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[PATCH] ASoC: dpcm: remove confusing trace in dpcm_get_be()
by Jerome Brunet 20 Feb '20

20 Feb '20

Now that dpcm_get_be() is used in dpcm_end_walk_at_be(), it is not a error if this function does not find a BE for the provided widget. Remove the related dev_err() trace which is confusing since things might be working as expected. When called from dpcm_add_paths(), it is an error if dpcm_get_be() fails to find a BE for the provided widget. The necessary error trace is already done in this case. Fixes: 027a48387183 ("ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077..aff27c8599ef 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1270,9 +1270,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, } } - /* dai link name and stream name set correctly ? */ - dev_err(card->dev, "ASoC: can't get %s BE for %s\n", - stream ? "capture" : "playback", widget->name); + /* Widget provided is not a BE */ return NULL; } -- 2.24.1

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[PATCH] ASoC: tas2562: Add support for ISENSE and VSENSE
by Dan Murphy 20 Feb '20

20 Feb '20

Add additional support for ISENSE and VSENSE feature for the TAS2562. This feature monitors the output to the loud speaker attempts to eliminate IR drop errors due to packaging. This feature is defined in Section 8.4.5 IV Sense of the data sheet. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 32 +++++++++++++++++++++++++++----- sound/soc/codecs/tas2562.h | 6 +++--- 2 files changed, 30 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 5b4803a76719..b2682c2360b6 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -383,18 +383,34 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; switch (event) { case SND_SOC_DAPM_POST_PMU: - dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_MUTE); + if (ret) + goto end; break; case SND_SOC_DAPM_PRE_PMD: - dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_SHUTDOWN); + if (ret) + goto end; break; default: - break; + dev_err(tas2562->dev, "Not supported evevt\n"); + return -EINVAL; } +end: + if (ret < 0) + return ret; + return 0; } @@ -416,7 +432,6 @@ static const struct snd_kcontrol_new tas2562_snd_controls[] = { static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux), - SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch), @@ -431,7 +446,7 @@ static const struct snd_soc_dapm_route tas2562_audio_map[] = { {"ASI1 Sel", "Left", "ASI1"}, {"ASI1 Sel", "Right", "ASI1"}, {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, - { "DAC", NULL, "DAC IN" }, + { "DAC", NULL, "ASI1 Sel" }, { "OUT", NULL, "DAC" }, {"ISENSE", "Switch", "IMON"}, {"VSENSE", "Switch", "VMON"}, @@ -472,6 +487,13 @@ static struct snd_soc_dai_driver tas2562_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = TAS2562_FORMATS, }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2562_FORMATS, + }, .ops = &tas2562_speaker_dai_ops, }, }; diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 62e659ab786d..6f55ebcf19ea 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -40,7 +40,7 @@ #define TAS2562_RESET BIT(0) -#define TAS2562_MODE_MASK 0x3 +#define TAS2562_MODE_MASK GENMASK(1,0) #define TAS2562_ACTIVE 0x0 #define TAS2562_MUTE 0x1 #define TAS2562_SHUTDOWN 0x2 @@ -73,8 +73,8 @@ #define TAS2562_TDM_CFG2_RXWLEN_24B BIT(3) #define TAS2562_TDM_CFG2_RXWLEN_32B (BIT(2) | BIT(3)) -#define TAS2562_VSENSE_POWER_EN BIT(2) -#define TAS2562_ISENSE_POWER_EN BIT(3) +#define TAS2562_VSENSE_POWER_EN 2 +#define TAS2562_ISENSE_POWER_EN 3 #define TAS2562_TDM_CFG5_VSNS_EN BIT(6) #define TAS2562_TDM_CFG5_VSNS_SLOT_MASK GENMASK(5, 0) -- 2.25.0

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[alsa-devel] Implicit feedback solution for Boss GT-1 and maybe other devices
by Mike Oliphant 20 Feb '20

20 Feb '20

I received a very helpful email response to my previous thread about trying to get the Boss GT-1 pedalboard working reliably, and as a result I now have a working solution. The issue seems to be that the GT-1 does not send any capture data until it gets some playback data first. This block in the "snd_usb_endpoint_start()" function in endpoint.c: "if (snd_usb_endpoint_implicit_feedback_sink(ep)) { ... }" causes the system to wait to send output until some capture urbs are received. As a result, both sides end up waiting on the other, and nothing happens. Simply commenting out that "if" block makes the GT-1 work perfectly with implicit feedback enabled. I plan to submit a patch for this after I do a bit of legwork to see if the fix works for other recent Boss devices (Katana, other GT devices, etc.). Perhaps making the relevant "if" block conditional on a flag, with a USB_ID switch statement to set the flag? Any feedback would be welcome. Thanks, Mike

