June 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH 05/14] ALSA: usb-audio: scarlett2: Remove interrupt debug message
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Just ignore instead of printing an error if the interrupt data is not the expected length. This check was for development and the condition has not been observed. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index a5f5c537e344..3225ec709d98 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -2018,19 +2018,14 @@ static void scarlett2_mixer_interrupt(struct urb *urb) int ustatus = urb->status; u32 data; - if (ustatus != 0) + if (ustatus != 0 || len != 8) goto requeue; - if (len == 8) { - data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); - if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) - scarlett2_mixer_interrupt_vol_change(mixer); - if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) - scarlett2_mixer_interrupt_button_change(mixer); - } else { - usb_audio_err(mixer->chip, - "scarlett mixer interrupt length %d\n", len); - } + data = le32_to_cpu(*(__le32 *)urb->transfer_buffer); + if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE) + scarlett2_mixer_interrupt_vol_change(mixer); + if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE) + scarlett2_mixer_interrupt_button_change(mixer); requeue: if (ustatus != -ENOENT && -- 2.31.1

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[PATCH 04/14] ALSA: usb-audio: scarlett2: Remove unused/useless code
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Remove #define not used. Remove useless assignments and copies. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 26 +++++--------------------- 1 file changed, 5 insertions(+), 21 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 8026847d27b6..a5f5c537e344 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -271,7 +271,6 @@ static const struct scarlett2_device_info s6i6_gen2_info = { .id = 0x000, .num = { 1, 0, 8, 8, 8 }, .src_descr = "Off", - .src_num_offset = 0, }, [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, @@ -327,7 +326,6 @@ static const struct scarlett2_device_info s18i8_gen2_info = { .id = 0x000, .num = { 1, 0, 8, 8, 4 }, .src_descr = "Off", - .src_num_offset = 0, }, [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, @@ -393,7 +391,6 @@ static const struct scarlett2_device_info s18i20_gen2_info = { .id = 0x000, .num = { 1, 0, 8, 8, 6 }, .src_descr = "Off", - .src_num_offset = 0, }, [SCARLETT2_PORT_TYPE_ANALOGUE] = { .id = 0x080, @@ -571,8 +568,6 @@ struct scarlett2_usb_packet { u8 data[]; }; -#define SCARLETT2_USB_PACKET_LEN (sizeof(struct scarlett2_usb_packet)) - static void scarlett2_fill_request_header(struct scarlett2_mixer_data *private, struct scarlett2_usb_packet *req, u32 cmd, u16 req_size) @@ -595,8 +590,8 @@ static int scarlett2_usb( struct scarlett2_mixer_data *private = mixer->private_data; u16 req_buf_size = sizeof(struct scarlett2_usb_packet) + req_size; u16 resp_buf_size = sizeof(struct scarlett2_usb_packet) + resp_size; - struct scarlett2_usb_packet *req = NULL, *resp = NULL; - int err = 0; + struct scarlett2_usb_packet *req, *resp = NULL; + int err; req = kmalloc(req_buf_size, GFP_KERNEL); if (!req) { @@ -1026,10 +1021,6 @@ static int scarlett2_usb_set_mux(struct usb_mixer_interface *mixer) ports, private->mux[j] ) << 12 ); - - /* skip private->mux[j] entries not output */ - j += ports[port_type].num[SCARLETT2_PORT_OUT] - - ports[port_type].num[port_dir_rate]; } err = scarlett2_usb(mixer, SCARLETT2_USB_SET_MUX, @@ -1681,7 +1672,7 @@ static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) int num_inputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_OUT]; int num_outputs = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; - for (i = 0, index = 0; i < num_outputs; i++) { + for (i = 0, index = 0; i < num_outputs; i++) for (j = 0; j < num_inputs; j++, index++) { snprintf(s, sizeof(s), "Mix %c Input %02d Playback Volume", @@ -1691,7 +1682,6 @@ static int scarlett2_add_mixer_ctls(struct usb_mixer_interface *mixer) if (err < 0) return err; } - } return 0; } @@ -1925,8 +1915,6 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) ports[SCARLETT2_PORT_TYPE_ANALOGUE].num[SCARLETT2_PORT_OUT]; int num_mixer_out = ports[SCARLETT2_PORT_TYPE_MIX].num[SCARLETT2_PORT_IN]; - u8 level_switches[SCARLETT2_LEVEL_SWITCH_MAX]; - u8 pad_switches[SCARLETT2_PAD_SWITCH_MAX]; struct scarlett2_usb_volume_status volume_status; int err, i; @@ -1935,11 +1923,9 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) mixer, SCARLETT2_CONFIG_LEVEL_SWITCH, info->level_input_count, - level_switches); + private->level_switch); if (err < 0) return err; - for (i = 0; i < info->level_input_count; i++) - private->level_switch[i] = level_switches[i]; } if (info->pad_input_count) { @@ -1947,11 +1933,9 @@ static int scarlett2_read_configs(struct usb_mixer_interface *mixer) mixer, SCARLETT2_CONFIG_PAD_SWITCH, info->pad_input_count, - pad_switches); + private->pad_switch); if (err < 0) return err; - for (i = 0; i < info->pad_input_count; i++) - private->pad_switch[i] = pad_switches[i]; } err = scarlett2_usb_get_volume_status(mixer, &volume_status); -- 2.31.1

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[PATCH 03/14] ALSA: usb-audio: scarlett2: Coding style improvements
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

