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

[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

2 2

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

1 0

[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

1 0

[PATCH AUTOSEL 4.4 20/38] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 366813f1a5f80..59f3a9492d74b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -242,21 +242,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -489,6 +474,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

1 0

[PATCH AUTOSEL 4.9 47/48] 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 ba0c3a381933e..53eb05d0722b0 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -610,11 +610,11 @@ 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

1 0

[PATCH AUTOSEL 4.9 22/48] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 95d02e25a3133..77c704847e4ac 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -242,21 +242,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -499,6 +484,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

1 0

[PATCH AUTOSEL 4.14 65/66] 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 21d9b7d96eb0f..35da926507ac3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -642,11 +642,11 @@ 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

1 0

[PATCH AUTOSEL 4.14 31/66] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 6caf94581a0e8..1ed4a7c094c48 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -242,21 +242,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -499,6 +484,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

1 0

[PATCH AUTOSEL 4.19 85/87] 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 f86b9b0a0607e..88d7b10375221 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -642,11 +642,11 @@ 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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[PATCH AUTOSEL 4.19 67/87] ASoC: Intel: bytcr_rt5640: Add quirk for ARCHOS Cesium 140
by Sasha Levin 23 Dec '20

23 Dec '20

From: Chris Chiu <chiu(a)endlessos.org> [ Upstream commit 1bea2256aa96a2d7b1b576eb74e29d79edc9bea8 ] Tha ARCHOS Cesium 140 tablet has problem with the jack-sensing, thus the heaset functions are not working. Add quirk for this model to select the correct input map, jack-detect options and channel map to enable jack sensing and headset microphone. This device uses IN1 for its internal MIC and JD2 for jack-detect. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20201208060414.27646-1-chiu@endlessos.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index ec630127ef2f3..f23ddb9f810ae 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -431,6 +431,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 140 CESIUM"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.27.0

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[PATCH AUTOSEL 4.19 61/87] ALSA: rawmidi: Access runtime->avail always in spinlock
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 88a06d6fd6b369d88cec46c62db3e2604a2f50d5 ] The runtime->avail field may be accessed concurrently while some places refer to it without taking the runtime->lock spinlock, as detected by KCSAN. Usually this isn't a big problem, but for consistency and safety, we should take the spinlock at each place referencing this field. Reported-by: syzbot+a23a6f1215c84756577c(a)syzkaller.appspotmail.com Reported-by: syzbot+3d367d1df1d2b67f5c19(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20201206083527.21163-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/rawmidi.c | 49 +++++++++++++++++++++++++++++++------------- 1 file changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 9b26973fe697a..f4f855d7a7910 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -87,11 +87,21 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) } } -static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) +static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) +{ + return runtime->avail >= runtime->avail_min; +} + +static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + bool ready; - return runtime->avail >= runtime->avail_min; + spin_lock_irqsave(&runtime->lock, flags); + ready = __snd_rawmidi_ready(runtime); + spin_unlock_irqrestore(&runtime->lock, flags); + return ready; } static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream, @@ -960,7 +970,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, if (result > 0) { if (runtime->event) schedule_work(&runtime->event_work); - else if (snd_rawmidi_ready(substream)) + else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } spin_unlock_irqrestore(&runtime->lock, flags); @@ -1039,7 +1049,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun result = 0; while (count > 0) { spin_lock_irq(&runtime->lock); - while (!snd_rawmidi_ready(substream)) { + while (!__snd_rawmidi_ready(runtime)) { wait_queue_entry_t wait; if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { @@ -1056,9 +1066,11 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_read1(substream, @@ -1196,7 +1208,7 @@ int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int coun runtime->avail += count; substream->bytes += count; if (count > 0) { - if (runtime->drain || snd_rawmidi_ready(substream)) + if (runtime->drain || __snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } return count; @@ -1363,9 +1375,11 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail && !timeout) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail && !timeout) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); @@ -1445,6 +1459,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; + unsigned long buffer_size, avail, xruns; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1463,13 +1478,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" " Avail : %lu\n", runtime->oss ? "OSS compatible" : "native", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail); + buffer_size, avail); } } } @@ -1487,13 +1505,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + xruns = runtime->xruns; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" " Overruns : %lu\n", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail, - (unsigned long) runtime->xruns); + buffer_size, avail, xruns); } } } -- 2.27.0

1 0

[PATCH AUTOSEL 4.19 37/87] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e4d2fcc89c306..5ff51c2983a19 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -243,21 +243,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -501,6 +486,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 126/130] 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 df4771b9eff24..0faee80e28b66 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -645,11 +645,11 @@ 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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[PATCH AUTOSEL 5.4 098/130] ASoC: Intel: bytcr_rt5640: Add quirk for ARCHOS Cesium 140
by Sasha Levin 23 Dec '20

23 Dec '20

From: Chris Chiu <chiu(a)endlessos.org> [ Upstream commit 1bea2256aa96a2d7b1b576eb74e29d79edc9bea8 ] Tha ARCHOS Cesium 140 tablet has problem with the jack-sensing, thus the heaset functions are not working. Add quirk for this model to select the correct input map, jack-detect options and channel map to enable jack sensing and headset microphone. This device uses IN1 for its internal MIC and JD2 for jack-detect. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20201208060414.27646-1-chiu@endlessos.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 6012367f6fe48..6b6749550fc4f 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -422,6 +422,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 140 CESIUM"), + }, + .driver_data = (void *)(BYT_RT5640_IN1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_2000UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."), -- 2.27.0

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[PATCH AUTOSEL 5.4 095/130] ASoC: Intel: cht_bsw_nau8824: Change SSP2-Codec DAI id to 0
by Sasha Levin 23 Dec '20

23 Dec '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 748e72e869718db8d735d773040bce95158c98c6 ] The snd-soc-sst-acpi driver does not care about the id specified for the SSP2-Codec DAI, but it does matter for the snd-sof-acpi driver; and when it is not 0 then the snd-sof-acpi driver does not work. Set the SSP2-Codec DAI id to 0, fixing the snd-sof-acpi driver not working on devices using the cht_bsw_nau8824 machine-driver. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20201206122436.13553-3-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 501bad3976fbf..bea1372b416e2 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -218,7 +218,7 @@ static struct snd_soc_dai_link cht_dailink[] = { { /* SSP2 - Codec */ .name = "SSP2-Codec", - .id = 1, + .id = 0, .no_pcm = 1, .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS, -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 087/130] ALSA: rawmidi: Access runtime->avail always in spinlock
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 88a06d6fd6b369d88cec46c62db3e2604a2f50d5 ] The runtime->avail field may be accessed concurrently while some places refer to it without taking the runtime->lock spinlock, as detected by KCSAN. Usually this isn't a big problem, but for consistency and safety, we should take the spinlock at each place referencing this field. Reported-by: syzbot+a23a6f1215c84756577c(a)syzkaller.appspotmail.com Reported-by: syzbot+3d367d1df1d2b67f5c19(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20201206083527.21163-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/rawmidi.c | 49 +++++++++++++++++++++++++++++++------------- 1 file changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 94db4683cfaff..6a3543b8455fc 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -72,11 +72,21 @@ static inline unsigned short snd_rawmidi_file_flags(struct file *file) } } -static inline int snd_rawmidi_ready(struct snd_rawmidi_substream *substream) +static inline bool __snd_rawmidi_ready(struct snd_rawmidi_runtime *runtime) +{ + return runtime->avail >= runtime->avail_min; +} + +static bool snd_rawmidi_ready(struct snd_rawmidi_substream *substream) { struct snd_rawmidi_runtime *runtime = substream->runtime; + unsigned long flags; + bool ready; - return runtime->avail >= runtime->avail_min; + spin_lock_irqsave(&runtime->lock, flags); + ready = __snd_rawmidi_ready(runtime); + spin_unlock_irqrestore(&runtime->lock, flags); + return ready; } static inline int snd_rawmidi_ready_append(struct snd_rawmidi_substream *substream, @@ -945,7 +955,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, if (result > 0) { if (runtime->event) schedule_work(&runtime->event_work); - else if (snd_rawmidi_ready(substream)) + else if (__snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } spin_unlock_irqrestore(&runtime->lock, flags); @@ -1024,7 +1034,7 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun result = 0; while (count > 0) { spin_lock_irq(&runtime->lock); - while (!snd_rawmidi_ready(substream)) { + while (!__snd_rawmidi_ready(runtime)) { wait_queue_entry_t wait; if ((file->f_flags & O_NONBLOCK) != 0 || result > 0) { @@ -1041,9 +1051,11 @@ static ssize_t snd_rawmidi_read(struct file *file, char __user *buf, size_t coun return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_read1(substream, @@ -1181,7 +1193,7 @@ int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int coun runtime->avail += count; substream->bytes += count; if (count > 0) { - if (runtime->drain || snd_rawmidi_ready(substream)) + if (runtime->drain || __snd_rawmidi_ready(runtime)) wake_up(&runtime->sleep); } return count; @@ -1370,9 +1382,11 @@ static ssize_t snd_rawmidi_write(struct file *file, const char __user *buf, return -ENODEV; if (signal_pending(current)) return result > 0 ? result : -ERESTARTSYS; - if (!runtime->avail && !timeout) - return result > 0 ? result : -EIO; spin_lock_irq(&runtime->lock); + if (!runtime->avail && !timeout) { + spin_unlock_irq(&runtime->lock); + return result > 0 ? result : -EIO; + } } spin_unlock_irq(&runtime->lock); count1 = snd_rawmidi_kernel_write1(substream, buf, NULL, count); @@ -1452,6 +1466,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; + unsigned long buffer_size, avail, xruns; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1470,13 +1485,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Mode : %s\n" " Buffer size : %lu\n" " Avail : %lu\n", runtime->oss ? "OSS compatible" : "native", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail); + buffer_size, avail); } } } @@ -1494,13 +1512,16 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Owner PID : %d\n", pid_vnr(substream->pid)); runtime = substream->runtime; + spin_lock_irq(&runtime->lock); + buffer_size = runtime->buffer_size; + avail = runtime->avail; + xruns = runtime->xruns; + spin_unlock_irq(&runtime->lock); snd_iprintf(buffer, " Buffer size : %lu\n" " Avail : %lu\n" " Overruns : %lu\n", - (unsigned long) runtime->buffer_size, - (unsigned long) runtime->avail, - (unsigned long) runtime->xruns); + buffer_size, avail, xruns); } } } -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 056/130] ALSA: usb-audio: Don't call usb_set_interface() at trigger callback
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4974b7950929e4a28d4eaee48e4ad07f168ac132 ] The PCM trigger callback is atomic, hence we must not call a function like usb_set_interface() there. Calling it from there would lead to a kernel Oops. Fix it by moving the usb_set_interface() call to set_sync_endpoint(). Also, apply the snd_usb_set_interface_quirk() for consistency, too. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 28 +++++++++++++--------------- 1 file changed, 13 insertions(+), 15 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 49ad4e7bb70b5..87389ab69b5ee 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -232,21 +232,6 @@ static int start_endpoints(struct snd_usb_substream *subs) !test_and_set_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags)) { struct snd_usb_endpoint *ep = subs->sync_endpoint; - if (subs->data_endpoint->iface != subs->sync_endpoint->iface || - subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { - err = usb_set_interface(subs->dev, - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting); - if (err < 0) { - clear_bit(SUBSTREAM_FLAG_SYNC_EP_STARTED, &subs->flags); - dev_err(&subs->dev->dev, - "%d:%d: cannot set interface (%d)\n", - subs->sync_endpoint->iface, - subs->sync_endpoint->altsetting, err); - return -EIO; - } - } - dev_dbg(&subs->dev->dev, "Starting sync EP @%p\n", ep); ep->sync_slave = subs->data_endpoint; @@ -512,6 +497,19 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, subs->data_endpoint->sync_master = subs->sync_endpoint; + if (subs->data_endpoint->iface != subs->sync_endpoint->iface || + subs->data_endpoint->altsetting != subs->sync_endpoint->altsetting) { + err = usb_set_interface(subs->dev, + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + if (err < 0) + return err; + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + subs->sync_endpoint->iface, + subs->sync_endpoint->altsetting); + snd_usb_set_interface_quirk(dev); + } + return 0; } -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 055/130] ALSA: usb-audio: Handle discrete rates properly in hw constraints
by Sasha Levin 23 Dec '20

