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

[PATCH AUTOSEL 6.0 28/44] ASoC: SOF: add quirk to override topology mclk_id
by Sasha Levin 10 Oct '22

10 Oct '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit d136949dd8e2e309dc2f186507486b71cbe9acdb ] Some Intel-based platforms rely on a topology file that hard-codes the use of MCLK0. This is incorrect in 10% of the cases. Rather than generating yet another set of topology files, this patch adds a kernel module parameter to override the topology value. In hindsight, we should never have allowed mclks to be specified in topology, this is a hardware-level information that should not have been visible in the topology. Future patches will try to set this value automagically, e.g. by parsing the NHLT content. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20220919115350.43104-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda.c | 11 +++++++++++ sound/soc/sof/ipc3-topology.c | 7 +++++++ sound/soc/sof/sof-priv.h | 4 ++++ 3 files changed, 22 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 6d4ecbe14adf..ada2e6775749 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -376,6 +376,10 @@ static int dmic_num_override = -1; module_param_named(dmic_num, dmic_num_override, int, 0444); MODULE_PARM_DESC(dmic_num, "SOF HDA DMIC number"); +static int mclk_id_override = -1; +module_param_named(mclk_id, mclk_id_override, int, 0444); +MODULE_PARM_DESC(mclk_id, "SOF SSP mclk_id"); + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) static bool hda_codec_use_common_hdmi = IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI); module_param_named(use_common_hdmi, hda_codec_use_common_hdmi, bool, 0444); @@ -1565,6 +1569,13 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) sof_pdata->tplg_filename = tplg_filename; } + + /* check if mclk_id should be modified from topology defaults */ + if (mclk_id_override >= 0) { + dev_info(sdev->dev, "Overriding topology with MCLK %d from kernel_parameter\n", mclk_id_override); + sdev->mclk_id_override = true; + sdev->mclk_id_quirk = mclk_id_override; + } } /* diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 65923e7a5976..a39b43850f0e 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -1249,6 +1249,7 @@ static int sof_link_afe_load(struct snd_soc_component *scomp, struct snd_sof_dai static int sof_link_ssp_load(struct snd_soc_component *scomp, struct snd_sof_dai_link *slink, struct sof_ipc_dai_config *config, struct snd_sof_dai *dai) { + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(scomp); struct snd_soc_tplg_hw_config *hw_config = slink->hw_configs; struct sof_dai_private_data *private = dai->private; u32 size = sizeof(*config); @@ -1273,6 +1274,12 @@ static int sof_link_ssp_load(struct snd_soc_component *scomp, struct snd_sof_dai config[i].hdr.size = size; + if (sdev->mclk_id_override) { + dev_dbg(scomp->dev, "tplg: overriding topology mclk_id %d by quirk %d\n", + config[i].ssp.mclk_id, sdev->mclk_id_quirk); + config[i].ssp.mclk_id = sdev->mclk_id_quirk; + } + /* copy differentiating hw configs to ipc structs */ config[i].ssp.mclk_rate = le32_to_cpu(hw_config[i].mclk_rate); config[i].ssp.bclk_rate = le32_to_cpu(hw_config[i].bclk_rate); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 823583086279..828c74bb75f8 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -594,6 +594,10 @@ struct snd_sof_dev { /* to protect the ipc_rx_handler_list and dsp_state_handler_list list */ struct mutex client_event_handler_mutex; + /* quirks to override topology values */ + bool mclk_id_override; + u16 mclk_id_quirk; /* same size as in IPC3 definitions */ + void *private; /* core does not touch this */ }; -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 27/44] ASoC: sunxi: sun4i-codec: set debugfs_prefix for CPU DAI component
by Sasha Levin 10 Oct '22

10 Oct '22

From: Mikhail Rudenko <mike.rudenko(a)gmail.com> [ Upstream commit 717a8ff20f32792d6a94f2883e771482c37d844b ] At present, succesfull probing of H3 Codec results in an error debugfs: Directory '1c22c00.codec' with parent 'H3 Audio Codec' already present! This is caused by a directory name conflict between codec components. Fix it by setting debugfs_prefix for the CPU DAI component. Signed-off-by: Mikhail Rudenko <mike.rudenko(a)gmail.com> Link: https://lore.kernel.org/r/20220913212256.151799-2-mike.rudenko@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sunxi/sun4i-codec.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 830beb38bf15..fdf3165acd70 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1232,6 +1232,9 @@ static const struct snd_soc_component_driver sun8i_a23_codec_codec = { static const struct snd_soc_component_driver sun4i_codec_component = { .name = "sun4i-codec", .legacy_dai_naming = 1, +#ifdef CONFIG_DEBUG_FS + .debugfs_prefix = "cpu", +#endif }; #define SUN4I_CODEC_RATES SNDRV_PCM_RATE_CONTINUOUS -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 26/44] ASoC: SOF: pci: Change DMI match info to support all Chrome platforms
by Sasha Levin 10 Oct '22

