August 2023 - Alsa-devel - mailman.alsa-project.org

[RFC PATCH 0/7] sun4i-i2s: Support channel remapping
by John Watts 31 Aug '23

31 Aug '23

Hi there, A while back I put out this thread: How do I set up multiple codecs on one I2S - without TDM?" https://lore.kernel.org/all/ZMBRMnv0GQF4wyfQ@titan/ This is my specific use case and motivation for this patch series. I posted how I managed to configure the audio-graph-card2 to allow doing this in conjunction with some register hacking, but I now have an initial patch series to support this in the sun4i-i2s driver. Now I've tested this on the T113-s3 with 3 wm8782s and it works well, but I'm marking this as RFC as I think I'm out of my depth here and I'd like to know the direction I need to take this: I'm new to kernel development and I'm genuinely unsure what are best practices and which are bad practices design-wise. My main concerns are the following: First, I split up channel-dins and channel-slots. This is mainly because I implemented one first but both of them only make sense together. The registers themselves use a format of a byte per channel with the upper nibble being the din and the lower being the slot. Perhaps this is a better format to copy? Second, I use u8 arrays on the device tree. This is done so I can just read the array easily, but it also means I can't make this property contingent on a compatible string in the schema as $ref doesn't seem to be allowed there. Third, channel-slots is available on all sun4i-i2s controllers, but I only have it implemented on the R329 variant for now when there are multiple din pins. I could add support for this on older controllers but there's not really a use case for manual configuration as there's no DIN and I don't have hardware to test it on. Fourth, I don't limit the readable channels to the channels listed. Reading more channels than you have currently results in the controller assuming you want to use TDM, but that's not the case here and you can have strange duplicate channels show up. Fifth, it might be a good idea to increase the maximum channels from 8 to 16, especially if people are going to be running multiple TDM streams on one controller. Sixth, the channel-slots only apply to capture, not playback. This is something I just realized now when writing and forgot to document in the patch set. Thanks for your time, John. John Watts (7): ASoC: sunxi: sun4i-i2s: Prepare for runtime DIN pin selection ASoC: sunxi: sun4i-i2s: Use channel-dins device tree property ASoC: sunxi: sun4i-i2s: Prepare for runtime channel slot selection ASoC: sunxi: sun4i-i2s: Use channel-slots device tree property ASoC: sunxi: sun4i-i2s: Detect TDM slots based on channel slots dt-bindings: sound: sun4i-i2s: Add channel-dins property dt-bindings: sound: sun4i-i2s: Add channel-slots property .../sound/allwinner,sun4i-a10-i2s.yaml | 30 +++++ sound/soc/sunxi/sun4i-i2s.c | 106 +++++++++++++++++- 2 files changed, 132 insertions(+), 4 deletions(-) -- 2.41.0

3 10

linux-6.5/sound/soc/codecs/rt722-sdca.c:612: some confused logic ?
by David Binderman 31 Aug '23

31 Aug '23

Hello there, I just ran cppcheck over the source code of linux-6.5. It said: linux-6.5/sound/soc/codecs/rt722-sdca.c:612:8: style: Expression is always true because 'else if' condition is opposite to previous condition at line 609. [multiCondition] Source code is if (!adc_vol_flag) /* boost gain */ ctl = regvalue / boost_step; else { /* ADC gain */ if (adc_vol_flag) ctl = p->max - (((vol_max - regvalue) & 0xffff) / interval_offset); else ctl = p->max - (((0 - regvalue) & 0xffff) / interval_offset); } This code might benefit from some rework. Regards David Binderman

1 0

[PATCH V2 1/3] ASoC: dt-bindings: fsl_easrc: Add support for imx8mp-easrc
by Adam Ford 31 Aug '23

31 Aug '23

The i.MX8MP appears to have the same easrc support as the Nano, so add imx8mp as an option with a fallback to imx8mn. Signed-off-by: Adam Ford <aford173(a)gmail.com> --- V2: Fixed errors: ['fsl,imx8mn-easrc'] is too short 'fsl,imx8mn-easrc' is not one of ['fsl,imx8mp-easrc'] diff --git a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml index bdde68a1059c..a680d7aff237 100644 --- a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml @@ -14,7 +14,13 @@ properties: pattern: "^easrc@.*" compatible: - const: fsl,imx8mn-easrc + oneOf: + - enum: + - fsl,imx8mn-easrc + - items: + - enum: + - fsl,imx8mp-easrc + - const: fsl,imx8mn-easrc reg: maxItems: 1 -- 2.39.2