23 Dec '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit bc4e94aa8e72e79598e63a0b73febdcd8aeb541f ] In the current code, when the device provides the discrete sample rate tables with unusual sample rates, the driver tries to gather the whole values from the audioformat entries and create a hw-constraint rule to restrict with this single rate list. This is rather inefficient and may overlook the rates that are associated only with the certain audioformat entries. This patch improves the hw constraint setup by rewriting the existing hw_rule_rate(). The discrete sample rates (identified by rate_table and nr_rates of format entry) are checked in the existing hw_rule_rate() instead of extra rules; in the case of discrete rates, the function compares with each rate table entry and calculates the min/max values from there. For the contiguous rates, the behavior doesn't change. Along with it, snd_usb_pcm_check_knot() and snb_usb_substream rate_list field become superfluous, thus those are dropped. Tested-by: Keith Milner <kamilner(a)superlative.org> Tested-by: Dylan Robinson <dylan_robinson(a)motu.com> Link: https://lore.kernel.org/r/20201123085347.19667-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.h | 1 - sound/usb/pcm.c | 73 ++++++++++------------------------------------ sound/usb/stream.c | 1 - 3 files changed, 15 insertions(+), 60 deletions(-) diff --git a/sound/usb/card.h b/sound/usb/card.h index d8ec5caf464de..d619e5e77a305 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -153,7 +153,6 @@ struct snd_usb_substream { u64 formats; /* format bitmasks (all or'ed) */ unsigned int num_formats; /* number of supported audio formats (list) */ struct list_head fmt_list; /* format list */ - struct snd_pcm_hw_constraint_list rate_list; /* limited rates */ spinlock_t lock; int last_frame_number; /* stored frame number */ diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 1a5e555002b2b..49ad4e7bb70b5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1034,27 +1034,31 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_usb_substream *subs = rule->private; struct audioformat *fp; struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); - unsigned int rmin, rmax; + unsigned int rmin, rmax, r; int changed; + int i; hwc_debug("hw_rule_rate: (%d,%d)\n", it->min, it->max); - changed = 0; - rmin = rmax = 0; + rmin = UINT_MAX; + rmax = 0; list_for_each_entry(fp, &subs->fmt_list, list) { if (!hw_check_valid_format(subs, params, fp)) continue; - if (changed++) { - if (rmin > fp->rate_min) - rmin = fp->rate_min; - if (rmax < fp->rate_max) - rmax = fp->rate_max; + if (fp->rate_table && fp->nr_rates) { + for (i = 0; i < fp->nr_rates; i++) { + r = fp->rate_table[i]; + if (!snd_interval_test(it, r)) + continue; + rmin = min(rmin, r); + rmax = max(rmax, r); + } } else { - rmin = fp->rate_min; - rmax = fp->rate_max; + rmin = min(rmin, fp->rate_min); + rmax = max(rmax, fp->rate_max); } } - if (!changed) { + if (rmin > rmax) { hwc_debug(" --> get empty\n"); it->empty = 1; return -EINVAL; @@ -1200,50 +1204,6 @@ static int hw_rule_period_time(struct snd_pcm_hw_params *params, return changed; } -/* - * If the device supports unusual bit rates, does the request meet these? - */ -static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, - struct snd_usb_substream *subs) -{ - struct audioformat *fp; - int *rate_list; - int count = 0, needs_knot = 0; - int err; - - kfree(subs->rate_list.list); - subs->rate_list.list = NULL; - - list_for_each_entry(fp, &subs->fmt_list, list) { - if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) - return 0; - count += fp->nr_rates; - if (fp->rates & SNDRV_PCM_RATE_KNOT) - needs_knot = 1; - } - if (!needs_knot) - return 0; - - subs->rate_list.list = rate_list = - kmalloc_array(count, sizeof(int), GFP_KERNEL); - if (!subs->rate_list.list) - return -ENOMEM; - subs->rate_list.count = count; - subs->rate_list.mask = 0; - count = 0; - list_for_each_entry(fp, &subs->fmt_list, list) { - int i; - for (i = 0; i < fp->nr_rates; i++) - rate_list[count++] = fp->rate_table[i]; - } - err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &subs->rate_list); - if (err < 0) - return err; - - return 0; -} - /* * set up the runtime hardware information. @@ -1333,9 +1293,6 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; } - err = snd_usb_pcm_check_knot(runtime, subs); - if (err < 0) - return err; return snd_usb_autoresume(subs->stream->chip); } diff --git a/sound/usb/stream.c b/sound/usb/stream.c index d01edd5da6cf8..49c1b8a208582 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -47,7 +47,6 @@ static void free_substream(struct snd_usb_substream *subs) return; /* not initialized */ list_for_each_entry_safe(fp, n, &subs->fmt_list, list) audioformat_free(fp); - kfree(subs->rate_list.list); kfree(subs->str_pd); snd_media_stream_delete(subs); } -- 2.27.0

1 0

[PATCH AUTOSEL 5.4 051/130] ASoC: SOF: IPC: fix implicit type overflow
by Sasha Levin 23 Dec '20

23 Dec '20

From: Curtis Malainey <cujomalainey(a)chromium.org> [ Upstream commit 7c1d0e554a359cca77bfabd2a29b06f5322d172d ] Implicit values may have a length of 15bits (s16) so we need to declare the proper size so we don't get undefined behaviour. This appears to be arch and compiler dependent. This commit is to keep the headers aligned between the firmware and kernel. UBSan discovered this bug in the firmware. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20201120144025.2166023-1-kai.vehmanen@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/sof/header.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index 10f00c08dbb7a..aae673b2bb5e2 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -30,12 +30,12 @@ /* Global Message - Generic */ #define SOF_GLB_TYPE_SHIFT 28 -#define SOF_GLB_TYPE_MASK (0xf << SOF_GLB_TYPE_SHIFT) +#define SOF_GLB_TYPE_MASK (0xfL << SOF_GLB_TYPE_SHIFT) #define SOF_GLB_TYPE(x) ((x) << SOF_GLB_TYPE_SHIFT) /* Command Message - Generic */ #define SOF_CMD_TYPE_SHIFT 16 -#define SOF_CMD_TYPE_MASK (0xfff << SOF_CMD_TYPE_SHIFT) +#define SOF_CMD_TYPE_MASK (0xfffL << SOF_CMD_TYPE_SHIFT) #define SOF_CMD_TYPE(x) ((x) << SOF_CMD_TYPE_SHIFT) /* Global Message Types */ -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 070/217] ASoC: Intel: sof_sdw: add quirk for new TigerLake-SDCA device
by Sasha Levin 23 Dec '20