2 1

Enable Headset Mic for Dell desktop
by Kailang 20 Feb '20

20 Feb '20

Hi Takashi, Attach patch was supported headset Mic. BR, Kailang

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes that can be read using a snd_rawmidi_read. (port is opened for input only) It seems get_avail always returns 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes that can be read using a snd_rawmidi_read. (port is opened for read only) It seems get_avail always returns 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes that can be read via snd_rawmidi_read. (port is opened for read only) It seems get_avail always returns 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes than can read via snd_rawmidi_read. (port is opened for read only) It seems get_avail always returns 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes than can read via snd_rawmidi_read. (port is opened for read only) It seems always return 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - opened 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was opened from MusicMaker: In non blocked raw MIDI mode, is snd_rawmidi_status_get_avail supposed to return the number of available bytes than can read via snd_rawmidi_read. (port is opened for read only) It seems always return 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sytem reboots when initializing Edirol FA-101 firewire audio interface
by Steve Morris 20 Feb '20

20 Feb '20

Hi All, I'm running Arch Linux x86_64 My system consistently reboots when I power on my FA-101 when running kernels 5.5.1-4. Downgrading to 5.4.15 allows everything to work properly. Here's the outpu of: journalctl | grep -E 'Reboot|firewire|fw|bebob|alsa|jack' This is a good init under 5.4.15: -- Reboot -- powered on interface while running 5.4.15, initialized properly Feb 18 18:35:37 hostname kernel: firewire_ohci 0000:05:00.0: enabling device (0080 -> 0083) Feb 18 18:35:37 hostname kernel: firewire_ohci 0000:05:00.0: added OHCI v1.10 device as card 0, 4 IR + 8 IT contexts, quirks 0x11 Feb 18 18:35:37 hostname kernel: firewire_core 0000:05:00.0: created device fw0: GUID 0011223333666677, S400 Feb 18 18:35:37 hostname systemd-udevd[541]: controlC0: Process '/usr/bin/alsactl restore 0' failed with exit code 99. Feb 18 18:35:37 hostname systemd-udevd[541]: controlC1: Process '/usr/bin/alsactl restore 1' failed with exit code 99. Feb 18 18:35:38 hostname kernel: amdgpu: [powerplay] smu driver if version = 0x00000033, smu fw if version = 0x00000035, smu fw version = 0x002a3200 (42.50.0) Feb 18 18:36:08 hostname kernel: firewire_ohci 0000:05:00.0: isochronous cycle inconsistent Feb 18 18:36:08 hostname kernel: firewire_core 0000:05:00.0: created device fw1: GUID 0040ab0000c20bc1, S400 Feb 18 18:36:08 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc1, gap_count=5 Feb 18 18:36:10 hostname kernel: firewire_core 0000:05:00.0: BM lock failed (timeout), making local node (ffc0) root Feb 18 18:36:10 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc0, gap_count=5 Feb 18 18:36:12 hostname systemd-udevd[1532]: controlC2: Process '/usr/bin/alsactl restore 2' failed with exit code 99. Feb 18 18:36:15 hostname kernel: snd-bebob fw1.0: transaction failed: no ack Feb 18 18:36:15 hostname kernel: snd-bebob fw1.0: fail to get an input for MSU in plug 7: -5 Feb 18 18:36:15 hostname kernel: snd-bebob fw1.0: transaction failed: no ack Feb 18 18:36:15 hostname kernel: snd-bebob fw1.0: fail to get an input for MSU in plug 7: -5 This is a bad init under 5.5.4: -- Reboot -- Powered on interface when machine was running. Feb 18 18:13:37 hostname kernel: firewire_ohci 0000:05:00.0: enabling device (0080 -> 0083) Feb 18 18:13:37 hostname kernel: firewire_ohci 0000:05:00.0: added OHCI v1.10 device as card 0, 4 IR + 8 IT contexts, quirks 0x11 Feb 18 18:13:37 hostname kernel: firewire_core 0000:05:00.0: created device fw0: GUID 0011223333666677, S400 Feb 18 18:13:37 hostname kernel: audit: type=1130 audit(1582078417.360:3): pid=1 uid=0 auid=4294967295 ses=4294967295 msg='unit=ufw comm="systemd" exe="/usr/lib/systemd/systemd" hostname=? addr=? terminal=? res=success' Feb 18 18:13:37 hostname audit[1]: SERVICE_START pid=1 uid=0 auid=4294967295 ses=4294967295 msg='unit=ufw comm="systemd" exe="/usr/lib/systemd/systemd" hostname=? addr=? terminal=? res=success' Feb 18 18:13:37 hostname systemd-udevd[539]: controlC0: Process '/usr/bin/alsactl restore 0' failed with exit code 99. Feb 18 18:13:37 hostname systemd-udevd[530]: controlC1: Process '/usr/bin/alsactl restore 1' failed with exit code 99. Feb 18 18:13:38 hostname kernel: amdgpu: [powerplay] smu driver if version = 0x00000033, smu fw if version = 0x00000035, smu fw version = 0x002a3200 (42.50.0) Feb 18 18:19:45 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc1, gap_count=5 Feb 18 18:19:48 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc1, gap_count=5 Feb 18 18:19:48 hostname kernel: firewire_core 0000:05:00.0: created device fw1: GUID 0040ab0000c20bc1, S400 Feb 18 18:19:52 hostname systemd-udevd[1682]: controlC2: Process '/usr/bin/alsactl restore 2' failed with exit code 99. Feb 18 18:20:08 hostname kernel: snd-bebob fw1.0: transaction failed: no ack Feb 18 18:20:08 hostname kernel: snd-bebob fw1.0: fail to get an input for MSU in plug 7: -5 Feb 18 18:20:08 hostname kernel: snd-bebob fw1.0: transaction failed: no ack Feb 18 18:20:08 hostname kernel: snd-bebob fw1.0: fail to get an input for MSU in plug 7: -5 Feb 18 18:20:12 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc1, gap_count=5 Feb 18 18:21:24 hostname kernel: firewire_ohci 0000:05:00.0: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x0001 address=0xd5080000 flags=0x0000] Followed by 125 more O_PAGE_FAULT and then the reboot, I'll be happy to provide additional information if needed. Steve

