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

[PATCH v1] ASoC: rsnd: add interrupt support for SSI BUSIF buffer
by Yongbo Zhang 11 May '20

11 May '20

SSI BUSIF buffer is possible to overflow or underflow, especially in a hypervisor environment. If there is no interrupt support, it will eventually lead to errors in pcm data. This patch adds overflow and underflow interrupt support for SSI BUSIF buffer. Reported-by: Chen Li <licheng0822(a)thundersoft.com> Signed-off-by: Yongbo Zhang <giraffesnn123(a)gmail.com> Tested-by: Chen Li <licheng0822(a)thundersoft.com> Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/gen.c | 8 +++ sound/soc/sh/rcar/rsnd.h | 9 +++ sound/soc/sh/rcar/ssi.c | 142 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 159 insertions(+) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index af19010b9d88..8bd49c8a9517 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -224,6 +224,14 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_S_REG(SSI_SYS_STATUS5, 0x884), RSND_GEN_S_REG(SSI_SYS_STATUS6, 0x888), RSND_GEN_S_REG(SSI_SYS_STATUS7, 0x88c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE0, 0x850), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE1, 0x854), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE2, 0x858), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE3, 0x85c), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE4, 0x890), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE5, 0x894), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE6, 0x898), + RSND_GEN_S_REG(SSI_SYS_INT_ENABLE7, 0x89c), RSND_GEN_S_REG(HDMI0_SEL, 0x9e0), RSND_GEN_S_REG(HDMI1_SEL, 0x9e4), diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ea6cbaa9743e..d47608ff5fac 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -189,6 +189,14 @@ enum rsnd_reg { SSI_SYS_STATUS5, SSI_SYS_STATUS6, SSI_SYS_STATUS7, + SSI_SYS_INT_ENABLE0, + SSI_SYS_INT_ENABLE1, + SSI_SYS_INT_ENABLE2, + SSI_SYS_INT_ENABLE3, + SSI_SYS_INT_ENABLE4, + SSI_SYS_INT_ENABLE5, + SSI_SYS_INT_ENABLE6, + SSI_SYS_INT_ENABLE7, HDMI0_SEL, HDMI1_SEL, SSI9_BUSIF0_MODE, @@ -237,6 +245,7 @@ enum rsnd_reg { #define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i)) #define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i)) #define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) +#define SSI_SYS_INT_ENABLE(i) (SSI_SYS_INT_ENABLE0 + (i)) struct rsnd_priv; diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4a7d3413917f..87a2182a4514 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -372,6 +372,9 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, u32 wsr = ssi->wsr; int width; int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; is_tdm = rsnd_runtime_is_tdm(io); is_tdm_split = rsnd_runtime_is_tdm_split(io); @@ -447,6 +450,38 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } + /* enable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable |= 0xf << (id * 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable |= 0xf << 4; + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + } + init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; @@ -496,6 +531,13 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, { struct rsnd_ssi *ssi = rsnd_mod_to_ssi(mod); struct device *dev = rsnd_priv_to_dev(priv); + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + int i; + u32 sys_int_enable = 0; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (!rsnd_ssi_is_run_mods(mod, io)) return 0; @@ -517,6 +559,37 @@ static int rsnd_ssi_quit(struct rsnd_mod *mod, ssi->wsr = 0; } + /* disable busif buffer over/under run interrupt. */ + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE(i * 2)); + sys_int_enable &= ~(0xf << (id * 4)); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE(i * 2), + sys_int_enable); + } + + break; + case 9: + for (i = 0; i < 4; i++) { + sys_int_enable = rsnd_mod_read(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1)); + sys_int_enable &= ~(0xf << 4); + rsnd_mod_write(mod, + SSI_SYS_INT_ENABLE((i * 2) + 1), + sys_int_enable); + } + + break; + } + return 0; } @@ -622,6 +695,11 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, int enable) { u32 val = 0; + int is_tdm, is_tdm_split; + int id = rsnd_mod_id(mod); + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); if (rsnd_is_gen1(priv)) return 0; @@ -635,6 +713,19 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod, if (enable) val = rsnd_ssi_is_dma_mode(mod) ? 0x0e000000 : 0x0f000000; + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + case 9: + val |= 0x0000ff00; + break; + } + } + rsnd_mod_write(mod, SSI_INT_ENABLE, val); return 0; @@ -651,6 +742,10 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, u32 status; bool elapsed = false; bool stop = false; + int is_tdm, is_tdm_split; + + is_tdm = rsnd_runtime_is_tdm(io); + is_tdm_split = rsnd_runtime_is_tdm_split(io); spin_lock(&priv->lock); @@ -672,6 +767,53 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, stop = true; } + status = 0; + + if (is_tdm || is_tdm_split) { + switch (id) { + case 0: + case 1: + case 2: + case 3: + case 4: + for (i = 0; i < 4; i++) { + status = rsnd_mod_read(mod, + SSI_SYS_STATUS(i * 2)); + status &= 0xf << (id * 4); + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS(i * 2), + 0xf << (id * 4)); + stop = true; + break; + } + } + break; + case 9: + for (i = 0; i < 4; i++) { + status = rsnd_mod_write(mod, + SSI_SYS_STATUS((i * 2) + 1)); + status &= 0xf << 4; + + if (status) { + rsnd_dbg_irq_status(dev, + "%s err status : 0x%08x\n", + rsnd_mod_name(mod), status); + rsnd_mod_write(mod, + SSI_SYS_STATUS((i * 2) + 1), + 0xf << 4); + stop = true; + break; + } + } + break; + } + } + rsnd_ssi_status_clear(mod); rsnd_ssi_interrupt_out: spin_unlock(&priv->lock); -- 2.26.2

