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

[alsa-devel] [GIT PULL] ASoC fixes for v5.5
by Mark Brown 16 Jan '20

16 Jan '20

The following changes since commit ff3b57417012fcc963ec281f5705bed837e4b1ac: ASoC: Intel: boards: Fix compile-testing RT1011/RT5682 (2020-01-03 00:56:47 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.5-rc6 for you to fetch changes up to 85578bbd642f65065039b1765ebe1a867d5435b0: ASoC: msm8916-wcd-digital: Reset RX interpolation path after use (2020-01-13 15:12:28 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.5 This is mostly driver specific fixes, plus an error handling fix in the core. There is a rather large diffstat for the stm32 SAI driver, this is a very large but mostly mechanical update which wraps every register access in the driver to allow a fix to the locking which avoids circular locks, the active change is much smaller and more reasonably sized. ---------------------------------------------------------------- Arnaud Pouliquen (1): ASoC: sti: fix possible sleep-in-atomic Bard liao (1): ASoC: SOF: Intel: lower print level to dbg if we will reinit DSP Dmitry Osipenko (1): ASoC: rt5640: Fix NULL dereference on module unload Hans de Goede (1): ASoC: Intel: bytcht_es8316: Fix Irbis NB41 netbook quirk Kai Vehmanen (2): ASoC: SOF: Intel: fix HDA codec driver probe with multiple controllers ASoC: hdac_hda: Fix error in driver removal after failed probe Olivier Moysan (2): ASoC: stm32: sai: fix possible circular locking ASoC: stm32: dfsdm: fix 16 bits record Stephan Gerhold (3): ASoC: msm8916-wcd-analog: Fix selected events for MIC BIAS External1 ASoC: msm8916-wcd-analog: Fix MIC BIAS Internal1 ASoC: msm8916-wcd-digital: Reset RX interpolation path after use Takashi Iwai (1): ASoC: Fix NULL dereference at freeing Yu-Hsuan Hsu (1): ASoC: cros_ec_codec: Make the device acpi compatible sound/soc/codecs/cros_ec_codec.c | 8 ++ sound/soc/codecs/hdac_hda.c | 4 +- sound/soc/codecs/msm8916-wcd-analog.c | 20 +++- sound/soc/codecs/msm8916-wcd-digital.c | 6 + sound/soc/codecs/rt5640.c | 7 ++ sound/soc/intel/boards/bytcht_es8316.c | 3 +- sound/soc/soc-component.c | 3 + sound/soc/sof/intel/hda-codec.c | 19 ++-- sound/soc/sof/intel/hda-loader.c | 6 +- sound/soc/sti/uniperif_player.c | 7 +- sound/soc/stm/stm32_adfsdm.c | 12 +- sound/soc/stm/stm32_sai_sub.c | 194 ++++++++++++++++++++++++--------- 12 files changed, 209 insertions(+), 80 deletions(-)

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[alsa-devel] [PATCH] dt-bindings: soundwire: fix example
by Srinivas Kandagatla 16 Jan '20

16 Jan '20

As wsa881x schema mentions #sound-dai-cells as required property, so update soundwire-controller.yaml example so that dt_bindings_check does not fail as below: Documentation/devicetree/bindings/soundwire/soundwire-controller.example.dt.yaml: speaker@0,1: '#sound-dai-cells' is a required property Documentation/devicetree/bindings/soundwire/soundwire-controller.example.dt.yaml: speaker@0,2: '#sound-dai-cells' is a required property Reported-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../devicetree/bindings/soundwire/soundwire-controller.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml b/Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml index 1b43993bccdb..330924b8618e 100644 --- a/Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml +++ b/Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml @@ -69,6 +69,7 @@ examples: reg = <0 1>; powerdown-gpios = <&wcdpinctrl 2 0>; #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; }; speaker@0,2 { @@ -76,6 +77,7 @@ examples: reg = <0 2>; powerdown-gpios = <&wcdpinctrl 2 0>; #thermal-sensor-cells = <0>; + #sound-dai-cells = <0>; }; }; -- 2.21.0