4 6

[PATCH 1/7] ASoC: cs35l45: Add support for Chip ID 0x35A460
by Vlad Karpovich 30 Aug '23

30 Aug '23

The 0x35A460 chip is a different variant of the cs35l45. Signed-off-by: Vlad Karpovich <vkarpovi(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 8f480df7f29a..7f116ae97acd 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -1078,6 +1078,7 @@ static int cs35l45_initialize(struct cs35l45_private *cs35l45) switch (dev_id[0]) { case 0x35A450: + case 0x35A460: break; default: dev_err(cs35l45->dev, "Bad DEVID 0x%x\n", dev_id[0]); -- 2.25.1

3 16

[PATCH] ASoC: cs35l56: Waiting for firmware to boot must be tolerant of I/O errors
by Richard Fitzgerald 29 Aug '23

29 Aug '23

From: Simon Trimmer <simont(a)opensource.cirrus.com> Ignore failure to read from the cs35l56 when polling as the device will NAK i2c accesses until it has booted and this would terminate the poll of regmap_read_poll_timeout(). Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l56-shared.c | 27 ++++++++++++++------------- 1 file changed, 14 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index ae373f335ea8..98b1e63360ae 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -243,26 +243,27 @@ int cs35l56_wait_for_firmware_boot(struct cs35l56_base *cs35l56_base) { unsigned int reg; unsigned int val; - int ret; + int read_ret, poll_ret; if (cs35l56_base->rev < CS35L56_REVID_B0) reg = CS35L56_DSP1_HALO_STATE_A1; else reg = CS35L56_DSP1_HALO_STATE; - ret = regmap_read_poll_timeout(cs35l56_base->regmap, reg, - val, - (val < 0xFFFF) && (val >= CS35L56_HALO_STATE_BOOT_DONE), - CS35L56_HALO_STATE_POLL_US, - CS35L56_HALO_STATE_TIMEOUT_US); + /* + * This can't be a regmap_read_poll_timeout() because cs35l56 will NAK + * I2C until it has booted which would terminate the poll + */ + poll_ret = read_poll_timeout(regmap_read, read_ret, + (val < 0xFFFF) && (val >= CS35L56_HALO_STATE_BOOT_DONE), + CS35L56_HALO_STATE_POLL_US, + CS35L56_HALO_STATE_TIMEOUT_US, + false, + cs35l56_base->regmap, reg, &val); - if ((ret < 0) && (ret != -ETIMEDOUT)) { - dev_err(cs35l56_base->dev, "Failed to read HALO_STATE: %d\n", ret); - return ret; - } - - if ((ret == -ETIMEDOUT) || (val != CS35L56_HALO_STATE_BOOT_DONE)) { - dev_err(cs35l56_base->dev, "Firmware boot fail: HALO_STATE=%#x\n", val); + if (poll_ret) { + dev_err(cs35l56_base->dev, "Firmware boot timed out(%d): HALO_STATE=%#x\n", + read_ret, val); return -EIO; } -- 2.30.2

2 1

[PATCH v2 1/1] ALSA: control: Use list_for_each_entry_safe()
by Andy Shevchenko 29 Aug '23

29 Aug '23

Instead of reiterating the list, use list_for_each_entry_safe() that allows to continue without starting over. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> --- v2: fixed a typo (Jaroslav), added tag (Jaroslav) sound/core/control_led.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index a78eb48927c7..43ea995c095a 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -297,16 +297,13 @@ static void snd_ctl_led_clean(struct snd_card *card) { unsigned int group; struct snd_ctl_led *led; - struct snd_ctl_led_ctl *lctl; + struct snd_ctl_led_ctl *lctl, _lctl; for (group = 0; group < MAX_LED; group++) { led = &snd_ctl_leds[group]; -repeat: - list_for_each_entry(lctl, &led->controls, list) - if (!card || lctl->card == card) { + list_for_each_entry_safe(lctl, _lctl, &led->controls, list) + if (!card || lctl->card == card) snd_ctl_led_ctl_destroy(lctl); - goto repeat; - } } } @@ -314,7 +311,7 @@ static int snd_ctl_led_reset(int card_number, unsigned int group) { struct snd_card *card; struct snd_ctl_led *led; - struct snd_ctl_led_ctl *lctl; + struct snd_ctl_led_ctl *lctl, _lctl; struct snd_kcontrol_volatile *vd; bool change = false; @@ -329,14 +326,12 @@ static int snd_ctl_led_reset(int card_number, unsigned int group) return -ENXIO; } led = &snd_ctl_leds[group]; -repeat: - list_for_each_entry(lctl, &led->controls, list) + list_for_each_entry_safe(lctl, _lctl, &led->controls, list) if (lctl->card == card) { vd = &lctl->kctl->vd[lctl->index_offset]; vd->access &= ~group_to_access(group); snd_ctl_led_ctl_destroy(lctl); change = true; - goto repeat; } mutex_unlock(&snd_ctl_led_mutex); if (change) -- 2.40.0.1.gaa8946217a0b

