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

[PATCH AUTOSEL 5.7 138/388] ALSA: usb-audio: Improve frames size computation
by Sasha Levin 18 Jun '20

18 Jun '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ] For computation of the the next frame size current value of fs/fps and accumulated fractional parts of fs/fps are used, where values are stored in Q16.16 format. This is quite natural for computing frame size for asynchronous endpoints driven by explicit feedback, since in this case fs/fps is a value provided by the feedback endpoint and it's already in the Q format. If an error is accumulated over time, the device can adjust fs/fps value to prevent buffer overruns/underruns. But for synchronous endpoints the accuracy provided by these computations is not enough. Due to accumulated error the driver periodically produces frames with incorrect size (+/- 1 audio sample). This patch fixes this issue by implementing a different algorithm for frame size computation. It is based on accumulating of the remainders from division fs/fps and it doesn't accumulate errors over time. This new method is enabled for synchronous and adaptive playback endpoints. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 4 ++++ sound/usb/endpoint.c | 43 ++++++++++++++++++++++++++++++++++++++----- sound/usb/endpoint.h | 1 + sound/usb/pcm.c | 2 ++ 4 files changed, 45 insertions(+), 5 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index 395403a2d33f..820e564656ed 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -84,6 +84,10 @@ struct snd_usb_endpoint { dma_addr_t sync_dma; /* DMA address of syncbuf */ unsigned int pipe; /* the data i/o pipe */ + unsigned int framesize[2]; /* small/large frame sizes in samples */ + unsigned int sample_rem; /* remainder from division fs/fps */ + unsigned int sample_accum; /* sample accumulator */ + unsigned int fps; /* frames per second */ unsigned int freqn; /* nominal sampling rate in fs/fps in Q16.16 format */ unsigned int freqm; /* momentary sampling rate in fs/fps in Q16.16 format */ int freqshift; /* how much to shift the feedback value to get Q16.16 */ diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 4a9a2f6ef5a4..d8dc7cb56d43 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -124,12 +124,12 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep) /* * For streaming based on information derived from sync endpoints, - * prepare_outbound_urb_sizes() will call next_packet_size() to + * prepare_outbound_urb_sizes() will call slave_next_packet_size() to * determine the number of samples to be sent in the next packet. * - * For implicit feedback, next_packet_size() is unused. + * For implicit feedback, slave_next_packet_size() is unused. */ -int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep) { unsigned long flags; int ret; @@ -146,6 +146,29 @@ int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) return ret; } +/* + * For adaptive and synchronous endpoints, prepare_outbound_urb_sizes() + * will call next_packet_size() to determine the number of samples to be + * sent in the next packet. + */ +int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep) +{ + int ret; + + if (ep->fill_max) + return ep->maxframesize; + + ep->sample_accum += ep->sample_rem; + if (ep->sample_accum >= ep->fps) { + ep->sample_accum -= ep->fps; + ret = ep->framesize[1]; + } else { + ret = ep->framesize[0]; + } + + return ret; +} + static void retire_outbound_urb(struct snd_usb_endpoint *ep, struct snd_urb_ctx *urb_ctx) { @@ -190,6 +213,8 @@ static void prepare_silent_urb(struct snd_usb_endpoint *ep, if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); @@ -874,10 +899,17 @@ int snd_usb_endpoint_set_params(struct snd_usb_endpoint *ep, ep->maxpacksize = fmt->maxpacksize; ep->fill_max = !!(fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX); - if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_FULL) { ep->freqn = get_usb_full_speed_rate(rate); - else + ep->fps = 1000; + } else { ep->freqn = get_usb_high_speed_rate(rate); + ep->fps = 8000; + } + + ep->sample_rem = rate % ep->fps; + ep->framesize[0] = rate / ep->fps; + ep->framesize[1] = (rate + (ep->fps - 1)) / ep->fps; /* calculate the frequency in 16.16 format */ ep->freqm = ep->freqn; @@ -936,6 +968,7 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; + ep->sample_accum = 0; snd_usb_endpoint_start_quirk(ep); diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 63a39d4fa8d8..d23fa0a8c11b 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -28,6 +28,7 @@ void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free(struct snd_usb_endpoint *ep); int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); +int snd_usb_endpoint_slave_next_packet_size(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep); void snd_usb_handle_sync_urb(struct snd_usb_endpoint *ep, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a4e4064f9aee..b50965ab3b3a 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1579,6 +1579,8 @@ static void prepare_playback_urb(struct snd_usb_substream *subs, for (i = 0; i < ctx->packets; i++) { if (ctx->packet_size[i]) counts = ctx->packet_size[i]; + else if (ep->sync_master) + counts = snd_usb_endpoint_slave_next_packet_size(ep); else counts = snd_usb_endpoint_next_packet_size(ep); -- 2.25.1

