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

[RFC] Add duplex sound support for USB devices using implicit feedback
by Erwin Burema 10 May '20

10 May '20

Hi, Since I had problems getting duplex audio to work on my motu m2 I created the following patch to get duplex audio to work. This is based on my understanding that when implicit feedback is used the size of the next output is determined by the previous input, for this the content of the input is ignored but in case of duplex audio the content can actually contain audio information, so long as both input and output are configured the same both sould be usable. With the above in mind an endpoint can be opened/configured a second time so long as it is an implicit sync endpoint and the config does not change. So that is what this patch does, the check if the config does not change might be a bit more complex then needed, but it seems to work on my motu m2. Fixes bug 207023 and should fix 103751 as well Kind Regards, Erwin Burema --- sound/usb/card.h | 1 + sound/usb/endpoint.c | 195 ++++++++++++++++++++++++++++++++++++++++++- sound/usb/pcm.c | 5 ++ 3 files changed, 197 insertions(+), 4 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 820e564656ed..d6219fba9699 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -108,6 +108,7 @@ struct snd_usb_endpoint { int iface, altsetting; int skip_packets; /* quirks for devices to ignore the first n packets in a stream */ + bool is_implicit_feedback; /* This endpoint is used as implicit feedback */ spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 50104f658ed4..9bea7d3f99f8 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -522,6 +522,8 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, list_add_tail(&ep->list, &chip->ep_list); + ep->is_implicit_feedback = 0; + __exit_unlock: mutex_unlock(&chip->mutex); @@ -621,6 +623,178 @@ static void release_urbs(struct snd_usb_endpoint *ep, int force) ep->nurbs = 0; } +/* + * Check data endpoint for format differences + */ +static bool check_ep_params(struct snd_usb_endpoint *ep, + snd_pcm_format_t pcm_format, + unsigned int channels, + unsigned int period_bytes, + unsigned int frames_per_period, + unsigned int periods_per_buffer, + struct audioformat *fmt, + struct snd_usb_endpoint *sync_ep) +{ + unsigned int maxsize, minsize, packs_per_ms, max_packs_per_urb; + unsigned int max_packs_per_period, urbs_per_period, urb_packs; + unsigned int max_urbs; + int frame_bits = snd_pcm_format_physical_width(pcm_format) * channels; + int tx_length_quirk = (ep->chip->tx_length_quirk && + usb_pipeout(ep->pipe)); + bool ret = 1; + + if (pcm_format == SNDRV_PCM_FORMAT_DSD_U16_LE && fmt->dsd_dop) { + /* + * When operating in DSD DOP mode, the size of a sample frame + * in hardware differs from the actual physical format width + * because we need to make room for the DOP markers. + */ + frame_bits += channels << 3; + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->stride == frame_bits >> 3); + + switch (pcm_format) { + case SNDRV_PCM_FORMAT_U8: + ret = ret && (ep->silence_value == 0x80); + break; + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + ret = ret && (ep->silence_value == 0x69); + break; + default: + ret = ret && (ep->silence_value == 0); + } + + /* assume max. frequency is 50% higher than nominal */ + ret = ret && (ep->freqmax == ep->freqn + (ep->freqn >> 1)); + /* Round up freqmax to nearest integer in order to calculate maximum + * packet size, which must represent a whole number of frames. + * This is accomplished by adding 0x0.ffff before converting the + * Q16.16 format into integer. + * In order to accurately calculate the maximum packet size when + * the data interval is more than 1 (i.e. ep->datainterval > 0), + * multiply by the data interval prior to rounding. For instance, + * a freqmax of 41 kHz will result in a max packet size of 6 (5.125) + * frames with a data interval of 1, but 11 (10.25) frames with a + * data interval of 2. + * (ep->freqmax << ep->datainterval overflows at 8.192 MHz for the + * maximum datainterval value of 3, at USB full speed, higher for + * USB high speed, noting that ep->freqmax is in units of + * frames per packet in Q16.16 format.) + */ + maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) * + (frame_bits >> 3); + if (tx_length_quirk) + maxsize += sizeof(__le32); /* Space for length descriptor */ + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { + /* whatever fits into a max. size packet */ + unsigned int data_maxsize = maxsize = ep->maxpacksize; + + if (tx_length_quirk) + /* Need to remove the length descriptor to calc freq */ + data_maxsize -= sizeof(__le32); + ret = ret && (ep->freqmax == (data_maxsize / (frame_bits >> 3)) + << (16 - ep->datainterval)); + } + + if (ep->fill_max) + ret = ret && (ep->curpacksize == ep->maxpacksize); + else + ret = ret && (ep->curpacksize == maxsize); + + if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { + packs_per_ms = 8 >> ep->datainterval; + max_packs_per_urb = MAX_PACKS_HS; + } else { + packs_per_ms = 1; + max_packs_per_urb = MAX_PACKS; + } + if (sync_ep && !snd_usb_endpoint_implicit_feedback_sink(ep)) + max_packs_per_urb = min(max_packs_per_urb, + 1U << sync_ep->syncinterval); + max_packs_per_urb = max(1u, max_packs_per_urb >> ep->datainterval); + + /* + * Capture endpoints need to use small URBs because there's no way + * to tell in advance where the next period will end, and we don't + * want the next URB to complete much after the period ends. + * + * Playback endpoints with implicit sync much use the same parameters + * as their corresponding capture endpoint. + */ + if (usb_pipein(ep->pipe) || + snd_usb_endpoint_implicit_feedback_sink(ep)) { + + urb_packs = packs_per_ms; + /* + * Wireless devices can poll at a max rate of once per 4ms. + * For dataintervals less than 5, increase the packet count to + * allow the host controller to use bursting to fill in the + * gaps. + */ + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) { + int interval = ep->datainterval; + + while (interval < 5) { + urb_packs <<= 1; + ++interval; + } + } + /* make capture URBs <= 1 ms and smaller than a period */ + urb_packs = min(max_packs_per_urb, urb_packs); + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; + ret = ret && (ep->nurbs == MAX_URBS); + + /* + * Playback endpoints without implicit sync are adjusted so that + * a period fits as evenly as possible in the smallest number of + * URBs. The total number of URBs is adjusted to the size of the + * ALSA buffer, subject to the MAX_URBS and MAX_QUEUE limits. + */ + } else { + /* determine how small a packet can be */ + minsize = (ep->freqn >> (16 - ep->datainterval)) * + (frame_bits >> 3); + /* with sync from device, assume it can be 12% lower */ + if (sync_ep) + minsize -= minsize >> 3; + minsize = max(minsize, 1u); + + /* how many packets will contain an entire ALSA period? */ + max_packs_per_period = DIV_ROUND_UP(period_bytes, minsize); + + /* how many URBs will contain a period? */ + urbs_per_period = DIV_ROUND_UP(max_packs_per_period, + max_packs_per_urb); + /* how many packets are needed in each URB? */ + urb_packs = DIV_ROUND_UP(max_packs_per_period, urbs_per_period); + + /* limit the number of frames in a single URB */ + ret = ret && (ep->max_urb_frames == + DIV_ROUND_UP(frames_per_period, urbs_per_period)); + + /* try to use enough URBs to contain an entire ALSA buffer */ + max_urbs = min((unsigned) MAX_URBS, + MAX_QUEUE * packs_per_ms / urb_packs); + ret = ret && (ep->nurbs == min(max_urbs, + urbs_per_period * periods_per_buffer)); + } + + ret = ret && (ep->datainterval == fmt->datainterval); + ret = ret && (ep->maxpacksize == fmt->maxpacksize); + ret = ret && + (ep->fill_max == !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX)); + + return ret; +} + /* * configure a data endpoint */ @@ -886,10 +1060,23 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, int err; if (ep->use_count != 0) { - usb_audio_warn(ep->chip, - "Unable to change format on ep #%x: already in use\n", - ep->ep_num); - return -EBUSY; + bool check = ep->is_implicit_feedback && + check_ep_params(ep, pcm_format, + channels, period_bytes, + period_frames, buffer_periods, + fmt, sync_ep); + + if (!check) { + usb_audio_warn(ep->chip, + "Unable to change format on ep #%x: already in use\n", + ep->ep_num); + return -EBUSY; + } + + usb_audio_dbg(ep->chip, + "Ep #%x already in use as implicit feedback but format not changed\n", + ep->ep_num); + return 0; } /* release old buffers, if any */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b50965ab3b3a..d61c2f1095b5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -404,6 +404,8 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, if (!subs->sync_endpoint) return -EINVAL; + subs->sync_endpoint->is_implicit_feedback = 1; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 1; @@ -502,12 +504,15 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, implicit_fb ? SND_USB_ENDPOINT_TYPE_DATA : SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) { if (is_playback && attr == USB_ENDPOINT_SYNC_NONE) return 0; return -EINVAL; } + subs->sync_endpoint->is_implicit_feedback = implicit_fb; + subs->data_endpoint->sync_master = subs->sync_endpoint; return 0; -- 2.26.2