2 1

[PATCH 0/3] ASoC: snprintf() usage fixes
by Takashi Iwai 19 Feb '20

19 Feb '20

Hi, this is a series of patches I spotted during debugging other code. Intel skl and ASoC PCM core invokes succeeding calls of snprintf() with the incorrect expectation of the return value, which may lead to a buffer overflow. The first patch is yet another bug I found during working on this, and it's rather a real bug that can be seen clearly. Takashi === Takashi Iwai (3): ASoC: intel: skl: Fix pin debug prints ASoC: intel: skl: Fix possible buffer overflow in debug outputs ASoC: pcm: Fix possible buffer overflow in dpcm state sysfs output sound/soc/intel/skylake/skl-debug.c | 32 +++++++++++++++++--------------- sound/soc/soc-pcm.c | 16 ++++++++-------- 2 files changed, 25 insertions(+), 23 deletions(-) -- 2.16.4

3 7

[PATCH v2] ASoC: MT6660 update to 1.0.8_G
by Jeff Chang 19 Feb '20

19 Feb '20

From: Jeff Chang <jeff_chang(a)richtek.com> 1. add mt6660_component_settign for Component INIT Setting Signed-off-by: Jeff Chang <jeff_chang(a)richtek.com> --- sound/soc/codecs/mt6660.c | 78 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 75 insertions(+), 3 deletions(-) changelogs between v2 & v1 - remove parse DT and apply function about Chip INIT SETTING. - use hard coding for MT6660 Component INIT SETTING. diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index a36c416..88cf228 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -9,7 +9,6 @@ #include <linux/i2c.h> #include <linux/pm_runtime.h> #include <linux/delay.h> -#include <linux/debugfs.h> #include <sound/soc.h> #include <sound/tlv.h> #include <sound/pcm_params.h> @@ -225,14 +224,87 @@ static int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) 0x01, on_off ? 0x00 : 0x01); } +struct reg_table { + uint32_t addr; + uint32_t mask; + uint32_t val; +}; + +static const struct reg_table mt6660_setting_table[] = { + { 0x20, 0x80, 0x00 }, + { 0x30, 0x01, 0x00 }, + { 0x50, 0x1c, 0x04 }, + { 0xB1, 0x0c, 0x00 }, + { 0xD3, 0x03, 0x03 }, + { 0xE0, 0x01, 0x00 }, + { 0x98, 0x44, 0x04 }, + { 0xB9, 0xff, 0x82 }, + { 0xB7, 0x7777, 0x7273 }, + { 0xB6, 0x07, 0x03 }, + { 0x6B, 0xe0, 0x20 }, + { 0x07, 0xff, 0x70 }, + { 0xBB, 0xff, 0x20 }, + { 0x69, 0xff, 0x40 }, + { 0xBD, 0xffff, 0x17f8 }, + { 0x70, 0xff, 0x15 }, + { 0x7C, 0xff, 0x00 }, + { 0x46, 0xff, 0x1d }, + { 0x1A, 0xffffffff, 0x7fdb7ffe }, + { 0x1B, 0xffffffff, 0x7fdb7ffe }, + { 0x51, 0xff, 0x58 }, + { 0xA2, 0xff, 0xce }, + { 0x33, 0xffff, 0x7fff }, + { 0x4C, 0xffff, 0x0116 }, + { 0x16, 0x1800, 0x0800 }, + { 0x68, 0x1f, 0x07 }, +}; + +static int mt6660_component_setting(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + size_t i = 0; + + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(component->dev, "%s chip power on failed\n", __func__); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(mt6660_setting_table); i++) { + ret = snd_soc_component_update_bits(component, + mt6660_setting_table[i].addr, + mt6660_setting_table[i].mask, + mt6660_setting_table[i].val); + if (ret < 0) { + dev_err(component->dev, "%s update 0x%02x failed\n", + __func__, mt6660_setting_table[i].addr); + return ret; + } + } + + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(component->dev, "%s chip power off failed\n", __func__); + return ret; + } + + return 0; +} + static int mt6660_component_probe(struct snd_soc_component *component) { struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret; dev_dbg(component->dev, "%s\n", __func__); snd_soc_component_init_regmap(component, chip->regmap); - return 0; + ret = mt6660_component_setting(component); + if (ret < 0) + dev_err(chip->dev, "mt6660 component setting failed\n"); + + return ret; } static void mt6660_component_remove(struct snd_soc_component *component) @@ -506,4 +578,4 @@ module_i2c_driver(mt6660_i2c_driver); MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); MODULE_LICENSE("GPL"); -MODULE_VERSION("1.0.7_G"); +MODULE_VERSION("1.0.8_G"); -- 2.7.4

