_rate() ASoC: rsnd: SRC TIMSEL support for Capture ASoC: rsnd: CMD TIMSEL support for Capture sound/soc/sh/rcar/adg.c | 209 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----------------------------------------------------------- sound/soc/sh/rcar/rsnd.h | 18 ++++++------ sound/soc/sh/rcar/src.c | 95 +++++++++++++++++++++++++++++++------------------------------ sound/soc/sh/rcar/ssi.c | 5 ++-- 4 files changed, 177 insertions(+), 150 deletions(-) Best regards --- Kuninori Morimoto

2 14

[alsa-devel] [RFT][PATCH 0/4] ALSA: dice: enabled to handle several streams
by Takashi Sakamoto 07 Mar '16

07 Mar '16

Hi, This patchset adds support for whole available isochronous streams to ALSA dice driver. As a result, for some models, additional PCM substreams are available via subdevice of ALSA PCM character device. In configuration space of alsa-lib, it's represented like 'hw:0,0,1'(card/device/subdevice). It's soon in the last week of this developing cycle, while I hope to finish my work for ALSA dice driver by this patchset. Unfortunately, this patchset adds huge changes and includes possibilities of regression. I'm happy to receive your reports to test with any of your Dice units. For testers, I push whole patches to 'dice-several-streams' branch in my private repository. The changed driver is available from this branch. I recommend to use DKMS to maintain the driver, instead of installing by your own. https://github.com/takaswie/snd-firewire-improve/tree/dice-several-streams To check whether your units support several isochronous streams, please read output from /proc/asound/cardX/dice. When seeing 'tx1' (input) or 'rx1' (output), the additional subdevices are available. For example, $ arecord -l ... card 1: Pro26000f5a [Pro26-000f5a], device 0: DICE [Pro26-000f5a] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 In above case, you can use two subdevices for PCM capture, thus: * arecord -D plughw:1,0,0 file1.wav * arecord -D plughw:1,0,1 file2.wav In the middle of next week, I plan to post this patchset again for merging. So it's nice to test till then. Regards Takashi Sakamoto (4): ALSA: dice: have two sets of isochronous resources/streams ALSA: dice: handle whole available isochronous streams ALSA: dice: handle several PCM substreams when any isochronous streams are available ALSA: dice: force to add two pcm devices for listed models sound/firewire/dice/dice-midi.c | 8 +- sound/firewire/dice/dice-pcm.c | 110 +++++++---- sound/firewire/dice/dice-stream.c | 400 ++++++++++++++++++++++++-------------- sound/firewire/dice/dice.c | 41 ++++ sound/firewire/dice/dice.h | 33 +++- 5 files changed, 401 insertions(+), 191 deletions(-) -- 2.7.0

2 10

[alsa-devel] [PATCH] ASoC: rt5616: add kcontrol for headphone channel switch
by John Lin 07 Mar '16

07 Mar '16

Add kcontrol for headphone channel switch. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5616.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index 3704d07..b370993 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -309,6 +309,8 @@ static const struct snd_kcontrol_new rt5616_snd_controls[] = { /* Headphone Output Volume */ SOC_DOUBLE("HP Playback Switch", RT5616_HP_VOL, RT5616_L_MUTE_SFT, RT5616_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("HP Channel Switch", RT5616_HP_VOL, + RT5616_VOL_L_SFT, RT5616_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5616_HP_VOL, RT5616_L_VOL_SFT, RT5616_R_VOL_SFT, 39, 1, out_vol_tlv), /* OUTPUT Control */ -- 1.8.1.1.439.g50a6b54

3 6

[alsa-devel] [PATCH] Revert "ASoC: fsl_ssi: remove explicit register defaults"
by Fabio Estevam 07 Mar '16

07 Mar '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit 5c408fee2546 ("ASoC: fsl_ssi: remove explicit register defaults") causes the driver to fail to probe: fsl-ssi-dai 2028000.ssi: No cache defaults, reading back from HW fsl-ssi-dai 2028000.ssi: Failed to init register map fsl-ssi-dai: probe of 2028000.ssi failed with error -22 , so revert this commit. Cc: Maciej S. Szmigiero <mail(a)maciej.szmigiero.name> Reported-by: Mika Penttilä <mika.penttila(a)nextfour.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Maciej, As this currently breaks SSI on 4.5-rc we should better revert it for this cycle. We can then add it back to 4.7 when Mark's patch 3245d460a1eb55 ("regmap: cache: Fall back to register by register read for cache defaults") will be available and then the driver will work fine with your patch. sound/soc/fsl/fsl_ssi.c | 42 ++++++++++++++++++++---------------------- 1 file changed, 20 insertions(+), 22 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ed8de10..40dfd8a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -112,6 +112,20 @@ struct fsl_ssi_rxtx_reg_val { struct fsl_ssi_reg_val tx; }; +static const struct reg_default fsl_ssi_reg_defaults[] = { + {CCSR_SSI_SCR, 0x00000000}, + {CCSR_SSI_SIER, 0x00003003}, + {CCSR_SSI_STCR, 0x00000200}, + {CCSR_SSI_SRCR, 0x00000200}, + {CCSR_SSI_STCCR, 0x00040000}, + {CCSR_SSI_SRCCR, 0x00040000}, + {CCSR_SSI_SACNT, 0x00000000}, + {CCSR_SSI_STMSK, 0x00000000}, + {CCSR_SSI_SRMSK, 0x00000000}, + {CCSR_SSI_SACCEN, 0x00000000}, + {CCSR_SSI_SACCDIS, 0x00000000}, +}; + static bool fsl_ssi_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -176,7 +190,8 @@ static const struct regmap_config fsl_ssi_regconfig = { .val_bits = 32, .reg_stride = 4, .val_format_endian = REGMAP_ENDIAN_NATIVE, - .num_reg_defaults_raw = CCSR_SSI_SACCDIS / sizeof(uint32_t) + 1, + .reg_defaults = fsl_ssi_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(fsl_ssi_reg_defaults), .readable_reg = fsl_ssi_readable_reg, .volatile_reg = fsl_ssi_volatile_reg, .precious_reg = fsl_ssi_precious_reg, @@ -186,7 +201,6 @@ static const struct regmap_config fsl_ssi_regconfig = { struct fsl_ssi_soc_data { bool imx; - bool imx21regs; /* imx21-class SSI - no SACC{ST,EN,DIS} regs */ bool offline_config; u32 sisr_write_mask; }; @@ -289,7 +303,6 @@ static struct fsl_ssi_soc_data fsl_ssi_mpc8610 = { static struct fsl_ssi_soc_data fsl_ssi_imx21 = { .imx = true, - .imx21regs = true, .offline_config = true, .sisr_write_mask = 0, }; @@ -573,12 +586,8 @@ static void fsl_ssi_setup_ac97(struct fsl_ssi_private *ssi_private) */ regmap_write(regs, CCSR_SSI_SACNT, CCSR_SSI_SACNT_AC97EN | CCSR_SSI_SACNT_FV); - - /* no SACC{ST,EN,DIS} regs on imx21-class SSI */ - if (!ssi_private->soc->imx21regs) { - regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); - regmap_write(regs, CCSR_SSI_SACCEN, 0x300); - } + regmap_write(regs, CCSR_SSI_SACCDIS, 0xff); + regmap_write(regs, CCSR_SSI_SACCEN, 0x300); /* * Enable SSI, Transmit and Receive. AC97 has to communicate with the @@ -1388,7 +1397,6 @@ static int fsl_ssi_probe(struct platform_device *pdev) struct resource *res; void __iomem *iomem; char name[64]; - struct regmap_config regconfig = fsl_ssi_regconfig; of_id = of_match_device(fsl_ssi_ids, &pdev->dev); if (!of_id || !of_id->data) @@ -1436,25 +1444,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) return PTR_ERR(iomem); ssi_private->ssi_phys = res->start; - if (ssi_private->soc->imx21regs) { - /* - * According to datasheet imx21-class SSI - * don't have SACC{ST,EN,DIS} regs. - */ - regconfig.max_register = CCSR_SSI_SRMSK; - regconfig.num_reg_defaults_raw = - CCSR_SSI_SRMSK / sizeof(uint32_t) + 1; - } - ret = of_property_match_string(np, "clock-names", "ipg"); if (ret < 0) { ssi_private->has_ipg_clk_name = false; ssi_private->regs = devm_regmap_init_mmio(&pdev->dev, iomem, - &regconfig); + &fsl_ssi_regconfig); } else { ssi_private->has_ipg_clk_name = true; ssi_private->regs = devm_regmap_init_mmio_clk(&pdev->dev, - "ipg", iomem, &regconfig); + "ipg", iomem, &fsl_ssi_regconfig); } if (IS_ERR(ssi_private->regs)) { dev_err(&pdev->dev, "Failed to init register map\n"); -- 1.9.1

3 6

[alsa-devel] Trouble about Roland UA-11 UA-33 UA-55 and snd-usb-audio module
by FirebUntU 07 Mar '16

07 Mar '16

Hi guys, I write you about support of snd-usb-audio module and a relatively-new gamma of devices from Roland/Edirol: DUO CAPTURE, TRI CAPTURE and QUAD CAPTURE respectively: UA-11, UA-33, UA-55 These aren't not listed in the Alsa db. I have searched a solution for a week and I haven't found a trick to make UA-11 to work under Alsa 1.25.0 then I downloaded the driver for try to patch it and I found these devices is not listed in the tricks table In despair I have played with the switch button called "EXT * **" in the bottom of the unit of UA-11 and magically the device ID is transformed from * 0582:012b to ** 0582:0137 Image: https://dl.dropbox.com/u/32759898/output.jpg (Rear of the UA-11 unit, from the manual) *Before* (Recognized from udev - Unknown from alsa): http://www.alsa-project.org/db/?f=502ee68cf31882ba9ef23525fd66452c34d329e1 *After* (Recognized from udev - Recognized from alsa): http://www.alsa-project.org/db/?f=b2ae3e4331fb3fe613ba71e7aba1ad2e0e6dcd29 In the tricks table header of ALSA the 0582:0137 stay for PM-1000 in /TODO/ statement. I understand the EXT switch, undiscovered in the manual of the DUO CAPTURE is - as the fact - the Advanced Mode. Now the peripherical running both in playback and recording in 16 and 24 bit and without any mixer options in alsamixer. Can you update future releases with this gamma of devices? Awaiting for your positive response :) Thanks. Best regards, Leonardo

9 16

[alsa-devel] [PATCH 1/2] ASoC: cs35l32: avoid uninitialized variable access
by Arnd Bergmann 06 Mar '16

06 Mar '16

gcc warns about the possibilty of accessing a property read from devicetree in cs35l32_i2c_probe() when it has not been initialized because CONFIG_OF is disabled: sound/soc/codecs/cs35l32.c: In function 'cs35l32_i2c_probe': sound/soc/codecs/cs35l32.c:278:2: warning: 'val' may be used uninitialized in this function [-Wmaybe-uninitialized] The code is actually correct because it checks the dev->of_node variable first and we know this is NULL here, but by adding a check for IS_ENABLED(CONFIG_OF), we can let the compiler know as well, and also generate smaller object code. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cs35l32.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l32.c b/sound/soc/codecs/cs35l32.c index 44c30fe3e315..52ffaa8eb02b 100644 --- a/sound/soc/codecs/cs35l32.c +++ b/sound/soc/codecs/cs35l32.c @@ -372,7 +372,7 @@ static int cs35l32_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "could not allocate pdata\n"); return -ENOMEM; } - if (i2c_client->dev.of_node) { + if (IS_ENABLED(CONFIG_OF) && i2c_client->dev.of_node) { ret = cs35l32_handle_of_data(i2c_client, &cs35l32->pdata); if (ret != 0) -- 2.7.0

5 7

[alsa-devel] [PATCH - pcm 1/1] pcm: fix "unable to create IPC shm instance" in some case
by IceBsi 05 Mar '16

05 Mar '16

From: Qing Cai <caiqing(a)neusoft.com> As stated in manpage SHMCTL(2), shm_nattch is "No. of current attaches" (i.e., number of processes attached to the shared memeory). If an application uses alsa-lib and invokes fork() at some point, there should be the following execution sequence: 1. execute the following statement: pcm_direct.c:110: dmix->shmptr = shmat(dmix->shmid, 0, 0) (shm_nattch becomes 1) 2. invoke fork() in some thread. (shm_nattch becomes 2) 3. execute the following statement: pcm_direct.c:122: if (buf.shm_nattch == 1) 4. execute the following statement: pcm_direct.c:131: if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) (As stated in manpage SHMGET(2), "When a new shared memory segment is created, its contents are initialized to zero values", so dmix->shmptr->magic is 0) 5. execute the following statements: pcm_direct.c:132: snd_pcm_direct_shm_discard(dmix) pcm_direct.c:133: return -EINVAL The above execution sequence will cause the following error: unable to create IPC shm instance This error causes multimedia application has no sound. This error rarely occurs, probability is about 1%. Because the first user of the shared memory will get that dmix->shmptr->magic is 0, check dmix->shmptr->magic's value to determine if "we're the first user" is OK. Tests have been made 400+ times after this fix, and the issue no longer exists. Signed-off-by: Qing Cai <bsiice(a)msn.com> Signed-off-by: Qing Cai <caiqing(a)neusoft.com> diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index fd3877c..93d6f3c 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -119,7 +119,7 @@ retryget: snd_pcm_direct_shm_discard(dmix); return err; } - if (buf.shm_nattch == 1) { /* we're the first user, clear the segment */ + if (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) { /* we're the first user, clear the segment */ memset(dmix->shmptr, 0, sizeof(snd_pcm_direct_share_t)); if (dmix->ipc_gid >= 0) { buf.shm_perm.gid = dmix->ipc_gid; -- 2.7.2

3 10

[alsa-devel] Applied "ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616" to the asoc tree
by Mark Brown 05 Mar '16

05 Mar '16

The patch ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616 has been applied to the asoc tree at git://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 288bc356a881fcdce05f9782736a6a3d9db359a9 Mon Sep 17 00:00:00 2001 From: Caesar Wang <wxt(a)rock-chips.com> Date: Fri, 4 Mar 2016 14:54:50 +0800 Subject: [PATCH] ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616 This patch fixes that hasn't built the rt5616 driver with 'CONFIG_SND_SOC_RT5616=y' in .config. The tristate is the prompt on the 'make menuconfig', in other words, that can't show the prompt and select it if we don't say what's the tristate. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 50693c867e71..1de259860bc9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -596,7 +596,7 @@ config SND_SOC_RT298 depends on I2C config SND_SOC_RT5616 - tristate + tristate "Realtek RT5616 CODEC" config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" -- 2.7.0

1 0

[alsa-devel] [PATCH 00/14] drivers: use __maybe_unused to hide pm functions
by Arnd Bergmann 05 Mar '16

05 Mar '16

I found many variations of the bug in these device drivers (and some USB drivers I already send patches for in a separate series). In each case, the power management operations structure conditionally references suspend/resume functions, but the functions are hidden in an incorrect #ifdef or not hidden at all. We could try to correct the #ifdefs, but it seems easier to just mark those functions as __maybe_unused, which has the same effect but provides better compile-time test coverage and (subjectively) looks a bit nicer. I have a patch series that avoids all warnings in ARM randconfig builds, and I have verified that all these patches fix a warning that is still present in today's linux-next, and that they do not introduce new warnings in any configuration I found. Note that all these drivers are ARM specific, so I assume that all portable drivers got fixed already when someone rand into the problem on x86. There are no dependencies between the patches, so I'd appreciate subsystem maintainers to put them directly into their git trees. Arnd Arnd Bergmann (14): pinctrl: at91: use __maybe_unused to hide pm functions irqchip: st: use __maybe_unused to hide st_irq_syscfg_resume power: ipaq-micro-battery: use __maybe_unused to hide pm functions power: pm2301-charger: use __maybe_unused to hide pm functions mfd: ipaq-micro: use __maybe_unused to hide pm functions dma: sirf: use __maybe_unused to hide pm functions hw_random: exynos: use __maybe_unused to hide pm functions scsi: mvumi: use __maybe_unused to hide pm functions amd-xgbe: use __maybe_unused to hide pm functions wireless: cw1200: use __maybe_unused to hide pm functions_ input: spear-keyboard: use __maybe_unused to hide pm functions keyboard: snvs-pwrkey: use __maybe_unused to hide pm functions [media] omap3isp: use IS_ENABLED() to hide pm functions ASoC: rockchip: use __maybe_unused to hide st_irq_syscfg_resume drivers/char/hw_random/exynos-rng.c | 10 ++++------ drivers/dma/sirf-dma.c | 10 ++++------ drivers/input/keyboard/snvs_pwrkey.c | 4 ++-- drivers/input/keyboard/spear-keyboard.c | 6 ++---- drivers/irqchip/irq-st.c | 2 +- drivers/media/platform/omap3isp/isp.c | 13 +------------ drivers/mfd/ipaq-micro.c | 2 +- drivers/net/ethernet/amd/xgbe/xgbe-main.c | 6 ++---- drivers/net/wireless/st/cw1200/cw1200_spi.c | 9 ++------- drivers/net/wireless/st/cw1200/pm.h | 9 +++++++-- drivers/pinctrl/pinctrl-at91-pio4.c | 4 ++-- drivers/power/ipaq_micro_battery.c | 4 ++-- drivers/power/pm2301_charger.c | 22 ++++++---------------- drivers/scsi/mvumi.c | 4 ++-- sound/soc/rockchip/rockchip_spdif.c | 4 ++-- 15 files changed, 40 insertions(+), 69 deletions(-) -- 2.7.0 Cc: herbert(a)gondor.apana.org.au Cc: k.kozlowski(a)samsung.com Cc: dan.j.williams(a)intel.com Cc: vinod.koul(a)intel.com Cc: baohua(a)kernel.org Cc: dmitry.torokhov(a)gmail.com Cc: tglx(a)linutronix.de Cc: jason(a)lakedaemon.net Cc: marc.zyngier(a)arm.com Cc: laurent.pinchart(a)ideasonboard.com Cc: mchehab(a)osg.samsung.com Cc: lee.jones(a)linaro.org Cc: kvalo(a)codeaurora.org Cc: ludovic.desroches(a)atmel.com Cc: linus.walleij(a)linaro.org Cc: sre(a)kernel.org Cc: dbaryshkov(a)gmail.com Cc: JBottomley(a)odin.com Cc: martin.petersen(a)oracle.com Cc: broonie(a)kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: dmaengine(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-media(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-wireless(a)vger.kernel.org Cc: linux-gpio(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-rockchip(a)lists.infradead.org

2 2

[alsa-devel] [PATCH] ASoC: pcm3168a: remove unused variable
by Sudip Mukherjee 05 Mar '16

05 Mar '16

While building we were getting build warning about: sound/soc/codecs/pcm3168a.c:403:21: warning: variable 'channels' set but not used The variable channels were being assigned some value but that was never reused. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- Its upto you now. If you dont want to accept then I will add an extra From: in the commit. But since that will not be a part of my scripts so there will be chances of error. If you are worried about missing signoff, then I can assure you that my scripts will catch that before i can send it. sound/soc/codecs/pcm3168a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 0c7248a..992a77e 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -400,13 +400,12 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, struct pcm3168a_priv *pcm3168a = snd_soc_codec_get_drvdata(codec); bool tx, master_mode; u32 val, mask, shift, reg; - unsigned int rate, channels, fmt, ratio, max_ratio; + unsigned int rate, fmt, ratio, max_ratio; int i, min_frame_size; snd_pcm_format_t format; rate = params_rate(params); format = params_format(params); - channels = params_channels(params); ratio = pcm3168a->sysclk / rate; -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: rockchip: add bindings for spdif controller
by Sugar Zhang 05 Mar '16

05 Mar '16

this patch add compatible for rk3366/rk3368/rk3399 spdif, these three spdifs share the same type. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Documentation/devicetree/bindings/sound/rockchip-spdif.txt | 8 ++++++-- sound/soc/rockchip/rockchip_spdif.c | 13 ++++++++++--- 2 files changed, 16 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt index e64dbde..1104642 100644 --- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt +++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt @@ -7,8 +7,12 @@ a fibre cable. Required properties: - compatible: should be one of the following: - - "rockchip,rk3288-spdif", "rockchip,rk3188-spdif" or - "rockchip,rk3066-spdif" + - "rockchip,rk3066-spdif" + - "rockchip,rk3188-spdif" + - "rockchip,rk3288-spdif" + - "rockchip,rk3366-spdif" + - "rockchip,rk3368-spdif" + - "rockchip,rk3399-spdif" - reg: physical base address of the controller and length of memory mapped region. - interrupts: should contain the SPDIF interrupt. diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 5a806da..c589f6f 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -28,6 +28,7 @@ enum rk_spdif_type { RK_SPDIF_RK3066, RK_SPDIF_RK3188, RK_SPDIF_RK3288, + RK_SPDIF_RK3366, }; #define RK3288_GRF_SOC_CON2 0x24c @@ -45,11 +46,17 @@ struct rk_spdif_dev { static const struct of_device_id rk_spdif_match[] = { { .compatible = "rockchip,rk3066-spdif", - .data = (void *) RK_SPDIF_RK3066 }, + .data = (void *)RK_SPDIF_RK3066 }, { .compatible = "rockchip,rk3188-spdif", - .data = (void *) RK_SPDIF_RK3188 }, + .data = (void *)RK_SPDIF_RK3188 }, { .compatible = "rockchip,rk3288-spdif", - .data = (void *) RK_SPDIF_RK3288 }, + .data = (void *)RK_SPDIF_RK3288 }, + { .compatible = "rockchip,rk3366-spdif", + .data = (void *)RK_SPDIF_RK3366 }, + { .compatible = "rockchip,rk3368-spdif", + .data = (void *)RK_SPDIF_RK3366 }, + { .compatible = "rockchip,rk3399-spdif", + .data = (void *)RK_SPDIF_RK3366 }, {}, }; MODULE_DEVICE_TABLE(of, rk_spdif_match); -- 1.9.1

3 2

[alsa-devel] Applied "ASoC: rsnd: add CTU support" to the asoc tree
by Mark Brown 05 Mar '16

05 Mar '16

The patch ASoC: rsnd: add CTU support has been applied to the asoc tree at git://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 ec2ac01afe001360b911e27c915579fd003339e8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 25 Feb 2016 05:55:23 +0000 Subject: [PATCH] ASoC: rsnd: add CTU support This patch adds CTU (= Channel Transfer Unit) support on Renesas R-Car sound driver. It can Down/Up mixing and splitter. You need to check R-Car datasheet especially CTUn_CPMDR/CTUn_SV0xR/CTUn_SV1xR/CTUn_SV2xR/CTUn_SV3xR for setting parameter. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ctu.c | 234 ++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 197 insertions(+), 37 deletions(-) diff --git a/sound/soc/sh/rcar/ctu.c b/sound/soc/sh/rcar/ctu.c index b326966ea407..9dcc1f9db026 100644 --- a/sound/soc/sh/rcar/ctu.c +++ b/sound/soc/sh/rcar/ctu.c @@ -12,8 +12,74 @@ #define CTU_NAME_SIZE 16 #define CTU_NAME "ctu" +/* + * User needs to setup CTU by amixer, and its settings are + * based on below registers + * + * CTUn_CPMDR : amixser set "CTU Pass" + * CTUn_SV0xR : amixser set "CTU SV0" + * CTUn_SV1xR : amixser set "CTU SV1" + * CTUn_SV2xR : amixser set "CTU SV2" + * CTUn_SV3xR : amixser set "CTU SV3" + * + * [CTU Pass] + * 0000: default + * 0001: Connect input data of channel 0 + * 0010: Connect input data of channel 1 + * 0011: Connect input data of channel 2 + * 0100: Connect input data of channel 3 + * 0101: Connect input data of channel 4 + * 0110: Connect input data of channel 5 + * 0111: Connect input data of channel 6 + * 1000: Connect input data of channel 7 + * 1001: Connect calculated data by scale values of matrix row 0 + * 1010: Connect calculated data by scale values of matrix row 1 + * 1011: Connect calculated data by scale values of matrix row 2 + * 1100: Connect calculated data by scale values of matrix row 3 + * + * [CTU SVx] + * [Output0] = [SV00, SV01, SV02, SV03, SV04, SV05, SV06, SV07] + * [Output1] = [SV10, SV11, SV12, SV13, SV14, SV15, SV16, SV17] + * [Output2] = [SV20, SV21, SV22, SV23, SV24, SV25, SV26, SV27] + * [Output3] = [SV30, SV31, SV32, SV33, SV34, SV35, SV36, SV37] + * [Output4] = [ 0, 0, 0, 0, 0, 0, 0, 0 ] + * [Output5] = [ 0, 0, 0, 0, 0, 0, 0, 0 ] + * [Output6] = [ 0, 0, 0, 0, 0, 0, 0, 0 ] + * [Output7] = [ 0, 0, 0, 0, 0, 0, 0, 0 ] + * + * [SVxx] + * Plus Minus + * value time dB value time dB + * ----------------------------------------------------------------------- + * H'7F_FFFF 2 6 H'80_0000 2 6 + * ... + * H'40_0000 1 0 H'C0_0000 1 0 + * ... + * H'00_0001 2.38 x 10^-7 -132 + * H'00_0000 0 Mute H'FF_FFFF 2.38 x 10^-7 -132 + * + * + * Ex) Input ch -> Output ch + * 1ch -> 0ch + * 0ch -> 1ch + * + * amixer set "CTU Reset" on + * amixer set "CTU Pass" 9,10 + * amixer set "CTU SV0" 0,4194304 + * amixer set "CTU SV1" 4194304,0 + * or + * amixer set "CTU Reset" on + * amixer set "CTU Pass" 2,1 + */ + struct rsnd_ctu { struct rsnd_mod mod; + struct rsnd_kctrl_cfg_m pass; + struct rsnd_kctrl_cfg_m sv0; + struct rsnd_kctrl_cfg_m sv1; + struct rsnd_kctrl_cfg_m sv2; + struct rsnd_kctrl_cfg_m sv3; + struct rsnd_kctrl_cfg_s reset; int channels; }; @@ -58,51 +124,91 @@ static int rsnd_ctu_probe_(struct rsnd_mod *mod, static void rsnd_ctu_value_init(struct rsnd_dai_stream *io, struct rsnd_mod *mod) { + struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); + u32 cpmdr = 0; + u32 scmdr = 0; + int i; + + for (i = 0; i < RSND_MAX_CHANNELS; i++) { + u32 val = ctu->pass.val[i]; + + cpmdr |= val << (28 - (i * 4)); + + if ((val > 0x8) && (scmdr < (val - 0x8))) + scmdr = val - 0x8; + } + rsnd_mod_write(mod, CTU_CTUIR, 1); rsnd_mod_write(mod, CTU_ADINR, rsnd_runtime_channel_original(io)); - rsnd_mod_write(mod, CTU_CPMDR, 0); - rsnd_mod_write(mod, CTU_SCMDR, 0); - rsnd_mod_write(mod, CTU_SV00R, 0); - rsnd_mod_write(mod, CTU_SV01R, 0); - rsnd_mod_write(mod, CTU_SV02R, 0); - rsnd_mod_write(mod, CTU_SV03R, 0); - rsnd_mod_write(mod, CTU_SV04R, 0); - rsnd_mod_write(mod, CTU_SV05R, 0); - rsnd_mod_write(mod, CTU_SV06R, 0); - rsnd_mod_write(mod, CTU_SV07R, 0); - - rsnd_mod_write(mod, CTU_SV10R, 0); - rsnd_mod_write(mod, CTU_SV11R, 0); - rsnd_mod_write(mod, CTU_SV12R, 0); - rsnd_mod_write(mod, CTU_SV13R, 0); - rsnd_mod_write(mod, CTU_SV14R, 0); - rsnd_mod_write(mod, CTU_SV15R, 0); - rsnd_mod_write(mod, CTU_SV16R, 0); - rsnd_mod_write(mod, CTU_SV17R, 0); - - rsnd_mod_write(mod, CTU_SV20R, 0); - rsnd_mod_write(mod, CTU_SV21R, 0); - rsnd_mod_write(mod, CTU_SV22R, 0); - rsnd_mod_write(mod, CTU_SV23R, 0); - rsnd_mod_write(mod, CTU_SV24R, 0); - rsnd_mod_write(mod, CTU_SV25R, 0); - rsnd_mod_write(mod, CTU_SV26R, 0); - rsnd_mod_write(mod, CTU_SV27R, 0); - - rsnd_mod_write(mod, CTU_SV30R, 0); - rsnd_mod_write(mod, CTU_SV31R, 0); - rsnd_mod_write(mod, CTU_SV32R, 0); - rsnd_mod_write(mod, CTU_SV33R, 0); - rsnd_mod_write(mod, CTU_SV34R, 0); - rsnd_mod_write(mod, CTU_SV35R, 0); - rsnd_mod_write(mod, CTU_SV36R, 0); - rsnd_mod_write(mod, CTU_SV37R, 0); + rsnd_mod_write(mod, CTU_CPMDR, cpmdr); + + rsnd_mod_write(mod, CTU_SCMDR, scmdr); + + if (scmdr > 0) { + rsnd_mod_write(mod, CTU_SV00R, ctu->sv0.val[0]); + rsnd_mod_write(mod, CTU_SV01R, ctu->sv0.val[1]); + rsnd_mod_write(mod, CTU_SV02R, ctu->sv0.val[2]); + rsnd_mod_write(mod, CTU_SV03R, ctu->sv0.val[3]); + rsnd_mod_write(mod, CTU_SV04R, ctu->sv0.val[4]); + rsnd_mod_write(mod, CTU_SV05R, ctu->sv0.val[5]); + rsnd_mod_write(mod, CTU_SV06R, ctu->sv0.val[6]); + rsnd_mod_write(mod, CTU_SV07R, ctu->sv0.val[7]); + } + if (scmdr > 1) { + rsnd_mod_write(mod, CTU_SV10R, ctu->sv1.val[0]); + rsnd_mod_write(mod, CTU_SV11R, ctu->sv1.val[1]); + rsnd_mod_write(mod, CTU_SV12R, ctu->sv1.val[2]); + rsnd_mod_write(mod, CTU_SV13R, ctu->sv1.val[3]); + rsnd_mod_write(mod, CTU_SV14R, ctu->sv1.val[4]); + rsnd_mod_write(mod, CTU_SV15R, ctu->sv1.val[5]); + rsnd_mod_write(mod, CTU_SV16R, ctu->sv1.val[6]); + rsnd_mod_write(mod, CTU_SV17R, ctu->sv1.val[7]); + } + if (scmdr > 2) { + rsnd_mod_write(mod, CTU_SV20R, ctu->sv2.val[0]); + rsnd_mod_write(mod, CTU_SV21R, ctu->sv2.val[1]); + rsnd_mod_write(mod, CTU_SV22R, ctu->sv2.val[2]); + rsnd_mod_write(mod, CTU_SV23R, ctu->sv2.val[3]); + rsnd_mod_write(mod, CTU_SV24R, ctu->sv2.val[4]); + rsnd_mod_write(mod, CTU_SV25R, ctu->sv2.val[5]); + rsnd_mod_write(mod, CTU_SV26R, ctu->sv2.val[6]); + rsnd_mod_write(mod, CTU_SV27R, ctu->sv2.val[7]); + } + if (scmdr > 3) { + rsnd_mod_write(mod, CTU_SV30R, ctu->sv3.val[0]); + rsnd_mod_write(mod, CTU_SV31R, ctu->sv3.val[1]); + rsnd_mod_write(mod, CTU_SV32R, ctu->sv3.val[2]); + rsnd_mod_write(mod, CTU_SV33R, ctu->sv3.val[3]); + rsnd_mod_write(mod, CTU_SV34R, ctu->sv3.val[4]); + rsnd_mod_write(mod, CTU_SV35R, ctu->sv3.val[5]); + rsnd_mod_write(mod, CTU_SV36R, ctu->sv3.val[6]); + rsnd_mod_write(mod, CTU_SV37R, ctu->sv3.val[7]); + } rsnd_mod_write(mod, CTU_CTUIR, 0); } +static void rsnd_ctu_value_reset(struct rsnd_dai_stream *io, + struct rsnd_mod *mod) +{ + struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); + int i; + + if (!ctu->reset.val) + return; + + for (i = 0; i < RSND_MAX_CHANNELS; i++) { + ctu->pass.val[i] = 0; + ctu->sv0.val[i] = 0; + ctu->sv1.val[i] = 0; + ctu->sv2.val[i] = 0; + ctu->sv3.val[i] = 0; + } + ctu->reset.val = 0; +} + static int rsnd_ctu_init(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) @@ -164,12 +270,66 @@ static int rsnd_ctu_hw_params(struct rsnd_mod *mod, return 0; } +static int rsnd_ctu_pcm_new(struct rsnd_mod *mod, + struct rsnd_dai_stream *io, + struct snd_soc_pcm_runtime *rtd) +{ + struct rsnd_ctu *ctu = rsnd_mod_to_ctu(mod); + int ret; + + /* CTU Pass */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU Pass", + NULL, + &ctu->pass, RSND_MAX_CHANNELS, + 0xC); + + /* ROW0 */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV0", + NULL, + &ctu->sv0, RSND_MAX_CHANNELS, + 0x00FFFFFF); + if (ret < 0) + return ret; + + /* ROW1 */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV1", + NULL, + &ctu->sv1, RSND_MAX_CHANNELS, + 0x00FFFFFF); + if (ret < 0) + return ret; + + /* ROW2 */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV2", + NULL, + &ctu->sv2, RSND_MAX_CHANNELS, + 0x00FFFFFF); + if (ret < 0) + return ret; + + /* ROW3 */ + ret = rsnd_kctrl_new_m(mod, io, rtd, "CTU SV3", + NULL, + &ctu->sv3, RSND_MAX_CHANNELS, + 0x00FFFFFF); + if (ret < 0) + return ret; + + /* Reset */ + ret = rsnd_kctrl_new_s(mod, io, rtd, "CTU Reset", + rsnd_ctu_value_reset, + &ctu->reset, 1); + + return ret; +} + static struct rsnd_mod_ops rsnd_ctu_ops = { .name = CTU_NAME, .probe = rsnd_ctu_probe_, .init = rsnd_ctu_init, .quit = rsnd_ctu_quit, .hw_params = rsnd_ctu_hw_params, + .pcm_new = rsnd_ctu_pcm_new, }; struct rsnd_mod *rsnd_ctu_mod_get(struct rsnd_priv *priv, int id) -- 2.7.0

1 0

[alsa-devel] [RFC v2 0/6] ASoC: Add mediatek HDMI codec support
by Philipp Zabel 05 Mar '16

05 Mar '16

Hi, this time we add an interface for Jyri's generic hdmi-codec driver [1] to the mediatek hdmi bridge driver. In place of a separate machine driver, the existing mt8173-rt5650-rt5676 driver is extended to link to the hdmi-codec instance provided by the hdmi bridge driver. The alsa jack support, previously implemented by the custom mediatek hdmi codec and machine drivers, is bolted on top of hdmi-codec using Russell's HDMI notifier prototype. It's all still a bit rough, but I could use some feedback at this point. Have we achieved consensus to go forward with the hdmi-codec approach? Is it ok to use an alsa jack for HDMI HPD status reporting? The mediatek drm patches apply on top of https://patchwork.kernel.org/patch/7949021/ ("drm/mediatek: Add HDMI support") the asoc patches depend on these two patches: https://patchwork.kernel.org/patch/7215121/ ("ALSA: pcm: add IEC958 channel status helper for hw_params") [1] https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") best regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the machine driver Philipp Zabel (5): drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver ASoC: mediatek: address dai link array entries by enum video: rmk's HDMI notification prototype drm/mediatek: hdmi: issue notifications ASoC: hdmi-codec: Use HDMI notifications to add jack support .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- drivers/gpu/drm/mediatek/Kconfig | 1 + drivers/gpu/drm/mediatek/mtk_cec.c | 5 + drivers/gpu/drm/mediatek/mtk_drm_hdmi_drv.c | 192 +++++++++++++++++++++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 26 +++ drivers/gpu/drm/mediatek/mtk_hdmi.h | 4 + drivers/video/Makefile | 2 +- drivers/video/hdmi-not.c | 61 +++++++ include/linux/hdmi-not.h | 39 +++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/hdmi-codec.c | 72 +++++++- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 66 ++++++- 12 files changed, 470 insertions(+), 9 deletions(-) create mode 100644 drivers/video/hdmi-not.c create mode 100644 include/linux/hdmi-not.h -- 2.6.2