2 2

[PATCH 0/6] ALSA: fireface: add support for Fireface 802 and UFX
by Takashi Sakamoto 10 May '20

10 May '20

Hi, Fireface 802 was shipped by RME GmbH in 2014. This model is one of latter models of Fireface series and support both of IEEE 1394 bus and USB. Furthermore, it supports two types of remote control unit (Basic Remote and Advanced Remote Control) with by 9pin mini-din connector. This patchset adds support for this model as a part of devices available by ALSA firewire stack (therefore it's not a device of USB). Userspace applications can transfer PCM frames and MIDI messages via ALSA PCM and Rawmidi interfaces. Fireface UFX is also supported since its internal design is almost the same as Fireface 802 in a point of packet communication. The support is untested and the status is 'request for test'. As well as the other models of RME Fireface series, audio output includes periodical hissing noise. This is not solved yet. Furthermore, capture of MIDI messages requires assist of userspace application. For the detail, please read comment in 'sound/firewire/fireface/ff-protocol-latter.c'. Regards Takashi Sakamoto (6): ALSA: fireface: fix configuration error for nominal sampling transfer frequency ALSA: fireface: start IR context immediately ALSA: fireface: code refactoring to add enumeration constants for model identification ALSA: fireface: code refactoring for name of sound card ALSA: fireface: add support for RME FireFace 802 ALSA: fireface: add support for Fireface UFX (untested) sound/firewire/Kconfig | 2 + sound/firewire/fireface/ff-protocol-latter.c | 58 +++++++++++++------ sound/firewire/fireface/ff-stream.c | 10 +--- sound/firewire/fireface/ff.c | 61 +++++++++++++++++--- sound/firewire/fireface/ff.h | 11 +++- 5 files changed, 103 insertions(+), 39 deletions(-) -- 2.25.1

2 9

snd_dice: Clicking artifacts with TC StudioKonnekt 48
by Mathias Buhr 10 May '20

10 May '20

Hi, thanks to this group, both of my firewire interfaces are (almost) working! Big thank you! While the TC Konnekt 24D works fine (playback and recording), the StudioKonnekt 48 produces clicking artifacts during playback when using snd_dice. This interface is working flawless on Windows and on a Jack/FFADO combination. This artifact occurs in all use cases (alsa via aplay, pulseaudio and jack) and does not seem to be in recorded streams. After recording the playback with another device, it looks like the level of the artifact is scaling with the signal and its interval is interestingly 256ms. Is there anything I can do to further debug this issue? Capture firewire packets? I would love to get this device fully working. I'm using kernel version 5.5.4 but this issue has been there for a long time. Thanks, Mathias

3 9

[PATCH] ALSA: hda/realtek: Add quirk for Samsung Notebook
by Mike Pozulp 10 May '20

10 May '20

Some models of the Samsung Notebook 9 have very quiet and distorted headphone output. This quirk changes the VREF value of the ALC298 codec NID 0x1a from default HIZ to new 100. Signed-off-by: Mike Pozulp <pozulp.kernel(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 63e1a56f705b..9b14ed5c75d7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5923,6 +5923,7 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, + ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, }; static const struct hda_fixup alc269_fixups[] = { @@ -7061,6 +7062,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc5 }, + { } + }, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7336,6 +7344,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), + SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), #if 0 /* Below is a quirk table taken from the old code. -- 2.24.1