23 Dec '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 488cdbd8931fe4bc7f374a8b429e81d0e4b7ac76 ] Add quirks for jack detection, rt715 DAI and number of speakers. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Link: https://lore.kernel.org/r/20201111214318.150529-2-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b29946eb43551..ca968901ac96f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -48,6 +48,16 @@ static int sof_sdw_quirk_cb(const struct dmi_system_id *id) } static const struct dmi_system_id sof_sdw_quirk_table[] = { + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0A32") + }, + .driver_data = (void *)(SOF_RT711_JD_SRC_JD2 | + SOF_RT715_DAI_ID_FIX | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 066/217] ASoC: Fix vaud18 power leakage of mt6359
by Sasha Levin 23 Dec '20

23 Dec '20

From: "Shane.Chien" <shane.chien(a)mediatek.com> [ Upstream commit 64a70744b77898a15d7a5b2b4dc0fa9523a75cde ] vaud18 is power of mt6359 audio path. It should only enable when audio is used, instead of in boot up stage. Once mt6359 audio path is enabled or disabled, vaud18 is controlled by regulator supply widget "LDO_VAUD18". Due to vaud18 is controlled by regulator dapm macro instead of regmap, the macro MT6359_LDO_VAUD18_CON0 and variable avdd_reg is no used and removed from mt6359.h. Signed-off-by: Shane.Chien <shane.chien(a)mediatek.com> Link: https://lore.kernel.org/r/1604975492-6142-2-git-send-email-shane.chien@medi… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/mt6359.c | 25 +------------------------ sound/soc/codecs/mt6359.h | 8 -------- 2 files changed, 1 insertion(+), 32 deletions(-) diff --git a/sound/soc/codecs/mt6359.c b/sound/soc/codecs/mt6359.c index 81aafb553bdd9..64b26ab969ed8 100644 --- a/sound/soc/codecs/mt6359.c +++ b/sound/soc/codecs/mt6359.c @@ -1833,9 +1833,7 @@ static const struct snd_soc_dapm_widget mt6359_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY_S("CLK_BUF", SUPPLY_SEQ_CLK_BUF, MT6359_DCXO_CW12, RG_XO_AUDIO_EN_M_SFT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY_S("LDO_VAUD18", SUPPLY_SEQ_LDO_VAUD18, - MT6359_LDO_VAUD18_CON0, - RG_LDO_VAUD18_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("LDO_VAUD18", 0, 0), SND_SOC_DAPM_SUPPLY_S("AUDGLB", SUPPLY_SEQ_AUD_GLB, MT6359_AUDDEC_ANA_CON13, RG_AUDGLB_PWRDN_VA32_SFT, 1, NULL, 0), @@ -2697,20 +2695,6 @@ static int mt6359_platform_driver_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, priv); priv->dev = &pdev->dev; - priv->avdd_reg = devm_regulator_get(&pdev->dev, "vaud18"); - if (IS_ERR(priv->avdd_reg)) { - dev_err(&pdev->dev, "%s(), have no vaud18 supply: %ld", - __func__, PTR_ERR(priv->avdd_reg)); - return PTR_ERR(priv->avdd_reg); - } - - ret = regulator_enable(priv->avdd_reg); - if (ret) { - dev_err(&pdev->dev, "%s(), failed to enable regulator!\n", - __func__); - return ret; - } - ret = mt6359_parse_dt(priv); if (ret) { dev_warn(&pdev->dev, "%s() failed to parse dts\n", __func__); @@ -2731,13 +2715,6 @@ static int mt6359_platform_driver_remove(struct platform_device *pdev) dev_dbg(&pdev->dev, "%s(), dev name %s\n", __func__, dev_name(&pdev->dev)); - ret = regulator_disable(priv->avdd_reg); - if (ret) { - dev_err(&pdev->dev, "%s(), failed to disable regulator!\n", - __func__); - return ret; - } - return 0; } diff --git a/sound/soc/codecs/mt6359.h b/sound/soc/codecs/mt6359.h index 3792e534a91b6..e0aebf639e461 100644 --- a/sound/soc/codecs/mt6359.h +++ b/sound/soc/codecs/mt6359.h @@ -135,11 +135,6 @@ /* MT6359_DCXO_CW12 */ #define RG_XO_AUDIO_EN_M_SFT 13 -/* LDO_VAUD18_CON0 */ -#define RG_LDO_VAUD18_EN_SFT 0 -#define RG_LDO_VAUD18_EN_MASK 0x1 -#define RG_LDO_VAUD18_EN_MASK_SFT (0x1 << 0) - /* AUD_TOP_CKPDN_CON0 */ #define RG_VOW13M_CK_PDN_SFT 13 #define RG_VOW13M_CK_PDN_MASK 0x1 @@ -2132,7 +2127,6 @@ #define MT6359_DCXO_CW11 0x7a6 #define MT6359_DCXO_CW12 0x7a8 -#define MT6359_LDO_VAUD18_CON0 0x1c98 #define MT6359_GPIO_MODE0 0xcc #define MT6359_GPIO_MODE0_SET 0xce @@ -2469,7 +2463,6 @@ enum { enum { /* common */ SUPPLY_SEQ_CLK_BUF, - SUPPLY_SEQ_LDO_VAUD18, SUPPLY_SEQ_AUD_GLB, SUPPLY_SEQ_HP_PULL_DOWN, SUPPLY_SEQ_CLKSQ, @@ -2629,7 +2622,6 @@ struct mt6359_priv { int hp_gain_ctl; int hp_hifi_mode; int mtkaif_protocol; - struct regulator *avdd_reg; }; #define CODEC_MT6359_NAME "mtk-codec-mt6359" -- 2.27.0

1 0

[PATCH AUTOSEL 5.10 013/217] ASoC: intel: sof_rt5682: Add quirk for Dooly
by Sasha Levin 23 Dec '20

23 Dec '20

From: Brent Lu <brent.lu(a)intel.com> [ Upstream commit bdd088ce5bfd32b95ab1bd90b49405e7c1f1fff5 ] This DMI product family string of this board is "Google_Hatch" so the DMI quirk will take place. However, this board is using rt1015 speaker amp instead of max98357a specified in the quirk. Therefore, we need an new DMI quirk for this board. Signed-off-by: Brent Lu <brent.lu(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20201030170559.20370-3-brent.lu@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ddbb9fe7cc06b..c6c33910bc354 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -99,6 +99,24 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(1)), }, + { + /* + * Dooly is hatch family but using rt1015 amp so it + * requires a quirk before "Google_Hatch". + */ + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HP"), + DMI_MATCH(DMI_PRODUCT_NAME, "Dooly"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1015_SPEAKER_AMP_PRESENT | + SOF_RT1015_SPEAKER_AMP_100FS | + SOF_RT5682_SSP_AMP(1)), + }, { .callback = sof_rt5682_quirk_cb, .matches = { -- 2.27.0

1 0

UBSAN: shift-out-of-bounds in snd_usbmidi_get_ms_info
by syzbot 22 Dec '20

22 Dec '20

Hello, syzbot found the following issue on: HEAD commit: 8653b778 Merge tag 'clk-for-linus' of git://git.kernel.org.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=109531a3500000 kernel config: https://syzkaller.appspot.com/x/.config?x=98408202fed1f636 dashboard link: https://syzkaller.appspot.com/bug?extid=92e45ae45543f89e8c88 compiler: gcc (GCC) 10.1.0-syz 20200507 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=1421ec47500000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=12bfa077500000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+92e45ae45543f89e8c88(a)syzkaller.appspotmail.com usb 1-1: config 0 interface 0 altsetting 0 bulk endpoint 0x8A has invalid maxpacket 31 usb 1-1: New USB device found, idVendor=324b, idProduct=5963, bcdDevice= 0.d2 usb 1-1: New USB device strings: Mfr=0, Product=0, SerialNumber=0 usb 1-1: config 0 descriptor?? usb 1-1: MIDIStreaming interface descriptor not found ================================================================================ UBSAN: shift-out-of-bounds in sound/usb/midi.c:1928:8 shift exponent 244 is too large for 32-bit type 'int' CPU: 1 PID: 17 Comm: kworker/1:0 Not tainted 5.10.0-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:79 [inline] dump_stack+0x107/0x163 lib/dump_stack.c:120 ubsan_epilogue+0xb/0x5a lib/ubsan.c:148 __ubsan_handle_shift_out_of_bounds.cold+0xb1/0x181 lib/ubsan.c:395 snd_usbmidi_get_ms_info.cold+0x83/0xd4 sound/usb/midi.c:1928 __snd_usbmidi_create+0x380/0x2320 sound/usb/midi.c:2404 snd_usb_create_quirk+0xa5/0xe0 sound/usb/quirks.c:562 usb_audio_probe+0x623/0x2ab0 sound/usb/card.c:795 --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