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[alsa-devel] [PATCH v2 0/2] ASoC: ads117x: Add device tree support for TI ADS117x
by Florian Vaussard 05 Mar '16

05 Mar '16

Hello, This series adds device tree support for Texas Instruments ADS117x ADC chip. The first patch adds the binding documentation, while the second patch adds the compatible string into the driver. No further properties are necessary. Best regards, Florian --- Since v1: - Updated topic for the documentation patch - Removed wildcard in compatible string --- Florian Vaussard (2): ASoC: ads117x: Add bindings documentation for TI ADS117x ADC ASoC: ads117x: Add device tree compatible string Documentation/devicetree/bindings/sound/ti,ads117x.txt | 11 +++++++++++ sound/soc/codecs/ads117x.c | 12 ++++++++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,ads117x.txt -- 2.1.0

3 7

[alsa-devel] [PATCH] ASoC: dapm: Remove unneeded variable ret returning 0
by Rajan Vaja 05 Mar '16

05 Mar '16

This patch fixes coccicheck report: Unneeded variable: "ret". Return "0" on line 2817 Remove unneccesary variable ret created to return zero. Signed-off-by: Rajan Vaja <rajan.vaja(a)gmail.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5a2812f..8dc395f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2805,7 +2805,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret = 0; + int i; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { @@ -2814,7 +2814,7 @@ int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, } mutex_unlock(&dapm->card->dapm_mutex); - return ret; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_del_routes); -- 1.7.9.5

2 1

[alsa-devel] I created a wiki page about bug reporting, review requested
by Tanu Kaskinen 05 Mar '16

05 Mar '16

Hi, When talking with PulseAudio users, I sometimes need to tell people to file ALSA bugs, and it would be nice to have a web page that explains how to do that. (Also, I found that the alsa packages in OpenEmbedded point to a non-functional bug tracker, and I wanted to fix the packages, but I couldn't find a substitute link.) Therefore I created a wiki page: http://alsa-project.org/main/index.php/Bug_Tracking The Bug_Tracking wiki page points to another new page, explaining alsa- info.sh: http://alsa-project.org/main/index.php/AlsaInfo Are the ALSA maintainers happy with the contents? I'm not really an expert on how you prefer bugs to be reported. Also, the "Bug Tracker" link in the wiki sidebar points to https://bugtrack.alsa-project.org/alsa-bug which doesn't work. I think that link should be changed to point to the new wiki page, but I don't have edit rights for the sidebar. -- Tanu

2 2

[alsa-devel] [PATCH 0/6] ASoC: rsnd: add CTU support
by Kuninori Morimoto 05 Mar '16

05 Mar '16

Hi Mark These patches adds CTU (= Channel Transfer Unit) support to Renesas R-Car sound driver. It can exchange channel order ex) channel {0,1,2,3} -> channel {3,2,1,0} And it can add/reduce each channels sound ex) (input 0th ch x 0.5) + (input 2nd ch x 0.5) = output 1st ch You need to check R-Car datasheet for each detail settings, or check 6) patch Kuninori Morimoto (6): 1) ASoC: rsnd: don't call update callback if it was NULL 2) ASoC: rsrc-card: add convert channels support 3) ASoC: rsnd: tidyup rsnd_ssiu_init_gen2() 4) ASoC: rsnd: add CTU basic support 5) ASoC: rsnd: add rsnd_runtime_channel_xxx() 6) ASoC: rsnd: add CTU support Documentation/devicetree/bindings/sound/renesas,rsrc-card.txt | 1 + sound/soc/sh/rcar/core.c | 71 ++++++++++++++---------- sound/soc/sh/rcar/ctu.c | 287 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++-------------- sound/soc/sh/rcar/dvc.c | 2 +- sound/soc/sh/rcar/mix.c | 2 +- sound/soc/sh/rcar/rsnd.h | 9 +++- sound/soc/sh/rcar/rsrc-card.c | 22 +++++--- sound/soc/sh/rcar/src.c | 2 +- sound/soc/sh/rcar/ssi.c | 15 +++--- sound/soc/sh/rcar/ssiu.c | 11 ++-- 10 files changed, 330 insertions(+), 92 deletions(-) Best regards --- Kuninori Morimoto

3 14

[alsa-devel] [PATCH] ASoC: wm8994: codec driver should manage its own MCLK
by Zidan Wang 05 Mar '16

05 Mar '16

codec driver should manage its own master clock, MCLK1 and MCLK2. When bias_level change from standby to prepare, enable codec MCLK. When bias_level change from prepare to standby, disable codec MCLK. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- drivers/mfd/wm8994-core.c | 7 +++++++ include/linux/mfd/wm8994/pdata.h | 3 +++ sound/soc/codecs/wm8994.c | 17 +++++++++++++++-- 3 files changed, 25 insertions(+), 2 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 7eec619..ace2cea 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -14,6 +14,7 @@ #include <linux/kernel.h> #include <linux/module.h> +#include <linux/clk.h> #include <linux/slab.h> #include <linux/i2c.h> #include <linux/err.h> @@ -270,6 +271,7 @@ static int wm8994_set_pdata_from_of(struct wm8994 *wm8994) { struct device_node *np = wm8994->dev->of_node; struct wm8994_pdata *pdata = &wm8994->pdata; + char tmp[8]; int i; if (!np) @@ -310,6 +312,11 @@ static int wm8994_set_pdata_from_of(struct wm8994 *wm8994) if (pdata->ldo[1].enable < 0) pdata->ldo[1].enable = 0; + for (i = 0; i < WM8994_NUM_MCLK; i++) { + sprintf(tmp, "MCLK%d", i + 1); + pdata->mclk[i] = devm_clk_get(wm8994->dev, tmp); + } + return 0; } #else diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 90c6052..657a828 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -18,6 +18,7 @@ #define WM8994_NUM_LDO 2 #define WM8994_NUM_GPIO 11 #define WM8994_NUM_AIF 3 +#define WM8994_NUM_MCLK 2 struct wm8994_ldo_pdata { /** GPIOs to enable regulator, 0 or less if not available */ @@ -233,6 +234,8 @@ struct wm8994_pdata { * GPIO for the IRQ pin if host only supports edge triggering */ int irq_gpio; + + struct clk *mclk[WM8994_NUM_MCLK]; }; #endif diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2ccbb32..7379eae 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gcd.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> @@ -2474,6 +2475,8 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + struct wm8994_pdata *pdata = &control->pdata; + int i; wm_hubs_set_bias_level(codec, level); @@ -2495,8 +2498,13 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; } - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) { active_reference(codec); + + for (i = 0; i < WM8994_NUM_MCLK; i++) + if (!IS_ERR(pdata->mclk[i])) + clk_prepare_enable(pdata->mclk[i]); + } break; case SND_SOC_BIAS_STANDBY: @@ -2524,8 +2532,13 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, WM8994_LINEOUT2_DISCH); } - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE) + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_PREPARE) { + for (i = 0; i < WM8994_NUM_MCLK; i++) + if (!IS_ERR(pdata->mclk[i])) + clk_disable_unprepare(pdata->mclk[i]); + active_dereference(codec); + } /* MICBIAS into bypass mode on newer devices */ switch (control->type) { -- 1.9.1

3 2

[alsa-devel] [PATCH] ALSA: compress: fix more typos
by Vinod Koul 04 Mar '16

04 Mar '16

More inspection of code revealed few more typos so fix them as well Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c4b14b5f1b35..7d59dbda8897 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -703,7 +703,7 @@ static int snd_compress_wait_for_drain(struct snd_compr_stream *stream) /* * We are called with lock held. So drop the lock while we wait for - * drain complete notfication from the driver + * drain complete notification from the driver * * It is expected that driver will notify the drain completion and then * stream will be moved to SETUP state, even if draining resulted in an @@ -758,7 +758,7 @@ static int snd_compr_next_track(struct snd_compr_stream *stream) if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) return -EPERM; - /* you can signal next track isf this is intended to be a gapless stream + /* you can signal next track if this is intended to be a gapless stream * and current track metadata is set */ if (stream->metadata_set == false) -- 1.9.1

2 1

Re: [alsa-devel] [PATCH v3] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Nicolin Chen 04 Mar '16

04 Mar '16

On Fri, Mar 04, 2016 at 08:54:10AM +0000, Zidan Wang wrote: > On Friday, March 04, 2016 3:40 PM, Nicolin Chen wrote: > > > > > cpu-dai1 : The phandle of an CPU DAI controller for AIF1 > > > > cpu-dai2 : The phandle of an CPU DAI controller for AIF2 > > > > cpu-dai3 : The phandle of an CPU DAI controller for AIF3 > > > > > > > > > + data->card.num_links = DAI_LINK_NUM; > > > > > + data->card.dai_link = data->dai_link; > > > > > > > > Then, rather than hard coding the total DAI link number and creating > > > > useless dummy links, you may create the links dynamically according > > > > to the presence of the three properties above. > > > > > > I don't know how to using cpu-dai1/2/3. There don't have voice and bt > > controller on the IMX cpu, don't have such cpu nodes. > > > Do you means our hardware connection are wrong, and we must have voice > > and bt controller on the cpu? > > > > Okay. I was trying to give you an excuse to have all three aifs included, > > as I thought you might eventually need them to *connect* something via DT. > > If you think you don't have any idea to pre-allocate these links, you > > probably should ignore my suggestion here and just follow Mark's comments: > > I am just confused by our hardward connection. If dummy cpu dai is not ok, I think I should follow your suggestion. > > If we want to keep our hardware connection, which bt and voice directly connect to codec, and don't have bt and voice controller. > What's the best way? The only way to keep them while not having BT and voice controllers is just what I suggested you to make it through DT. Devicetree is a script to reflect your exact hardware connections. You can dis-link them for now in the DT as they are not "connected" in your current design and then link them whenever you get them connected -- I think there should be connections since you are using a validation board. You just don't have the extension board or something. So you may try to reach your hardware team, especially designer, for detail. The idea here is similar to the ASRC links in fsl-asoc-card driver: users can drop ASRC by merely removing the asrc phandle in the DT, depending on their board design or requirement. The drawback of this way is that you need to figure out an accurate DT binding: you might need to figure out a better name rather than "cpu-dai" because it sounds like WM8958 may not act as a Codec at all when it connects your BT route. That means WM8958 will become a cpu-dai somehow? You can send a patch with a decent DT binding. I believe Mark will give some better advice.

1 0

[alsa-devel] [PATCH 0/4] Fix headphone detection on Acer Chromebook 13
by Jonathan Tinkham 04 Mar '16

04 Mar '16

Headphone output works, but headphone detection does not. The problem is improper inversion of the jack gpio, and improper naming of the DAPM switch. With these changes, headphone insertion and removal is picked up by userspace, and the switch properly toggled. Jonathan Tinkham (4): soc/sound: tegra_max98090: do not invert headphone jack sound/soc: tegra_max98090: rename headphone jack DAPM sound/soc: tegra_max98090: document headphone jack rename soc/sound: tegra_max98090: update dts affected by rename .../devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt | 6 +++--- arch/arm/boot/dts/tegra124-nyan.dtsi | 4 ++-- arch/arm/boot/dts/tegra124-venice2.dts | 4 ++-- sound/soc/tegra/tegra_max98090.c | 10 +++++----- 4 files changed, 12 insertions(+), 12 deletions(-) -- 2.7.0

4 37

[alsa-devel] CM6206-LX usb sound card
by Nikita Orlov 04 Mar '16

04 Mar '16

Hello! First I sent these questions to alsa-user, but I think this mailing list is more proper place to ask such questions. I have cheap chinese usb audio card, based on CM6206-LX chip from CMEDIA. CM6206-LX is similar to CM6206, but not 7.1 channels output, it is 5.1 channels. I use ubuntu 15.* with kernel 4.2.0-30-generic. dmesg show me some warnings about sound card: [31010.394487] usb 1-1: new full-speed USB device number 24 using xhci_hcd [31010.523953] usb 1-1: New USB device found, idVendor=0d8c, idProduct=0102 [31010.523956] usb 1-1: New USB device strings: Mfr=0, Product=2, SerialNumber=0 [31010.523957] usb 1-1: Product: USB Sound Device [31010.526235] input: USB Sound Device as /devices/pci0000:00/0000:00:14.0/usb1/1-1/1-1:1.3/0003:0D8C:0102.0010/input/input33 [31010.578876] hid-generic 0003:0D8C:0102.0010: input,hidraw0: USB HID v1.00 Device [USB Sound Device ] on usb-0000:00:14.0-1/input3 [31010.643908] usb 1-1: current rate 30464 is different from the runtime rate 96000 [31010.646123] usb 1-1: Warning! Unlikely big volume range (=8065), cval->res is probably wrong. [31010.646126] usb 1-1: [9] FU [Mic Playback Volume] ch = 2, val = -6144/1921/1 [31010.646485] usb 1-1: Warning! Unlikely big volume range (=8065), cval->res is probably wrong. [31010.646486] usb 1-1: [11] FU [Line Playback Volume] ch = 2, val = -6144/1921/1 [31010.647884] usb 1-1: Warning! Unlikely big volume range (=6928), cval->res is probably wrong. [31010.647887] usb 1-1: [8] FU [Mic Capture Volume] ch = 2, val = -4096/2832/1 [31010.648291] usb 1-1: Warning! Unlikely big volume range (=6928), cval->res is probably wrong. [31010.648295] usb 1-1: [15] FU [Line Capture Volume] ch = 2, val = -4096/2832/1 [31010.648662] usb 1-1: Warning! Unlikely big volume range (=6928), cval->res is probably wrong. [31010.648664] usb 1-1: [2] FU [PCM Capture Volume] ch = 2, val = -4096/2832/1 [31010.648887] usbcore: registered new interface driver snd-usb-audio There are two types of errors, first that current rate 30464 is different from the runtime rate 96000. Unfortunately I have no idea about this warning. And warnings like Unlikely big volume range (=8065), cval->res is probably wrong. As I understand it is because auto value range for some mixers are not proper detected. I have test capturing with simple test app. It seems only one avalible sample rate is 48000 Hz. printf("Sample rate = %d\n", SampleRate); snd_pcm_hw_params_set_rate_near(Handle, params, &SampleRate, &dir); printf("Sample rate = %d\n", SampleRate); After calling these function SampleRate becomes 48000, even if before it was 44100. Maybe it is related to first error in dmesg. Anybody knows? But, according to cat /proc/asound/card2/stream0 USB Sound Device at usb-0000:00:14.0-1, full speed : USB Audio Playback: Status: Stop Interface 1 Altset 1 Format: S16_LE Channels: 8 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 1 Altset 2 Format: S16_LE Channels: 2 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 1 Altset 3 Format: S16_LE Channels: 4 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 1 Altset 4 Format: S16_LE Channels: 6 Endpoint: 6 OUT (ADAPTIVE) Rates: 44100, 48000 Interface 1 Altset 5 Format: S16_LE Channels: 2 Endpoint: 6 OUT (ADAPTIVE) Rates: 96000 Capture: Status: Stop Interface 2 Altset 1 Format: S16_LE Channels: 2 Endpoint: 5 IN (ASYNC) Rates: 44100, 48000 It is able to capture 44100. I have tried to capture via arecord: arecord -D hw:2,0 -f cd -c 2 > ./test2.wav and VLC said it is 44100, but I don`t have microphone at the moment to test is there proper sound recorded or not. I will try it later. I have found that init of sound card is related to snd_usb_audio.ko module and learn a bit how it works. There is a quirk that write internal registers. According to mailing list archives dated in 2009, values were found by usb traffic sniffing under Windows. I have found CM6206 datasheet and check description of internal registers. http://wenku.baidu.com/link?url=vkQLGhKUuvVmHC7vyobjKlF0yixpjn3ueAdD81OO6Ul… It is link to datasheet. Website is chinese, but datasheet itsel is in english. To see it after 3rd page click 下载文档到电脑，查找使用更方便. But there are no interesting things about registers and mostly values that are exist in current snd_usb_audio must work ok. Also in datasheet there some info about values of volume ranges that can be related to second dmesg warnings. I have uploaded images of related info from datasheet here https://yadi.sk/d/1kdCFs7Gpv8L8 I have try with mixer.c in sound/usb/ dir of kernel tree (4.2.0) and tried to add experimental code in volume_control_quirks function. But it seems function is not called for my device for some reason. It is more complicated that I think first. So another question is how to make quirk for proper mixer values? -- Nikita Orlov Skype: nik_stet QQ: 2717846083

1 0

[alsa-devel] compress offload patch
by Eric Laurent 04 Mar '16

04 Mar '16

Takashi, Vinod, Can you help with merging the attached patch for compress offload core upstream? Thanks. Eric.

3 4

[alsa-devel] [PATCH 1/2] ALSA: compress: Add SNDRV_PCM_STATE_PREPARED state explanation
by Vinod Koul 04 Mar '16

04 Mar '16

Stream states were explained in the code comments but SNDRV_PCM_STATE_PREPARED was missed so add it Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/core/compress_offload.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 7fac3cae8abd..51a8eb495f54 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -74,6 +74,9 @@ struct snd_compr_file { * SNDRV_PCM_STATE_SETUP: When stream has been initialized. This is done by * calling SNDRV_COMPRESS_SET_PARAMS. running streams will come to this * state at stop by calling SNDRV_COMPRESS_STOP, or at end of drain. + * SNDRV_PCM_STATE_PREPARED: When a stream has been written to (for + * playback only). User after setting up stream writes the data buffer + * before starting the stream. * SNDRV_PCM_STATE_RUNNING: When stream has been started and is * decoding/encoding and rendering/capturing data. * SNDRV_PCM_STATE_DRAINING: When stream is draining current data. This is done -- 1.9.1

2 2

[alsa-devel] USB audio2.0 status interrupt endpoint
by Puneet Sharma 04 Mar '16

04 Mar '16

Hello all, I am using the uac2 kernel driver(drivers/usb/gadget/f_uac2.c) having kernel version 3.10.31 for an embedded board. The board is presented as a USB device to a Linux PC having USB host controller in it. I have added an interrupt endpoint in the uac2 kernel driver to support the status interrupt endpoint request as per usb audio 2.0 specification. For the interrupt transfer to happen, I prepared the interrupt packet structure as per audio 2.0 specification and queued the endpoint to transfer it to the host driver. The interrupt structure is prepared as follows: bInfo : 0 (Interface request) bAttribute : 0x1 (CUR request) wValue : 0x200 (VOLUME feature unit control selector) wIndex : 0x700 (FEATURE UNIT ID) As per my understanding, when the above interrupt endpoint is transferred to host in response to Interrupt IN request, the host should send me a GET CUR volume class-specific request to further qualify the cause of interrupt. However, I am not getting any class-specific request from host in response to the status interrupt transferred to host. My query is do i need to implement a mixer interface in the uac2 kernel driver to get the request or i am doing something wrong in the process. I am stuck onto this and cannot proceed further. Thanks Puneet

2 2

[alsa-devel] writing an alsa driver
by Michele Curti 04 Mar '16

04 Mar '16

Hi all, I'm going to *try* to "write" a driver for the sound card on my new laptop, an asus x205ta. To be fair I have almost no experience but I'm hoping that all the code is already present and it only needs some adaptations. The cpu is a intel bay trail z3735f, the sound card is a realtek rt5648 that has the same product Id of the rt5645 (0x6308). I see in kernel sources: - the codec driver sound/soc/codecs/rt5645.c - a baytrail board that uses the rt5640 sound/soc/intel/boards/byt-rt5640.c - a braswell board that uses the rt5645 sound/soc/intel/boards/cht_bsw_rt5645.c What do you think? Is it feasible, given a good amount of free-time? :p Thanks, Michele

10 46

[alsa-devel] [PATCH v2] ASoC: pcm3168a: remove unused variable
by Sudip Mukherjee 04 Mar '16

04 Mar '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> While building we were getting build warning about: sound/soc/codecs/pcm3168a.c:403:21: warning: variable 'channels' set but not used The variable channels were being assigned some value but that was never reused. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- v2: From: address changed sound/soc/codecs/pcm3168a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 0c7248a..992a77e 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -400,13 +400,12 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, struct pcm3168a_priv *pcm3168a = snd_soc_codec_get_drvdata(codec); bool tx, master_mode; u32 val, mask, shift, reg; - unsigned int rate, channels, fmt, ratio, max_ratio; + unsigned int rate, fmt, ratio, max_ratio; int i, min_frame_size; snd_pcm_format_t format; rate = params_rate(params); format = params_format(params); - channels = params_channels(params); ratio = pcm3168a->sysclk / rate; -- 1.9.1

1 0

[alsa-devel] [PATCH] Make alsa asound.h header more portable.
by Thomas Klausner 04 Mar '16

04 Mar '16

From: Thomas Klausner <wiz(a)NetBSD.org> This header is used as-is in the alsa-lib userland library, which is portable to other operating systems. For this reason, include linux/types.h only on Linux systems. Add sys/ioctl.h for _IOR/_IOW/etc. (works at least on *BSD and Solaris). Signed-off-by: Thomas Klausner <wiz(a)NetBSD.org> --- include/uapi/sound/asound.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index a82108e..3d581b5 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -23,7 +23,11 @@ #ifndef _UAPI__SOUND_ASOUND_H #define _UAPI__SOUND_ASOUND_H +#ifdef __linux__ #include <linux/types.h> +#else +#include <sys/ioctl.h> +#endif #ifndef __KERNEL__ #include <stdlib.h> -- 2.7.1

2 4

[alsa-devel] [PATCH 1/2] ALSA: hda - hdmi add wmb barrier for audio component
by libin.yang＠linux.intel.com 04 Mar '16

04 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> To make sure audio_ptr is set before intel_audio_codec_enable() or intel_audio_codec_disable() calling pin_eld_notify(), this patch adds wmb barrier to prevent optimizing. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index aa2d6a2..b5190e0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2939,6 +2939,11 @@ static int patch_generic_hdmi(struct hda_codec *codec) if (codec_has_acomp(codec)) { codec->depop_delay = 0; spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; snd_hdac_i915_register_notifier(&spec->i915_audio_ops); } -- 1.9.1

2 1

Re: [alsa-devel] [PATCH v3] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Nicolin Chen 04 Mar '16

04 Mar '16

On Fri, Mar 04, 2016 at 07:15:44AM +0000, Zidan Wang wrote: > > cpu-dai1 : The phandle of an CPU DAI controller for AIF1 > > cpu-dai2 : The phandle of an CPU DAI controller for AIF2 > > cpu-dai3 : The phandle of an CPU DAI controller for AIF3 > > > > > + data->card.num_links = DAI_LINK_NUM; > > > + data->card.dai_link = data->dai_link; > > > > Then, rather than hard coding the total DAI link number and creating > > useless dummy links, you may create the links dynamically according to > > the presence of the three properties above. > > I don't know how to using cpu-dai1/2/3. There don't have voice and bt controller on the IMX cpu, don't have such cpu nodes. > Do you means our hardware connection are wrong, and we must have voice and bt controller on the cpu? Okay. I was trying to give you an excuse to have all three aifs included, as I thought you might eventually need them to *connect* something via DT. If you think you don't have any idea to pre-allocate these links, you probably should ignore my suggestion here and just follow Mark's comments: [ Why are you mapping in dummy DAIs? If these devices aren't connected then they're not connected and you shouldn't represent them. If they are connected to something then describe those connections, possibly in followup patches if you have other devices you need to support upstream first. ]

1 0

[alsa-devel] [PATCH] ASoC: rt5616: allow to build with CONFIG_SND_SOC_RT5616
by Caesar Wang 04 Mar '16

04 Mar '16

This patch fixes that hasn't built the rt5616 driver with 'CONFIG_SND_SOC_RT5616=y' in .config. The tristate is the prompt on the 'make menuconfig', in other words, that can't show the prompt and select it if we don't say what's the tristate. Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4383966..14eb5f3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -623,7 +623,7 @@ config SND_SOC_RT5514 tristate config SND_SOC_RT5616 - tristate + tristate "Realtek RT5616 CODEC" config SND_SOC_RT5631 tristate "Realtek ALC5631/RT5631 CODEC" -- 1.9.1

1 0

[alsa-devel] [PATCH] sound/usb: Use meaninful names for goto labels
by Shuah Khan 04 Mar '16

04 Mar '16

Fix to use meaningful names instead of numbered goto labels Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- sound/usb/media.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/usb/media.c b/sound/usb/media.c index b87d5b3..ab6a642 100644 --- a/sound/usb/media.c +++ b/sound/usb/media.c @@ -84,21 +84,21 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, ret = media_device_register_entity(mctl->media_dev, &mctl->media_entity); if (ret) - goto err1; + goto free_mctl; mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, MAJOR(pcm_dev->devt), MINOR(pcm_dev->devt)); if (!mctl->intf_devnode) { ret = -ENOMEM; - goto err2; + goto unregister_entity; } mctl->intf_link = media_create_intf_link(&mctl->media_entity, &mctl->intf_devnode->intf, MEDIA_LNK_FL_ENABLED); if (!mctl->intf_link) { ret = -ENOMEM; - goto err3; + goto devnode_remove; } /* create link between mixer and audio */ @@ -109,7 +109,7 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, &mctl->media_entity, 0, MEDIA_LNK_FL_ENABLED); if (ret) - goto err4; + goto remove_intf_link; break; } } @@ -117,13 +117,13 @@ int media_snd_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, subs->media_ctl = mctl; return 0; -err4: +remove_intf_link: media_remove_intf_link(mctl->intf_link); -err3: +devnode_remove: media_devnode_remove(mctl->intf_devnode); -err2: +unregister_entity: media_device_unregister_entity(&mctl->media_entity); -err1: +free_mctl: kfree(mctl); return ret; } -- 2.5.0

2 1

[alsa-devel] [PATCH 2/2] ALSA: hda - hdmi defer to register acomp eld notifier
by libin.yang＠linux.intel.com 04 Mar '16

04 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> Defer to register acomp eld notifier until hdmi audio driver is fully ready. After registering eld notifier, gfx driver can use this callback function to notify audio driver the monitor connection event. However this action may happen when audio driver is adding the pins or doing other initialization. This is not always safe, however. For example, using per_pin->lock before the lock is initialized. Let's register the eld notifier after the initialization is done. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b5190e0..9bdb301 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2936,18 +2936,6 @@ static int patch_generic_hdmi(struct hda_codec *codec) is_broxton(codec)) codec->core.link_power_control = 1; - if (codec_has_acomp(codec)) { - codec->depop_delay = 0; - spec->i915_audio_ops.audio_ptr = codec; - /* intel_audio_codec_enable() or intel_audio_codec_disable() - * will call pin_eld_notify with using audio_ptr pointer - * We need make sure audio_ptr is really setup - */ - wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); - } - if (hdmi_parse_codec(codec) < 0) { if (spec->i915_bound) snd_hdac_i915_exit(&codec->bus->core); @@ -2969,6 +2957,18 @@ static int patch_generic_hdmi(struct hda_codec *codec) init_channel_allocations(); + if (codec_has_acomp(codec)) { + codec->depop_delay = 0; + spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); + spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + } + WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec)); return 0; } -- 1.9.1

1 0

[alsa-devel] Creative ca0132 on Gigabyte Z170X Gaming 7 motherboard: no sound
by Scott Doty 04 Mar '16

04 Mar '16

Hello, Didn't see a response on alsa-user, so thought I'd better roll up my sleeves and get to work. :) The ca0132 codec seems to do everything on my Gigabyte Z170X Gaming 7 motherboard but make sound. I can see the mixer settings, change them, even see the DSP settings -- but no sound out, either with HP or SPDIF. http://www.alsa-project.org/db/?f=6373811de293125e4e8c345c0c2d4a736c1fc2b8 I do see this in the startup messages: [ 9.314893] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 9.321116] snd_hda_intel 0000:00:1f.3: failed to add i915 component master (-19) I've tried all the fixes I could find reference to, as well as using iecset, but no joy. I also tried editing the driver to disable the "vga_switcheroo" flag, but that didn't work either. (Which is too bad, I thought that would be a quick fix that I could report. :/ ) If someone could point me in the right direction, I think I can investigate this and get it working. Any assistance greatly appreciated. Thanks, -Scott

2 2

[alsa-devel] [PATCH] Provide sequencer sound card number / PID via alsa-lib
by Martin Koegler 03 Mar '16

03 Mar '16

From: Martin Koegler <martin.koegler(a)chello.at> rawmidi devices expose the card number via IOCTLs, which allows to find the corresponding device in sysfs. The sequencer provides no identifing data. Chromium works around this issue by scanning rawmidi as well as sequencer devices and matching them by using assumtions, how the kernel register sequencer devices. This changes adds support for exposing the card number for kernel clients as well as the PID for user client. It supports kernels with and without the required support. Signed-off-by: Martin Koegler <martin.koegler(a)chello.at> --- include/seq.h | 2 ++ include/sound/asequencer.h | 6 ++++-- src/seq/seq.c | 26 ++++++++++++++++++++++++++ src/seq/seq_hw.c | 12 +++++++++++- 4 files changed, 43 insertions(+), 3 deletions(-) diff --git a/include/seq.h b/include/seq.h index 9576822..d05940e 100644 --- a/include/seq.h +++ b/include/seq.h @@ -143,6 +143,8 @@ snd_seq_client_type_t snd_seq_client_info_get_type(const snd_seq_client_info_t * const char *snd_seq_client_info_get_name(snd_seq_client_info_t *info); int snd_seq_client_info_get_broadcast_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info); +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info); const unsigned char *snd_seq_client_info_get_event_filter(const snd_seq_client_info_t *info); int snd_seq_client_info_get_num_ports(const snd_seq_client_info_t *info); int snd_seq_client_info_get_event_lost(const snd_seq_client_info_t *info); diff --git a/include/sound/asequencer.h b/include/sound/asequencer.h index 09c8a00..3ac70fd 100644 --- a/include/sound/asequencer.h +++ b/include/sound/asequencer.h @@ -24,7 +24,7 @@ /** version of the sequencer */ -#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 1) +#define SNDRV_SEQ_VERSION SNDRV_PROTOCOL_VERSION (1, 0, 2) /** * definition of sequencer event types @@ -356,7 +356,9 @@ struct snd_seq_client_info { unsigned char event_filter[32]; /* event filter bitmap */ int num_ports; /* RO: number of ports */ int event_lost; /* number of lost events */ - char reserved[64]; /* for future use */ + int card; /* RO: card number[kernel] */ + int pid; /* RO: pid[user] */ + char reserved[56]; /* for future use */ }; diff --git a/src/seq/seq.c b/src/seq/seq.c index 620ca3f..4405e68 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1522,6 +1522,32 @@ int snd_seq_client_info_get_error_bounce(const snd_seq_client_info_t *info) } /** + * \brief Get the sound card number. + * \param info client_info container + * \return card number or -1 if value is not available. + * + * Only available for SND_SEQ_KERNEL_CLIENT clients. + */ +int snd_seq_client_info_get_card(const snd_seq_client_info_t *info) +{ + assert(info); + return info->card; +} + +/** + * \brief Get the owning PID. + * \param info client_info container + * \return pid or -1 if value is not available. + * + * Only available for SND_SEQ_USER_CLIENT clients. + */ +int snd_seq_client_info_get_pid(const snd_seq_client_info_t *info) +{ + assert(info); + return info->pid; +} + +/** * \brief (DEPRECATED) Get the event filter bitmap of a client_info container * \param info client_info container * \return NULL if no event filter, or pointer to event filter bitmap diff --git a/src/seq/seq_hw.c b/src/seq/seq_hw.c index d033367..578ef12 100644 --- a/src/seq/seq_hw.c +++ b/src/seq/seq_hw.c @@ -32,10 +32,11 @@ const char *_snd_module_seq_hw = ""; #ifndef DOC_HIDDEN #define SNDRV_FILE_SEQ ALSA_DEVICE_DIRECTORY "seq" #define SNDRV_FILE_ALOADSEQ ALOAD_DEVICE_DIRECTORY "aloadSEQ" -#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 1) +#define SNDRV_SEQ_VERSION_MAX SNDRV_PROTOCOL_VERSION(1, 0, 2) typedef struct { int fd; + int version; } snd_seq_hw_t; #endif /* DOC_HIDDEN */ @@ -100,6 +101,10 @@ static int snd_seq_hw_get_client_info(snd_seq_t *seq, snd_seq_client_info_t * in /*SYSERR("SNDRV_SEQ_IOCTL_GET_CLIENT_INFO failed");*/ return -errno; } + if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -368,6 +373,10 @@ static int snd_seq_hw_query_next_client(snd_seq_t *seq, snd_seq_client_info_t *i /*SYSERR("SNDRV_SEQ_IOCTL_QUERY_NEXT_CLIENT failed");*/ return -errno; } + if (hw->version < SNDRV_PROTOCOL_VERSION(1, 0, 2)) { + info->card = -1; + info->pid = -1; + } return 0; } @@ -480,6 +489,7 @@ int snd_seq_hw_open(snd_seq_t **handle, const char *name, int streams, int mode) return -ENOMEM; } hw->fd = fd; + hw->version = ver; if (streams & SND_SEQ_OPEN_OUTPUT) { seq->obuf = (char *) malloc(seq->obufsize = SND_SEQ_OBUF_SIZE); if (!seq->obuf) { -- 2.1.4

1 0

[alsa-devel] [PATCH v3] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 03 Mar '16

03 Mar '16

This is the initial imx-wm8958 device-tree-only machine driver working with fsl_sai driver. This sound card has three dai link, HIFI, VOICE and BT dai. HIFI dai link will support codec master and slave mode. VOICE an BT dai link have dummy cpu dai, and just support codec master mode. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v2->v3: 1. move codec mclk enable to codec driver 2. remove rate constraints 3. remove GPIO configurations, which can be configured in device tree .../devicetree/bindings/sound/imx-audio-wm8958.txt | 49 +++ arch/arm/configs/imx_v6_v7_defconfig | 1 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 327 +++++++++++++++++++++ 5 files changed, 393 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..7a2b8a9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,49 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - cpu-dai : The phandle of an CPU DAI controller + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + + - fsl,hifi-dai-master : If present, hifi dai works as master, and will + provide bit clock and frame clock. Otherwise, + hifi dai works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8958-audio"; + cpu-dai = <&sai1>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; + + fsl,hifi-dai-master; +}; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..7ac5856 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -251,6 +251,7 @@ CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_PHYCORE_AC97=y CONFIG_SND_SOC_EUKREA_TLV320=y +CONFIG_SND_SOC_IMX_WM8958=y CONFIG_SND_SOC_IMX_WM8962=y CONFIG_SND_SOC_IMX_SGTL5000=y CONFIG_SND_SOC_IMX_SPDIF=y diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..01e7044 --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,327 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link *dai_link; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_hifi_dai_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + if (!data->is_hifi_dai_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (!data->is_hifi_dai_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == HIFI_DAI && !data->is_hifi_dai_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + .ops = &imx_voice_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + int ret, i; + + cpu_np = of_parse_phandle(np, "cpu-dai", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + return -EINVAL; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find cpu dai platform device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->dai_link = imx_wm8958_dai_link; + + /* + * AIF1 support codec master and slave mode + * AIF2 and AIF3 just support codec master mode + */ + if (of_property_read_bool(pdev->dev.of_node, "fsl,hifi-dai-master")) { + data->is_hifi_dai_master = true; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + data->is_hifi_dai_master = false; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "MCLK%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (!IS_ERR(data->mclk[i])) + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + + if (!data->mclk_freq[0] && !data->mclk_freq[1]) { + dev_err(&pdev->dev, "failed to get mclk clock\n"); + ret = -EINVAL; + goto fail; + } + + data->dai_link[HIFI_DAI].cpu_dai_name = dev_name(&cpu_pdev->dev); + data->dai_link[HIFI_DAI].platform_of_node = cpu_np; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = DAI_LINK_NUM; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ASoC: cht_bsw_rt5645: Fix writing to string literal
by Carlo Caione 03 Mar '16

03 Mar '16

From: Carlo Caione <carlo(a)endlessm.com> We cannot use strcpy() to write to a const char * location. This is causing a 'BUG: unable to handle kernel paging request' error at boot when using the cht-bsw-rt5645 driver. With this patch we also fix a wrong indexing in the driver where the codec_name of the wrong dai_link is being overwritten. Signed-off-by: Carlo Caione <carlo(a)endlessm.com> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 2d3afdd..a7b96a9 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -367,8 +367,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev) } card->dev = &pdev->dev; sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id); + /* set correct codec name */ - strcpy((char *)card->dai_link[2].codec_name, codec_name); + for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) + if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00")) + card->dai_link[i].codec_name = kstrdup(codec_name, GFP_KERNEL); + snd_soc_card_set_drvdata(card, drv); ret_val = devm_snd_soc_register_card(&pdev->dev, card); if (ret_val) { -- 2.5.0

