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[PATCH] ALSA: hda: Drop power save deny list entry for Clevo W65_67SB
by Takashi Iwai 02 Feb '21

02 Feb '21

As the runtime PM issue was addressed by the recent fix 4961167bf748 ("ALSA: hda/via: Apply the workaround generically for Clevo machines") for VIA codecs, we need no longer to keep the Clevo device off from the power saving as default. Drop the deny list entry accordingly. Depends: 4961167bf748 ("ALSA: hda/via: Apply the workaround generically for Clevo machines") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5a50d3a46445..ef1f3aabc4a3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2221,8 +2221,6 @@ static const struct snd_pci_quirk power_save_denylist[] = { /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1043, 0x8733, "Asus Prime X370-Pro", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ - SND_PCI_QUIRK(0x1558, 0x6504, "Clevo W65_67SB", 0), - /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ SND_PCI_QUIRK(0x1028, 0x0497, "Dell Precision T3600", 0), /* https://bugzilla.redhat.com/show_bug.cgi?id=1525104 */ /* Note the P55A-UD3 and Z87-D3HP share the subsys id for the HDA dev */ -- 2.26.2

1 0

[bug report] potential deadlock in snd_emu10k1_voice_alloc()
by Dan Carpenter 02 Feb '21

02 Feb '21

Hello ALSA devs, The patch 1da177e4c3f4: "Linux-2.6.12-rc2" from Apr 16, 2005, leads to the following static checker warning: sound/pci/emu10k1/voice.c:112 snd_emu10k1_voice_alloc() warn: called with lock held. '&emu->voice_lock' sound/pci/emu10k1/voice.c 101 int snd_emu10k1_voice_alloc(struct snd_emu10k1 *emu, int type, int number, 102 struct snd_emu10k1_voice **rvoice) 103 { 104 unsigned long flags; 105 int result; 106 107 if (snd_BUG_ON(!rvoice)) 108 return -EINVAL; 109 if (snd_BUG_ON(!number)) 110 return -EINVAL; 111 112 spin_lock_irqsave(&emu->voice_lock, flags); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ This lock is already held in the caller. 113 for (;;) { 114 result = voice_alloc(emu, type, number, rvoice); 115 if (result == 0 || type == EMU10K1_SYNTH || type == EMU10K1_MIDI) 116 break; 117 118 /* free a voice from synth */ 119 if (emu->get_synth_voice) { The call tree is: snd_emux_note_on() takes the lock: spin_lock_irqsave(&emu->voice_lock, flags); calls vp = emu->ops.get_voice(emu, port); --> get_voice() --> snd_emu10k1_voice_alloc() regards, dan carpenter