2 1

crackling microphone
by GitHub issues - reopened 09 May '20

09 May '20

alsa-project/alsa-plugins issue #10 was reopened from riddicc: Hi! When I send a 800Hz sine wave with rate 12345 to pcm.outmic and when I arecord from pcm.micend at rate 8000 with 1 channel ([/etc/asound.conf](https://github.com/alsa-project/alsa-plugins/files/4597703/etc-asound-conf.txt)), then I get a lot of tiny little gaps (see attachment) and a lot of messages like "overrun!!! (at least 0.014 ms long)" (the "ms" vary from 0.008 to 0.043; and arecord continues to record) and sometimes even this error: "arecord: xrun:1672: read/write error, state = RUNNING" (it causes arecord to terminate). 1. > speaker-test -D outmic -t sine -f 800 -c2 -r 12345 2. > arecord -D micend -r 8000 -c 1 -f S16_LE /tmp/bla.wav twinkle makes even bigger gaps, when I let it use pcm.micend as microphone... But with hw:Generic,0,0 as mic device twinkle works fine... https://github.com/LubosD/twinkle/issues/207 "samplerate_best" and "samplerate_order" seems to make it worse... How can I fix that? Thx. Bye. ![audacity screenshot](https://user-images.githubusercontent.com/33246439/81384628-7ad… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/10 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[alsa-devel] [alsa-lib][RFC PATCH] ucm: reset config id of condition items
by libin.yang＠linux.intel.com 09 May '20

09 May '20

From: Libin Yang <libin.yang(a)intel.com> UCMv2 supports "If" statement and will merge the same items with compound_merge(). If the items have the same id, it will fail to add the config items. And the id of the item in an array is automatically generated with the increased number. It is probably that some items to be merged have the same id. Let's add prefix in the id to avoid such situation. For example: If.seq1 { Condition { Type ControlExists Control "name='PGA1.0 1 Master Playback Volume'" } True { EnableSequence [ cset "name='PGA1.0 1 Master Playback Volume' 50" ] } } If.seq2 { Condition { Type ControlExists Control "name='PGA2.0 2 Master Playback Volume'" } True { EnableSequence [ cset "name='PGA2.0 2 Master Playback Volume' 50" ] } } If.seq3 { Condition { Type ControlExists Control "name='PGA3.0 3 Master Playback Volume'" } True { EnableSequence [ cset "name='PGA3.0 3 Master Playback Volume' 50" ] } } If seq1, seq2 and seq3 conditions are true, UCM will fail to initialize. This patch rename the config id to avoid the same id conflict. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- include/conf.h | 2 +- src/conf.c | 21 +++++++++++++++++++++ src/ucm/ucm_cond.c | 28 ++++++++++++++++++++++++---- 3 files changed, 46 insertions(+), 5 deletions(-) diff --git a/include/conf.h b/include/conf.h index 456b272..adb3d84 100644 --- a/include/conf.h +++ b/include/conf.h @@ -139,7 +139,7 @@ int snd_config_imake_safe_string(snd_config_t **config, const char *key, const c int snd_config_imake_pointer(snd_config_t **config, const char *key, const void *ptr); snd_config_type_t snd_config_get_type(const snd_config_t *config); - +int snd_config_of_array(const snd_config_t *config); int snd_config_set_id(snd_config_t *config, const char *id); int snd_config_set_integer(snd_config_t *config, long value); int snd_config_set_integer64(snd_config_t *config, long long value); diff --git a/src/conf.c b/src/conf.c index 50d0403..43d565b 100644 --- a/src/conf.c +++ b/src/conf.c @@ -435,6 +435,8 @@ struct _snd_config { char *id; snd_config_type_t type; int refcount; /* default = 0 */ + /* member of an array */ + int of_array; union { long integer; long long integer64; @@ -1123,6 +1125,7 @@ static int _snd_config_make(snd_config_t **config, char **id, snd_config_type_t *id = NULL; } n->type = type; + n->of_array = 0; if (type == SND_CONFIG_TYPE_COMPOUND) INIT_LIST_HEAD(&n->u.compound.fields); *config = n; @@ -1316,6 +1319,8 @@ static int parse_array_def(snd_config_t *parent, input_t *input, int *idx, int s default: unget_char(c, input); err = parse_value(&n, parent, input, &id, skip); + /* this is a member of an array */ + n->of_array = 1; if (err < 0) goto __end; break; @@ -1784,6 +1789,22 @@ snd_config_type_t snd_config_get_type(const snd_config_t *config) } /** + * \brief Returns the of_array of a configuration node. + * \param config Handle to the configuration node. + * \return of_array of the node + * + * The returned value indicates whether the node is a member of an array. + * + * \par Conforming to: + * LSB 3.2 + */ +int snd_config_of_array(const snd_config_t *config) +{ + assert(config); + return config->of_array; +} + +/** * \brief Returns the id of a configuration node. * \param[in] config Handle to the configuration node. * \param[out] id The function puts the pointer to the id string at the diff --git a/src/ucm/ucm_cond.c b/src/ucm/ucm_cond.c index 22b418d..725a69e 100644 --- a/src/ucm/ucm_cond.c +++ b/src/ucm/ucm_cond.c @@ -347,14 +347,15 @@ static void config_dump(snd_config_t *cfg) } #endif -static int compound_merge(const char *id, +static int compound_merge(const char *id, const char *cname, snd_config_t *dst, snd_config_t *src, snd_config_t *before, snd_config_t *after) { snd_config_iterator_t i, next; snd_config_t *n, *_before = NULL, *_after = NULL; const char *s; - int err; + char s1[32]; + int err, cnt; if (snd_config_get_type(src) != SND_CONFIG_TYPE_COMPOUND) { uc_error("compound type expected for If True/False block"); @@ -387,8 +388,22 @@ static int compound_merge(const char *id, return -EINVAL; } + cnt = 0; snd_config_for_each(i, next, src) { n = snd_config_iterator_entry(i); + /* + * If n is an array member, n->id is automatically generated. + * It is prossible that n->id is used by other array member, + * which will be merged with this one. So let's add prefix + * to the id to avoid the conflict. + */ + if (snd_config_of_array(n)) { + err = snd_config_get_id(n, &s); + if (err < 0) + return err; /* FIXME: this will never happen */ + snprintf(s1, sizeof(s1), "%s-%d-%s", cname, cnt++, s); + snd_config_set_id(n, (const char *)s1); + } err = snd_config_remove(n); if (err < 0) return err; @@ -422,7 +437,7 @@ int uc_mgr_evaluate_condition(snd_use_case_mgr_t *uc_mgr, { snd_config_iterator_t i, i2, next, next2; snd_config_t *a, *n, *n2, *parent2, *before, *after; - const char *id; + const char *id, *cname; int err; if (uc_mgr->conf_format < 2) { @@ -437,6 +452,10 @@ int uc_mgr_evaluate_condition(snd_use_case_mgr_t *uc_mgr, snd_config_for_each(i, next, cond) { n = snd_config_iterator_entry(i); + err = snd_config_get_id(n, &cname); + if (err < 0) + return err; /* FIXME: this will never happen */ + before = after = NULL; err = if_eval_one(uc_mgr, n, &a, &before, &after); if (err < 0) @@ -469,7 +488,8 @@ __add: err = snd_config_search(parent, id, &parent2); if (err == -ENOENT) goto __add; - err = compound_merge(id, parent2, n2, before, after); + err = compound_merge(id, cname, parent2, n2, + before, after); if (err < 0) return err; } -- 2.7.4

