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

Re: [GIT PULL] sound updates for 5.14-rc1
by Linus Torvalds 03 Jul '21

03 Jul '21

On Fri, Jul 2, 2021 at 9:37 PM Linus Torvalds <torvalds(a)linux-foundation.org> wrote: > > But I thought I'd report this as a likely candidate. Confirmed. The watchdog hang bisects right down to commit 9ce650a75a3b ("ALSA: usb-audio: Reduce latency at playback start"). And reverting it on top of my tree also fixes the hang, so it's not some bisection fluke. I have no idea what is actually wrong with that commit, but it most definitely is the problem, and I have reverted it in my tree so that I can continue merging stuff tomorrow. Linus

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[PATCH] ALSA: hda: Enable runtime PM when codec probe fails
by Kai-Heng Feng 02 Jul '21

02 Jul '21

When codec probe fails, it doesn't enable runtime suspend, and can prevent graphics card from getting powered down: [ 4.280991] snd_hda_intel 0000:01:00.1: no codecs initialized $ cat /sys/bus/pci/devices/0000:01:00.1/power/runtime_status active So enable runtime PM when codec probe fails, to let graphics card be able to runtime suspend again. Merge azx_probe_continue() into azx_probe() and just let probe fail for this case could be a better approach. However that's a much bigger task so let's settle with a quirk workaround. BugLink: https://bugs.launchpad.net/bugs/1907212 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6852668f1bcb..3fd920069268 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2328,7 +2328,7 @@ static int azx_probe_continue(struct azx *chip) if (bus->codec_mask) { err = azx_probe_codecs(chip, azx_max_codecs[chip->driver_type]); if (err < 0) - goto out_free; + goto out_enable_rpm; } #ifdef CONFIG_SND_HDA_PATCH_LOADER @@ -2360,6 +2360,7 @@ static int azx_probe_continue(struct azx *chip) set_default_power_save(chip); +out_enable_rpm: if (azx_has_pm_runtime(chip)) { pm_runtime_use_autosuspend(&pci->dev); pm_runtime_allow(&pci->dev); -- 2.29.2

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timestamp
by folkert 02 Jul '21