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[alsa-devel] [PATCH] soundwire: intel: report slave_ids for each link to SOF driver
by Pierre-Louis Bossart 16 Jan '20

16 Jan '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> The existing link_mask flag is no longer sufficient to detect the hardware and identify which topology file and a machine driver to load. By reporting the slave_ids exposed in ACPI tables, the parent SOF driver will be able to compare against a set of static configurations. This patch only adds the interface change, the functionality is added in future patches. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/linux/soundwire/sdw_intel.h | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 93b83bdf8035..979b41b5dcb4 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -5,6 +5,7 @@ #define __SDW_INTEL_H #include <linux/irqreturn.h> +#include <linux/soundwire/sdw.h> /** * struct sdw_intel_stream_params_data: configuration passed during @@ -93,6 +94,11 @@ struct sdw_intel_link_res; */ #define SDW_INTEL_CLK_STOP_BUS_RESET BIT(3) +struct sdw_intel_slave_id { + int link_id; + struct sdw_slave_id id; +}; + /** * struct sdw_intel_ctx - context allocated by the controller * driver probe @@ -101,9 +107,12 @@ struct sdw_intel_link_res; * hardware capabilities after all power dependencies are settled. * @link_mask: bit-wise mask listing SoundWire links reported by the * Controller + * @num_slaves: total number of devices exposed across all enabled links * @handle: ACPI parent handle * @links: information for each link (controller-specific and kept * opaque here) + * @ids: array of slave_id, representing Slaves exposed across all enabled + * links * @link_list: list to handle interrupts across all links * @shim_lock: mutex to handle concurrent rmw access to shared SHIM registers. */ @@ -111,8 +120,10 @@ struct sdw_intel_ctx { int count; void __iomem *mmio_base; u32 link_mask; + int num_slaves; acpi_handle handle; struct sdw_intel_link_res *links; + struct sdw_intel_slave_id *ids; struct list_head link_list; struct mutex shim_lock; /* lock for access to shared SHIM registers */ }; -- 2.20.1

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[alsa-devel] [PATCH V2 1/2] regulator: core: Add regulator_is_equal() helper
by Marek Vasut 16 Jan '20

16 Jan '20

Add regulator_is_equal() helper to compare whether two regulators are the same. This is useful for checking whether two separate regulators in a driver are actually the same supply. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Igor Opaniuk <igor.opaniuk(a)toradex.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> --- V2: New patch --- drivers/regulator/helpers.c | 14 ++++++++++++++ include/linux/regulator/consumer.h | 7 +++++++ 2 files changed, 21 insertions(+) diff --git a/drivers/regulator/helpers.c b/drivers/regulator/helpers.c index ca3dc3f3bb292..bb16c465426ef 100644 --- a/drivers/regulator/helpers.c +++ b/drivers/regulator/helpers.c @@ -13,6 +13,8 @@ #include <linux/regulator/driver.h> #include <linux/module.h> +#include "internal.h" + /** * regulator_is_enabled_regmap - standard is_enabled() for regmap users * @@ -881,3 +883,15 @@ void regulator_bulk_set_supply_names(struct regulator_bulk_data *consumers, consumers[i].supply = supply_names[i]; } EXPORT_SYMBOL_GPL(regulator_bulk_set_supply_names); + +/** + * regulator_is_equal - test whether two regulators are the same + * + * @reg1: first regulator to operate on + * @reg2: second regulator to operate on + */ +bool regulator_is_equal(struct regulator *reg1, struct regulator *reg2) +{ + return reg1->rdev == reg2->rdev; +} +EXPORT_SYMBOL_GPL(regulator_is_equal); diff --git a/include/linux/regulator/consumer.h b/include/linux/regulator/consumer.h index 337a463915278..2c89d886595cb 100644 --- a/include/linux/regulator/consumer.h +++ b/include/linux/regulator/consumer.h @@ -287,6 +287,8 @@ void regulator_bulk_set_supply_names(struct regulator_bulk_data *consumers, const char *const *supply_names, unsigned int num_supplies); +bool regulator_is_equal(struct regulator *reg1, struct regulator *reg2); + #else /* @@ -593,6 +595,11 @@ regulator_bulk_set_supply_names(struct regulator_bulk_data *consumers, { } +static inline bool +regulator_is_equal(struct regulator *reg1, struct regulator *reg2); +{ + return false; +} #endif static inline int regulator_set_voltage_triplet(struct regulator *regulator, -- 2.24.1