10 Oct '22

From: Jairaj Arava <jairaj.arava(a)intel.com> [ Upstream commit c1c1fc8103f794a10c5c15e3c17879caf4f42c8f ] In some Chrome platforms if OEM's use their own string as SYS_VENDOR than "Google", it leads to firmware load failure from intel/sof/community path. Hence, changing SYS_VENDOR to PRODUCT_FAMILY in which "Google" is used as common prefix and is supported in all Chrome platforms. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Chao Song <chao.song(a)intel.com> Reviewed-by: Curtis Malainey <curtis(a)malainey.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220919114429.42700-1-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/sof-pci-dev.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d627092b399d..643fd1036d60 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -138,7 +138,7 @@ static const struct dmi_system_id community_key_platforms[] = { .ident = "Google Chromebooks", .callback = chromebook_use_community_key, .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google"), } }, {}, -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 25/44] ALSA: intel-dspconfig: add ES8336 support for AlderLake-PS
by Sasha Levin 10 Oct '22

10 Oct '22

From: Muralidhar Reddy <muralidhar.reddy(a)intel.com> [ Upstream commit 9db1c9fa214ef41d098633ff40a87284ca6e1870 ] added quirks for ESS8336 for AlderLake-PS Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Muralidhar Reddy <muralidhar.reddy(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220919114548.42769-1-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/intel-dsp-config.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 5a478649f338..b9eb3208f288 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -427,6 +427,11 @@ static const struct config_entry config_table[] = { .device = 0x51cd, }, /* Alderlake-PS */ + { + .flags = FLAG_SOF, + .device = 0x51c9, + .codec_hid = &essx_83x6, + }, { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x51c9, -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 17/44] ALSA: usb-audio: Register card at the last interface
by Sasha Levin 10 Oct '22