Improve alignment and readability with: - Whitespace fixes - Add leading zeros to 32-bit flag values - Rename SCARLETT2_USB_GET_METER_LEVELS to SCARLETT2_USB_GET_METER - Rename SCARLETT2_PORT_DIRECTIONS to SCARLETT2_PORT_DIRNS Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 53 +++++++++++++++++---------------- 1 file changed, 27 insertions(+), 26 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index e327f15b89ce..8026847d27b6 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -175,23 +175,23 @@ static const u16 scarlett2_mixer_values[SCARLETT2_MIXER_VALUE_COUNT] = { * - PCM I/O */ enum { - SCARLETT2_PORT_TYPE_NONE = 0, + SCARLETT2_PORT_TYPE_NONE = 0, SCARLETT2_PORT_TYPE_ANALOGUE = 1, - SCARLETT2_PORT_TYPE_SPDIF = 2, - SCARLETT2_PORT_TYPE_ADAT = 3, - SCARLETT2_PORT_TYPE_MIX = 4, - SCARLETT2_PORT_TYPE_PCM = 5, - SCARLETT2_PORT_TYPE_COUNT = 6, + SCARLETT2_PORT_TYPE_SPDIF = 2, + SCARLETT2_PORT_TYPE_ADAT = 3, + SCARLETT2_PORT_TYPE_MIX = 4, + SCARLETT2_PORT_TYPE_PCM = 5, + SCARLETT2_PORT_TYPE_COUNT = 6, }; /* Count of total I/O and number available at each sample rate */ enum { - SCARLETT2_PORT_IN = 0, - SCARLETT2_PORT_OUT = 1, - SCARLETT2_PORT_OUT_44 = 2, - SCARLETT2_PORT_OUT_88 = 3, + SCARLETT2_PORT_IN = 0, + SCARLETT2_PORT_OUT = 1, + SCARLETT2_PORT_OUT_44 = 2, + SCARLETT2_PORT_OUT_88 = 3, SCARLETT2_PORT_OUT_176 = 4, - SCARLETT2_PORT_DIRECTIONS = 5, + SCARLETT2_PORT_DIRNS = 5, }; /* Hardware buttons on the 18i20 */ @@ -210,7 +210,7 @@ static const char *const scarlett2_button_names[SCARLETT2_BUTTON_MAX] = { */ struct scarlett2_ports { u16 id; - int num[SCARLETT2_PORT_DIRECTIONS]; + int num[SCARLETT2_PORT_DIRNS]; const char * const src_descr; int src_num_offset; const char * const dst_descr; @@ -458,22 +458,23 @@ static int scarlett2_get_port_start_num(const struct scarlett2_ports *ports, #define SCARLETT2_USB_INTERRUPT_INTERVAL 3 /* Interrupt flags for volume and mute/dim button changes */ -#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x400000 -#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x200000 +#define SCARLETT2_USB_INTERRUPT_VOL_CHANGE 0x00400000 +#define SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE 0x00200000 /* Commands for sending/receiving requests/responses */ -#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 +#define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_REQ 2 #define SCARLETT2_USB_VENDOR_SPECIFIC_CMD_RESP 3 -#define SCARLETT2_USB_INIT_SEQ 0x00000000 -#define SCARLETT2_USB_GET_METER_LEVELS 0x00001001 -#define SCARLETT2_USB_GET_MIX 0x00002001 -#define SCARLETT2_USB_SET_MIX 0x00002002 -#define SCARLETT2_USB_GET_MUX 0x00003001 -#define SCARLETT2_USB_SET_MUX 0x00003002 -#define SCARLETT2_USB_GET_DATA 0x00800000 -#define SCARLETT2_USB_SET_DATA 0x00800001 -#define SCARLETT2_USB_DATA_CMD 0x00800002 +#define SCARLETT2_USB_INIT_SEQ 0x00000000 +#define SCARLETT2_USB_GET_METER 0x00001001 +#define SCARLETT2_USB_GET_MIX 0x00002001 +#define SCARLETT2_USB_SET_MIX 0x00002002 +#define SCARLETT2_USB_GET_MUX 0x00003001 +#define SCARLETT2_USB_SET_MUX 0x00003002 +#define SCARLETT2_USB_GET_DATA 0x00800000 +#define SCARLETT2_USB_SET_DATA 0x00800001 +#define SCARLETT2_USB_DATA_CMD 0x00800002 + #define SCARLETT2_USB_CONFIG_SAVE 6 #define SCARLETT2_USB_VOLUME_STATUS_OFFSET 0x31 @@ -835,7 +836,7 @@ static int scarlett2_usb_get_mix(struct usb_mixer_interface *mixer, * (values obtained from private->mix[]) */ static int scarlett2_usb_set_mix(struct usb_mixer_interface *mixer, - int mix_num) + int mix_num) { struct scarlett2_mixer_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; @@ -1056,7 +1057,7 @@ static int scarlett2_usb_get_meter_levels(struct usb_mixer_interface *mixer, req.pad = 0; req.num_meters = cpu_to_le16(SCARLETT2_NUM_METERS); req.magic = cpu_to_le32(SCARLETT2_USB_METER_LEVELS_GET_MAGIC); - err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER_LEVELS, + err = scarlett2_usb(mixer, SCARLETT2_USB_GET_METER, &req, sizeof(req), resp, sizeof(resp)); if (err < 0) return err; -- 2.31.1

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[PATCH 02/14] ALSA: usb-audio: scarlett2: Fix 18i8 Gen 2 PCM Input count
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The 18i8 Gen 2 has 8 PCM Inputs, not 20. Fix the ports entry in s18i8_gen2_info. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 962050436a9f..e327f15b89ce 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -358,7 +358,7 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, [SCARLETT2_PORT_TYPE_PCM] = { .id = 0x600, - .num = { 20, 18, 18, 14, 10 }, + .num = { 8, 18, 18, 14, 10 }, .src_descr = "PCM %d", .src_num_offset = 1, .dst_descr = "PCM %02d Capture" -- 2.31.1