2 1

[PATCH] Add support to expose controls of ladspa plugin
by Camel Guo 02 Feb '21

02 Feb '21

From: Camel Guo <camelg(a)axis.com> In order for external software components to adjust ladspa plugin dynamically, this commit adds an option to exposes the control array of input control ports of a ladspa plugin to a file, through which any applications with proper permission can control this plugin. Signed-off-by: Camel Guo <camelg(a)axis.com> --- src/pcm/pcm_ladspa.c | 157 ++++++++++++++++++++++++++++++++++++++----- 1 file changed, 139 insertions(+), 18 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index ad73347d..40b5d38f 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -32,6 +32,9 @@ * http://www.medianet.ag */ +#include <stdbool.h> +#include <sys/mman.h> +#include <sys/stat.h> #include <dirent.h> #include <locale.h> #include <math.h> @@ -93,7 +96,8 @@ typedef struct { unsigned int port_bindings_size; /* size of array */ unsigned int *port_bindings; /* index = channel number, value = LADSPA port */ unsigned int controls_size; /* size of array */ - unsigned char *controls_initialized; /* initialized by ALSA user */ + bool controls_new; /* if controls is new, it can be overrided by alsa config */ + bool controls_shared; /* if controls is shared memory map */ LADSPA_Data *controls; /* index = LADSPA control port index */ } snd_pcm_ladspa_plugin_io_t; @@ -101,6 +105,7 @@ typedef struct { struct list_head list; snd_pcm_ladspa_policy_t policy; char *filename; + char *controls_path; void *dl_handle; const LADSPA_Descriptor *desc; snd_pcm_ladspa_plugin_io_t input; @@ -110,6 +115,73 @@ typedef struct { #endif /* DOC_HIDDEN */ +static LADSPA_Data* mmap_ladspa_controls(const char* filename, unsigned long length, bool *is_new) +{ + LADSPA_Data *ptr = NULL; + int fd = -1; + int ret = 0; + struct stat statbuf = { 0 }; + int prot = PROT_READ; + + if (filename == NULL || is_new == NULL) return NULL; + + *is_new = false; + + fd = open(filename, O_RDONLY); + if(fd < 0) { + if (errno == ENOENT) { + fd = open(filename, O_RDWR | O_CREAT, 0666); + if(fd < 0) { + SNDERR("Failed to open controls file: %s", filename); + return NULL; + } + + prot |= PROT_WRITE; + *is_new = true; + } else { + SNDERR("Failed to open %s due to '%s'", filename, strerror(errno)); + return NULL; + } + } + + ret = fstat(fd, &statbuf); + if (ret == -1) { + SNDERR("Failed to get status of '%s' due to '%s'", filename, strerror(errno)); + goto out; + } + + if (statbuf.st_size < (off_t) length) { + /* this file is invalid, must be truncated and re-initialized later on */ + close(fd); + + fd = open(filename, O_RDWR | O_CREAT, 0666); + if(fd < 0) { + SNDERR("Failed to open controls file: %s", filename); + /* no need to close(fd), so just return */ + return NULL; + } + + ret = ftruncate(fd, length); + if (ret == -1) { + SNDERR("Failed to increase the size of '%s' due to '%s'", filename, strerror(errno)); + goto out; + } + + prot |= PROT_WRITE; + *is_new = true; + } + + ptr = (LADSPA_Data*) mmap(NULL, length, prot, MAP_SHARED, fd, 0); + if(ptr == MAP_FAILED) { + SNDERR("Failed to mmap %s due to %s", filename, strerror(errno)); + ptr = NULL; + } + +out: + close (fd); + return ptr; +} + static unsigned int snd_pcm_ladspa_count_ports(snd_pcm_ladspa_plugin_t *lplug, LADSPA_PortDescriptor pdesc) { @@ -174,8 +246,11 @@ static int snd_pcm_ladspa_find_port_idx(unsigned int *res, static void snd_pcm_ladspa_free_io(snd_pcm_ladspa_plugin_io_t *io) { - free(io->controls); - free(io->controls_initialized); + if (io->controls_shared) { + munmap(io->controls, io->controls_size * sizeof(LADSPA_Data)); + } else { + free(io->controls); + } } static void snd_pcm_ladspa_free_plugins(struct list_head *plugins) @@ -186,6 +261,8 @@ static void snd_pcm_ladspa_free_plugins(struct list_head *plugins) snd_pcm_ladspa_free_io(&plugin->output); if (plugin->dl_handle) dlclose(plugin->dl_handle); + if (plugin->controls_path) + free(plugin->controls_path); free(plugin->filename); list_del(&plugin->list); free(plugin); @@ -574,8 +651,6 @@ static int snd_pcm_ladspa_connect_controls(snd_pcm_ladspa_plugin_t *plugin, for (idx = midx = 0; idx < plugin->desc->PortCount; idx++) if ((plugin->desc->PortDescriptors[idx] & (io->pdesc | LADSPA_PORT_CONTROL)) == (io->pdesc | LADSPA_PORT_CONTROL)) { if (io->controls_size > midx) { - if (!io->controls_initialized[midx]) - snd_pcm_ladspa_get_default_cvalue(plugin->desc, idx, &io->controls[midx]); plugin->desc->connect_port(instance->handle, idx, &io->controls[midx]); } else { return -EINVAL; @@ -878,6 +953,10 @@ snd_pcm_ladspa_write_areas(snd_pcm_t *pcm, snd_pcm_ladspa_plugin_t *plugin = list_entry(pos, snd_pcm_ladspa_plugin_t, list); list_for_each(pos1, &plugin->instances) { instance = list_entry(pos1, snd_pcm_ladspa_instance_t, list); + if (plugin->input.controls_shared) { + (void) snd_pcm_ladspa_connect_controls(plugin, &plugin->input, instance); + } + /* Skip output controls since they can not be changed dynamically */ for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->input.channels.array[idx]; data = instance->input.data[idx]; @@ -939,6 +1018,10 @@ snd_pcm_ladspa_read_areas(snd_pcm_t *pcm, snd_pcm_ladspa_plugin_t *plugin = list_entry(pos, snd_pcm_ladspa_plugin_t, list); list_for_each(pos1, &plugin->instances) { instance = list_entry(pos1, snd_pcm_ladspa_instance_t, list); + if (plugin->input.controls_shared) { + (void) snd_pcm_ladspa_connect_controls(plugin, &plugin->input, instance); + } + /* Skip output controls since they can not be changed dynamically */ for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->input.channels.array[idx]; data = instance->input.data[idx]; @@ -1227,26 +1310,50 @@ static int snd_pcm_ladspa_add_default_controls(snd_pcm_ladspa_plugin_t *lplug, { unsigned int count = 0; LADSPA_Data *array; - unsigned char *initialized; unsigned long idx; + unsigned long midx; + bool controls_shared = false; + bool is_new = true; for (idx = 0; idx < lplug->desc->PortCount; idx++) if ((lplug->desc->PortDescriptors[idx] & (io->pdesc | LADSPA_PORT_CONTROL)) == (io->pdesc | LADSPA_PORT_CONTROL)) count++; - array = (LADSPA_Data *)calloc(count, sizeof(LADSPA_Data)); - if (!array) - return -ENOMEM; - initialized = (unsigned char *)calloc(count, sizeof(unsigned char)); - if (!initialized) { - free(array); - return -ENOMEM; + + /* + * Only support to expose ladspa control array for input control ports + */ + if ((io->pdesc == LADSPA_PORT_INPUT) && lplug->controls_path && count != 0) { + array = mmap_ladspa_controls(lplug->controls_path, count * sizeof(LADSPA_Data), &is_new); + if (!array) + return -ENOMEM; + controls_shared = true; + } else { + array = (LADSPA_Data *)calloc(count, sizeof(LADSPA_Data)); + if (!array) + return -ENOMEM; + } + + if (is_new) { + /* + * Initialize array of controls of this ladspa plugin with its + * default values defined in this plugin implementation. This + * array can be overrided by its alsa configuration files. + */ + for (idx = midx = 0; idx < lplug->desc->PortCount; idx++) { + if ((lplug->desc->PortDescriptors[idx] & (io->pdesc | LADSPA_PORT_CONTROL)) == (io->pdesc | LADSPA_PORT_CONTROL)) { + snd_pcm_ladspa_get_default_cvalue(lplug->desc, idx, &array[midx]); + midx++; + } + } } + io->controls_size = count; - io->controls_initialized = initialized; + io->controls_new = is_new; + io->controls_shared = controls_shared; io->controls = array; return 0; -} +} static int snd_pcm_ladspa_parse_controls(snd_pcm_ladspa_plugin_t *lplug, snd_pcm_ladspa_plugin_io_t *io, @@ -1287,8 +1394,10 @@ static int snd_pcm_ladspa_parse_controls(snd_pcm_ladspa_plugin_t *lplug, SNDERR("internal error"); return err; } - io->controls_initialized[uval] = 1; - io->controls[uval] = (LADSPA_Data)dval; + + if (io->controls_new) { + io->controls[uval] = (LADSPA_Data)dval; + } } return 0; @@ -1429,7 +1538,7 @@ static int snd_pcm_ladspa_add_plugin(struct list_head *list, int reverse) { snd_config_iterator_t i, next; - const char *label = NULL, *filename = NULL; + const char *label = NULL, *filename = NULL, *controls_path = NULL; long ladspa_id = 0; int err; snd_pcm_ladspa_plugin_t *lplug; @@ -1467,6 +1576,12 @@ static int snd_pcm_ladspa_add_plugin(struct list_head *list, output = n; continue; } + if (strcmp(id, "controls") == 0) { + err = snd_config_get_string(n, &controls_path); + if (err < 0) + return err; + continue; + } if (strcmp(id, "policy") == 0) { const char *str; err = snd_config_get_string(n, &str); @@ -1496,6 +1611,11 @@ static int snd_pcm_ladspa_add_plugin(struct list_head *list, lplug->input.pdesc = LADSPA_PORT_INPUT; lplug->output.pdesc = LADSPA_PORT_OUTPUT; INIT_LIST_HEAD(&lplug->instances); + if (controls_path) { + lplug->controls_path = strdup(controls_path); + } else { + lplug->controls_path = NULL; + } if (filename) { err = snd_pcm_ladspa_check_file(lplug, filename, label, ladspa_id); if (err < 0) { @@ -1692,6 +1812,7 @@ pcm.name { [label STR] # LADSPA plugin label (for example 'delay_5s') [filename STR] # Full filename of .so library with LADSPA plugin code [policy STR] # Policy can be 'none' or 'duplicate' + [controls STR] # Path (directory) with controls for this plugin input | output { bindings { C INT or STR # C - channel, INT - audio port index, STR - audio port name -- 2.20.1