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[PATCH AUTOSEL 5.7 124/388] ALSA: usb-audio: RME Babyface Pro mixer patch
by Sasha Levin 18 Jun '20

18 Jun '20

From: Thomas Ebeling <penguins(a)bollie.de> [ Upstream commit 3e8f3bd047163d30fb1ad32ca7e4628921555c09 ] Added mixer quirks to allow controlling the internal DSP of the RME Babyface Pro and its successor Babyface Pro FS. Signed-off-by: Thomas Ebeling <penguins(a)bollie.de> Link: https://lore.kernel.org/r/20200414211019.qprg7whepg2y7nei@bollie.ca9.eu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer_quirks.c | 418 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 418 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index a5f65a9a0254..bdff8674942e 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2185,6 +2185,421 @@ static int snd_rme_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * RME Babyface Pro (FS) + * + * These devices exposes a couple of DSP functions via request to EP0. + * Switches are available via control registers, while routing is controlled + * by controlling the volume on each possible crossing point. + * Volume control is linear, from -inf (dec. 0) to +6dB (dec. 46341) with + * 0dB being at dec. 32768. + */ +enum { + SND_BBFPRO_CTL_REG1 = 0, + SND_BBFPRO_CTL_REG2 +}; + +#define SND_BBFPRO_CTL_REG_MASK 1 +#define SND_BBFPRO_CTL_IDX_MASK 0xff +#define SND_BBFPRO_CTL_IDX_SHIFT 1 +#define SND_BBFPRO_CTL_VAL_MASK 1 +#define SND_BBFPRO_CTL_VAL_SHIFT 9 +#define SND_BBFPRO_CTL_REG1_CLK_MASTER 0 +#define SND_BBFPRO_CTL_REG1_CLK_OPTICAL 1 +#define SND_BBFPRO_CTL_REG1_SPDIF_PRO 7 +#define SND_BBFPRO_CTL_REG1_SPDIF_EMPH 8 +#define SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL 10 +#define SND_BBFPRO_CTL_REG2_48V_AN1 0 +#define SND_BBFPRO_CTL_REG2_48V_AN2 1 +#define SND_BBFPRO_CTL_REG2_SENS_IN3 2 +#define SND_BBFPRO_CTL_REG2_SENS_IN4 3 +#define SND_BBFPRO_CTL_REG2_PAD_AN1 4 +#define SND_BBFPRO_CTL_REG2_PAD_AN2 5 + +#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff +#define SND_BBFPRO_MIXER_VAL_SHIFT 9 +#define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf +#define SND_BBFPRO_MIXER_VAL_MAX 46341 // +6dB + +#define SND_BBFPRO_USBREQ_CTL_REG1 0x10 +#define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_MIXER 0x12 + +static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, + u8 index, u8 value) +{ + int err; + u16 usb_req, usb_idx, usb_val; + struct snd_usb_audio *chip = mixer->chip; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + if (reg == SND_BBFPRO_CTL_REG1) { + usb_req = SND_BBFPRO_USBREQ_CTL_REG1; + if (index == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) { + usb_idx = 3; + usb_val = value ? 3 : 0; + } else { + usb_idx = 1 << index; + usb_val = value ? usb_idx : 0; + } + } else { + usb_req = SND_BBFPRO_USBREQ_CTL_REG2; + usb_idx = 1 << index; + usb_val = value ? usb_idx : 0; + } + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), usb_req, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + usb_val, usb_idx, 0, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u8 reg, idx, val; + int pv; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + val = kcontrol->private_value >> SND_BBFPRO_CTL_VAL_SHIFT; + + if ((reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) || + (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) { + ucontrol->value.enumerated.item[0] = val; + } else { + ucontrol->value.integer.value[0] = val; + } + return 0; +} + +static int