5 6

Re: [alsa-devel] Compute Stick baytrail audio problem
by Giacomo Comes 03 Mar '16

03 Mar '16

On Tue, Feb 23, 2016 at 12:32:23PM +0900, Toyo Abe wrote: > > Hi, > > I'd struggled a similar underrun problem on my baytrail device. The dmesg showed > the following warnings. > > [15356.757938] had: Driver detected 2 missed buffer done interrupt(s)!!!! > [15356.761594] had: Driver detected 1 missed buffer done interrupt(s)!!!! > [15356.774775] had: Unable to clear UNDERRUN bits > [15356.791312] had: Driver detected 1 missed buffer done interrupt(s)!!!! > [15356.808148] had: Unable to clear UNDERRUN bits > [15356.824692] had: Driver detected 2 missed buffer done interrupt(s)!!!! > > After some investigation, I found some problems on the driver and fixed a few of > them. I pushed the fix to my github. I hope it might help you. > https://raw.githubusercontent.com/toyoabe/baytrailaudio/fix-underrun/hdmi_a… > > Best Regards, > -toyo Hi, that was it!. I added your changes and now the audio works. I'm happy. I got finally every subsystem of the Compute Stick working. I have a minor question tought. Alsamixer says: This sound device does not have any controls. Is there a way to change the volume without enabling pulseaudio? Just for documentation, here is the small change that Toyo did to make the underrun disappear: diff -Nraub a/drivers/gpu/drm/i915/i915_irq.c b/drivers/gpu/drm/i915/i915_irq.c --- a/drivers/gpu/drm/i915/i915_irq.c 2016-02-23 08:03:42.406016576 -0400 +++ b/drivers/gpu/drm/i915/i915_irq.c 2016-02-23 08:03:58.714016576 -0400 @@ -2717,11 +2717,14 @@ int pipe = 1; spin_lock_irqsave(&dev_priv->irq_lock, irqflags); + i915_enable_lpe_pipestat(dev_priv, pipe); + imr = I915_READ(VLV_IMR); /* Audio is on Stream A */ imr &= ~I915_LPE_PIPE_A_INTERRUPT; I915_WRITE(VLV_IMR, imr); - i915_enable_lpe_pipestat(dev_priv, pipe); + I915_WRITE(VLV_IER, ~imr); + POSTING_READ(VLV_IER); spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); return 0; @@ -2769,7 +2772,10 @@ spin_lock_irqsave(&dev_priv->irq_lock, irqflags); imr = I915_READ(VLV_IMR); imr |= I915_LPE_PIPE_A_INTERRUPT; + I915_WRITE(VLV_IER, ~imr); I915_WRITE(VLV_IMR, imr); + POSTING_READ(VLV_IMR); + i915_disable_lpe_pipestat(dev_priv, pipe); spin_unlock_irqrestore(&dev_priv->irq_lock, irqflags); diff -Nraub a/sound/hdmi_audio/intel_mid_hdmi_audio.c b/sound/hdmi_audio/intel_mid_hdmi_audio.c --- a/sound/hdmi_audio/intel_mid_hdmi_audio.c 2016-02-23 08:03:42.414016576 -0400 +++ b/sound/hdmi_audio/intel_mid_hdmi_audio.c 2016-02-23 08:03:58.722016576 -0400 @@ -1128,7 +1128,6 @@ pr_debug("HDMI status =0x%x\n", hdmi_status); if (hdmi_status & AUD_CONFIG_MASK_UNDERRUN) { i++; - hdmi_status &= ~AUD_CONFIG_MASK_UNDERRUN; had_write_register(AUD_HDMI_STATUS_v2, hdmi_status); } else break; Thanks again. Giacomo > > Pierre-Louis Bossart wrote: > > On 02/19/2016 02:39 PM, Giacomo Comes wrote: > >> Hi, > >> I have recently bought a Intel Compute Stick and I am having trouble with= > > the audio. > >> Intel provides a custom Ubuntu release that support audio, but I want to = > > run the > >> linux distribution of my choice on it (openSUSE 13.2 for the moment). > >> > >> In order to have HDMI audio support it is necessary to compile a kernel w= > > ith this patch > >> (https://raw.githubusercontent.com/01org/baytrailaudio/master/hdmi_audio_= > > 20150319.patch) > >> provided by Intel for kernel 3.16.7 (https://01.org/ubuntu-hdmi) > >> > >> I compiled the openSUSE kernel using the default openSUSE .config plus th= > > e following: > >> CONFIG_SUPPORT_HDMI=3Dy > >> CONFIG_SND_SOC=3Dm > >> CONFIG_SND_SOC_INTEL_SST=3Dm > >> CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH=3Dm > >> > >> I booted the new kernel and aplay -l gives this output: > >> **** List of PLAYBACK Hardware Devices **** > >> card 0: IntelHDMI [IntelHDMI], device 0: IntelHDMI [IntelHDMI] > >> Subdevices: 0/1 > >> Subdevice #0: subdevice #0 > >> > >> but when I run: > >> aplay /usr/share/sounds/alsa/test.wav > >> the audio plays for .2 seconds and then nothing more. > >> Sometime it will play later a little bit more and stop again. > >> At some point these messages appears: > >> underrun!!! (at least 29325.946 ms long) > >> underrun!!! (at least 1535.889 ms long) > >> underrun!!! (at least 3196.213 ms long) > >> > >> > >> I have looked around on google for a hint on how to fix this issue but I = > > found nothing. > >> Any idea where to look for a solution? > >> > >> Attached there are the dmesg and lsmod output for ubuntu > >> with the working audio and openSUSE 13.2 plus patch > >> without the working audio. > >> I can provide any other information you need. > > > > There are enough messages that make me believe something is very wrong > > from the start > > > > 19.664486] had: ******** HAD DRIVER loading.. Ver: 0.01.003 > > [ 19.664549] hdmi_audio_probe dma_mask: 0 > > [ 19.664937] card->dev is NULL!!!!! Should not be this case > > [ 19.668176] [drm] mid_hdmi_audio_register: Scheduling HDMI audio work qu= > > eue > > [ 19.670745] [drm:i915_had_wq] *ERROR* Checking for HDMI connection at bo= > > ot > > > > Some of us at Intel are trying to clean this up but it'll take time. > > > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

3 4

[alsa-devel] feeding back pkgsrc changs
by Thomas Klausner 03 Mar '16

03 Mar '16

Hi! I've just updated the pkgsrc packages for alsa-lib, alsa-utils and alsa-plugins to 1.1.0 and noticed that there are quite a number of local patches we have. I'd like to get them integrated. When I visited the alsa-projects web page, I noticed that the link to the bug tracker is dead. What is the recommended method to provide the patches? Thanks, Thomas

3 29

[alsa-devel] [PATCH 1/5] Introduce bswap.h for portable definitions of byte swap macros.
by Thomas Klausner 03 Mar '16

03 Mar '16

From: Thomas Klausner <wiz(a)NetBSD.org> Signed-off-by: Thomas Klausner <wiz(a)NetBSD.org> --- include/Makefile.am | 2 +- include/bswap.h | 39 +++++++++++++++++++++++++++++++++++++++ src/pcm/pcm_adpcm.c | 2 +- src/pcm/pcm_alaw.c | 2 +- src/pcm/pcm_copy.c | 2 +- src/pcm/pcm_dmix_generic.c | 2 +- src/pcm/pcm_file.c | 2 +- src/pcm/pcm_iec958.c | 2 +- src/pcm/pcm_lfloat.c | 2 +- src/pcm/pcm_linear.c | 2 +- src/pcm/pcm_meter.c | 2 +- src/pcm/pcm_misc.c | 2 +- src/pcm/pcm_mulaw.c | 2 +- src/pcm/pcm_null.c | 2 +- src/pcm/pcm_rate.c | 2 +- src/pcm/pcm_rate_linear.c | 2 +- src/pcm/pcm_route.c | 2 +- src/pcm/pcm_softvol.c | 2 +- 18 files changed, 56 insertions(+), 17 deletions(-) create mode 100644 include/bswap.h diff --git a/include/Makefile.am b/include/Makefile.am index 8c83c11..31a3f74 100644 --- a/include/Makefile.am +++ b/include/Makefile.am @@ -59,7 +59,7 @@ alsainclude_HEADERS += alisp.h endif noinst_HEADERS = alsa sys.h search.h list.h aserver.h local.h alsa-symbols.h \ - asoundlib-head.h asoundlib-tail.h type_compat.h + asoundlib-head.h asoundlib-tail.h bswap.h type_compat.h DISTCLEANFILES = stamp-vh version.h alsa asoundlib.h diff --git a/include/bswap.h b/include/bswap.h new file mode 100644 index 0000000..68adce3 --- /dev/null +++ b/include/bswap.h @@ -0,0 +1,39 @@ +/* + * ALSA lib - compatibility header for providing byte swapping macros + * Copyright (c) 2016 by Thomas Klausner <wiz(a)NetBSD.org> + * + * + * This library is free software; you can redistribute it and/or modify + * it under the terms of the GNU Lesser General Public License as + * published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#ifndef __BSWAP_H +#define __BSWAP_H + +#if defined(__FreeBSD__) || defined(__DragonFly__) || defined(__NetBSD__) +#include <sys/endian.h> +#define bswap_16 bswap16 +#define bswap_32 bswap32 +#define bswap_64 bswap64 +#elif defined (__sun) +#include <sys/byteorder.h> +#define bswap_16 BSWAP_16 +#define bswap_32 BSWAP_32 +#define bswap_64 BSWAP_64 +#else +#include <byteswap.h> +#endif + +#endif diff --git a/src/pcm/pcm_adpcm.c b/src/pcm/pcm_adpcm.c index 1a83c5a..1c88c83 100644 --- a/src/pcm/pcm_adpcm.c +++ b/src/pcm/pcm_adpcm.c @@ -56,7 +56,7 @@ IMA compatibility project proceedings, Vol 2, Issue 2, May 1992. come across a good description of XA yet. */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_alaw.c b/src/pcm/pcm_alaw.c index db759e3..fa58441 100644 --- a/src/pcm/pcm_alaw.c +++ b/src/pcm/pcm_alaw.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_copy.c b/src/pcm/pcm_copy.c index 66d3a47..0dbd5bd 100644 --- a/src/pcm/pcm_copy.c +++ b/src/pcm/pcm_copy.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_dmix_generic.c b/src/pcm/pcm_dmix_generic.c index 9e9d3c3..40c0874 100644 --- a/src/pcm/pcm_dmix_generic.c +++ b/src/pcm/pcm_dmix_generic.c @@ -125,7 +125,7 @@ static void mix_select_callbacks(snd_pcm_direct_t *dmix) (1ULL << SND_PCM_FORMAT_S24_LE) | (1ULL << SND_PCM_FORMAT_S24_3LE) | \ (1ULL << SND_PCM_FORMAT_U8)) -#include <byteswap.h> +#include "bswap.h" static void generic_mix_areas_16_native(unsigned int size, volatile signed short *dst, diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 9ab6964..92eb072 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <ctype.h> #include <string.h> #include "pcm_local.h" diff --git a/src/pcm/pcm_iec958.c b/src/pcm/pcm_iec958.c index 38c4ce7..86ac9cf 100644 --- a/src/pcm/pcm_iec958.c +++ b/src/pcm/pcm_iec958.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_lfloat.c b/src/pcm/pcm_lfloat.c index 2f3e578..22201f8 100644 --- a/src/pcm/pcm_lfloat.c +++ b/src/pcm/pcm_lfloat.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_linear.c b/src/pcm/pcm_linear.c index 9a92abd..50df779 100644 --- a/src/pcm/pcm_linear.c +++ b/src/pcm/pcm_linear.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index 034f582..1b0ccb4 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -27,7 +27,7 @@ */ -#include <byteswap.h> +#include "bswap.h" #include <time.h> #include <pthread.h> #include <dlfcn.h> diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index 5870f82..7d2b05d 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -23,7 +23,7 @@ #include <stdlib.h> #include <unistd.h> #include <string.h> -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" diff --git a/src/pcm/pcm_mulaw.c b/src/pcm/pcm_mulaw.c index 011b2a5..15dbdc1 100644 --- a/src/pcm/pcm_mulaw.c +++ b/src/pcm/pcm_mulaw.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index 5e63caa..685618e 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <limits.h> #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index c76db25..41bddac 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -28,7 +28,7 @@ * */ #include <inttypes.h> -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_rate.h" diff --git a/src/pcm/pcm_rate_linear.c b/src/pcm/pcm_rate_linear.c index 7481b38..70399e0 100644 --- a/src/pcm/pcm_rate_linear.c +++ b/src/pcm/pcm_rate_linear.c @@ -21,7 +21,7 @@ */ #include <inttypes.h> -#include <byteswap.h> +#include "bswap.h" #include "pcm_local.h" #include "pcm_plugin.h" #include "pcm_rate.h" diff --git a/src/pcm/pcm_route.c b/src/pcm/pcm_route.c index 646517d..3611603 100644 --- a/src/pcm/pcm_route.c +++ b/src/pcm/pcm_route.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <math.h> #include "pcm_local.h" #include "pcm_plugin.h" diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index c6cfd88..802aa4b 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -26,7 +26,7 @@ * */ -#include <byteswap.h> +#include "bswap.h" #include <math.h> #include "pcm_local.h" #include "pcm_plugin.h" -- 2.7.1

1 4

[alsa-devel] [PATCH v2 0/7] ALSA: hda: Move chmap helpers to core
by Subhransu S. Prusty 03 Mar '16

03 Mar '16

The HDMI channel map helper APIs and control registration can be reused by ASoC skylake HDMI drivers. A common chmap object is created, channel map control callback handlers, helper APIs and data structures are moved to the core. Driver object store an instance of this chmap object to register chmap controls and program channel allocation for HDMI infoframe. Driver registers ops for specific operations. changes in v2: - Splitted patches for easier review - Access most of chmap helpers through ops, thus reducing number of exported APIs for chmap support Subhransu S. Prusty (7): ALSA: hda - Create common chmap object ALSA: hda - Register chmap obj as priv data instead of codec ALSA: hda - Use hdmi name space for CEA spk alloc structure ALSA: hda - Add hdmi chmap verb programming ops to chmap object ALSA: hda - Use ops instead of directly accessing chmap helpers ALSA: hda - Move chmap support helpers/ops to core ALSA: hda - Use snd_hdmi namespace prefix for printing ch alloc api include/sound/hdmi_chmap.h | 75 ++++ sound/hda/Makefile | 2 +- sound/hda/hdmi_chmap.c | 791 ++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_eld.c | 31 +- sound/pci/hda/patch_hdmi.c | 830 +++++---------------------------------------- 5 files changed, 957 insertions(+), 772 deletions(-) create mode 100644 include/sound/hdmi_chmap.h create mode 100644 sound/hda/hdmi_chmap.c -- 1.9.1

3 14

[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Fix mic issues on Acer Aspire E1-472
by Simon South 03 Mar '16

03 Mar '16

This patch applies the microphone-related fix created for the Acer Aspire E1-572 to the E1-472 as well, as it uses the same Realtek ALC282 CODEC and demonstrates the same issues. This patch allows an external, headset microphone to be used and limits the gain on the (quite noisy) internal microphone. Signed-off-by: Simon South <simon(a)simonsouth.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1527fc..5c672f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5224,6 +5224,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), + SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), -- 2.5.0

2 1

[alsa-devel] [PATCH 0/5] ASoC: pxa: Fix enum ctl accesses
by Takashi Iwai 02 Mar '16

02 Mar '16

Hi, here is a patchset to correct the wrong type accesses in pxa drivers for enum ctls. I didn't put Cc to stable for them, supposing these are only for 32bit systems. If not, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (5): ASoC: pxa: corgi: Fix enum ctl accesses in a wrong type ASoC: pxa: magician: Fix enum ctl accesses in a wrong type ASoC: pxa: poodle: Fix enum ctl accesses in a wrong type ASoC: pxa: spitz: Fix enum ctl accesses in a wrong type ASoC: pxa: tosa: Fix enum ctl accesses in a wrong type sound/soc/pxa/corgi.c | 12 ++++++------ sound/soc/pxa/magician.c | 6 +++--- sound/soc/pxa/poodle.c | 12 ++++++------ sound/soc/pxa/spitz.c | 12 ++++++------ sound/soc/pxa/tosa.c | 12 ++++++------ 5 files changed, 27 insertions(+), 27 deletions(-) -- 2.7.2

3 11

[alsa-devel] [PATCH v4 22/22] sound/usb: Use Media Controller API to share media resources
by Shuah Khan 02 Mar '16

02 Mar '16

Change ALSA driver to use Media Controller API to share media resources with DVB and V4L2 drivers on a AU0828 media device. Media Controller specific initialization is done after sound card is registered. ALSA creates Media interface and entity function graph nodes for Control, Mixer, PCM Playback, and PCM Capture devices. snd_usb_hw_params() will call Media Controller enable source handler interface to request the media resource. If resource request is granted, it will release it from snd_usb_hw_free(). If resource is busy, -EBUSY is returned. Media specific cleanup is done in usb_audio_disconnect(). Signed-off-by: Shuah Khan <shuahkh(a)osg.samsung.com> --- Changes since v3: - Fixed Kconfig to handle the following 1. CONFIG_MEDIA_SUPPORT and CONFIG_SND_USB_AUDIO are both modules CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER is selected 2. CONFIG_MEDIA_SUPPORT=y and CONFIG_SND_USB_AUDIO=m CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER is selected 3. CONFIG_MEDIA_SUPPORT=y and CONFIG_SND_USB_AUDIO=y CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER is selected 4. CONFIG_MEDIA_SUPPORT=m and CONFIG_SND_USB_AUDIO=y This is when we don't want CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER selected sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 +++ sound/usb/card.h | 3 + sound/usb/media.c | 318 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/media.h | 72 +++++++++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 28 ++++- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + 11 files changed, 448 insertions(+), 5 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index a452ad7..d14bf41 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -15,6 +15,7 @@ config SND_USB_AUDIO select SND_RAWMIDI select SND_PCM select BITREVERSE + select SND_USB_AUDIO_USE_MEDIA_CONTROLLER if MEDIA_CONTROLLER && (MEDIA_SUPPORT=y || MEDIA_SUPPORT=SND_USB_AUDIO) help Say Y here to include support for USB audio and USB MIDI devices. @@ -22,6 +23,9 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. +config SND_USB_AUDIO_USE_MEDIA_CONTROLLER + bool + config SND_USB_UA101 tristate "Edirol UA-101/UA-1000 driver" select SND_PCM diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 2d2d122..8dca3c4 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -15,6 +15,8 @@ snd-usb-audio-objs := card.o \ quirks.o \ stream.o +snd-usb-audio-$(CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER) += media.o + snd-usbmidi-lib-objs := midi.o # Toplevel Module Dependency diff --git a/sound/usb/card.c b/sound/usb/card.c index 1f09d95..35fe256 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -66,6 +66,7 @@ #include "format.h" #include "power.h" #include "stream.h" +#include "media.h" MODULE_AUTHOR("Takashi Iwai <tiwai(a)suse.de>"); MODULE_DESCRIPTION("USB Audio"); @@ -561,6 +562,11 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) goto __error; + if (quirk->media_device) { + /* don't want to fail when media_device_create() fails */ + media_device_create(chip, intf); + } + usb_chip[chip->index] = chip; chip->num_interfaces++; usb_set_intfdata(intf, chip); @@ -617,6 +623,14 @@ static void usb_audio_disconnect(struct usb_interface *intf) list_for_each(p, &chip->midi_list) { snd_usbmidi_disconnect(p); } + /* + * Nice to check quirk && quirk->media_device + * need some special handlings. Doesn't look like + * we have access to quirk here + * Acceses mixer_list + */ + media_device_delete(chip); + /* release mixer resources */ list_for_each_entry(mixer, &chip->mixer_list, list) { snd_usb_mixer_disconnect(mixer); diff --git a/sound/usb/card.h b/sound/usb/card.h index 71778ca..34a0898 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -105,6 +105,8 @@ struct snd_usb_endpoint { struct list_head list; }; +struct media_ctl; + struct snd_usb_substream { struct snd_usb_stream *stream; struct usb_device *dev; @@ -156,6 +158,7 @@ struct snd_usb_substream { } dsd_dop; bool trigger_tstamp_pending_update; /* trigger timestamp being updated from initial estimate */ + struct media_ctl *media_ctl; }; struct snd_usb_stream { diff --git a/sound/usb/media.c b/sound/usb/media.c new file mode 100644 index 0000000..cff1459 --- /dev/null +++ b/sound/usb/media.c @@ -0,0 +1,318 @@ +/* + * media.c - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ + +#include <linux/init.h> +#include <linux/list.h> +#include <linux/mutex.h> +#include <linux/slab.h> +#include <linux/usb.h> + +#include <sound/pcm.h> +#include <sound/core.h> + +#include "usbaudio.h" +#include "card.h" +#include "mixer.h" +#include "media.h" + +static int media_enable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->enable_source) + return mctl->media_dev->enable_source(&mctl->media_entity, + &mctl->media_pipe); + return 0; +} + +static void media_disable_source(struct media_ctl *mctl) +{ + if (mctl && mctl->media_dev->disable_source) + mctl->media_dev->disable_source(&mctl->media_entity); +} + +int media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream) +{ + struct media_device *mdev; + struct media_ctl *mctl; + struct device *pcm_dev = &pcm->streams[stream].dev; + u32 intf_type; + int ret = 0; + u16 mixer_pad; + struct media_entity *entity; + + mdev = subs->stream->chip->media_dev; + if (!mdev) + return -ENODEV; + + if (subs->media_ctl) + return 0; + + /* allocate media_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + intf_type = MEDIA_INTF_T_ALSA_PCM_PLAYBACK; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_PLAYBACK; + mctl->media_pad.flags = MEDIA_PAD_FL_SOURCE; + mixer_pad = 1; + } else { + intf_type = MEDIA_INTF_T_ALSA_PCM_CAPTURE; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_CAPTURE; + mctl->media_pad.flags = MEDIA_PAD_FL_SINK; + mixer_pad = 2; + } + mctl->media_entity.name = pcm->name; + media_entity_pads_init(&mctl->media_entity, 1, &mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) + goto err1; + + mctl->intf_devnode = media_devnode_create(mdev, intf_type, 0, + MAJOR(pcm_dev->devt), + MINOR(pcm_dev->devt)); + if (!mctl->intf_devnode) { + ret = -ENOMEM; + goto err2; + } + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &mctl->intf_devnode->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + ret = -ENOMEM; + goto err3; + } + + /* create link between mixer and audio */ + media_device_for_each_entity(entity, mdev) { + switch (entity->function) { + case MEDIA_ENT_F_AUDIO_MIXER: + ret = media_create_pad_link(entity, mixer_pad, + &mctl->media_entity, 0, + MEDIA_LNK_FL_ENABLED); + if (ret) + goto err4; + break; + } + } + + subs->media_ctl = mctl; + return 0; + +err4: + media_remove_intf_link(mctl->intf_link); +err3: + media_devnode_remove(mctl->intf_devnode); +err2: + media_device_unregister_entity(&mctl->media_entity); +err1: + kfree(mctl); + return ret; +} + +void media_stream_delete(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl && mctl->media_dev) { + struct media_device *mdev; + + mdev = subs->stream->chip->media_dev; + if (mdev && media_devnode_is_registered(&mdev->devnode)) { + media_devnode_remove(mctl->intf_devnode); + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + subs->media_ctl = NULL; + } +} + +int media_start_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + return media_enable_source(mctl); + return 0; +} + +void media_stop_pipeline(struct snd_usb_substream *subs) +{ + struct media_ctl *mctl = subs->media_ctl; + + if (mctl) + media_disable_source(mctl); +} + +int media_mixer_init(struct snd_usb_audio *chip) +{ + struct device *ctl_dev = &chip->card->ctl_dev; + struct media_intf_devnode *ctl_intf; + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + struct media_mixer_ctl *mctl; + u32 intf_type = MEDIA_INTF_T_ALSA_CONTROL; + int ret; + + if (!mdev) + return -ENODEV; + + ctl_intf = chip->ctl_intf_media_devnode; + if (!ctl_intf) { + ctl_intf = media_devnode_create(mdev, intf_type, 0, + MAJOR(ctl_dev->devt), + MINOR(ctl_dev->devt)); + if (!ctl_intf) + return -ENOMEM; + chip->ctl_intf_media_devnode = ctl_intf; + } + + list_for_each_entry(mixer, &chip->mixer_list, list) { + + if (mixer->media_mixer_ctl) + continue; + + /* allocate media_mixer_ctl */ + mctl = kzalloc(sizeof(*mctl), GFP_KERNEL); + if (!mctl) + return -ENOMEM; + + mctl->media_dev = mdev; + mctl->media_entity.function = MEDIA_ENT_F_AUDIO_MIXER; + mctl->media_entity.name = chip->card->mixername; + mctl->media_pad[0].flags = MEDIA_PAD_FL_SINK; + mctl->media_pad[1].flags = MEDIA_PAD_FL_SOURCE; + mctl->media_pad[2].flags = MEDIA_PAD_FL_SOURCE; + media_entity_pads_init(&mctl->media_entity, MEDIA_MIXER_PAD_MAX, + mctl->media_pad); + ret = media_device_register_entity(mctl->media_dev, + &mctl->media_entity); + if (ret) { + kfree(mctl); + return ret; + } + + mctl->intf_link = media_create_intf_link(&mctl->media_entity, + &ctl_intf->intf, + MEDIA_LNK_FL_ENABLED); + if (!mctl->intf_link) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + kfree(mctl); + return -ENOMEM; + } + mctl->intf_devnode = ctl_intf; + mixer->media_mixer_ctl = mctl; + } + return 0; +} + +static void media_mixer_delete(struct snd_usb_audio *chip) +{ + struct usb_mixer_interface *mixer; + struct media_device *mdev = chip->media_dev; + + if (!mdev) + return; + + list_for_each_entry(mixer, &chip->mixer_list, list) { + struct media_mixer_ctl *mctl; + + mctl = mixer->media_mixer_ctl; + if (!mixer->media_mixer_ctl) + continue; + + if (media_devnode_is_registered(&mdev->devnode)) { + media_device_unregister_entity(&mctl->media_entity); + media_entity_cleanup(&mctl->media_entity); + } + kfree(mctl); + mixer->media_mixer_ctl = NULL; + } + if (media_devnode_is_registered(&mdev->devnode)) + media_devnode_remove(chip->ctl_intf_media_devnode); + chip->ctl_intf_media_devnode = NULL; +} + +int media_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) +{ + struct media_device *mdev; + struct usb_device *usbdev = interface_to_usbdev(iface); + int ret; + + mdev = media_device_get_devres(&usbdev->dev); + if (!mdev) + return -ENOMEM; + if (!mdev->dev) { + /* register media device */ + mdev->dev = &usbdev->dev; + if (usbdev->product) + strlcpy(mdev->model, usbdev->product, + sizeof(mdev->model)); + if (usbdev->serial) + strlcpy(mdev->serial, usbdev->serial, + sizeof(mdev->serial)); + strcpy(mdev->bus_info, usbdev->devpath); + mdev->hw_revision = le16_to_cpu(usbdev->descriptor.bcdDevice); + media_device_init(mdev); + } + if (!media_devnode_is_registered(&mdev->devnode)) { + ret = media_device_register(mdev); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't register media device. Error: %d\n", + ret); + return ret; + } + } + + /* save media device - avoid lookups */ + chip->media_dev = mdev; + + /* Create media entities for mixer and control dev */ + ret = media_mixer_init(chip); + if (ret) { + dev_err(&usbdev->dev, + "Couldn't create media mixer entities. Error: %d\n", + ret); + + /* clear saved media_dev */ + chip->media_dev = NULL; + + return ret; + } + return 0; +} + +void media_device_delete(struct snd_usb_audio *chip) +{ + struct media_device *mdev = chip->media_dev; + + media_mixer_delete(chip); + + if (mdev) { + if (media_devnode_is_registered(&mdev->devnode)) + media_device_unregister(mdev); + chip->media_dev = NULL; + } +} diff --git a/sound/usb/media.h b/sound/usb/media.h new file mode 100644 index 0000000..f66b4d7 --- /dev/null +++ b/sound/usb/media.h @@ -0,0 +1,72 @@ +/* + * media.h - Media Controller specific ALSA driver code + * + * Copyright (c) 2016 Shuah Khan <shuahkh(a)osg.samsung.com> + * Copyright (c) 2016 Samsung Electronics Co., Ltd. + * + * This file is released under the GPLv2. + */ + +/* + * This file adds Media Controller support to ALSA driver + * to use the Media Controller API to share tuner with DVB + * and V4L2 drivers that control media device. Media device + * is created based on existing quirks framework. Using this + * approach, the media controller API usage can be added for + * a specific device. +*/ +#ifndef __MEDIA_H + +#ifdef CONFIG_SND_USB_AUDIO_USE_MEDIA_CONTROLLER + +#include <media/media-device.h> +#include <media/media-entity.h> +#include <sound/asound.h> + +struct media_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad; + struct media_pipeline media_pipe; +}; + +/* + * One source pad each for SNDRV_PCM_STREAM_CAPTURE and + * SNDRV_PCM_STREAM_PLAYBACK. One for sink pad to link + * to AUDIO Source +*/ +#define MEDIA_MIXER_PAD_MAX (SNDRV_PCM_STREAM_LAST + 2) + +struct media_mixer_ctl { + struct media_device *media_dev; + struct media_entity media_entity; + struct media_intf_devnode *intf_devnode; + struct media_link *intf_link; + struct media_pad media_pad[MEDIA_MIXER_PAD_MAX]; + struct media_pipeline media_pipe; +}; + +int media_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface); +void media_device_delete(struct snd_usb_audio *chip); +int media_stream_init(struct snd_usb_substream *subs, struct snd_pcm *pcm, + int stream); +void media_stream_delete(struct snd_usb_substream *subs); +int media_start_pipeline(struct snd_usb_substream *subs); +void media_stop_pipeline(struct snd_usb_substream *subs); +#else +static inline int media_device_create(struct snd_usb_audio *chip, + struct usb_interface *iface) + { return 0; } +static inline void media_device_delete(struct snd_usb_audio *chip) { } +static inline int media_stream_init(struct snd_usb_substream *subs, + struct snd_pcm *pcm, int stream) + { return 0; } +static inline void media_stream_delete(struct snd_usb_substream *subs) { } +static inline int media_start_pipeline(struct snd_usb_substream *subs) + { return 0; } +static inline void media_stop_pipeline(struct snd_usb_substream *subs) { } +#endif +#endif /* __MEDIA_H */ diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index 3417ef3..f378944 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -3,6 +3,8 @@ #include <sound/info.h> +struct media_mixer_ctl; + struct usb_mixer_interface { struct snd_usb_audio *chip; struct usb_host_interface *hostif; @@ -22,6 +24,7 @@ struct usb_mixer_interface { struct urb *rc_urb; struct usb_ctrlrequest *rc_setup_packet; u8 rc_buffer[6]; + struct media_mixer_ctl *media_mixer_ctl; }; #define MAX_CHANNELS 16 /* max logical channels */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 9245f52..c3b8486 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -35,6 +35,7 @@ #include "pcm.h" #include "clock.h" #include "power.h" +#include "media.h" #define SUBSTREAM_FLAG_DATA_EP_STARTED 0 #define SUBSTREAM_FLAG_SYNC_EP_STARTED 1 @@ -715,10 +716,14 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, struct audioformat *fmt; int ret; + ret = media_start_pipeline(subs); + if (ret) + return ret; + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, params_buffer_bytes(hw_params)); if (ret < 0) - return ret; + goto err_ret; subs->pcm_format = params_format(hw_params); subs->period_bytes = params_period_bytes(hw_params); @@ -732,22 +737,27 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, dev_dbg(&subs->dev->dev, "cannot set format: format = %#x, rate = %d, channels = %d\n", subs->pcm_format, subs->cur_rate, subs->channels); - return -EINVAL; + ret = -EINVAL; + goto err_ret; } ret = snd_usb_lock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; ret = set_format(subs, fmt); snd_usb_unlock_shutdown(subs->stream->chip); if (ret < 0) - return ret; + goto err_ret; subs->interface = fmt->iface; subs->altset_idx = fmt->altset_idx; subs->need_setup_ep = true; return 0; + +err_ret: + media_stop_pipeline(subs); + return ret; } /* @@ -759,6 +769,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs = substream->runtime->private_data; + media_stop_pipeline(subs); subs->cur_audiofmt = NULL; subs->cur_rate = 0; subs->period_bytes = 0; @@ -1219,6 +1230,7 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usb_substream *subs = &as->substream[direction]; + int ret; subs->interface = -1; subs->altset_idx = 0; @@ -1232,7 +1244,12 @@ static int snd_usb_pcm_open(struct snd_pcm_substream *substream, int direction) subs->dsd_dop.channel = 0; subs->dsd_dop.marker = 1; - return setup_hw_info(runtime, subs); + ret = setup_hw_info(runtime, subs); + if (ret == 0) + ret = media_stream_init(subs, as->pcm, direction); + if (ret) + snd_usb_autosuspend(subs->stream->chip); + return ret; } static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) @@ -1241,6 +1258,7 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) struct snd_usb_substream *subs = &as->substream[direction]; stop_endpoints(subs, true); + media_stop_pipeline(subs); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c60a776..9d087b1 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2886,6 +2886,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .product_name = pname, \ .ifnum = QUIRK_ANY_INTERFACE, \ .type = QUIRK_AUDIO_ALIGN_TRANSFER, \ + .media_device = 1, \ } \ } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index c4dc577..6c7f5c1 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -36,6 +36,7 @@ #include "format.h" #include "clock.h" #include "stream.h" +#include "media.h" /* * free a substream @@ -52,6 +53,7 @@ static void free_substream(struct snd_usb_substream *subs) kfree(fp); } kfree(subs->rate_list.list); + media_stream_delete(subs); } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b665d85..a161c7c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -30,6 +30,9 @@ * */ +struct media_device; +struct media_intf_devnode; + struct snd_usb_audio { int index; struct usb_device *dev; @@ -60,6 +63,8 @@ struct snd_usb_audio { bool autoclock; /* from the 'autoclock' module param */ struct usb_host_interface *ctrl_intf; /* the audio control interface */ + struct media_device *media_dev; + struct media_intf_devnode *ctl_intf_media_devnode; }; #define usb_audio_err(chip, fmt, args...) \ @@ -110,6 +115,7 @@ struct snd_usb_audio_quirk { const char *product_name; int16_t ifnum; uint16_t type; + bool media_device; const void *data; }; -- 2.5.0

3 4

[alsa-devel] [PATCH] ASoC: intel: remove unused variable
by Sudip Mukherjee 02 Mar '16

02 Mar '16

The variable cmd_id is only assigned some value and is never used. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/soc/intel/atom/sst/sst_ipc.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3dc7358..8afa6fe 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -318,7 +318,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, union ipc_header_high msg_high; u32 msg_low; struct ipc_dsp_hdr *dsp_hdr; - unsigned int cmd_id; msg_high = msg->mrfld_header.p.header_high; msg_low = msg->mrfld_header.p.header_low_payload; @@ -357,7 +356,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, return; /* Copy command id so that we can use to put sst to reset */ dsp_hdr = (struct ipc_dsp_hdr *)data; - cmd_id = dsp_hdr->cmd_id; dev_dbg(sst_drv_ctx->dev, "cmd_id %d\n", dsp_hdr->cmd_id); if (sst_wake_up_block(sst_drv_ctx, msg_high.part.result, msg_high.part.drv_id, -- 1.9.1

4 13

[alsa-devel] [PATCH] ALSA: portman2x4: fix NULL pointer dereference
by Sudip Mukherjee 02 Mar '16

