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

[PATCH 01/18] ALSA: core: Use DIV_ROUND_UP() instead of open-coding it
by Lars-Peter Clausen 25 Dec '20

25 Dec '20

Use DIV_ROUND_UP() instead of open-coding it. This documents intent and makes it more clear what is going on for the casual reviewer. Generated using the following the Coccinelle semantic patch. // <smpl> @@ expression x, y; @@ -(((x) + (y) - 1) / (y)) +DIV_ROUND_UP(x, y) // </smpl> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/core/control.c | 4 ++-- sound/core/seq/seq_memory.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 3b44378b9dec..1571c7f7c43b 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -836,7 +836,7 @@ static void fill_remaining_elem_value(struct snd_ctl_elem_value *control, { size_t offset = value_sizes[info->type] * info->count; - offset = (offset + sizeof(u32) - 1) / sizeof(u32); + offset = DIV_ROUND_UP(offset, sizeof(u32)); memset32((u32 *)control->value.bytes.data + offset, pattern, sizeof(control->value) / sizeof(u32) - offset); } @@ -928,7 +928,7 @@ static int sanity_check_elem_value(struct snd_card *card, /* check whether the remaining area kept untouched */ offset = value_sizes[info->type] * info->count; - offset = (offset + sizeof(u32) - 1) / sizeof(u32); + offset = DIV_ROUND_UP(offset, sizeof(u32)); p = (u32 *)control->value.bytes.data + offset; for (; offset < sizeof(control->value) / sizeof(u32); offset++, p++) { if (*p != pattern) { diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c index 65db1a7c77b7..e245bb6ba533 100644 --- a/sound/core/seq/seq_memory.c +++ b/sound/core/seq/seq_memory.c @@ -290,7 +290,7 @@ int snd_seq_event_dup(struct snd_seq_pool *pool, struct snd_seq_event *event, extlen = 0; if (snd_seq_ev_is_variable(event)) { extlen = event->data.ext.len & ~SNDRV_SEQ_EXT_MASK; - ncells = (extlen + sizeof(struct snd_seq_event) - 1) / sizeof(struct snd_seq_event); + ncells = DIV_ROUND_UP(extlen, sizeof(struct snd_seq_event)); } if (ncells >= pool->total_elements) return -ENOMEM; -- 2.20.1

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[PATCH] ALSA: pcm: use bit field on snd_pcm
by Kuninori Morimoto 25 Dec '20

25 Dec '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_pcm's internal, nonatomic, no_device_suspend are defined as bool. But we can reduce struct size if we use bit field for these. This patch converts these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/pcm.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 2336bf9243e1..e836741d2708 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -520,12 +520,13 @@ struct snd_pcm { wait_queue_head_t open_wait; void *private_data; void (*private_free) (struct snd_pcm *pcm); - bool internal; /* pcm is for internal use only */ - bool nonatomic; /* whole PCM operations are in non-atomic context */ - bool no_device_suspend; /* don't invoke device PM suspend */ #if IS_ENABLED(CONFIG_SND_PCM_OSS) struct snd_pcm_oss oss; #endif + /* bit field */ + unsigned int internal:1; /* pcm is for internal use only */ + unsigned int nonatomic:1; /* whole PCM operations are in non-atomic context */ + unsigned int no_device_suspend:1; /* don't invoke device PM suspend */ }; /* -- 2.25.1

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Fwd: question about alsa tracepoints and alsa debugging
by Bert Schiettecatte 25 Dec '20

25 Dec '20

Hi all, I have a rk3288 board I am working with, it has two ADCs and one DAC. The ADCs are connected via an FPGA to the rk3288. The DAC is directly connected to the I2S port of the rk3288. The rk3288 generates the master clock. I am using buildroot and busybox and kernel 5.9.12 (stable). I have a driver which receives the ADC data from the FPGA via SDIO. When I run speaker-test in the background and use arecord to record simultaneously in the foreground, both arecord and speaker-test seem to lock up. When I run speaker-test alone I do not have a problem. The playback device uses the rockchip_i2s.c driver while the capture device uses my SDIO driver. My driver based on dw_mmc.c calls snd_pcm_period_elapsed() for each buffer of frames that is ready. I can see that this is being called for a while when I start arecord in the foreground, until the lockup occurs. The moment I start arecord, the console output from speaker-test stops, so it's locked up by starting arecord. Previously this was working fine on kernel 4.11. However, since I updated to kernel 5.9.12 this no longer works. Has anything changed between 4.11 and 5.9.12 that could be the cause of the problem? commands used for starting playback / trace capture and for recording - ./perf record -F 99 -g -e snd_pcm:* -e sched:sched_switch -e sched:sched_wakeup -e irq:* speaker-test -D hw:CARD=ak4458ak5558aud,DEV=0 -c8 -r192000 -FS32_LE -tsine & arecord -Dhw:CARD=ak4458ak5558aud,DEV=1 -fS32_LE -c16 -r192000 /mnt/ramdisk/rec.wav I enabled the kernel options as described at https://www.kernel.org/doc/html/v5.9/sound/designs/tracepoints.html and have used perf to record a trace, which can be access here - https://kernel-debugging.s3-us-west-1.amazonaws.com/perf.data.script.211220… (exported using "perf script") The trace file shows "lockdep_recursion" at some point so I am wondering if my problem has to do with a deadlock somehow being caused. However, I also see handle_mm_fault, and do_page_fault, which has to do with memory mapping? So I'm not sure what the problem exactly is and what to do next to find out what's wrong. Thanks in advance for your help and interest and patience, Bert