snd_bbfpro_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + u8 reg, idx; + int pv; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + + if (reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) { + static const char * const texts[2] = { + "AutoSync", + "Internal" + }; + return snd_ctl_enum_info(uinfo, 1, 2, texts); + } else if (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4)) { + static const char * const texts[2] = { + "-10dBV", + "+4dBu" + }; + return snd_ctl_enum_info(uinfo, 1, 2, texts); + } + + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + return 0; +} + +static int snd_bbfpro_ctl_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int err; + u8 reg, idx; + int old_value, pv, val; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + old_value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK; + + if ((reg == SND_BBFPRO_CTL_REG1 && + idx == SND_BBFPRO_CTL_REG1_CLK_OPTICAL) || + (reg == SND_BBFPRO_CTL_REG2 && + (idx == SND_BBFPRO_CTL_REG2_SENS_IN3 || + idx == SND_BBFPRO_CTL_REG2_SENS_IN4))) { + val = ucontrol->value.enumerated.item[0]; + } else { + val = ucontrol->value.integer.value[0]; + } + + if (val > 1) + return -EINVAL; + + if (val == old_value) + return 0; + + kcontrol->private_value = reg + | ((idx & SND_BBFPRO_CTL_IDX_MASK) << SND_BBFPRO_CTL_IDX_SHIFT) + | ((val & SND_BBFPRO_CTL_VAL_MASK) << SND_BBFPRO_CTL_VAL_SHIFT); + + err = snd_bbfpro_ctl_update(mixer, reg, idx, val); + return err < 0 ? err : 1; +} + +static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) +{ + u8 reg, idx; + int value, pv; + + pv = list->kctl->private_value; + reg = pv & SND_BBFPRO_CTL_REG_MASK; + idx = (pv >> SND_BBFPRO_CTL_IDX_SHIFT) & SND_BBFPRO_CTL_IDX_MASK; + value = (pv >> SND_BBFPRO_CTL_VAL_SHIFT) & SND_BBFPRO_CTL_VAL_MASK; + + return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); +} + +static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, + u32 value) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + u16 idx; + u16 usb_idx, usb_val; + u32 v; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + idx = index & SND_BBFPRO_MIXER_IDX_MASK; + // 18 bit linear volume, split so 2 bits end up in index. + v = value & SND_BBFPRO_MIXER_VAL_MASK; + usb_idx = idx | (v & 0x3) << 14; + usb_val = (v >> 2) & 0xffff; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_MIXER, + USB_DIR_OUT | USB_TYPE_VENDOR | + USB_RECIP_DEVICE, + usb_val, usb_idx, 0, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = + kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT; + return 0; +} + +static int snd_bbfpro_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_MIXER_VAL_MIN; + uinfo->value.integer.max = SND_BBFPRO_MIXER_VAL_MAX; + return 0; +} + +static int snd_bbfpro_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int err; + u16 idx; + u32 new_val, old_value, uvalue; + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + uvalue = ucontrol->value.integer.value[0]; + idx = kcontrol->private_value & SND_BBFPRO_MIXER_IDX_MASK; + old_value = kcontrol->private_value >> SND_BBFPRO_MIXER_VAL_SHIFT; + + if (uvalue > SND_BBFPRO_MIXER_VAL_MAX) + return -EINVAL; + + if (uvalue == old_value) + return 0; + + new_val = uvalue & SND_BBFPRO_MIXER_VAL_MASK; + + kcontrol->private_value = idx + | (new_val << SND_BBFPRO_MIXER_VAL_SHIFT); + + err = snd_bbfpro_vol_update(mixer, idx, new_val); + return err < 0 ? err : 1; +} + +static int snd_bbfpro_vol_resume(struct usb_mixer_elem_list *list) +{ + int pv = list->kctl->private_value; + u16 idx = pv & SND_BBFPRO_MIXER_IDX_MASK; + u32 val = (pv >> SND_BBFPRO_MIXER_VAL_SHIFT) + & SND_BBFPRO_MIXER_VAL_MASK; + return snd_bbfpro_vol_update(list->mixer, idx, val); +} + +// Predfine elements +static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_ctl_info, + .get = snd_bbfpro_ctl_get, + .put = snd_bbfpro_ctl_put +}; + +static const struct snd_kcontrol_new snd_bbfpro_vol_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_vol_info, + .get = snd_bbfpro_vol_get, + .put = snd_bbfpro_vol_put +}; + +static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, + u8 index, char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_ctl_control; + + knew.name = name; + knew.private_value = (reg & SND_BBFPRO_CTL_REG_MASK) + | ((index & SND_BBFPRO_CTL_IDX_MASK) + << SND_BBFPRO_CTL_IDX_SHIFT); + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_ctl_resume, + &knew, NULL); +} + +static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_vol_control; + + knew.name = name; + knew.private_value = index & SND_BBFPRO_MIXER_IDX_MASK; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_vol_resume, + &knew, NULL); +} + +static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i, o; + char name[48]; + + static const char * const input[] = { + "AN1", "AN2", "IN3", "IN4", "AS1", "AS2", "ADAT3", + "ADAT4", "ADAT5", "ADAT6", "ADAT7", "ADAT8"}; + + static const char * const output[] = { + "AN1", "AN2", "PH3", "PH4", "AS1", "AS2", "ADAT3", "ADAT4", + "ADAT5", "ADAT6", "ADAT7", "ADAT8"}; + + for (o = 0 ; o < 12 ; ++o) { + for (i = 0 ; i < 12 ; ++i) { + // Line routing + snprintf(name, sizeof(name), + "%s-%s-%s Playback Volume", + (i < 2 ? "Mic" : "Line"), + input[i], output[o]); + err = snd_bbfpro_vol_add(mixer, (26 * o + i), name); + if (err < 0) + return err; + + // PCM routing... yes, it is output remapping + snprintf(name, sizeof(name), + "PCM-%s-%s Playback Volume", + output[i], output[o]); + err = snd_bbfpro_vol_add(mixer, (26 * o + 12 + i), + name); + if (err < 0) + return err; + } + } + + // Control Reg 1 + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_CLK_OPTICAL, + "Sample Clock Source"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_PRO, + "IEC958 Pro Mask"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_EMPH, + "IEC958 Emphasis"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, + SND_BBFPRO_CTL_REG1_SPDIF_OPTICAL, + "IEC958 Switch"); + if (err < 0) + return err; + + // Control Reg 2 + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_48V_AN1, + "Mic-AN1 48V"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_48V_AN2, + "Mic-AN2 48V"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_SENS_IN3, + "Line-IN3 Sens."); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_SENS_IN4, + "Line-IN4 Sens."); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_PAD_AN1, + "Mic-AN1 PAD"); + if (err < 0) + return err; + + err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG2, + SND_BBFPRO_CTL_REG2_PAD_AN2, + "Mic-AN2 PAD"); + if (err < 0) + return err; + + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2286,6 +2701,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ err = snd_sc1810_init_mixer(mixer); break; + case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ + err = snd_bbfpro_controls_create(mixer); + break; } return err; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 103/388] ASoC: meson: add missing free_irq() in error path
by Sasha Levin 18 Jun '20