2 2

[PATCH next] ASoC: cs42l43: Fix missing error code in cs42l43_codec_probe()
by Harshit Mogalapalli 29 Aug '23

29 Aug '23

When clk_get_optional() fails, the error handling code does a 'goto err_pm' with ret = 0, which is resturning success on a failure path. Fix this by assigning the PTR_ERR(priv->mclk) to ret variable. Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43") Signed-off-by: Harshit Mogalapalli <harshit.m.mogalapalli(a)oracle.com> --- This is based on static analysis with smatch, only compile tested. --- sound/soc/codecs/cs42l43.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 24e718e51174..1a95c370fc4c 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2205,7 +2205,8 @@ static int cs42l43_codec_probe(struct platform_device *pdev) // Don't use devm as we need to get against the MFD device priv->mclk = clk_get_optional(cs42l43->dev, "mclk"); if (IS_ERR(priv->mclk)) { - dev_err_probe(priv->dev, PTR_ERR(priv->mclk), "Failed to get mclk\n"); + ret = PTR_ERR(priv->mclk); + dev_err_probe(priv->dev, ret, "Failed to get mclk\n"); goto err_pm; } -- 2.39.3

3 2

[PATCH] ASoC: codecs: wcd93xx: fix object added to multiple drivers
by Krzysztof Kozlowski 29 Aug '23

29 Aug '23

Three Qualcomm audio codecs (WCD9355, WCD934x and WCD938x) use the same object file wcd-clsh-v2.o leading to warnings: Makefile: wcd-clsh-v2.o is added to multiple modules: snd-soc-wcd9335 snd-soc-wcd934x snd-soc-wcd938x Convert the wcd-clsh-v2.o to a module to solve it. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/Kconfig | 6 ++++++ sound/soc/codecs/Makefile | 8 +++++--- sound/soc/codecs/wcd-clsh-v2.c | 8 ++++++++ 3 files changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 95b5bd883215..f1e1dbc509f6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1968,11 +1968,15 @@ config SND_SOC_UDA1380 tristate depends on I2C +config SND_SOC_WCD_CLASSH + tristate + config SND_SOC_WCD9335 tristate "WCD9335 Codec" depends on SLIMBUS select REGMAP_SLIMBUS select REGMAP_IRQ + select SND_SOC_WCD_CLASSH help The WCD9335 is a standalone Hi-Fi audio CODEC IC, supports Qualcomm Technologies, Inc. (QTI) multimedia solutions, @@ -1987,6 +1991,7 @@ config SND_SOC_WCD934X depends on SLIMBUS select REGMAP_IRQ select REGMAP_SLIMBUS + select SND_SOC_WCD_CLASSH select SND_SOC_WCD_MBHC depends on MFD_WCD934X || COMPILE_TEST help @@ -1997,6 +2002,7 @@ config SND_SOC_WCD938X depends on SND_SOC_WCD938X_SDW tristate depends on SOUNDWIRE || !SOUNDWIRE + select SND_SOC_WCD_CLASSH config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c8502a49b40a..a87e56938ce5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -303,10 +303,11 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o snd-soc-uda1334-objs := uda1334.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wcd-classh-objs := wcd-clsh-v2.o snd-soc-wcd-mbhc-objs := wcd-mbhc-v2.o -snd-soc-wcd9335-objs := wcd-clsh-v2.o wcd9335.o -snd-soc-wcd934x-objs := wcd-clsh-v2.o wcd934x.o -snd-soc-wcd938x-objs := wcd938x.o wcd-clsh-v2.o +snd-soc-wcd9335-objs := wcd9335.o +snd-soc-wcd934x-objs := wcd934x.o +snd-soc-wcd938x-objs := wcd938x.o snd-soc-wcd938x-sdw-objs := wcd938x-sdw.o snd-soc-wl1273-objs := wl1273.o snd-soc-wm-adsp-objs := wm_adsp.o @@ -685,6 +686,7 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA1334) += snd-soc-uda1334.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WCD_CLASSH) += snd-soc-wcd-classh.o obj-$(CONFIG_SND_SOC_WCD_MBHC) += snd-soc-wcd-mbhc.o obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o obj-$(CONFIG_SND_SOC_WCD934X) += snd-soc-wcd934x.o diff --git a/sound/soc/codecs/wcd-clsh-v2.c b/sound/soc/codecs/wcd-clsh-v2.c index a75db27e5205..d96e23ec43d4 100644 --- a/sound/soc/codecs/wcd-clsh-v2.c +++ b/sound/soc/codecs/wcd-clsh-v2.c @@ -355,6 +355,7 @@ void wcd_clsh_set_hph_mode(struct wcd_clsh_ctrl *ctrl, int mode) wcd_clsh_v2_set_hph_mode(comp, mode); } +EXPORT_SYMBOL_GPL(wcd_clsh_set_hph_mode); static void wcd_clsh_set_flyback_current(struct snd_soc_component *comp, int mode) @@ -869,11 +870,13 @@ int wcd_clsh_ctrl_set_state(struct wcd_clsh_ctrl *ctrl, return 0; } +EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_set_state); int wcd_clsh_ctrl_get_state(struct wcd_clsh_ctrl *ctrl) { return ctrl->state; } +EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_get_state); struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp, int version) @@ -890,8 +893,13 @@ struct wcd_clsh_ctrl *wcd_clsh_ctrl_alloc(struct snd_soc_component *comp, return ctrl; } +EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_alloc); void wcd_clsh_ctrl_free(struct wcd_clsh_ctrl *ctrl) { kfree(ctrl); } +EXPORT_SYMBOL_GPL(wcd_clsh_ctrl_free); + +MODULE_DESCRIPTION("WCD93XX Class-H driver"); +MODULE_LICENSE("GPL"); -- 2.34.1