2 1

[PATCH] ASoC: fix card registration regression.
by Jerome Brunet 19 Feb '20

19 Feb '20

This reverts commit b2354e4009a773c00054b964d937e1b81cb92078. This change might have been desirable to ensure the uniqueness of the component name. It would have helped to better support linux devices which register multiple components, something is which more common than initially thought. However, some card driver are directly using dev_name() to fill the component names of the dai_link which is a problem if want to change the way ASoC generates the component names. Until we figure out the appropriate way to deal with this, revert the change and keep the names as they were. There might be a couple of warning related to debugfs (which were already present before the change) but it is still better than breaking working audio cards. Cc: Marek Szyprowski <m.szyprowski(a)samsung.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-core.c | 29 +---------------------------- 1 file changed, 1 insertion(+), 28 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30c17fde14ca..518b652cf872 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2442,33 +2442,6 @@ static int snd_soc_register_dais(struct snd_soc_component *component, return ret; } -static char *snd_soc_component_unique_name(struct device *dev, - struct snd_soc_component *component) -{ - struct snd_soc_component *pos; - int count = 0; - char *name, *unique; - - name = fmt_single_name(dev, &component->id); - if (!name) - return name; - - /* Count the number of components registred by the device */ - for_each_component(pos) { - if (dev == pos->dev) - count++; - } - - /* Keep naming as it is for the 1st component */ - if (!count) - return name; - - unique = devm_kasprintf(dev, GFP_KERNEL, "%s-%d", name, count); - devm_kfree(dev, name); - - return unique; -} - static int snd_soc_component_initialize(struct snd_soc_component *component, const struct snd_soc_component_driver *driver, struct device *dev) { @@ -2477,7 +2450,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->card_list); mutex_init(&component->io_mutex); - component->name = snd_soc_component_unique_name(dev, component); + component->name = fmt_single_name(dev, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.24.1

3 2

[RFC PATCH 0/2] ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL
by Tzung-Bi Shih 19 Feb '20

19 Feb '20

In previous discussion[1], Mark thinks we should have something general pinctrl integration with DAPM. SND_SOC_DAPM_PINCTRL is exactly what we want but I failed to find existing usage (maybe in topology files?). To be equivalent to mt8183-da7219's original implementation, the 1st patch selects sleep-state at the beginning. The 2nd patch attaches the DAPM widget to playback stream. When the stream is about to start, the power on sequence selects the active-state. When the stream is about to stop, the power off sequence selects the sleep-state. Note: the DAI links will connect to hdmi-codec, that's why the playback stream name is "I2S Playback". Note: this is a RFC series. If it makes sense, will also apply to mt8183-mt6358. [1]: https://patchwork.kernel.org/patch/11379735/ Tzung-Bi Shih (2): ASoC: dapm: select sleep_state when initializing PINCTRL widget ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL in TDM out .../mediatek/mt8183/mt8183-da7219-max98357.c | 98 ++----------------- sound/soc/soc-dapm.c | 3 + 2 files changed, 13 insertions(+), 88 deletions(-) -- 2.25.0.265.gbab2e86ba0-goog