4 8

crackling microphone
by GitHub issues - reopened 09 May '20

09 May '20

alsa-project/alsa-plugins issue #10 was reopened from riddicc: Hi! When I send a 800Hz sine wave with rate 12345 to pcm.outmic and when I arecord from pcm.micend at rate 8000 with 1 channel ([/etc/asound.conf](https://github.com/alsa-project/alsa-plugins/files/4597703/etc-asound-conf.txt)), then I get a lot of tiny little gaps (see attachment) and a lot of messages like "overrun!!! (at least 0.014 ms long)" (the "ms" vary from 0.008 to 0.043; and arecord continues to record) and sometimes even this error: "arecord: xrun:1672: read/write error, state = RUNNING" (it causes arecord to terminate). 1. > speaker-test -D outmic -t sine -f 800 -c2 -r 12345 2. > arecord -D micend -r 8000 -c 1 -f S16_LE /tmp/bla.wav twinkle makes even bigger gaps, when I let it use pcm.micend as microphone... But with hw:Generic,0,0 as mic device twinkle works fine... https://github.com/LubosD/twinkle/issues/207 "samplerate_best" and "samplerate_order" seems to make it worse... How can I fix that? Thx. Bye. ![audacity screenshot](https://user-images.githubusercontent.com/33246439/81384628-7ad… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/10 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH 1/2] dt-bindings: Added device tree binding for max98390
by Steve Lee 09 May '20