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[alsa-devel] [PATCH] ALSA: usb-audio: add implicit fb quirk for MOTU M Series
by Alexander Tsoy 16 Jan '20

16 Jan '20

This fixes crackling sound during playback. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- MOTU is known for reusing Product IDs for different devices or different generations of the device (e.g. MicroBook I/II/IIc shares a single Product ID). This patch was only tested with M4 audio interface, but the same Product ID is also used by M2. Hope it will work for M2 as well. sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index a11c8150af58..4f1c00615d94 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + ep = 0x81; + ifnum = 2; + goto add_sync_ep_from_ifnum; case USB_ID(0x0582, 0x01d8): /* BOSS Katana */ /* BOSS Katana amplifiers do not need quirks */ return 0; -- 2.24.1

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[alsa-devel] Question about runtime PM for HDA codecs
by Sridharan, Ranjani 16 Jan '20

16 Jan '20

Hi Takashi, Can I please bother you with a question about how to handle the situation with runtime PM for HDA codec on one of the Intel CML-based platforms with an ALC285 codec. Basically, the problem we're facing is that when using the SOF driver, jack detection doesn't seem to work as intended because the codec gets suspended right after the headset is plugged in. After a bit of experimenting, what we found was that calling snd_hda_set_power_save() with a delay of 0 or something > 0, both help with fixing the problem. But, I have a basic question about this. What is the expectation for enabling the codec runtime PM? I see that the legacy driver calls snd_hda_set_power_save() based on the CONFIG_SND_HDA_POWER_SAVE_DEFAULT value. In the case of SOF, we do not explicitly set this value so it is 0 by default. Also, when the codec is registered, the runtime PM for the codec is enabled by default without checking if the power_save delay is set to 0 or not. And later when the snd_hda_set_power_save() is called from the legacy HDA driver probe, it sets the use of auto suspend and the delay to be used based on the config. Would it be correct to remove the pm_runtime_enable() call in snd_hda_codec_register() and let the codec runtime PM enabling be set with the call to snd_hda_set_power_save() for both the legacy HDA driver and the SOF driver? Thanks, Ranjani

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[alsa-devel] [PATCH] ALSA: seq: Fix racy access for queue timer in proc read
by Takashi Iwai 15 Jan '20

15 Jan '20

snd_seq_info_timer_read() reads the information of the timer assigned for each queue, but it's done in a racy way which may lead to UAF as spotted by syzkaller. This patch applies the missing q->timer_mutex lock while accessing the timer object as well as a slight code change to adapt the standard coding style. Reported-by: syzbot+2b2ef983f973e5c40943(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_timer.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 63dc7bdb622d..be59b59c9be4 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -471,15 +471,19 @@ void snd_seq_info_timer_read(struct snd_info_entry *entry, q = queueptr(idx); if (q == NULL) continue; - if ((tmr = q->timer) == NULL || - (ti = tmr->timeri) == NULL) { - queuefree(q); - continue; - } + mutex_lock(&q->timer_mutex); + tmr = q->timer; + if (!tmr) + goto unlock; + ti = tmr->timeri; + if (!ti) + goto unlock; snd_iprintf(buffer, "Timer for queue %i : %s\n", q->queue, ti->timer->name); resolution = snd_timer_resolution(ti) * tmr->ticks; snd_iprintf(buffer, " Period time : %lu.%09lu\n", resolution / 1000000000, resolution % 1000000000); snd_iprintf(buffer, " Skew : %u / %u\n", tmr->skew, tmr->skew_base); +unlock: + mutex_unlock(&q->timer_mutex); queuefree(q); } } -- 2.16.4