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[PATCH 01/14] ALSA: usb-audio: scarlett2: Remove incorrect S/PDIF comment
by Geoffrey D. Bennett 20 Jun '21

20 Jun '21

The 18i8 Gen 2 S/PDIF outputs are available at 192kHz, unlike the 18i20 Gen 2. Remove the comment that says otherwise. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_scarlett_gen2.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index a461317dc8c6..962050436a9f 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -338,10 +338,6 @@ static const struct scarlett2_device_info s18i8_gen2_info = { }, [SCARLETT2_PORT_TYPE_SPDIF] = { .id = 0x180, - /* S/PDIF outputs aren't available at 192kHz - * but are included in the USB mux I/O - * assignment message anyway - */ .num = { 2, 2, 2, 2, 2 }, .src_descr = "S/PDIF %d", .src_num_offset = 1, -- 2.31.1

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[PATCH AUTOSEL 5.12 11/33] ASoC: AMD Renoir - add DMI entry for Lenovo 2020 AMD platforms
by Sasha Levin 20 Jun '21

20 Jun '21

From: Mark Pearson <markpearson(a)lenovo.com> [ Upstream commit 19a0aa9b04c5ab9a063b6ceaf7211ee7d9a9d24d ] More laptops identified where the AMD ACP bridge needs to be blocked or the microphone will not work when connected to HDMI. Use DMI to block the microphone PCM device for these platforms. Suggested-by: Gabriel Craciunescu <nix.or.die(a)gmail.com> Signed-off-by: Mark Pearson <markpearson(a)lenovo.com> Link: https://lore.kernel.org/r/20210531145502.6079-1-markpearson@lenovo.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 35 +++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 050a61fe9693..47a4dfd81a46 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -199,6 +199,41 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), } }, + { + /* Lenovo ThinkPad P14s Gen 1 (20Y1) */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20Y1"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UH"), + } + }, + { + /* Lenovo ThinkPad T14s Gen1 Campus */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UJ"), + } + }, + { + /* Lenovo ThinkPad T14 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UD"), + } + }, + { + /* Lenovo ThinkPad X13 Gen 1*/ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_BOARD_NAME, "20UF"), + } + }, {} }; -- 2.30.2

2 2

[PATCH] soundwire: stream: Fix test for DP prepare complete
by Richard Fitzgerald 20 Jun '21

20 Jun '21

In sdw_prep_deprep_slave_ports(), after the wait_for_completion() the DP prepare status register is read. If this indicates that the port is now prepared, the code should continue with the port setup. It is irrelevant whether the wait_for_completion() timed out if the port is now ready. The previous implementation would always fail if the wait_for_completion() timed out, even if the port was reporting successful prepare. This patch also fixes a minor bug where the return from sdw_read() was not checked for error - any error code with LSBits clear could be misinterpreted as a successful port prepare. Fixes: 79df15b7d37c ("soundwire: Add helpers for ports operations") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/stream.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 1eaedaaba094..1a18308f4ef4 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -422,7 +422,6 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, struct completion *port_ready; struct sdw_dpn_prop *dpn_prop; struct sdw_prepare_ch prep_ch; - unsigned int time_left; bool intr = false; int ret = 0, val; u32 addr; @@ -479,15 +478,15 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, /* Wait for completion on port ready */ port_ready = &s_rt->slave->port_ready[prep_ch.num]; - time_left = wait_for_completion_timeout(port_ready, - msecs_to_jiffies(dpn_prop->ch_prep_timeout)); + wait_for_completion_timeout(port_ready, + msecs_to_jiffies(dpn_prop->ch_prep_timeout)); val = sdw_read(s_rt->slave, SDW_DPN_PREPARESTATUS(p_rt->num)); - val &= p_rt->ch_mask; - if (!time_left || val) { + if ((val < 0) || (val & p_rt->ch_mask)) { + ret = (val < 0) ? val : -ETIMEDOUT; dev_err(&s_rt->slave->dev, - "Chn prep failed for port:%d\n", prep_ch.num); - return -ETIMEDOUT; + "Chn prep failed for port %d: %d\n", prep_ch.num, ret); + return ret; } } -- 2.20.1

3 2

[PATCH v2] soundwire: bus: Make sdw_nwrite() data pointer argument const
by Richard Fitzgerald 20 Jun '21

20 Jun '21

Idiomatically, write functions should take const pointers to the data buffer, as they don't change the data. They are also likely to be called from functions that receive a const data pointer. Internally the pointer is passed to function/structs shared with the read functions, requiring a cast, but this is an implementation detail that should be hidden by the public API. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 8 ++++---- include/linux/soundwire/sdw.h | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a9e0aa72654d..49c66deba106 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -394,13 +394,13 @@ sdw_nread_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) } static int -sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +sdw_nwrite_no_pm(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { struct sdw_msg msg; int ret; ret = sdw_fill_msg(&msg, slave, addr, count, - slave->dev_num, SDW_MSG_FLAG_WRITE, val); + slave->dev_num, SDW_MSG_FLAG_WRITE, (u8 *)val); if (ret < 0) return ret; @@ -535,9 +535,9 @@ EXPORT_SYMBOL(sdw_nread); * @slave: SDW Slave * @addr: Register address * @count: length - * @val: Buffer for values to be read + * @val: Buffer for values to be written */ -int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, u8 *val) +int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val) { int ret; diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index ced07f8fde87..ee45d2984334 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -1040,7 +1040,7 @@ int sdw_write(struct sdw_slave *slave, u32 addr, u8 value); int sdw_write_no_pm(struct sdw_slave *slave, u32 addr, u8 value); int sdw_read_no_pm(struct sdw_slave *slave, u32 addr); int sdw_nread(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); -int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, u8 *val); +int sdw_nwrite(struct sdw_slave *slave, u32 addr, size_t count, const u8 *val); int sdw_compare_devid(struct sdw_slave *slave, struct sdw_slave_id id); void sdw_extract_slave_id(struct sdw_bus *bus, u64 addr, struct sdw_slave_id *id); -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek: Fix bass speaker DAC mapping for Asus UM431D
by Takashi Iwai 20 Jun '21

