May 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: cs5535audio: fix typo in comment
by Julia Lawall 23 May '22

23 May '22

Spelling mistake (triple letters) in comment. Detected with the help of Coccinelle. Signed-off-by: Julia Lawall <Julia.Lawall(a)inria.fr> --- sound/pci/cs5535audio/cs5535audio_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 5ff10fec7b90..0db24cc4d916 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -129,7 +129,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, return 0; /* the u32 cast is okay because in snd*create we successfully told - pci alloc that we're only 32 bit capable so the uppper will be 0 */ + pci alloc that we're only 32 bit capable so the upper will be 0 */ addr = (u32) substream->runtime->dma_addr; desc_addr = (u32) dma->desc_buf.addr; for (i = 0; i < periods; i++) {

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[PATCH] ALSA: ctxfi: Add SB046x PCI ID
by Edward Matijevic 23 May '22

23 May '22

Adds the PCI ID for X-Fi cards sold under the Platnum and XtremeMusic names Before: snd_ctxfi 0000:05:05.0: chip 20K1 model Unknown (1102:0021) is found After: snd_ctxfi 0000:05:05.0: chip 20K1 model SB046x (1102:0021) is found Signed-off-by: Edward Matijevic <motolav(a)gmail.com> --- sound/pci/ctxfi/ctatc.c | 2 ++ sound/pci/ctxfi/cthardware.h | 3 ++- 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 78f35e88aed6..fbdb8a3d5b8e 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -36,6 +36,7 @@ | ((IEC958_AES3_CON_FS_48000) << 24)) static const struct snd_pci_quirk subsys_20k1_list[] = { + SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0021, "SB046x", CTSB046X), SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0022, "SB055x", CTSB055X), SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x002f, "SB055x", CTSB055X), SND_PCI_QUIRK(PCI_VENDOR_ID_CREATIVE, 0x0029, "SB073x", CTSB073X), @@ -64,6 +65,7 @@ static const struct snd_pci_quirk subsys_20k2_list[] = { static const char *ct_subsys_name[NUM_CTCARDS] = { /* 20k1 models */ + [CTSB046X] = "SB046x", [CTSB055X] = "SB055x", [CTSB073X] = "SB073x", [CTUAA] = "UAA", diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h index f406b626a28c..2875cec83b8f 100644 --- a/sound/pci/ctxfi/cthardware.h +++ b/sound/pci/ctxfi/cthardware.h @@ -26,8 +26,9 @@ enum CHIPTYP { enum CTCARDS { /* 20k1 models */ + CTSB046X, + CT20K1_MODEL_FIRST = CTSB046X, CTSB055X, - CT20K1_MODEL_FIRST = CTSB055X, CTSB073X, CTUAA, CT20K1_UNKNOWN, -- 2.36.1

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Is it possible to set the CM6533 so it has 96 kHz sample rate on capture and 48 kHz for playback using ALSA?
by Rob Frohne 22 May '22

22 May '22

Hi All, I found this thread <https://www.alsa-project.org/pipermail/alsa-devel/2017-November/127289.html> on the list that explains why I cannot get 96 kHz sampling with the C-media CM6533. Reading the datasheet <cid:part1.SZL6HGoZ.u1f9OjPs@wallawalla.edu> it is tempting to think the mode I need for my class QSD SDR receiver project, (96 kHz sample rate on capture and 48 kHz for playback could be selected perhaps by ALSA in the HID interface). Is this possible to do with ALSA somehow? The sound cards I purchased for my class on Amazon from Syba, used to use Via VT1620a codecs, but now have been redesigned to use the C-media CM6533, and these have the limitation that will not allow 96 kHz sampling on both capture and playback. I need to change the present setting (96 kHz playback, 48 kHz capture) to the opposite. Page 7 of the datasheet has a note about this. Thanks, Rob -- Rob Frohne, PhD, PE Professor EF Cross School of Engineering Walla Walla University 100 SW 4th Street College Place, WA 99324 (509) 527-2075

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Is it possible to set the CM6533 so it has 96 kHz sample rate on capture and 48 kHz for playback using ALSA?
by Rob Frohne 22 May '22

22 May '22

Hi All, I found this thread <https://www.alsa-project.org/pipermail/alsa-devel/2017-November/127289.html> on the list that explains why I cannot get 96 kHz sampling with the C-media CM6533. Reading the datasheet <cid:part1.yl7hLKYg.W0fZhG6J@wallawalla.edu> it is tempting to think the mode I need for my class QSD SDR receiver project, (96 kHz sample rate on capture and 48 kHz for playback could be selected perhaps by ALSA in the HID interface). Is this possible to do with ALSA somehow? The sound cards I purchased for my class on Amazon from Syba, used to use Via VT1620a codecs, but now have been redesigned to use the C-media CM6533, and these have the limitation that will not allow 96 kHz sampling on both capture and playback. I need to change the present setting (96 kHz playback, 48 kHz capture) to the opposite. Page 7 of the datasheet has a note about this. Thanks, Rob -- Rob Frohne, PhD, PE Professor EF Cross School of Engineering Walla Walla University 100 SW 4th Street College Place, WA 99324 (509) 527-2075

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[PATCH] ALSA: usb-audio: Add missing ep_idx in fixed EP quirks
by Takashi Iwai 21 May '22