3 2

[bug report] ALSA: rawmidi: Avoid racy info ioctl via ctl device
by Dan Carpenter 02 Feb '21

02 Feb '21

Hello Takashi Iwai, The patch c1cfd9025cc3: "ALSA: rawmidi: Avoid racy info ioctl via ctl device" from Dec 14, 2017, leads to the following static checker warning: sound/core/rawmidi.c:651 snd_rawmidi_info_select() warn: called with lock held. '&register_mutex' sound/core/seq/seq_midi.c 297 mutex_lock(&register_mutex); ^^^^^^^^^^^^^^^ Holding this lock. 298 client = synths[card->number]; 299 if (client == NULL) { 300 newclient = 1; 301 client = kzalloc(sizeof(*client), GFP_KERNEL); 302 if (client == NULL) { 303 mutex_unlock(&register_mutex); 304 kfree(info); 305 return -ENOMEM; 306 } 307 client->seq_client = 308 snd_seq_create_kernel_client( 309 card, 0, "%s", card->shortname[0] ? 310 (const char *)card->shortname : "External MIDI"); 311 if (client->seq_client < 0) { 312 kfree(client); 313 mutex_unlock(&register_mutex); 314 kfree(info); 315 return -ENOMEM; 316 } 317 } 318 319 msynth = kcalloc(ports, sizeof(struct seq_midisynth), GFP_KERNEL); 320 port = kmalloc(sizeof(*port), GFP_KERNEL); 321 if (msynth == NULL || port == NULL) 322 goto __nomem; 323 324 for (p = 0; p < ports; p++) { 325 ms = &msynth[p]; 326 327 if (snd_seq_midisynth_new(ms, card, device, p) < 0) 328 goto __nomem; 329 330 /* declare port */ 331 memset(port, 0, sizeof(*port)); 332 port->addr.client = client->seq_client; 333 port->addr.port = device * (256 / SNDRV_RAWMIDI_DEVICES) + p; 334 port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; 335 memset(info, 0, sizeof(*info)); 336 info->device = device; 337 if (p < output_count) 338 info->stream = SNDRV_RAWMIDI_STREAM_OUTPUT; 339 else 340 info->stream = SNDRV_RAWMIDI_STREAM_INPUT; 341 info->subdevice = p; 342 if (snd_rawmidi_info_select(card, info) >= 0) ^^^^^^^^^^^^^^^^^^^^^^^ We can't call this function when we're holding the lock or it leads to a deadlock. Before the patch, we used to rely on the callers to take the lock before calling snd_rawmidi_info_select() but the patch moved the lock inside the function. 343 strcpy(port->name, info->subname); 344 if (! port->name[0]) { 345 if (info->name[0]) { 346 if (ports > 1) 347 snprintf(port->name, sizeof(port->name), "%s-%u", info->name, p); 348 else 349 snprintf(port->name, sizeof(port->name), "%s", info->name); regards, dan carpenter