2 1

Front 3.5mm panel not recognized (MSI X370 Gaming Pro & MSI B450-A Pro)
by GitHub issues - opened 29 Aug '23

29 Aug '23

alsa-project/alsa-ucm-conf issue #344 was opened from fiftydinar: Realtek ALC892 soundcard on both. alsa-info.sh (MSI X370 Gaming Pro) http://alsa-project.org/db/?f=26d6d5ef7701fabf260fd75e0d2bd135976019dd alsa.info.sh (MSI B450-A Pro) http://alsa-project.org/db/?f=621f405abe6c94f4a3069caba7ab7cb802d03563 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/344 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1 1/1] ALSA: control: Use list_for_each_entry_safe()
by Andy Shevchenko 29 Aug '23

29 Aug '23

Instead of reiterating the list, use list_for_each_entry_safe() that allows to continue without starting over. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- Takashi, if you have anybody or want yourself to spend some time, I believe you can simplify a lot the parser in this file with the help of lib/cmdline.c APIs. sound/core/control_led.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index a78eb48927c7..afc9ffc388e3 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -297,16 +297,13 @@ static void snd_ctl_led_clean(struct snd_card *card) { unsigned int group; struct snd_ctl_led *led; - struct snd_ctl_led_ctl *lctl; + struct snd_ctl_led_ctl *lctl, _lctl; for (group = 0; group < MAX_LED; group++) { led = &snd_ctl_leds[group]; -repeat: - list_for_each_entry(lctl, &led->controls, list) - if (!card || lctl->card == card) { + list_for_each_entry_safe(lctl, _lctl, &led->controls, list) + if (!card || lctl->card == card) snd_ctl_led_ctl_destroy(lctl); - goto repeat; - } } } @@ -314,7 +311,7 @@ static int snd_ctl_led_reset(int card_number, unsigned int group) { struct snd_card *card; struct snd_ctl_led *led; - struct snd_ctl_led_ctl *lctl; + struct snd_ctl_led_ctl *lctl, _lctl; struct snd_kcontrol_volatile *vd; bool change = false; @@ -329,14 +326,12 @@ static int snd_ctl_led_reset(int card_number, unsigned int group) return -ENXIO; } led = &snd_ctl_leds[group]; -repeat: - list_for_each_entry(lctl, &led->controls, list) + list_for_each_entry(lctl, _lctl, &led->controls, list) if (lctl->card == card) { vd = &lctl->kctl->vd[lctl->index_offset]; vd->access &= ~group_to_access(group); snd_ctl_led_ctl_destroy(lctl); change = true; - goto repeat; } mutex_unlock(&snd_ctl_led_mutex); if (change) -- 2.40.0.1.gaa8946217a0b