02 Mar '16

While registering pardev, the irq_func was also registered. As a result when we tried to probe for the card, an interrupt was generated and in the ISR we tried to dereference private_data. But private_data is still NULL as we have not yet done portman_create(). Lets probe for the device after card is created. Fixes: e6a1b7e88046 ("ALSA: portman2x4 - use new parport device model") Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/drivers/portman2x4.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 362533f..189e3e7 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -762,11 +762,6 @@ static int snd_portman_probe(struct platform_device *pdev) err = -EIO; goto free_pardev; } - err = portman_probe(p); - if (err) { - err = -EIO; - goto release_pardev; - } if ((err = portman_create(card, pardev, &pm)) < 0) { snd_printd("Cannot create main component\n"); @@ -774,6 +769,12 @@ static int snd_portman_probe(struct platform_device *pdev) } card->private_data = pm; card->private_free = snd_portman_card_private_free; + + err = portman_probe(p); + if (err) { + err = -EIO; + goto __err; + } if ((err = snd_portman_rawmidi_create(card)) < 0) { snd_printd("Creating Rawmidi component failed\n"); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: distinguish the different SSC
by Songjun Wu 02 Mar '16

02 Mar '16

Cpu_dai id always equals 0, can't distinguish the different SSC. Use platform_device id to record and distinguish the different SSC. Signed-off-by: Songjun Wu <songjun.wu(a)atmel.com> --- drivers/misc/atmel-ssc.c | 1 + sound/soc/atmel/atmel_ssc_dai.c | 27 ++++++++++++++++++--------- 2 files changed, 19 insertions(+), 9 deletions(-) diff --git a/drivers/misc/atmel-ssc.c b/drivers/misc/atmel-ssc.c index e11a0bd..0516ecd 100644 --- a/drivers/misc/atmel-ssc.c +++ b/drivers/misc/atmel-ssc.c @@ -34,6 +34,7 @@ struct ssc_device *ssc_request(unsigned int ssc_num) if (ssc->pdev->dev.of_node) { if (of_alias_get_id(ssc->pdev->dev.of_node, "ssc") == ssc_num) { + ssc->pdev->id = ssc_num; ssc_valid = 1; break; } diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index ba8def5..2768970 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -285,7 +285,8 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, static int atmel_ssc_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; int ret; @@ -346,7 +347,8 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir, dir_mask; @@ -392,7 +394,8 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; ssc_p->daifmt = fmt; return 0; @@ -404,7 +407,8 @@ static int atmel_ssc_set_dai_fmt(struct snd_soc_dai *cpu_dai, static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { - struct atmel_ssc_info *ssc_p = &ssc_info[cpu_dai->id]; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; switch (div_id) { case ATMEL_SSC_CMR_DIV: @@ -445,7 +449,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int id = dai->id; + struct platform_device *pdev = to_platform_device(dai->dev); + int id = pdev->id; struct atmel_ssc_info *ssc_p = &ssc_info[id]; struct ssc_device *ssc = ssc_p->ssc; struct atmel_pcm_dma_params *dma_params; @@ -772,7 +777,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, static int atmel_ssc_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir; @@ -795,7 +801,8 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, static int atmel_ssc_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct platform_device *pdev = to_platform_device(dai->dev); + struct atmel_ssc_info *ssc_p = &ssc_info[pdev->id]; struct atmel_pcm_dma_params *dma_params; int dir; @@ -824,11 +831,12 @@ static int atmel_ssc_trigger(struct snd_pcm_substream *substream, static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) { struct atmel_ssc_info *ssc_p; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); if (!cpu_dai->active) return 0; - ssc_p = &ssc_info[cpu_dai->id]; + ssc_p = &ssc_info[pdev->id]; /* Save the status register before disabling transmit and receive */ ssc_p->ssc_state.ssc_sr = ssc_readl(ssc_p->ssc->regs, SR); @@ -852,12 +860,13 @@ static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai) { struct atmel_ssc_info *ssc_p; + struct platform_device *pdev = to_platform_device(cpu_dai->dev); u32 cr; if (!cpu_dai->active) return 0; - ssc_p = &ssc_info[cpu_dai->id]; + ssc_p = &ssc_info[pdev->id]; /* restore SSC register settings */ ssc_writel(ssc_p->ssc->regs, TFMR, ssc_p->ssc_state.ssc_tfmr); -- 1.7.9.5

4 4

[alsa-devel] [PATCH 1/2] ASoC: sh: rcar: core: don't open code of_device_get_match_data()
by Wolfram Sang 02 Mar '16

02 Mar '16

From: Wolfram Sang <wsa+renesas(a)sang-engineering.com> This change will also make Coverity happy by avoiding a theoretical NULL pointer dereference; yet another reason is to use the above helper function to tighten the code and make it more readable. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- Compile tested only. I am on the road and can't test the multimedia drivers because of no cables :( If someone could test it, that would be much appreciated. Or I'll do it next week. The pattern worked for other drivers I could actually test, though. sound/soc/sh/rcar/core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 5227aad43e3854..3dae30710c34f2 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1119,7 +1119,6 @@ static int rsnd_probe(struct platform_device *pdev) struct rsnd_priv *priv; struct device *dev = &pdev->dev; struct rsnd_dai *rdai; - const struct of_device_id *of_id = of_match_device(rsnd_of_match, dev); int (*probe_func[])(struct rsnd_priv *priv) = { rsnd_gen_probe, rsnd_dma_probe, @@ -1145,7 +1144,7 @@ static int rsnd_probe(struct platform_device *pdev) } priv->pdev = pdev; - priv->flags = (unsigned long)of_id->data; + priv->flags = (unsigned long)of_device_get_match_data(dev); spin_lock_init(&priv->lock); /* -- 2.7.0

3 6

[alsa-devel] [PATCH v3 0/4] add IEC958 channel status control helper
by Arnaud Pouliquen 02 Mar '16

02 Mar '16

- Define helpers function to handle 'IEC958 Playback Default' control - add "pcm_new" ops for DAI initialisations that need pcm runtime context. This patchset is needed to be able to associate control to the PCM device. The iec control is linked to the DAI but also to the PCM device. Furthermore, some platforms like sti support both HDMI and SPDIF outputs in parallel. Logical way to differentiate them is to link them to pcm device index. V3: This patchset is extracted from following RFC http://permalink.gmane.org/gmane.linux.alsa.devel/149876 (sti: add audio interface to the hdmi driver) Notice that "ASoC: hdmi-codec: add IEC control" patch depends on integration of the hdmi-codec driver Patches update: - ALSA: pcm: add IEC958 channel status control helper - make mutex usage mandatory. - Set control index to pcm device id ( needed to be supported by iecset) - ASoC: core: add code to complete dai init after pcm creation - fix dai param of pcm_new ops for cpu dai. - ASoC: sti: use iec channel status control helper new patch: implementation of the helper for sti cpu-dai V2: - patch: ALSA: pcm: add IEC958 channel status control helper - Return 1 instead of 0 in snd_pcm_iec958_put - Add .access field in control structure - I have kept condition on mutex for flexibility (but could be cleaned to force user to use a mutex) V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") Arnaud Pouliquen (4): ALSA: pcm: add IEC958 channel status control helper ASoC: core: add code to complete dai init after pcm creation ASoC: sti: use iec channel status control helper ASoC: hdmi-codec: add IEC control. include/sound/hdmi-codec.h | 1 + include/sound/pcm_iec958.h | 15 ++++++ include/sound/soc-dai.h | 7 +++ sound/core/pcm_iec958.c | 107 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdmi-codec.c | 59 ++++++++++++++++------ sound/soc/soc-core.c | 14 ++++++ sound/soc/sti/Kconfig | 1 + sound/soc/sti/uniperif_player.c | 79 ++++++++++++----------------- 8 files changed, 219 insertions(+), 62 deletions(-) -- 1.9.1

3 11

[alsa-devel] sound: uninterruptible hang in snd_seq_oss_writeq_sync
by Dmitry Vyukov 02 Mar '16

02 Mar '16

Hello, The following program creates an unkillable process: // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> #ifndef SYS_mmap #define SYS_mmap 9 #endif #ifndef SYS_syz_open_dev #define SYS_syz_open_dev 1000001 #endif #ifndef SYS_write #define SYS_write 1 #endif #ifndef SYS_read #define SYS_read 0 #endif #ifndef SYS_sync_file_range #define SYS_sync_file_range 277 #endif long r[61]; int main() { memset(r, -1, sizeof(r)); r[0] = syscall(SYS_mmap, 0x20000000ul, 0x5e000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); r[2] = syscall(SYS_open, "/dev/sequencer", 0x202ul, 0, 0, 0); *(uint8_t*)0x20057000 = (uint8_t)0x7; *(uint8_t*)0x20057001 = (uint8_t)0x6; *(uint8_t*)0x20057002 = (uint8_t)0x3; *(uint8_t*)0x20057003 = (uint8_t)0x99be; *(uint32_t*)0x2005700c = (uint32_t)0x4; *(uint8_t*)0x20057010 = (uint8_t)0x5; *(uint8_t*)0x20057011 = (uint8_t)0xffffffffffffffff; *(uint8_t*)0x20057012 = (uint8_t)0x100; *(uint8_t*)0x20057013 = (uint8_t)0x1; *(uint8_t*)0x20057024 = (uint8_t)0x8; *(uint8_t*)0x20057025 = (uint8_t)0xabe4; *(uint16_t*)0x20057026 = (uint16_t)0x3; *(uint64_t*)0x20057028 = (uint64_t)0x2005055c; *(uint8_t*)0x20057030 = (uint8_t)0xffffffffffffffe0; *(uint8_t*)0x20057031 = (uint8_t)0xffffffffffffffb8; *(uint8_t*)0x20057032 = (uint8_t)0x8; *(uint8_t*)0x20057033 = (uint8_t)0x7; *(uint32_t*)0x2005703c = (uint32_t)0x5; *(uint8_t*)0x20057040 = (uint8_t)0x5; *(uint8_t*)0x20057041 = (uint8_t)0x401; *(uint8_t*)0x20057042 = (uint8_t)0x8; *(uint8_t*)0x20057043 = (uint8_t)0x4; *(uint8_t*)0x2005704c = (uint8_t)0x0; *(uint8_t*)0x2005704d = (uint8_t)0x3; *(uint8_t*)0x2005704e = (uint8_t)0x200; *(uint8_t*)0x2005704f = (uint8_t)0xee; *(uint8_t*)0x20057050 = (uint8_t)0x2; *(uint8_t*)0x20057051 = (uint8_t)0x8; *(uint8_t*)0x20057052 = (uint8_t)0x800; *(uint8_t*)0x20057053 = (uint8_t)0xfffffffffffff000; *(uint64_t*)0x20057068 = (uint64_t)0x77359400; *(uint64_t*)0x20057070 = (uint64_t)0x0; *(uint8_t*)0x20057078 = (uint8_t)0x10000; *(uint8_t*)0x20057079 = (uint8_t)0x946; *(uint8_t*)0x2005707a = (uint8_t)0x1; *(uint8_t*)0x2005707b = (uint8_t)0x1ff; *(uint8_t*)0x2005708c = (uint8_t)0x3; *(uint32_t*)0x20057090 = (uint32_t)0x7; *(uint32_t*)0x20057094 = (uint32_t)0x1; *(uint8_t*)0x20057098 = (uint8_t)0x2; *(uint8_t*)0x20057099 = (uint8_t)0x7ff; *(uint8_t*)0x2005709a = (uint8_t)0x8; *(uint8_t*)0x2005709b = (uint8_t)0xdd8; *(uint64_t*)0x200570b0 = (uint64_t)0x1; *(uint64_t*)0x200570b8 = (uint64_t)0x0; *(uint8_t*)0x200570c0 = (uint8_t)0x9; *(uint8_t*)0x200570c1 = (uint8_t)0x3; *(uint8_t*)0x200570c2 = (uint8_t)0x7ff; *(uint8_t*)0x200570c3 = (uint8_t)0x8; *(uint32_t*)0x200570d4 = (uint32_t)0xec; *(uint64_t*)0x200570d8 = (uint64_t)0x20053f14; memcpy((void*)0x2005055c, "\xa4\x62\xd2\xb0\x88\xe0\x2b\xbd\xea\xa1\xa5\xf7\x7b\x89\xca" "\xb0\x43\xca\x19\x6d\x70\xb6\xa6\xcd\x30\xaa\x98\x07\xb3\xf3" "\xea\xf8\x44\x19\xc8\x83\xbc\xb7\x33\xd1\xd0\x5e\x8f\x34\x12" "\x82\xb4\xf8\xbd\x02\x63\x0b\xe7\x45\x23\x40\xa4\x51\x40\x23" "\x01\xcc\x57\x89\x40\x9c\xc1\x29\x36\xe2\xc0\x70\xe5\xda\xc9" "\x1f\x40\x25\xe0\x74\x98\x07\x75\xe9\x5a\xfc\x0e\x10\x3f\xd9" "\x4b\xc3\xc2\x48\xfa\x18\xb8\xda\x49\x3e\xeb\x69\x2c\x4b\xd2" "\xb7\xac\xc6\x5a\x1c\xfd", 111); memcpy((void*)0x20053f14, "\x1b\x4b\xa5\x94\xeb\xe1\xd9\x4b\x57\xf2\xeb\xab\xdd\xe5\xba" "\x24\xf2\xc8\x36\x90\x88\x21\x2c\xec\x8e\x5a\x0d\x24\x0f\x48" "\xcc\x0d\x3c\xab\x21\x8e\xfa\x8f\x0d\xcd\x8d\x14\x99\xc3\x64" "\xea\x7f\xf3\x88\xe9\xf7\x4f\x4c\x12\xe9\x0e\x23\xad\x73\x7d" "\xa3\x4c\x45\x3f\xb3\x48\xb7\x44\x89\x41\x20\x7b\x2c\xbd\xf2" "\x89\x16\x1f\xbd\x80\x24\x72\xa1\x8d\x4b\x13\x74\xb2\xf4\xdf" "\x49\xe0\x6c\x40\x0d\xf2\x7c\x8a\xbf\xf2\xb1\x62\xe0\x7e\x51" "\x9a\xd4\x60\x64\x49\x96\x4b\x83\x91\xe5\x51\x50\x94\x3c\x41" "\xf6\xd1\xef\x1f\x90\x5a\x48\xa5\xe2\xf8\xe3\xd9\xf9\x82\x5e" "\xa1\x5c\x85\x1c\xbe\x47\xa2\x29\xb7\x5a\xf4\x14\x24\x92\x11" "\x4b\x6b\x54\x95\x5e\x9f\x1b\xda\x02\xa3\x84\xb3\x2b\x07\x13" "\x18\x6b\x31\xb4\xc1\x04\x30\x5f\xa0\xa2\xe9\xe6\x50\x6f\xed" "\x15\x10\xf5\xda\xa1\xe5\x06\xaf\x28\xe0\x53\xea\x88\x49\xf8" "\x97\x52\x4b\xe8\xa2\xce\x9f\xc7\x47\x4f\xd6\xc0\xe2\xb7\xc1" "\x7f\xa9\x54\x3d\x11\xe4\x99\x28\x65\x39\xb1\xf3\xec\x48\x43" "\x58\x19\x12\xf6\x38\x5b\xee\xca\x33\x8e\x4d", 236); r[56] = syscall(SYS_write, r[2], 0x20057000ul, 0x9cul, 0, 0, 0); memcpy((void*)0x2005bfa7, "\x2f\x64\x65\x76\x2f\x76\x68\x63\x69\x00", 10); return 0; } The hang stack is: [<ffffffff85309f77>] snd_seq_oss_writeq_sync+0x327/0x790 sound/core/seq/oss/seq_oss_writeq.c:121 [<ffffffff852fa353>] snd_seq_oss_drain_write+0x113/0x160 sound/core/seq/oss/seq_oss_init.c:448 [<ffffffff852f8d04>] odev_release+0x44/0x80 sound/core/seq/oss/seq_oss.c:152 [<ffffffff817ccdc6>] __fput+0x236/0x780 fs/file_table.c:208 [<ffffffff817cd395>] ____fput+0x15/0x20 fs/file_table.c:244 [<ffffffff813b8db0>] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [<ffffffff81363ca0>] do_exit+0xaf0/0x2d20 kernel/exit.c:748 [<ffffffff81366048>] do_group_exit+0x108/0x330 kernel/exit.c:878 [< inline >] SYSC_exit_group kernel/exit.c:889 [<ffffffff8136628d>] SyS_exit_group+0x1d/0x20 kernel/exit.c:887 [<ffffffff866a3236>] entry_SYSCALL_64_fastpath+0x16/0x7a arch/x86/entry/entry_64.S:185 On commit fc77dbd34c5c99bce46d40a2491937c3bcbd10af (4.5-rc6).

2 7

[alsa-devel] [PATCH] ALSA: hda - hdmi defer to register acomp eld notifier
by libin.yang＠linux.intel.com 02 Mar '16

02 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> Defer to register acomp eld notifier until hdmi audio driver is fully ready. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index aa2d6a2..a3de50b 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2936,13 +2936,6 @@ static int patch_generic_hdmi(struct hda_codec *codec) is_broxton(codec)) codec->core.link_power_control = 1; - if (codec_has_acomp(codec)) { - codec->depop_delay = 0; - spec->i915_audio_ops.audio_ptr = codec; - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); - } - if (hdmi_parse_codec(codec) < 0) { if (spec->i915_bound) snd_hdac_i915_exit(&codec->bus->core); @@ -2964,6 +2957,13 @@ static int patch_generic_hdmi(struct hda_codec *codec) init_channel_allocations(); + if (codec_has_acomp(codec)) { + codec->depop_delay = 0; + spec->i915_audio_ops.audio_ptr = codec; + spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + } + WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec)); return 0; } -- 1.9.1

3 2

[alsa-devel] [PATCH v2] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 02 Mar '16

02 Mar '16

This is the initial imx-wm8958 device-tree-only machine driver working with fsl_sai driver. This sound card has three dai link, HIFI, VOICE and BT dai. HIFI dai link will support codec master and slave mode. VOICE an BT dai link have dummy cpu dai, and just support codec master mode. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- .../devicetree/bindings/sound/imx-audio-wm8958.txt | 49 +++ arch/arm/configs/imx_v6_v7_defconfig | 1 + sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 432 +++++++++++++++++++++ 5 files changed, 498 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..50bad12 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,49 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - cpu-dai : The phandle of an CPU DAI controller + + - audio-codec : The phandle of the WM8958 audio codec + + - audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being + the connection's sink, the second being the + connection's source. Valid names could be power + supplies, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + + - fsl,hifi-dai-master : If present, hifi dai works as master, and will + provide bit clock and frame clock. Otherwise, + hifi dai works as slave. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8960-audio"; + cpu-dai = <&sai1>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; + + fsl,hifi-dai-master; +}; diff --git a/arch/arm/configs/imx_v6_v7_defconfig b/arch/arm/configs/imx_v6_v7_defconfig index 978c5de..7ac5856 100644 --- a/arch/arm/configs/imx_v6_v7_defconfig +++ b/arch/arm/configs/imx_v6_v7_defconfig @@ -251,6 +251,7 @@ CONFIG_SND_SOC_FSL_ASRC=y CONFIG_SND_IMX_SOC=y CONFIG_SND_SOC_PHYCORE_AC97=y CONFIG_SND_SOC_EUKREA_TLV320=y +CONFIG_SND_SOC_IMX_WM8958=y CONFIG_SND_SOC_IMX_WM8962=y CONFIG_SND_SOC_IMX_SGTL5000=y CONFIG_SND_SOC_IMX_SPDIF=y diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..f6e5d96 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..24eb66f --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,432 @@ +/* + * Copyright (C) 2015-2016 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1 : WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link *dai_link; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_hifi_dai_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static u32 imx_wm8958_rates[] = { 8000, 16000, 32000, 48000 }; + +static struct snd_pcm_hw_constraint_list imx_wm8958_rate_constraints = { + .count = ARRAY_SIZE(imx_wm8958_rates), + .list = imx_wm8958_rates, +}; + +static int imx_wm8958_hifi_startup(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + + if (!data->is_hifi_dai_master) + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &imx_wm8958_rate_constraints); + + return ret; +} +static int imx_wm8958_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + /* + * Set GPIO1 pin function to reserve, so that DAC1 and ADC1 + * using shared LRCLK from DACLRCK1. + */ + snd_soc_update_bits(codec, WM8994_GPIO_1, 0x1f, 0x2); + + if (!data->is_hifi_dai_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, + 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (!data->is_hifi_dai_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, + "failed to set codec sysclk: %d\n", + ret); + return ret; + } + + return 0; + } + } else if (id == VOICE_DAI) { + /* + * Set GPIO6 pin function to reserve, so that DAC2 and ADC2 + * using shared LRCLK from DACLRCK2. + */ + snd_soc_update_bits(codec, WM8994_GPIO_6, 0x1f, 0x2); + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == HIFI_DAI && !data->is_hifi_dai_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, + * and disconnect AIF1CLK source to FLL before disable FLL, + * otherwise FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static int imx_wm8958_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + int ret, i; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (card->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* need to enable mclk to write/read wm8958 register */ + for (i = 0; i < WM8958_MCLK_MAX; i++) { + if (!IS_ERR(data->mclk[i])) { + ret = clk_prepare_enable(data->mclk[i]); + if (ret) + dev_warn(card->dev, + "Failed to enable MCLK%d: %d\n", + i + 1, ret); + } + } + } + break; + default: + break; + } + + return 0; +} + +static int imx_wm8958_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + int i; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + for (i = 0; i < WM8958_MCLK_MAX; i++) { + if (!IS_ERR(data->mclk[i])) + clk_disable_unprepare(data->mclk[i]); + } + break; + default: + break; + } + + card->dapm.bias_level = level; + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, + .startup = imx_wm8958_hifi_startup, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + .ops = &imx_voice_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + int ret, i; + + cpu_np = of_parse_phandle(np, "cpu-dai", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + return -EINVAL; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find cpu dai platform device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->dai_link = imx_wm8958_dai_link; + + /* + * AIF1 support codec master and slave mode + * AIF2 and AIF3 just support codec master mode + */ + if (of_property_read_bool(pdev->dev.of_node, "fsl,hifi-dai-master")) { + data->is_hifi_dai_master = true; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else { + data->is_hifi_dai_master = false; + data->dai_link[HIFI_DAI].dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (IS_ERR(data->mclk[i])) { + ret = PTR_ERR(data->mclk[i]); + dev_err(&pdev->dev, "failed to get mclk%d clock: %d\n", + i + 1, ret); + } else { + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + } + + data->dai_link[HIFI_DAI].cpu_dai_name = dev_name(&cpu_pdev->dev); + data->dai_link[HIFI_DAI].platform_of_node = cpu_np; + + data->card.set_bias_level = imx_wm8958_set_bias_level; + data->card.set_bias_level_post = imx_wm8958_set_bias_level_post; + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = DAI_LINK_NUM; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + data->card.owner = THIS_MODULE; + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: mediatek: Add machine driver for rt5650 rt5514 codec
by PC Liao 02 Mar '16

02 Mar '16

From: Koro Chen <koro.chen(a)mediatek.com> This is the DPCM based machine driver with rt5650 and rt5514. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- .../bindings/sound/mt8173-rt5650-rt5514.txt | 15 ++ sound/soc/mediatek/Kconfig | 28 ++- sound/soc/mediatek/Makefile | 3 +- sound/soc/mediatek/mt8173-rt5650-rt5514.c | 258 ++++++++++++++++++++ 4 files changed, 294 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt create mode 100644 sound/soc/mediatek/mt8173-rt5650-rt5514.c diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt new file mode 100644 index 0000000..e8b3c80 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5514.txt @@ -0,0 +1,15 @@ +MT8173 with RT5650 RT5514 CODECS + +Required properties: +- compatible : "mediatek,mt8173-rt5650-rt5514" +- mediatek,audio-codec: the phandles of rt5650 and rt5514 codecs +- mediatek,platform: the phandle of MT8173 ASoC platform + +Example: + + sound { + compatible = "mediatek,mt8173-rt5650-rt5514"; + mediatek,audio-codec = <&rt5650 &rt5514>; + mediatek,platform = <&afe>; + }; + diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 6c88087..f7e789e 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -18,14 +18,25 @@ config SND_SOC_MT8173_MAX98090 If unsure select "N". config SND_SOC_MT8173_RT5650 - tristate "ASoC Audio driver for MT8173 with RT5650 codec" - depends on SND_SOC_MEDIATEK && I2C - select SND_SOC_RT5645 - help - This adds ASoC driver for Mediatek MT8173 boards - with the RT5650 audio codec. - Select Y if you have such device. - If unsure select "N". + tristate "ASoC Audio driver for MT8173 with RT5650 codec" + depends on SND_SOC_MEDIATEK && I2C + select SND_SOC_RT5645 + help + This adds ASoC driver for Mediatek MT8173 boards + with the RT5650 audio codec. + Select Y if you have such device. + If unsure select "N". + +config SND_SOC_MT8173_RT5650_RT5514 + tristate "ASoC Audio driver for MT8173 with RT5650 RT5514 codecs" + depends on SND_SOC_MEDIATEK && I2C + select SND_SOC_RT5645 + select SND_SOC_RT5514 + help + This adds ASoC driver for Mediatek MT8173 boards + with the RT5650 and RT5514 codecs. + Select Y if you have such device. + If unsure select "N". config SND_SOC_MT8173_RT5650_RT5676 tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs" @@ -37,4 +48,3 @@ config SND_SOC_MT8173_RT5650_RT5676 with the RT5650 and RT5676 codecs. Select Y if you have such device. If unsure select "N". - diff --git a/sound/soc/mediatek/Makefile b/sound/soc/mediatek/Makefile index bcdcb06..d486860 100644 --- a/sound/soc/mediatek/Makefile +++ b/sound/soc/mediatek/Makefile @@ -2,5 +2,6 @@ obj-$(CONFIG_SND_SOC_MEDIATEK) += mtk-afe-pcm.o # Machine support obj-$(CONFIG_SND_SOC_MT8173_MAX98090) += mt8173-max98090.o -obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o obj-$(CONFIG_SND_SOC_MT8173_RT5650) += mt8173-rt5650.o +obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5514) += mt8173-rt5650-rt5514.o +obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o diff --git a/sound/soc/mediatek/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173-rt5650-rt5514.c new file mode 100644 index 0000000..58e0836 --- /dev/null +++ b/sound/soc/mediatek/mt8173-rt5650-rt5514.c @@ -0,0 +1,258 @@ +/* + * mt8173-rt5650-rt5514.c -- MT8173 machine driver with RT5650/5514 codecs + * + * Copyright (c) 2016 MediaTek Inc. + * Author: Koro Chen <koro.chen(a)mediatek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5645.h" + +#define MCLK_FOR_CODECS 12288000 + +static const struct snd_soc_dapm_widget mt8173_rt5650_rt5514_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route mt8173_rt5650_rt5514_routes[] = { + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + {"Sub DMIC1L", NULL, "Int Mic"}, + {"Sub DMIC1R", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Headset Mic", NULL, "micbias1"}, + {"Headset Mic", NULL, "micbias2"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, +}; + +static const struct snd_kcontrol_new mt8173_rt5650_rt5514_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static int mt8173_rt5650_rt5514_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int i, ret; + + for (i = 0; i < rtd->num_codecs; i++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; + + /* pll from mclk 12.288M */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, + params_rate(params) * 512); + if (ret) + return ret; + + /* sysclk from pll */ + ret = snd_soc_dai_set_sysclk(codec_dai, 1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret) + return ret; + } + return 0; +} + +static struct snd_soc_ops mt8173_rt5650_rt5514_ops = { + .hw_params = mt8173_rt5650_rt5514_hw_params, +}; + +static struct snd_soc_jack mt8173_rt5650_rt5514_jack; + +static int mt8173_rt5650_rt5514_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct snd_soc_codec *codec = runtime->codec_dais[0]->codec; + int ret; + + rt5645_sel_asrc_clk_src(codec, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* enable jack detection */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &mt8173_rt5650_rt5514_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "Can't new Headset Jack %d\n", ret); + return ret; + } + + return rt5645_set_jack_detect(codec, + &mt8173_rt5650_rt5514_jack, + &mt8173_rt5650_rt5514_jack, + &mt8173_rt5650_rt5514_jack); +} + +static struct snd_soc_dai_link_component mt8173_rt5650_rt5514_codecs[] = { + { + .dai_name = "rt5645-aif1", + }, + { + .dai_name = "rt5514-aif1", + }, +}; + +enum { + DAI_LINK_PLAYBACK, + DAI_LINK_CAPTURE, + DAI_LINK_CODEC_I2S, +}; + +/* Digital audio interface glue - connects codec <---> CPU */ +static struct snd_soc_dai_link mt8173_rt5650_rt5514_dais[] = { + /* Front End DAI links */ + [DAI_LINK_PLAYBACK] = { + .name = "rt5650_rt5514 Playback", + .stream_name = "rt5650_rt5514 Playback", + .cpu_dai_name = "DL1", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_playback = 1, + }, + [DAI_LINK_CAPTURE] = { + .name = "rt5650_rt5514 Capture", + .stream_name = "rt5650_rt5514 Capture", + .cpu_dai_name = "VUL", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dynamic = 1, + .dpcm_capture = 1, + }, + /* Back End DAI links */ + [DAI_LINK_CODEC_I2S] = { + .name = "Codec", + .cpu_dai_name = "I2S", + .no_pcm = 1, + .codecs = mt8173_rt5650_rt5514_codecs, + .num_codecs = 2, + .init = mt8173_rt5650_rt5514_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &mt8173_rt5650_rt5514_ops, + .ignore_pmdown_time = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +static struct snd_soc_codec_conf mt8173_rt5650_rt5514_codec_conf[] = { + { + .name_prefix = "Sub", + }, +}; + +static struct snd_soc_card mt8173_rt5650_rt5514_card = { + .name = "mtk-rt5650-rt5514", + .owner = THIS_MODULE, + .dai_link = mt8173_rt5650_rt5514_dais, + .num_links = ARRAY_SIZE(mt8173_rt5650_rt5514_dais), + .codec_conf = mt8173_rt5650_rt5514_codec_conf, + .num_configs = ARRAY_SIZE(mt8173_rt5650_rt5514_codec_conf), + .controls = mt8173_rt5650_rt5514_controls, + .num_controls = ARRAY_SIZE(mt8173_rt5650_rt5514_controls), + .dapm_widgets = mt8173_rt5650_rt5514_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8173_rt5650_rt5514_widgets), + .dapm_routes = mt8173_rt5650_rt5514_routes, + .num_dapm_routes = ARRAY_SIZE(mt8173_rt5650_rt5514_routes), +}; + +static int mt8173_rt5650_rt5514_dev_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &mt8173_rt5650_rt5514_card; + struct device_node *platform_node; + int i, ret; + + platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!platform_node) { + dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + if (mt8173_rt5650_rt5514_dais[i].platform_name) + continue; + mt8173_rt5650_rt5514_dais[i].platform_of_node = platform_node; + } + + mt8173_rt5650_rt5514_codecs[0].of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); + if (!mt8173_rt5650_rt5514_codecs[0].of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + mt8173_rt5650_rt5514_codecs[1].of_node = + of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1); + if (!mt8173_rt5650_rt5514_codecs[1].of_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + mt8173_rt5650_rt5514_codec_conf[0].of_node = + mt8173_rt5650_rt5514_codecs[1].of_node; + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + return ret; +} + +static const struct of_device_id mt8173_rt5650_rt5514_dt_match[] = { + { .compatible = "mediatek,mt8173-rt5650-rt5514", }, + { } +}; +MODULE_DEVICE_TABLE(of, mt8173_rt5650_rt5514_dt_match); + +static struct platform_driver mt8173_rt5650_rt5514_driver = { + .driver = { + .name = "mtk-rt5650-rt5514", + .of_match_table = mt8173_rt5650_rt5514_dt_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, + .probe = mt8173_rt5650_rt5514_dev_probe, +}; + +module_platform_driver(mt8173_rt5650_rt5514_driver); + +/* Module information */ +MODULE_DESCRIPTION("MT8173 RT5650 and RT5514 SoC machine driver"); +MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:mtk-rt5650-rt5514"); + -- 1.7.9.5

2 1

[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Update the IVI feedback path
by Vinod Koul 02 Mar '16

02 Mar '16

This patch series fixes the blob query direction and then uses that in IVI feedback path in nau88125_ssm4567 machine. Jeeja KP (3): ASoC: Intel: Skylake: correct direction in blob query ASoC: Intel: Skylake: Add IVI feedback path in nau88125_ssm4567 machine ASoC: Intel: Skylake: remove IVI codec loop in skl_nau88l25_ssm4567 machine sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 21 +++------------------ sound/soc/intel/skylake/skl-topology.c | 15 ++++++++------- 2 files changed, 11 insertions(+), 25 deletions(-) -- 1.9.1

2 6

Re: [alsa-devel] [RFD] Add hw_params support for hostless stream BE DAIs
by Liam Girdwood 02 Mar '16