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Applied "ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ce83baca8526689c332135c0abca2667ba056009 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:55:53 +0900 Subject: [PATCH] ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87o8tvjcbc.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/soundwire/qcom.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 1c6c6a2e0def..fb30bbec999a 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -594,6 +594,7 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dai->dev); struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sdw_stream_runtime *sruntime; + struct snd_soc_dai *codec_dai; int ret, i; sruntime = sdw_alloc_stream(dai->name); @@ -602,12 +603,12 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, ctrl->sruntime[dai->id] = sruntime; - for (i = 0; i < rtd->num_codecs; i++) { - ret = snd_soc_dai_set_sdw_stream(rtd->codec_dais[i], sruntime, + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_set_sdw_stream(codec_dai, sruntime, substream->stream); if (ret < 0 && ret != -ENOTSUPP) { dev_err(dai->dev, "Failed to set sdw stream on %s", - rtd->codec_dais[i]->name); + codec_dai->name); sdw_release_stream(sruntime); return ret; } -- 2.20.1

1 0

Applied "ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a4eb41eef331d31b8593defa10b249e155e0314f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:01 +0900 Subject: [PATCH] ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87mu9fjcb4.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 3b5547a27aad..5a23597261ac 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -43,14 +43,14 @@ static int sdm845_slim_snd_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0, i; - for (i = 0 ; i < dai_link->num_codecs; i++) { - ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i], + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_get_channel_map(codec_dai, &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); if (ret != 0 && ret != -ENOTSUPP) { @@ -77,6 +77,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int ret = 0, j; int channels, slot_width; @@ -125,8 +126,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, } } - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { ret = snd_soc_dai_set_tdm_slot( @@ -214,7 +214,6 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); struct snd_jack *jack; - struct snd_soc_dai_link *dai_link = rtd->dai_link; /* * Codec SLIMBUS configuration * RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13 @@ -266,8 +265,8 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) } break; case SLIMBUS_0_RX...SLIMBUS_6_TX: - for (i = 0 ; i < dai_link->num_codecs; i++) { - rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i], + for_each_rtd_codec_dai(rtd, i, codec_dai) { + rval = snd_soc_dai_set_channel_map(codec_dai, ARRAY_SIZE(tx_ch), tx_ch, ARRAY_SIZE(rx_ch), @@ -275,7 +274,7 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) if (rval != 0 && rval != -ENOTSUPP) return rval; - snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0, + snd_soc_dai_set_sysclk(codec_dai, 0, WCD934X_DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); } @@ -345,8 +344,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B; - for (j = 0; j < rtd->num_codecs; j++) { - codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { -- 2.20.1

1 0

Applied "ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From cf4dae032096f1299cf390fd55da489cb445dcbb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:09 +0900 Subject: [PATCH] ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87lfozjcaw.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/apq8016_sbc.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index ac75838bbfab..2d064f3bc9b6 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -34,8 +34,8 @@ struct apq8016_sbc_data { static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; struct snd_soc_component *component; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card); int i, rval; @@ -90,10 +90,9 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } - for (i = 0 ; i < dai_link->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { - component = dai->component; + component = codec_dai->component; /* Set default mclk for internal codec */ rval = snd_soc_component_set_sysclk(component, 0, 0, DEFAULT_MCLK_RATE, SND_SOC_CLOCK_IN); -- 2.20.1

1 0

Applied "ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 225c53a8cfb6fdd8defbbf72e8dcfb3801f7f51e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:15 +0900 Subject: [PATCH] ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87k14jjcaq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index dd80d0186a6c..02aa18d24319 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -164,8 +164,7 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, srate = params_rate(params); - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* 100 Fs to drive 24 bit data */ ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, -- 2.20.1

1 0

Applied "ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 89a2870f6be6aa75de2df22f8baa982c2d7d86e8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:20 +0900 Subject: [PATCH] ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87imk3jcal.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 7a13e9b35187..88f69e3697d2 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -176,10 +176,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *runtime = substream->private_data; + struct snd_soc_dai *codec_dai; int ret, j; - for (j = 0; j < runtime->num_codecs; j++) { - struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + for_each_rtd_codec_dai(runtime, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); @@ -221,10 +221,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int j, ret; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { const char *name = codec_dai->component->name; struct snd_soc_component *component = codec_dai->component; struct snd_soc_dapm_context *dapm = -- 2.20.1