18 Jun '20

From: "Pavel Machek (CIP)" <pavel(a)denx.de> [ Upstream commit 3b8a299a58b2afce464ae11324b59dcf0f1d10a7 ] free_irq() is missing in case of error, fix that. Signed-off-by: Pavel Machek (CIP) <pavel(a)denx.de> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200606153103.GA17905@amd Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/axg-fifo.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index 2e9b56b29d31..b2e867113226 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -249,7 +249,7 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Enable pclk to access registers and clock the fifo ip */ ret = clk_prepare_enable(fifo->pclk); if (ret) - return ret; + goto free_irq; /* Setup status2 so it reports the memory pointer */ regmap_update_bits(fifo->map, FIFO_CTRL1, @@ -269,8 +269,14 @@ int axg_fifo_pcm_open(struct snd_soc_component *component, /* Take memory arbitror out of reset */ ret = reset_control_deassert(fifo->arb); if (ret) - clk_disable_unprepare(fifo->pclk); + goto free_clk; + + return 0; +free_clk: + clk_disable_unprepare(fifo->pclk); +free_irq: + free_irq(fifo->irq, ss); return ret; } EXPORT_SYMBOL_GPL(axg_fifo_pcm_open); -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 094/388] ASoC: meson: fix memory leak of links if allocation of ldata fails
by Sasha Levin 18 Jun '20

18 Jun '20

From: Colin Ian King <colin.king(a)canonical.com> [ Upstream commit 6e801dc411329aff592fbd48fb116183d0acdb00 ] Currently if the allocation of ldata fails the error return path does not kfree the allocated links object. Fix this by adding an error exit return path that performs the necessary kfree'ing. Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> Addresses-Coverity: ("Resource leak") Link: https://lore.kernel.org/r/20200604171216.60043-1-colin.king@canonical.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/meson/meson-card-utils.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 2ca8c98e204f..5a4a91c88734 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -49,19 +49,26 @@ int meson_card_reallocate_links(struct snd_soc_card *card, links = krealloc(priv->card.dai_link, num_links * sizeof(*priv->card.dai_link), GFP_KERNEL | __GFP_ZERO); + if (!links) + goto err_links; + ldata = krealloc(priv->link_data, num_links * sizeof(*priv->link_data), GFP_KERNEL | __GFP_ZERO); - - if (!links || !ldata) { - dev_err(priv->card.dev, "failed to allocate links\n"); - return -ENOMEM; - } + if (!ldata) + goto err_ldata; priv->card.dai_link = links; priv->link_data = ldata; priv->card.num_links = num_links; return 0; + +err_ldata: + kfree(links); +err_links: + dev_err(priv->card.dev, "failed to allocate links\n"); + return -ENOMEM; + } EXPORT_SYMBOL_GPL(meson_card_reallocate_links); -- 2.25.1

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[PATCH AUTOSEL 5.7 085/388] ASoC: qcom: q6asm-dai: kCFI fix
by Sasha Levin 18 Jun '20