10 Oct '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 6392dcd1d0c7034ccf630ec55fc9e5810ecadf3b ] The USB-audio driver matches per interface, and as default, it registers the card instance at the very first instance. This can be a problem for the devices that have multiple interfaces to be probed, as the udev rule isn't applied properly for the later appearing interfaces. Although we introduced the delayed_register option and the quirks for covering those shortcomings, it's nothing but a workaround for specific devices. This patch is an another attempt to fix the problem in a more generic way. Now the driver checks the whole USB device descriptor at the very first time when an interface is attached to a sound card. It looks at each matching interface in the descriptor and remembers the last matching one. The snd_card_register() is invoked only when this last interface is probed. After this change, the quirks for the delayed registration become superfluous, hence they are removed along with the patch. OTOH, the delayed_register option is still kept, as it might be useful for some corner cases (e.g. a special driver overtakes the interface probe from the standard driver, and the last interface probe may miss). Link: https://lore.kernel.org/r/20220904161247.16461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.c | 32 +++++++++++++++++++++++++------- sound/usb/quirks.c | 42 ------------------------------------------ sound/usb/quirks.h | 2 -- sound/usb/usbaudio.h | 1 + 4 files changed, 26 insertions(+), 51 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 706d249a9ad6..3aea241435fb 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -690,7 +690,7 @@ static bool get_alias_id(struct usb_device *dev, unsigned int *id) return false; } -static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) +static int check_delayed_register_option(struct snd_usb_audio *chip) { int i; unsigned int id, inum; @@ -699,14 +699,31 @@ static bool check_delayed_register_option(struct snd_usb_audio *chip, int iface) if (delayed_register[i] && sscanf(delayed_register[i], "%x:%x", &id, &inum) == 2 && id == chip->usb_id) - return iface < inum; + return inum; } - return false; + return -1; } static const struct usb_device_id usb_audio_ids[]; /* defined below */ +/* look for the last interface that matches with our ids and remember it */ +static void find_last_interface(struct snd_usb_audio *chip) +{ + struct usb_host_config *config = chip->dev->actconfig; + struct usb_interface *intf; + int i; + + if (!config) + return; + for (i = 0; i < config->desc.bNumInterfaces; i++) { + intf = config->interface[i]; + if (usb_match_id(intf, usb_audio_ids)) + chip->last_iface = intf->altsetting[0].desc.bInterfaceNumber; + } + usb_audio_dbg(chip, "Found last interface = %d\n", chip->last_iface); +} + /* look for the corresponding quirk */ static const struct snd_usb_audio_quirk * get_alias_quirk(struct usb_device *dev, unsigned int id) @@ -813,6 +830,7 @@ static int usb_audio_probe(struct usb_interface *intf, err = -ENODEV; goto __error; } + find_last_interface(chip); } if (chip->num_interfaces >= MAX_CARD_INTERFACES) { @@ -862,11 +880,11 @@ static int usb_audio_probe(struct usb_interface *intf, chip->need_delayed_register = false; /* clear again */ } - /* we are allowed to call snd_card_register() many times, but first - * check to see if a device needs to skip it or do anything special + /* register card if we reach to the last interface or to the specified + * one given via option */ - if (!snd_usb_registration_quirk(chip, ifnum) && - !check_delayed_register_option(chip, ifnum)) { + if (check_delayed_register_option(chip) == ifnum || + chip->last_iface == ifnum) { err = snd_card_register(chip->card); if (err < 0) goto __error; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 194c75c45628..eadac586bcc8 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2030,48 +2030,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, } } -/* - * registration quirk: - * the registration is skipped if a device matches with the given ID, - * unless the interface reaches to the defined one. This is for delaying - * the registration until the last known interface, so that the card and - * devices appear at the same time. - */ - -struct registration_quirk { - unsigned int usb_id; /* composed via USB_ID() */ - unsigned int interface; /* the interface to trigger register */ -}; - -#define REG_QUIRK_ENTRY(vendor, product, iface) \ - { .usb_id = USB_ID(vendor, product), .interface = (iface) } - -static const struct registration_quirk registration_quirks[] = { - REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ - REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */ - REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f47, 2), /* JBL Quantum 800 */ - REG_QUIRK_ENTRY(0x0ecb, 0x1f4c, 2), /* JBL Quantum 400 */ - REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ - REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ - REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ - { 0 } /* terminator */ -}; - -/* return true if skipping registration */ -bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface) -{ - const struct registration_quirk *q; - - for (q = registration_quirks; q->usb_id; q++) - if (chip->usb_id == q->usb_id) - return iface < q->interface; - - /* Register as normal */ - return false; -} - /* * driver behavior quirk flags */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index 31abb7cb01a5..f9bfd5ac7bab 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -48,8 +48,6 @@ void snd_usb_audioformat_attributes_quirk(struct snd_usb_audio *chip, struct audioformat *fp, int stream); -bool snd_usb_registration_quirk(struct snd_usb_audio *chip, int iface); - void snd_usb_init_quirk_flags(struct snd_usb_audio *chip); #endif /* __USBAUDIO_QUIRKS_H */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ffbb4b0d09a0..2c6575029b1c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -37,6 +37,7 @@ struct snd_usb_audio { unsigned int quirk_flags; unsigned int need_delayed_register:1; /* warn for delayed registration */ int num_interfaces; + int last_iface; int num_suspended_intf; int sample_rate_read_error; -- 2.35.1

1 0

[PATCH AUTOSEL 6.0 08/44] ALSA: usb-audio: Add quirk to enable Avid Mbox 3 support
by Sasha Levin 10 Oct '22