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[alsa-devel] [PATCH] ALSA: usb-audio: unlock on error in probe
by Dan Carpenter 15 Jan '20

15 Jan '20

We need to unlock before we returning on this error path. Fixes: 73ac9f5e5b43 ("ALSA: usb-audio: Add boot quirk for MOTU M Series") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2f582ac7cf78..827fb0bc8b56 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -602,7 +602,7 @@ static int usb_audio_probe(struct usb_interface *intf, if (! chip) { err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); if (err < 0) - return err; + goto __error; /* it's a fresh one. * now look for an empty slot and create a new card instance -- 2.11.0

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[alsa-devel] [PATCH] ASoC: dt-bindings: stm32: convert spdfirx to json-schema
by Olivier Moysan 15 Jan '20

15 Jan '20

Convert the STM32 SPDIFRX bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- .../bindings/sound/st,stm32-spdifrx.txt | 56 ------------- .../bindings/sound/st,stm32-spdifrx.yaml | 80 +++++++++++++++++++ 2 files changed, 80 insertions(+), 56 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt deleted file mode 100644 index 33826f2459fa..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt +++ /dev/null @@ -1,56 +0,0 @@ -STMicroelectronics STM32 S/PDIF receiver (SPDIFRX). - -The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with -IEC-60958 and IEC-61937. - -Required properties: - - compatible: should be "st,stm32h7-spdifrx" - - reg: cpu DAI IP base address and size - - clocks: must contain an entry for kclk (used as S/PDIF signal reference) - - clock-names: must contain "kclk" - - interrupts: cpu DAI interrupt line - - dmas: DMA specifiers for audio data DMA and iec control flow DMA - See STM32 DMA bindings, Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: two dmas have to be defined, "rx" and "rx-ctrl" - -Optional properties: - - resets: Reference to a reset controller asserting the SPDIFRX - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -spdifrx: spdifrx@40004000 { - compatible = "st,stm32h7-spdifrx"; - reg = <0x40004000 0x400>; - clocks = <&rcc SPDIFRX_CK>; - clock-names = "kclk"; - interrupts = <97>; - dmas = <&dmamux1 2 93 0x400 0x0>, - <&dmamux1 3 94 0x400 0x0>; - dma-names = "rx", "rx-ctrl"; - pinctrl-0 = <&spdifrx_pins>; - pinctrl-names = "default"; - - spdifrx_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - }; - }; -}; - -spdif_in: spdif-in { - compatible = "linux,spdif-dir"; - - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; - -soundcard { - compatible = "audio-graph-card"; - dais = <&spdifrx_port>; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml new file mode 100644 index 000000000000..ab8e9d74ac3c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-spdifrx.yaml @@ -0,0 +1,80 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-spdifrx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 S/PDIF receiver (SPDIFRX) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: | + The SPDIFRX peripheral, is designed to receive an S/PDIF flow compliant with + IEC-60958 and IEC-61937. + +properties: + compatible: + enum: + - st,stm32h7-spdifrx + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: kclk + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio data capture DMA + - description: IEC status bits capture DMA + minItems: 1 + maxItems: 2 + + dma-names: + items: + - const: rx + - const: rx-ctrl + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + spdifrx: spdifrx@40004000 { + compatible = "st,stm32h7-spdifrx"; + #sound-dai-cells = <0>; + reg = <0x40004000 0x400>; + clocks = <&rcc SPDIF_K>; + clock-names = "kclk"; + interrupts = <GIC_SPI 97 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 2 93 0x400 0x0>, + <&dmamux1 3 94 0x400 0x0>; + dma-names = "rx", "rx-ctrl"; + pinctrl-0 = <&spdifrx_pins>; + pinctrl-names = "default"; + }; + +... -- 2.17.1

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[alsa-devel] Simultaneous play/record on implicit feedback device causes 'endpoint in use' error
by Robert Giaraffa 15 Jan '20