18 Jun '20

From: John Stultz <john.stultz(a)linaro.org> [ Upstream commit a6b675a89e51a1cdad0481b809b7840d3f86e4b5 ] Fixes the following kCFI crash seen on db845c, caused by the function prototypes not matching the callback function prototype. [ 82.585661] Unable to handle kernel NULL pointer dereference at virtual address 0000000000000001 [ 82.595387] Mem abort info: [ 82.599463] ESR = 0x96000005 [ 82.602658] EC = 0x25: DABT (current EL), IL = 32 bits [ 82.608177] SET = 0, FnV = 0 [ 82.611829] EA = 0, S1PTW = 0 [ 82.615369] Data abort info: [ 82.618751] ISV = 0, ISS = 0x00000005 [ 82.622641] CM = 0, WnR = 0 [ 82.625774] user pgtable: 4k pages, 39-bit VAs, pgdp=0000000174259000 [ 82.632292] [0000000000000001] pgd=0000000000000000, pud=0000000000000000 [ 82.639167] Internal error: Oops: 96000005 [#1] PREEMPT SMP [ 82.644795] Modules linked in: hci_uart btqca xhci_plat_hcd xhci_pci_renesas xhci_pci xhci_hcd wcn36xx wcnss_ctrl wcd934x vctrl_regulator ufs_qcom syscon_reboot_e [ 82.644927] qcom_apcs_ipc_mailbox q6asm_dai q6routing q6asm q6afe_dai q6adm q6afe q6core q6dsp_common pm8941_pwrkey pm8916_wdt platform_mhu pinctrl_spmi_mpp pine [ 82.812982] CPU: 3 PID: 240 Comm: kworker/u16:4 Tainted: G W 5.6.0-rc7-mainline-00960-g0c34353d11b9-dirty #1 [ 82.824201] Hardware name: Thundercomm Dragonboard 845c (DT) [ 82.829937] Workqueue: qcom_apr_rx apr_rxwq [apr] [ 82.834698] pstate: 80c00005 (Nzcv daif +PAN +UAO) [ 82.839553] pc : __cfi_check_fail+0x4/0x1c [q6asm_dai] [ 82.844754] lr : __cfi_check+0x3a8/0x3b0 [q6asm_dai] [ 82.849767] sp : ffffffc0105f3c20 [ 82.853123] x29: ffffffc0105f3c30 x28: 0000000000000020 [ 82.858489] x27: ffffff80f4588400 x26: ffffff80f458ec94 [ 82.863854] x25: ffffff80f458ece8 x24: ffffffe3670c7000 [ 82.869220] x23: ffffff8094bb7b34 x22: ffffffe367137000 [ 82.874585] x21: bd07909b332eada6 x20: 0000000000000001 [ 82.879950] x19: ffffffe36713863c x18: ffffff80f8df4430 [ 82.885316] x17: 0000000000000001 x16: ffffffe39d15e660 [ 82.890681] x15: 0000000000000001 x14: 0000000000000027 [ 82.896047] x13: 0000000000000000 x12: ffffffe39e6465a0 [ 82.901413] x11: 0000000000000051 x10: 000000000000ffff [ 82.906779] x9 : 000ffffffe366c19 x8 : c3c5f18762d1ceef [ 82.912145] x7 : 0000000000000000 x6 : ffffffc010877698 [ 82.917511] x5 : ffffffc0105f3c00 x4 : 0000000000000000 [ 82.922877] x3 : 0000000000000000 x2 : 0000000000000001 [ 82.928243] x1 : ffffffe36713863c x0 : 0000000000000001 [ 82.933610] Call trace: [ 82.936099] __cfi_check_fail+0x4/0x1c [q6asm_dai] [ 82.940955] q6asm_srvc_callback+0x22c/0x618 [q6asm] [ 82.945973] apr_rxwq+0x1a8/0x27c [apr] [ 82.949861] process_one_work+0x2e8/0x54c [ 82.953919] worker_thread+0x27c/0x4d4 [ 82.957715] kthread+0x144/0x154 [ 82.960985] ret_from_fork+0x10/0x18 [ 82.964603] Code: a8c37bfd f85f8e5e d65f03c0 b40000a0 (39400008) [ 82.970762] ---[ end trace 410accb839617143 ]--- [ 82.975429] Kernel panic - not syncing: Fatal exception Signed-off-by: John Stultz <john.stultz(a)linaro.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Patrick Lai <plai(a)codeaurora.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Stephan Gerhold <stephan(a)gerhold.net> Cc: Sami Tolvanen <samitolvanen(a)google.com> Cc: Todd Kjos <tkjos(a)google.com> Cc: Alistair Delva <adelva(a)google.com> Cc: Amit Pundir <amit.pundir(a)linaro.org> Cc: Sumit Semwal <sumit.semwal(a)linaro.org> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20200529213823.98812-1-john.stultz@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 125af00bba53..4640804aab7f 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -176,7 +176,7 @@ static const struct snd_compr_codec_caps q6asm_compr_caps = { }; static void event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + void *payload, void *priv) { struct q6asm_dai_rtd *prtd = priv; struct snd_pcm_substream *substream = prtd->substream; @@ -490,7 +490,7 @@ static int q6asm_dai_hw_params(struct snd_soc_component *component, } static void compress_event_handler(uint32_t opcode, uint32_t token, - uint32_t *payload, void *priv) + void *payload, void *priv) { struct q6asm_dai_rtd *prtd = priv; struct snd_compr_stream *substream = prtd->cstream; -- 2.25.1

