July 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: ops: Don't modify the driver's plaform_max when reading state
by Mark Brown 22 Aug '22

22 Aug '22

Currently snd_soc_info_volsw() will set a platform_max based on the limit the control has if one is not already set. This isn't really great, we shouldn't be modifying the passed in driver data especially in a path like this which may not ever be executed or where we may execute other callbacks before this one. Instead make this function leave the data unchanged, and clarify things a bit by referring to max rather than platform_max within the function. platform_max is now applied as a limit after working out the natural maximum value for the control. This means that platform_max is no longer treated as a direct register value for controls were min is non-zero. The put() callbacks already validate on this basis, and there do not appear to be any in tree users that would be affected. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-ops.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 2d39370ddeca..93e72a016b4d 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -176,20 +176,21 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, { struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - int platform_max; + int max; - if (!mc->platform_max) - mc->platform_max = mc->max; - platform_max = mc->platform_max; + max = uinfo->value.integer.max = mc->max - mc->min; + if (mc->platform_max && mc->platform_max < max) + max = mc->platform_max; - if (platform_max == 1 && !strstr(kcontrol->id.name, " Volume")) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max - mc->min; + uinfo->value.integer.max = max; + return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); -- 2.30.2

2 6

[PATCH] Add quirk to enable Avid Mbox 3 support
by connerknoxpublic＠gmail.com 22 Aug '22

22 Aug '22

From: Conner Knox <connerknoxpublic(a)gmail.com> Hi all, I believe these are the right emails to send this patch to, but if I've missed something please let me know. This patch enables the Avid Mbox 3 audio interface to be used through ALSA. It locks the device at a 48khz sample rate. Then it mutes and raises the appropriate channels/inputs to allow for expected functionality. Thanks for your time. --- sound/usb/quirks-table.h | 76 ++++++++++ sound/usb/quirks.c | 311 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 387 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0ea39565e623..5f43863e5a5f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2978,6 +2978,82 @@ 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 ab9f3da49941..e7caa7eeb955 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1020,6 +1020,313 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } +static void mbox3_setup_48_24_magic(struct usb_device *dev) +{ + u8 com_buff[4]; + + /* Load 48000Hz rate into buffer */ + com_buff[0] = 0x80; + com_buff[1] = 0xbb; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + + /* Set 48000Hz sample rate */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x0001, &com_buff, 0x0004); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &com_buff, 0x0004); + + /* Set clock source to Internal (as opposed to S/PDIF) */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x01; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + 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. + * The Mbox 3 is little endian and + * the data written here seems a bit odd. + * 0x8000 (shown in big endian form) is muted + * and volume increases to 0xffff it seems. + * I've yet to observe 0xffff being sent. + * Instead, full volume seems to be 0x0000. + * Per my understanding of 16 bit integers, this is seems strange, but maybe I'm missing something + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* 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 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue software return 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4001, &com_buff, 2); + /* Analogue software return 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4001, &com_buff, 2); + /* Analogue software return 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4001, &com_buff, 2); + /* Analogue software return 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4001, &com_buff, 2); + /* 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 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* 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 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x02; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + + /* Do not dim fx returns */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0002, 0x2001, &com_buff, 1); + + /* Do not set fx returns to mono */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 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 + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* 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); + /* S/PDIF software return 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010a, 0x4001, &com_buff, 2); + /* 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); + /* S/PDIF software return 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010f, 0x4001, &com_buff, 2); + /* 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 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0200, 0x4301, &com_buff, 1); + + /* Set the effect duration to 0 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 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 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 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 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x05; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0005, 0x2001, &com_buff, 1); + + /* Use dim LEDs for button of state */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0004, 0x2001, &com_buff, 1); + + return; +} + +#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 +1631,10 @@ 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

4 8

[PATCH 1/2] ASoC: Intel: Skylake: fix error message of NHLT blob selection
by Icenowy Zheng 18 Aug '22

18 Aug '22

The query to select NHLT blob had been changed to use parameters of the pipeline instead of PCM, however the error message returned when it fails still uses PCM properties. Fix this inconsistency. Fixes: 87b265260046 ("ASoC: Intel: Skylake: Select proper format for NHLT blob") Signed-off-by: Icenowy Zheng <uwu(a)icenowy.me> --- sound/soc/intel/skylake/skl-topology.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 9bdf020a2b64..19994ec8bba1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -1864,7 +1864,8 @@ static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai, } else { dev_err(dai->dev, "Blob NULL for id:%d type:%d dirn:%d ch:%d, freq:%d, fmt:%d\n", mconfig->vbus_id, link_type, params->stream, - params->ch, params->s_freq, params->s_fmt); + pipe_fmt->channels, pipe_fmt->freq, + pipe_fmt->bps); return -EINVAL; } -- 2.37.1

3 9

[PATCH 0/4] ASoC/soundwire: log actual PING status on resume issues
by Bard Liao 17 Aug '22