1 0

seq: user_client: use constant qualifier for read-only parameter
by GitHub pull_request - opened 11 May '20

11 May '20

alsa-project/alsa-gobject pull request #20 was opened from takaswie: The 'port_id' parameters of 'alsaseq_user_client_create_port()' is a pointer to storage of unsigned char type, due to optional parameter in g-i interface. It's read-only if having value and it's better to have const qualifier. Request URL : https://github.com/alsa-project/alsa-gobject/pull/20 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/20.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

Fix pour Asus Xonar U5 card SPIF in USB-Audio.conf
by GitHub issues - edited 11 May '20

11 May '20

alsa-project/alsa-lib issue #48 was edited from burningd0g: It would be greatly apreciated to add the iec958 correct device in the default USB-Audio.conf for the Xonar USB sound card : USB-Audio.pcm.iec958_device { ... "ASUS XONAR U5" 1 "XONAR U5" 1 ... } It would prevent me (and maybe some other people) to freak out on why it doesn't work out of the box! :) I know the issue is also present on the U7 but i don't have all the possible device name for it. Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/48 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fix pour Asus Xonar U5 card SPIF in USB-Audio.conf
by GitHub issues - opened 11 May '20

11 May '20

alsa-project/alsa-lib issue #48 was opened from burningd0g: Issue URL : https://github.com/alsa-project/alsa-lib/issues/48 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: firewire-motu: fulfill missing entries in Kconfig
by Takashi Sakamoto 11 May '20

11 May '20

This is trivial commit but fulfill missing entries in Kconfig for devices supported by ALSA firewire-motu driver. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index f7ed69d60310..25778765cbfe 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -150,8 +150,12 @@ config SND_FIREWIRE_MOTU Say Y here to enable support for FireWire devices which MOTU produced: * 828mk2 * Traveler - * 828mk3 + * Ultralite + * 8pre + * 828mk3 (FireWire only) + * 828mk3 (Hybrid) * Audio Express + * 4pre To compile this driver as a module, choose M here: the module will be called snd-firewire-motu. -- 2.25.1

2 1

[PATCH v2 0/2] soundwire: remove multiple superfluous "else" statements
by Guennadi Liakhovetski 11 May '20

11 May '20

Two patches cleaning up the same issue for soundwire core and Intel- specific code. v2: split into 2 for core and Intel code Guennadi Liakhovetski (2): soundwire: (cosmetic) remove multiple superfluous "else" statements soundwire: intel: (cosmetic) remove multiple superfluous "else" statements drivers/soundwire/bus.c | 17 ++++++++++------- drivers/soundwire/cadence_master.c | 8 ++++++-- drivers/soundwire/intel_init.c | 4 +++- 3 files changed, 19 insertions(+), 10 deletions(-) -- 1.9.3

2 3

[PATCH -next] soundwire: qcom: Use IRQF_ONESHOT
by Samuel Zou 11 May '20

11 May '20

Fixes coccicheck error: drivers/soundwire/qcom.c:815:7-32: ERROR: Threaded IRQ with no primary handler requested without IRQF_ONESHOT Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Samuel Zou <zou_wei(a)huawei.com> --- drivers/soundwire/qcom.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index e08a17c..a2a35f7 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -814,7 +814,8 @@ static int qcom_swrm_probe(struct platform_device *pdev) ret = devm_request_threaded_irq(dev, ctrl->irq, NULL, qcom_swrm_irq_handler, - IRQF_TRIGGER_RISING, + IRQF_TRIGGER_RISING | + IRQF_ONESHOT, "soundwire", ctrl); if (ret) { dev_err(dev, "Failed to request soundwire irq\n"); -- 2.6.2

2 1

[PATCH] ASoC: core: Make sure component driver names are unique
by Ranjani Sridharan 11 May '20

11 May '20

When registering a component, make sure that the driver names are unique. This will ensure that the snd_soc_rtdcom_lookup() function returns the right component based on the name. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-core.c | 31 +++++++++++++++++++++++-------- 1 file changed, 23 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 8321e75ff244..00c1f8ce46af 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -286,14 +286,6 @@ struct snd_soc_component *snd_soc_rtdcom_lookup(struct snd_soc_pcm_runtime *rtd, if (!driver_name) return NULL; - /* - * NOTE - * - * snd_soc_rtdcom_lookup() will find component from rtd by using - * specified driver name. - * But, if many components which have same driver name are connected - * to 1 rtd, this function will return 1st found component. - */ for_each_rtd_components(rtd, i, component) { const char *component_name = component->driver->name; @@ -2640,6 +2632,22 @@ int snd_soc_add_component(struct device *dev, } EXPORT_SYMBOL_GPL(snd_soc_add_component); +static bool +snd_soc_component_driver_name_is_unique(const struct snd_soc_component_driver *component_driver) +{ + struct snd_soc_component *component; + + mutex_lock(&client_mutex); + for_each_component(component) + if (!strcmp(component->driver->name, component_driver->name)) { + mutex_unlock(&client_mutex); + return false; + } + + mutex_unlock(&client_mutex); + return true; +} + int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *component_driver, struct snd_soc_dai_driver *dai_drv, @@ -2647,6 +2655,13 @@ int snd_soc_register_component(struct device *dev, { struct snd_soc_component *component; + if (component_driver->name && + !snd_soc_component_driver_name_is_unique(component_driver)) { + dev_err(dev, "component driver name %s is not unique\n", + component_driver->name); + return -EINVAL; + } + component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL); if (!component) return -ENOMEM; -- 2.17.1

3 6

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

10 May '20

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

2 2

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

10 May '20

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

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