20 Jun '21

Asus Zenbook 14 UM431D has two speaker pins and a headphone pin, and the auto-parser ends up assigning the bass to the third DAC 0x06. Although the tone comes out, it's inconvenient because this DAC has no volume control unlike two other DACs. For obtaining the volume control for the bass speaker, this patch enforces the mapping to let both front and bass speaker pins sharing the same DAC. It's not ideal but a little bit of improvement. Since we've already applied the same workaround for another ASUS machine, we just need to hook the chain to the existing quirk. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212547 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 07eabcf22b5f..49f4cac8b05e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7839,6 +7839,8 @@ static const struct hda_fixup alc269_fixups[] = { { 0x20, AC_VERB_SET_PROC_COEF, 0x4e4b }, { } }, + .chained = true, + .chain_id = ALC289_FIXUP_ASUS_GA401, }, [ALC285_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, -- 2.26.2

1 0

[PATCH] ALSA: hda/realtek: Improve fixup for HP Spectre x360 15-df0xxx
by Elia Devito 20 Jun '21

20 Jun '21

On HP Spectre x360 15-df0xxx, after system boot with plugged headset, the headset mic are not detected. Moving pincfg and DAC's config to single fixup function fix this. Signed-off-by: Elia Devito <eliadevito(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 27 ++++++++++++++++++++------- 1 file changed, 20 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59d0936377eb..07eabcf22b5f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6354,6 +6354,24 @@ static void alc_fixup_no_int_mic(struct hda_codec *codec, } } +static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02 }; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* rear speaker */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + /* force front speaker to DAC1 */ + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + break; + } +} + /* for hda_fixup_thinkpad_acpi() */ #include "thinkpad_helper.c" @@ -8138,13 +8156,8 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC269_FIXUP_HP_LINE1_MIC1_LED, }, [ALC285_FIXUP_HP_SPECTRE_X360] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x90170110 }, /* enable top speaker */ - {} - }, - .chained = true, - .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1, + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360, }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, -- 2.31.1

2 1

[PATCH] ALSA: bebob: add support for ToneWeal FW66
by Takashi Sakamoto 20 Jun '21

20 Jun '21

A user of FFADO project reported the issue of ToneWeal FW66. As a result, the device is identified as one of applications of BeBoB solution. I note that in the report the device returns contradictory result in plug discovery process for audio subunit. Fortunately ALSA BeBoB driver doesn't perform it thus it's likely to handle the device without issues. I receive no reaction to test request for this patch yet, however it would be worth to add support for it. daniel@gibbonmoon:/sys/bus/firewire/devices/fw1$ grep -r . * Binary file config_rom matches dev:244:1 guid:0x0023270002000000 hardware_version:0x000002 is_local:0 model:0x020002 model_name:FW66 power/runtime_active_time:0 power/runtime_active_kids:0 power/runtime_usage:0 power/runtime_status:unsupported power/async:disabled power/runtime_suspended_time:0 power/runtime_enabled:disabled power/control:auto subsystem/drivers_autoprobe:1 uevent:MAJOR=244 uevent:MINOR=1 uevent:DEVNAME=fw1 units:0x00a02d:0x010001 vendor:0x002327 vendor_name:ToneWeal fw1.0/uevent:MODALIAS=ieee1394:ven00002327mo00020002sp0000A02Dver00010001 fw1.0/power/runtime_active_time:0 fw1.0/power/runtime_active_kids:0 fw1.0/power/runtime_usage:0 fw1.0/power/runtime_status:unsupported fw1.0/power/async:disabled fw1.0/power/runtime_suspended_time:0 fw1.0/power/runtime_enabled:disabled fw1.0/power/control:auto fw1.0/model:0x020002 fw1.0/rom_index:15 fw1.0/specifier_id:0x00a02d fw1.0/model_name:FW66 fw1.0/version:0x010001 fw1.0/modalias:ieee1394:ven00002327mo00020002sp0000A02Dver00010001 Cc: Daniel Jozsef <daniel.jozsef(a)gmail.com> Reference: https://lore.kernel.org/alsa-devel/20200119164335.GA11974@workstation/ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 62658e2b111a..fd109bea4c53 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -127,6 +127,7 @@ config SND_BEBOB * M-Audio Ozonic/NRV10/ProfireLightBridge * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro + * ToneWeal FW66 To compile this driver as a module, choose M here: the module will be called snd-bebob. diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 7ca452efb133..42980da45fbf 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -56,6 +56,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_FOCUSRITE 0x0000130e #define VEN_MAUDIO 0x00000d6c #define VEN_DIGIDESIGN 0x00a07e +#define OUI_SHOUYO 0x002327 #define MODEL_FOCUSRITE_SAFFIRE_BOTH 0x00000000 #define MODEL_MAUDIO_AUDIOPHILE_BOTH 0x00010060 @@ -469,6 +470,8 @@ static const struct ieee1394_device_id bebob_id_table[] = { &maudio_special_spec), /* Digidesign Mbox 2 Pro */ SND_BEBOB_DEV_ENTRY(VEN_DIGIDESIGN, 0x0000a9, &spec_normal), + // Toneweal FW66. + SND_BEBOB_DEV_ENTRY(OUI_SHOUYO, 0x020002, &spec_normal), /* IDs are unknown but able to be supported */ /* Apogee, Mini-ME Firewire */ /* Apogee, Mini-DAC Firewire */ -- 2.30.2

