June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: firewire-motu: wait for notification when changing clock configuration for protocol v3
by Takashi Sakamoto 14 Jun '20

14 Jun '20

This patch is for development period of v5.9 kernel. It costs expensive to change clock configuration for models of protocol version 3. In current implementation, speculative strategy is used to finish the operation; just waiting for 4 seconds. As long as I investigate, when accepting and changing clock status actually, the device sends notification with mask. This commit uses wise way to wait for the notification after changing sampling clock rate during 4 seconds. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 01a47ac7bb2d..4e6b0e449ee4 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -24,6 +24,9 @@ #define V3_NO_ADAT_OPT_OUT_IFACE_A 0x00040000 #define V3_NO_ADAT_OPT_OUT_IFACE_B 0x00400000 +#define V3_MSG_FLAG_CLK_CHANGED 0x00000002 +#define V3_CLK_WAIT_MSEC 4000 + int snd_motu_protocol_v3_get_clock_rate(struct snd_motu *motu, unsigned int *rate) { @@ -79,9 +82,16 @@ int snd_motu_protocol_v3_set_clock_rate(struct snd_motu *motu, return err; if (need_to_wait) { - /* Cost expensive. */ - if (msleep_interruptible(4000) > 0) - return -EINTR; + int result; + + motu->msg = 0; + result = wait_event_interruptible_timeout(motu->hwdep_wait, + motu->msg & V3_MSG_FLAG_CLK_CHANGED, + msecs_to_jiffies(V3_CLK_WAIT_MSEC)); + if (result < 0) + return result; + if (result == 0) + return -ETIMEDOUT; } return 0; -- 2.25.1

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alsa-gobject: add Python 3 sample scripts
by GitHub pull_request - opened 14 Jun '20

14 Jun '20

alsa-project/alsa-gobject pull request #37 was opened from takaswie: PyGObject is a loader for GObject Introspection in Python 3 runtime. This patchset adds Python 3 sample scripts to use ALSACtl, ALSAHwdep, ALSARawmidi, ALSATimer, and ALSASeq loaded by PyGObject. Request URL : https://github.com/alsa-project/alsa-gobject/pull/37 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/37.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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alsa-gobject: seq: fix port creation API
by GitHub pull_request - opened 14 Jun '20

14 Jun '20

alsa-project/alsa-gobject pull request #36 was opened from takaswie: ALSASeq.UserClient.create_port() has nullable argument for the numerical ID of port to use SNDRV_SEQ_PORT_FLG_GIVEN_PORT flag properly. However, the nullable is not inconvenient for PyGObject and causes SIGSEGV. This patchset adds an alternative API, ALSASeq.UserClient.create_port_at() for the flag, then ALSASeq.UserClient.create_port() hs no argument for explicit port ID. Additionally, this patchset includes some fixes for the other parts. Request URL : https://github.com/alsa-project/alsa-gobject/pull/36 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/36.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH] ALSA: usb-audio: add quirk for Denon DCD-1500RE
by Yick W. Tse 14 Jun '20

14 Jun '20

fix error "clock source 41 is not valid, cannot use" [] New USB device found, idVendor=154e, idProduct=1002, bcdDevice= 1.00 [] New USB device strings: Mfr=1, Product=2, SerialNumber=0 [] Product: DCD-1500RE [] Manufacturer: D & M Holdings Inc. [] [] clock source 41 is not valid, cannot use [] usbcore: registered new interface driver snd-usb-audio Signed-off-by: Yick W. Tse <y_w_tse(a)yahoo.com.hk> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index bca0179a0ef8..c495e720e2f1 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1532,6 +1532,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) static bool is_itf_usb_dsd_dac(unsigned int id) { switch (id) { + case USB_ID(0x154e, 0x1002): /* Denon DCD-1500RE */ case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ --

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Cannot get Lenovo P520/Realtek ALC662 (rev 3) to produce sound
by Jan Pokorny 14 Jun '20

14 Jun '20

Hello list, this time within wider circle (and with more experience so far). I am seeking for help with shiny new Lenovo P520 machine that only lacks one thing for me to be entirely happy it. I believe this is relatively a new batch -- intentionally mentioning it since the topic of this very machine appears to be a settled case already: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… With newest BIOS, running the most recent kernel from Fedora Rawhide (but I did try 5.5.10 kernel from Fedora 30 as well), booted via UEFI (incl. secure boot), the only way to get it to produce any sound whatsoever is to manually load snd-pcsp [*1]. With that done, virtual terminal starts to beep as expected (as in printf '\a'), I am able to run "beep" program, and even "aplay -D hw:pcsp /usr/share/sound/alsa/*.wav" is producing the respective sounds, despite suffering from surrounding noise[*]. That validates that basic function of the built-in speaker works just fine (and previously, it was validated under Windows that it worked just fine). As the linked commit indicates, the proper model parameter to snd-hda-audio module is dual-codecs. I've tried that, and a handful of other models, to no avail. I considered the possibility the problem is just that I cannot work with "mute" state properly, so tried everything I could (incl. alsaunmute), again, without any change. Next dimension to the matrix was attaching external speaker to the front connector. Nothing has helped. Comparing outputs from alsa-info.sh for both states, "no model specified" and "dual-codecs" specified, there was no observable change other than this sole "model" parameter not/being (null). I am not sure whether this means this particular predefined quirk was applied properly, or the same observation is to be expected in general. Note that said quirk is, IIUIC, normally hooked onto the presence of "Subsystem: Lenovo Device 1036" (as in from lspci), and it is what this machine assuredly satisfies. I am attaching the output from the plain/default case, and looking forward to being able to move at least a bit with this. I'd be fine, for the time being, to use the sketched "PC speaker" workaround, if only it wasn't spoiled with so much noise/hum (no audio connector was used at that time). But that won't solve the urging general teleconferencing problem of these days, apparently. [*1] at a matter of fact, even it should play no role here, there was a change between this 5.7.0.rc7 kernel and the other tested, 5.5.10, in that the respective (another, IIRC) module got loaded automatically in the latter/older, whereas it needed to be loaded manually with the former/newer [*2] perhaps resembled https://bugzilla.redhat.com/show_bug.cgi?id=1660581 that I found when trying to do an extensive search around similar problems Thanks -- Jan