21 May '22

The quirk entry for Focusrite Saffire 6 had no proper ep_idx for the capture endpoint, and this confused the driver, resulting in the broken sound. This patch adds the missing ep_idx in the entry. While we are at it, a couple of other entries (for Digidesign MBox and MOTU MicroBook II) seem to have the same problem, and those are covered as well. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Reported-by: André Kapelrud <a.kapelrud(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 40a5e3eb4ef2..78eb41b621d6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2672,6 +2672,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .altset_idx = 1, .attributes = 0, .endpoint = 0x82, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC, .datainterval = 1, .maxpacksize = 0x0126, @@ -2875,6 +2876,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .altset_idx = 1, .attributes = 0x4, .endpoint = 0x81, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, .maxpacksize = 0x130, @@ -3391,6 +3393,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .altset_idx = 1, .attributes = 0, .endpoint = 0x03, + .ep_idx = 1, .rates = SNDRV_PCM_RATE_96000, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, -- 2.35.3

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[PATCH] ALSA: lola: Bounds check loop iterator against streams array size
by Kees Cook 21 May '22

21 May '22

GCC 12 sees that it's technically possible for num_streams to be larger than ARRAY_SIZE(pcm->streams). Bounds-check the iterator. ../sound/pci/lola/lola_pcm.c: In function 'lola_pcm_update': ../sound/pci/lola/lola_pcm.c:567:64: warning: array subscript [0, 31] is outside array bounds of 'struct lola_stream[16]' [-Warray-bounds] 567 | struct lola_stream *str = &pcm->streams[i]; | ~~~~~~~~~~~~^~~ In file included from ../sound/pci/lola/lola_pcm.c:15: ../sound/pci/lola/lola.h:307:28: note: while referencing 'streams' 307 | struct lola_stream streams[MAX_STREAM_COUNT]; | ^~~~~~~ Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Kees Cook <keescook(a)chromium.org> --- sound/pci/lola/lola_pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 738ec987000a..32193fae978d 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -561,8 +561,9 @@ static snd_pcm_uframes_t lola_pcm_pointer(struct snd_pcm_substream *substream) void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits) { int i; + u8 num_streams = min_t(u8, pcm->num_streams, ARRAY_SIZE(pcm->streams)); - for (i = 0; bits && i < pcm->num_streams; i++) { + for (i = 0; bits && i < num_streams; i++) { if (bits & (1 << i)) { struct lola_stream *str = &pcm->streams[i]; if (str->substream && str->running) -- 2.32.0

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[GIT PULL] ASoC fixes for v5.18-rc7
by Mark Brown 21 May '22

21 May '22

The following changes since commit 5e02fb590e83684f63217f93a9cdeabd6a925f9c: ASoC: cs35l41: Fix DSP mbox start command and global enable order (2022-03-07 13:12:38 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.18-rc7 for you to fetch changes up to 353bb6a5f2ac495f289b7c7a528c7d134c9a8ec4: ASoC: wm_adsp: Compressed stream DSP memory structs should be __packed (2022-03-09 17:50:26 +0000) ---------------------------------------------------------------- ASoC: Fix for v5.17 This is rather late and at this point I'm expecting it to get merged in the merge window rather than as a fix but if we get a -rc8 it's a small, driver specific fix which should be fine to send. ---------------------------------------------------------------- Simon Trimmer (1): ASoC: wm_adsp: Compressed stream DSP memory structs should be __packed sound/soc/codecs/wm_adsp.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-)

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[PATCH 0/6] OPE support on Tegra210 and later
by Sameer Pujar 21 May '22

21 May '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 51 + .../bindings/sound/nvidia,tegra210-ope.yaml | 91 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 52 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2920 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

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[PATCH] ALSA: usb-audio: Workaround for clock setup on TEAC devices
by Takashi Iwai 21 May '22

21 May '22

Maris reported that TEAC UD-501 (0644:8043) doesn't work with the typical "clock source 41 is not valid, cannot use" errors on the recent kernels. The currently known workaround so far is to restore (partially) what we've done unconditionally at the clock setup; namely, re-setup the USB interface immediately after the clock is changed. This patch re-introduces the behavior conditionally for TEAC devices. Further notes: - The USB interface shall be set later in snd_usb_endpoint_configure(), but this seems to be too late. - Even calling usb_set_interface() right after sne_usb_init_sample_rate() doesn't help; so this must be related with the clock validation, too. - The device may still spew the "clock source 41 is not valid" error at the first clock setup. This seems happening at the very first try of clock setup, but it disappears at later attempts. The error is likely harmless because the driver retries the clock setup (such an error is more or less expected on some devices). Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Reported-and-tested-by: Maris Abele <maris7abele(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 4dfe76416794..3c435d379306 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -572,6 +572,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, /* continue processing */ } + /* FIXME - TEAC devices require the immediate interface setup */ + if (rate != prev_rate && USB_ID_VENDOR(chip->usb_id) == 0x0644) { + usb_set_interface(chip->dev, fmt->iface, fmt->altsetting); + if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY) + msleep(50); + } + validation: /* validate clock after rate change */ if (!uac_clock_source_is_valid(chip, fmt, clock)) -- 2.35.3

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Complete support for "Pro" USB multichannel cards
by GitHub issues - edited 20 May '22

20 May '22

alsa-project/alsa-ucm-conf issue #154 was edited from perexg: - [x] channel split using alsa-lib - [x] channel split inside app - [x] channel split hide the specific configs (alsa-lib, app side support) - [x] mixer control remap using alsa-lib - [x] simplify the mixer control remap Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/154 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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