1 0

Applied "ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 56f1003f65830697bd68ae1e26b5561e4cbe6523 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:25 +0900 Subject: [PATCH] ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87h7znjcag.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 03d104fbe185..4a5ef07e956b 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -40,6 +40,7 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; unsigned int rate = params_rate(params); unsigned int mclk_fs_ratio = 256; unsigned int mclk_fs = rate * mclk_fs_ratio; @@ -51,8 +52,7 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) dev_err(rtd->dev, "failed to set cpu dai sysclk\n"); - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_sysclk(codec_dai, @@ -82,10 +82,10 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, static int mt8183_da7219_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int ret = 0, j; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_pll(codec_dai, -- 2.20.1

1 0

Applied "ASoC: dt-bindings: Make RK3328 codec GPIO explicit" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: dt-bindings: Make RK3328 codec GPIO explicit has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e14980976534d9d94f5cddd70033707965482ede Mon Sep 17 00:00:00 2001 From: Robin Murphy <robin.murphy(a)arm.com> Date: Tue, 18 Feb 2020 21:31:58 +0000 Subject: [PATCH] ASoC: dt-bindings: Make RK3328 codec GPIO explicit Existing RK3328 codec drivers have overloaded the GRF phandle to assume implicit control of the limited-function GPIO_MUTE pin, which is usually used to enable an external audio line driver IC. Since this pin has a proper binding of its own (see gpio/rockchip,rk3328-grf-gpio.txt), make a GPIO explicit in the codec binding too. This will help avoid ambiguity on boards that use that pin for some other purpose. (and while touching the example, enforce the "don't include status" rule) Signed-off-by: Robin Murphy <robin.murphy(a)arm.com> Link: https://lore.kernel.org/r/5f7a399dea8a9dedef57f6f99f0f6ab1c1fdc56a.15813767… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/rockchip,rk3328-codec.txt | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt index 2469588c7ccb..1ecd75d2032a 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt @@ -10,6 +10,11 @@ Required properties: - clock-names: should be "pclk". - spk-depop-time-ms: speak depop time msec. +Optional properties: + +- mute-gpios: GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + Example for rk3328 internal codec: codec: codec@ff410000 { @@ -18,6 +23,6 @@ codec: codec@ff410000 { rockchip,grf = <&grf>; clocks = <&cru PCLK_ACODEC>; clock-names = "pclk"; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; spk-depop-time-ms = 100; - status = "disabled"; }; -- 2.20.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: topology: Fix memory leaks with ABI version mismatch
by Dragos Tarcatu 19 Feb '20

19 Feb '20

The topology loader in soc-topology.c is currently accepting ABI version mismatches for some topology components in order to keep backwards compatibility with ABIv4. When dealing with these ABI mismatch situations, temporary copies of the v4 objects are allocated and passed on to the right driver topology handlers. After that they're supposed to be freed. However, memory leaks are currently possible in soc_tplg_link_elems_load() and also soc_tplg_manifest_load(). These patches fix these. I've originally created a PR on github for SOF at: https://github.com/thesofproject/linux/pull/1771 as my impression was this can affect older SOF toplogies. But as it turns out this may impact Skylake instead. Dragos Tarcatu (2): ASoC: topology: Fix memleak in soc_tplg_link_elems_load() ASoC: topology: Fix memleak in soc_tplg_manifest_load() sound/soc/soc-topology.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) -- 2.17.1

2 4

Applied "ASoC: meson: add t9015 internal codec binding documentation" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: meson: add t9015 internal codec binding documentation has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5c36abcd2621adc3d50d05628f0ef0be6e7840a9 Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Wed, 19 Feb 2020 18:35:02 +0100 Subject: [PATCH] ASoC: meson: add t9015 internal codec binding documentation Add the DT binding documention of the internal DAC found in the Amlogic gxl, g12a and sm1 SoC family. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200219173503.1112561-2-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/amlogic,t9015.yaml | 58 +++++++++++++++++++ 1 file changed, 58 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml diff --git a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml new file mode 100644 index 000000000000..b7c38c2b5b54 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,t9015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic T9015 Internal Audio DAC + +maintainers: + - Jerome Brunet <jbrunet(a)baylibre.com> + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 0 + + compatible: + items: + - const: amlogic,t9015 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: pclk + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - resets + +examples: + - | + #include <dt-bindings/clock/g12a-clkc.h> + #include <dt-bindings/reset/amlogic,meson-g12a-reset.h> + + acodec: audio-controller@32000 { + compatible = "amlogic,t9015"; + reg = <0x0 0x32000 0x0 0x14>; + #sound-dai-cells = <0>; + clocks = <&clkc CLKID_AUDIO_CODEC>; + clock-names = "pclk"; + resets = <&reset RESET_AUDIO_CODEC>; + }; + -- 2.20.1