[PATCH 1/2] ALSA: hda/realtek: Apply jack fixup for Quanta NL3
by Chris Chiu 22 Dec '20

22 Dec '20

The Quanta NL3 laptop has both a headphone output jack and a headset jack, on the right edge of the chassis. The pin information suggests that both of these are at the Front. The PulseAudio is confused to differentiate them so one of the jack can neither get the jack sense working nor the audio output. The ALC269_FIXUP_LIFEBOOK chained with ALC269_FIXUP_QUANTA_MUTE can help to differentiate 2 jacks and get the 'Auto-Mute Mode' working correctly. Signed-off-by: Chris Chiu <chiu(a)endlessos.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8616c5624870..bb1010d78717 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8013,6 +8013,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1325, "System76 Darter Pro (darp5)", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.20.1

2 3

[PATCH] pcm: Fix a typo in SND_PCM_AUDIO_TSTAMP_TYPE_LAST definition
by Takashi Iwai 22 Dec '20

22 Dec '20

It was wrongly defined with SNDRV_ prefix. Fix it. Fixes: ea02dbd20a53 ("pcm: Add snd_pcm_audio_tstamp_type_t constants") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/pcm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/pcm.h b/include/pcm.h index cf1eea8bbc9b..e300b951c076 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -361,7 +361,7 @@ typedef enum _snd_pcm_audio_tstamp_type { SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ABSOLUTE = 3, /**< link time reported by sample or wallclock counter, not reset on startup */ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_ESTIMATED = 4, /**< link time estimated indirectly */ SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED = 5, /**< link time synchronized with system time */ - SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED + SND_PCM_AUDIO_TSTAMP_TYPE_LAST = SND_PCM_AUDIO_TSTAMP_TYPE_LINK_SYNCHRONIZED } snd_pcm_audio_tstamp_type_t; typedef struct _snd_pcm_audio_tstamp_config { -- 2.26.2

1 0

[PATCH 0/2] Publish snd_pcm_audio_tstamp_type_t constants
by David Henningsson 22 Dec '20

22 Dec '20

Hi, Sorry if this message reaches you twice. Apparently I tried to send this earlier from an address which I no longer has access to. :-( So if you sent a reply I could never read it. Hopefully it'll work better this time around. Anyhow, testing of these patches are limited to running "gitcompile", which does not seem to compile the tests, is there an easy command to compile them as well? David Henningsson (2): pcm: Add snd_pcm_audio_tstamp_type_t constants test/audio_time: Make use of SND_PCM_AUDIO_TSTAMP_TYPE constants include/pcm.h | 14 ++++++++++++++ test/audio_time.c | 26 +++++++++++++------------- 2 files changed, 27 insertions(+), 13 deletions(-) -- 2.25.1

2 3

[PATCH] ALSA: usb-audio: Add implicit feeback support for the BOSS GT-1
by Mike Oliphant 22 Dec '20

22 Dec '20

The BOSS GT-1 (USB ID 0582:01d6) requires implicit feedback like other similar BOSS devices. This patch adds this support. Signed-off-by: Mike Oliphant <oliphant(a)nostatic.org> --- sound/usb/implicit.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 4e911d200..d44997e49 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -77,6 +77,7 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { IMPLICIT_FB_SKIP_DEV(0x0582, 0x0189), /* BOSS GT-100v2 */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d8), /* BOSS Katana */ IMPLICIT_FB_SKIP_DEV(0x0582, 0x01e5), /* BOSS GT-001 */ + IMPLICIT_FB_SKIP_DEV(0x0582, 0x01d6), /* BOSS GT-1 */ {} /* terminator */ }; @@ -87,6 +88,7 @@ static const struct snd_usb_implicit_fb_match capture_implicit_fb_quirks[] = { IMPLICIT_FB_FIXED_DEV(0x0582, 0x0189, 0x0d, 0x01), /* BOSS GT-100v2 */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d8, 0x0d, 0x01), /* BOSS Katana */ IMPLICIT_FB_FIXED_DEV(0x0582, 0x01e5, 0x0d, 0x01), /* BOSS GT-001 */ + IMPLICIT_FB_FIXED_DEV(0x0582, 0x01d6, 0x0d, 0x01), /* BOSS GT-1 */ {} /* terminator */ }; -- 2.29.2

3 2

[PATCH v3 0/6] Add support for Rpi4b + Cirrus Lochnagar2 and CS47L15
by Richard Fitzgerald 21 Dec '20

21 Dec '20

This set of patches provides support for using the Raspberry Pi 4b with a Cirrus Logic Lochnagar 2 audio development platform plus Cirrus Logic CS47L15 codec. Patches are needed to audio-graph-card to enable support for setting the component sysclks and plls. These are not specific to CS47L15 - several Cirrus and non-Cirrus codecs also require component clock setup, typically for larger codecs with multiple internal clocking options and clock domains. The codec sysclks and plls cannot be placed under the clock framework because they are typically I2C/SPI/Soundwire-connected peripherals and access to the registers would cause a nested get of the bus clock. The clock framework does not support this and it would result in a deadlock. Richard Fitzgerald (6): of: base: Add of_count_phandle_with_fixed_args() ASoC: audio-graph-card: Add plls and sysclks DT bindings ASoC: audio-graph-card: Support setting component plls and sysclks ASoC: madera: Allow codecs to be selected from kernel config ASoC: madera: Export clock config defines to dt-bindings ARM: dts: Add dts for RPi4b + Cirrus Logic Lochnagar2 + CS47L15 .../bindings/sound/audio-graph-card.txt | 72 ++++++ MAINTAINERS | 1 + arch/arm/boot/dts/Makefile | 1 + ...bcm2711-rpi4b-cirrus-lochnagar-cs47l15.dts | 186 ++++++++++++++ .../dts/bcm2711-rpi4b-cirrus-lochnagar.dtsi | 201 +++++++++++++++ drivers/of/base.c | 73 ++++-- include/dt-bindings/sound/madera.h | 60 +++++ include/linux/of.h | 9 + include/sound/simple_card_utils.h | 25 ++ sound/soc/codecs/Kconfig | 10 +- sound/soc/codecs/madera.h | 56 +---- sound/soc/generic/audio-graph-card.c | 16 +- sound/soc/generic/simple-card-utils.c | 236 ++++++++++++++++++ 13 files changed, 861 insertions(+), 85 deletions(-) create mode 100644 arch/arm/boot/dts/bcm2711-rpi4b-cirrus-lochnagar-cs47l15.dts create mode 100644 arch/arm/boot/dts/bcm2711-rpi4b-cirrus-lochnagar.dtsi -- 2.20.1

4 10

[PATCH] ALSA: usb-audio: Add alias entry for ASUS PRIME TRX40 PRO-S
by Takashi Iwai 21 Dec '20

21 Dec '20

ASUS PRIME TRX40 PRO-S mobo with 0b05:1918 needs the same quirk alias for another ASUS mobo (0b05:1917) for the proper mixer mapping, etc. Add the corresponding entry. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210783 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index cb0b6582dfee..d731ca62d599 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -383,6 +383,9 @@ static const struct usb_audio_device_name usb_audio_names[] = { /* ASUS ROG Strix */ PROFILE_NAME(0x0b05, 0x1917, "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), + /* ASUS PRIME TRX40 PRO-S */ + PROFILE_NAME(0x0b05, 0x1918, + "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), /* Dell WD15 Dock */ PROFILE_NAME(0x0bda, 0x4014, "Dell", "WD15 Dock", "Dell-WD15-Dock"), -- 2.26.2

1 0

[PATCH 2/2] sound: Remove redundant comments
by hui yang 21 Dec '20

21 Dec '20

From: YangHui <yanghui.def(a)gmail.com> Remove redundant comments Signed-off-by: YangHui <yanghui.def(a)gmail.com> --- sound/core/init.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 764dbe6..75aec71 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -149,8 +149,6 @@ static void release_card_device(struct device *dev) * @extra_size: allocate this extra size after the main soundcard structure * @card_ret: the pointer to store the created card instance * - * Creates and initializes a soundcard structure. - * * The function allocates snd_card instance via kzalloc with the given * space for the driver to use freely. The allocated struct is stored * in the given card_ret pointer. -- 2.7.4

2 1

[PATCH] ALSA: hda/realtek: Add quirk for MSI-GP73
by Takashi Iwai 20 Dec '20

20 Dec '20

MSI-GP73 (with SSID 1462:1229) requires yet again ALC1220_FIXUP_CLEVO_P950 quirk like other MSI models. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210793 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c203eba8d9ff..5478a89d94e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2516,6 +2516,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0ce, "Gigabyte X570 Aorus Xtreme", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x11f7, "MSI-GE63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1229, "MSI-GP73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1275, "MSI-GL63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), -- 2.26.2

1 0

[RFC][PATCH v2 0/1] design a way to change audio Jack state by software
by Hui Wang 19 Dec '20