2 2

[PATCH] ASoC: fsl_xcvr: remove unneeded semicolon
by Yang Li 02 Feb '21

02 Feb '21

Eliminate the following coccicheck warning: ./sound/soc/fsl/fsl_xcvr.c:739:2-3: Unneeded semicolon Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/fsl/fsl_xcvr.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 3d58c88..65b388a 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -736,7 +736,7 @@ static int fsl_xcvr_load_firmware(struct fsl_xcvr *xcvr) /* clean current page, including data memory */ memset_io(xcvr->ram_addr, 0, size); } - }; + } err_firmware: release_firmware(fw); -- 1.8.3.1

2 1

conf: HDA-Intel: increase the max number of HDMI/DP output to 11
by GitHub pull_request - opened 02 Feb '21

02 Feb '21

alsa-project/alsa-lib pull request #118 was opened from jason77-wang: Intel TGL HDMI/DP codec provides 9 pins (Linux kernel, 9a11ba7388f16: ALSA: hda: hdmi - add Tigerlake support), and with the DP MST enabled, the linux kernel will build 11 output devices (3, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16), and current alsa-lib only maps the first 8 output devices, if users plug the HDMI/DP monitor to the last 4 ports, the users will not see the output device from pulseaudio or gnome. We have experienced this issue on a dell TGL machine with a dock, we plugged 2 HDMP/DP monitors on the dock, but we could only see 1 HDMI/DP output device from pulseaudio or gnome, through investigation, we found one monitor is plugged in the 2nd port from last. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/118 Patch URL : https://github.com/alsa-project/alsa-lib/pull/118.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 2/2] ALSA: usb-audio: Add DJM-450 to the quirks table
by Olivia Mackintosh 02 Feb '21

02 Feb '21

As with most Pioneer devices, the device descriptor is vendor specific and as such, the number of channels, the PCM format, endpoints and sample rate need to be specified. This device has 8 inputs and 8 outputs and a sample rate of 48000 only. The PCM format is S24_3LE like other devices. There seems to be an appetite for reducing duplication amongs these Pioneer patches but again, I feel this is a step to be taken after support has been added as it's not completely clear where the commonalities are. Signed-off-by: Olivia Mackintosh <livvy(a)base.nu> --- sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 93d55cd1a5a4..1165a5ac60f2 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3817,6 +3817,63 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } } }, +{ + /* + * Pioneer DJ DJM-450 + * PCM is 8 channels out @ 48 fixed (endpoint 0x01) + * and 8 channels in @ 48 fixed (endpoint 0x82). + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0013), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // outputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE -- 2.30.0

1 0

[PATCH 0/2] ALSA: usb-audio: Add Pioneer DJM-450 support
by Olivia Mackintosh 02 Feb '21

02 Feb '21

Here are a couple of patches to support the DJM-450. Un-suprisingly, a device quirk is required just like the other DJM models. Testing was done by "xfz" from #xwax on Freenode. Perhaps this is useful for reference or if anyone requires information about this device. They were hesitant to have their full details in the commit log. Olivia Mackintosh (2): ALSA: usb-audio: Add DJM450 to Pioneer format quirk ALSA: usb-audio: Add DJM-450 to the quirks table sound/usb/quirks-table.h | 57 ++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.c | 20 ++++++++++++++ 2 files changed, 77 insertions(+) -- 2.30.0