17 Aug '22

we've been stuck with problems in the dual-amplifier configurations where one of the two devices seems to become UNATTACHED and never regains sync, see https://github.com/thesofproject/linux/issues/3638. This is a rather infrequent issue that may happen once or twice per month, but still it remains a concern. One possibility is that the device does lose sync but somehow our hardware detection fails to see it resync. This series just adds a basic read directly from the PING frames to help confirm if yes/no the device regain sync. The change is mainly on soundwire. @Mark, Could you ack the ASoC patch if it looks good to you? Pierre-Louis Bossart (4): soundwire: add read_ping_status helper definition in manager ops soundwire: intel/cadence: expose PING status in manager ops soundwire: add sdw_show_ping_status() helper ASoC: codecs: show PING status on resume failures drivers/soundwire/bus.c | 32 ++++++++++++++++++++++++++++++ drivers/soundwire/cadence_master.c | 8 ++++++++ drivers/soundwire/cadence_master.h | 2 ++ drivers/soundwire/intel.c | 1 + include/linux/soundwire/sdw.h | 5 +++++ sound/soc/codecs/max98373-sdw.c | 2 ++ sound/soc/codecs/rt1308-sdw.c | 2 ++ sound/soc/codecs/rt1316-sdw.c | 2 ++ sound/soc/codecs/rt5682-sdw.c | 2 ++ sound/soc/codecs/rt700-sdw.c | 2 ++ sound/soc/codecs/rt711-sdca-sdw.c | 2 ++ sound/soc/codecs/rt715-sdca-sdw.c | 2 ++ sound/soc/codecs/rt715-sdw.c | 2 ++ 13 files changed, 64 insertions(+) -- 2.25.1

4 10

[PATCH] soundwire: qcom: Check device status before reading devid
by Srinivas Kandagatla 13 Aug '22

13 Aug '22

As per hardware datasheet its recommended that we check the device status before reading devid assigned by auto-enumeration. Without this patch we see SoundWire devices with invalid enumeration addresses on the bus. Cc: stable(a)vger.kernel.org Fixes: a6e6581942ca ("soundwire: qcom: add auto enumeration support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index dbf793e7e5d7..9df970eeca45 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -480,6 +480,10 @@ static int qcom_swrm_enumerate(struct sdw_bus *bus) char *buf1 = (char *)&val1, *buf2 = (char *)&val2; for (i = 1; i <= SDW_MAX_DEVICES; i++) { + /* do not continue if the status is Not Present */ + if (!ctrl->status[i]) + continue; + /*SCP_Devid5 - Devid 4*/ ctrl->reg_read(ctrl, SWRM_ENUMERATOR_SLAVE_DEV_ID_1(i), &val1); -- 2.25.1