4 5

[PATCH v2] mfd: madera-i2c: Simplify probe()
by Biju Das 29 Aug '23

29 Aug '23

Simplify probe() by replacing of_device_get_match_data() and ID lookup for retrieving match data by i2c_get_match_data(). Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Note: This patch is only compile tested. v1->v2: * Added Ack tag from Charles Keepax. * Dropped blank line before switch statement. --- drivers/mfd/madera-i2c.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/drivers/mfd/madera-i2c.c b/drivers/mfd/madera-i2c.c index a404ea26bc79..33781c631345 100644 --- a/drivers/mfd/madera-i2c.c +++ b/drivers/mfd/madera-i2c.c @@ -18,21 +18,14 @@ static int madera_i2c_probe(struct i2c_client *i2c) { - const struct i2c_device_id *id = i2c_client_get_device_id(i2c); struct madera *madera; const struct regmap_config *regmap_16bit_config = NULL; const struct regmap_config *regmap_32bit_config = NULL; - const void *of_data; unsigned long type; const char *name; int ret; - of_data = of_device_get_match_data(&i2c->dev); - if (of_data) - type = (unsigned long)of_data; - else - type = id->driver_data; - + type = (unsigned long)i2c_get_match_data(i2c); switch (type) { case CS47L15: if (IS_ENABLED(CONFIG_MFD_CS47L15)) { -- 2.25.1

2 2

[PATCH v4 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 29 Aug '23

29 Aug '23

Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (3): xhci: add support to allocate several interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (29): usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: Add USB backend ASoC driver for Q6 sound: usb: card: Introduce USB SND platform op callbacks sound: usb: Export USB SND APIs for modules dt-bindings: usb: dwc3: Add snps,num-hc-interrupters definition usb: dwc3: Add DT parameter to specify maximum number of interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present sound: usb: pcm: Export fixed rate check USB SND API sound: usb: qcom: Add USB QMI definitions sound: usb: Introduce QC USB SND offloading support sound: usb: card: Check for support for requested audio format sound: soc: soc-usb: Add PCM format check API for USB backend sound: soc: qcom: qusb6: Ensure PCM format is supported by USB audio device sound: usb: Prevent starting of audio stream if in use ASoC: dt-bindings: Add Q6USB backend bindings ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: q6afe: Split USB AFE dev_token param into separate API sound: Pass USB SND card and PCM information to SOC USB sound: soc: qdsp6: Add SND kcontrol to select offload device sound: soc: qdsp6: Add SND kcontrol for fetching offload status sound: soc: qcom: q6usb: Add headphone jack for offload connection status sound: usb: qc_audio_offload: Use card and PCM index from QMI request sound: usb: card: Allow for rediscovery of connected USB SND devices sound: soc: soc-usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,q6usb-dais.yaml | 49 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 13 + arch/arm64/boot/dts/qcom/sm8350.dtsi.rej | 14 + arch/arm64/configs/defconfig.rej | 24 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 4 + drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-hub.c | 29 +- drivers/usb/host/xhci-mem.c | 167 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 2 +- drivers/usb/host/xhci-sideband.c | 195 ++ drivers/usb/host/xhci.c | 109 +- drivers/usb/host/xhci.h | 102 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 86 + include/linux/usb/xhci-sideband.h | 90 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 39 + sound/core/oss/pcm_oss.c | 4 +- sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 50 + sound/soc/qcom/qdsp6/q6afe.c | 208 +- sound/soc/qcom/qdsp6/q6afe.h | 47 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 458 +++++ sound/soc/soc-usb.c | 197 ++ sound/usb/Kconfig | 15 + sound/usb/Makefile | 2 +- sound/usb/card.c | 85 + sound/usb/card.h | 32 + sound/usb/endpoint.c | 2 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 29 +- sound/usb/pcm.h | 12 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1820 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 47 files changed, 4870 insertions(+), 179 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb-dais.yaml create mode 100644 arch/arm64/boot/dts/qcom/sm8350.dtsi.rej create mode 100644 arch/arm64/configs/defconfig.rej create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-intr.h create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

10 68

[PATCH] ALSA: pcm: Fix missing fixup call in compat hw_refine ioctl
by Takashi Iwai 29 Aug '23

