February 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] sound: pci/rme9652 - implement and expose controls for output
by fassl 02 Feb '21

02 Feb '21

>From 980aa253c17d233966e05a43f3611693e2e02040 Mon Sep 17 00:00:00 2001 From: Jasmin Fazlic <superfassl(a)gmail.com> Date: Sun, 31 Jan 2021 22:17:22 +0100 Subject: [PATCH] sound: pci/rme9652 - implement and expose controls for output loopback - so far only tested and enabled for RME HDSP9632 Signed-off-by: Jasmin Fazlic <superfassl(a)gmail.com> --- sound/pci/rme9652/hdsp.c | 85 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 84 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index cea53a878c36..7e832a502cea 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -469,6 +469,7 @@ struct hdsp { unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; + u32 io_loopback; /* output loopback channel states*/ struct snd_dma_buffer capture_dma_buf; struct snd_dma_buffer playback_dma_buf; @@ -3253,6 +3254,71 @@ static const struct snd_kcontrol_new snd_hdsp_96xx_aeb = HDSP_AnalogExtensionBoard); static struct snd_kcontrol_new snd_hdsp_adat_sync_check = HDSP_ADAT_SYNC_CHECK; + +static bool hdsp_loopback_get(struct hdsp *const hdsp, const u8 channel) +{ + return hdsp->io_loopback & (1 << channel); +} + +static int hdsp_loopback_set(struct hdsp *const hdsp, const u8 channel, const bool enable) +{ + if (hdsp_loopback_get(hdsp, channel) == enable) + return 0; + + hdsp->io_loopback ^= (1 << channel); + + hdsp_write(hdsp, HDSP_inputEnable + (4 * (hdsp->max_channels + channel)), enable); + + return 1; +} + +static int snd_hdsp_loopback_info(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_info *const uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_hdsp_loopback_get(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_value *const ucontrol) +{ + struct hdsp *const hdsp = snd_kcontrol_chip(kcontrol); + const u8 channel = kcontrol->id.index; + + if (channel >= hdsp->max_channels) + return -ENOENT; + + ucontrol->value.integer.value[0] = hdsp_loopback_get(hdsp, channel); + + return 0; +} + +static int snd_hdsp_loopback_put(struct snd_kcontrol *const kcontrol, + struct snd_ctl_elem_value *const ucontrol) +{ + struct hdsp *const hdsp = snd_kcontrol_chip(kcontrol); + const u8 channel = ucontrol->id.index; + const bool enable = ucontrol->value.integer.value[0] & 1; + + if (channel >= hdsp->max_channels) + return -ENOENT; + + return hdsp_loopback_set(hdsp, channel, enable); +} + +static struct snd_kcontrol_new snd_hdsp_loopback_control = { + .iface = SNDRV_CTL_ELEM_IFACE_HWDEP, + .name = "Output Loopback", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_hdsp_loopback_info, + .get = snd_hdsp_loopback_get, + .put = snd_hdsp_loopback_put +}; + static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) { unsigned int idx; @@ -3297,6 +3363,17 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) } } + /* Output loopback controls for H9632 cards */ + if (hdsp->io_type == H9632) { + for (idx = 0; idx < hdsp->max_channels; idx++) { + snd_hdsp_loopback_control.index = idx; + kctl = snd_ctl_new1(&snd_hdsp_loopback_control, hdsp); + err = snd_ctl_add(card, kctl); + if (err < 0) + return err; + } + } + /* AEB control for H96xx card */ if (hdsp->io_type == H9632 || hdsp->io_type == H9652) { if ((err = snd_ctl_add(card, kctl = snd_ctl_new1(&snd_hdsp_96xx_aeb, hdsp))) < 0) @@ -4956,7 +5033,7 @@ static int snd_hdsp_enable_io (struct hdsp *hdsp) static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { - int status, aebi_channels, aebo_channels; + int status, aebi_channels, aebo_channels, i; switch (hdsp->io_type) { case Digiface: @@ -4983,6 +5060,12 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_out_channels = H9632_SS_CHANNELS+aebo_channels; hdsp->ds_out_channels = H9632_DS_CHANNELS+aebo_channels; hdsp->qs_out_channels = H9632_QS_CHANNELS+aebo_channels; + /* Disable loopback of output channels, as the set function + * only sets on a change we fake all bits (channels) as enabled. + */ + hdsp->io_loopback = 0xffffffff; + for (i = 0; i < hdsp->max_channels; ++i) + hdsp_loopback_set(hdsp, i, false); break; case Multiface: -- 2.27.0

2 5

add one Advantech SSID in the ALC886
by Pshou 02 Feb '21

02 Feb '21

Hi Alsa-devel ML: Can you help me add some source code to patch_realtek.c? Add one Advantech SSID in the alc886. Best regards Pshou

2 1

[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

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[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

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