19 Dec '20

the changes in the V2: - using debugfs instead of sysfs - using jack_ctrl to create a folder instead of snd_jack, since ASoC drivers could create multi jack_ctrls within a snd_jack - create a folder for each jack_ctrl instead for all jack_ctrls This is the layout of folder on a machine with 2 sound cards: root@hwang4-ThinkPad-P1-Gen-3:/sys/kernel/debug# tree /sys/kernel/debug/sound-core/ /sys/kernel/debug/sound-core/ ├── card0 │ ├── HDMI!DP,pcm=10 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=11 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=3 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=7 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ ├── HDMI!DP,pcm=8 Jack │ │ ├── jackin_inject │ │ └── sw_inject_enable │ └── HDMI!DP,pcm=9 Jack │ ├── jackin_inject │ └── sw_inject_enable └── card1 ├── HDMI!DP,pcm=3 Jack │ ├── jackin_inject │ └── sw_inject_enable ├── HDMI!DP,pcm=4 Jack │ ├── jackin_inject │ └── sw_inject_enable ├── HDMI!DP,pcm=5 Jack │ ├── jackin_inject │ └── sw_inject_enable ├── Headphone Jack │ ├── jackin_inject │ └── sw_inject_enable ├── Headset Jack │ ├── jackin_inject │ └── sw_inject_enable └── Mic Jack ├── jackin_inject └── sw_inject_enable The sw_inject_enable has rw mode, the jackin_inject has write_only mode. If users want to inject a plugin to Headphone Jack, they could run: echo 1 > 'Headphone Jack'/sw_inject_enable /* now, this jack state can't be changed by hw events */ cat 'Headphone Jack'/sw_inject_enable ack: Headphone Jack Inject Enabled: 1 echo 1 > 'Headphone Jack'/jackin_inject /* this will inject a plugin to Headphone Jack */ After testing, run echo 0 > 'Headphone Jack'/sw_inject_enable /* the hw events will change the jack state */ Hui Wang (1): alsa: jack: implement software jack injection via debugfs include/sound/core.h | 2 + sound/core/init.c | 7 ++ sound/core/jack.c | 202 ++++++++++++++++++++++++++++++++++++------- sound/core/sound.c | 8 ++ 4 files changed, 188 insertions(+), 31 deletions(-) -- 2.25.1

3 5

[PATCH] Fix mic sound on Jieli webcam
by Marco Giunta 19 Dec '20

19 Dec '20

Hi, recently I've bought a usb webcam with integrated mic: Jieli Technology USB PHY 2.0 (1224:2a25) The video part works well, but the mic sound is speedups, "like minions" (cit.). When I connect the camera, these are the dmesg messages: kernel: usb 1-8: current rate 0 is different from the runtime rate 8000 kernel: usb 1-8: current rate 0 is different from the runtime rate 16000 kernel: usb 1-8: current rate 0 is different from the runtime rate 44100 kernel: usb 1-8: current rate 0 is different from the runtime rate 48000 kernel: usb 1-8: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. kernel: usb 1-8: [3] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 kernel: usbcore: registered new interface driver snd-usb-audio and after a while, dmesg log is filled, every 5 seconds, with: kernel: retire_capture_urb: 84 callbacks suppressed kernel: retire_capture_urb: 1714 callbacks suppressed A guy reports on ArchLinux bug website the same problem (https://bugs.archlinux.org/task/68141?opened=12995&status%5B0%5D=) and provides a patch to fix the sound issue. I've applied the patch on kernel 5.9.13 (Fedora 33 x86_64) and now the mic works, no more minions voice effect. Now dmesg messages are only: kernel: usb 1-8: Warning! Unlikely big volume range (=4096), cval->res is probably wrong. kernel: usb 1-8: [3] FU [Mic Capture Volume] ch = 1, val = 0/4096/1 kernel: usbcore: registered new interface driver snd-usb-audio the retire_capture_urb messages are gone. All credits for the patch go to him but I don't know how to contact that guy nor I don't know if he has already contacted you, so my question is if you could review his patch and finally apply upstream. If you need other information or you need a tester, I'm here. Thanks, Marco --- a/sound/usb/format.c 2020-10-01 18:36:35.000000000 +0300 +++ b/sound/usb/format.c 2020-10-04 02:10:21.678685952 +0300 @@ -217,6 +217,21 @@ (chip->usb_id == USB_ID(0x041e, 0x4064) || chip->usb_id == USB_ID(0x041e, 0x4068))) rate = 8000; + + // hack for "Jieli Technology USB PHY 2.0" webcam + if (chip->usb_id == USB_ID(0x1224, 0x2a25)) { + switch (rate) { + case 8000: + fp->datainterval += 4; + break; + case 16000: + fp->datainterval += 3; + break; + default: + fp->datainterval += 1; + break; + } + } fp->rate_table[fp->nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) --- a/sound/usb/endpoint.c 2020-10-01 18:36:35.000000000 +0300 +++ b/sound/usb/endpoint.c 2020-10-04 02:09:09.471978982 +0300 @@ -882,6 +882,8 @@ if (snd_usb_get_speed(ep->chip->dev) != USB_SPEED_FULL) { packs_per_ms = 8 >> ep->datainterval; max_packs_per_urb = MAX_PACKS_HS; + if (!packs_per_ms) + packs_per_ms = 1; } else { packs_per_ms = 1; max_packs_per_urb = MAX_PACKS; --- a/sound/usb/quirks.c 2020-10-01 18:36:35.000000000 +0300 +++ b/sound/usb/quirks.c 2020-10-04 02:14:04.532196519 +0300 @@ -1516,6 +1516,7 @@ case USB_ID(0x1901, 0x0191): /* GE B850V3 CP2114 audio interface */ case USB_ID(0x21b4, 0x0081): /* AudioQuest DragonFly */ case USB_ID(0x2912, 0x30c8): /* Audioengine D1 */ + case USB_ID(0x1224, 0x2a25): /* Jieli Technology USB PHY 2.0 */ return true; }

2 8

[PATCH] ALSA: pcm: oss: Fix a few more UBSAN fixes
by Takashi Iwai 18 Dec '20

18 Dec '20

There are a few places that call round{up|down}_pow_of_two() with the value zero, and this causes undefined behavior warnings. Avoid calling those macros if such a nonsense value is passed; it's a minor optimization as well, as we handle it as either an error or a value to be skipped, instead. Reported-by: syzbot+33ef0b6639a8d2d42b4c(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index de1917484647..142fc751a847 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -693,6 +693,8 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, oss_buffer_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, NULL)) * oss_frame_size; + if (!oss_buffer_size) + return -EINVAL; oss_buffer_size = rounddown_pow_of_two(oss_buffer_size); if (atomic_read(&substream->mmap_count)) { if (oss_buffer_size > runtime->oss.mmap_bytes) @@ -728,17 +730,21 @@ static int snd_pcm_oss_period_size(struct snd_pcm_substream *substream, min_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_min(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - min_period_size *= oss_frame_size; - min_period_size = roundup_pow_of_two(min_period_size); - if (oss_period_size < min_period_size) - oss_period_size = min_period_size; + if (min_period_size) { + min_period_size *= oss_frame_size; + min_period_size = roundup_pow_of_two(min_period_size); + if (oss_period_size < min_period_size) + oss_period_size = min_period_size; + } max_period_size = snd_pcm_plug_client_size(substream, snd_pcm_hw_param_value_max(slave_params, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, NULL)); - max_period_size *= oss_frame_size; - max_period_size = rounddown_pow_of_two(max_period_size); - if (oss_period_size > max_period_size) - oss_period_size = max_period_size; + if (max_period_size) { + max_period_size *= oss_frame_size; + max_period_size = rounddown_pow_of_two(max_period_size); + if (oss_period_size > max_period_size) + oss_period_size = max_period_size; + } oss_periods = oss_buffer_size / oss_period_size; -- 2.26.2

1 0

[PATCH] ALSA: pcm: Remove snd_pcm_lib_preallocate_dma_free()
by Lars-Peter Clausen 18 Dec '20

18 Dec '20

Since commit d4cfb30fce03 ("ALSA: pcm: Set per-card upper limit of PCM buffer allocations") snd_pcm_lib_preallocate_dma_free() is a single line function that has one caller, which is another single line function. Clean this up a bit and remove snd_pcm_lib_preallocate_dma_free() and directly call do_free_pages() from snd_pcm_lib_preallocate_free(). This is a bit less boilerplate. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/core/pcm_memory.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 4f03ba8ed0ae..ee6e9c5eec45 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -89,14 +89,6 @@ static int preallocate_pcm_pages(struct snd_pcm_substream *substream, size_t siz return 0; } -/* - * release the preallocated buffer if not yet done. - */ -static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream) -{ - do_free_pages(substream->pcm->card, &substream->dma_buffer); -} - /** * snd_pcm_lib_preallocate_free - release the preallocated buffer of the specified substream. * @substream: the pcm substream instance @@ -105,7 +97,7 @@ static void snd_pcm_lib_preallocate_dma_free(struct snd_pcm_substream *substream */ void snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream) { - snd_pcm_lib_preallocate_dma_free(substream); + do_free_pages(substream->pcm->card, &substream->dma_buffer); } /** -- 2.20.1