1 0

[bug report] ASoC: cpcap: new codec
by Dan Carpenter 01 Feb '21

01 Feb '21

Hello Sebastian Reichel, The patch f6cdf2d3445d: "ASoC: cpcap: new codec" from Feb 23, 2018, leads to the following static checker warning: sound/soc/codecs/cpcap.c:1276 cpcap_voice_hw_params() warn: 'CPCAP_BIT_MIC1_RX_TIMESLOT0' is a shifter (not for '|='). sound/soc/codecs/cpcap.c:1279 cpcap_voice_hw_params() warn: 'CPCAP_BIT_MIC2_TIMESLOT0' is a shifter (not for '|='). sound/soc/codecs/cpcap.c 1253 static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, 1254 struct snd_pcm_hw_params *params, 1255 struct snd_soc_dai *dai) 1256 { 1257 struct snd_soc_component *component = dai->component; 1258 struct device *dev = component->dev; 1259 struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); 1260 static const u16 reg_cdi = CPCAP_REG_CDI; 1261 int rate = params_rate(params); 1262 int channels = params_channels(params); 1263 int direction = substream->stream; 1264 u16 val, mask; 1265 int err; 1266 1267 dev_dbg(dev, "Voice setup HW params: rate=%d, direction=%d, chan=%d", 1268 rate, direction, channels); 1269 1270 err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, rate); 1271 if (err) 1272 return err; 1273 1274 if (direction == SNDRV_PCM_STREAM_CAPTURE) { 1275 mask = 0x0000; 1276 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT0; ^^^^^^^^^^^^^^^^^^^^^^^^^^^ This should probably be BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0). 1277 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT1; 1278 mask |= CPCAP_BIT_MIC1_RX_TIMESLOT2; 1279 mask |= CPCAP_BIT_MIC2_TIMESLOT0; ^^^^^^^^^^^^^^^^^^^^^^^^ Same for all the others as well I guess. 1280 mask |= CPCAP_BIT_MIC2_TIMESLOT1; 1281 mask |= CPCAP_BIT_MIC2_TIMESLOT2; 1282 val = 0x0000; 1283 if (channels >= 2) 1284 val = BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Here it's BIT(CPCAP_BIT_MIC1_RX_TIMESLOT0). 1285 err = regmap_update_bits(cpcap->regmap, reg_cdi, mask, val); 1286 if (err) 1287 return err; 1288 } 1289 1290 return 0; 1291 } regards, dan carpenter

4 3

[PATCH] ASoC: tegra: SND_SOC_TEGRA_AUDIO_GRAPH_CARD should depend on SND_SOC_TEGRA
by Geert Uytterhoeven 01 Feb '21