02 Jul '21

Hi, Please consider including the following patch. This patch adds a '-T' flag which prepends MIDI messages outputted by '--dump' with a timestamp. E.g.: 1625245188.993861810) C964 1625245188.994318675) F8 1625245188.995468101) 99647F 1625245189.494490183) F8 1625245189.495448424) 99647F 1625245189.995470260) F8 1625245189.996461381) 99647F 1625245190.494519924) F8 [...] diff -uNr t3/alsa-utils-1.2.3/amidi/amidi.1 alsa-utils-1.2.3/amidi/amidi.1 --- t3/alsa-utils-1.2.3/amidi/amidi.1 2020-06-07 19:19:42.000000000 +0200 +++ alsa-utils-1.2.3/amidi/amidi.1 2021-07-02 14:35:25.416330576 +0200 @@ -6,6 +6,7 @@ .SH SYNOPSIS \fBamidi\fP [\fI\-p port\fP] [\fI\-s file\fP | \fI\-S data\fP] [\fI\-r file\fP] [\fI\-d\fP] [\fI\-t seconds\fP] [\fI\-a\fP] +[\fI\-T\fP] .SH DESCRIPTION .B amidi @@ -41,6 +42,10 @@ options to specify what data to send or receive. .TP +.I \-T +Adds a timestamp in front of each MIDI message. + +.TP .I \-h, \-\-help Help: prints a list of options. diff -uNr t3/alsa-utils-1.2.3/amidi/amidi.c alsa-utils-1.2.3/amidi/amidi.c --- t3/alsa-utils-1.2.3/amidi/amidi.c 2020-06-07 19:19:42.000000000 +0200 +++ alsa-utils-1.2.3/amidi/amidi.c 2021-07-02 19:41:34.476470438 +0200 @@ -36,6 +36,7 @@ #include <unistd.h> #include <fcntl.h> #include <alsa/asoundlib.h> +#include <time.h> #include "aconfig.h" #include "version.h" @@ -80,6 +81,7 @@ "-r, --receive=file write received data into a file\n" "-S, --send-hex=\"...\" send hexadecimal bytes\n" "-d, --dump print received data as hexadecimal bytes\n" + "-T, --timestamp adds a timestamp in front of each dumped message\n" "-t, --timeout=seconds exits when no data has been received\n" " for the specified duration\n" "-a, --active-sensing include active sensing bytes\n" @@ -356,7 +358,7 @@ /* * prints MIDI commands, formatting them nicely */ -static void print_byte(unsigned char byte) +static void print_byte(unsigned char byte, struct timespec *ts) { static enum { STATE_UNKNOWN, @@ -426,7 +428,18 @@ if (running_status) fputs("\n ", stdout); } - printf("%c%02X", newline ? '\n' : ' ', byte); + + if (newline) { + printf("\n"); + + /* Nanoseconds does not make a lot of sense for serial MIDI (the + * 31250 bps one) but I'm not sure about MIDI over USB. + */ + if (ts) + printf("%lld.%.9ld) ", (long long)ts->tv_sec, ts->tv_nsec); + } + + printf("%02X", byte); } static void sig_handler(int dummy) @@ -454,7 +467,7 @@ int main(int argc, char *argv[]) { - static const char short_options[] = "hVlLp:s:r:S::dt:aci:"; + static const char short_options[] = "hVlLp:s:r:S::dt:aci:T"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, @@ -465,6 +478,7 @@ {"receive", 1, NULL, 'r'}, {"send-hex", 2, NULL, 'S'}, {"dump", 0, NULL, 'd'}, + {"timestamp", 0, NULL, 'T'}, {"timeout", 1, NULL, 't'}, {"active-sensing", 0, NULL, 'a'}, {"clock", 0, NULL, 'c'}, @@ -475,6 +489,7 @@ int ignore_active_sensing = 1; int ignore_clock = 1; int do_send_hex = 0; + int do_print_timestamp = 0; struct itimerspec itimerspec = { .it_interval = { 0, 0 } }; while ((c = getopt_long(argc, argv, short_options, @@ -509,6 +524,9 @@ case 'd': dump = 1; break; + case 'T': + do_print_timestamp = 1; + break; case 't': if (optarg) timeout = atof(optarg); @@ -611,6 +629,7 @@ } if (inputp) { + int need_timestamp = 0; int read = 0; int npfds; struct pollfd *pfds; @@ -648,6 +667,7 @@ unsigned char buf[256]; int i, length; unsigned short revents; + struct timespec ts; err = poll(pfds, npfds, -1); if (stop || (err < 0 && errno == EINTR)) @@ -657,6 +677,11 @@ break; } + if (clock_gettime(CLOCK_REALTIME, &ts) < 0) { + error("clock_getres failed: %s", strerror(errno)); + break; + } + err = snd_rawmidi_poll_descriptors_revents(input, &pfds[1], npfds - 1, &revents); if (err < 0) { error("cannot get poll events: %s", snd_strerror(errno)); @@ -692,7 +717,8 @@ write(receive_file, buf, length); if (dump) { for (i = 0; i < length; ++i) - print_byte(buf[i]); + print_byte(buf[i], do_print_timestamp ? &ts : NULL); + fflush(stdout); } diff -uNr t3/alsa-utils-1.2.3/configure.ac alsa-utils-1.2.3/configure.ac --- t3/alsa-utils-1.2.3/configure.ac 2021-07-02 19:41:46.000000000 +0200 +++ alsa-utils-1.2.3/configure.ac 2021-07-02 15:55:13.245585248 +0200 @@ -97,7 +97,7 @@ if test x$bat = xtrue; then saved_CFLAGS="$CFLAGS" - saved_LDFLAGS="$LDFLAGS" + saved_LDFLAGS="$LDFLAGS -lrt" saved_LIBS="$LIBS" FFTW_INC="" FFTW_LIB="" @@ -116,7 +116,7 @@ FFTW_CFLAGS="$CFLAGS" FFTW_LIB="$LIBS" CFLAGS="$saved_CFLAGS" - LDFLAGS="$saved_LDFLAGS" + LDFLAGS="$saved_LDFLAGS -lrt" LIBS="$saved_LIBS" AC_SUBST(FFTW_INC) AC_SUBST(FFTW_LIB) @@ -292,11 +292,11 @@ fi saved_CFLAGS="$CFLAGS" - saved_LDFLAGS="$LDFLAGS" + saved_LDFLAGS="$LDFLAGS -lrt" saved_LIBS="$LIBS" CFLAGS="$CFLAGS $CURSES_CFLAGS" if test -n "$CURSESLIBDIR"; then - LDFLAGS="$LDFLAGS -L$CURSESLIBDIR" + LDFLAGS="$LDFLAGS -L$CURSESLIBDIR -lrt" fi LIBS="$CURSESLIB $LIBS" @@ -342,7 +342,7 @@ [AC_MSG_ERROR([form$NCURSESLIBSUFFIX library not found])]) CFLAGS="$saved_CFLAGS" - LDFLAGS="$saved_LDFLAGS" + LDFLAGS="$saved_LDFLAGS -lrt" LIBS="$saved_LIBS" if test -n "$CURSESLIBDIR"; then