2 1

[PATCH 0/2] ALSA: Clear PCM buffers more properly
by Takashi Iwai 18 Dec '20

18 Dec '20

Hi, here are two small fix patches for PCM buffer allocations. Basically those address two things that have been discussed in the recent thread [1]: * Clear the remaining bytes in PCM buffer pages that are exposed via mmap * Assure the page aligned allocations for genalloc This doesn't cover the cases where non-standard PCM buffer allocation is used. Those should handle the buffer clearance by themselves. Takashi [1] https://lore.kernel.org/r/05c824e5-0c33-4182-26fa-b116a42b10d6@metafoo.de === Takashi Iwai (2): ALSA: memalloc: Align buffer allocations in page size ALSA: pcm: Clear the full allocated memory at hw_params sound/core/memalloc.c | 1 + sound/core/pcm_native.c | 9 +++++++-- 2 files changed, 8 insertions(+), 2 deletions(-) -- 2.26.2

2 4

[PATCH] ALSA: usb-audio: Disable sample read check if firmware doesn't give back
by Takashi Iwai 18 Dec '20

18 Dec '20

Some buggy firmware don't give the current sample rate but leaves zero. Handle this case more gracefully without warning but just skip the current rate verification from the next time. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index e940dcee792b..31051f2be46d 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -534,6 +534,12 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, } crate = data[0] | (data[1] << 8) | (data[2] << 16); + if (!crate) { + dev_info(&dev->dev, "failed to read current rate; disabling the check\n"); + chip->sample_rate_read_error = 3; /* three strikes, see above */ + return 0; + } + if (crate != rate) { dev_warn(&dev->dev, "current rate %d is different from the runtime rate %d\n", crate, rate); // runtime->rate = crate; -- 2.26.2

1 0

Re: [PATCH 06/18] arm64: dts: renesas: beacon: Configure Audio CODEC clocks
by Geert Uytterhoeven 18 Dec '20

18 Dec '20

Hi Adam, CC alsa-devel On Sun, Dec 13, 2020 at 7:38 PM Adam Ford <aford173(a)gmail.com> wrote: > With the newly added configurable clock options, the audio CODEC can > configure the mclk automatically. Add the reference to the versaclock. > Since the devices on I2C5 can communicate at 400KHz, let's also increase > that too > > Signed-off-by: Adam Ford <aford173(a)gmail.com> Thanks for your patch! > --- a/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > +++ b/arch/arm64/boot/dts/renesas/beacon-renesom-baseboard.dtsi > @@ -424,13 +424,15 @@ &i2c0 { > > &i2c5 { > status = "okay"; > - clock-frequency = <100000>; > + clock-frequency = <400000>; > pinctrl-0 = <&i2c5_pins>; > pinctrl-names = "default"; > > codec: wm8962@1a { > compatible = "wlf,wm8962"; > reg = <0x1a>; > + clocks = <&versaclock6_bb 3>; > + clock-names = "mclk"; While the driver does get the (nameless) clock, the DT bindings lack any mention of a clocks property. It would be good to update the bindings. Note that arch/arm/boot/dts/imx6-logicpd-baseboard.dtsi and arch/arm64/boot/dts/freescale/imx8mm-beacon-baseboard.dtsi (both by your hand) use "xclk" instead of "mclk"? > DCVDD-supply = <&reg_audio>; > DBVDD-supply = <&reg_audio>; > AVDD-supply = <&reg_audio>; Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

2 4

Move SDM845 out of Qualcomm folder and add C360
by GitHub pull_request - opened 18 Dec '20

18 Dec '20

alsa-project/alsa-ucm-conf pull request #70 was opened from Srinivas-Kandagatla: With driver_name fix in sdm845 machine driver https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/commit/?… /proc/asound/cards get populated correctly with proper driver name and card long name. With this in place we should be able to rearrange sdm845 directory to be inline with others and reuse some of the common code. Without this all the sdm845 based machines will get a generic ucm description which will not work, one of such instance is Lenovo C360 Laptop. This patchset also add support to Lenovo C360 support Thanks, srini Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/70.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: meson: axg-tdmin: fix axg skew offset
by Jerome Brunet 18 Dec '20

18 Dec '20

The signal captured on from tdm decoder of the AXG SoC is incorrect. It appears amplified. The skew offset of the decoder is wrong. Setting the skew offset to 3, like the g12 and sm1 SoCs, solves and gives correct data. Fixes: 13a22e6a98f8 ("ASoC: meson: add tdm input driver") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdmin.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/meson/axg-tdmin.c b/sound/soc/meson/axg-tdmin.c index 88ed95ae886b..b4faf9d5c1aa 100644 --- a/sound/soc/meson/axg-tdmin.c +++ b/sound/soc/meson/axg-tdmin.c @@ -224,15 +224,6 @@ static const struct axg_tdm_formatter_ops axg_tdmin_ops = { }; static const struct axg_tdm_formatter_driver axg_tdmin_drv = { - .component_drv = &axg_tdmin_component_drv, - .regmap_cfg = &axg_tdmin_regmap_cfg, - .ops = &axg_tdmin_ops, - .quirks = &(const struct axg_tdm_formatter_hw) { - .skew_offset = 2, - }, -}; - -static const struct axg_tdm_formatter_driver g12a_tdmin_drv = { .component_drv = &axg_tdmin_component_drv, .regmap_cfg = &axg_tdmin_regmap_cfg, .ops = &axg_tdmin_ops, @@ -247,10 +238,10 @@ static const struct of_device_id axg_tdmin_of_match[] = { .data = &axg_tdmin_drv, }, { .compatible = "amlogic,g12a-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, { .compatible = "amlogic,sm1-tdmin", - .data = &g12a_tdmin_drv, + .data = &axg_tdmin_drv, }, {} }; MODULE_DEVICE_TABLE(of, axg_tdmin_of_match); -- 2.29.2

2 1

[PATCH] ASoC: meson: axg-tdm-interface: fix loopback
by Jerome Brunet 18 Dec '20

18 Dec '20

When the axg-tdm-interface was introduced, the backend DAI was marked as an endpoint when DPCM was walking the DAPM graph to find a its BE. It is no longer the case since this commit 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Because of this, when DPCM finds a BE it does everything it needs on the DAIs but it won't power up the widgets between the FE and the BE if there is no actual endpoint after the BE. On meson-axg HWs, the loopback is a special DAI of the tdm-interface BE. It is only linked to the dummy codec since there no actual HW after it. >From the DAPM perspective, the DAI has no endpoint. Because of this, the TDM decoder, which is a widget between the FE and BE is not powered up. >From the user perspective, everything seems fine but no data is produced. Connecting the Loopback DAI to a dummy DAPM endpoint solves the problem. Fixes: 8dd26dff00c0 ("ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks") Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index c8664ab80d45..87cac440b369 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -467,8 +467,20 @@ static int axg_tdm_iface_set_bias_level(struct snd_soc_component *component, return ret; } +static const struct snd_soc_dapm_widget axg_tdm_iface_dapm_widgets[] = { + SND_SOC_DAPM_SIGGEN("Playback Signal"), +}; + +static const struct snd_soc_dapm_route axg_tdm_iface_dapm_routes[] = { + { "Loopback", NULL, "Playback Signal" }, +}; + static const struct snd_soc_component_driver axg_tdm_iface_component_drv = { - .set_bias_level = axg_tdm_iface_set_bias_level, + .dapm_widgets = axg_tdm_iface_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(axg_tdm_iface_dapm_widgets), + .dapm_routes = axg_tdm_iface_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(axg_tdm_iface_dapm_routes), + .set_bias_level = axg_tdm_iface_set_bias_level, }; static const struct of_device_id axg_tdm_iface_of_match[] = { -- 2.29.2

3 2

[PATCH] ASoC: Intel: haswell: Add missing pm_ops
by Cezary Rojewski 18 Dec '20

18 Dec '20

haswell machine board is missing pm_ops what prevents it from undergoing suspend-resume procedure successfully. Assign default snd_soc_pm_ops so this is no longer the case. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/boards/haswell.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index c55d1239e705..c763bfeb1f38 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -189,6 +189,7 @@ static struct platform_driver haswell_audio = { .probe = haswell_audio_probe, .driver = { .name = "haswell-audio", + .pm = &snd_soc_pm_ops, }, }; -- 2.17.1

2 1

[PATCH] ASoC: rt711: mutex between calibration and power state changes
by shumingf＠realtek.com 18 Dec '20

18 Dec '20

From: Shuming Fan <shumingf(a)realtek.com> To avoid calibration time-out, this patch adds the mutex between calibration and power state changes Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 5771c02c3459..85f744184a60 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -462,6 +462,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, unsigned int read_ll, read_rl; int i; + mutex_lock(&rt711->calibrate_mutex); + /* Can't use update bit function, so read the original value first */ addr_h = mc->reg; addr_l = mc->rreg; @@ -547,6 +549,8 @@ static int rt711_set_amp_gain_put(struct snd_kcontrol *kcontrol, if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + + mutex_unlock(&rt711->calibrate_mutex); return 0; } @@ -859,9 +863,11 @@ static int rt711_set_bias_level(struct snd_soc_component *component, break; case SND_SOC_BIAS_STANDBY: + mutex_lock(&rt711->calibrate_mutex); regmap_write(rt711->regmap, RT711_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + mutex_unlock(&rt711->calibrate_mutex); break; default: -- 2.29.0