02 Mar '16

On Tue, 2016-02-16 at 21:47 +0530, Jeeja KP wrote: > We have no mechanism to configure hw_params for DAIs which are hostless. > Like a loop or DAIs connected to internal sinks/sources. > Fwiw, a few years back Peter, Sebastien and I were considering a similar mechanism for the ABE on OMAP4. We never got around to implementing or upstreaming though.... One comment below. > Problem Statement: > During a BE-BE loop of DSP or a SRC->SINK path, we cannot program > hw_params. There are no hw_params from user as these are hostless. > This scenario can happens in below 2 cases: > > Case 1: BE-BE loop aka hostless stream through SoC DSP > Hostless stream can be defined by BE to BE connection and in this case > the path is enabled or disabled by the state of the DAPM graph. This > usually means there is a mixer control that can be used to connect or > disconnect the path between both DAIs. > Example: > |--------------/----------------| > | | > | | > codec out modem_in > | | > | | > ssp0_TX ssp1_RX > | | > | | > AIFI_Playback dummy_capture > > In the above case (SSP0,codec playback,SSp1 dai needs to be configured > and turned on with the selected hw_params). > > Case 2: SoC Internal Sink or Source > Example: > WOV SINK (DAPM Sink) > | > | > | > Switch > | > | > w1 > | > dmic rx (BE dai widget) > | > Capture(codec dai widget) > | > DMIC AIF > | > DMic > | > Soc DMic (MIC Widget) (SRC) > > In the above case BE (dmic RX) dai hw_params needs to applied. > > Approach: > 1. Allow params to be defined for other BE dai link also. this > currently applicable only for codec_codec link. > > 2. when params is defined for BE dai link, create params kcontrols > and associate it with the dai widgets. this will allow params to be > selected via Kcontrol for each dai defined in the dai link. > The only problem we have with using a kcontrol to define params is that we already have an existing kernel ioctl and public C API to define SW/HW params. This would add another mechanism, unless each new kcontrol allows modification of a single param ? Liam > 3. Before powering on the dai widget and if the dai is not powered > with the stream event, then dai_ops(hw_params) for the dai widget will > be called.This will apply the selected hw_params. > > 4. with the above, fix_up can be removed and can use .params to > select the parameter. > > Core Changes: > 1. params definition needs to changed, currently this is defined to used > only for codec codec link. define new flag to identifying codec > codec link. > > 2. if params is defined for any BE dai link, then create kcontrol for > each dai defined in the dai_link and associated the kcontrol to dai > widget. > > 3. In the powering sequence check if the dai widget and if the event is > not a stream event, SND_SOC_DAPM_STREAM_NOP , then call dai ops with > the selected params. This will apply the hw_params for dai. > > 1. params definition: > - Add new flag in struct snd_soc_dai_link. > - Based on this flag create, the codec codec link instead of > params > struct snd_soc_dai_link { > const struct snd_soc_pcm_stream *params; > unsigned int num_params; > > + /* flag to create codec codec based on the flag*/ > + unsigned int codec_link:1; > + > unsigned int dai_fmt; /* format to set on init */ > > enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ > > - creation of dai link based on the codec_link flag > static int soc_probe_link_dais(struct snd_soc_card *card, > int num, int order) > } > } else { > > - if (!dai_link->params) { > + if (!dai_link->codec_link) { > /* create the pcm */ > ret = soc_new_pcm(rtd, num); > if (ret < 0) { > > 2. when params is defined for BE dai link, create params kcontrols and > associate it with the dai widgets. this will allow params to be selected > via Kcontrol for each dai defined in the dai link. > a. Move the param control creation from snd_soc_dapm_new_pcm and add new > API to created kcontrols with the params. > +snd_soc_dapm_create_param_control(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params) > { > /* create kcontrol based on the params */ > } > > - modify the snd_soc_dapm_new_pcm to create the param control > int snd_soc_dapm_new_pcm(struct snd_soc_card *card, > const struct snd_soc_pcm_stream *params, > unsigned int num_params, > struct snd_soc_dapm_widget *source, > struct snd_soc_dapm_widget *sink) > { > > template.kcontrol_news = snd_soc_dapm_create_param_control(card, > params, num_params); > if (!template.kcontrol_news) { > dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", > link_name); > ret = -ENOMEM; > - goto outfree_private_value; > + goto outfree_link_name; > } > > > b. add new API snd_soc_dapm_new_dai_params() to create kcontrol for dai and associated it to dai. > +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params, > + struct snd_soc_dapm_widget *dai_w) > +{ > + struct snd_kcontrol_new *kcontrol_news; > + > + kcontrol_news = snd_soc_dapm_create_param_control(card, params, > + num_params); > + if (!kcontrol_news) { > + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", > + dai_w->name); > + return -EINVAL; > + } > + > + dai_w->num_kcontrols = 1; > + dai_w->kcontrol_news = kcontrol_news; > + > + dai_w->params = params; > + dai_w->num_params = num_params; > + > + return 0; > + > +} > + > +++ b/include/sound/soc-dapm.h > +int snd_soc_dapm_new_dai_params(struct snd_soc_card *card, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params, > + struct snd_soc_dapm_widget *dai_w); > > > c. if the be dai link has .params defined, call new API snd_soc_dapm_new_dai_params > > +static int soc_link_be_params_widget(struct snd_soc_card *card, > + struct snd_soc_dai *dai, > + const struct snd_soc_pcm_stream *params, > + unsigned int num_params) > +{ > + int ret; > + struct snd_soc_dapm_widget *w; > + > + if (!strcmp(dai->name, "snd-soc-dummy")) > + return 0; > + > + if (dai->playback_widget) { > + w = dai->playback_widget; > + ret = snd_soc_dapm_new_dai_params(card, params, > + num_params, w); > + > + if (ret != 0) { > + dev_err(card->dev, > + "ASoC: Can't create params widget=%s %d\n", > + w->name, ret); > + return ret; > + } > + } > + > + if (dai->capture_widget) { > + w = dai->capture_widget; > + ret = snd_soc_dapm_new_dai_params(card, params, > + num_params, w); > + > + if (ret != 0) { > + dev_err(card->dev, > + "ASoC: Can't create params widget=%s %d\n", > + w->name, ret); > + return ret; > + } > + } > + > + return 0; > +} > + > +static int soc_link_be_params_dai_widgets(struct snd_soc_card *card, > + struct snd_soc_dai_link *dai_link, > + struct snd_soc_pcm_runtime *rtd) > +{ > + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; > + int ret, i; > + > + for (i = 0; i < rtd->num_codecs; i++) { > + struct snd_soc_dai *codec = rtd->codec_dais[i]; > + > + ret = soc_link_be_params_widget(card, codec, > + dai_link->params, > + dai_link->num_params); > + > + if (!ret) > + return ret; > + } > + > + return soc_link_be_params_widget(card, cpu_dai, dai_link->params, > + dai_link->num_params); > +} > + > > @@ -1651,6 +1714,13 @@ static int soc_probe_link_dais(struct snd_soc_card *card, > dai_link->stream_name, ret); > return ret; > } > + if (dai_link->params && dai_link->no_pcm) { > + /* create dai link controls */ > + ret = soc_link_be_params_dai_widgets(card, > + dai_link, rtd); > + if (ret) > + return ret; > + } > } else { > INIT_DELAYED_WORK(&rtd->delayed_work, > codec2codec_close_delayed_work); > > > 3. Applying the hw_params for dai widget > > --- a/sound/soc/soc-dapm.c > +++ b/sound/soc/soc-dapm.c > @@ -1398,9 +1398,64 @@ static void dapm_seq_check_event(struct snd_soc_card * > card, > } > } > > +static int snd_soc_dai_params(struct snd_soc_dapm_widget *w, int event) > +{ > + struct snd_soc_dai *dai; > + const struct snd_soc_pcm_stream *config = w->params + w->params_select; > + struct snd_pcm_substream substream; > + struct snd_pcm_hw_params *params = NULL; > + u64 fmt; > + int ret = 0; > + > + if (WARN_ON(!config)) > + return -EINVAL; > + > + dai = w->priv; > + > + if (config->formats) { > + fmt = ffs(config->formats) - 1; > + } else { > + dev_warn(w->dapm->dev, "ASoC: Invalid format %llx specified\n", > + config->formats); > + fmt = 0; > + } > + > + /* Currently very limited parameter selection */ > + params = kzalloc(sizeof(*params), GFP_KERNEL); > + if (!params) { > + ret = -ENOMEM; > + goto out; > + } > + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), fmt); > + > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->min = > + config->rate_min; > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE)->max = > + config->rate_max; > + > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->min > + = config->channels_min; > + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max > + = config->channels_max; > + > + memset(&substream, 0, sizeof(substream)); > + if (w->id == snd_soc_dapm_dai_in) > + substream.stream = SNDRV_PCM_STREAM_CAPTURE; > + else > + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; > + > + ret = soc_dai_hw_params(&substream, params, dai); > + if (ret < 0) > + goto out; > +out: > + kfree(params); > + return ret; > +} > + > /* Apply the coalesced changes from a DAPM sequence */ > static void dapm_seq_run_coalesced(struct snd_soc_card *card, > - struct list_head *pending) > + struct list_head *pending, > + int event) > { > struct snd_soc_dapm_context *dapm; > struct snd_soc_dapm_widget *w; > @@ -1413,6 +1468,14 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, > dapm = w->dapm; > > list_for_each_entry(w, pending, power_list) { > + if ((event == SND_SOC_DAPM_STREAM_NOP) && > + ((w->id == snd_soc_dapm_dai_out) || > + (w->id == snd_soc_dapm_dai_in))) { > + snd_soc_dai_params(w, SND_SOC_DAPM_PRE_PMU); > + } > + } > + > + list_for_each_entry(w, pending, power_list) { > WARN_ON(reg != w->reg || dapm != w->dapm); > w->power = w->new_power; > > @@ -1447,6 +1510,13 @@ static void dapm_seq_run_coalesced(struct snd_soc_card *card, > dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMU); > dapm_seq_check_event(card, w, SND_SOC_DAPM_POST_PMD); > } > + list_for_each_entry(w, pending, power_list) { > + if ((event == SND_SOC_DAPM_STREAM_NOP) && > + ((w->id == snd_soc_dapm_dai_out) || > + (w->id == snd_soc_dapm_dai_in))) { > + snd_soc_dai_params(w, SND_SOC_DAPM_POST_PMD); > + } > + } > } > > /* Apply a DAPM power sequence. > @@ -1482,7 +1552,7 @@ static void dapm_seq_run(struct snd_soc_card *card, > if (sort[w->id] != cur_sort || w->reg != cur_reg || > w->dapm != cur_dapm || w->subseq != cur_subseq) { > if (!list_empty(&pending)) > - dapm_seq_run_coalesced(card, &pending); > + dapm_seq_run_coalesced(card, &pending, event); > > if (cur_dapm && cur_dapm->seq_notifier) { > for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) > @@ -1545,7 +1615,7 @@ static void dapm_seq_run(struct snd_soc_card *card, > } > > if (!list_empty(&pending)) > - dapm_seq_run_coalesced(card, &pending); > + dapm_seq_run_coalesced(card, &pending, event); > > if (cur_dapm && cur_dapm->seq_notifier) { > for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++) > -- > 1.7.9.5 > > Comments welcome, based on discussion we will send patches. > > --Jeeja

3 2

Re: [alsa-devel] [PATCH v2] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Nicolin Chen 02 Mar '16

02 Mar '16

It looks like you replied me off the maillist. So I just add the maillist and related TO/CCs back. On Tue, Mar 01, 2016 at 05:43:22AM +0000, Zidan Wang wrote: > On Fri, Feb 26, 2016 at 11:42:42AM +0800, Zidan Wang wrote: > > This is the initial imx-wm8958 device-tree-only machine driver working > > with fsl_sai driver. This sound card has three dai link, HIFI, VOICE > > and BT dai. HIFI dai link will support codec master and slave mode. > > VOICE an BT dai link have dummy cpu dai, and just support codec master > > mode. > > Why VOICE and BT are using dummy cpu dai? Where can I find the schematics > for wm8958 from NXP website? > [Wang Zidan-B50113] I have attached the schematics for wm8958. In our > release, we can't support VOICE and BT, and just using AIF1. But in > order to upstream, I add the voice and bt dai. The way to support the other two AIFs are not very graceful -- looks like you will have to hard code the CPU DAIs in the driver. I personally suggest you to use fsl-asoc-card. Since the Codec IC provides three DAIs. And in the schematics, you do have three DAI links. It sounds plausible to me that we might treat these three DAI links as three different cards. Then use different compatible names: "imx-audio-wm8958-aif1" and so on. And each one can also has ASRC back-end support. If you really have big trouble to integrate it into fsl-asoc-card (not 100% sure if the 3 cards methodology is totally feasible), I will be okay with a dedicated driver for wm8958. But you will need to refine the DT binding to support three CPU DAIs in my opinion. > > +static u32 imx_wm8958_rates[] = { 8000, 16000, 32000, 48000 }; > > + > > +static struct snd_pcm_hw_constraint_list imx_wm8958_rate_constraints = { > > + .count = ARRAY_SIZE(imx_wm8958_rates), > > + .list = imx_wm8958_rates, > > +}; > > Where are these rate constraints coming from? > [Wang Zidan-B50113] these rates are restricted by the SAI MCLK. If the SAI MCLK runs at a different rate, it would be invalid. So there should be at least two constraints depending on the MCLK rate. And I think it might be better to have this in the SAI driver as a refinement since the current driver only errors out for those unsupported rates. > > + if (id == HIFI_DAI) { > > + /* > > + * Set GPIO1 pin function to reserve, so that DAC1 and ADC1 > > + * using shared LRCLK from DACLRCK1. > > + */ > > + snd_soc_update_bits(codec, WM8994_GPIO_1, 0x1f, 0x2); > > I don't see any reverse operation against this GPIO configuration here. > So is it really an operation that has to be done every hw_params()? > > > + } else if (id == VOICE_DAI) { > > + /* > > + * Set GPIO6 pin function to reserve, so that DAC2 and ADC2 > > + * using shared LRCLK from DACLRCK2. > > + */ > > + snd_soc_update_bits(codec, WM8994_GPIO_6, 0x1f, 0x2); > > Same here. Actually I just took a look at the DT binding doc of WM8994. It supports GPIO configurations in the codec DT nodes. So you may not need to do these over here at all. > > +static int imx_wm8958_set_bias_level_post(struct snd_soc_card *card, > > + struct snd_soc_dapm_context *dapm, > > + enum snd_soc_bias_level level) { > > + switch (level) { > > + case SND_SOC_BIAS_OFF: > > + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) > > + for (i = 0; i < WM8958_MCLK_MAX; i++) { > > + if (!IS_ERR(data->mclk[i])) > > + clk_disable_unprepare(data->mclk[i]); > > The set_bias() funcs are merely en/disabling mclk. > Would be possible to do it in the codec driver? > [Wang Zidan-B50113] It can be moved to codec driver, > but the wm8994 driver can support wm8958 wm8994 wm1811 codec, > it's hard for me to do like that, and I also can't test them. Sending a patch to the maillist is a way to let people be aware of the change and then test it. The driver maintainer will review it and give you feedback. If it really causes compatible issues, you can then limit the code for wm8958 only. > > +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { > > + [BT_DAI] = { > > + .name = "Bluetooth", > > + .stream_name = "Bluetooth", > > + .cpu_dai_name = "snd-soc-dummy-dai", > > + .codec_name = "wm8994-codec", > > + .codec_dai_name = "wm8994-aif3", > > + .platform_name = "snd-soc-dummy", > > + .ignore_pmdown_time = 1, > > + }, > > A little curious...how is the BT gonna work for you? > [Wang Zidan-B50113] I have not test the BT dai. The BT is directly > connect to AIF3, when AIF3 receive date from BT, it will route the > date to AIF1 or AIF2. And the date from AIF1 and AIF2 can also route > to AIF3 to BT. So we just need to set the amixer setting to open the > audio route. Sounds interesting. So the Codec totally becomes a router. If this design can not be supported by using three cards via the fsl-asoc-card, you may continue this driver as I can see the situation here. But I encourage you to try that first so as to avoid duplicated code/driver.

2 2

[alsa-devel] [PATCH] Add a quirk for Plantronics DA45
by Dennis Kadioglu 01 Mar '16

01 Mar '16

Plantronics DA45 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x4" and "cannot get freq at ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and avoids those error messages. Signed-off-by: Dennis Kadioglu <denk(a)post.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2585c17..5b03296 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1123,6 +1123,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ ---

2 1

[alsa-devel] [PATCH] Add a quirk for Plantronics DA45
by denk 01 Mar '16

01 Mar '16

Plantronics DA45 does not support reading the sample rate which leads to many lines of "cannot get freq at ep 0x4" and "cannot get freq at ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and avoids those error messages. --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 2585c17..5b03296 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1123,6 +1123,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x045E, 0x076F): /* MS Lifecam HD-6000 */ case USB_ID(0x045E, 0x0772): /* MS Lifecam Studio */ case USB_ID(0x045E, 0x0779): /* MS Lifecam HD-3000 */ + case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ --- Signed-off-by: Dennis Kadioglu <denk(a)post.com>

4 5

[alsa-devel] [PATCH 1/1] alsabat: truncate sample frames for faster FFT analysis
by han.lu＠intel.com 01 Mar '16

01 Mar '16

From: "Lu, Han" <han.lu(a)intel.com> Truncate the sample frames to powers of 2, since the FFTW algorithm runs especially fast in this case, and other sizes may be computed by means of a slow, general-purpose algorithm. In my test environment applying the patch, a sound clip of 33072 frames is cut off to 32768 frames before analysis, and the time cost is reduced from 6.128s to 0.224s. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/analyze.c b/bat/analyze.c index 60e2d1c..5cfdac3 100644 --- a/bat/analyze.c +++ b/bat/analyze.c @@ -256,6 +256,20 @@ static int reorder_data(struct bat *bat) return 0; } +/* truncate sample frames for faster FFT analysis process */ +static int truncate_frames(struct bat *bat) +{ + int shift = SHIFT_MAX; + + for (; shift > SHIFT_MIN; shift--) + if (bat->frames & (1 << shift)) { + bat->frames = 1 << shift; + return 0; + } + + return -EINVAL; +} + int analyze_capture(struct bat *bat) { int err = 0; @@ -263,6 +277,13 @@ int analyze_capture(struct bat *bat) int c; struct analyze a; + err = truncate_frames(bat); + if (err < 0) { + fprintf(bat->err, _("Invalid frame number for analysis: %d\n"), + bat->frames); + return err; + } + fprintf(bat->log, _("\nBAT analysis: signal has %d frames at %d Hz,"), bat->frames, bat->rate); fprintf(bat->log, _(" %d channels, %d bytes per sample.\n"), diff --git a/bat/common.h b/bat/common.h index b8f6670..ff03fc1 100644 --- a/bat/common.h +++ b/bat/common.h @@ -68,6 +68,12 @@ #define FOUND_DC (1<<1) #define FOUND_WRONG_PEAK (1<<0) +/* Truncate sample frames to (1 << N), for faster FFT analysis process. The + * valid range of N is (SHIFT_MIN, SHIFT_MAX). When N increases, the analysis + * will be more time-consuming, and the result will be more accurate. */ +#define SHIFT_MAX (sizeof(int) * 8 - 2) +#define SHIFT_MIN 8 + struct wav_header { unsigned int magic; /* 'RIFF' */ unsigned int length; /* file len */ -- 2.5.0

2 1

[alsa-devel] [PATCH] ALSA: hda - hdmi_find_pcm_slot return value bug fix
by libin.yang＠linux.intel.com 01 Mar '16

01 Mar '16

From: Libin Yang <libin.yang(a)linux.intel.com> hdmi_find_pcm_slot return -EBUSY when not no pcm slot found, not -ENODEV. So the caller should compare with -EBUSY. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d77509a..aa2d6a2 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1758,7 +1758,7 @@ static void hdmi_attach_hda_pcm(struct hdmi_spec *spec, if (per_pin->pcm) return; idx = hdmi_find_pcm_slot(spec, per_pin); - if (idx == -ENODEV) + if (idx == -EBUSY) return; per_pin->pcm_idx = idx; per_pin->pcm = get_hdmi_pcm(spec, idx); -- 1.9.1

2 1

[alsa-devel] [PATCH] teac ud-501/ud-503 usb delay
by Guillaume Fougnies 01 Mar '16

01 Mar '16

Hello, This is a small patch against sound/usb/quirks.c to correct "Teac Corp." products delay. Tested on Teac UD-501 and UD-503 products, could not harm other Teac products. It corrects the "clock source 41 is not valid, cannot use" error for both devices. And more critically on the new UD-503, a complete freeze of the teac usb interface when switching between different rate/format. (Freeze only solved by a power switch of the device...) The patch matches the problem of the "Playback Design" products. I just refactored a bit the code to include Teac products properly. Regards, Guillaume --- sound/usb/quirks.c | 58 +++++++++++++++++++++++++++++++----------------------- 1 file changed, 33 insertions(+), 25 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 23ea6d8..f3faaef 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1202,40 +1202,48 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) void snd_usb_set_interface_quirk(struct usb_device *dev) { /* - * "Playback Design" products need a 50ms delay after setting the + * Products needing a 50ms delay after setting the * USB interface. */ - if (le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) - mdelay(50); + switch (le16_to_cpu(dev->descriptor.idVendor)) { + case 0x23ba: /* Playback Design */ + case 0x0644: /* TEAC Corp. */ + mdelay(50); + } } void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size) { - /* - * "Playback Design" products need a 20ms delay after each - * class compliant request - */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x23ba) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); - - /* Marantz/Denon devices with USB DAC functionality need a delay - * after each class compliant request - */ - if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), - le16_to_cpu(dev->descriptor.idProduct))) - && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(20); + if ((requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) { + switch (le16_to_cpu(dev->descriptor.idVendor)) { + /* + * "Playback Design"/"TEAC Corp." products need a 20ms delay after each + * class compliant request + */ + case 0x23ba: /* "Playback Design" */ + case 0x0644: /* TEAC Corp. */ + mdelay(20); + return; + + /* Zoom R16/24 needs a tiny delay here, otherwise requests like + * get/set frequency return as failed despite actually succeeding. + */ + case 0x1686: + /* restrict to 0x00dd */ + if (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) + mdelay(1); + return; + } - /* Zoom R16/24 needs a tiny delay here, otherwise requests like - * get/set frequency return as failed despite actually succeeding. - */ - if ((le16_to_cpu(dev->descriptor.idVendor) == 0x1686) && - (le16_to_cpu(dev->descriptor.idProduct) == 0x00dd) && - (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) - mdelay(1); + /* Marantz/Denon devices with USB DAC functionality need a delay + * after each class compliant request + */ + if (is_marantz_denon_dac(USB_ID(le16_to_cpu(dev->descriptor.idVendor), + le16_to_cpu(dev->descriptor.idProduct)))) + mdelay(20); + } } /* -- 2.6.4

4 5

[alsa-devel] Applied "ASoC: intel: remove unused variable" to the asoc tree
by Mark Brown 01 Mar '16

01 Mar '16

The patch ASoC: intel: remove unused variable has been applied to the asoc tree at git://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 dbc77eed6553362e08c1874ff30df640c96eca92 Mon Sep 17 00:00:00 2001 From: Sudip Mukherjee <sudipm.mukherjee(a)gmail.com> Date: Mon, 29 Feb 2016 22:53:46 +0530 Subject: [PATCH] ASoC: intel: remove unused variable The variable cmd_id is only assigned some value and is never used. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/atom/sst/sst_ipc.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index 3dc7358828b3..8afa6fe7b0b0 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -318,7 +318,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, union ipc_header_high msg_high; u32 msg_low; struct ipc_dsp_hdr *dsp_hdr; - unsigned int cmd_id; msg_high = msg->mrfld_header.p.header_high; msg_low = msg->mrfld_header.p.header_low_payload; @@ -357,7 +356,6 @@ void sst_process_reply_mrfld(struct intel_sst_drv *sst_drv_ctx, return; /* Copy command id so that we can use to put sst to reset */ dsp_hdr = (struct ipc_dsp_hdr *)data; - cmd_id = dsp_hdr->cmd_id; dev_dbg(sst_drv_ctx->dev, "cmd_id %d\n", dsp_hdr->cmd_id); if (sst_wake_up_block(sst_drv_ctx, msg_high.part.result, msg_high.part.drv_id, -- 2.7.0

1 0

[alsa-devel] [PATCH] ASoC: dapm: Fix ctl value accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

snd_soc_dapm_dai_link_get() and _put() access the associated ctl values as value.integer.value[]. However, this is an enum ctl, and it has to be accessed via value.enumerated.item[]. The former is long while the latter is unsigned int, so they don't align. Fixes: c66150824b8a ('ASoC: dapm: add code to configure dai link parameters') Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0d3707987900..581175a51ecf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3573,7 +3573,7 @@ static int snd_soc_dapm_dai_link_get(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = w->params_select; + ucontrol->value.enumerated.item[0] = w->params_select; return 0; } @@ -3587,13 +3587,13 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, if (w->power) return -EBUSY; - if (ucontrol->value.integer.value[0] == w->params_select) + if (ucontrol->value.enumerated.item[0] == w->params_select) return 0; - if (ucontrol->value.integer.value[0] >= w->num_params) + if (ucontrol->value.enumerated.item[0] >= w->num_params) return -EINVAL; - w->params_select = ucontrol->value.integer.value[0]; + w->params_select = ucontrol->value.enumerated.item[0]; return 0; } -- 2.7.2

2 1

[alsa-devel] [PATCH 0/2] ASoC: omap: Fix enum ctl accesses
by Takashi Iwai 01 Mar '16

01 Mar '16

Hi, here is a patchset to correct the wrong type accesses in omap drivers for enum ctls. I didn't put Cc to stable for them, supposing these are only for 32bit systems. If not, feel free to put Cc to stable. thanks, Takashi === Takashi Iwai (2): ASoC: omap: n810: Fix enum ctl accesses in a wrong type ASoC: omap: rx51: Fix enum ctl accesses in a wrong type sound/soc/omap/n810.c | 18 +++++++++--------- sound/soc/omap/rx51.c | 18 +++++++++--------- 2 files changed, 18 insertions(+), 18 deletions(-) -- 2.7.2

3 6

[alsa-devel] [PATCH] ASoC: intel: mfld: Fix enum ctl accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

"Playback Switch" and "Lineout Mux" ctls in medfld machine driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- I didn't put Cc to stable, as I assume medfiled is 32bit only. If it also supports 64bit, put Cc to stable. sound/soc/intel/boards/mfld_machine.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c index 49c09a0add79..34f46c72a0e2 100644 --- a/sound/soc/intel/boards/mfld_machine.c +++ b/sound/soc/intel/boards/mfld_machine.c @@ -94,7 +94,7 @@ static const struct soc_enum lo_enum = static int headset_get_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = hs_switch; + ucontrol->value.enumerated.item[0] = hs_switch; return 0; } @@ -104,12 +104,12 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_context *dapm = &card->dapm; - if (ucontrol->value.integer.value[0] == hs_switch) + if (ucontrol->value.enumerated.item[0] == hs_switch) return 0; snd_soc_dapm_mutex_lock(dapm); - if (ucontrol->value.integer.value[0]) { + if (ucontrol->value.enumerated.item[0]) { pr_debug("hs_set HS path\n"); snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); @@ -123,7 +123,7 @@ static int headset_set_switch(struct snd_kcontrol *kcontrol, snd_soc_dapm_mutex_unlock(dapm); - hs_switch = ucontrol->value.integer.value[0]; + hs_switch = ucontrol->value.enumerated.item[0]; return 0; } @@ -148,7 +148,7 @@ static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm) static int lo_get_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = lo_dac; + ucontrol->value.enumerated.item[0] = lo_dac; return 0; } @@ -158,7 +158,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); struct snd_soc_dapm_context *dapm = &card->dapm; - if (ucontrol->value.integer.value[0] == lo_dac) + if (ucontrol->value.enumerated.item[0] == lo_dac) return 0; snd_soc_dapm_mutex_lock(dapm); @@ -168,7 +168,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, */ lo_enable_out_pins(dapm); - switch (ucontrol->value.integer.value[0]) { + switch (ucontrol->value.enumerated.item[0]) { case 0: pr_debug("set vibra path\n"); snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT"); @@ -202,7 +202,7 @@ static int lo_set_switch(struct snd_kcontrol *kcontrol, snd_soc_dapm_mutex_unlock(dapm); - lo_dac = ucontrol->value.integer.value[0]; + lo_dac = ucontrol->value.enumerated.item[0]; return 0; } -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: cs42l51: Fix enum ctl accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

"PCM channel mixer" ctl in cs42l51 codec driver is enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- like other patchsets, I didn't put Cc to stable. If this driver is used for 64bit archs, feel free to put Cc to stable. sound/soc/codecs/cs42l51.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index b3951524339f..35488f14e237 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -60,15 +60,15 @@ static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol, switch (value) { default: case 0: - ucontrol->value.integer.value[0] = 0; + ucontrol->value.enumerated.item[0] = 0; break; /* same value : (L+R)/2 and (R+L)/2 */ case 1: case 2: - ucontrol->value.integer.value[0] = 1; + ucontrol->value.enumerated.item[0] = 1; break; case 3: - ucontrol->value.integer.value[0] = 2; + ucontrol->value.enumerated.item[0] = 2; break; } @@ -85,7 +85,7 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); unsigned char val; - switch (ucontrol->value.integer.value[0]) { + switch (ucontrol->value.enumerated.item[0]) { default: case 0: val = CHAN_MIX_NORMAL; -- 2.7.2

2 1

[alsa-devel] [PATCH] ASoC: da732x: Fix enum ctl accesses in a wrong type
by Takashi Iwai 01 Mar '16

01 Mar '16

"DAC1 High Pass Filter Mode" & co in da732x codec driver are enum, while the current driver accesses wrongly via value.integer.value[]. They have to be via value.enumerated.item[] instead. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- like other patchsets, I didn't put Cc to stable. If this driver is used for 64bit archs, feel free to put Cc to stable. sound/soc/codecs/da732x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 1d5a89c5164b..461506a4ca6a 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -334,7 +334,7 @@ static int da732x_hpf_set(struct snd_kcontrol *kcontrol, struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; unsigned int reg = enum_ctrl->reg; - unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.enumerated.item[0]; unsigned int bits; switch (sel) { @@ -368,13 +368,13 @@ static int da732x_hpf_get(struct snd_kcontrol *kcontrol, switch (val) { case DA732X_HPF_VOICE_EN: - ucontrol->value.integer.value[0] = DA732X_HPF_VOICE; + ucontrol->value.enumerated.item[0] = DA732X_HPF_VOICE; break; case DA732X_HPF_MUSIC_EN: - ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC; + ucontrol->value.enumerated.item[0] = DA732X_HPF_MUSIC; break; default: - ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED; + ucontrol->value.enumerated.item[0] = DA732X_HPF_DISABLED; break; } -- 2.7.2

2 1

[alsa-devel] [PATCH 0/5] ASoC: Fix enum ctl accesses in various codec drivers
by Takashi Iwai 01 Mar '16

01 Mar '16

Hi, here are the rest of buggy drivers to be fixed for the wrong ctl access types to enum ctls. Again, put Cc to stable appropriately if needed. thanks, Takashi === Takashi Iwai (5): ASoC: ab8500: Fix enum ctl accesses in a wrong type ASoC: max98088: Fix enum ctl accesses in a wrong type ASoC: max98095: Fix enum ctl accesses in a wrong type ASoC: tlv320dac33: Fix enum ctl accesses in a wrong type ASoC: wl1273: Fix enum ctl accesses in a wrong type sound/soc/codecs/ab8500-codec.c | 8 ++++---- sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/codecs/tlv320dac33.c | 9 ++++----- sound/soc/codecs/wl1273.c | 13 ++++++------- 5 files changed, 17 insertions(+), 19 deletions(-) -- 2.7.2

2 10

[alsa-devel] [PATCH v2 0/6] ALSA core X32 ABI fixes
by Takashi Iwai 29 Feb '16

29 Feb '16

Hi, this is the revised patchset for fixing the incompatible ioctls in ALSA core with X32 ABI. Too bad that we have to deal in this way, but it's life... Takashi === Takashi Iwai (6): ALSA: ctl: Fix ioctls for X32 ABI ALSA: pcm: Fix ioctls for X32 ABI ALSA: rawmidi: Use comapt_put_timespec() ALSA: rawmidi: Fix ioctls X32 ABI ALSA: timer: Fix broken compat timer user status ioctl ALSA: timer: Fix ioctls for X32 ABI sound/core/control_compat.c | 90 ++++++++++++++++++---- sound/core/pcm_compat.c | 177 +++++++++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 56 +++++++++++++- sound/core/timer_compat.c | 18 ++++- 4 files changed, 320 insertions(+), 21 deletions(-) -- 2.7.2

2 8

[alsa-devel] [PATCH] ALSA: seq: Remove unimplemented ioctls
by Takashi Iwai 29 Feb '16

29 Feb '16

SNDRV_SEQ_IOCTL_{GET|SET}_QUEUE_OWNER and *_{GET|SET}_QUEUE_SYNC ioctls have been never implemented. Get rid of the definitions from uapi header file. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/uapi/sound/asequencer.h | 6 ------ 1 file changed, 6 deletions(-) diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 5a5fa4956ebd..af96f2044f91 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -594,14 +594,8 @@ struct snd_seq_query_subs { #define SNDRV_SEQ_IOCTL_GET_QUEUE_STATUS _IOWR('S', 0x40, struct snd_seq_queue_status) #define SNDRV_SEQ_IOCTL_GET_QUEUE_TEMPO _IOWR('S', 0x41, struct snd_seq_queue_tempo) #define SNDRV_SEQ_IOCTL_SET_QUEUE_TEMPO _IOW ('S', 0x42, struct snd_seq_queue_tempo) -#define SNDRV_SEQ_IOCTL_GET_QUEUE_OWNER _IOWR('S', 0x43, struct snd_seq_queue_owner) -#define SNDRV_SEQ_IOCTL_SET_QUEUE_OWNER _IOW ('S', 0x44, struct snd_seq_queue_owner) #define SNDRV_SEQ_IOCTL_GET_QUEUE_TIMER _IOWR('S', 0x45, struct snd_seq_queue_timer) #define SNDRV_SEQ_IOCTL_SET_QUEUE_TIMER _IOW ('S', 0x46, struct snd_seq_queue_timer) -/* XXX -#define SNDRV_SEQ_IOCTL_GET_QUEUE_SYNC _IOWR('S', 0x53, struct snd_seq_queue_sync) -#define SNDRV_SEQ_IOCTL_SET_QUEUE_SYNC _IOW ('S', 0x54, struct snd_seq_queue_sync) -*/ #define SNDRV_SEQ_IOCTL_GET_QUEUE_CLIENT _IOWR('S', 0x49, struct snd_seq_queue_client) #define SNDRV_SEQ_IOCTL_SET_QUEUE_CLIENT _IOW ('S', 0x4a, struct snd_seq_queue_client) #define SNDRV_SEQ_IOCTL_GET_CLIENT_POOL _IOWR('S', 0x4b, struct snd_seq_client_pool) -- 2.7.2

1 0

[alsa-devel] [PATCH] ASoC: qcom: add dependency on HAS_DMA
by Sudip Mukherjee 29 Feb '16

29 Feb '16

m32r allmodconfig build is failing with the error: ERROR: "bad_dma_ops" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! ERROR: "dma_common_mmap" [sound/soc/qcom/snd-soc-lpass-platform.ko] undefined! lpass-platform uses DMA so add HAS_DMA in the Kconfig. Also add the same for SND_SOC_LPASS_IPQ806X and SND_SOC_LPASS_APQ8016 as they will also select SND_SOC_LPASS_PLATFORM. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- m32r allmodconfig build for next-20160229 is at: https://travis-ci.org/sudipm-mukherjee/parport/jobs/112534168 sound/soc/qcom/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 3cc252e..1e7a783 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -11,15 +11,18 @@ config SND_SOC_LPASS_CPU config SND_SOC_LPASS_PLATFORM tristate + depends on HAS_DMA select REGMAP_MMIO config SND_SOC_LPASS_IPQ806X tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM config SND_SOC_LPASS_APQ8016 tristate + depends on HAS_DMA select SND_SOC_LPASS_CPU select SND_SOC_LPASS_PLATFORM -- 1.9.1

1 1

[alsa-devel] [PATCH] ALSA: mts64: fix NULL pointer dereference
by Sudip Mukherjee 29 Feb '16

29 Feb '16

While registering pardev, the irq_func was also registered. As a result when we tried to probe for the card, an interrupt was generated and in the ISR we tried to dereference private_data. But private_data is still NULL as we have not yet done snd_mts64_create(). Lets probe for the card after mts64 is created. Reported-by: Fengguang Wu <fengguang.wu(a)intel.com> Fixes: 94a573500d48 ("ALSA: mts64: use new parport device model") Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- sound/drivers/mts64.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index c76bd87..fd4d18d 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -964,11 +964,6 @@ static int snd_mts64_probe(struct platform_device *pdev) err = -EIO; goto free_pardev; } - err = mts64_probe(p); - if (err) { - err = -EIO; - goto release_pardev; - } if ((err = snd_mts64_create(card, pardev, &mts)) < 0) { snd_printd("Cannot create main component\n"); @@ -976,6 +971,12 @@ static int snd_mts64_probe(struct platform_device *pdev) } card->private_data = mts; card->private_free = snd_mts64_card_private_free; + + err = mts64_probe(p); + if (err) { + err = -EIO; + goto __err; + } if ((err = snd_mts64_rawmidi_create(card)) < 0) { snd_printd("Creating Rawmidi component failed\n"); -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: hdsp: Fix wrong boolean ctl value accesses
by Takashi Iwai 29 Feb '16

29 Feb '16

snd-hdsp driver accesses enum item values (int) instead of boolean values (long) wrongly for some ctl elements. This patch fixes them. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme9652/hdsp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 2875b4f6d8c9..7c8941b8b2de 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -2879,7 +2879,7 @@ static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); + ucontrol->value.integer.value[0] = hdsp_dds_offset(hdsp); return 0; } @@ -2891,7 +2891,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; - val = ucontrol->value.enumerated.item[0]; + val = ucontrol->value.integer.value[0]; spin_lock_irq(&hdsp->lock); if (val != hdsp_dds_offset(hdsp)) change = (hdsp_set_dds_offset(hdsp, val) == 0) ? 1 : 0; -- 2.7.2

1 0

[alsa-devel] [PATCH 1/2] ALSA: hdspm: Fix wrong boolean ctl value accesses
by Takashi Iwai 29 Feb '16

29 Feb '16

snd-hdspm driver accesses enum item values (int) instead of boolean values (long) wrongly for some ctl elements. This patch fixes them. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/rme9652/hdspm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8bc8016c173d..b047b1ba48fd 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2261,7 +2261,7 @@ static int snd_hdspm_put_system_sample_rate(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - hdspm_set_dds_value(hdspm, ucontrol->value.enumerated.item[0]); + hdspm_set_dds_value(hdspm, ucontrol->value.integer.value[0]); return 0; } @@ -4449,7 +4449,7 @@ static int snd_hdspm_get_tco_word_term(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = hdspm->tco->term; + ucontrol->value.integer.value[0] = hdspm->tco->term; return 0; } @@ -4460,8 +4460,8 @@ static int snd_hdspm_put_tco_word_term(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - if (hdspm->tco->term != ucontrol->value.enumerated.item[0]) { - hdspm->tco->term = ucontrol->value.enumerated.item[0]; + if (hdspm->tco->term != ucontrol->value.integer.value[0]) { + hdspm->tco->term = ucontrol->value.integer.value[0]; hdspm_tco_write(hdspm); -- 2.7.2

1 1

[alsa-devel] [PATCH] ALSA: ctl: Fix incompatible ioctls for X32 ABI
by Takashi Iwai 29 Feb '16