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[PATCH AUTOSEL 5.7 026/388] ALSA: isa/wavefront: prevent out of bounds write in ioctl
by Sasha Levin 18 Jun '20

18 Jun '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ] The "header->number" comes from the ioctl and it needs to be clamped to prevent out of bounds writes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/wavefront/wavefront_synth.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index c5b1d5900eed..d6420d224d09 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1171,7 +1171,10 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) "alias for %d\n", header->number, header->hdr.a.OriginalSample); - + + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + munge_int32 (header->number, &alias_hdr[0], 2); munge_int32 (header->hdr.a.OriginalSample, &alias_hdr[2], 2); munge_int32 (*((unsigned int *)&header->hdr.a.sampleStartOffset), @@ -1202,6 +1205,9 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) int num_samples; unsigned char *msample_hdr; + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + msample_hdr = kmalloc(WF_MSAMPLE_BYTES, GFP_KERNEL); if (! msample_hdr) return -ENOMEM; -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 025/388] ALSA: hda/realtek - Introduce polarity for micmute LED GPIO
by Sasha Levin 18 Jun '20

18 Jun '20

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit dbd13179780555ecd3c992dea1222ca31920e892 ] Currently mute LED and micmute LED share the same GPIO polarity. So split the polarity for mute and micmute, in case they have different polarities. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2c4575909441..e057ecb5a904 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -81,6 +81,7 @@ struct alc_spec { /* mute LED for HP laptops, see alc269_fixup_mic_mute_hook() */ int mute_led_polarity; + int micmute_led_polarity; hda_nid_t mute_led_nid; hda_nid_t cap_mute_led_nid; @@ -4080,11 +4081,9 @@ static void alc269_fixup_hp_mute_led_mic3(struct hda_codec *codec, /* update LED status via GPIO */ static void alc_update_gpio_led(struct hda_codec *codec, unsigned int mask, - bool enabled) + int polarity, bool enabled) { - struct alc_spec *spec = codec->spec; - - if (spec->mute_led_polarity) + if (polarity) enabled = !enabled; alc_update_gpio_data(codec, mask, !enabled); /* muted -> LED on */ } @@ -4095,7 +4094,8 @@ static void alc_fixup_gpio_mute_hook(void *private_data, int enabled) struct hda_codec *codec = private_data; struct alc_spec *spec = codec->spec; - alc_update_gpio_led(codec, spec->gpio_mute_led_mask, enabled); + alc_update_gpio_led(codec, spec->gpio_mute_led_mask, + spec->mute_led_polarity, enabled); } /* turn on/off mic-mute LED via GPIO per capture hook */ @@ -4104,6 +4104,7 @@ static void alc_gpio_micmute_update(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_update_gpio_led(codec, spec->gpio_mic_led_mask, + spec->micmute_led_polarity, spec->gen.micmute_led.led_value); } @@ -5808,7 +5809,8 @@ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, snd_hda_gen_hp_automute(codec, jack); /* mute_led_polarity is set to 0, so we pass inverted value here */ - alc_update_gpio_led(codec, 0x10, !spec->gen.hp_jack_present); + alc_update_gpio_led(codec, 0x10, spec->mute_led_polarity, + !spec->gen.hp_jack_present); } /* Manage GPIOs for HP EliteBook Folio 9480m. -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 020/388] ASoC: codecs: wm97xx: fix ac97 dependency
by Sasha Levin 18 Jun '20