2 1

[PATCH 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 19 Jun '21

19 Jun '21

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 408 ++++++++++++++++++++++++++++++++++ 4 files changed, 424 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index b4094f07214e..1b87225c39b0 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..78d512d93974 --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,408 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * ALSA SoC DMIC Audio Layer + * + * Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + * + */ + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct platform_device *pdev; + struct clk *dmic_clk; + struct clk *apb_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static void sun50i_snd_rxctrl_enable(struct snd_pcm_substream *substream, + struct sun50i_dmic_dev *host, bool enable) +{ + if (enable) { + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + } else { + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + } +} + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(rtd->cpu_dai); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_write(host->regmap, SUN50I_DMIC_INT_STA, + SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING | + SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct platform_device *pdev = host->pdev; + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(&pdev->dev, "Invalid format!\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + if (i == ARRAY_SIZE(dmic_rate_s)) { + dev_err(&pdev->dev, "Invalid rate!\n"); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sun50i_snd_rxctrl_enable(substream, host, true); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sun50i_snd_rxctrl_enable(substream, host, false); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_set_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (clk_set_rate(host->dmic_clk, freq)) { + dev_err(dai->dev, "Freq : %u not support\n", freq); + return -EINVAL; + } + + return 0; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, + .set_sysclk = sun50i_dmic_set_sysclk, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->apb_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->apb_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + host->pdev = pdev; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->apb_clk = devm_clk_get(&pdev->dev, "apb"); + if (IS_ERR(host->apb_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->apb_clk), + "failed to get apb clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "dmic"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst) && PTR_ERR(host->rst) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + dev_err(&pdev->dev, "Failed to get reset: %d\n", ret); + return ret; + } + if (!IS_ERR(host->rst)) + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.29.0

5 10

[PATCH] ALSA: hda/realtek: Add another ALC236 variant support
by Takashi Iwai 18 Jun '21

18 Jun '21

The codec chip 10ec:0230 is another variant of ALC236, combined with a card reader. Apply the equivalent setup as 10ec:0236. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1184869 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bbbe37f3e724..fac2f15d9e26 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -385,6 +385,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) alc_update_coef_idx(codec, 0x67, 0xf000, 0x3000); fallthrough; case 0x10ec0215: + case 0x10ec0230: case 0x10ec0233: case 0x10ec0235: case 0x10ec0236: @@ -3127,6 +3128,7 @@ static void alc_disable_headset_jack_key(struct hda_codec *codec) alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x48, 0x0); @@ -3154,6 +3156,7 @@ static void alc_enable_headset_jack_key(struct hda_codec *codec) alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x48, 0xd011); @@ -4689,6 +4692,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -4803,6 +4807,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, alc_process_coef_fw(codec, coef0255); snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x45, 0xc489); @@ -4952,6 +4957,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x1b, 0x0e4b); @@ -5050,6 +5056,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -5163,6 +5170,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, coef0255); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, coef0256); @@ -5263,6 +5271,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) val = alc_read_coef_idx(codec, 0x46); is_ctia = (val & 0x0070) == 0x0070; break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_write_coef_idx(codec, 0x1b, 0x0e4b); @@ -5556,6 +5565,7 @@ static void alc255_set_default_jack_type(struct hda_codec *codec) case 0x10ec0255: alc_process_coef_fw(codec, alc255fw); break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: alc_process_coef_fw(codec, alc256fw); @@ -6156,6 +6166,7 @@ static void alc_combo_jack_hp_jd_restart(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 0x8000, 1 << 15); /* Reset HP JD */ alc_update_coef_idx(codec, 0x4a, 0x8000, 0 << 15); break; + case 0x10ec0230: case 0x10ec0235: case 0x10ec0236: case 0x10ec0255: @@ -9130,6 +9141,7 @@ static int patch_alc269(struct hda_codec *codec) spec->shutup = alc256_shutup; spec->init_hook = alc256_init; break; + case 0x10ec0230: case 0x10ec0236: case 0x10ec0256: spec->codec_variant = ALC269_TYPE_ALC256; @@ -10421,6 +10433,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269), HDA_CODEC_ENTRY(0x10ec0222, "ALC222", patch_alc269), HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0230, "ALC236", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), -- 2.26.2

1 0

[PATCH v3 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From fdeab6f32dbdbfd83c9af6efc5a51f91021b6578 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:17:48 +0800 Subject: [PATCH v3 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8 The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- v2 -> v3: no changes. v1 -> v2: fix typo from HP ProBook 440 to HP ProBook 450 G8 (https://lkml.org/lkml/2021/6/18/325) sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab5113cccffa..70bc5b11acfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8336,6 +8336,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

2 1

[PATCH 1/1] ASoC: Intel: bdw-rt5677: remove unnecessary oom message
by Zhen Lei 18 Jun '21

18 Jun '21

Fixes scripts/checkpatch.pl warning: WARNING: Possible unnecessary 'out of memory' message Remove it can help us save a bit of memory. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> --- sound/soc/intel/boards/bdw-rt5677.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 021bc59aac80..e01b7a90ca6c 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -423,10 +423,8 @@ static int bdw_rt5677_probe(struct platform_device *pdev) /* Allocate driver private struct */ bdw_rt5677 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5677_priv), GFP_KERNEL); - if (!bdw_rt5677) { - dev_err(&pdev->dev, "Can't allocate bdw_rt5677\n"); + if (!bdw_rt5677) return -ENOMEM; - } /* override plaform name, if required */ mach = pdev->dev.platform_data; -- 2.25.1