15 Jan '20

My company (Silicon Labs) makes a CP2615 USB audio class 1.0 device which uses implicit feedback. I saw these two issues when testing with aplay/arecord/alsabat/alsaloop: * During playback-only operation, the implicit feedback (IN) endpoint was not active. * During play/record operation, the implicit feedback values were not acted on, i.e. the host always sent 192 bytes per OUT packet even if previous IN packets had been 188 or 196 bytes. This behavior was observed on these systems: RaspberryPi2: Raspbian 9 (stretch), kernel v4.14.72 x86-64: Ubuntu 18.04, kernel v4.15.0-36 I was able to get implicit feedback working properly in playback mode by adding a device-specific quirk to 'pcm.c' that is similar to existing quirks for implicit-feedback devices, e.g. M-Audio Fast Track Ultra. (Note: The following patch applies to kernel v4.14.72. In v4.18, the set_sync_ep_implicit_fb_quirk function was changed so that there is a single call to usb_ifnum_to_if(), rather than a call for each device-dependent quirk.) $ git diff pcm.c diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b9c9a19f..6fb5567b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -358,6 +358,21 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, alts = &iface->altsetting[1]; goto add_sync_ep; + case USB_ID(0x10C4, 0xEAC1): // Silicon Labs CP2615 + // Ensure CP2615 is configured in Async Mode + if (attr != USB_ENDPOINT_SYNC_ASYNC) + return 0; + + // In P16R16 async mode, AudioIN EP (IF:4 EP:0x83) used for implicit feedback + ep = 0x83; + iface = usb_ifnum_to_if(dev, 4); + + if (!iface || iface->num_altsetting == 0) + return -EINVAL; + + alts = &iface->altsetting[1]; + goto add_sync_ep; + } if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && After adding the patch, I verified that implicit feedback worked properly for playback, i.e. the OUT samples per packet tracked previous IN values. However, when attempting to perform simultaneous play and record via alsabat, alsaloop, or separate aplay/arecord, the system reports the 'endpoint already in use' error when attempting to set the format for the IN endpoint that is already in use (note that this dmesg log has customized printk messages): [Oct23 10:13] snd_usb_hw_params->set_format() [ +0.000014] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.001654] set_format()->usb_set_interface 3:1 [ +0.000006] Creating new playback data endpoint 03 [ +0.000004] Creating new capture data endpoint 83 <Begin Playback. SYNC endpoint also starts because feedback is implicit> [ +0.000027] snd_usb_pcm_prepare->set_format() [ +0.000005] set_format(): dev:b3856c00 alts:b67e8824 ifnum: 3 [ +0.000755] Setting params for EP 03 (type 0, 12 urbs), ret=0 [ +0.000017] Setting params for EP 83 (type 0, 12 urbs), ret=0 [ +0.000004] Starting DATA_EP 03 @b9224000 [ +0.000004] snd_usb_endpoint_implicit_feedback_sink (03) is TRUE [ +0.001495] Starting SYNC_EP 83 @b9222000 <After playing for 0.5s, start recording. Failure occurs because EP83 is already in use.> [ +0.493035] snd_usb_hw_params->set_format() [ +0.000016] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 [ +0.001549] set_format()->usb_set_interface 4:1 [ +0.000005] Re-using EP 83 in iface 4:1 [ +0.000028] snd_usb_pcm_prepare->set_format() [ +0.000004] set_format(): dev:b3856c00 alts:b67e88a4 ifnum: 4 *** [ +0.003196] *** snd_usb_endpoint_set_params(): EP #83: already in use [ +0.000003] configure_endpoint->endpoint_set_params: err=FFFFFFF0 I bought a used Fast Track Ultra device for comparison testing, and found that it behaved the same way as the CP2615 does with the implicit feedback quirk: in playback-only mode implicit feedback works properly, but play/record causes the 'EP already in use' error. Are there any suggestions on how I can resolve this issue? Any help would be greatly appreciated! Thanks, Bob Giaraffa

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