29 Feb '16

The X32 ABI takes the same alignment like x86-64, and this may result in the incompatible struct size from ia32. Unfortunately, we hit this in some control ABI: struct snd_ctl_elem_value differs between them due to the position of 64bit variable array. This ends up with the unknown ioctl (ENOTTY) error. The fix is to add the compat entries for the new aligned struct. Reported-and-tested-by: Steven Newbury <steve(a)snewbury.org.uk> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control_compat.c | 90 +++++++++++++++++++++++++++++++++++++-------- 1 file changed, 74 insertions(+), 16 deletions(-) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index b9c0910fb8c4..dff3671217c8 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -170,6 +170,19 @@ struct snd_ctl_elem_value32 { unsigned char reserved[128]; }; +#ifdef CONFIG_X86_X32 +/* x32 has a different alignment for 64bit values from ia32 */ +struct snd_ctl_elem_value_x32 { + struct snd_ctl_elem_id id; + unsigned int indirect; /* bit-field causes misalignment */ + union { + s32 integer[128]; + unsigned char data[512]; + s64 integer64[64]; + } value; + unsigned char reserved[128]; +}; +#endif /* CONFIG_X86_X32 */ /* get the value type and count of the control */ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, @@ -219,9 +232,11 @@ static int get_elem_size(int type, int count) static int copy_ctl_value_from_user(struct snd_card *card, struct snd_ctl_elem_value *data, - struct snd_ctl_elem_value32 __user *data32, + void __user *userdata, + void __user *valuep, int *typep, int *countp) { + struct snd_ctl_elem_value32 __user *data32 = userdata; int i, type, size; int uninitialized_var(count); unsigned int indirect; @@ -239,8 +254,9 @@ static int copy_ctl_value_from_user(struct snd_card *card, if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || type == SNDRV_CTL_ELEM_TYPE_INTEGER) { for (i = 0; i < count; i++) { + s32 __user *intp = valuep; int val; - if (get_user(val, &data32->value.integer[i])) + if (get_user(val, &intp[i])) return -EFAULT; data->value.integer.value[i] = val; } @@ -250,8 +266,7 @@ static int copy_ctl_value_from_user(struct snd_card *card, dev_err(card->dev, "snd_ioctl32_ctl_elem_value: unknown type %d\n", type); return -EINVAL; } - if (copy_from_user(data->value.bytes.data, - data32->value.data, size)) + if (copy_from_user(data->value.bytes.data, valuep, size)) return -EFAULT; } @@ -261,7 +276,8 @@ static int copy_ctl_value_from_user(struct snd_card *card, } /* restore the value to 32bit */ -static int copy_ctl_value_to_user(struct snd_ctl_elem_value32 __user *data32, +static int copy_ctl_value_to_user(void __user *userdata, + void __user *valuep, struct snd_ctl_elem_value *data, int type, int count) { @@ -270,22 +286,22 @@ static int copy_ctl_value_to_user(struct snd_ctl_elem_value32 __user *data32, if (type == SNDRV_CTL_ELEM_TYPE_BOOLEAN || type == SNDRV_CTL_ELEM_TYPE_INTEGER) { for (i = 0; i < count; i++) { + s32 __user *intp = valuep; int val; val = data->value.integer.value[i]; - if (put_user(val, &data32->value.integer[i])) + if (put_user(val, &intp[i])) return -EFAULT; } } else { size = get_elem_size(type, count); - if (copy_to_user(data32->value.data, - data->value.bytes.data, size)) + if (copy_to_user(valuep, data->value.bytes.data, size)) return -EFAULT; } return 0; } -static int snd_ctl_elem_read_user_compat(struct snd_card *card, - struct snd_ctl_elem_value32 __user *data32) +static int ctl_elem_read_user(struct snd_card *card, + void __user *userdata, void __user *valuep) { struct snd_ctl_elem_value *data; int err, type, count; @@ -294,7 +310,9 @@ static int snd_ctl_elem_read_user_compat(struct snd_card *card, if (data == NULL) return -ENOMEM; - if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0) + err = copy_ctl_value_from_user(card, data, userdata, valuep, + &type, &count); + if (err < 0) goto error; snd_power_lock(card); @@ -303,14 +321,15 @@ static int snd_ctl_elem_read_user_compat(struct snd_card *card, err = snd_ctl_elem_read(card, data); snd_power_unlock(card); if (err >= 0) - err = copy_ctl_value_to_user(data32, data, type, count); + err = copy_ctl_value_to_user(userdata, valuep, data, + type, count); error: kfree(data); return err; } -static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, - struct snd_ctl_elem_value32 __user *data32) +static int ctl_elem_write_user(struct snd_ctl_file *file, + void __user *userdata, void __user *valuep) { struct snd_ctl_elem_value *data; struct snd_card *card = file->card; @@ -320,7 +339,9 @@ static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, if (data == NULL) return -ENOMEM; - if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0) + err = copy_ctl_value_from_user(card, data, userdata, valuep, + &type, &count); + if (err < 0) goto error; snd_power_lock(card); @@ -329,12 +350,39 @@ static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, err = snd_ctl_elem_write(card, file, data); snd_power_unlock(card); if (err >= 0) - err = copy_ctl_value_to_user(data32, data, type, count); + err = copy_ctl_value_to_user(userdata, valuep, data, + type, count); error: kfree(data); return err; } +static int snd_ctl_elem_read_user_compat(struct snd_card *card, + struct snd_ctl_elem_value32 __user *data32) +{ + return ctl_elem_read_user(card, data32, &data32->value); +} + +static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, + struct snd_ctl_elem_value32 __user *data32) +{ + return ctl_elem_write_user(file, data32, &data32->value); +} + +#ifdef CONFIG_X86_X32 +static int snd_ctl_elem_read_user_x32_compat(struct snd_card *card, + struct snd_ctl_elem_value_x32 __user *data32) +{ + return ctl_elem_read_user(card, data32, &data32->value); +} + +static int snd_ctl_elem_write_user_x32_compat(struct snd_ctl_file *file, + struct snd_ctl_elem_value_x32 __user *data32) +{ + return ctl_elem_write_user(file, data32, &data32->value); +} +#endif /* CONFIG_X86_X32 */ + /* add or replace a user control */ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, struct snd_ctl_elem_info32 __user *data32, @@ -391,6 +439,10 @@ enum { SNDRV_CTL_IOCTL_ELEM_INFO32 = _IOWR('U', 0x11, struct snd_ctl_elem_info32), SNDRV_CTL_IOCTL_ELEM_READ32 = _IOWR('U', 0x12, struct snd_ctl_elem_value32), SNDRV_CTL_IOCTL_ELEM_WRITE32 = _IOWR('U', 0x13, struct snd_ctl_elem_value32), +#ifdef CONFIG_X86_X32 + SNDRV_CTL_IOCTL_ELEM_READ_X32 = _IOWR('U', 0x12, struct snd_ctl_elem_value_x32), + SNDRV_CTL_IOCTL_ELEM_WRITE_X32 = _IOWR('U', 0x13, struct snd_ctl_elem_value_x32), +#endif SNDRV_CTL_IOCTL_ELEM_ADD32 = _IOWR('U', 0x17, struct snd_ctl_elem_info32), SNDRV_CTL_IOCTL_ELEM_REPLACE32 = _IOWR('U', 0x18, struct snd_ctl_elem_info32), }; @@ -427,6 +479,12 @@ static inline long snd_ctl_ioctl_compat(struct file *file, unsigned int cmd, uns return snd_ctl_elem_read_user_compat(ctl->card, argp); case SNDRV_CTL_IOCTL_ELEM_WRITE32: return snd_ctl_elem_write_user_compat(ctl, argp); +#ifdef CONFIG_X86_X32 + case SNDRV_CTL_IOCTL_ELEM_READ_X32: + return snd_ctl_elem_read_user_x32_compat(ctl->card, argp); + case SNDRV_CTL_IOCTL_ELEM_WRITE_X32: + return snd_ctl_elem_write_user_x32_compat(ctl, argp); +#endif case SNDRV_CTL_IOCTL_ELEM_ADD32: return snd_ctl_elem_add_compat(ctl, argp, 0); case SNDRV_CTL_IOCTL_ELEM_REPLACE32: -- 2.7.2

1 1

[alsa-devel] ERROR: USB Sound card with unsafe characters
by Brian Jones 29 Feb '16

29 Feb '16

Newby alert - but I don't know how to get this bug to the developers I have been trying to work on a fix for Bug 1498129 - https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1498129 on Ubuntu My USB soundcard is an Asus Xonar U7 Echelon Edition <https://www.asus.com/Sound-Cards/Xonar_U7_Echelon_Edition/> which identifies itself to alsa as "Xonar U7 Echelon Ed." The card is supported by alsa and works well in analog mode but I cannot get digital output via alsa. It has iec958 as device 1 rather than device 0 and I can access digital output by referring to directly to hw:x,1 but cannot access digital output via alsa. Attempts to change the iec958 device in USB-Audio.conf fail because of the period char and apparently my level of OS (Mint 17.3 based on Ubuntu 14.04, kernel 3.19.0-32) does not have alsa supporting USB vendorid:productid naming so I don't have a workaround. If a patch exists, advice on applying/rebuilding alsa would be gratefully appreciated.

2 2

[alsa-devel] [PATCH] ASoC: nau8825: reduce stanby power consumption
by John Hsu 28 Feb '16

28 Feb '16

Fix unnecessary widget power up when standby; And decrease internal clock frequency to save power. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 49 +++++++++++++++++++++++++++++++++------------- 1 file changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 52882e4..f9752cc 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -262,6 +262,29 @@ static int nau8825_playback_event(struct snd_soc_dapm_widget *w, return 0; } +static int nau8825_dacr_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + break; + case SND_SOC_DAPM_POST_PMD: + if (!nau8825->irq) + regmap_update_bits(nau8825->regmap, + NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + static const char * const nau8825_adc_decimation[] = { "32", "64", "128", "256" @@ -337,16 +360,17 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { 0), /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), - SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC_E("DDACR", NULL, SND_SOC_NOPM, 0, 0, + nau8825_dacr_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACL_SFT, 0), SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), @@ -878,13 +902,6 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* The interrupt clock is gated by x1[10:8], - * one of them needs to be enabled all the time for - * interrupts to happen. - */ - snd_soc_dapm_force_enable_pin(dapm, "DDACR"); - snd_soc_dapm_sync(dapm); - /* Unmask interruptions. Handler uses dapm object so we can enable * interruptions only after dapm is fully initialized. */ @@ -1075,6 +1092,12 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, NAU8825_DCO_EN); regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; @@ -1309,9 +1332,7 @@ static int nau8825_setup_irq(struct nau8825 *nau8825) /* Enable internal VCO needed for interruptions */ nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - /* Enable DDACR needed for interrupts - * It is the same as force_enable_pin("DDACR") we do later - */ + /* Enable DDACR needed for interrupts */ regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); -- 2.6.4

2 5

[alsa-devel] [PATCH] ALSA: dice: drop duplex streams synchronization to transfer own time stamps
by Takashi Sakamoto 28 Feb '16

28 Feb '16

This commit drops implementation of duplex streams synchronization from ALSA dice driver, due to a reason of hardware design. This patch allows dice-based units to generate sounds correctly when isochronous packet streaming starts at first time. In IEC 61883-6:2005, CIP packetization layer for AM824 data format utilizes the value of SYT field in CIP header of received packet for a reference to phase lock loop. Figure 3 in clause 4.3 describes it. The value is an offset from cycle_time field of every cycle start packet from cycle master on IEEE 1394 bus. The time calculated with these two fields is called as 'presentation timestamp' which represents the time to play data included in the packet. Although, this idea includes some problems due to accuracy of timekeep in cycle master, accuracy of transmission of cycle start packet on the bus with the other units, accuracy of sampling clock in data transmitter side and accuracy of replay in data receiver side. In most case, these accuracies somewhat worse because there's no such ideal hardwares in this world. For the issues, ASICs for Dice include Jitter Elimination Technologies (JET) PLL. The PLL can handle several sources of clock and compensate it with high-precision internal clock source. The sequence of value in syt field of received AMDTP packets is one of the sources, therefore transmitters on IEEE 1394 bus should transfer it. On the other hand, current ALSA dice driver is programmed with a mode of duplex streams with synchronization. In this mode, the driver outputs packets after some incoming packets are handled, to re-use the value of SYT field in incoming packets to the value for outgoing packets. This mode is enabled when source signal of sampling clock is set to internal, and this is a major use case. Thus, in most cases, the unit receives no packets during a short time after packet streaming starts. As long as I experienced, this causes the units to generate no sounds at first time to receive packets. This issue occurs only with Dice II. I guess this is due to a quirk of the PLL. In short, the PLL cannot generate firm signals to ADCs/DACs or the other ICs when no packets are received in the beginning of packet streaming. While, on second time or later, the unit generates sound correctly. I guess that starting packet streaming at first time sets the PLL correctly. Well, still based on my hypothesis and no way to prove it, this commit drops duplex streams synchronization from this driver. At least, the PLL requires the sequence of value in SYT field of received AMDTP packets as one of source of clock signals with internal clock source. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-stream.c | 42 ++------------------------------------- 1 file changed, 2 insertions(+), 40 deletions(-) diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index a64b3cc..df035b1 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -191,53 +191,17 @@ end: return err; } -static int get_sync_mode(struct snd_dice *dice, enum cip_flags *sync_mode) -{ - u32 source; - int err; - - err = snd_dice_transaction_get_clock_source(dice, &source); - if (err < 0) - goto end; - - switch (source) { - /* So-called 'SYT Match' modes, sync_to_syt value of packets received */ - case CLOCK_SOURCE_ARX4: /* in 4th stream */ - case CLOCK_SOURCE_ARX3: /* in 3rd stream */ - case CLOCK_SOURCE_ARX2: /* in 2nd stream */ - err = -ENOSYS; - break; - case CLOCK_SOURCE_ARX1: /* in 1st stream, which this driver uses */ - *sync_mode = 0; - break; - default: - *sync_mode = CIP_SYNC_TO_DEVICE; - break; - } -end: - return err; -} - int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) { struct amdtp_stream *master, *slave; unsigned int curr_rate; - enum cip_flags sync_mode; int err = 0; if (dice->substreams_counter == 0) goto end; - err = get_sync_mode(dice, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &dice->tx_stream; - slave = &dice->rx_stream; - } else { - master = &dice->rx_stream; - slave = &dice->tx_stream; - } + master = &dice->rx_stream; + slave = &dice->tx_stream; /* Some packet queueing errors. */ if (amdtp_streaming_error(master) || amdtp_streaming_error(slave)) @@ -261,8 +225,6 @@ int snd_dice_stream_start_duplex(struct snd_dice *dice, unsigned int rate) stop_stream(dice, slave); snd_dice_transaction_clear_enable(dice); - amdtp_stream_set_sync(sync_mode, master, slave); - err = ensure_phase_lock(dice); if (err < 0) { dev_err(&dice->unit->device, -- 2.5.0

2 1

[alsa-devel] [PATCH 0/2] clk: imx6: add kpp clock for i.MX6UL
by Lothar Waßmann 28 Feb '16

28 Feb '16

This patchset adds the clock which is necessary to operate the KPP unit on i.MX6UL. The first patch removes bogus whitespace before TABs in indentation. The second patch adds the clock definition.

3 4

[alsa-devel] [PATCH 1/4] ASoC: wm9713: add binding for WM9713 codec
by Robert Jarzmik 27 Feb '16

27 Feb '16

This adds a binding for the Wolfson WM9713 audio codec. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Documentation/devicetree/bindings/sound/wm9713.txt | 14 ++++++++++++++ 1 file changed, 14 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm9713.txt diff --git a/Documentation/devicetree/bindings/sound/wm9713.txt b/Documentation/devicetree/bindings/sound/wm9713.txt new file mode 100644 index 000000000000..761001bc3201 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm9713.txt @@ -0,0 +1,14 @@ +WM9713 audio CODEC + +This devices supports I2C. + +Required properties: + - compatible : + "wlf,wm9713" + +Example: + wm9713: wm9713@0 { + compatible = "wlf,wm9713"; + #sound-dai-cells = <1>; + status = "okay"; + }; -- 2.1.4

3 16

[alsa-devel] [PATCH] ASoC: rt5640: add master clock handling for rt5640
by Sugar Zhang 27 Feb '16

27 Feb '16

enable/disable master clock when codec is active or not. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Documentation/devicetree/bindings/sound/rt5640.txt | 3 +++ sound/soc/codecs/rt5640.c | 31 ++++++++++++++++++++++ sound/soc/codecs/rt5640.h | 2 ++ 3 files changed, 36 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index 9e62f6e..57fe646 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -12,6 +12,9 @@ Required properties: Optional properties: +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + - realtek,in1-differential - realtek,in2-differential - realtek,in3-differential diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 11d032c..6cd84fb 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1949,7 +1949,33 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, static int rt5640_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* + * SND_SOC_BIAS_PREPARE is called while preparing for a + * transition to ON or away from ON. If current bias_level + * is SND_SOC_BIAS_ON, then it is preparing for a transition + * away from ON. Disable the clock in that case, otherwise + * enable it. + */ + if (IS_ERR(rt5640->mclk)) + break; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_ON) { + clk_disable_unprepare(rt5640->mclk); + } else { + ret = clk_prepare_enable(rt5640->mclk); + if (ret) + return ret; + } + break; + case SND_SOC_BIAS_STANDBY: if (SND_SOC_BIAS_OFF == snd_soc_codec_get_bias_level(codec)) { snd_soc_update_bits(codec, RT5640_PWR_ANLG1, @@ -2088,6 +2114,11 @@ static int rt5640_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + /* Check if MCLK provided */ + rt5640->mclk = devm_clk_get(codec->dev, "mclk"); + if (PTR_ERR(rt5640->mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + rt5640->codec = codec; snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 83a7150..1761c3a9 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -12,6 +12,7 @@ #ifndef _RT5640_H #define _RT5640_H +#include <linux/clk.h> #include <sound/rt5640.h> /* Info */ @@ -2097,6 +2098,7 @@ struct rt5640_priv { struct snd_soc_codec *codec; struct rt5640_platform_data pdata; struct regmap *regmap; + struct clk *mclk; int sysclk; int sysclk_src; -- 1.9.1

5 9

[alsa-devel] [PATCH] ALSA: hda - Autosuspend controller after probe even if codecs are already suspended
by ville.syrjala＠linux.intel.com 26 Feb '16

26 Feb '16

From: Ville Syrjälä <ville.syrjala(a)linux.intel.com> azx_probe_continue() uses pm_runtime_put_noidle() to drop the rpm usage_count, which means that if it's the last reference the autosuspend of the controller won't actually happen. So if the codecs autosuspend before the azx_probe_continue() drops the last reference we'll fail to autosuspend the controller. This does happen in practice, but not every time. As can be seen in [1] the controller autosuspend attempt fails due to the usage_count when suspending the codecs. A bit later we see the the contoller usage_count dropping to zero without further attempts at autosuspend. Fix the problem by using pm_runtime_put_autosuspend() instead, which will kick off the autosuspend of the controller even if the codecs are already asleep. As can be seen in [2] the controller autosuspend still fails while suspending the codecs, but later on we see another autosuspend attempt after dropping the usage_count to 0. I was also a bit worried that there might still be a race between the controller autosuspend and the rest of the code in azx_probe_continue(). So I also tried replacing the the put_noidle() with put_sync_suspend(). No explosions occurred, so I'm somewhat satisfied that there are no serious problems in this area. [1] kworker/1:2-122 [001] .... 63.661310: __pm_runtime_suspend: hdaudioC0D0 usage_count 0 kworker/1:2-122 [001] d..2 63.661316: rpm_suspend: hdaudioC0D0 flags-d cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:2-122 [001] d..1 63.661317: rpm_check_suspend_allowed: hdaudioC0D0 retval 0 kworker/1:2-122 [001] d..2 63.661332: rpm_return_int: rpm_suspend+0x406/0x5e8:hdaudioC0D0 ret=0 kworker/1:1-72 [001] d..2 63.661543: rpm_suspend: hdaudioC0D0 flags-a cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:1-72 [001] d..1 63.661544: rpm_check_suspend_allowed: hdaudioC0D0 retval 0 kworker/1:1-72 [001] .... 63.661545: hda_codec_runtime_suspend: hdaudioC0D0 suspend kworker/1:1-72 [001] d..2 63.661614: rpm_idle: 0000:00:03.0 flags-1 cnt-1 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:1-72 [001] d..1 63.661615: rpm_check_suspend_allowed: 0000:00:03.0 usage_count 1 kworker/1:1-72 [001] d..1 63.661615: rpm_check_suspend_allowed: 0000:00:03.0 retval -11 kworker/1:1-72 [001] d..2 63.661616: rpm_return_int: rpm_idle+0x249/0x487:0000:00:03.0 ret=-11 kworker/1:1-72 [001] d..2 63.661616: rpm_return_int: rpm_suspend+0x406/0x5e8:hdaudioC0D0 ret=0 kworker/1:2-122 [001] d..2 63.664834: rpm_idle: hdaudioC0D0 flags-8 cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:2-122 [001] d..1 63.664835: rpm_check_suspend_allowed: hdaudioC0D0 retval 1 kworker/1:2-122 [001] d..2 63.664836: rpm_return_int: rpm_idle+0x249/0x487:hdaudioC0D0 ret=-11 kworker/1:2-122 [001] d..2 63.664841: rpm_idle: hdaudioC0D0 flags-8 cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/1:2-122 [001] d..1 63.664841: rpm_check_suspend_allowed: hdaudioC0D0 retval 1 kworker/1:2-122 [001] d..2 63.664841: rpm_return_int: rpm_idle+0x249/0x487:hdaudioC0D0 ret=-11 kworker/1:2-122 [001] .... 63.664842: azx_probe_continue: 0000:00:03.0 usage_count=0 [2] kworker/0:0-4 [000] .... 50.354567: __pm_runtime_suspend: hdaudioC0D0 usage_count 0 kworker/0:0-4 [000] d..2 50.354574: rpm_suspend: hdaudioC0D0 flags-d cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.354575: rpm_check_suspend_allowed: hdaudioC0D0 retval 0 kworker/0:0-4 [000] d..2 50.354589: rpm_return_int: rpm_suspend+0x406/0x5e8:hdaudioC0D0 ret=0 kworker/0:2-135 [000] d..2 50.354809: rpm_suspend: hdaudioC0D0 flags-a cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:2-135 [000] d..1 50.354810: rpm_check_suspend_allowed: hdaudioC0D0 retval 0 kworker/0:2-135 [000] .... 50.354816: hda_codec_runtime_suspend: hdaudioC0D0 suspend kworker/0:2-135 [000] d..2 50.354908: rpm_idle: 0000:00:03.0 flags-1 cnt-1 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:2-135 [000] d..1 50.354909: rpm_check_suspend_allowed: 0000:00:03.0 usage_count 1 kworker/0:2-135 [000] d..1 50.354909: rpm_check_suspend_allowed: 0000:00:03.0 retval -11 kworker/0:2-135 [000] d..2 50.354909: rpm_return_int: rpm_idle+0x249/0x487:0000:00:03.0 ret=-11 kworker/0:2-135 [000] d..2 50.354910: rpm_return_int: rpm_suspend+0x406/0x5e8:hdaudioC0D0 ret=0 kworker/0:0-4 [000] d..2 50.373791: rpm_idle: hdaudioC0D0 flags-8 cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.373792: rpm_check_suspend_allowed: hdaudioC0D0 retval 1 kworker/0:0-4 [000] d..2 50.373793: rpm_return_int: rpm_idle+0x249/0x487:hdaudioC0D0 ret=-11 kworker/0:0-4 [000] d..2 50.373797: rpm_idle: hdaudioC0D0 flags-8 cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.373798: rpm_check_suspend_allowed: hdaudioC0D0 retval 1 kworker/0:0-4 [000] d..2 50.373798: rpm_return_int: rpm_idle+0x249/0x487:hdaudioC0D0 ret=-11 kworker/0:0-4 [000] .... 50.373799: __pm_runtime_suspend: 0000:00:03.0 usage_count 0 kworker/0:0-4 [000] d..2 50.373800: rpm_suspend: 0000:00:03.0 flags-d cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.373800: rpm_check_suspend_allowed: 0000:00:03.0 retval 0 kworker/0:0-4 [000] d..2 50.373803: rpm_return_int: rpm_suspend+0x406/0x5e8:0000:00:03.0 ret=0 kworker/0:0-4 [000] d..2 50.385164: rpm_suspend: 0000:00:03.0 flags-a cnt-0 dep-0 auto-1 p-0 irq-0 child-0 kworker/0:0-4 [000] d..1 50.385165: rpm_check_suspend_allowed: 0000:00:03.0 retval 0 kworker/0:0-4 [000] .... 50.385174: azx_runtime_suspend: 0000:00:03.0 azx suspend releaseing power well kworker/0:0-4 [000] .... 50.385179: azx_runtime_suspend: 0000:00:03.0 azx suspend kworker/0:0-4 [000] d..2 50.386872: rpm_return_int: rpm_suspend+0x406/0x5e8:0000:00:03.0 ret=0 Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Ville Syrjälä <ville.syrjala(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e5240cb3749f..2624cfe98884 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2145,7 +2145,7 @@ static int azx_probe_continue(struct azx *chip) azx_add_card_list(chip); snd_hda_set_power_save(&chip->bus, power_save * 1000); if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) - pm_runtime_put_noidle(&pci->dev); + pm_runtime_put_autosuspend(&pci->dev); out_free: if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL -- 2.4.10

2 1

[alsa-devel] [PATCH] ASoC: Extend FLL function
by John Hsu 26 Feb '16

26 Feb '16

Extend FFL clock source from MCLK/BCLK/FS; And modify some FLL parameter setting. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 100 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/nau8825.h | 14 ++++++- 2 files changed, 86 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 7fc7b4e..16da8ef 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -926,7 +926,8 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, struct nau8825_fll *fll_param) { regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, fll_param->mclk_src); + NAU8825_CLK_SRC_MASK | NAU8825_CLK_MCLK_SRC_MASK, + NAU8825_CLK_SRC_MCLK | fll_param->mclk_src); regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, NAU8825_FLL_RATIO_MASK, fll_param->ratio); /* FLL 16-bit fractional input */ @@ -939,10 +940,20 @@ static void nau8825_fll_apply(struct nau8825 *nau8825, NAU8825_FLL_REF_DIV_MASK, fll_param->clk_ref_div); /* select divided VCO input */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL5, - NAU8825_FLL_FILTER_SW_MASK, 0x0000); - /* FLL sigma delta modulator enable */ + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FILTER_SW_MASK, + NAU8825_FLL_PDB_DAC_EN | NAU8825_FLL_LOOP_FTR_EN | + NAU8825_FLL_FILTER_SW_REF); + /* Disable free-running mode */ regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + NAU8825_DCO_EN, 0); + /* FLL sigma delta modulator enable */ + if (fll_param->fll_frac) + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN_MASK, NAU8825_SDM_EN); + else + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN_MASK, 0); } /* freq_out must be 256*Fs in order to achieve the best performance */ @@ -970,6 +981,37 @@ static int nau8825_set_pll(struct snd_soc_codec *codec, int pll_id, int source, return 0; } +static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) +{ + int ret = 0; + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + goto done; + + if (!nau8825->mclk_freq) { + ret = clk_prepare_enable(nau8825->mclk); + if (ret) { + dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); + goto done; + } + } + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + ret = clk_set_rate(nau8825->mclk, freq); + if (ret) { + dev_err(nau8825->dev, "Unable to set mclk rate\n"); + goto done; + } + } + +done: + return ret; +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -981,29 +1023,9 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); - - /* We selected MCLK source but the clock itself managed externally */ - if (!nau8825->mclk) - break; - - if (!nau8825->mclk_freq) { - ret = clk_prepare_enable(nau8825->mclk); - if (ret) { - dev_err(nau8825->dev, "Unable to prepare codec mclk\n"); - return ret; - } - } - - if (nau8825->mclk_freq != freq) { - nau8825->mclk_freq = freq; - - freq = clk_round_rate(nau8825->mclk, freq); - ret = clk_set_rate(nau8825->mclk, freq); - if (ret) { - dev_err(nau8825->dev, "Unable to set mclk rate\n"); - return ret; - } - } + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; break; case NAU8825_CLK_INTERNAL: @@ -1018,6 +1040,30 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, } break; + case NAU8825_CLK_FLL_MCLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_MCLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_BLK: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_BLK); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; + case NAU8825_CLK_FLL_FS: + regmap_update_bits(regmap, NAU8825_REG_FLL3, + NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_FS); + ret = nau8825_mclk_prepare(nau8825, freq); + if (ret) + return ret; + + break; default: dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); return -EINVAL; diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index dff8edb..e8f7ca5 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -112,12 +112,21 @@ /* FLL3 (0x06) */ #define NAU8825_FLL_INTEGER_MASK (0x3ff << 0) +#define NAU8825_FLL_CLK_SRC_SFT 10 +#define NAU8825_FLL_CLK_SRC_MASK (0x3 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_MCLK (0 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_BLK (0x2 << NAU8825_FLL_CLK_SRC_SFT) +#define NAU8825_FLL_CLK_SRC_FS (0x3 << NAU8825_FLL_CLK_SRC_SFT) /* FLL4 (0x07) */ #define NAU8825_FLL_REF_DIV_MASK (0x3 << 10) /* FLL5 (0x08) */ -#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 14) +#define NAU8825_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8825_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8825_FLL_FILTER_SW_MASK (0x1 << 13) +#define NAU8825_FLL_FILTER_SW_N2 (0x1 << 13) +#define NAU8825_FLL_FILTER_SW_REF (0x0 << 13) /* FLL6 (0x9) */ #define NAU8825_DCO_EN_MASK (0x1 << 15) @@ -306,6 +315,9 @@ enum { NAU8825_CLK_MCLK = 0, NAU8825_CLK_INTERNAL, + NAU8825_CLK_FLL_MCLK, + NAU8825_CLK_FLL_BLK, + NAU8825_CLK_FLL_FS, }; struct nau8825 { -- 1.9.1

3 7

[alsa-devel] [PATCH RESEND] ASoC: pxa: remove unused variable
by Arnd Bergmann 26 Feb '16

26 Feb '16

As pointed out by Zhangfei Gao, the sspa_div variable in brownstone_wm8994_hw_params() is completely unused, so as a cleanup following a prior patch, this removes both the variable and the division. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- Submitted originally on Nov 23 2015, this apparently got lost, resending it now. sound/soc/pxa/brownstone.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 416ea646c3b1..ec522e94b0e2 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -52,7 +52,6 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; int freq_out, sspa_mclk, sysclk; - int sspa_div; if (params_rate(params) > 11025) { freq_out = params_rate(params) * 512; @@ -63,7 +62,6 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, sysclk = params_rate(params) * 512; sspa_mclk = params_rate(params) * 64; } - sspa_div = freq_out / sspa_mclk; snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk); -- 2.7.0

2 2

Re: [alsa-devel] Fwd: Lenovo Y700-17ISK notebook - multiple issues
by Takashi Iwai 26 Feb '16