18 Jun '20

From: Arnd Bergmann <arnd(a)arndb.de> [ Upstream commit ee2cbe06935bfa58f1fe07dc2a2283945f4b97dc ] A recent build fix got the dependency slightly wrong, breaking builds with CONFIG_AC97_BUS_NEW: WARNING: unmet direct dependencies detected for SND_SOC_WM9713 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AC97_BUS [=n] Selected by [m]: - SND_SOC_ZYLONITE [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_PXA2XX_SOC [=m] && MACH_ZYLONITE [=y] && AC97_BUS [=n]=n WARNING: unmet direct dependencies detected for SND_SOC_WM9712 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AC97_BUS [=n] Selected by [m]: - SND_PXA2XX_SOC_EM_X270 [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_PXA2XX_SOC [=m] && (MACH_EM_X270 [=n] || MACH_EXEDA [=n] || MACH_CM_X300 [=y]) && AC97_BUS [=n]=n Change the dependency to allow either version of the AC97 library code. Fixes: 5a309875787d ("ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200428212721.2877627-1-arnd@arndb.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/Kconfig | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e60e0b6a689c..8cdc68c141dc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1620,19 +1620,19 @@ config SND_SOC_WM9090 config SND_SOC_WM9705 tristate - depends on SND_SOC_AC97_BUS + depends on SND_SOC_AC97_BUS || AC97_BUS_NEW select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9712 tristate - depends on SND_SOC_AC97_BUS + depends on SND_SOC_AC97_BUS || AC97_BUS_NEW select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9713 tristate - depends on SND_SOC_AC97_BUS + depends on SND_SOC_AC97_BUS || AC97_BUS_NEW select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW -- 2.25.1

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[PATCH AUTOSEL 5.7 016/388] ASoC: davinci-mcasp: Fix dma_chan refcnt leak when getting dma type
by Sasha Levin 18 Jun '20

18 Jun '20

From: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> [ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ] davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a reference of the specified dma_chan object to "chan" with increased refcnt. When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes invalid, so the refcount should be decreased to keep refcount balanced. The reference counting issue happens in one exception handling path of davinci_mcasp_get_dma_type(). When chan device is NULL, the function forgets to decrease the refcnt increased by dma_request_chan(), causing a refcnt leak. Fix this issue by calling dma_release_channel() when chan device is NULL. Signed-off-by: Xiyu Yang <xiyuyang19(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fuda… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 734ffe925c4d..7a7db743dc5b 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -1896,8 +1896,10 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) PTR_ERR(chan)); return PTR_ERR(chan); } - if (WARN_ON(!chan->device || !chan->device->dev)) + if (WARN_ON(!chan->device || !chan->device->dev)) { + dma_release_channel(chan); return -EINVAL; + } if (chan->device->dev->of_node) ret = of_property_read_string(chan->device->dev->of_node, -- 2.25.1

1 0

[PATCH AUTOSEL 5.7 013/388] ASoC: fsl_esai: Disable exception interrupt before scheduling tasklet
by Sasha Levin 18 Jun '20

18 Jun '20

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 1fecbb71fe0e46b886f84e3b6decca6643c3af6d ] Disable exception interrupt before scheduling tasklet, otherwise if the tasklet isn't handled immediately, there will be endless xrun interrupt. Fixes: 7ccafa2b3879 ("ASoC: fsl_esai: recover the channel swap after xrun") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Link: https://lore.kernel.org/r/a8f2ad955aac9e52587beedc1133b3efbe746895.15879688… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_esai.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index c7a49d03463a..84290be778f0 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -87,6 +87,10 @@ static irqreturn_t esai_isr(int irq, void *devid) if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) && esai_priv->reset_at_xrun) { dev_dbg(&pdev->dev, "reset module for xrun\n"); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xEIE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xEIE_MASK, 0); tasklet_schedule(&esai_priv->task); } -- 2.25.1

1 0