10 Oct '22

From: Conner Knox <connerknoxpublic(a)gmail.com> [ Upstream commit b01104fc62b6194c852124f6c6df1c0a5c031fc1 ] Add support for Avid Mbox3 USB audio interface at 48kHz Signed-off-by: Conner Knox <connerknoxpublic(a)gmail.com> Link: https://lore.kernel.org/r/20220818201433.16360-1-mbarriolinares@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/quirks-table.h | 76 ++++++++++ sound/usb/quirks.c | 302 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 378 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f93201a830b5..06dfdd45cff8 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2985,6 +2985,82 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +/* DIGIDESIGN MBOX 3 */ +{ + USB_DEVICE(0x0dba, 0x5000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 3", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x00, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x81, + .attributes = 0x00, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .maxpacksize = 0x009c, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, { /* Tascam US122 MKII - playback-only support */ USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5b4d8f5eade2..194c75c45628 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1020,6 +1020,304 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } +static void mbox3_setup_48_24_magic(struct usb_device *dev) +{ + /* The Mbox 3 is "little endian" */ + /* max volume is: 0x0000. */ + /* min volume is: 0x0080 (shown in little endian form) */ + + + /* Load 48000Hz rate into buffer */ + u8 com_buff[4] = {0x80, 0xbb, 0x00, 0x00}; + + /* Set 48000Hz sample rate */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x0001, &com_buff, 4); //Is this really needed? + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &com_buff, 4); + + /* Deactivate Tuner */ + /* on = 0x01*/ + /* off = 0x00*/ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0003, 0x2001, &com_buff, 1); + + /* Set clock source to Internal (as opposed to S/PDIF) */ + com_buff[0] = 0x01; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x8001, &com_buff, 1); + + /* Mute the hardware loopbacks to start the device in a known state. */ + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue input 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0110, 0x4001, &com_buff, 2); + /* Analogue input 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0111, 0x4001, &com_buff, 2); + /* Analogue input 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0114, 0x4001, &com_buff, 2); + /* Analogue input 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0115, 0x4001, &com_buff, 2); + /* Analogue input 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0118, 0x4001, &com_buff, 2); + /* Analogue input 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0119, 0x4001, &com_buff, 2); + /* Analogue input 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011c, 0x4001, &com_buff, 2); + /* Analogue input 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011d, 0x4001, &com_buff, 2); + + /* Set software sends to output */ + com_buff[0] = 0x00; + com_buff[1] = 0x00; + /* Analogue software return 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4001, &com_buff, 2); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue software return 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4001, &com_buff, 2); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue software return 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4001, &com_buff, 2); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + /* Analogue software return 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue software return 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0108, 0x4001, &com_buff, 2); + /* Analogue software return 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0109, 0x4001, &com_buff, 2); + /* Analogue software return 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010c, 0x4001, &com_buff, 2); + /* Analogue software return 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010d, 0x4001, &com_buff, 2); + + /* Return to muting sends */ + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* Analogue fx return left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0120, 0x4001, &com_buff, 2); + /* Analogue fx return right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0121, 0x4001, &com_buff, 2); + + /* Analogue software input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4201, &com_buff, 2); + /* Analogue software input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4201, &com_buff, 2); + /* Analogue software input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4201, &com_buff, 2); + /* Analogue software input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4201, &com_buff, 2); + /* Analogue input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4201, &com_buff, 2); + /* Analogue input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4201, &com_buff, 2); + /* Analogue input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4201, &com_buff, 2); + /* Analogue input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4201, &com_buff, 2); + + /* Toggle allowing host control */ + com_buff[0] = 0x02; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + + /* Do not dim fx returns */ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0002, 0x2001, &com_buff, 1); + + /* Do not set fx returns to mono */ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0001, 0x2001, &com_buff, 1); + + /* Mute the S/PDIF hardware loopback + * same odd volume logic here as above + */ + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* S/PDIF hardware input 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0112, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0113, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0116, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0117, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011a, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011b, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011e, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011f, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x00; + /* S/PDIF software return 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010a, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* S/PDIF software return 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010b, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010e, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x00; + /* S/PDIF software return 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010f, 0x4001, &com_buff, 2); + + com_buff[0] = 0x00; + com_buff[1] = 0x80; + /* S/PDIF fx returns left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0122, 0x4001, &com_buff, 2); + /* S/PDIF fx returns right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0123, 0x4001, &com_buff, 2); + + /* Set the dropdown "Effect" to the first option */ + /* Room1 = 0x00 */ + /* Room2 = 0x01 */ + /* Room3 = 0x02 */ + /* Hall 1 = 0x03 */ + /* Hall 2 = 0x04 */ + /* Plate = 0x05 */ + /* Delay = 0x06 */ + /* Echo = 0x07 */ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0200, 0x4301, &com_buff, 1); /* max is 0xff */ + /* min is 0x00 */ + + + /* Set the effect duration to 0 */ + /* max is 0xffff */ + /* min is 0x0000 */ + com_buff[0] = 0x00; + com_buff[1] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0400, 0x4301, &com_buff, 2); + + /* Set the effect volume and feedback to 0 */ + /* max is 0xff */ + /* min is 0x00 */ + com_buff[0] = 0x00; + /* feedback: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0500, 0x4301, &com_buff, 1); + /* volume: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0300, 0x4301, &com_buff, 1); + + /* Set soft button hold duration */ + /* 0x03 = 250ms */ + /* 0x05 = 500ms DEFAULT */ + /* 0x08 = 750ms */ + /* 0x0a = 1sec */ + com_buff[0] = 0x05; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0005, 0x2001, &com_buff, 1); + + /* Use dim LEDs for button of state */ + com_buff[0] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0004, 0x2001, &com_buff, 1); +} + +#define MBOX3_DESCRIPTOR_SIZE 464 + +static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + int descriptor_size; + + descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); + + if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { + dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + return -ENODEV; + } + + dev_dbg(&dev->dev, "device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) + dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + + err = usb_reset_configuration(dev); + if (err < 0) + dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); + dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + mbox3_setup_48_24_magic(dev); + dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + + return 0; /* Successful boot */ +} #define MICROBOOK_BUF_SIZE 128 @@ -1324,6 +1622,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x0dba, 0x3000): /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); + case USB_ID(0x0dba, 0x5000): + /* Digidesign Mbox 3 */ + return snd_usb_mbox3_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ -- 2.35.1