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[tiwai/for-next] hard lockups with pipewire and snd-usb-audio
by Hector Martin 02 Jul '21

02 Jul '21

Hi Takashi, I've been testing builds from your for-next branch to get the recent firewire fixes. All that works fine, but instead I've been getting hard lockups with pipewire and snd-usb-audio. I'm currently on 9a7b7ec3c6. Backtrace: do_raw_spin_lock+0x8e/0x90 _raw_spin_lock_irqsave+0x3e/0x50 ? snd_pcm_period_elapsed+0x10/0x30 [snd_pcm] snd_pcm_period_elapsed+0x10/0x30 [snd_pcm] snd_usb_endpoint_start+0xd8/0x270 [snd_usb_audio] start_endpoints+0x50/0x70 [snd_usb_audio] snd_usb_substream_playback_trigger+0x58/0x100 [snd_usb_audio] snd_pcm_action+0x74/0xb0 [snd_pcm] snd_pcm_action_lock_irq+0x36/0x80 [snd_pcm] snd_pcm_ioctl+0x1e/0x30 [snd_pcm] __x64_sys_ioctl+0x7d/0xb0 do_syscall_64+0x45/0x80 entry_SYSCALL_64_after_hwframe+0x44/0xae To me it looks like it's recursively trying to lock the stream spinlock. I'm not sure what version this is a regression from; I've only triggered it with pipewire so far, which I wasn't previously using. It's easy to reproduce. Usually, pulling up pavucontrol on pipewire with the USB audio device on an active profile will do it. Just thought you might want to know. Let me know if you need any more info. -- Hector Martin (marcan(a)marcan.st) Public Key: https://mrcn.st/pub

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[PATCH 0/3] ALSA: firewire: media clock recovery for syt-aware devices
by Takashi Sakamoto 02 Jul '21

02 Jul '21

Hi, In a commit f9e5ecdfc2c2 ("ALSA: firewire-lib: add replay target to cache sequence of packet"), I categorize devices supported by drivers in ALSA firewire stack in terms of the way to deliver effective sampling transfer frequency. This patchset is for the devices in group 2. The devices are known to have problems when ALSA dice/bebob drivers handle. Many of them sometimes transfer packets with discontinued counter, corrupt at break of CMP connection, generates bus-reset voluntarily. The devices interpret presentation time to decide playback timing. The drivers process presentation time expressed in syt field of CIP header for outgoing packets. Current implementation of the drivers processes the sequence of outgoing packet by computation according to nominal sampling transfer frequency, assisted by ALSA IEC 61883-1/6 packet streaming engine. However, the ideal sequence is not adequate to the devices, actually. With this patchset, the drivers are going to replay the sequence of incoming packets for media clock recovery. For the detail of sequence replay, please refer to a commit 39c2649c71d8 ("ALSA: firewire-lib: replay sequence of incoming packets for outgoing packets"). Takashi Sakamoto (3): ALSA: dice: wait just for NOTIFY_CLOCK_ACCEPTED after GLOBAL_CLOCK_SELECT operation ALSA: dice: perform sequence replay for media clock recovery ALSA: bebob: perform sequence replay for media clock recovery sound/firewire/bebob/bebob_stream.c | 13 +++++++--- sound/firewire/dice/dice-stream.c | 35 +++++++++----------------- sound/firewire/dice/dice-transaction.c | 2 +- 3 files changed, 22 insertions(+), 28 deletions(-) -- 2.27.0