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[PATCH] ASoC: rt1015: reset cali_done when suspending
by Tzung-Bi Shih 25 Dec '20

25 Dec '20

Assuming the calibration state gets lost after system suspend. Resets cali_done when suspending. The rt1015_priv is alloced by kzalloc. No need to initialize cali_done to 0 in component probe callback. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- The idea inspired from previous discussion: (https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178229.…) The device could lose the calibration state after suspend/resume. It is better to reset the flag and get chance to calibrate again. The patch applies after the series: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-December/178608.… sound/soc/codecs/rt1015.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 4a9e2eaebe30..2dcb7b0fba60 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -1055,7 +1055,6 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; - rt1015->cali_done = 0; INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); @@ -1103,6 +1102,7 @@ static int rt1015_suspend(struct snd_soc_component *component) regcache_cache_only(rt1015->regmap, true); regcache_mark_dirty(rt1015->regmap); + rt1015->cali_done = 0; return 0; } -- 2.29.2.729.g45daf8777d-goog

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[PATCH] ALSA: usb-audio: Fix UBSAN warnings for MIDI jacks
by Takashi Iwai 23 Dec '20

23 Dec '20

The calculation of in_cables and out_cables bitmaps are done with the bit shift by the value from the descriptor, which is an arbitrary value, and can lead to UBSAN shift-out-of-bounds warnings. Fix it by filtering the bad descriptor values with the check of the upper bound 0x10 (the cable bitmaps are 16 bits). Reported-by: syzbot+92e45ae45543f89e8c88(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/midi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index c8213652470c..0c23fa6d8525 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1889,6 +1889,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ms_ep = find_usb_ms_endpoint_descriptor(hostep); if (!ms_ep) continue; + if (ms_ep->bNumEmbMIDIJack > 0x10) + continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { if (++epidx >= MIDI_MAX_ENDPOINTS) { @@ -2141,6 +2143,8 @@ static int snd_usbmidi_detect_roland(struct snd_usb_midi *umidi, cs_desc[1] == USB_DT_CS_INTERFACE && cs_desc[2] == 0xf1 && cs_desc[3] == 0x02) { + if (cs_desc[4] > 0x10 || cs_desc[5] > 0x10) + continue; endpoint->in_cables = (1 << cs_desc[4]) - 1; endpoint->out_cables = (1 << cs_desc[5]) - 1; return snd_usbmidi_detect_endpoints(umidi, endpoint, 1); -- 2.26.2

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Dell 0x0a58 change name
by Kailang 23 Dec '20

23 Dec '20

Hi Takashi, Dell want change platform name to 'Dell' only. BR, Kailang

2 1

Lenovo C940 speaker
by Kailang 23 Dec '20

23 Dec '20

Hi Takashi, C940 make i2s Amp(none Realtek) verb in BIOS verb table. It will enable internal speaker. This patch will solve it can't adjust speaker volume issue. BR, Kailang