2 1

[PATCH v6 0/2] Platform driver update to support playback recover after resume
by Srinivasa Rao Mandadapu 18 Dec '20

18 Dec '20

This patch set is to add support for playback recover after hard suspend and resume. It includes: 1. Reverting part of previous commit, which is for handling registers invalid state after hard suspend. 2. Adding pm ops in component driver and do regcache sync. Changes Since v1 and v2: -- Subject lines changed Changes Since v3: -- Patch is splitted into 2 patches Changes Since v4: -- Subject lines changed Changes Since v5: -- Removed redundant initialization of map variable in lpass-platform.c Srinivasa Rao Mandadapu (2): ASoC: qcom: Fix incorrect volatile registers ASoC: qcom: Add support for playback recover after resume sound/soc/qcom/lpass-cpu.c | 20 ++--------------- sound/soc/qcom/lpass-platform.c | 50 ++++++++++++++++++++++++++++------------- 2 files changed, 37 insertions(+), 33 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH v3] ASoC: max98373: don't access volatile registers in bias level off
by Bard Liao 18 Dec '20

18 Dec '20

We will set regcache_cache_only true in suspend. As a result, regmap_read will return error when we try to read volatile registers in suspend. Besides, it doesn't make sense to read feedback data when codec is not active. To make userspace happy, this patch returns a cached value shich should be a valid value. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v3: - Allocate cache dynamically. v2: - Return cached values instead of 0. --- sound/soc/codecs/max98373-i2c.c | 20 +++++++++++++++++++ sound/soc/codecs/max98373-sdw.c | 20 +++++++++++++++++++ sound/soc/codecs/max98373.c | 34 ++++++++++++++++++++++++++++++--- sound/soc/codecs/max98373.h | 8 ++++++++ 4 files changed, 79 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 92921e34f948..85f6865019d4 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -19,6 +19,12 @@ #include <sound/tlv.h> #include "max98373.h" +static const u32 max98373_i2c_cache_reg[] = { + MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, + MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, + MAX98373_R20B6_BDE_CUR_STATE_READBACK, +}; + static struct reg_default max98373_reg[] = { {MAX98373_R2000_SW_RESET, 0x00}, {MAX98373_R2001_INT_RAW1, 0x00}, @@ -472,6 +478,11 @@ static struct snd_soc_dai_driver max98373_dai[] = { static int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); regcache_mark_dirty(max98373->regmap); @@ -509,6 +520,7 @@ static int max98373_i2c_probe(struct i2c_client *i2c, { int ret = 0; int reg = 0; + int i; struct max98373_priv *max98373 = NULL; max98373 = devm_kzalloc(&i2c->dev, sizeof(*max98373), GFP_KERNEL); @@ -534,6 +546,14 @@ static int max98373_i2c_probe(struct i2c_client *i2c, return ret; } + max98373->cache_num = ARRAY_SIZE(max98373_i2c_cache_reg); + max98373->cache = devm_kcalloc(&i2c->dev, max98373->cache_num, + sizeof(*max98373->cache), + GFP_KERNEL); + + for (i = 0; i < max98373->cache_num; i++) + max98373->cache[i].reg = max98373_i2c_cache_reg[i]; + /* voltage/current slot & gpio configuration */ max98373_slot_config(&i2c->dev, max98373); diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index ec2e79c57357..b8d471d79e93 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -23,6 +23,12 @@ struct sdw_stream_data { struct sdw_stream_runtime *sdw_stream; }; +static const u32 max98373_sdw_cache_reg[] = { + MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, + MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, + MAX98373_R20B6_BDE_CUR_STATE_READBACK, +}; + static struct reg_default max98373_reg[] = { {MAX98373_R0040_SCP_INIT_STAT_1, 0x00}, {MAX98373_R0041_SCP_INIT_MASK_1, 0x00}, @@ -245,6 +251,11 @@ static const struct regmap_config max98373_sdw_regmap = { static __maybe_unused int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); @@ -757,6 +768,7 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) { struct max98373_priv *max98373; int ret; + int i; struct device *dev = &slave->dev; /* Allocate and assign private driver data structure */ @@ -768,6 +780,14 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) max98373->regmap = regmap; max98373->slave = slave; + max98373->cache_num = ARRAY_SIZE(max98373_sdw_cache_reg); + max98373->cache = devm_kcalloc(dev, max98373->cache_num, + sizeof(*max98373->cache), + GFP_KERNEL); + + for (i = 0; i < max98373->cache_num; i++) + max98373->cache[i].reg = max98373_sdw_cache_reg[i]; + /* Read voltage and slot configuration */ max98373_slot_config(dev, max98373); diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 929bb1798c43..31d571d4fac1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -168,6 +168,31 @@ static SOC_ENUM_SINGLE_DECL(max98373_adc_samplerate_enum, MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0, max98373_ADC_samplerate_text); +static int max98373_feedback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); + int i; + + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + /* + * Register values will be cached before suspend. The cached value + * will be a valid value and userspace will happy with that. + */ + for (i = 0; i < max98373->cache_num; i++) { + if (mc->reg == max98373->cache[i].reg) { + ucontrol->value.integer.value[0] = max98373->cache[i].val; + return 0; + } + } + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new max98373_snd_controls[] = { SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_VOL_SEL_SHIFT, 1, 0), @@ -209,8 +234,10 @@ SOC_SINGLE("ADC PVDD FLT Switch", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), -SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0), -SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0), +SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), +SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, 0, 0x3, 0), SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, @@ -226,7 +253,8 @@ SOC_SINGLE("BDE LVL1 Thresh", MAX98373_R2097_BDE_L1_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0), -SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0), +SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0, + max98373_feedback_get, NULL), SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0), SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0), SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h index 4ab29b9d51c7..71f5a5228f34 100644 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -203,6 +203,11 @@ /* MAX98373_R2000_SW_RESET */ #define MAX98373_SOFT_RESET (0x1 << 0) +struct max98373_cache { + u32 reg; + u32 val; +}; + struct max98373_priv { struct regmap *regmap; int reset_gpio; @@ -212,6 +217,9 @@ struct max98373_priv { bool interleave_mode; unsigned int ch_size; bool tdm_mode; + /* cache for reading a valid fake feedback value */ + struct max98373_cache *cache; + int cache_num; /* variables to support soundwire */ struct sdw_slave *slave; bool hw_init; -- 2.17.1

2 1

[PATCH v2] ASoC: rsnd: don't call clk_disable_unprepare() if can't use
by Kuninori Morimoto 18 Dec '20

18 Dec '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We need to care clock accessibility, because we might can't use clock for some reasons. It sets clk_rate for each clocks when enabled. This means it doesn't have clk_rate if we can't use. We can avoid to call clk_disable_unprepare() in such case. Link: https://lore.kernel.org/r/CAMuHMdWvB+p=2JqTsO7bR8uJqKqO5A2XgXFXsVAjHk3hcxgc… Reported-by: Geert Uytterhoeven <geert+renesas(a)glider.be> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - reset adg->clk_rate[i] = 0 sound/soc/sh/rcar/adg.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index b9aacf3d3b29..abdfd9cf91e2 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -366,25 +366,27 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable) struct rsnd_adg *adg = rsnd_priv_to_adg(priv); struct device *dev = rsnd_priv_to_dev(priv); struct clk *clk; - int i, ret; + int i; for_each_rsnd_clk(clk, adg, i) { - ret = 0; if (enable) { - ret = clk_prepare_enable(clk); + int ret = clk_prepare_enable(clk); /* * We shouldn't use clk_get_rate() under * atomic context. Let's keep it when * rsnd_adg_clk_enable() was called */ - adg->clk_rate[i] = clk_get_rate(adg->clk[i]); + adg->clk_rate[i] = 0; + if (ret < 0) + dev_warn(dev, "can't use clk %d\n", i); + else + adg->clk_rate[i] = clk_get_rate(clk); } else { - clk_disable_unprepare(clk); + if (adg->clk_rate[i]) + clk_disable_unprepare(clk); + adg->clk_rate[i] = 0; } - - if (ret < 0) - dev_warn(dev, "can't use clk %d\n", i); } } -- 2.25.1

3 2

[PATCH v2] ASoC: max98373: don't access volatile registers in bias level off
by Bard Liao 18 Dec '20