09 May '20

Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- .../devicetree/bindings/sound/max98390.txt | 26 +++++++++++++++++++ 1 file changed, 26 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98390.txt diff --git a/Documentation/devicetree/bindings/sound/max98390.txt b/Documentation/devicetree/bindings/sound/max98390.txt new file mode 100644 index 000000000000..147dfd88cd3b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98390.txt @@ -0,0 +1,26 @@ +Maxim Integrated MAX98390 Speaker Amplifier + +This device supports I2C. + +Required properties: + + - compatible : "maxim,max98390" + + - reg : the I2C address of the device. + +Optional properties: + + - maxim,temperature_calib + u32. The calculated temperature data was measured while doing the calibration. Data calibration : Temp / 100 * 2^12 + + - maxim,r0_calib + u32. This is r0 calibration data which was measured in factory mode. + +Example: + +codec: max98390@38 { + compatible = "maxim,max98390"; + reg = <0x38>; + maxim,temperature_calib = <1024>; + maxim,r0_calib = <0x224050>; +}; -- 2.17.1

1 0

[PATCH] ASoC: cros_ec_codec: allocate shash_desc dynamically
by Arnd Bergmann 08 May '20

08 May '20

The wov_hotword_model_put() function has multiple large variables on its stack, the largest of which is the result of SHASH_DESC_ON_STACK(). In total, this exceeds the warning limit for 32-bit architectures: sound/soc/codecs/cros_ec_codec.c:776:12: error: stack frame size of 1152 bytes in function 'wov_hotword_model_put' [-Werror,-Wframe-larger-than=] The function already has a dynamic crypto_alloc_shash() allocation, so using kmalloc() for the descriptor is correct as well and does not introduce any additional failure scenarios. With this, the stack usage of wov_hotword_model_put() gets reduced to 480 bytes in my test configuration. Fixes: b6bc07d4360d ("ASoC: cros_ec_codec: support WoV") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/cros_ec_codec.c | 17 +++++++++-------- 1 file changed, 9 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index d3dc42aa6825..1948bc6971f6 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -108,22 +108,23 @@ static int calculate_sha256(struct cros_ec_codec_priv *priv, uint8_t *buf, uint32_t size, uint8_t *digest) { struct crypto_shash *tfm; + struct shash_desc *desc; tfm = crypto_alloc_shash("sha256", CRYPTO_ALG_TYPE_SHASH, 0); if (IS_ERR(tfm)) { dev_err(priv->dev, "can't alloc shash\n"); return PTR_ERR(tfm); } - - { - SHASH_DESC_ON_STACK(desc, tfm); - - desc->tfm = tfm; - - crypto_shash_digest(desc, buf, size, digest); - shash_desc_zero(desc); + desc = kmalloc(sizeof(*desc) + crypto_shash_descsize(tfm), GFP_KERNEL); + if (!desc) { + crypto_free_shash(tfm); + return -ENOMEM; } + desc->tfm = tfm; + crypto_shash_digest(desc, buf, size, digest); + shash_desc_zero(desc); + kfree(desc); crypto_free_shash(tfm); #ifdef DEBUG -- 2.26.0

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[PATCH] ASoC: qcom: Use devm_platform_ioremap_resource_byname() to simplify code
by Tang Bin 08 May '20

08 May '20

Use devm_platform_ioremap_resource_byname() instead of platform_get_resource_byname() + devm_ioremap_resource(). Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/qcom/lpass-cpu.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index dbce7e92b..885c1f2e7 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -417,7 +417,6 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) { struct lpass_data *drvdata; struct device_node *dsp_of_node; - struct resource *res; struct lpass_variant *variant; struct device *dev = &pdev->dev; const struct of_device_id *match; @@ -442,9 +441,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) drvdata->variant = (struct lpass_variant *)match->data; variant = drvdata->variant; - res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "lpass-lpaif"); - - drvdata->lpaif = devm_ioremap_resource(&pdev->dev, res); + drvdata->lpaif = devm_platform_ioremap_resource_byname(pdev, + "lpass-lpaif") if (IS_ERR((void const __force *)drvdata->lpaif)) { dev_err(&pdev->dev, "error mapping reg resource: %ld\n", PTR_ERR((void const __force *)drvdata->lpaif)); -- 2.20.1.windows.1

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