1 0

[PATCH v3 0/2] ASoC: meson: add internal DAC support
by Jerome Brunet 19 Feb '20

19 Feb '20

This patchset adds support for the internal audio DAC provided on the Amlogic gxl, g12a and sm1 SoC families. On each of these SoC families, there is glue between this codec on the audio provider. The architecture is similar to the one used for the synopsys hdmi codec on these SoCs Changes since v2 [1]: * Fix Mute kcontrol name Changes since v1 [0]: * Change some kcontrol names * Move DAC sources to DAPM [0]: https://lore.kernel.org/r/20200219133646.1035506-1-jbrunet@baylibre.com [1]: https://lore.kernel.org/r/20200219161625.1078051-1-jbrunet@baylibre.com Jerome Brunet (2): ASoC: meson: add t9015 internal codec binding documentation ASoC: meson: add t9015 internal DAC driver .../bindings/sound/amlogic,t9015.yaml | 58 +++ sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 2 + sound/soc/meson/t9015.c | 333 ++++++++++++++++++ 4 files changed, 401 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml create mode 100644 sound/soc/meson/t9015.c -- 2.24.1

2 3

[alsa-devel] [PATCH v2] ASoC: dt-bindings: stm32: convert i2s to json-schema
by Olivier Moysan 19 Feb '20

19 Feb '20

Convert the STM32 I2S bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - Define items order for clock and dma properties --- .../bindings/sound/st,stm32-i2s.txt | 62 ------------- .../bindings/sound/st,stm32-i2s.yaml | 87 +++++++++++++++++++ 2 files changed, 87 insertions(+), 62 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt deleted file mode 100644 index cbf24bcd1b8d..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ /dev/null @@ -1,62 +0,0 @@ -STMicroelectronics STM32 SPI/I2S Controller - -The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. -Only some SPI instances support I2S. - -Required properties: - - compatible: Must be "st,stm32h7-i2s" - - reg: Offset and length of the device's register set. - - interrupts: Must contain the interrupt line id. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k". - "i2sclk": clock which feeds the internal clock generator - "pclk": clock which feeds the peripheral bus interface - "x8k": I2S parent clock for sampling rates multiple of 8kHz. - "x11k": I2S parent clock for sampling rates multiple of 11.025kHz. - - dmas: DMA specifiers for tx and rx dma. - See Documentation/devicetree/bindings/dma/stm32-dma.txt. - - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -Optional properties: - - resets: Reference to a reset controller asserting the reset controller - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&i2s2_port>; -}; - -i2s2: audio-controller@40003800 { - compatible = "st,stm32h7-i2s"; - reg = <0x40003800 0x400>; - interrupts = <36>; - clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "i2sclk", "x8k", "x11k"; - dmas = <&dmamux2 2 39 0x400 0x1>, - <&dmamux2 3 40 0x400 0x1>; - dma-names = "rx", "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2s2>; - - i2s2_port: port@0 { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; -}; - -audio-codec { - codec_port: port@0 { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml new file mode 100644 index 000000000000..f32410890589 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 SPI/I2S Controller + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. + Only some SPI instances support I2S. + +properties: + compatible: + enum: + - st,stm32h7-i2s + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock feeding the peripheral bus interface. + - description: clock feeding the internal clock generator. + - description: I2S parent clock for sampling rates multiple of 8kHz. + - description: I2S parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: i2sclk + - const: x8k + - const: x11k + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio capture DMA. + - description: audio playback DMA. + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + i2s2: audio-controller@4000b000 { + compatible = "st,stm32h7-i2s"; + #sound-dai-cells = <0>; + reg = <0x4000b000 0x400>; + clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 39 0x400 0x01>, + <&dmamux1 40 0x400 0x01>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pins_a>; + }; + +... -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: cpcap: Implement set_tdm_slot for voice call support
by Tony Lindgren 19 Feb '20