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[PATCH] ALSA: usb-audio: scarlett2: Fix for loop increment in scarlett2_usb_get_config
by Nathan Chancellor 01 Jul '21

01 Jul '21

Clang warns: sound/usb/mixer_scarlett_gen2.c:1189:32: warning: expression result unused [-Wunused-value] for (i = 0; i < count; i++, (u16 *)buf++) ^ ~~~~~ 1 warning generated. It appears the intention was to cast the void pointer to a u16 pointer so that the data could be iterated through like an array of u16 values. However, the cast happens after the increment because a cast is an rvalue, whereas the post-increment operator only works on lvalues, so the loop does not iterate as expected. Replace the post-increment shorthand with the full expression so the cast can be added in the right place and the look works as expected. Fixes: ac34df733d2d ("ALSA: usb-audio: scarlett2: Update get_config to do endian conversion") Link: https://github.com/ClangBuiltLinux/linux/issues/1408 Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/usb/mixer_scarlett_gen2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index fcba682cd422..c20c7f1ddc50 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1186,7 +1186,7 @@ static int scarlett2_usb_get_config( if (err < 0) return err; if (size == 2) - for (i = 0; i < count; i++, (u16 *)buf++) + for (i = 0; i < count; i++, buf = (u16 *)buf + 1) *(u16 *)buf = le16_to_cpu(*(__le16 *)buf); return 0; } base-commit: 5c89c2c7fbfa9124dd521c375b9c82b9ed75bc28 -- 2.32.0.93.g670b81a890

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Add ALC285 Headphone initial procedure
by Kailang 01 Jul '21

01 Jul '21

Hi Takashi, Attach patch was for ALC285 HP init. BR, Kailang

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[PATCH v2 1/3] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 01 Jul '21

01 Jul '21

The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 308a58a83f33..6c423172a1f4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8363,6 +8363,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

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[PATCH] ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits
by Kyle Russell 01 Jul '21

01 Jul '21

These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 81952984613d..2513922a0292 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- 2.25.1

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[PATCH 0/2] arm64: tegra: Enable audio IOMMU support on Tegra194
by Thierry Reding 01 Jul '21

01 Jul '21

From: Thierry Reding <treding(a)nvidia.com> This small series addresses a minor issue with how IOMMU support is wired up on various Tegra generations. Currently the virtual "card" device is used to allocate DMA memory for, but since that device does not actually exist, the path to memory cannot be correctly described. To address this, this series moves to using the ADMAIF as the DMA device for audio. This is a real device that can have a proper DMA mask set and with which a stream ID can be associated with in the SMMU. The memory accesses technically originate from the ADMA controller (that the ADMAIF uses), but DMA channel are dynamically allocated at runtime while DMA memory is allocated at driver load time, drivers won't have access to the ADMA device yet. Further patches will be required to correct this issue on Tegra186 and Tegra210, but I wanted to get feedback on this approach first. Thierry Thierry Reding (2): ASoC: tegra: Use ADMAIF component for DMA allocations arm64: tegra: Enable audio IOMMU support on Tegra194 arch/arm64/boot/dts/nvidia/tegra194.dtsi | 4 ++++ sound/soc/tegra/tegra_pcm.c | 21 +++++++++------------ 2 files changed, 13 insertions(+), 12 deletions(-) -- 2.32.0

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