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[PATCH AUTOSEL 5.4 057/130] ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 93db51d06b32227319dae2ac289029ccf1b33181 ] The current driver code assumes blindly that all found sample rates for the same endpoint from the UAC2 and UAC3 descriptors can be used no matter which altsetting, but actually this was wrong: some devices accept only limited sample rates in each altsetting. For determining which altsetting supports which rate, we need to verify each sample rate and check the validity via UAC2_AS_VAL_ALT_SETTINGS. This control reports back the available altsettings as a bitmap. This patch implements the missing piece above, the verification and reconstructs the sample rate tables based on the result. An open question is how to deal with the altsettings that ended up with no valid sample rates after verification. At least, there is a device that showed this problem although the sample rates did work in the later usage (see bug link). For now, we accept such an altset as is, assuming that it's a firmware bug. Reported-by: Dylan Robinson <dylan_robinson(a)motu.com> Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1178203 Link: https://lore.kernel.org/r/20201123085347.19667-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/clock.c | 106 ++++++++++++++++++++++++++------------------- sound/usb/clock.h | 4 ++ sound/usb/format.c | 93 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 158 insertions(+), 45 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b118cf97607f3..960174485a531 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -560,16 +560,60 @@ static int get_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, return le32_to_cpu(data); } +/* + * Try to set the given sample rate: + * + * Return 0 if the clock source is read-only, the actual rate on success, + * or a negative error code. + * + * This function gets called from format.c to validate each sample rate, too. + * Hence no message is shown upon error + */ +int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, + const struct audioformat *fmt, + int clock, int rate) +{ + bool writeable; + u32 bmControls; + __le32 data; + int err; + + if (fmt->protocol == UAC_VERSION_3) { + struct uac3_clock_source_descriptor *cs_desc; + + cs_desc = snd_usb_find_clock_source_v3(chip->ctrl_intf, clock); + bmControls = le32_to_cpu(cs_desc->bmControls); + } else { + struct uac_clock_source_descriptor *cs_desc; + + cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, clock); + bmControls = cs_desc->bmControls; + } + + writeable = uac_v2v3_control_is_writeable(bmControls, + UAC2_CS_CONTROL_SAM_FREQ); + if (!writeable) + return 0; + + data = cpu_to_le32(rate); + err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, + UAC2_CS_CONTROL_SAM_FREQ << 8, + snd_usb_ctrl_intf(chip) | (clock << 8), + &data, sizeof(data)); + if (err < 0) + return err; + + return get_sample_rate_v2v3(chip, fmt->iface, fmt->altsetting, clock); +} + static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, - struct usb_host_interface *alts, - struct audioformat *fmt, int rate) + struct usb_host_interface *alts, + struct audioformat *fmt, int rate) { struct usb_device *dev = chip->dev; - __le32 data; - int err, cur_rate, prev_rate; + int cur_rate, prev_rate; int clock; - bool writeable; - u32 bmControls; /* First, try to find a valid clock. This may trigger * automatic clock selection if the current clock is not @@ -592,50 +636,22 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, if (prev_rate == rate) goto validation; - if (fmt->protocol == UAC_VERSION_3) { - struct uac3_clock_source_descriptor *cs_desc; - - cs_desc = snd_usb_find_clock_source_v3(chip->ctrl_intf, clock); - bmControls = le32_to_cpu(cs_desc->bmControls); - } else { - struct uac_clock_source_descriptor *cs_desc; - - cs_desc = snd_usb_find_clock_source(chip->ctrl_intf, clock); - bmControls = cs_desc->bmControls; + cur_rate = snd_usb_set_sample_rate_v2v3(chip, fmt, clock, rate); + if (cur_rate < 0) { + usb_audio_err(chip, + "%d:%d: cannot set freq %d (v2/v3): err %d\n", + iface, fmt->altsetting, rate, cur_rate); + return cur_rate; } - writeable = uac_v2v3_control_is_writeable(bmControls, - UAC2_CS_CONTROL_SAM_FREQ); - if (writeable) { - data = cpu_to_le32(rate); - err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, - USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, - UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), - &data, sizeof(data)); - if (err < 0) { - usb_audio_err(chip, - "%d:%d: cannot set freq %d (v2/v3): err %d\n", - iface, fmt->altsetting, rate, err); - return err; - } - - cur_rate = get_sample_rate_v2v3(chip, iface, - fmt->altsetting, clock); - } else { + if (!cur_rate) cur_rate = prev_rate; - } if (cur_rate != rate) { - if (!writeable) { - usb_audio_warn(chip, - "%d:%d: freq mismatch (RO clock): req %d, clock runs @%d\n", - iface, fmt->altsetting, rate, cur_rate); - return -ENXIO; - } - usb_audio_dbg(chip, - "current rate %d is different from the runtime rate %d\n", - cur_rate, rate); + usb_audio_warn(chip, + "%d:%d: freq mismatch (RO clock): req %d, clock runs @%d\n", + fmt->iface, fmt->altsetting, rate, cur_rate); + return -ENXIO; } /* Some devices doesn't respond to sample rate changes while the diff --git a/sound/usb/clock.h b/sound/usb/clock.h index 68df0fbe09d00..97597f5a3c18a 100644 --- a/sound/usb/clock.h +++ b/sound/usb/clock.h @@ -9,4 +9,8 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, int snd_usb_clock_find_source(struct snd_usb_audio *chip, struct audioformat *fmt, bool validate); +int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, + const struct audioformat *fmt, + int clock, int rate); + #endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 1f9ea513230a6..442df4578182d 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -367,6 +367,97 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, return -ENODEV; } +/* check whether the given altsetting is supported for the already set rate */ +static bool check_valid_altsetting_v2v3(struct snd_usb_audio *chip, int iface, + int altsetting) +{ + struct usb_device *dev = chip->dev; + __le64 raw_data = 0; + u64 data; + int err; + + /* we assume 64bit is enough for any altsettings */ + if (snd_BUG_ON(altsetting >= 64 - 8)) + return false; + + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + UAC2_AS_VAL_ALT_SETTINGS << 8, + iface, &raw_data, sizeof(raw_data)); + if (err < 0) + return false; + + data = le64_to_cpu(raw_data); + /* first byte contains the bitmap size */ + if ((data & 0xff) * 8 < altsetting) + return false; + if (data & (1ULL << (altsetting + 8))) + return true; + + return false; +} + +/* + * Validate each sample rate with the altsetting + * Rebuild the rate table if only partial values are valid + */ +static int validate_sample_rate_table_v2v3(struct snd_usb_audio *chip, + struct audioformat *fp, + int clock) +{ + struct usb_device *dev = chip->dev; + unsigned int *table; + unsigned int nr_rates; + unsigned int rate_min = 0x7fffffff; + unsigned int rate_max = 0; + unsigned int rates = 0; + int i, err; + + table = kcalloc(fp->nr_rates, sizeof(*table), GFP_KERNEL); + if (!table) + return -ENOMEM; + + /* clear the interface altsetting at first */ + usb_set_interface(dev, fp->iface, 0); + + nr_rates = 0; + for (i = 0; i < fp->nr_rates; i++) { + err = snd_usb_set_sample_rate_v2v3(chip, fp, clock, + fp->rate_table[i]); + if (err < 0) + continue; + + if (check_valid_altsetting_v2v3(chip, fp->iface, fp->altsetting)) { + table[nr_rates++] = fp->rate_table[i]; + if (rate_min > fp->rate_table[i]) + rate_min = fp->rate_table[i]; + if (rate_max < fp->rate_table[i]) + rate_max = fp->rate_table[i]; + rates |= snd_pcm_rate_to_rate_bit(fp->rate_table[i]); + } + } + + if (!nr_rates) { + usb_audio_dbg(chip, + "No valid sample rate available for %d:%d, assuming a firmware bug\n", + fp->iface, fp->altsetting); + nr_rates = fp->nr_rates; /* continue as is */ + } + + if (fp->nr_rates == nr_rates) { + kfree(table); + return 0; + } + + kfree(fp->rate_table); + fp->rate_table = table; + fp->nr_rates = nr_rates; + fp->rate_min = rate_min; + fp->rate_max = rate_max; + fp->rates = rates; + return 0; +} + /* * parse the format descriptor and stores the possible sample rates * on the audioformat table (audio class v2 and v3). @@ -459,6 +550,8 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, * allocated, so the rates will be stored */ parse_uac2_sample_rate_range(chip, fp, nr_triplets, data); + ret = validate_sample_rate_table_v2v3(chip, fp, clock); + err_free: kfree(data); err: -- 2.27.0

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Sdca support
by GitHub pull_request - opened 23 Dec '20

23 Dec '20

alsa-project/alsa-ucm-conf pull request #71 was opened from libinyang: Add supports for SDW sdca codec devices Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/71 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/71.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH AUTOSEL 4.4 37/38] ALSA: hda/hdmi: packet buffer index must be set before reading value
by Sasha Levin 23 Dec '20

23 Dec '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 46c3bbd9827952f92e250fa6ee30a797a4c4e17e ] The check for infoframe transmit status in hdmi_infoframe_uptodate() makes the assumption that packet buffer index is set to zero. Align code with specification and explicitly set the index before AC_VERB_GET_HDMI_DIP_XMIT. The packet index setting affects both DIP-Data and DIP-XmitCtrl verbs. There are no known cases where the old implementation has caused driver to work incorrectly. This change is purely based on code review against the specification (HDA spec rev1.0a). Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20201211131613.3271407-1-kai.vehmanen@linux.intel… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b249b1b857464..bab2b06c62b1c 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1055,11 +1055,11 @@ static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, u8 val; int i; + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) != AC_DIPXMIT_BEST) return false; - hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); for (i = 0; i < size; i++) { val = snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_DATA, 0); -- 2.27.0

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