26 Feb '16

On Wed, 24 Feb 2016 21:20:06 +0100, Aljosa wrote: > > OS: Ubuntu 16.04 > I tested first the original Ubuntu kernel and now I'm running > 4.5-rc5-wily. > I didn't test any model option because I don't know how to do it (I'm > not an expert). > > Please find attached the 'alsa-info.sh output' file. Please don't top-post. Also, again, don't drop Cc to ML. Judging from alsa-info.sh output, there is no pin declared for the bass output by BIOS. So you'd need to identify the pin by yourself, e.g. via hdajackretask, and set it up manually. Once when you figuring out which pin corresponds to the bass output, let us know. We can add it statically in the driver fixup. thanks, Takashi > > Thanks, ciao :) > Aljosa > > > > On 2016-02-24 20:29, Takashi Iwai wrote: > > On Wed, 24 Feb 2016 19:00:36 +0100, > > Aljosa wrote: > >> > >> Hi Takashi, > >> thank you very much for your reply. > >> Notebook subwoofer doesn't work at all (it does not produce any > >> sound). > > > > OK, could you give more details? Which kernel did you test? Did you > > test any model option? > > > > Also give alsa-info.sh output (run it with --no-upload option and > > attach the output). This contains the h/w details. > > > > Last but not least, don't drop Cc to ML. > > > > > > thanks, > > > > Takashi > > > >> Regards, Aljosa > >> > >> > >> > >> On 2016-02-24 16:35, Takashi Iwai wrote: > >> > On Tue, 23 Feb 2016 20:37:21 +0100, > >> > Aljosa wrote: > >> >> > >> >> Hello, > >> >> advised by one of the kernel developers - I'm reporting you the > >> >> problem > >> >> nr.2 from the below link. > >> > > >> > Could you describe what are the exact issues, instead of pointing some > >> > URL? > >> > > >> > > >> > thanks, > >> > > >> > Takashi > >> > upload=true&script=true&cardinfo= > !!################################ > !!ALSA Information Script v 0.4.64 > !!################################ > > !!Script ran on: Wed Feb 24 20:16:58 UTC 2016 > > > !!Linux Distribution > !!------------------ > > Ubuntu Xenial Xerus (development branch) \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu Xenial Xerus (development branch)" NAME="Ubuntu" ID=ubuntu ID_LIKE=debian PRETTY_NAME="Ubuntu 16.04" HOME_URL="http://www.ubuntu.com/" SUPPORT_URL="http://help.ubuntu.com/" BUG_REPORT_URL="http://bugs.launchpad.net/ubuntu/" UBUNTU_CODENAME=xenial > > > !!DMI Information > !!--------------- > > Manufacturer: LENOVO > Product Name: 80Q0 > Product Version: Lenovo ideapad Y700-17ISK > Firmware Version: CDCN26WW > > > !!Kernel Information > !!------------------ > > Kernel release: 4.5.0-040500rc5-generic > Operating System: GNU/Linux > Architecture: x86_64 > Processor: x86_64 > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: k4.5.0-040500rc5-generic > Library version: 1.1.0 > Utilities version: 1.1.0 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [PCH ]: HDA-Intel - HDA Intel PCH > HDA Intel PCH at 0x94320000 irq 132 > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1f.3 Audio device: Intel Corporation Sunrise Point-H HD Audio (rev 31) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!------------------------------------------------------- > > 00:1f.3 0403: 8086:a170 (rev 31) > Subsystem: 17aa:3802 > > > !!Modprobe options (Sound related) > !!-------------------------------- > > snd_pcsp: index=-2 > snd_usb_audio: index=-2 > snd_atiixp_modem: index=-2 > snd_intel8x0m: index=-2 > snd_via82xx_modem: index=-2 > snd_atiixp_modem: index=-2 > snd_intel8x0m: index=-2 > snd_via82xx_modem: index=-2 > snd_usb_audio: index=-2 > snd_usb_caiaq: index=-2 > snd_usb_ua101: index=-2 > snd_usb_us122l: index=-2 > snd_usb_usx2y: index=-2 > snd_cmipci: mpu_port=0x330 fm_port=0x388 > snd_pcsp: index=-2 > snd_usb_audio: index=-2 > > > !!Loaded sound module options > !!--------------------------- > > !!Module: snd_hda_intel > align_buffer_size : -1 > bdl_pos_adj : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > beep_mode : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N > enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y > enable_msi : -1 > id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > jackpoll_ms : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) > position_fix : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > power_save : 0 > power_save_controller : Y > probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 > probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 > single_cmd : N > snoop : -1 > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: Realtek ALC233 > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x10ec0235 > Subsystem Id: 0x17aa3829 > Revision Id: 0x100002 > No Modem Function Group found > Default PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > State of AFG node 0x01: > Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS > Power: setting=D0, actual=D0 > GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 > IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 > Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out > Control: name="Speaker Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x50 0x50] > Converter: stream=1, channel=0 > PCM: > rates [0x60]: 44100 48000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out > Control: name="Headphone Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Device: name="ALC233 Analog", type="Audio", device=0 > Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 > Amp-Out vals: [0x00 0x00] > Converter: stream=1, channel=0 > PCM: > rates [0x60]: 44100 48000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x06 [Audio Output] wcaps 0x611: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x5e0]: 44100 48000 88200 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x08 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Control: name="Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Control: name="Capture Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Device: name="ALC233 Analog", type="Audio", device=0 > Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 > Amp-In vals: [0x30 0x30] > Converter: stream=1, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Connection: 1 > 0x23 > Node 0x09 [Audio Input] wcaps 0x10051b: Stereo Amp-In > Amp-In caps: ofs=0x17, nsteps=0x3f, stepsize=0x02, mute=1 > Amp-In vals: [0x97 0x97] > Converter: stream=0, channel=0 > SDI-Select: 0 > PCM: > rates [0x560]: 44100 48000 96000 192000 > bits [0xe]: 16 20 24 > formats [0x1]: PCM > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 1 > 0x22 > Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Control: name="Mic Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=1, ofs=0 > Control: name="Mic Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=1, ofs=0 > Control: name="Beep Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=4, ofs=0 > Control: name="Beep Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=4, ofs=0 > Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 5 > 0x18 0x19 0x1a 0x1b 0x1d > Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x02 0x0b > Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] [0x00 0x00] > Connection: 2 > 0x03 0x0b > Node 0x0e [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x0f [Audio Mixer] wcaps 0x20010a: Mono Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x00 0x00] > Connection: 1 > 0x0d > Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x12 [Pin Complex] wcaps 0x40040b: Stereo Amp-In > Control: name="Internal Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00000020: IN > Pin Default 0x90a60140: [Fixed] Mic at Int N/A > Conn = Digital, Color = Unknown > DefAssociation = 0x4, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x14 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > Control: name="Speaker Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x00010014: OUT EAPD Detect > EAPD 0x2: EAPD > Pin Default 0x90170110: [Fixed] Speaker at Int N/A > Conn = Analog, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D0, actual=D0 > Connection: 1 > 0x0c > Node 0x15 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x16 [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x17 [Pin Complex] wcaps 0x40050c: Mono Amp-Out > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80] > Pincap 0x00000010: OUT > Pin Default 0x40000000: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x0, Sequence = 0x0 > Pin-ctls: 0x00: > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 1 > 0x0f > Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00003724: IN Detect > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In > Control: name="Mic Boost Volume", index=0, device=0 > ControlAmp: chs=3, dir=In, idx=0, ofs=0 > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00003724: IN Detect > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x04a11030: [Jack] Mic at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x3, Sequence = 0x0 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=02, enabled=1 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Pincap 0x00003724: IN Detect > Vref caps: HIZ 50 GRD 80 100 > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out > Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 > Amp-In vals: [0x00 0x00] > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0001373c: IN OUT HP EAPD Detect > Vref caps: HIZ 50 GRD 80 100 > EAPD 0x2: EAPD > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN VREF_HIZ > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 2 > 0x0c* 0x0d > Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x1d [Pin Complex] wcaps 0x400400: Mono > Pincap 0x00000020: IN > Pin Default 0x40f79a2d: [N/A] Other at Ext N/A > Conn = Analog, Color = Pink > DefAssociation = 0x2, Sequence = 0xd > Pin-ctls: 0x20: IN > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Node 0x1e [Pin Complex] wcaps 0x400781: Stereo Digital > Pincap 0x00000014: OUT Detect > Pin Default 0x411111f0: [N/A] Speaker at Ext Rear > Conn = 1/8, Color = Black > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x40: OUT > Unsolicited: tag=00, enabled=0 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 1 > 0x06 > Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono > Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono > Processing caps: benign=0, ncoeff=76 > Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out > Control: name="Headphone Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x80 0x80] > Pincap 0x0001001c: OUT HP EAPD Detect > EAPD 0x2: EAPD > Pin Default 0x04211020: [Jack] HP Out at Ext Right > Conn = 1/8, Color = Black > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=01, enabled=1 > Power states: D0 D1 D2 D3 EPSS > Power: setting=D3, actual=D3 > Connection: 2 > 0x0c 0x0d* > Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] > Connection: 6 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b > Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In > Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x00 0x00] > Connection: 7 > 0x18 0x19 0x1a 0x1b 0x1d 0x0b 0x12 > Codec: Intel Skylake HDMI > Address: 2 > AFG Function Id: 0x1 (unsol 0) > Vendor Id: 0x80862809 > Subsystem Id: 0x80860101 > Revision Id: 0x100000 > No Modem Function Group found > Default PCM: > rates [0x0]: > bits [0x0]: > formats [0x0]: > Default Amp-In caps: N/A > Default Amp-Out caps: N/A > State of AFG node 0x01: > Power states: D0 D3 CLKSTOP EPSS > Power: setting=D0, actual=D0, Clock-stop-OK > GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 > Node 0x02 [Audio Output] wcaps 0x6611: 8-Channels Digital > Converter: stream=0, channel=0 > Digital: Enabled KAE > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1a]: 16 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x03 [Audio Output] wcaps 0x6611: 8-Channels Digital > Converter: stream=0, channel=0 > Digital: Enabled KAE > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1a]: 16 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x04 [Audio Output] wcaps 0x6611: 8-Channels Digital > Converter: stream=0, channel=0 > Digital: Enabled KAE > Digital category: 0x0 > IEC Coding Type: 0x0 > PCM: > rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 > bits [0x1a]: 16 24 32 > formats [0x5]: PCM AC3 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Control: name="IEC958 Playback Con Mask", index=0, device=0 > Control: name="IEC958 Playback Pro Mask", index=0, device=0 > Control: name="IEC958 Playback Default", index=0, device=0 > Control: name="IEC958 Playback Switch", index=0, device=0 > Control: name="ELD", index=0, device=3 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0b000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Devices: 0 > Connection: 0 > In-driver Connection: 3 > 0x02 0x03 0x04 > Node 0x06 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Control: name="IEC958 Playback Con Mask", index=1, device=0 > Control: name="IEC958 Playback Pro Mask", index=1, device=0 > Control: name="IEC958 Playback Default", index=1, device=0 > Control: name="IEC958 Playback Switch", index=1, device=0 > Control: name="ELD", index=0, device=7 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0b000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Devices: 0 > Connection: 0 > In-driver Connection: 3 > 0x02 0x03 0x04 > Node 0x07 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP > Control: name="IEC958 Playback Con Mask", index=2, device=0 > Control: name="IEC958 Playback Pro Mask", index=2, device=0 > Control: name="IEC958 Playback Default", index=2, device=0 > Control: name="IEC958 Playback Switch", index=2, device=0 > Control: name="ELD", index=0, device=8 > Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 > Amp-Out vals: [0x00 0x00] > Pincap 0x0b000094: OUT Detect HBR HDMI DP > Pin Default 0x18560010: [Jack] Digital Out at Int HDMI > Conn = Digital, Color = Unknown > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Power states: D0 D3 EPSS > Power: setting=D0, actual=D0 > Devices: 0 > Connection: 0 > In-driver Connection: 3 > 0x02 0x03 0x04 > Node 0x08 [Vendor Defined Widget] wcaps 0xf00000: Mono > --endcollapse-- > > > !!ALSA Device nodes > !!----------------- > > crw-rw----+ 1 root audio 116, 2 Feb 24 21:02 /dev/snd/controlC0 > crw-rw----+ 1 root audio 116, 8 Feb 24 21:02 /dev/snd/hwC0D0 > crw-rw----+ 1 root audio 116, 9 Feb 24 21:02 /dev/snd/hwC0D2 > crw-rw----+ 1 root audio 116, 4 Feb 24 21:02 /dev/snd/pcmC0D0c > crw-rw----+ 1 root audio 116, 3 Feb 24 21:02 /dev/snd/pcmC0D0p > crw-rw----+ 1 root audio 116, 5 Feb 24 21:02 /dev/snd/pcmC0D3p > crw-rw----+ 1 root audio 116, 6 Feb 24 21:02 /dev/snd/pcmC0D7p > crw-rw----+ 1 root audio 116, 7 Feb 24 21:02 /dev/snd/pcmC0D8p > crw-rw----+ 1 root audio 116, 1 Feb 24 21:02 /dev/snd/seq > crw-rw----+ 1 root audio 116, 33 Feb 24 21:02 /dev/snd/timer > > /dev/snd/by-path: > total 0 > drwxr-xr-x 2 root root 60 Feb 24 21:02 . > drwxr-xr-x 3 root root 260 Feb 24 21:02 .. > lrwxrwxrwx 1 root root 12 Feb 24 21:02 pci-0000:00:1f.3 -> ../controlC0 > > > !!Aplay/Arecord output > !!-------------------- > > APLAY > > **** List of PLAYBACK Hardware Devices **** > card 0: PCH [HDA Intel PCH], device 0: ALC233 Analog [ALC233 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > ARECORD > > **** List of CAPTURE Hardware Devices **** > card 0: PCH [HDA Intel PCH], device 0: ALC233 Analog [ALC233 Analog] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > !!Amixer output > !!------------- > > !!-------Mixer controls for card 0 [PCH] > > Card hw:0 'PCH'/'HDA Intel PCH at 0x94320000 irq 132' > Mixer name : 'Realtek ALC233' > Components : 'HDA:10ec0235,17aa3829,00100002 HDA:80862809,80860101,00100000' > Controls : 44 > Simple ctrls : 14 > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > Playback channels: Mono > Limits: Playback 0 - 87 > Mono: Playback 80 [92%] [-5.25dB] [on] > Simple mixer control 'Headphone',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 87 > Mono: > Front Left: Playback 0 [0%] [-65.25dB] [off] > Front Right: Playback 0 [0%] [-65.25dB] [off] > Simple mixer control 'Speaker',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 87 > Mono: > Front Left: Playback 87 [100%] [0.00dB] [on] > Front Right: Playback 87 [100%] [0.00dB] [on] > Simple mixer control 'PCM',0 > Capabilities: pvolume > Playback channels: Front Left - Front Right > Limits: Playback 0 - 255 > Mono: > Front Left: Playback 255 [100%] [0.00dB] > Front Right: Playback 255 [100%] [0.00dB] > Simple mixer control 'Mic',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 0 [0%] [-34.50dB] [off] > Front Right: Playback 0 [0%] [-34.50dB] [off] > Simple mixer control 'Mic Boost',0 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'IEC958',0 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [off] > Simple mixer control 'IEC958',1 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [on] > Simple mixer control 'IEC958',2 > Capabilities: pswitch pswitch-joined > Playback channels: Mono > Mono: Playback [on] > Simple mixer control 'Beep',0 > Capabilities: pvolume pswitch > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 0 [0%] [-34.50dB] [off] > Front Right: Playback 0 [0%] [-34.50dB] [off] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch > Capture channels: Front Left - Front Right > Limits: Capture 0 - 63 > Front Left: Capture 48 [76%] [18.75dB] [on] > Front Right: Capture 48 [76%] [18.75dB] [on] > Simple mixer control 'Auto-Mute Mode',0 > Capabilities: enum > Items: 'Disabled' 'Enabled' > Item0: 'Enabled' > Simple mixer control 'Internal Mic Boost',0 > Capabilities: volume > Playback channels: Front Left - Front Right > Capture channels: Front Left - Front Right > Limits: 0 - 3 > Front Left: 0 [0%] [0.00dB] > Front Right: 0 [0%] [0.00dB] > Simple mixer control 'Loopback Mixing',0 > Capabilities: enum > Items: 'Disabled' 'Enabled' > Item0: 'Enabled' > > > !!Alsactl output > !!-------------- > > --startcollapse-- > state.PCH { > control.1 { > iface MIXER > name 'Headphone Playback Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 87' > dbmin -6525 > dbmax 0 > dbvalue.0 -6525 > dbvalue.1 -6525 > } > } > control.2 { > iface MIXER > name 'Headphone Playback Switch' > value.0 false > value.1 false > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.3 { > iface MIXER > name 'Speaker Playback Volume' > value.0 87 > value.1 87 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 87' > dbmin -6525 > dbmax 0 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.4 { > iface MIXER > name 'Speaker Playback Switch' > value.0 true > value.1 true > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.5 { > iface MIXER > name 'Loopback Mixing' > value Enabled > comment { > access 'read write' > type ENUMERATED > count 1 > item.0 Disabled > item.1 Enabled > } > } > control.6 { > iface MIXER > name 'Mic Playback Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 31' > dbmin -3450 > dbmax 1200 > dbvalue.0 -3450 > dbvalue.1 -3450 > } > } > control.7 { > iface MIXER > name 'Mic Playback Switch' > value.0 false > value.1 false > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.8 { > iface MIXER > name 'Auto-Mute Mode' > value Enabled > comment { > access 'read write' > type ENUMERATED > count 1 > item.0 Disabled > item.1 Enabled > } > } > control.9 { > iface MIXER > name 'Capture Volume' > value.0 48 > value.1 48 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 63' > dbmin -1725 > dbmax 3000 > dbvalue.0 1875 > dbvalue.1 1875 > } > } > control.10 { > iface MIXER > name 'Capture Switch' > value.0 true > value.1 true > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.11 { > iface MIXER > name 'Mic Boost Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.12 { > iface MIXER > name 'Internal Mic Boost Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 3' > dbmin 0 > dbmax 3000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.13 { > iface MIXER > name 'Master Playback Volume' > value 80 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 87' > dbmin -6525 > dbmax 0 > dbvalue.0 -525 > } > } > control.14 { > iface MIXER > name 'Master Playback Switch' > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.15 { > iface CARD > name 'Mic Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.16 { > iface CARD > name 'Internal Mic Phantom Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.17 { > iface CARD > name 'Headphone Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.18 { > iface CARD > name 'Speaker Phantom Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.19 { > iface MIXER > name 'Beep Playback Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 31' > dbmin -3450 > dbmax 1200 > dbvalue.0 -3450 > dbvalue.1 -3450 > } > } > control.20 { > iface MIXER > name 'Beep Playback Switch' > value.0 false > value.1 false > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.21 { > iface PCM > name 'Playback Channel Map' > value.0 0 > value.1 0 > comment { > access read > type INTEGER > count 2 > range '0 - 36' > } > } > control.22 { > iface PCM > name 'Capture Channel Map' > value.0 0 > value.1 0 > comment { > access read > type INTEGER > count 2 > range '0 - 36' > } > } > control.23 { > iface CARD > name 'HDMI/DP,pcm=3 Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.24 { > iface MIXER > name 'IEC958 Playback Con Mask' > value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.25 { > iface MIXER > name 'IEC958 Playback Pro Mask' > value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.26 { > iface MIXER > name 'IEC958 Playback Default' > value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access 'read write' > type IEC958 > count 1 > } > } > control.27 { > iface MIXER > name 'IEC958 Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.28 { > iface PCM > device 3 > name ELD > value '' > comment { > access 'read volatile' > type BYTES > count 0 > } > } > control.29 { > iface CARD > name 'HDMI/DP,pcm=7 Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.30 { > iface MIXER > name 'IEC958 Playback Con Mask' > index 1 > value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.31 { > iface MIXER > name 'IEC958 Playback Pro Mask' > index 1 > value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.32 { > iface MIXER > name 'IEC958 Playback Default' > index 1 > value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access 'read write' > type IEC958 > count 1 > } > } > control.33 { > iface MIXER > name 'IEC958 Playback Switch' > index 1 > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.34 { > iface PCM > device 7 > name ELD > value '' > comment { > access 'read volatile' > type BYTES > count 0 > } > } > control.35 { > iface CARD > name 'HDMI/DP,pcm=8 Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.36 { > iface MIXER > name 'IEC958 Playback Con Mask' > index 2 > value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.37 { > iface MIXER > name 'IEC958 Playback Pro Mask' > index 2 > value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access read > type IEC958 > count 1 > } > } > control.38 { > iface MIXER > name 'IEC958 Playback Default' > index 2 > value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' > comment { > access 'read write' > type IEC958 > count 1 > } > } > control.39 { > iface MIXER > name 'IEC958 Playback Switch' > index 2 > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.40 { > iface PCM > device 8 > name ELD > value '' > comment { > access 'read volatile' > type BYTES > count 0 > } > } > control.41 { > iface PCM > device 3 > name 'Playback Channel Map' > value.0 0 > value.1 0 > value.2 0 > value.3 0 > value.4 0 > value.5 0 > value.6 0 > value.7 0 > comment { > access 'read write' > type INTEGER > count 8 > range '0 - 36' > } > } > control.42 { > iface PCM > device 7 > name 'Playback Channel Map' > value.0 0 > value.1 0 > value.2 0 > value.3 0 > value.4 0 > value.5 0 > value.6 0 > value.7 0 > comment { > access 'read write' > type INTEGER > count 8 > range '0 - 36' > } > } > control.43 { > iface PCM > device 8 > name 'Playback Channel Map' > value.0 0 > value.1 0 > value.2 0 > value.3 0 > value.4 0 > value.5 0 > value.6 0 > value.7 0 > comment { > access 'read write' > type INTEGER > count 8 > range '0 - 36' > } > } > control.44 { > iface MIXER > name 'PCM Playback Volume' > value.0 255 > value.1 255 > comment { > access 'read write user' > type INTEGER > count 2 > range '0 - 255' > tlv '0000000100000008ffffec1400000014' > dbmin -5100 > dbmax 0 > dbvalue.0 0 > dbvalue.1 0 > } > } > } > --endcollapse-- > > > !!All Loaded Modules > !!------------------ > > Module > drbg > ansi_cprng > ctr > ccm > cmac > rfcomm > bnep > bbswitch > arc4 > binfmt_misc > nls_iso8859_1 > snd_hda_codec_hdmi > snd_hda_codec_realtek > snd_hda_codec_generic > intel_rapl > x86_pkg_temp_thermal > intel_powerclamp > coretemp > iwlmvm > kvm_intel > mac80211 > kvm > irqbypass > crct10dif_pclmul > snd_hda_intel > crc32_pclmul > uvcvideo > snd_hda_codec > ghash_clmulni_intel > videobuf2_vmalloc > videobuf2_memops > snd_hda_core > aesni_intel > videobuf2_v4l2 > snd_hwdep > aes_x86_64 > videobuf2_core > lrw > iwlwifi > gf128mul > glue_helper > ablk_helper > snd_pcm > videodev > cryptd > snd_seq_midi > media > snd_seq_midi_event > cfg80211 > input_leds > snd_rawmidi > joydev > snd_seq > serio_raw > snd_seq_device > snd_timer > btusb > btrtl > snd > soundcore > mei_me > shpchp > mei > ideapad_laptop > sparse_keymap > hci_uart > btbcm > btqca > btintel > bluetooth > intel_lpss_acpi > wmi > intel_lpss > mac_hid > tpm_crb > acpi_pad > ip6t_REJECT > nf_reject_ipv6 > nf_log_ipv6 > xt_hl > ip6t_rt > nf_conntrack_ipv6 > nf_defrag_ipv6 > ipt_REJECT > nf_reject_ipv4 > nf_log_ipv4 > nf_log_common > xt_LOG > xt_limit > xt_tcpudp > xt_addrtype > nf_conntrack_ipv4 > nf_defrag_ipv4 > xt_conntrack > ip6table_filter > ip6_tables > nf_conntrack_netbios_ns > nf_conntrack_broadcast > nf_nat_ftp > nf_nat > nf_conntrack_ftp > nf_conntrack > iptable_filter > parport_pc > ip_tables > ppdev > x_tables > lp > parport > autofs4 > hid_generic > usbhid > i915 > i2c_algo_bit > drm_kms_helper > psmouse > syscopyarea > sysfillrect > sysimgblt > fb_sys_fops > r8169 > drm > sdhci_pci > ahci > mii > sdhci > libahci > i2c_hid > pinctrl_sunrisepoint > video > hid > pinctrl_intel > fjes > > > !!Sysfs Files > !!----------- > > /sys/class/sound/hwC0D0/init_pin_configs: > 0x12 0x90a60140 > 0x14 0x90170110 > 0x17 0x40000000 > 0x18 0x411111f0 > 0x19 0x04a11030 > 0x1a 0x411111f0 > 0x1b 0x411111f0 > 0x1d 0x40f79a2d > 0x1e 0x411111f0 > 0x21 0x04211020 > > /sys/class/sound/hwC0D0/driver_pin_configs: > > /sys/class/sound/hwC0D0/user_pin_configs: > > /sys/class/sound/hwC0D0/init_verbs: > > /sys/class/sound/hwC0D0/hints: > > /sys/class/sound/hwC0D2/init_pin_configs: > 0x05 0x18560010 > 0x06 0x18560010 > 0x07 0x18560010 > > /sys/class/sound/hwC0D2/driver_pin_configs: > > /sys/class/sound/hwC0D2/user_pin_configs: > > /sys/class/sound/hwC0D2/init_verbs: > > /sys/class/sound/hwC0D2/hints: > > > !!ALSA/HDA dmesg > !!-------------- > > [ 3.118546] iwlwifi 0000:08:00.0: loaded firmware version 16.242414.0 op_mode iwlmvm > [ 3.119214] snd_hda_intel 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) > [ 3.119989] uvcvideo 1-6:1.0: Entity type for entity Realtek Extended Controls Unit was not initialized! > -- > [ 3.159964] Bluetooth: hci0: Failed to send firmware data (-38) > [ 3.161319] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC233: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker > [ 3.161321] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > [ 3.161323] snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) > [ 3.161324] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 > [ 3.161325] snd_hda_codec_realtek hdaudioC0D0: inputs: > [ 3.161326] snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 > [ 3.161327] snd_hda_codec_realtek hdaudioC0D0: Internal Mic=0x12 > [ 3.170162] nvidia_uvm: Loaded the UVM driver, major device number 245 > [ 3.172370] Adding 32529404k swap on /dev/sda3. Priority:-1 extents:1 across:32529404k SSFS > [ 3.181718] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1f.3/sound/card0/input12 > [ 3.181763] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1f.3/sound/card0/input13 > [ 3.181802] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input14 > [ 3.181846] input: HDA Intel PCH HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input15 > [ 3.181905] input: HDA Intel PCH HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:1f.3/sound/card0/input16 > [ 3.259924] audit: type=1400 audit(1456344123.829:2): apparmor="STATUS" operation="profile_load" name="/usr/lib/lightdm/lightdm-guest-session" pid=676 comm="apparmor_parser" > >

1 0

[alsa-devel] USB Sound card with unsafe characters
by Brian Jones 26 Feb '16

26 Feb '16

I have been trying to work on a fix for Bug 1498129 - https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1498129 on Ubuntu My USB soundcard is an Asus Xonar U7 Echelon Edition <https://www.asus.com/Sound-Cards/Xonar_U7_Echelon_Edition/> which identifies itself to alsa as "Xonar U7 Echelon Ed." The card is supported by alsa and works well in analog mode but I cannot get digital output via alsa or pulseaudio. It has iec958 as device 1 rather than device 0 and I can access digital output by referring to directly to hw:x,1 but cannot access digital output via alsa. The fix I believe is to add "Xonar U7 Echelon Ed." 1 to the USB-Audio.pcm.iec958_device section of USB-Audio.conf. This does not work. As I suspected, and now have confirmed by Raymond on the bug report, this is because the alsa name for this card contains the period character '.' Is there a workaround for this or do I need to await an alsa code change?

1 0

[alsa-devel] [RFC: PATCH] ALSA: hda - clear IRQ statu twice on some Intel platforms
by libin.yang＠linux.intel.com 26 Feb '16

26 Feb '16

From: Libin Yang <libin.yang(a)linux.intel.com> On some Intel platforms, we found the interrupt issue in the below scenario: 1. driver is in irq handler 2. there is another interrupt from HW after interrupt status is cleared and before exiting from interrupt handler 3. exit from the current irq handling After exiting the irq handler, it should raise another interrupt for driver to handle the new interrupt. But actually, it failed to raise the interrupt and driver will never have chance to clear the interrupt status. The patch clears the interrupt status again just before exiting for interrupt handler. This can reduce the contest dramatically. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/pci/hda/hda_controller.c | 3 +++ sound/pci/hda/hda_controller.h | 3 +++ sound/pci/hda/hda_intel.c | 40 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 46 insertions(+) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 37cf9ce..2ca95c7 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -963,6 +963,9 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } + if (chip->post_irq) + chip->post_irq(chip); + spin_unlock(&bus->reg_lock); return IRQ_HANDLED; diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index ec63bbf..ce59997 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -146,6 +146,9 @@ struct azx { const struct firmware *fw; #endif + /* callback at the end of interrupt handler */ + void (*post_irq)(struct azx *); + /* flags */ int bdl_pos_adj; int poll_count; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4045dca..6a47d88 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1663,6 +1663,43 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, return 0; } +/* HSW, BDW, SKL and BXT need do post_irq() */ +#define INTEL_IRQ_POST(chip) \ + (((chip)->pci->vendor == PCI_VENDOR_ID_INTEL) && \ + (((chip)->pci->device == 0x0a0c) || \ + ((chip)->pci->device == 0x0c0c) || \ + ((chip)->pci->device == 0x0d0c) || \ + ((chip)->pci->device == 0x160c) || \ + ((chip)->pci->device == 0xa170) || \ + ((chip)->pci->device == 0x9d70) || \ + ((chip)->pci->device == 0x5a98))) + + +/* on some intel platforms, if there occurs an interrupt + * when irq is being handled, interrupt signal will not be raised + * after the irq handler returns. And the interrupt status may never + * be cleared. + * So let's clear all the interrupt status before return from irq handler. + * This can help to reduce the contest between the irq handler and the signal. + */ +static void intel_post_irq(struct azx *chip) +{ + struct hdac_bus *bus = azx_bus(chip); + struct hdac_stream *azx_dev; + + list_for_each_entry(azx_dev, &bus->stream_list, list) + snd_hdac_stream_writeb(azx_dev, SD_STS, SD_INT_MASK); + + /* clear STATESTS */ + snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK); + + /* clear rirb status */ + snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK); + + /* clear int status */ + snd_hdac_chip_writel(bus, INTSTS, AZX_INT_CTRL_EN | AZX_INT_ALL_STREAM); +} + static int azx_first_init(struct azx *chip) { int dev = chip->dev_index; @@ -1717,6 +1754,9 @@ static int azx_first_init(struct azx *chip) if (chip->pci->vendor == PCI_VENDOR_ID_AMD) dma_bits = 40; + if (INTEL_IRQ_POST(chip)) + chip->post_irq = intel_post_irq; + /* disable SB600 64bit support for safety */ if (chip->pci->vendor == PCI_VENDOR_ID_ATI) { struct pci_dev *p_smbus; -- 1.9.1

3 7

[alsa-devel] snd-hda-intel runtime PM fail after module reload
by Ville Syrjälä 26 Feb '16

26 Feb '16

Hi, My investigation into some sporadic i915 runtime PM failures seem to point the finger at snd-hda-intel. I just tried to play around unloding and reloading snd-hda-intel and sometimes I get snd-hda-intel loaded with runtime PM supposedly enabled, but actually the device won't runtime suspend. At which point frobbing with power/control is enough to kick it back into submission. # for i in `seq 1 3`; do sleep 1 ; grep . /sys/bus/pci/devices/0000\:00\:03.0/power/* ; done /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:117045 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:222657 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:117045 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:223666 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:117045 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:224674 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 # rmmod snd-hda-intel # modprobe snd-hda-intel # for i in `seq 1 3`; do sleep 1 ; grep . /sys/bus/pci/devices/0000\:00\:03.0/power/* ; done /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:121454 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:active /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:228560 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:122463 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:active /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:228560 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:123472 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:active /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:228560 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 # echo on > /sys/bus/pci/devices/0000:00:03.0/power/control # echo auto > /sys/bus/pci/devices/0000:00:03.0/power/control # for i in `seq 1 3`; do sleep 1 ; grep . /sys/bus/pci/devices/0000\:00\:03.0/power/* ; done /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:141459 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:231699 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:141459 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:232706 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 /sys/bus/pci/devices/0000:00:03.0/power/async:enabled grep: /sys/bus/pci/devices/0000:00:03.0/power/autosuspend_delay_ms: Input/output error /sys/bus/pci/devices/0000:00:03.0/power/control:auto /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_kids:0 /sys/bus/pci/devices/0000:00:03.0/power/runtime_active_time:141459 /sys/bus/pci/devices/0000:00:03.0/power/runtime_enabled:enabled /sys/bus/pci/devices/0000:00:03.0/power/runtime_status:suspended /sys/bus/pci/devices/0000:00:03.0/power/runtime_suspended_time:233713 /sys/bus/pci/devices/0000:00:03.0/power/runtime_usage:0 -- Ville Syrjälä Intel OTC

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[alsa-devel] [PATCH 0/4 RESEND] Add I2S/ADV7511 audio support for ARC AXS10x boards
by Jose Abreu 26 Feb '16

26 Feb '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by ADV7511 transmitter. This patch set adds audio for the ADV7511 transmitter and I2S audio for the AXS10x platform. Jose Abreu (4): [adv7511] Add audio support [adv7511] Update defconfigs so that audio is enabled [dwc] Add I2S HDMI audio support using custom platform driver [dwc] Update defconfigs so that I2S is enabled arch/arc/boot/dts/axs10x_mb.dtsi | 20 +- arch/arc/configs/axs101_defconfig | 4 + arch/arc/configs/axs103_defconfig | 6 + arch/arc/configs/axs103_smp_defconfig | 6 + drivers/gpu/drm/i2c/Kconfig | 8 + drivers/gpu/drm/i2c/Makefile | 2 + drivers/gpu/drm/i2c/adv7511.c | 1024 --------------------------------- drivers/gpu/drm/i2c/adv7511.h | 41 ++ drivers/gpu/drm/i2c/adv7511_audio.c | 310 ++++++++++ drivers/gpu/drm/i2c/adv7511_core.c | 1005 ++++++++++++++++++++++++++++++++ include/sound/soc-dai.h | 1 + sound/soc/dwc/Kconfig | 1 + sound/soc/dwc/Makefile | 4 +- sound/soc/dwc/designware_i2s.c | 139 ++++- sound/soc/dwc/designware_pcm.c | 264 +++++++++ sound/soc/dwc/designware_pcm.h | 21 + 16 files changed, 1813 insertions(+), 1043 deletions(-) delete mode 100644 drivers/gpu/drm/i2c/adv7511.c create mode 100644 drivers/gpu/drm/i2c/adv7511_audio.c create mode 100644 drivers/gpu/drm/i2c/adv7511_core.c create mode 100644 sound/soc/dwc/designware_pcm.c create mode 100644 sound/soc/dwc/designware_pcm.h -- 1.9.1

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[alsa-devel] [PATCH 0/15 v2][RFC] cleanup regmap write functions
by Kuninori Morimoto 26 Feb '16

26 Feb '16

Hi Mark Current regmap has many similar functions regmap_update_bits() regmap_update_bits_async() regmap_update_bits_check() regmap_update_bits_check_async() But difference is very few. And I would like to have _force_ feature on it too. So, these patches add new regmap_update_bits_base() which has _check, _async, and _force option. Above functions are now defined as macro. I used [RFC], because regmap has big effects. 1), 6), 9) adds new regmap_xxx_update_bits_base() functions and current each functions are merged into it by macro. Please review carefully especially 7), 10). I think using ~0 mask is correct for regmap_field_write() <-> regmap_field_update_bits_base() 1) - 5) for regmap_xxx() 6) - 8) for regmap_field_xxx() 9) - 11) for regmap_fields_xxx() 12) - 13) adds _force_ macro 14) remove unused regmap_write_bits() which was one of _force_ function 15) rsnd driver uses _force_ function for register access BTW, I noticed #if - #else - #endif on ${LINUX}/include/linux/regmap.h are strange. Many functions/struct/macro are defined under #ifdef CONFIG_REGMAP, but few are defined under #else. It can be trouble ? Do we really need this #if ? Kuninori Morimoto (15): 1) regmap: add regmap_update_bits_base() 2) regmap: merge regmap_update_bits() into macro 3) regmap: merge regmap_update_bits_async() into macro 4) regmap: merge regmap_update_bits_check() into macro 5) regmap: merge regmap_update_bits_check_async() into macro 6) regmap: add regmap_field_update_bits_base() 7) regmap: merge regmap_field_write() into macro 8) regmap: merge regmap_field_update_bits() into macro 9) regmap: add regmap_fields_update_bits_base() 10) regmap: merge regmap_fields_write() into macro 11) regmap: merge regmap_fields_update_bits() into macro 12) regmap: add regmap_field_force_xxx() macros 13) regmap: add regmap_fields_force_xxx() macros 14) regmpa: remove regmap_write_bits() 15) ASoC: rsnd: rsnd_write() / rsnd_bset() uses regmap _force_ function drivers/base/regmap/regmap.c | 217 ++++++++++++++++++++++++----------------------------------------------------------------------------------------------------------- include/linux/regmap.h | 94 ++++++++++++++++++++++++++++----------------------------- sound/soc/sh/rcar/gen.c | 21 ++----------- sound/soc/sh/rcar/rsnd.h | 2 -- 4 files changed, 87 insertions(+), 247 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/6][RFC] cleanup regmap write functions
by Kuninori Morimoto 26 Feb '16

26 Feb '16

Hi Mark Current regmap has many similar functions regmap_update_bits() regmap_update_bits_async() regmap_update_bits_check() regmap_update_bits_check_async() But difference is very few. And I would like to have _force_ feature on it. So, these patches add new regmap_raw_update_bits() which has _check, _async, _force option. Above functions are now defined as macro. 4), 5) adds _force_ feature. We can easily add _check, _async. I used [RFC], because regmap has many effect. BTW, I noticed #if - #else - #endif on ${LINUX}/include/linux/regmap.h are strange. Many functions/struct/macro are defined under #ifdef CONFIG_REGMAP, but few are defined under #else. It can be trouble ? Do we really need this #if ? Kuninori Morimoto (7): 1) regmap: add regmap_raw_update_bits() and merge all regmap_update_bits_xxx() 2) regmap: regmap_field_xxx() uses regmap_raw_update_bits() 3) regmap: regmap_fields_xxx() uses regmap_raw_update_bits() 4) regmap: add regmap_field_force_xxx() functions 5) regmap: add regmap_fields_force_xxx() functions 6) regmpa: remove regmap_write_bits() 7) ASoC: rsnd: rsnd_write() / rsnd_bset() uses regmap _force_ function drivers/base/regmap/regmap.c | 191 +++++++++++++++++++++++++++++------------------------------------------------------------------------------------------------------ include/linux/regmap.h | 102 ++++++++++++++++++++++++++++++---------------------------------------- sound/soc/sh/rcar/gen.c | 21 ++------------- sound/soc/sh/rcar/rsnd.h | 2 -- 4 files changed, 88 insertions(+), 228 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: mediatek: Add I03/I04 widgets and corresponding routes
by PC Liao 26 Feb '16

26 Feb '16

From: Koro Chen <koro.chen(a)mediatek.com> Add these widgets to allow another path from I2S input. Change-Id: Ib6804bbbde388830d8359245fe13bbb0f2bb36ba Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 689c51f..5de878f 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -40,6 +40,7 @@ #define AFE_CONN1 0x0024 #define AFE_CONN2 0x0028 +#define AFE_CONN3 0x002c #define AFE_CONN7 0x0460 #define AFE_CONN8 0x0464 #define AFE_HDMI_CONN0 0x0390 @@ -63,6 +64,7 @@ #define AFE_HDMI_OUT_CUR 0x0378 #define AFE_HDMI_OUT_END 0x037c +#define AFE_ADDA_TOP_CON0 0x0120 #define AFE_ADDA2_TOP_CON0 0x0600 #define AFE_HDMI_OUT_CON0 0x0370 @@ -259,6 +261,7 @@ static int mtk_afe_set_i2s(struct mtk_afe *afe, unsigned int rate) return -EINVAL; /* from external ADC */ + regmap_update_bits(afe->regmap, AFE_ADDA_TOP_CON0, 0x1, 0x1); regmap_update_bits(afe->regmap, AFE_ADDA2_TOP_CON0, 0x1, 0x1); /* set input */ @@ -903,15 +906,19 @@ static const struct snd_kcontrol_new mtk_afe_o04_mix[] = { }; static const struct snd_kcontrol_new mtk_afe_o09_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I03 Switch", AFE_CONN3, 0, 1, 0), SOC_DAPM_SINGLE_AUTODISABLE("I17 Switch", AFE_CONN7, 30, 1, 0), }; static const struct snd_kcontrol_new mtk_afe_o10_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I04 Switch", AFE_CONN3, 3, 1, 0), SOC_DAPM_SINGLE_AUTODISABLE("I18 Switch", AFE_CONN8, 0, 1, 0), }; static const struct snd_soc_dapm_widget mtk_afe_pcm_widgets[] = { /* inter-connections */ + SND_SOC_DAPM_MIXER("I03", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I04", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("I05", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("I06", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("I17", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -934,12 +941,16 @@ static const struct snd_soc_dapm_route mtk_afe_pcm_routes[] = { {"I2S Playback", NULL, "O04"}, {"VUL", NULL, "O09"}, {"VUL", NULL, "O10"}, + {"I03", NULL, "I2S Capture"}, + {"I04", NULL, "I2S Capture"}, {"I17", NULL, "I2S Capture"}, {"I18", NULL, "I2S Capture"}, { "O03", "I05 Switch", "I05" }, { "O04", "I06 Switch", "I06" }, { "O09", "I17 Switch", "I17" }, + { "O09", "I03 Switch", "I03" }, { "O10", "I18 Switch", "I18" }, + { "O10", "I04 Switch", "I04" }, }; static const struct snd_soc_dapm_route mtk_afe_hdmi_routes[] = { -- 1.7.9.5