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dlmisc.c: fix uclibc build
by GitHub pull_request - opened 13 Jun '20

13 Jun '20

alsa-project/alsa-lib pull request #58 was opened from bkuhls: RTLD_DL_LINKMAP & RTLD_DI_ORIGIN, are unsupported on uClibc: https://cgit.uclibc-ng.org/cgi/cgit/uclibc-ng.git/tree/include/dlfcn.h#n106 This patch adds detection for uClibc because uClibc also defines __GLIBC__: https://lists.gnu.org/archive/html/bug-gnulib/2010-11/msg00280.html Request URL : https://github.com/alsa-project/alsa-lib/pull/58 Patch URL : https://github.com/alsa-project/alsa-lib/pull/58.patch Repository URL: https://github.com/alsa-project/alsa-lib

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alsa-gobject: obsolete function with two arguments
by GitHub pull_request - opened 13 Jun '20

13 Jun '20

alsa-project/alsa-gobject pull request #35 was opened from takaswie: According to guideline of GObject Introspection, API with multiple out parameters is not preferrable[1]. This patchset replaces such functions with more suitable ones. [1] https://gi.readthedocs.io/en/latest/writingbindableapis.html#multiple-out-p… Request URL : https://github.com/alsa-project/alsa-gobject/pull/35 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/35.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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alsa-gobject: misc fixes and documentation for system calls
by GitHub pull_request - opened 13 Jun '20

13 Jun '20

alsa-project/alsa-gobject pull request #34 was opened from takaswie: This patchset includes misc fixes and documentation for system calls for each methods. Request URL : https://github.com/alsa-project/alsa-gobject/pull/34 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/34.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH] ASoC: fsl_ssi: Fix bclk calculation for mono channel
by Shengjiu Wang 12 Jun '20

12 Jun '20

For mono channel, ssi will switch to normal mode. In normal mode, the Word Length Control bits control the word length divider in clock generator, which is different with I2S master mode, the word length is fixed to 32bit. So we refine the famula for mono channel, otherwise there will be sound issue for S24_LE. Fixes: b0a7043d5c2c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bad89b0d129e..e347776590f7 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -695,6 +695,11 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, /* Generate bit clock based on the slot number and slot width */ freq = slots * slot_width * params_rate(hw_params); + /* The slot_width is not fixed to 32 for normal mode */ + if (params_channels(hw_params) == 1) + freq = (slots <= 1 ? 2 : slots) * params_width(hw_params) * + params_rate(hw_params); + /* Don't apply it to any non-baudclk circumstance */ if (IS_ERR(ssi->baudclk)) return -EINVAL; -- 2.21.0

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[alsa-devel] [PATCH 0/2 v4] ALSA: pcm: anonymous dup implementation
by Jaroslav Kysela 12 Jun '20

12 Jun '20

This patchset contains the anonymous dup implementation with permissions checking for the ALSA's PCM interface in kernel to enable the restricted DMA sound buffer sharing for the restricted tasks. The code was tested through qemu and it seems to be pretty stable. The initial tinyalsa implementation can be found here: https://github.com/perexg/tinyalsa/commits/anondup v2 of the patches: - change clone parameter to subdevice number for the pcm attach - change SNDRV_PCM_PERM_MAX to SNDRV_PCM_PERM_MASK - the tinyalsa implementation was a little updated (restructured) v3 of the patches: - group integer declarations in snd_pcm_anonymous_dup() - replaced substream->pcm with pcm in snd_pcm_anonymous_dup() - added SNDRV_PCM_PERM_RW check for read/write/readv/writev syscalls v4 of the patches: - more simple restriction control (only two modes - full/buffer) - the tinyalsa implementation follows this change Cc: Phil Burk <philburk(a)google.com> Cc: Zach Riggle <riggle(a)google.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Leo Yan <leo.yan(a)linaro.org> Cc: Baolin Wang <baolin.wang(a)linaro.org> Jaroslav Kysela (2): ALSA: pcm: implement the anonymous dup (inode file descriptor) ALSA: pcm: implement the mmap buffer mode for the anonymous dup include/sound/pcm.h | 10 +++-- include/uapi/sound/asound.h | 6 ++- sound/core/oss/pcm_oss.c | 2 +- sound/core/pcm.c | 13 +++--- sound/core/pcm_compat.c | 1 + sound/core/pcm_native.c | 97 ++++++++++++++++++++++++++++++++++++++++++--- 6 files changed, 110 insertions(+), 19 deletions(-) -- 2.13.6

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