29 Aug '23

The update of rate_num/den and msbits were factored out to fixup_unreferenced_params() function to be called explicitly after the hw_refine or hw_params procedure. It's called from snd_pcm_hw_refine_user(), but it's forgotten in the PCM compat ioctl. This ended up with the incomplete rate_num/den and msbits parameters when 32bit compat ioctl is used. This patch adds the missing call in snd_pcm_ioctl_hw_params_compat(). Reported-by: Meng_Cai(a)novatek.com.cn Fixes: f9a076bff053 ("ALSA: pcm: calculate non-mask/non-interval parameters always when possible") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_compat.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 42c2ada8e888..c96483091f30 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -253,10 +253,14 @@ static int snd_pcm_ioctl_hw_params_compat(struct snd_pcm_substream *substream, goto error; } - if (refine) + if (refine) { err = snd_pcm_hw_refine(substream, data); - else + if (err < 0) + goto error; + err = fixup_unreferenced_params(substream, data); + } else { err = snd_pcm_hw_params(substream, data); + } if (err < 0) goto error; if (copy_to_user(data32, data, sizeof(*data32)) || -- 2.35.3

3 2

[PATCH AUTOSEL 4.14 5/5] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 29 Aug '23

29 Aug '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index cc8f06638edca..7226c03f15934 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -50,6 +50,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -184,6 +185,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -332,14 +334,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -349,14 +353,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -381,13 +385,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -401,10 +409,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -423,6 +430,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.40.1

1 0

[PATCH AUTOSEL 4.19 5/5] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 29 Aug '23

29 Aug '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 838c3c8b403cb..2ddfd6fed122e 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -50,6 +50,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -184,6 +185,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -332,14 +334,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -349,14 +353,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -381,13 +385,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -401,10 +409,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -423,6 +430,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.40.1

1 0

[PATCH AUTOSEL 5.4 6/6] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 29 Aug '23

29 Aug '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index f73ee0798aeab..be80ce72e0c72 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -171,6 +172,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -319,14 +321,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -336,14 +340,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -368,13 +372,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -388,10 +396,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -410,6 +417,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.40.1

1 0

[PATCH AUTOSEL 5.10 7/7] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 29 Aug '23

29 Aug '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index f73ee0798aeab..be80ce72e0c72 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -171,6 +172,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strlcpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -319,14 +321,16 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -336,14 +340,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -368,13 +372,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -388,10 +396,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) if ((mdev = get_mididev(dp, dev)) == NULL) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -410,6 +417,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.40.1

1 0

[PATCH AUTOSEL 5.15 9/9] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 29 Aug '23

29 Aug '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 07efb38f58ac1..f2940b29595f0 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -172,6 +173,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strscpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -322,15 +324,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -340,14 +344,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -372,13 +376,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -393,10 +401,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -415,6 +422,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.40.1

1 0

[PATCH AUTOSEL 6.1 15/15] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Sasha Levin 29 Aug '23

29 Aug '23

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 297224fc0922e7385573a30c29ffdabb67f27b7d ] Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Link: https://lore.kernel.org/r/20230612125533.27461-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 07efb38f58ac1..f2940b29595f0 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -172,6 +173,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strscpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -322,15 +324,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -340,14 +344,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -372,13 +376,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -393,10 +401,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -415,6 +422,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 12/17] ASoC: rt1308-sdw: fix random louder sound
by Sasha Levin 29 Aug '23

29 Aug '23

From: Shuming Fan <shumingf(a)realtek.com> [ Upstream commit 37aba3190891d4de189bd5192ee95220e295f34d ] This patch uses a vendor register to check whether the system hibernated ever. The driver will only set the preset when the driver brings up or the system hibernated. It will avoid the unknown issue that makes the speaker output louder and can't control the volume. Signed-off-by: Shuming Fan <shumingf(a)realtek.com Link: https://lore.kernel.org/r/20230811093822.37573-1-shumingf@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt1308-sdw.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 1797af824f60b..e2699c0b117be 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -52,6 +52,7 @@ static bool rt1308_volatile_register(struct device *dev, unsigned int reg) case 0x300a: case 0xc000: case 0xc710: + case 0xcf01: case 0xc860 ... 0xc863: case 0xc870 ... 0xc873: return true; @@ -213,7 +214,7 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) { struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(dev); int ret = 0; - unsigned int tmp; + unsigned int tmp, hibernation_flag; if (rt1308->hw_init) return 0; @@ -242,6 +243,10 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) pm_runtime_get_noresume(&slave->dev); + regmap_read(rt1308->regmap, 0xcf01, &hibernation_flag); + if ((hibernation_flag != 0x00) && rt1308->first_hw_init) + goto _preset_ready_; + /* sw reset */ regmap_write(rt1308->regmap, RT1308_SDW_RESET, 0); @@ -282,6 +287,12 @@ static int rt1308_io_init(struct device *dev, struct sdw_slave *slave) regmap_write(rt1308->regmap, 0xc100, 0xd7); regmap_write(rt1308->regmap, 0xc101, 0xd7); + /* apply BQ params */ + rt1308_apply_bq_params(rt1308); + + regmap_write(rt1308->regmap, 0xcf01, 0x01); + +_preset_ready_: if (rt1308->first_hw_init) { regcache_cache_bypass(rt1308->regmap, false); regcache_mark_dirty(rt1308->regmap); -- 2.40.1