1 0

[PATCH 0/4] ALSA: usb-audio: More EP-related fixes
by Takashi Iwai 09 Oct '22

09 Oct '22

This is a series of small fixes for USB-audio endpoint management after the latest change to split prepare/hw_params. Takashi === Takashi Iwai (4): ALSA: usb-audio: Avoid unnecessary interface change at EP close ALSA: usb-audio: Apply mutex around snd_usb_endpoint_set_params() ALSA: usb-audio: Correct the return code from snd_usb_endpoint_set_params() ALSA: usb-audio: Avoid superfluous endpoint setup sound/usb/card.h | 3 ++- sound/usb/endpoint.c | 32 +++++++++++++++++++++++++------- 2 files changed, 27 insertions(+), 8 deletions(-) -- 2.35.3

1 4

[PATCH] ALSA: hda: Update register polling macros
by Amadeusz Sławiński 09 Oct '22

09 Oct '22

Recent commit d91857059def ("ALSA: hda: Rework snd_hdac_stream_reset() to use macros") missed that on some devices register access needs to be done with unaligned access helper. Change polling macros to use read_poll_timeout_atomic() in order to specify register read function. Reported-by: Jon Hunter <jonathanh(a)nvidia.com> Link: https://lore.kernel.org/alsa-devel/20220818141517.109280-1-amadeuszx.slawin… Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- include/sound/hdaudio.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index ddff03e546e9..35778f953a3f 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -592,11 +592,11 @@ int snd_hdac_get_stream_stripe_ctl(struct hdac_bus *bus, #define snd_hdac_stream_readb(dev, reg) \ snd_hdac_reg_readb((dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg) #define snd_hdac_stream_readb_poll(dev, reg, val, cond, delay_us, timeout_us) \ - readb_poll_timeout((dev)->sd_addr + AZX_REG_ ## reg, val, cond, \ - delay_us, timeout_us) + read_poll_timeout_atomic(snd_hdac_reg_readb, val, cond, delay_us, timeout_us, \ + false, (dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg) #define snd_hdac_stream_readl_poll(dev, reg, val, cond, delay_us, timeout_us) \ - readl_poll_timeout((dev)->sd_addr + AZX_REG_ ## reg, val, cond, \ - delay_us, timeout_us) + read_poll_timeout_atomic(snd_hdac_reg_readl, val, cond, delay_us, timeout_us, \ + false, (dev)->bus, (dev)->sd_addr + AZX_REG_ ## reg) /* update a register, pass without AZX_REG_ prefix */ #define snd_hdac_stream_updatel(dev, reg, mask, val) \ -- 2.25.1

2 1

[PATCH v2] sound: soc: codecs: wcd-mbhc-v2: expose ALSA control for jack
by Dylan Van Assche 09 Oct '22

