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

[PATCH AUTOSEL 5.10 06/25] ASoC: Intel: Skylake: Zero snd_ctl_elem_value
by Sasha Levin 02 Feb '21

02 Feb '21

From: Ricardo Ribalda <ribalda(a)chromium.org> [ Upstream commit 1d8fe0648e118fd495a2cb393a34eb8d428e7808 ] Clear struct snd_ctl_elem_value before calling ->put() to avoid any data leak. Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Link: https://lore.kernel.org/r/20210121171644.131059-2-ribalda@chromium.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 40bee10b0c65a..1bde9ce04de33 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3631,7 +3631,7 @@ static void skl_tplg_complete(struct snd_soc_component *component) sprintf(chan_text, "c%d", mach->mach_params.dmic_num); for (i = 0; i < se->items; i++) { - struct snd_ctl_elem_value val; + struct snd_ctl_elem_value val = {}; if (strstr(texts[i], chan_text)) { val.value.enumerated.item[0] = i; -- 2.27.0

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[PATCH AUTOSEL 5.10 03/25] ASoC: wm_adsp: Fix control name parsing for multi-fw
by Sasha Levin 02 Feb '21

02 Feb '21

From: James Schulman <james.schulman(a)cirrus.com> [ Upstream commit a8939f2e138e418c2b059056ff5b501eaf2eae54 ] When switching between firmware types, the wrong control can be selected when requesting control in kernel API. Use the currently selected DSP firwmare type to select the proper mixer control. Signed-off-by: James Schulman <james.schulman(a)cirrus.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20210115201105.14075-1-james.schulman@cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index dec8716aa8ef5..985b2dcecf138 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2031,11 +2031,14 @@ static struct wm_coeff_ctl *wm_adsp_get_ctl(struct wm_adsp *dsp, unsigned int alg) { struct wm_coeff_ctl *pos, *rslt = NULL; + const char *fw_txt = wm_adsp_fw_text[dsp->fw]; list_for_each_entry(pos, &dsp->ctl_list, list) { if (!pos->subname) continue; if (strncmp(pos->subname, name, pos->subname_len) == 0 && + strncmp(pos->fw_name, fw_txt, + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0 && pos->alg_region.alg == alg && pos->alg_region.type == type) { rslt = pos; -- 2.27.0

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hdspmixer: Hardware output loopback toggle buttons
by GitHub pull_request - edited 02 Feb '21

02 Feb '21

alsa-project/alsa-tools pull request #4 was edited 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

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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

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[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

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[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

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[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

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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

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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

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[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

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