1 0

[PATCH AUTOSEL 6.4 10/17] ALSA: hda/cs8409: Support new Dell Dolphin Variants
by Sasha Levin 29 Aug '23

29 Aug '23

From: Stefan Binding <sbinding(a)opensource.cirrus.com> [ Upstream commit 7c761166399bedfc89c928bef8015546d85a9099 ] Add 4 new Dell Dolphin Systems, same configuration as older systems. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20230811123044.1045651-1-sbinding@opensource.cirr… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_cs8409-tables.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index b288874e401e5..36b411d1a9609 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -550,6 +550,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C73, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C75, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C7D, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C7F, "Dolphin", CS8409_DOLPHIN), {} /* terminator */ }; -- 2.40.1

1 0

[PATCH v2 0/4] ASoC: amd: acp: Add sound support for a line of HUAWEI laptops
by Marian Postevca 29 Aug '23

29 Aug '23

This series adds support for a line of HUAWEI laptops with AMD CPUs that connect using the ACP3x module to a ES8336 CODEC. The CODEC driver must be extended to support the S32 LE format and the MCLK div by 2 option. MCLK div by 2 is needed for one specific SKU, which uses a 48Mhz MCLK, which seems to be too high of a frequency for the CODEC and must be divided by 2. The acp legacy driver must also be extended by using callbacks so that the more complicated handling of this specific CODEC can be moved outside the more generic ACP code. Changes in v2: - Removed patch 4: "ASoC: amd: acp: Improve support for speaker power events". May be resubmitted separately. - Split the first commit that enabled support for S32 LE format and the MCLK div by 2 option into two separate commits. - Removed the MCLK div by 2 DT property that was previously enabled by the machine driver. Now it's enabled by the CODEC driver if the MCLK frequency is equal or greater than 48Mhz. - Used normal conditional statements. - Removed constraint rates from the machine driver, the CODEC should be able to set them. - Moved the DAI format configuration. - Uncoupled the speaker and headphone GPIOs. Now they can be handled independently Marian Postevca (4): ASoC: es8316: Enable support for S32 LE format ASoC: es8316: Enable support for MCLK div by 2 ASoC: amd: acp: Add support for splitting the codec specific code from the ACP driver ASoC: amd: acp: Add machine driver that enables sound for systems with a ES8336 codec sound/soc/amd/acp-config.c | 70 +++ sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-legacy-mach.c | 102 +++- sound/soc/amd/acp/acp-mach-common.c | 8 + sound/soc/amd/acp/acp-mach.h | 67 +++ sound/soc/amd/acp/acp-renoir.c | 4 + sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 449 ++++++++++++++++++ sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h | 12 + sound/soc/codecs/es8316.c | 22 +- sound/soc/codecs/es8316.h | 3 + 10 files changed, 724 insertions(+), 15 deletions(-) create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h -- 2.41.0

2 13

[PATCH v3 00/28] Introduce QC USB SND audio offloading support
by Wesley Cheng 29 Aug '23