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[alsa-devel] [patch 2/2] ASoC: max9867: silence and array overflow warning
by Dan Carpenter 26 Feb '16

26 Feb '16

Smatch complains that we might reach the end of this loop without finding what we're looking for leading to a buffer overflow. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 4fb6fc7..9fb03c6 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -156,6 +156,8 @@ static inline int get_ni_value(int mclk, int rate) if (ni_div[i].mclk >= mclk) break; } + if (i == ARRAY_SIZE(ni_div)) + return -EINVAL; switch (rate) { case 8000:

3 2

[alsa-devel] [PATCH] sound: hdmi: avoid dereferencing uninitialized 'jack' pointer
by Arnd Bergmann 26 Feb '16

26 Feb '16

When CONFIG_SND_JACK is disabled, the hdmi driver tries to create a new output jack structure and dereferences it even though the pointer is never assigned: sound/pci/hda/patch_hdmi.c: In function 'generic_hdmi_build_jack': sound/pci/hda/patch_hdmi.c:2534:30: error: 'jack' may be used uninitialized in this function [-Werror=maybe-uninitialized] sound/pci/hda/patch_hdmi.c:2526:19: note: 'jack' was declared here This uses a compile-time check to avoid calling a nonworking snd_jack_new() function, and sets an explicit NULL pointer instead. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 788d441a164c ("ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling") --- sound/pci/hda/patch_hdmi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f4443b5fbf6e..c0872ad80aa0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2526,6 +2526,11 @@ static int add_hdmi_jack_kctl(struct hda_codec *codec, struct snd_jack *jack; int err; + if (!IS_ENABLED(CONFIG_SND_JACK)) { + spec->pcm_rec[pcm_idx].jack = NULL; + return 0; + } + err = snd_jack_new(codec->card, name, SND_JACK_AVOUT, &jack, true, false); if (err < 0) -- 2.7.0

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[alsa-devel] [PATCH 1/2] ASoC: rt5645: improve headphone pop when system resumes from S3
by John Lin 26 Feb '16

26 Feb '16

When playing audio through headphone, headphone makes pop noise if system resumes from S3 to S0. We modify the sequence of writing register for avoiding pop sound. Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 93e8c90..d57119d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3029,13 +3029,18 @@ static int rt5645_set_bias_level(struct snd_soc_codec *codec, RT5645_PWR_BG | RT5645_PWR_VREF2, RT5645_PWR_VREF1 | RT5645_PWR_MB | RT5645_PWR_BG | RT5645_PWR_VREF2); + mdelay(10); snd_soc_update_bits(codec, RT5645_PWR_ANLG1, RT5645_PWR_FV1 | RT5645_PWR_FV2, RT5645_PWR_FV1 | RT5645_PWR_FV2); - if (rt5645->en_button_func && - snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) - queue_delayed_work(system_power_efficient_wq, - &rt5645->jack_detect_work, msecs_to_jiffies(0)); + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140); + msleep(40); + if (rt5645->en_button_func) + queue_delayed_work(system_power_efficient_wq, + &rt5645->jack_detect_work, + msecs_to_jiffies(0)); + } break; case SND_SOC_BIAS_OFF: -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: pxa: remove incorrect do_div() call
by Arnd Bergmann 26 Feb '16

26 Feb '16

The new optimized do_div implementation (now in asm-generic/next) exposes a glitch in the brownstone audio driver by producing a compile-time warning: sound/soc/pxa/brownstone.c: In function 'brownstone_wm8994_hw_params': sound/soc/pxa/brownstone.c:67:85: warning: comparison of distinct pointer types lacks a cast sound/soc/pxa/brownstone.c:67:10125: warning: right shift count >= width of type [-Wshift-count-overflow] sound/soc/pxa/brownstone.c:67:10254: warning: passing argument 1 of '__div64_32' from incompatible pointer type [-Wincompatible-pointer-types] The driver just divides two plain integer values, so it should not use do_div to start with, but has apparently done so ever since the code was first merged. This replaces do_div with a simple division operator. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- I think this one is harmless, so we only need it in 4.5 to avoid the warning, but not backported to earlier kernels. Found today on ARM allmodconfig diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 6147e86e9b0f..416ea646c3b1 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -63,8 +63,7 @@ static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, sysclk = params_rate(params) * 512; sspa_mclk = params_rate(params) * 64; } - sspa_div = freq_out; - do_div(sspa_div, sspa_mclk); + sspa_div = freq_out / sspa_mclk; snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);

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[alsa-devel] [PATCH] ASoC: mediatek: remove soft reset and add second I2S clock
by PC Liao 26 Feb '16

26 Feb '16

To improve I2S flow, this patch removes soft reset and adds second I2S clock to use. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index 689c51f..3456bfa 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -283,20 +283,13 @@ static void mtk_afe_set_i2s_enable(struct mtk_afe *afe, bool enable) regmap_read(afe->regmap, AFE_I2S_CON2, &val); if (!!(val & AFE_I2S_CON2_EN) == enable) - return; /* must skip soft reset */ - - /* I2S soft reset begin */ - regmap_update_bits(afe->regmap, AUDIO_TOP_CON1, 0x4, 0x4); + return; /* input */ regmap_update_bits(afe->regmap, AFE_I2S_CON2, 0x1, enable); /* output */ regmap_update_bits(afe->regmap, AFE_I2S_CON1, 0x1, enable); - - /* I2S soft reset end */ - udelay(1); - regmap_update_bits(afe->regmap, AUDIO_TOP_CON1, 0x4, 0); } static int mtk_afe_dais_enable_clks(struct mtk_afe *afe, @@ -365,6 +358,7 @@ static int mtk_afe_i2s_startup(struct snd_pcm_substream *substream, return 0; mtk_afe_dais_enable_clks(afe, afe->clocks[MTK_CLK_I2S1_M], NULL); + mtk_afe_dais_enable_clks(afe, afe->clocks[MTK_CLK_I2S2_M], NULL); regmap_update_bits(afe->regmap, AUDIO_TOP_CON0, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M, 0); return 0; @@ -384,6 +378,7 @@ static void mtk_afe_i2s_shutdown(struct snd_pcm_substream *substream, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M); mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S1_M], NULL); + mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S2_M], NULL); } static int mtk_afe_i2s_prepare(struct snd_pcm_substream *substream, @@ -397,6 +392,9 @@ static int mtk_afe_i2s_prepare(struct snd_pcm_substream *substream, mtk_afe_dais_set_clks(afe, afe->clocks[MTK_CLK_I2S1_M], runtime->rate * 256, NULL, 0); + mtk_afe_dais_set_clks(afe, + afe->clocks[MTK_CLK_I2S2_M], runtime->rate * 256, + NULL, 0); /* config I2S */ ret = mtk_afe_set_i2s(afe, substream->runtime->rate); if (ret) -- 1.7.9.5

2 1

[alsa-devel] [RESEND PATCH] ASoC: cs4271: add regulator consumer support
by pascal.huerst＠gmail.com 26 Feb '16

26 Feb '16

From: Pascal Huerst <pascal.huerst(a)gmail.com> The cs4271 has three power domains: vd, vl and va. Enable them all, as long as the codec is in use. While at it, factored out the reset code into its own function. Signed-off-by: Pascal Huerst <pascal.huerst(a)gmail.com> --- Documentation/devicetree/bindings/sound/cs4271.txt | 7 +++ sound/soc/codecs/cs4271.c | 69 +++++++++++++++++++--- 2 files changed, 68 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt index e2cd1d7..6e699ce 100644 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ b/Documentation/devicetree/bindings/sound/cs4271.txt @@ -33,12 +33,19 @@ Optional properties: Note that this is not needed in case the clocks are stable throughout the entire runtime of the codec. + - vd-supply: Digital power + - vl-supply: Logic power + - va-supply: Analog Power + Examples: codec_i2c: cs4271@10 { compatible = "cirrus,cs4271"; reg = <0x10>; reset-gpio = <&gpio 23 0>; + vd-supply = <&vdd_3v3_reg>; + vl-supply = <&vdd_3v3_reg>; + va-supply = <&vdd_3v3_reg>; }; codec_spi: cs4271@0 { diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index e770ee6..0c0010b 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -26,6 +26,7 @@ #include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> #include <sound/pcm.h> #include <sound/soc.h> #include <sound/tlv.h> @@ -157,6 +158,10 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) return reg == CS4271_CHIPID; } +static const char * const supply_names[] = { + "vd", "vl", "va" +}; + struct cs4271_private { unsigned int mclk; bool master; @@ -170,6 +175,7 @@ struct cs4271_private { int gpio_disable; /* enable soft reset workaround */ bool enable_soft_reset; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = { @@ -487,6 +493,20 @@ static struct snd_soc_dai_driver cs4271_dai = { .symmetric_rates = 1, }; +static int cs4271_reset(struct snd_soc_codec *codec) +{ + struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(cs4271->gpio_nreset)) { + gpio_set_value(cs4271->gpio_nreset, 0); + mdelay(1); + gpio_set_value(cs4271->gpio_nreset, 1); + mdelay(1); + } + + return 0; +} + #ifdef CONFIG_PM static int cs4271_soc_suspend(struct snd_soc_codec *codec) { @@ -499,6 +519,9 @@ static int cs4271_soc_suspend(struct snd_soc_codec *codec) if (ret < 0) return ret; + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; } @@ -507,6 +530,16 @@ static int cs4271_soc_resume(struct snd_soc_codec *codec) int ret; struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + + /* Do a proper reset after power up */ + cs4271_reset(codec); + /* Restore codec state */ ret = regcache_sync(cs4271->regmap); if (ret < 0) @@ -553,19 +586,24 @@ static int cs4271_codec_probe(struct snd_soc_codec *codec) } #endif + ret = regulator_bulk_enable(ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable regulators: %d\n", ret); + return ret; + } + if (cs4271plat) { amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_is_valid(cs4271->gpio_nreset)) { - /* Reset codec */ - gpio_direction_output(cs4271->gpio_nreset, 0); - mdelay(1); - gpio_set_value(cs4271->gpio_nreset, 1); - /* Give the codec time to wake up */ - mdelay(1); - } + /* Reset codec */ + cs4271_reset(codec); + + ret = regcache_sync(cs4271->regmap); + if (ret < 0) + return ret; ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, @@ -595,6 +633,9 @@ static int cs4271_codec_remove(struct snd_soc_codec *codec) /* Set codec to the reset state */ gpio_set_value(cs4271->gpio_nreset, 0); + regcache_mark_dirty(cs4271->regmap); + regulator_bulk_disable(ARRAY_SIZE(cs4271->supplies), cs4271->supplies); + return 0; }; @@ -617,6 +658,7 @@ static int cs4271_common_probe(struct device *dev, { struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; + int i, ret; cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); if (!cs4271) @@ -638,6 +680,17 @@ static int cs4271_common_probe(struct device *dev, return ret; } + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + cs4271->supplies[i].supply = supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(cs4271->supplies), + cs4271->supplies); + + if (ret < 0) { + dev_err(dev, "Failed to get regulators: %d\n", ret); + return ret; + } + *c = cs4271; return 0; } -- 2.4.3

2 1

[alsa-devel] [PATCH 0/1] ALSA: hda - Use acpi_dev_present()
by Lukas Wunner 25 Feb '16

25 Feb '16

Hi Takashi, the acpi_dev_present() API has now landed in Linus' tree. Thus, after Linus' tree gets merged back into yours, it would be possible to use the API in the Thinkpad hda drivers as per the following patch. I've also pushed it to GitHub in case anyone prefers perusing it in a browser: https://github.com/l1k/linux/commit/a1473d726b57eaf97c4de8812c5967603068e261 An ack for this patch was kindly provided by Hui Wang with: Message-ID: <5653C291.9090607(a)canonical.com> http://mailman.alsa-project.org/pipermail/alsa-devel/2015-November/100962.h… Thanks, Lukas Lukas Wunner (1): ALSA: hda - Use acpi_dev_present() sound/pci/hda/thinkpad_helper.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) -- 2.1.0

3 8

Re: [alsa-devel] [PATCH] ASoC: nau8825: Modify power management and add interrupt wakeup
by John Hsu 25 Feb '16

25 Feb '16

Hi Mark, This patch has cancel and only submit enhance codec suspend and resume patch. Please review the following link for the patch. Many thanks. http://mailman.alsa-project.org/pipermail/alsa-devel/2016-January/102924.ht… [alsa-devel] [PATCH] ASoC: nau8825: fix interrupt fails and unstable after resume On 1/8/2016 9:14 PM, Mark Brown wrote: > On Fri, Jan 08, 2016 at 03:11:22PM +0800, John Hsu wrote: > >> 1. Enhance codec suspend and resume sequence 2. Add interrupt wakeup >> function >> > > If this is two or more separate changes you should be sending two or more separate patches. You also need a better changelog which describes what the patch is supposed to do, in what way is suspend and resume enhanced for example? > > >> +/** >> + * nau8825_init_wakeup - set wakeup capability for codec >> + * >> + * @codec: codec device component >> + * >> + * After this function done, codec can support system wakeup by button. >> + */ >> +int nau8825_init_wakeup(struct snd_soc_codec *codec) { >> + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); >> + >> + device_init_wakeup(nau8825->dev, true); >> + >> + return 0; >> +} >> +EXPORT_SYMBOL_GPL(nau8825_init_wakeup); >> > > You need a bit more explanation as to what's going on here. Why is this not something the driver just does or if this should be done elsewhere why do we need the wrapper function? > > >> +int nau8825_irq_wakeup(struct snd_soc_codec *codec, int on) { >> + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); >> > > Same thing here, this looks like an unclear interface. >

2 2

[alsa-devel] Headset doesn't work for HP EliteBook 755 G2
by Derick Eddington 25 Feb '16

25 Feb '16

Hi, first time here, following suggestion [1] to report non-working default configuration, which I found how to make work. I checked using the latest stable kernel 4.4.2 and my headset jack doesn't work by default but my fix does work, on Scientific Linux 7.2 using the 4.4.2 kernel from the ELRepo repository. I also checked using Ubuntu 16.04 alpha with its 4.4.0 kernel and the problem and fix are the same. My laptop is an HP EliteBook 755 G2 from mid 2015 [2] that has a combined headset (headphone + mic) jack that doesn't work by default. The speakers work fine by default and they mute when a headset is plugged in, but the headphones don't work. The snd-hda-intel module handles my device. The codec is Realtek ALC3228. I have the latest BIOS update from HP. I got it working by passing model=hp,headset-mic to snd-hda-intel. I found this fix in an alsa-user message about a different HP laptop with the same codec [3], but I had to modify the message's fix by only including the module option for model= and omitting the other options and aliases that seem to be specific to Arch Linux. I tried using only "hp" or only "headset-mic" as the value for model= but neither worked. Attached are my outputs of alsa-info.sh from before and after I did model=hp,headset-mic. Also, there's another problem after enabling model=hp,headset-mic, which I found how to fix: There is a very annoying squeaky background noise in the headphones that's independent of adjusting the volume level but it does stop when the headphones are muted. I tried a different headset and it's the same. I found how to suppress this by using the white_noise_fix.py script from [4] as described at [5,6], which apparently does some ioctl verbs to adjust the amp gain. I thought this might be something that should be fixed/adjusted in the kernel module(s) that handle this device, so that the white_noise_fix.py isn't needed. If so, maybe it's worth mentioning that when my laptop resumes from sleep, the background noise comes back, so I have to run white_noise_fix.py again. I'm willing to help investigate my hardware more or try other configurations or distros if requested. If it's worth mentioning, I found someone else with the same laptop who has the same non-working headset problem, see [7]. I'd like to help get this fixed in Ubuntu 16.04 before its release. If the proper fix is in the kernel ALSA, I'm willing to go through Ubuntu's process for asking them to incorporate it in 16.04. Apologies if I've misdirected this report. Thanks for your work on ALSA! -- Derick [1] linux-doc/sound/alsa/ALSA-Configuration.txt [2] Product number L9H51UT#ABA [3] https://sourceforge.net/p/alsa/mailman/message/33087916/ [4] http://xps13-9333.appspot.com/root/usr/local/bin/white_noise_fix.py [5] http://xps13-9333.appspot.com/#background_noise [6] http://askubuntu.com/questions/524344/noise-from-headphones [7] http://askubuntu.com/questions/714920/audio-card-alc3228-issue

2 3

[alsa-devel] [PATCH] hda-emu: add acpi_dev_present
by David Henningsson 25 Feb '16

25 Feb '16

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/acpi/acpi.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/acpi/acpi.h b/include/acpi/acpi.h index d7d6cb9..9fd9c86 100644 --- a/include/acpi/acpi.h +++ b/include/acpi/acpi.h @@ -33,4 +33,6 @@ acpi_get_devices(const char *HID, return user_function(NULL, 0, context, NULL); } +static inline bool acpi_dev_present(const char *hid) { return true; } + #endif /* __ACPI_H__ */ -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ALSA: hda - Fixup speaker pass-through control for nid 0x14 on ALC225
by David Henningsson 25 Feb '16

25 Feb '16

On one of the machines we enable, we found that the actual speaker volume did not always correspond to the volume set in alsamixer. This patch fixes that problem. This patch was orginally written by Kailang @ Realtek, I've rebased it to fit sound git master. Cc: stable(a)vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1549660 Co-Authored-By: Kailang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 72fa58d..7fded69 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3801,6 +3801,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, static void alc_headset_mode_default(struct hda_codec *codec) { + static struct coef_fw coef0225[] = { + UPDATE_COEF(0x45, 0x3f<<10, 0x34<<10), + {} + }; static struct coef_fw coef0255[] = { WRITE_COEF(0x45, 0xc089), WRITE_COEF(0x45, 0xc489), @@ -3842,6 +3846,9 @@ static void alc_headset_mode_default(struct hda_codec *codec) }; switch (codec->core.vendor_id) { + case 0x10ec0225: + alc_process_coef_fw(codec, coef0225); + break; case 0x10ec0255: case 0x10ec0256: alc_process_coef_fw(codec, coef0255); @@ -4750,6 +4757,7 @@ enum { ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, + ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5375,6 +5383,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable pass-through path for FRONT 14h */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x36 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5646,10 +5665,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x21, 0x03211020} static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701a0}), - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701b0}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fixup speaker pass-through control for nid 0x14 on ALC225
by David Henningsson 25 Feb '16

25 Feb '16

On one of the machines we enable, we found that the actual speaker volume did not always correspond to the volume set in alsamixer. This patch fixes that problem. This patch was orginally written by Kailang @ Realtek, I've rebased it to fit sound git master. BugLink: https://bugs.launchpad.net/bugs/1549660 Co-Authored-By: Kailang <kailang(a)realtek.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++-- 1 file changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index efd4980..ab353b0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3801,6 +3801,10 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, static void alc_headset_mode_default(struct hda_codec *codec) { + static struct coef_fw coef0225[] = { + UPDATE_COEF(0x45, 0x3f<<10, 0x34<<10), + {} + }; static struct coef_fw coef0255[] = { WRITE_COEF(0x45, 0xc089), WRITE_COEF(0x45, 0xc489), @@ -3842,6 +3846,9 @@ static void alc_headset_mode_default(struct hda_codec *codec) }; switch (codec->core.vendor_id) { + case 0x10ec0225: + alc_process_coef_fw(codec, coef0225); + break; case 0x10ec0255: case 0x10ec0256: alc_process_coef_fw(codec, coef0255); @@ -4749,6 +4756,7 @@ enum { ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, + ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5368,6 +5376,17 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_lenovo_line2_mic_hotkey, }, + [ALC225_FIXUP_DELL1_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Disable pass-through path for FRONT 14h */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x36 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x57d7 }, + {} + }, + .chained = true, + .chain_id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5638,10 +5657,10 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {0x21, 0x03211020} static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701a0}), - SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + SND_HDA_PIN_QUIRK(0x10ec0225, 0x1028, "Dell", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x14, 0x901701b0}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE, -- 1.9.1

2 2

[alsa-devel] [patch 1/2] ASoC: max9867: signedness bug in max9867_dai_hw_params()
by Dan Carpenter 25 Feb '16

25 Feb '16

"value" needs to be signed for the error handling to work. Fixes: 805d132dcbbd ('ASoC: Add max9867 codec driver') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index ab7a94d..4fb6fc7 100755 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -180,7 +180,8 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct max9867_priv *max9867 = snd_soc_codec_get_drvdata(codec); - unsigned int ni_h, ni_l, value; + unsigned int ni_h, ni_l; + int value; value = get_ni_value(max9867->sysclk, params_rate(params)); if (value < 0)

2 1

[alsa-devel] [PATCH] ALSA: hda - Fixing background noise on Dell Inspiron 3162
by Kai-Heng Feng 25 Feb '16

25 Feb '16

After login to the desktop on Dell Inspiron 3162, there's a very loud background noise comes from the builtin speaker. The noise does not go away even if the speaker is muted. The noise disappears after using the aamix fixup. Codec: Realtek ALC3234 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0255 Subsystem Id: 0x10280725 Revision Id: 0x100002 No Modem Function Group found BugLink: http://bugs.launchpad.net/bugs/1549620 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index efd4980..72fa58d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4749,6 +4749,7 @@ enum { ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, + ALC255_FIXUP_DELL_SPK_NOISE, }; static const struct hda_fixup alc269_fixups[] = { @@ -5368,6 +5369,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_lenovo_line2_mic_hotkey, }, + [ALC255_FIXUP_DELL_SPK_NOISE] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_disable_aamix, + .chained = true, + .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -5410,6 +5417,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.5.0

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[alsa-devel] [PATCH 0/2 v2] ALSA: oxfw: implement retries for scs1x at transaction failure
by Takashi Sakamoto 25 Feb '16

25 Feb '16

Hi, This patchset updates my former post below. Changes are just adding my sign (I forgot it...) and improvements of commit messages. [alsa-devel] [PATCH 0/2] ALSA: oxfw: implement retries for scs1x at transaction failure http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104579.h… In thread of the post, I got some comments about work to extend ALSA rawmidi core. I've investigated it in a few days ago and realized that it may not so light work, mainly because of a lack of unified state management[0]. In this development cycle, I'd like to give it up and use my time for the other work within my plan for the cycle. Please just apply these two patches. [0] In short, the state management is understandable for developers who can understand it, and I'm not. Regards Takashi Sakamoto (2): ALSA: oxfw: retry MIDI transferring for scs1x at transaction failure ALSA: oxfw: discontinue MIDI substream for scs1x at transaction failure sound/firewire/oxfw/oxfw-scs1x.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH] ASoC: arizona: Make logging of FLL calculations clearer
by Richard Fitzgerald 25 Feb '16

25 Feb '16

The debug logging of FLL calculations was confusing. Values were printed in hex without indicating this by a leading 0x, and despite these normally being required in decimal. Also where the register value isn't the actual value (it s a power-of-two or 0 means 1, 1 means 2, ...) it was unclear whether the actual or register value was shown. This patch changes the log print so that all mathematical values are shown in decimal, all register values are shown in hex with a 0x prefix, and where the register value isn't the actual integer value the decimal integer value is shown in () after the hex register value. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 913cfa8..92d22a0 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2149,11 +2149,12 @@ static int arizona_calc_fll(struct arizona_fll *fll, return -EINVAL; } - arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", + arizona_fll_dbg(fll, "N=%d THETA=%d LAMBDA=%d\n", cfg->n, cfg->theta, cfg->lambda); - arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", - cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); - arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain); + arizona_fll_dbg(fll, "FRATIO=0x%x(%d) OUTDIV=%d REFCLK_DIV=0x%x(%d)\n", + cfg->fratio, ratio, cfg->outdiv, + cfg->refdiv, 1 << cfg->refdiv); + arizona_fll_dbg(fll, "GAIN=0x%x(%d)\n", cfg->gain, 1 << cfg->gain); return 0; -- 1.9.1

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Re: [alsa-devel] Fwd: Lenovo Y700-17ISK notebook - multiple issues
by Takashi Iwai 24 Feb '16

24 Feb '16

On Wed, 24 Feb 2016 19:00:36 +0100, Aljosa wrote: > > Hi Takashi, > thank you very much for your reply. > Notebook subwoofer doesn't work at all (it does not produce any sound). OK, could you give more details? Which kernel did you test? Did you test any model option? Also give alsa-info.sh output (run it with --no-upload option and attach the output). This contains the h/w details. Last but not least, don't drop Cc to ML. thanks, Takashi > Regards, Aljosa > > > > On 2016-02-24 16:35, Takashi Iwai wrote: > > On Tue, 23 Feb 2016 20:37:21 +0100, > > Aljosa wrote: > >> > >> Hello, > >> advised by one of the kernel developers - I'm reporting you the > >> problem > >> nr.2 from the below link. > > > > Could you describe what are the exact issues, instead of pointing some > > URL? > > > > > > thanks, > > > > Takashi >

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[alsa-devel] Codec "virtual multiplexer" representation
by Petr Kulhavy 24 Feb '16

24 Feb '16

Hi, I'm writing an ASoC driver for the TAS5721 amplifier. This 2.1 amplifier provides a complex configuration of the audio channels (3), PWMs(4) and output(4) mappings. However only few (about 6) configurations really make sense. I'm trying to model the output configurations in ALSA. In order to avoid a complicated set of switches and multiplexers I would like to add just one multiplexer like control which selects one of these 6 output configurations and a callback function which sets the proper registers accordingly (there are 5 affected registers). Is there a way to create such a "virtual multiplexer" in ALSA? Thanks Petr

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[alsa-devel] Fwd: Lenovo Y700-17ISK notebook - multiple issues
by Aljosa 24 Feb '16

24 Feb '16

Hello, advised by one of the kernel developers - I'm reporting you the problem nr.2 from the below link. Best regards, Aljosa -------- Original Message -------- Subject: Lenovo Y700-17ISK notebook - multiple issues Date: 2016-02-21 11:18 From: Aljosa <aljosa(a)cromalternativemoney.org> To: jkosina(a)suse.com Reply-To: aljosa(a)cromalternativemoney.org Hi, please excuse me if I contact you in this way - I found your email by googling "Lenovo" inside the Linux kernel mailing list archive. Few days ago I purchased a new Lenovo Y700-17ISK notebook, and after installing "current Ubuntu 16.04 daily build" I'm facing some issues which I think are kernel related: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1547934 Consequently, I would like to inform kernel developers about these problems. If you are not the right person to contact on this subject, please route this e-mail to whomever can help me or let me know who to contact - thank you very much in advance. Best regards, Aljosa PS. I have just tried freshly released kernel v4.5-rc5-wily (http://kernel.ubuntu.com/~kernel-ppa/mainline/?C=N;O=D), and with it my wireless connection now works, but there are still plenty problematic lines inside DMESG output.

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[alsa-devel] [BUG] [REGRESSION] [BISECTED] -rc1 breaks audio over HDMI for i915
by Martin Kepplinger 24 Feb '16

24 Feb '16

The following change: 788d441 ALSA: hda - Use component ops for i915 HDMI/DP audio jack handling breaks audio over HDMI on my snd_hda_intel laptop. It is the first bad commit. This was merged for -rc1 and isn't fixed until now, so I got nervous. There are no errors in the log that stand out. I'm no sound or alsa developer, and just reverting it isn't easy since there is stuff depending on this. I'm happy to test possible fixes and hope v4.5 not to ship with such a big regression ;)

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[alsa-devel] ASoC: simultaneous start of input/output channels
by Jonah Petri 24 Feb '16

24 Feb '16

Hello! Our application is very low-latency full duplex audio on a Freescale SoC. ALSA is configured with 2 64-frame buffers, with DSP happening on a realtime user thread. With small buffer sizes, we're obviously very tight on timing, and I've run into an issue which seems architectural in nature, and seemingly is worthy of consulting the list. What we're seeing is a large temporal latency between the DMA completion IRQ for the audio input DMA and audio output DMA. This delay is on the order of 200 microseconds, but also somewhat variable (ok, "large" is relative here, but it matters to us!). In the ideal case, these two interrupts would be perfectly aligned in time, allowing us to run DSP for the full buffer's duration. However, since the interrupts are offset by about 20% of a buffer, we effectively lose that time for our DSP. And, since the timing is variable, and we're in a realtime application, this produces scared engineers. Working backwards from the hardware drivers, I see that snd_soc_dai_ops's trigger callback is used to start and stop the DMA. The fundamental "start" action comes down to a single register write which can easily start both input and output streams in the same write. However, it seems that the snd_soc_dai_ops does not provide for simultaneous start of both input and output streams. Tracing back through the kernel interface and into user space, it seems that starting both the input and output streams necessitates two separate calls from userspace, which could explain the delay, and some of the variability in timing. My question is: * Did I miss some API or technique which could allow simultaneous start of full duplex streams? * If not, I would certainly appreciate advice on where we could fit this into the existing architecture! We would be glad to do the kernel work to make this change, as this capability seems generally useful! Thanks much for any help, Jonah

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[alsa-devel] [asoc:topic/intel 18/22] sound/soc/intel/boards/skl_nau88l25_max98357a.c:25:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory
by kbuild test robot 24 Feb '16

24 Feb '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 8be3838af0e992846d4725fae152b818560c1068 commit: 0ab338ff33762d50e5005d8d28b30b25f23aa934 [18/22] ASoC: Intel: boards: Enable HDMI and DP on SKL nau88l25_max98357 machine config: i386-allmodconfig (attached as .config) reproduce: git checkout 0ab338ff33762d50e5005d8d28b30b25f23aa934 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): >> sound/soc/intel/boards/skl_nau88l25_max98357a.c:25:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory compilation terminated. vim +25 sound/soc/intel/boards/skl_nau88l25_max98357a.c 19 #include <sound/core.h> 20 #include <sound/jack.h> 21 #include <sound/pcm.h> 22 #include <sound/pcm_params.h> 23 #include <sound/soc.h> 24 #include "../../codecs/nau8825.h" > 25 #include "../../codecs/hdac_hdmi.h" 26 27 #define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" 28 #define SKL_MAXIM_CODEC_DAI "HiFi" --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/intel 16/22] sound/soc/intel/boards/skl_rt286.c:29:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory
by kbuild test robot 24 Feb '16

24 Feb '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 8be3838af0e992846d4725fae152b818560c1068 commit: 23905cd1ca48cbc852a670a3dbbfc5e7d694580f [16/22] ASoC: Intel: boards: Enable HDMI and DP on skl_rt286 machine config: i386-allmodconfig (attached as .config) reproduce: git checkout 23905cd1ca48cbc852a670a3dbbfc5e7d694580f # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): >> sound/soc/intel/boards/skl_rt286.c:29:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory compilation terminated. vim +29 sound/soc/intel/boards/skl_rt286.c 23 #include <sound/core.h> 24 #include <sound/pcm.h> 25 #include <sound/soc.h> 26 #include <sound/jack.h> 27 #include <sound/pcm_params.h> 28 #include "../../codecs/rt286.h" > 29 #include "../../codecs/hdac_hdmi.h" 30 31 static struct snd_soc_jack skylake_headset; 32 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [asoc:topic/intel 17/22] sound/soc/intel/boards/skl_nau88l25_ssm4567.c:29:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory
by kbuild test robot 24 Feb '16

24 Feb '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: 8be3838af0e992846d4725fae152b818560c1068 commit: bc5f6ac9a66f90f8e16e568ebd463cbe44f46b81 [17/22] ASoC: Intel: boards: Enable HDMI and DP on nau88l2 machine config: i386-allmodconfig (attached as .config) reproduce: git checkout bc5f6ac9a66f90f8e16e568ebd463cbe44f46b81 # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): >> sound/soc/intel/boards/skl_nau88l25_ssm4567.c:29:36: fatal error: ../../codecs/hdac_hdmi.h: No such file or directory compilation terminated. vim +29 sound/soc/intel/boards/skl_nau88l25_ssm4567.c 23 #include <sound/core.h> 24 #include <sound/pcm.h> 25 #include <sound/soc.h> 26 #include <sound/jack.h> 27 #include <sound/pcm_params.h> 28 #include "../../codecs/nau8825.h" > 29 #include "../../codecs/hdac_hdmi.h" 30 31 #define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" 32 #define SKL_SSM_CODEC_DAI "ssm4567-hifi" --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH 1/3] ASoC: rt298: enable IRQ for jack detection
by Bard Liao 24 Feb '16

24 Feb '16

There are some registers needed for enabling rt298 IRQ missed on current driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 6 ++++++ sound/soc/codecs/rt298.h | 8 ++++++++ 2 files changed, 14 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 30c6de6..e67ef5f 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1225,6 +1225,12 @@ static int rt298_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt298->regmap, RT298_WIND_FILTER_CTRL, 0x0082, 0x0082); regmap_update_bits(rt298->regmap, RT298_IRQ_CTRL, 0x2, 0x2); + + regmap_write(rt298->regmap, RT298_UNSOLICITED_INLINE_CMD, 0x81); + regmap_write(rt298->regmap, RT298_UNSOLICITED_HP_OUT, 0x82); + regmap_write(rt298->regmap, RT298_UNSOLICITED_MIC1, 0x84); + regmap_update_bits(rt298->regmap, RT298_IRQ_FLAG_CTRL, 0x2, 0x2); + rt298->is_hp_in = -1; if (rt298->i2c->irq) { diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index 31da162..d66f884 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -34,6 +34,7 @@ #define RT298_HP_OUT 0x21 #define RT298_MIXER_IN1 0x22 #define RT298_MIXER_IN2 0x23 +#define RT298_INLINE_CMD 0x55 #define RT298_SET_PIN_SFT 6 #define RT298_SET_PIN_ENABLE 0x40 @@ -124,6 +125,12 @@ VERB_CMD(AC_VERB_SET_COEF_INDEX, RT298_VENDOR_REGISTERS, 0) #define RT298_PROC_COEF\ VERB_CMD(AC_VERB_SET_PROC_COEF, RT298_VENDOR_REGISTERS, 0) +#define RT298_UNSOLICITED_INLINE_CMD\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_INLINE_CMD, 0) +#define RT298_UNSOLICITED_HP_OUT\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_HP_OUT, 0) +#define RT298_UNSOLICITED_MIC1\ + VERB_CMD(AC_VERB_SET_UNSOLICITED_ENABLE, RT298_MIC1, 0) /* Index registers */ #define RT298_A_BIAS_CTRL1 0x01 @@ -148,6 +155,7 @@ #define RT298_DEPOP_CTRL2 0x67 #define RT298_DEPOP_CTRL3 0x68 #define RT298_DEPOP_CTRL4 0x69 +#define RT298_IRQ_FLAG_CTRL 0x7c /* SPDIF (0x06) */ #define RT298_SPDIF_SEL_SFT 0 -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/7] ASoC: Intel: boards: Enabled HDMI/DP and update WOV on SKL machines
by Vinod Koul 24 Feb '16

24 Feb '16

This series adds the multiple port support for HDMI and DP on respective SKL machine drivers. Also the WoV sink is removed as that needs to be associated with PCM device and not a DAPM Sink Jeeja KP (5): ASoC: Intel: boards: Enable HDMI and DP on skl_rt286 machine ASoC: Intel: boards: Enable HDMI and DP on nau88l2 machine ASoC: Intel: boards: Remove WoV sink in skl_rt286 ASoC: Intel: boards: Remove WoV sink in nau88125_ssm4567 machine ASoC: Intel: boards: Remove WoV sink in nau88125_max98357a machine Sathya Prakash M R (1): ASoC: Intel: boards: Add IV feedback route for nau88l25_ssm4567 machine Sathyanarayana Nujella (1): ASoC: Intel: boards: Enable HDMI and DP on SKL nau88l25_max98357 machine sound/soc/intel/Kconfig | 3 + sound/soc/intel/boards/skl_nau88l25_max98357a.c | 122 +++++++++++++++++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 145 +++++++++++++++++++++--- sound/soc/intel/boards/skl_rt286.c | 113 ++++++++++++++++-- 4 files changed, 339 insertions(+), 44 deletions(-) -- 1.9.1

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