3 2

[PATCH] ASoC: fsi: fix spelling mistake
by Flavio Suligoi 18 Jun '21

18 Jun '21

Fix "thse" --> "these" in struct fsi_stream declaration. Signed-off-by: Flavio Suligoi <f.suligoi(a)asem.it> --- sound/soc/sh/fsi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3c934f87c242..cdf3b7f69ba7 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -219,7 +219,7 @@ struct fsi_stream { u32 bus_option; /* - * thse are initialized by fsi_handler_init() + * these are initialized by fsi_handler_init() */ struct fsi_stream_handler *handler; struct fsi_priv *priv; -- 2.25.1

2 1

[PATCH -next] ASoC: rk817: Constify static struct snd_soc_dai_ops
by Wei Yongjun 18 Jun '21

18 Jun '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 17e672b85ee5..9a896e4326c3 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -382,7 +382,7 @@ static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -static struct snd_soc_dai_ops rk817_dai_ops = { +static const struct snd_soc_dai_ops rk817_dai_ops = { .hw_params = rk817_hw_params, .set_fmt = rk817_set_dai_fmt, .set_sysclk = rk817_set_dai_sysclk,

3 2

[PATCH] soundwire: stream: Use polling for DP Prepare completion
by Richard Fitzgerald 18 Jun '21

18 Jun '21

sdw_prep_deprep_slave_ports() cannot use the port interrupt to signal CP_SM completion because it is called while holding the bus lock, which blocks the alert handler from running. Change to polling the PREPARESTATUS register and remove the infrastructure that was implemented for the previous interrupt waiting. A new configuration field 'ch_prep_poll_us' is added to struct sdw_dpn_prop so that the peripheral driver may select a polling interval. If this is left at zero a default interval of 500 usec is used. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/soundwire/bus.c | 2 -- drivers/soundwire/slave.c | 4 ---- drivers/soundwire/stream.c | 24 ++++++++++++++---------- include/linux/soundwire/sdw.h | 8 ++++++-- 4 files changed, 20 insertions(+), 18 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index adcbf3969110..606fc26d407f 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1351,7 +1351,6 @@ static int sdw_handle_dp0_interrupt(struct sdw_slave *slave, u8 *slave_status) */ if (status & SDW_DP0_INT_PORT_READY) { - complete(&slave->port_ready[0]); clear |= SDW_DP0_INT_PORT_READY; } @@ -1429,7 +1428,6 @@ static int sdw_handle_port_interrupt(struct sdw_slave *slave, * for ports implementing CP_SM. */ if (status & SDW_DPN_INT_PORT_READY) { - complete(&slave->port_ready[port]); clear |= SDW_DPN_INT_PORT_READY; } diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 669d7573320b..55ca884ea951 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -26,7 +26,6 @@ int sdw_slave_add(struct sdw_bus *bus, { struct sdw_slave *slave; int ret; - int i; slave = kzalloc(sizeof(*slave), GFP_KERNEL); if (!slave) @@ -62,9 +61,6 @@ int sdw_slave_add(struct sdw_bus *bus, slave->probed = false; slave->first_interrupt_done = false; - for (i = 0; i < SDW_MAX_PORTS; i++) - init_completion(&slave->port_ready[i]); - mutex_lock(&bus->bus_lock); list_add_tail(&slave->node, &bus->slaves); mutex_unlock(&bus->bus_lock); diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 1eaedaaba094..bd6b3b64de90 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -8,11 +8,13 @@ #include <linux/delay.h> #include <linux/device.h> #include <linux/init.h> +#include <linux/iopoll.h> #include <linux/module.h> #include <linux/mod_devicetable.h> #include <linux/slab.h> #include <linux/soundwire/sdw_registers.h> #include <linux/soundwire/sdw.h> +#include <linux/time.h> #include <sound/soc.h> #include "bus.h" @@ -419,11 +421,10 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, struct sdw_port_runtime *p_rt, bool prep) { - struct completion *port_ready; struct sdw_dpn_prop *dpn_prop; struct sdw_prepare_ch prep_ch; - unsigned int time_left; bool intr = false; + unsigned long prep_poll_us, prep_timeout_us; int ret = 0, val; u32 addr; @@ -478,16 +479,19 @@ static int sdw_prep_deprep_slave_ports(struct sdw_bus *bus, } /* Wait for completion on port ready */ - port_ready = &s_rt->slave->port_ready[prep_ch.num]; - time_left = wait_for_completion_timeout(port_ready, - msecs_to_jiffies(dpn_prop->ch_prep_timeout)); + prep_timeout_us = dpn_prop->ch_prep_timeout * USEC_PER_MSEC; + if (dpn_prop->ch_prep_poll_us) + prep_poll_us = dpn_prop->ch_prep_poll_us; + else + prep_poll_us = SDW_DEFAULT_DP_PREP_POLL_US; - val = sdw_read(s_rt->slave, SDW_DPN_PREPARESTATUS(p_rt->num)); - val &= p_rt->ch_mask; - if (!time_left || val) { + ret = read_poll_timeout(sdw_read, val, ((val & p_rt->ch_mask) == 0), + prep_poll_us, prep_timeout_us, false, + s_rt->slave, SDW_DPN_PREPARESTATUS(p_rt->num)); + if (ret < 0) { dev_err(&s_rt->slave->dev, - "Chn prep failed for port:%d\n", prep_ch.num); - return -ETIMEDOUT; + "Chn prep failed for port %d: %d\n", prep_ch.num, ret); + return ret; } } diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index ddbeb00799e4..4e5290b083bf 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -67,6 +67,9 @@ enum { #define SDW_BLOCK_PACKG_PER_PORT BIT(0) #define SDW_BLOCK_PACKG_PER_CH BIT(1) +/* Default interval to poll for DP prepare completion */ +#define SDW_DEFAULT_DP_PREP_POLL_US 500 + /** * enum sdw_slave_status - Slave status * @SDW_SLAVE_UNATTACHED: Slave is not attached with the bus. @@ -295,6 +298,8 @@ struct sdw_dpn_audio_mode { * @simple_ch_prep_sm: If the port supports simplified channel prepare state * machine * @ch_prep_timeout: Port-specific timeout value, in milliseconds + * @ch_prep_poll_us: Interval to poll for CP_SM prepare completion. Zero means + * poll at SDW_DEFAULT_DP_PREP_POLL_US intervals. * @imp_def_interrupts: If set, each bit corresponds to support for * implementation-defined interrupts * @max_ch: Maximum channels supported @@ -321,6 +326,7 @@ struct sdw_dpn_prop { u32 max_grouping; bool simple_ch_prep_sm; u32 ch_prep_timeout; + u32 ch_prep_poll_us; u32 imp_def_interrupts; u32 max_ch; u32 min_ch; @@ -641,7 +647,6 @@ struct sdw_slave_ops { * @prop: Slave properties * @debugfs: Slave debugfs * @node: node for bus list - * @port_ready: Port ready completion flag for each Slave port * @m_port_map: static Master port map for each Slave port * @dev_num: Current Device Number, values can be 0 or dev_num_sticky * @dev_num_sticky: one-time static Device Number assigned by Bus @@ -673,7 +678,6 @@ struct sdw_slave { struct dentry *debugfs; #endif struct list_head node; - struct completion port_ready[SDW_MAX_PORTS]; unsigned int m_port_map[SDW_MAX_PORTS]; u16 dev_num; u16 dev_num_sticky; -- 2.20.1