18 Dec '20

We will set regcache_cache_only true in suspend. As a result, regmap_read will return error when we try to read volatile registers in suspend. Besides, it doesn't make sense to read feedback data when codec is not active. To make userspace happy, this patch returns a cached value shich should be a valid value. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- v2: - Return cached values instead of 0. --- sound/soc/codecs/max98373-i2c.c | 8 ++++++++ sound/soc/codecs/max98373-sdw.c | 8 ++++++++ sound/soc/codecs/max98373.c | 34 ++++++++++++++++++++++++++++++--- sound/soc/codecs/max98373.h | 14 ++++++++++++++ 4 files changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max98373-i2c.c b/sound/soc/codecs/max98373-i2c.c index 92921e34f948..09890478b120 100644 --- a/sound/soc/codecs/max98373-i2c.c +++ b/sound/soc/codecs/max98373-i2c.c @@ -472,6 +472,11 @@ static struct snd_soc_dai_driver max98373_dai[] = { static int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); regcache_mark_dirty(max98373->regmap); @@ -534,6 +539,9 @@ static int max98373_i2c_probe(struct i2c_client *i2c, return ret; } + max98373->cache = max98373_cache; + max98373->cache_num = ARRAY_SIZE(max98373_cache); + /* voltage/current slot & gpio configuration */ max98373_slot_config(&i2c->dev, max98373); diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index ec2e79c57357..0575d323df8a 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -245,6 +245,11 @@ static const struct regmap_config max98373_sdw_regmap = { static __maybe_unused int max98373_suspend(struct device *dev) { struct max98373_priv *max98373 = dev_get_drvdata(dev); + int i; + + /* cache feedback register values before suspend */ + for (i = 0; i < max98373->cache_num; i++) + regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); @@ -768,6 +773,9 @@ static int max98373_init(struct sdw_slave *slave, struct regmap *regmap) max98373->regmap = regmap; max98373->slave = slave; + max98373->cache = max98373_cache; + max98373->cache_num = ARRAY_SIZE(max98373_cache); + /* Read voltage and slot configuration */ max98373_slot_config(dev, max98373); diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index 929bb1798c43..31d571d4fac1 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -168,6 +168,31 @@ static SOC_ENUM_SINGLE_DECL(max98373_adc_samplerate_enum, MAX98373_R2051_MEAS_ADC_SAMPLING_RATE, 0, max98373_ADC_samplerate_text); +static int max98373_feedback_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct max98373_priv *max98373 = snd_soc_component_get_drvdata(component); + int i; + + if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) { + /* + * Register values will be cached before suspend. The cached value + * will be a valid value and userspace will happy with that. + */ + for (i = 0; i < max98373->cache_num; i++) { + if (mc->reg == max98373->cache[i].reg) { + ucontrol->value.integer.value[0] = max98373->cache[i].val; + return 0; + } + } + } + + return snd_soc_put_volsw(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new max98373_snd_controls[] = { SOC_SINGLE("Digital Vol Sel Switch", MAX98373_R203F_AMP_DSP_CFG, MAX98373_AMP_VOL_SEL_SHIFT, 1, 0), @@ -209,8 +234,10 @@ SOC_SINGLE("ADC PVDD FLT Switch", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), SOC_SINGLE("ADC TEMP FLT Switch", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, MAX98373_FLT_EN_SHIFT, 1, 0), -SOC_SINGLE("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0), -SOC_SINGLE("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0), +SOC_SINGLE_EXT("ADC PVDD", MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), +SOC_SINGLE_EXT("ADC TEMP", MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0, 0xFF, 0, + max98373_feedback_get, NULL), SOC_SINGLE("ADC PVDD FLT Coeff", MAX98373_R2052_MEAS_ADC_PVDD_FLT_CFG, 0, 0x3, 0), SOC_SINGLE("ADC TEMP FLT Coeff", MAX98373_R2053_MEAS_ADC_THERM_FLT_CFG, @@ -226,7 +253,8 @@ SOC_SINGLE("BDE LVL1 Thresh", MAX98373_R2097_BDE_L1_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL2 Thresh", MAX98373_R2098_BDE_L2_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL3 Thresh", MAX98373_R2099_BDE_L3_THRESH, 0, 0xFF, 0), SOC_SINGLE("BDE LVL4 Thresh", MAX98373_R209A_BDE_L4_THRESH, 0, 0xFF, 0), -SOC_SINGLE("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0), +SOC_SINGLE_EXT("BDE Active Level", MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0, 8, 0, + max98373_feedback_get, NULL), SOC_SINGLE("BDE Clip Mode Switch", MAX98373_R2092_BDE_CLIPPER_MODE, 0, 1, 0), SOC_SINGLE("BDE Thresh Hysteresis", MAX98373_R209B_BDE_THRESH_HYST, 0, 0xFF, 0), SOC_SINGLE("BDE Hold Time", MAX98373_R2090_BDE_LVL_HOLD, 0, 0xFF, 0), diff --git a/sound/soc/codecs/max98373.h b/sound/soc/codecs/max98373.h index 4ab29b9d51c7..6206c6c3a773 100644 --- a/sound/soc/codecs/max98373.h +++ b/sound/soc/codecs/max98373.h @@ -203,6 +203,17 @@ /* MAX98373_R2000_SW_RESET */ #define MAX98373_SOFT_RESET (0x1 << 0) +struct max98373_cache { + u32 reg; + u32 val; +}; + +struct max98373_cache max98373_cache[] = { + {MAX98373_R2054_MEAS_ADC_PVDD_CH_READBACK, 0}, + {MAX98373_R2055_MEAS_ADC_THERM_CH_READBACK, 0}, + {MAX98373_R20B6_BDE_CUR_STATE_READBACK, 0}, +}; + struct max98373_priv { struct regmap *regmap; int reset_gpio; @@ -212,6 +223,9 @@ struct max98373_priv { bool interleave_mode; unsigned int ch_size; bool tdm_mode; + /* cache for reading a valid fake feedback value */ + struct max98373_cache *cache; + int cache_num; /* variables to support soundwire */ struct sdw_slave *slave; bool hw_init; -- 2.17.1

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[PATCH v5 0/2] Platform driver update to support playback recover after resume
by Srinivasa Rao Mandadapu 18 Dec '20

18 Dec '20

This patch set is to add support for playback recover after hard suspend and resume. It includes: 1. Reverting part of previous commit, which is for handling registers invalid state after hard suspend. 2. Adding pm ops in component driver and do regcache sync. Changes Since v1 and v2: -- Subject lines changed Changes Since v3: -- Patch is splitted into 2 patches Changes Since v4: -- Subject lines changed Srinivasa Rao Mandadapu (2): ASoC: qcom: Fix incorrect volatile registers ASoC: qcom: Add support for playback recover after resume sound/soc/qcom/lpass-cpu.c | 20 ++---------------- sound/soc/qcom/lpass-platform.c | 46 +++++++++++++++++++++++++++++++---------- 2 files changed, 37 insertions(+), 29 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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UCM fails as CardDriver may not be defined
by GitHub issues - opened 18 Dec '20

18 Dec '20

alsa-project/alsa-ucm-conf issue #69 was opened from coolstar: UCM doesn't load as CardDriver isn't set on some platforms Changing https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/ucm.conf#L69 to `Directory "${CardLongName}"` resolves the issue ``` I: [pulseaudio] source.c: Created source 1 "alsa_input.platform-sound.analog-stereo" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: [pulseaudio] source.c: alsa.resolution_bits = "16" I: [pulseaudio] source.c: device.api = "alsa" I: [pulseaudio] source.c: device.class = "sound" I: [pulseaudio] source.c: alsa.class = "generic" I: [pulseaudio] source.c: alsa.subclass = "generic-mix" I: [pulseaudio] source.c: alsa.name = "" I: [pulseaudio] source.c: alsa.id = "ADMAIF1 CIF ADMAIF1-0" I: [pulseaudio] source.c: alsa.subdevice = "0" I: [pulseaudio] source.c: alsa.subdevice_name = "subdevice #0" I: [pulseaudio] source.c: alsa.device = "0" I: [pulseaudio] source.c: alsa.card = "1" I: [pulseaudio] source.c: alsa.card_name = "tegra-snd-t210ref-mobile-rt565x" I: [pulseaudio] source.c: alsa.long_card_name = "tegra-snd-t210ref-mobile-rt565x" I: [pulseaudio] source.c: device.bus_path = "platform-sound" I: [pulseaudio] source.c: sysfs.path = "/devices/sound/sound/card1" I: [pulseaudio] source.c: device.form_factor = "internal" I: [pulseaudio] source.c: device.string = "front:1" I: [pulseaudio] source.c: device.buffering.buffer_size = "32768" I: [pulseaudio] source.c: device.buffering.fragment_size = "16384" I: [pulseaudio] source.c: device.access_mode = "mmap+timer" I: [pulseaudio] source.c: device.profile.name = "analog-stereo" I: [pulseaudio] source.c: device.profile.description = "Analog Stereo" I: [pulseaudio] source.c: device.description = "Built-in Audio Analog Stereo" I: [pulseaudio] source.c: module-udev-detect.discovered = "1" I: [pulseaudio] source.c: device.icon_name = "audio-card" ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/69 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v1 ] ALSA: core: memalloc: add page alignment for iram
by Robin Gong 17 Dec '20

17 Dec '20

Since mmap for userspace is based on page alignment, add page alignment for iram alloc from pool, otherwise, some good data located in the same page of dmab->area maybe touched wrongly by userspace like pulseaudio. Signed-off-by: Robin Gong <yibin.gong(a)nxp.com> --- sound/core/memalloc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 0aeeb62..0f33516 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -77,7 +77,8 @@ static void snd_malloc_dev_iram(struct snd_dma_buffer *dmab, size_t size) /* Assign the pool into private_data field */ dmab->private_data = pool; - dmab->area = gen_pool_dma_alloc(pool, size, &dmab->addr); + dmab->area = gen_pool_dma_alloc_align(pool, size, &dmab->addr, + PAGE_SIZE); } /** -- 2.7.4

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