09 Oct '22

Jack detection is currently fully functional via the input interface together with multimedia buttons, but is not exposed as an ALSA control. Therefore, ALSA clients such as PulseAudio do not pick up the jack detection events as they only support one of the possible interface (ALSA control or input interface, but not both). Expose the jack events as an ALSA control and input interface to provide ALSA clients both interfaces. Signed-off-by: Dylan Van Assche <me(a)dylanvanassche.be> --- sound/soc/codecs/wcd-mbhc-v2.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/wcd-mbhc-v2.c b/sound/soc/codecs/wcd-mbhc-v2.c index 1911750f7445..1b75f49b2858 100644 --- a/sound/soc/codecs/wcd-mbhc-v2.c +++ b/sound/soc/codecs/wcd-mbhc-v2.c @@ -725,6 +725,10 @@ static int wcd_mbhc_initialise(struct wcd_mbhc *mbhc) mutex_lock(&mbhc->lock); + ret = snd_jack_add_new_kctl(mbhc->jack->jack, "Headset Jack", WCD_MBHC_JACK_MASK); + if (ret) + dev_warn(component->dev, "failed creating Headset Jack kctl with err: %d\n", ret); + /* enable HS detection */ if (mbhc->mbhc_cb->hph_pull_up_control_v2) mbhc->mbhc_cb->hph_pull_up_control_v2(component, -- 2.37.3

3 2

alsa-lib-1.2.7.2/src/pcm/pcm.c:3238 triggers SIGSEGV
by GitHub issues - edited 07 Oct '22