2 3

[PATCH v3 2/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 445 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From f6d2556f8f6bcdd2261b48604b8086fb67ae9a03 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:23:06 +0800 Subject: [PATCH v3 2/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 445 G8 The HP ProBook 445 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- v2 -> v3: no changes. v1 -> v2: no changes. sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 70bc5b11acfd..a1e62b2de167 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8344,6 +8344,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.25.1

1 0

[PATCH v2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From fdeab6f32dbdbfd83c9af6efc5a51f91021b6578 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:17:48 +0800 Subject: [PATCH v2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8 The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab5113cccffa..70bc5b11acfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8336,6 +8336,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

1 0

[PATCH 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8
by Andy Chi 18 Jun '21

18 Jun '21

>From fdeab6f32dbdbfd83c9af6efc5a51f91021b6578 Mon Sep 17 00:00:00 2001 From: Andy Chi <andy.chi(a)canonical.com> Date: Fri, 18 Jun 2021 16:17:48 +0800 Subject: [PATCH 1/2] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 450 G8 The HP ProBook 450 G8 using ALC236 codec which using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi <andy.chi(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab5113cccffa..70bc5b11acfd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8336,6 +8336,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.25.1

1 1

[bug report] ASoC: codecs: wcd938x: add capture dapm widgets
by Dan Carpenter 18 Jun '21

18 Jun '21

Hello Srinivas Kandagatla, The patch d5add08fcbce: "ASoC: codecs: wcd938x: add capture dapm widgets" from Jun 9, 2021, leads to the following static checker warning: sound/soc/codecs/wcd938x.c:2189 wcd938x_tx_channel_config() error: uninitialized symbol 'reg'. sound/soc/codecs/wcd938x.c 2165 static void wcd938x_tx_channel_config(struct snd_soc_component *component, 2166 int channel, int mode) 2167 { 2168 int reg, mask; 2169 2170 switch (channel) { 2171 case 0: 2172 reg = WCD938X_ANA_TX_CH2; 2173 mask = WCD938X_HPF1_INIT_MASK; 2174 break; 2175 case 1: 2176 reg = WCD938X_ANA_TX_CH2; 2177 mask = WCD938X_HPF2_INIT_MASK; 2178 break; 2179 case 2: 2180 reg = WCD938X_ANA_TX_CH4; 2181 mask = WCD938X_HPF3_INIT_MASK; 2182 break; 2183 case 3: 2184 reg = WCD938X_ANA_TX_CH4; 2185 mask = WCD938X_HPF4_INIT_MASK; 2186 break; We could add a default case? default: return; 2187 } 2188 2189 snd_soc_component_write_field(component, reg, mask, mode); 2190 } regards, dan carpenter

1 0

[PATCH] ALSA: firewire-motu: fix rx packet format at higher rate for MOTU 828 mk3 Hybrid
by Takashi Sakamoto 18 Jun '21