29 Aug '23

Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Mathias Nyman (1): xhci: Add support to allocate several interrupters Wesley Cheng (27): usb: xhci: Add XHCI APIs to support USB offloading usb: host: xhci-mem: Cleanup pending secondary event ring events ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: Add USB backend ASoC driver for Q6 sound: usb: card: Introduce USB SND platform op callbacks sound: usb: Export USB SND APIs for modules dt-bindings: usb: dwc3: Add snps,num-hc-interrupters definition usb: dwc3: Add DT parameter to specify maximum number of interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present sound: usb: pcm: Export fixed rate check USB SND API sound: usb: Introduce QC USB SND offloading support sound: usb: card: Check for support for requested audio format sound: soc: soc-usb: Add PCM format check API for USB backend sound: soc: qcom: qusb6: Ensure PCM format is supported by USB audio device sound: usb: Prevent starting of audio stream if in use ASoC: dt-bindings: Add Q6USB backend bindings ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: q6afe: Split USB AFE dev_token param into separate API sound: Pass USB SND card and PCM information to SOC USB sound: soc: qdsp6: Add SND kcontrol to select offload device sound: soc: qdsp6: Add SND kcontrol for fetching offload status sound: usb: qc_audio_offload: Use card and PCM index from QMI request sound: usb: card: Allow for rediscovery of connected USB SND devices sound: soc: soc-usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,q6usb-dais.yaml | 49 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 13 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-mem.c | 171 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 2 +- drivers/usb/host/xhci.c | 181 +- drivers/usb/host/xhci.h | 2 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 94 + include/sound/pcm_params.h | 4 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 39 + sound/core/oss/pcm_oss.c | 58 - sound/core/pcm_lib.c | 65 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 50 + sound/soc/qcom/qdsp6/q6afe.c | 208 +- sound/soc/qcom/qdsp6/q6afe.h | 47 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 445 ++++ sound/soc/soc-usb.c | 197 ++ sound/usb/Kconfig | 14 + sound/usb/Makefile | 2 +- sound/usb/card.c | 85 + sound/usb/card.h | 31 + sound/usb/endpoint.c | 2 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 29 +- sound/usb/pcm.h | 12 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1790 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 42 files changed, 4646 insertions(+), 100 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb-dais.yaml create mode 100644 include/linux/usb/xhci-intr.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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[PATCH v1] ASoC: dwc: Fix the warnnings about JH7110 ops
by Xingyu Wu 29 Aug '23

29 Aug '23

Add a check of CONFIG_OF before the JH7110 ops to fix the warnings that the JH7110 ops are defined but not used when CONFIG_OF is disabled. Fixes: 1a512d13837a ("Add I2S support for the StarFive JH7110 SoC") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202308250433.nhzrhYEo-lkp@intel.com/ Signed-off-by: Xingyu Wu <xingyu.wu(a)starfivetech.com> --- Hi Mark, This patch fixes the issue about the warning of JH7110 ops when CONFIG_OF is disabled. So add a check of CONFIG_OF before defining JH7110 ops. Thanks, Xingyu Wu --- sound/soc/dwc/dwc-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 5ab1b3eb2d28..329015ef1d1e 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -729,6 +729,7 @@ static int dw_configure_dai_by_dt(struct dw_i2s_dev *dev, } +#ifdef CONFIG_OF /* clocks initialization with master mode on JH7110 SoC */ static int jh7110_i2s_crg_master_init(struct dw_i2s_dev *dev) { @@ -912,6 +913,7 @@ static int jh7110_i2stx0_clk_cfg(struct i2s_clk_config_data *config) return clk_set_rate(dev->clk, bclk_rate); } +#endif static int dw_i2s_probe(struct platform_device *pdev) { -- 2.25.1

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[GIT PULL] ASoC fixes for v6.5-merge-window
by Mark Brown 28 Aug '23

28 Aug '23

The following changes since commit 1f1ef7e5bbe2de125da413e238915f6047ea4ba2: ASoC: intel: sof_sdw: Fixup typo in device link checking (2023-06-19 12:58:22 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-merge-window for you to fetch changes up to d900d9a435ca95a386f49424f3689cd17ec201da: ASoC: tegra: Fix SFC conversion for few rates (2023-06-24 01:57:16 +0100) ---------------------------------------------------------------- ASoC: Fixes that got left after v6.4 These were some changes in my v6.4 branch that never got sent as fixes, none of them super urgent thankfully. ---------------------------------------------------------------- Phil Elwell (1): ASoC: cs43130: Fix numerator/denominator mixup Sheetal (1): ASoC: tegra: Fix SFC conversion for few rates yixuanjiang (1): ASoC: soc-compress: Fix deadlock in soc_compr_open_fe sound/soc/codecs/cs43130.h | 138 ++++++++++++++++++++--------------------- sound/soc/soc-compress.c | 1 + sound/soc/tegra/tegra210_sfc.c | 31 ++++++++- sound/soc/tegra/tegra210_sfc.h | 4 +- 4 files changed, 102 insertions(+), 72 deletions(-)

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