3 3

[PATCH] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 10 Aug '22

10 Aug '22

Convert the NXP SAI binding to DT schema format using json-schema. The Synchronous Audio Interface (SAI) provides an interface that supports full-duplex serial interfaces with frame synchronization formats such as I2S, AC97, TDM, and codec/DSP interfaces. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../devicetree/bindings/sound/fsl,sai.yaml | 175 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 ---------- 2 files changed, 175 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..adcd77531eba --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,175 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + minItems: 1 + maxItems: 2 + items: + enum: + - fsl,vf610-sai + - fsl,imx6sx-sai + - fsl,imx6ul-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - fsl,imx8ulp-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + minItems: 2 + maxItems: 2 + description: + Must contain a list of pairs of references to DMA specifiers, one for + transmission, and one for reception. + + dma-names: + minItems: 2 + maxItems: 2 + items: + enum: + - tx + - rx + + clocks: + minItems: 4 + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted, compatible for old dts) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + + clock-names: + minItems: 4 + maxItems: 7 + items: + enum: + - bus + - mclk0 + - mclk1 + - mclk2 + - mclk3 + - pll8k + - pll11k + + lsb-first: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Boolean property, required if all the SAI + registers are big-endian rather than little-endian. + + fsl,sai-synchronous-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, indicating + that SAI will work in the synchronous mode (sync Tx + with Rx) which means both the transmitter and the + receiver will send and receive data by following + receiver's bit clocks and frame sync clocks. + fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. + + fsl,sai-asynchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, indicating + that SAI will work in the asynchronous mode, which + means both transmitter and receiver will send and + receive data by following their own bit clocks and + frame sync clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transmitter and receiver will send and receive data by following clocks + of transmitter. + fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + configure the dataline. it has 3 value for each configuration + first one means the type: I2S(1) or PDM(2) + second one is dataline mask for 'rx' + third one is dataline mask for 'tx'. + for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type + rx mask is 0xff, tx mask is 0x11 (dataline 1 and 5 enabled). + + fsl,sai-mclk-direction-output: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, + indicates that SAI will output the SAI MCLK clock. + + fsl,shared-interrupt: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, + indicates that interrupt is shared with other modules. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/vf610-clock.h> + sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "rx", "tx"; + dmas = <&edma0 0 20>, + <&edma0 0 21>; + big-endian; + lsb-first; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt deleted file mode 100644 index fbdefc3fade7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ /dev/null @@ -1,95 +0,0 @@ -Freescale Synchronous Audio Interface (SAI). - -The SAI is based on I2S module that used communicating with audio codecs, -which provides a synchronous audio interface that supports fullduplex -serial interfaces with frame synchronization such as I2S, AC97, TDM, and -codec/DSP interfaces. - -Required properties: - - - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai", - "fsl,imx8qm-sai", "fsl,imx8mm-sai", - "fsl,imx8mn-sai", "fsl,imx8mp-sai", or - "fsl,imx8ulp-sai". - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "bus" for register access and - "mclk1", "mclk2", "mclk3" for bit clock and frame - clock providing. - "pll8k", "pll11k" are optional, they are the clock - source for root clock, one is for 8kHz series rates - another one is for 11kHz series rates. - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - pinctrl-names : Must contain a "default" entry. - - - pinctrl-NNN : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. - - - lsb-first : Configures whether the LSB or the MSB is transmitted - first for the fifo data. If this property is absent, - the MSB is transmitted first as default, or the LSB - is transmitted first. - - - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transmitter and the - receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. - - - fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which - means both transmitter and receiver will send and - receive data by following their own bit clocks and - frame sync clocks separately. - - - fsl,dataline : configure the dataline. it has 3 value for each configuration - first one means the type: I2S(1) or PDM(2) - second one is dataline mask for 'rx' - third one is dataline mask for 'tx'. - for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; - it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type - rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). - -Optional properties: - - - big-endian : Boolean property, required if all the SAI - registers are big-endian rather than little-endian. - -Optional properties (for mx6ul): - - - fsl,sai-mclk-direction-output: This is a boolean property. If present, - indicates that SAI will output the SAI MCLK clock. - -Note: -- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the - default synchronous mode (sync Rx with Tx) will be used, which means both - transmitter and receiver will send and receive data by following clocks - of transmitter. -- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. - -Example: -sai2: sai@40031000 { - compatible = "fsl,vf610-sai"; - reg = <0x40031000 0x1000>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_PLATFORM_BUS>, - <&clks VF610_CLK_SAI2>, - <&clks 0>, <&clks 0>; - clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, - <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; - big-endian; - lsb-first; -}; -- 2.34.1

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[PATCH v2] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Biju Das 05 Aug '22

05 Aug '22

We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- v1->v2: Replaced the label err_dma->err_reset. --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 0d0594a0e4f6..7ace0c0db5b1 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1017,32 +1017,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.25.1

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[PATCH] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 05 Aug '22

05 Aug '22

In commit <c3ecef21c3f26> ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ffc24afb5a7a..f0602077b385 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1054,7 +1054,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

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[PATCH] ASoC: fsl_asrc_dma: fully initialize structs
by Sascha Hauer 05 Aug '22

05 Aug '22

The driver uses two statically ininitialized struct dma_slave_config, but only one of them is initialized to zero. Initialize config_be to zero as well to make sure that no fields are filled with random values. Let the compiler do this instead of explicitly calling memset() which makes it easier to read. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_asrc_dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5038faf035cba..fec56dbb64e60 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -138,7 +138,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct dma_chan *tmp_chan = NULL, *be_chan = NULL; struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; - struct dma_slave_config config_fe, config_be; + struct dma_slave_config config_fe = {}, config_be = {}; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; struct device_node *of_dma_node; @@ -181,7 +181,6 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, return -EINVAL; } - memset(&config_fe, 0, sizeof(config_fe)); ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe); if (ret) { dev_err(dev, "failed to prepare DMA config for Front-End\n"); -- 2.30.2

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[PATCH] soc: sof: fix clang -Wformat warnings
by Justin Stitt 03 Aug '22

03 Aug '22

When building with Clang we encounter these warnings: | sound/soc/sof/ipc3-topology.c:2343:4: error: format specifies type | 'unsigned char' but the argument has type 'int' [-Werror,-Wformat] | SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); | ^~~~~~~~~~~~~~~^~~~~~~~~~~~~~~^~~~~~~~~~~~~ Use correct format specifier `%d` since args are of type int. Link: https://github.com/ClangBuiltLinux/linux/issues/378 Reported-by: Nathan Chancellor <nathan(a)kernel.org> Suggested-by: Nathan Chancellor <nathan(a)kernel.org> Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- Reported by Nathan here: https://lore.kernel.org/all/YtmrCJjQrSbv8Aj1@dev-arch.thelio-3990X/ sound/soc/sof/ipc3-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index b2cc046b9f60..65923e7a5976 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2338,7 +2338,7 @@ static int sof_ipc3_parse_manifest(struct snd_soc_component *scomp, int index, } dev_info(scomp->dev, - "Topology: ABI %d:%d:%d Kernel ABI %hhu:%hhu:%hhu\n", + "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", man->priv.data[0], man->priv.data[1], man->priv.data[2], SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); -- 2.37.1.359.gd136c6c3e2-goog

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