18 Jun '21

I assumed that the combination of packet formats for MOTU 828 mk3 Hybrid is the same as MOTU 828 mk3 FireWire. However at higher sampling rate, it is different. MOTU 828 mk3 Hybrid has additional 4 dummy data chunks for rx packet. This commit fixes the issue to which I address at a commit f2ac3b839540 ("ALSA: firewire-motu: sequence replay for source packet header"). Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu-protocol-v3.c | 15 ++++++++++++--- sound/firewire/motu/motu.c | 4 ++-- sound/firewire/motu/motu.h | 3 ++- 3 files changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/firewire/motu/motu-protocol-v3.c b/sound/firewire/motu/motu-protocol-v3.c index 4e6b0e449ee4..77e61e89770b 100644 --- a/sound/firewire/motu/motu-protocol-v3.c +++ b/sound/firewire/motu/motu-protocol-v3.c @@ -185,7 +185,7 @@ int snd_motu_protocol_v3_get_clock_source(struct snd_motu *motu, return err; data = be32_to_cpu(reg) & V3_CLOCK_SOURCE_MASK; - if (motu->spec == &snd_motu_spec_828mk3) + if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid) return detect_clock_source_828mk3(motu, data, src); else return v3_detect_clock_source(motu, data, src); @@ -284,14 +284,14 @@ int snd_motu_protocol_v3_cache_packet_formats(struct snd_motu *motu) motu->spec->rx_fixed_pcm_chunks, sizeof(motu->rx_packet_formats.pcm_chunks)); - if (motu->spec == &snd_motu_spec_828mk3) + if (motu->spec == &snd_motu_spec_828mk3_fw || motu->spec == &snd_motu_spec_828mk3_hybrid) return detect_packet_formats_828mk3(motu, data); else return 0; } -const struct snd_motu_spec snd_motu_spec_828mk3 = { +const struct snd_motu_spec snd_motu_spec_828mk3_fw = { .name = "828mk3", .protocol_version = SND_MOTU_PROTOCOL_V3, .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | @@ -300,6 +300,15 @@ const struct snd_motu_spec snd_motu_spec_828mk3 = { .rx_fixed_pcm_chunks = {14, 14, 10}, }; +const struct snd_motu_spec snd_motu_spec_828mk3_hybrid = { + .name = "828mk3", + .protocol_version = SND_MOTU_PROTOCOL_V3, + .flags = SND_MOTU_SPEC_RX_MIDI_3RD_Q | + SND_MOTU_SPEC_TX_MIDI_3RD_Q, + .tx_fixed_pcm_chunks = {18, 18, 14}, + .rx_fixed_pcm_chunks = {14, 14, 14}, // Additional 4 dummy chunks at higher rate. +}; + const struct snd_motu_spec snd_motu_spec_ultralite_mk3 = { .name = "UltraLiteMk3", .protocol_version = SND_MOTU_PROTOCOL_V3, diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 0cae670d711c..543136578c70 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -156,10 +156,10 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000009, &snd_motu_spec_traveler), SND_MOTU_DEV_ENTRY(0x00000d, &snd_motu_spec_ultralite), SND_MOTU_DEV_ENTRY(0x00000f, &snd_motu_spec_8pre), - SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3), // FireWire only. + SND_MOTU_DEV_ENTRY(0x000015, &snd_motu_spec_828mk3_fw), // FireWire only. SND_MOTU_DEV_ENTRY(0x000019, &snd_motu_spec_ultralite_mk3), // FireWire only. SND_MOTU_DEV_ENTRY(0x000030, &snd_motu_spec_ultralite_mk3), // Hybrid. - SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3), // Hybrid. + SND_MOTU_DEV_ENTRY(0x000035, &snd_motu_spec_828mk3_hybrid), // Hybrid. SND_MOTU_DEV_ENTRY(0x000033, &snd_motu_spec_audio_express), SND_MOTU_DEV_ENTRY(0x000045, &snd_motu_spec_4pre), { } diff --git a/sound/firewire/motu/motu.h b/sound/firewire/motu/motu.h index a3deabdf9e34..73f36d1be515 100644 --- a/sound/firewire/motu/motu.h +++ b/sound/firewire/motu/motu.h @@ -130,7 +130,8 @@ extern const struct snd_motu_spec snd_motu_spec_traveler; extern const struct snd_motu_spec snd_motu_spec_ultralite; extern const struct snd_motu_spec snd_motu_spec_8pre; -extern const struct snd_motu_spec snd_motu_spec_828mk3; +extern const struct snd_motu_spec snd_motu_spec_828mk3_fw; +extern const struct snd_motu_spec snd_motu_spec_828mk3_hybrid; extern const struct snd_motu_spec snd_motu_spec_ultralite_mk3; extern const struct snd_motu_spec snd_motu_spec_audio_express; extern const struct snd_motu_spec snd_motu_spec_4pre; -- 2.30.2

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[PATCH] ALSA: bebob: fix rx packet format for Yamaha GO44/GO46, Terratec Phase 24/x24
by Takashi Sakamoto 18 Jun '21

18 Jun '21

Below devices reports zero as the number of channels for external output plug with MIDI type: * Yamaha GO44/GO46 * Terratec Phase 24/X24 As a result, rx packet format is invalid and they generate silent sound. This is a regression added in v5.13. This commit fixes the bug, addressed at a commit 1bd1b3be8655 ("ALSA: bebob: perform sequence replay for media clock recovery"). Cc: <stable(a)vger.kernel.org> Fixes: 5c6ea94f2b7c ("ALSA: bebob: detect the number of available MIDI ports") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_stream.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index e3e23e42add3..8629b14ded76 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -856,6 +856,11 @@ static int detect_midi_ports(struct snd_bebob *bebob, err = avc_bridgeco_get_plug_ch_count(bebob->unit, addr, &ch_count); if (err < 0) break; + // Yamaha GO44, GO46, Terratec Phase 24, Phase x24 reports 0 for the number of + // channels in external output plug 3 (MIDI type) even if it has a pair of physical + // MIDI jacks. As a workaround, assume it as one. + if (ch_count == 0) + ch_count = 1; *midi_ports += ch_count; } @@ -934,12 +939,12 @@ int snd_bebob_stream_discover(struct snd_bebob *bebob) if (err < 0) goto end; - err = detect_midi_ports(bebob, bebob->rx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_IN, + err = detect_midi_ports(bebob, bebob->tx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_IN, plugs[2], &bebob->midi_input_ports); if (err < 0) goto end; - err = detect_midi_ports(bebob, bebob->tx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_OUT, + err = detect_midi_ports(bebob, bebob->rx_stream_formations, addr, AVC_BRIDGECO_PLUG_DIR_OUT, plugs[3], &bebob->midi_output_ports); if (err < 0) goto end; -- 2.30.2

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[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP EliteBook x360 830 G8
by Jeremy Szu 18 Jun '21

18 Jun '21

The HP EliteBook x360 830 G8 using ALC285 codec which using 0x04 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dcd9bceace68..ae5963345093 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8330,6 +8330,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), -- 2.31.1

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