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

Hardware output loopback toggle buttons
by GitHub pull_request - opened 02 Feb '21

02 Feb '21

alsa-project/alsa-tools pull request #4 was opened from pfandl: Hi, as promised in the sound/pci/rme9652 patch here are the user land changes which provide the user the ability to toggle the hardware output loopback functionality per output channel. I tried to hide the buttons for devices which do not expose the snd controls yet, but due to probably unnecessary struggle i cancelled that. Please let me know though should you want them removed for not yet enabled devices. Request URL : https://github.com/alsa-project/alsa-tools/pull/4 Patch URL : https://github.com/alsa-project/alsa-tools/pull/4.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH 1/2] ALSA: usb-audio: Add DJM450 to Pioneer format quirk
by Olivia Mackintosh 02 Feb '21

02 Feb '21

Like the DJM-750, ensure that the format control message is passed to the device when opening a stream. It seems as though fmt->sync_ep is not always set when this function is called hence the passing of the value at the call site. If this can be fixed, fmt->sync_up should be used as the wvalue. --- sound/usb/quirks.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e196e364cef1..08300954a50e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1470,6 +1470,23 @@ static void set_format_emu_quirk(struct snd_usb_substream *subs, subs->pkt_offset_adj = (emu_samplerate_id >= EMU_QUIRK_SR_176400HZ) ? 4 : 0; } +static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, + u16 windex) +{ + unsigned int cur_rate = subs->data_endpoint->cur_rate; + u8 sr[3]; + // Convert to little endian + sr[0] = cur_rate&0xff; + sr[1] = (cur_rate>>8)&0xff; + sr[2] = (cur_rate>>16)&0xff; + usb_set_interface(subs->dev, 0, 1); + // we should derive windex from fmt-sync_ep but it's not set + snd_usb_ctl_msg(subs->stream->chip->dev, + usb_rcvctrlpipe(subs->stream->chip->dev, 0), + 0x01, 0x22, 0x0100, windex, &sr, 0x0003); + return 0; +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1483,6 +1500,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ + pioneer_djm_set_format_quirk(subs, 0x0082); + break; } } -- 2.30.0

2 2

[PATCH v9 0/1] audio jack software injection
by Hui Wang 02 Feb '21

02 Feb '21

the changes in the v9: - move the "Phantom" jack comparison into the snd_jack_debugfs_add_inject_node(), then the comparison is only executed when SND_JACK_INJECTION_DEBUG is enabled. the changes in the v8: - change "for (i = 0; i < strlen(tname); i++)" to "for (i = 0; tname[i]; i++)" - use CONFIG_SND_DEBUG to wrap some debugfs variable and struct members. the changes in the v7: - change the format of the last part in jack-injection.rst - add dependence SND_DEBUG in the Kconfig - create debugfs_mount_dir/sound and debugfs_mount_dir/sound/cardN only SND_DEBUG is enabled - change simple_write_to_buffer(buf, count, ppos, from, count) to simple_write_to_buffer(buf, sizeof(buf) - 1, ppos, from, count) - rewrite the function parse_mask_bits() according to v6's comment the changes in the v6: - use the sizeof(buf) to replace the digits in the scnprintf - squash the document patch into the 0001-xxx.patch. the changes in the v5: - add a Kconfig to enable/disable the jack injection feature - replace all kzalloc with local char[] in the debugfs fops - replace the chars of !isalnum() to '_' for the jack folder's name - squash all .c files patches into one patch - add a document to explain jack injection, for easier review, put the document in a standalone patch. If needed, will squash this patch to the previous one. the changes in the v4: - change the sound-core to sound and change the sound_core_debugfs_root to sound_debugfs_root in the 0001-xxx.patch - change kzalloc/kfree to kvzalloc/kvfree in the debugfs fops for 0001-xxx.patch and 0003-xxx.patch - And if needed, I could squash 4 patches into 1 patch before merging. the changes in the v3 (for easy to review, divide change into 4 patches): - address the comment about the snd_jack_report() and _snd_jack_report(), the v2 design is hard to understand and is hard to review, in the v3, separate the jack_report to snd_jack_report() and snd_jack_inject_report(), hw jack events call snd_jack_report() as before, if a jack contains multi jack_kctl and the jack_kctl's sw_inject is enabled, the status and the related input-dev's events will not be reproted. The injection events call snd_jack_inject_report(). This change is squashed to 0001-xxx.patch, the rest part of 0001-xxx.patch is same as v2. - address the comment about folders'name in the 0002-xxx.patch, so far, drop the '/', ',', '=' and ' ' from the folders' name. - address the comment about adding more debugfs nodes in the 0003-xxx.patch, it adds kctl_id, mask_bits, status and type. - address the comment about save-n-restore jack's hw status in the 0004-xxx.patch, adding a hw_status_cache and save the last reported jack hw event, once the sw_inject is disabled, will restore all jack_kctl's state under the same snd_jack with hw_status_cache. [snip] the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls [ snip ] Hui Wang (1): alsa: jack: implement software jack injection via debugfs Documentation/sound/designs/index.rst | 1 + .../sound/designs/jack-injection.rst | 166 ++++++++++ include/sound/core.h | 6 + include/sound/jack.h | 1 + sound/core/Kconfig | 9 + sound/core/init.c | 16 + sound/core/jack.c | 304 +++++++++++++++++- sound/core/sound.c | 13 + 8 files changed, 512 insertions(+), 4 deletions(-) create mode 100644 Documentation/sound/designs/jack-injection.rst -- 2.25.1

2 3

[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

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[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

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