07 Oct '22

alsa-project/alsa-lib issue #274 was edited from mmokrejs: Hi, I tried an example command `espeak-ng "This is a test"` command but it crahes inside `libasound2`. ``` [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `espeak-ng This is a test'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f672678ea6b in __memmove_avx_unaligned_erms () from /lib64/libc.so.6 [Current thread is 1 (Thread 0x7f6725d4d640 (LWP 9533))] (gdb) where #0 0x00007f672678ea6b in __memmove_avx_unaligned_erms () from /lib64/libc.so.6 #1 0x00007f6726492ffd in snd_pcm_area_copy (dst_area=0x7f672001fc90, dst_offset=18, src_area=0x7f672002a4c0, src_offset=0, samples=4026531840, format=SND_PCM_FORMAT_S16_LE) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm.c:3238 #2 0x00007f67264934af in snd_pcm_areas_copy (dst_areas=0x7f672001fca0, dst_offset=18, src_areas=0x7f672002a4d0, src_offset=0, channels=1, frames=18446744073709551608, format=SND_PCM_FORMAT_S16_LE) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm.c:3389 #3 0x00007f67264ba368 in snd_pcm_rate_commit_area (pcm=0x7f6720061cf0, rate=0x7f6720061b00, appl_offset=7508, size=10, slave_size=22) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm_rate.c:817 #4 0x00007f67264bb217 in snd_pcm_rate_drain (pcm=0x7f6720061cf0) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm_rate.c:1161 #5 0x00007f6726490548 in snd_pcm_drain (pcm=0x7f6720061700) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm.c:1359 #6 0x00007f67266347e2 in alsa_object_drain () from /usr/lib64/libpcaudio.so.0 #7 0x00007f6726873c4e in sync_espeak_Synth () from /usr/lib64/libespeak-ng.so.1 #8 0x00007f672688e813 in process_espeak_command () from /usr/lib64/libespeak-ng.so.1 #9 0x00007f672688f708 in say_thread () from /usr/lib64/libespeak-ng.so.1 #10 0x00007f67266c2485 in start_thread () from /lib64/libc.so.6 #11 0x00007f672674610c in clone3 () from /lib64/libc.so.6 (gdb) where #0 0x00007f672678ea6b in __memmove_avx_unaligned_erms () from /lib64/libc.so.6 #1 0x00007f6726492ffd in snd_pcm_area_copy (dst_area=0x7f672001fc90, dst_offset=18, src_area=0x7f672002a4c0, src_offset=0, samples=4026531840, format=SND_PCM_FORMAT_S16_LE) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm.c:3238 #2 0x00007f67264934af in snd_pcm_areas_copy (dst_areas=0x7f672001fca0, dst_offset=18, src_areas=0x7f672002a4d0, src_offset=0, channels=1, frames=18446744073709551608, format=SND_PCM_FORMAT_S16_LE) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm.c:3389 #3 0x00007f67264ba368 in snd_pcm_rate_commit_area (pcm=0x7f6720061cf0, rate=0x7f6720061b00, appl_offset=7508, size=10, slave_size=22) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm_rate.c:817 #4 0x00007f67264bb217 in snd_pcm_rate_drain (pcm=0x7f6720061cf0) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm_rate.c:1161 #5 0x00007f6726490548 in snd_pcm_drain (pcm=0x7f6720061700) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm.c:1359 #6 0x00007f67266347e2 in alsa_object_drain () from /usr/lib64/libpcaudio.so.0 #7 0x00007f6726873c4e in sync_espeak_Synth () from /usr/lib64/libespeak-ng.so.1 #8 0x00007f672688e813 in process_espeak_command () from /usr/lib64/libespeak-ng.so.1 #9 0x00007f672688f708 in say_thread () from /usr/lib64/libespeak-ng.so.1 #10 0x00007f67266c2485 in start_thread () from /lib64/libc.so.6 #11 0x00007f672674610c in clone3 () from /lib64/libc.so.6 (gdb) bt full #0 0x00007f672678ea6b in __memmove_avx_unaligned_erms () from /lib64/libc.so.6 No symbol table info available. #1 0x00007f6726492ffd in snd_pcm_area_copy (dst_area=0x7f672001fc90, dst_offset=18, src_area=0x7f672002a4c0, src_offset=0, samples=4026531840, format=SND_PCM_FORMAT_S16_LE) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm.c:3238 bytes = 536870896 src = 0x7f672006ba50 "A\003+\002\205\001\252\001M\001\025\001P\002\317\001\227\001\324\375`\001\341\001Y\377P\002k\377\\" dst = 0x7f6720038814 "P\002\317\001\227\001\324\375`\001\341\001Y\377P\002k\377\\" width = 16 src_step = 642318657 dst_step = 32615 #2 0x00007f67264934af in snd_pcm_areas_copy (dst_areas=0x7f672001fca0, dst_offset=18, src_areas=0x7f672002a4d0, src_offset=0, channels=1, frames=18446744073709551608, format=SND_PCM_FORMAT_S16_LE) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm.c:3389 step = 16 src_addr = 0x7f672006ba50 src_start = 0x7f672002a4c0 channels1 = 0 chns = 1 dst_addr = 0x7f67200387f0 dst_start = 0x7f672001fc90 width = 16 #3 0x00007f67264ba368 in snd_pcm_rate_commit_area (pcm=0x7f6720061cf0, rate=0x7f6720061b00, appl_offset=7508, size=10, slave_size=22) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm_rate.c:817 cont = 18 areas = 0x7f672002a4c0 slave_areas = 0x38c slave_offset = 13422172416 xfer = 2199023255787 slave_frames = 18446744073709551615 result = 140080984498990 #4 0x00007f67264bb217 in snd_pcm_rate_drain (pcm=0x7f6720061cf0) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm_rate.c:1161 psize = 10 spsize = 22 err = 1 sw_params = {tstamp_mode = 0, period_step = 1, sleep_min = 0, avail_min = 1, xfer_align = 1, start_threshold = 2, stop_threshold = 16384, silence_threshold = 0, silence_size = 0, boundary = 4611686018427387904, proto = 131087, tstamp_type = 1, reserved = '\000' <repeats 55 times>} size = 10 ofs = 7508 saved_avail_min = 1024 commit_err = 0 rate = 0x7f6720061b00 #5 0x00007f6726490548 in snd_pcm_drain (pcm=0x7f6720061700) at /mnt/nocrypt/var/tmp/portage/portage/media-libs/alsa-lib-1.2.7.2-r1/work/alsa-lib-1.2.7.2/src/pcm/pcm.c:1359 err = 0 #6 0x00007f67266347e2 in alsa_object_drain () from /usr/lib64/libpcaudio.so.0 No symbol table info available. #7 0x00007f6726873c4e in sync_espeak_Synth () from /usr/lib64/libespeak-ng.so.1 No symbol table info available. #8 0x00007f672688e813 in process_espeak_command () from /usr/lib64/libespeak-ng.so.1 No symbol table info available. #9 0x00007f672688f708 in say_thread () from /usr/lib64/libespeak-ng.so.1 No symbol table info available. #10 0x00007f67266c2485 in start_thread () from /lib64/libc.so.6 No symbol table info available. #11 0x00007f672674610c in clone3 () from /lib64/libc.so.6 No symbol table info available. (gdb) quit ``` Does it help? Issue URL : https://github.com/alsa-project/alsa-lib/issues/274 Repository URL: https://github.com/alsa-project/alsa-lib

1 0