01 Feb '21

Audio Graph Card based Tegra driver is only useful on NVIDIA Tegra SoCs. Hence add a dependency on SND_SOC_TEGRA, to prevent asking the user about this driver when configuring a kernel without Tegra sound support. Wrap all Tegra sound config options inside a big if/endif block, instead of just adding the dependency to the single config option that does not have it yet, to preventing similar future mistakes. Fixes: 202e2f7745437aa5 ("ASoC: tegra: Add audio graph based card driver") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/tegra/Kconfig | 33 +++++++++++++-------------------- 1 file changed, 13 insertions(+), 20 deletions(-) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 6dc83ad8607a1de2..a4e6760944d028cf 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -9,9 +9,10 @@ config SND_SOC_TEGRA help Say Y or M here if you want support for SoC audio on Tegra. +if SND_SOC_TEGRA + config SND_SOC_TEGRA20_AC97 tristate "Tegra20 AC97 interface" - depends on SND_SOC_TEGRA select SND_SOC_AC97_BUS select SND_SOC_TEGRA20_DAS help @@ -21,7 +22,6 @@ config SND_SOC_TEGRA20_AC97 config SND_SOC_TEGRA20_DAS tristate "Tegra20 DAS module" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 DAS module. You will also need to select the individual machine drivers to @@ -29,7 +29,6 @@ config SND_SOC_TEGRA20_DAS config SND_SOC_TEGRA20_I2S tristate "Tegra20 I2S interface" - depends on SND_SOC_TEGRA select SND_SOC_TEGRA20_DAS help Say Y or M if you want to add support for codecs attached to the @@ -38,7 +37,6 @@ config SND_SOC_TEGRA20_I2S config SND_SOC_TEGRA20_SPDIF tristate "Tegra20 SPDIF interface" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra20 SPDIF interface. You will also need to select the individual machine drivers to support @@ -46,7 +44,6 @@ config SND_SOC_TEGRA20_SPDIF config SND_SOC_TEGRA30_AHUB tristate "Tegra30 AHUB module" - depends on SND_SOC_TEGRA help Say Y or M if you want to add support for the Tegra30 AHUB module. You will also need to select the individual machine drivers to @@ -54,7 +51,6 @@ config SND_SOC_TEGRA30_AHUB config SND_SOC_TEGRA30_I2S tristate "Tegra30 I2S interface" - depends on SND_SOC_TEGRA select SND_SOC_TEGRA30_AHUB help Say Y or M if you want to add support for codecs attached to the @@ -63,7 +59,6 @@ config SND_SOC_TEGRA30_I2S config SND_SOC_TEGRA210_AHUB tristate "Tegra210 AHUB module" - depends on SND_SOC_TEGRA help Config to enable Audio Hub (AHUB) module, which comprises of a switch called Audio Crossbar (AXBAR) used to configure or modify @@ -73,7 +68,6 @@ config SND_SOC_TEGRA210_AHUB config SND_SOC_TEGRA210_DMIC tristate "Tegra210 DMIC module" - depends on SND_SOC_TEGRA help Config to enable the Digital MIC (DMIC) controller which is used to interface with Pulse Density Modulation (PDM) input devices. @@ -84,7 +78,6 @@ config SND_SOC_TEGRA210_DMIC config SND_SOC_TEGRA210_I2S tristate "Tegra210 I2S module" - depends on SND_SOC_TEGRA help Config to enable the Inter-IC Sound (I2S) Controller which implements full-duplex and bidirectional and single direction @@ -94,7 +87,6 @@ config SND_SOC_TEGRA210_I2S config SND_SOC_TEGRA186_DSPK tristate "Tegra186 DSPK module" - depends on SND_SOC_TEGRA help Config to enable the Digital Speaker Controller (DSPK) which converts the multi-bit Pulse Code Modulation (PCM) audio input to @@ -107,7 +99,6 @@ config SND_SOC_TEGRA186_DSPK config SND_SOC_TEGRA210_ADMAIF tristate "Tegra210 ADMAIF module" - depends on SND_SOC_TEGRA help Config to enable ADMAIF which is the interface between ADMA and Audio Hub (AHUB). Each ADMA channel that sends/receives data to/ @@ -128,7 +119,7 @@ config SND_SOC_TEGRA_AUDIO_GRAPH_CARD config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_RT5640 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -136,7 +127,7 @@ config SND_SOC_TEGRA_RT5640 config SND_SOC_TEGRA_WM8753 tristate "SoC Audio support for Tegra boards using a WM8753 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_WM8753 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -144,7 +135,7 @@ config SND_SOC_TEGRA_WM8753 config SND_SOC_TEGRA_WM8903 tristate "SoC Audio support for Tegra boards using a WM8903 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_WM8903 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -153,7 +144,7 @@ config SND_SOC_TEGRA_WM8903 config SND_SOC_TEGRA_WM9712 tristate "SoC Audio support for Tegra boards using a WM9712 codec" - depends on SND_SOC_TEGRA && GPIOLIB + depends on GPIOLIB select SND_SOC_TEGRA20_AC97 select SND_SOC_WM9712 help @@ -162,7 +153,7 @@ config SND_SOC_TEGRA_WM9712 config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" - depends on SND_SOC_TEGRA && I2C + depends on I2C select SND_SOC_TLV320AIC23_I2C help Say Y or M here if you want to add support for SoC audio on the @@ -170,7 +161,7 @@ config SND_SOC_TEGRA_TRIMSLICE config SND_SOC_TEGRA_ALC5632 tristate "SoC Audio support for Tegra boards using an ALC5632 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_ALC5632 help Say Y or M here if you want to add support for SoC audio on the @@ -178,7 +169,7 @@ config SND_SOC_TEGRA_ALC5632 config SND_SOC_TEGRA_MAX98090 tristate "SoC Audio support for Tegra boards using a MAX98090 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_MAX98090 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -186,7 +177,7 @@ config SND_SOC_TEGRA_MAX98090 config SND_SOC_TEGRA_RT5677 tristate "SoC Audio support for Tegra boards using a RT5677 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_RT5677 help Say Y or M here if you want to add support for SoC audio on Tegra @@ -194,9 +185,11 @@ config SND_SOC_TEGRA_RT5677 config SND_SOC_TEGRA_SGTL5000 tristate "SoC Audio support for Tegra boards using a SGTL5000 codec" - depends on SND_SOC_TEGRA && I2C && GPIOLIB + depends on I2C && GPIOLIB select SND_SOC_SGTL5000 help Say Y or M here if you want to add support for SoC audio on Tegra boards using the SGTL5000 codec, such as Apalis T30, Apalis TK1 or Colibri T30. + +endif -- 2.25.1

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