19 Feb '20

For using cpcap for voice calls, we need to route audio directly from the modem to cpcap for TDM (Time Division Multiplexing). The voice call is direct data between the modem and cpcap with no CPU involvment. In this mode, the cpcap related audio mixer controls work for the speaker selection and volume though. To do this, we need to implement standard snd_soc_dai_set_tdm_slot() for cpcap. Then the modem codec driver can use snd_soc_dai_set_sysclk(), snd_soc_dai_set_fmt(), and snd_soc_dai_set_tdm_slot() to configure a voice call. Let's add cpcap_voice_set_tdm_slot() for this, and cpcap_voice_call() helper to configure the additional registers needed for voice call. Let's also clear CPCAP_REG_VAUDIOC on init in case we have the bit for CPCAP_BIT_VAUDIO_MODE0 set on init. Cc: Arthur D. <spinal.by(a)gmail.com> Cc: Merlijn Wajer <merlijn(a)wizzup.org> Cc: Pavel Machek <pavel(a)ucw.cz> Cc: Sebastian Reichel <sre(a)kernel.org> Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- sound/soc/codecs/cpcap.c | 123 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 123 insertions(+) diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -16,6 +16,14 @@ #include <sound/soc.h> #include <sound/tlv.h> +/* Register 512 CPCAP_REG_VAUDIOC --- Audio Regulator and Bias Voltage */ +#define CPCAP_BIT_AUDIO_LOW_PWR 6 +#define CPCAP_BIT_AUD_LOWPWR_SPEED 5 +#define CPCAP_BIT_VAUDIOPRISTBY 4 +#define CPCAP_BIT_VAUDIO_MODE1 2 +#define CPCAP_BIT_VAUDIO_MODE0 1 +#define CPCAP_BIT_V_AUDIO_EN 0 + /* Register 513 CPCAP_REG_CC --- CODEC */ #define CPCAP_BIT_CDC_CLK2 15 #define CPCAP_BIT_CDC_CLK1 14 @@ -221,6 +229,7 @@ struct cpcap_reg_info { }; static const struct cpcap_reg_info cpcap_default_regs[] = { + { CPCAP_REG_VAUDIOC, 0x003F, 0x0000 }, { CPCAP_REG_CC, 0xFFFF, 0x0000 }, { CPCAP_REG_CC, 0xFFFF, 0x0000 }, { CPCAP_REG_CDI, 0xBFFF, 0x0000 }, @@ -1370,6 +1379,119 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +/* + * Configure codec for voice call if requested. + * + * We can configure most with snd_soc_dai_set_sysclk(), snd_soc_dai_set_fmt() + * and snd_soc_dai_set_tdm_slot(). This function configures the rest of the + * cpcap related hardware as CPU is not involved in the voice call. + */ +static int cpcap_voice_call(struct cpcap_audio *cpcap, struct snd_soc_dai *dai, + bool voice_call) +{ + int mask, err; + + /* Modem to codec VAUDIO_MODE1 */ + mask = BIT(CPCAP_BIT_VAUDIO_MODE1); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_VAUDIOC, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Clear MIC1_MUX for call */ + mask = BIT(CPCAP_BIT_MIC1_MUX); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI, + mask, voice_call ? 0 : mask); + if (err) + return err; + + /* Set MIC2_MUX for call */ + mask = BIT(CPCAP_BIT_MB_ON1L) | BIT(CPCAP_BIT_MB_ON1R) | + BIT(CPCAP_BIT_MIC2_MUX) | BIT(CPCAP_BIT_MIC2_PGA_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Enable LDSP for call */ + mask = BIT(CPCAP_BIT_A2_LDSP_L_EN) | BIT(CPCAP_BIT_A2_LDSP_R_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXOA, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Enable CPCAP_BIT_PGA_CDC_EN for call */ + mask = BIT(CPCAP_BIT_PGA_CDC_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXCOA, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Unmute voice for call */ + if (dai) { + err = snd_soc_dai_digital_mute(dai, !voice_call, + SNDRV_PCM_STREAM_PLAYBACK); + if (err) + return err; + } + + /* Set modem to codec mic CDC and HPF for call */ + mask = BIT(CPCAP_BIT_MIC2_CDC_EN) | BIT(CPCAP_BIT_CDC_EN_RX) | + BIT(CPCAP_BIT_AUDOHPF_1) | BIT(CPCAP_BIT_AUDOHPF_0) | + BIT(CPCAP_BIT_AUDIHPF_1) | BIT(CPCAP_BIT_AUDIHPF_0); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CC, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Enable modem to codec CDC for call*/ + mask = BIT(CPCAP_BIT_CDC_CLK_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, + mask, voice_call ? mask : 0); + + return err; +} + +static int cpcap_voice_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); + int err, ts_mask, mask; + bool voice_call; + + /* + * Primitive test for voice call, probably needs more checks + * later on for 16-bit calls detected, Bluetooth headset etc. + */ + if (tx_mask == 0 && rx_mask == 1 && slot_width == 8) + voice_call = true; + else + voice_call = false; + + ts_mask = 0x7 << CPCAP_BIT_MIC2_TIMESLOT0; + ts_mask |= 0x7 << CPCAP_BIT_MIC1_RX_TIMESLOT0; + + mask = (tx_mask & 0x7) << CPCAP_BIT_MIC2_TIMESLOT0; + mask |= (rx_mask & 0x7) << CPCAP_BIT_MIC1_RX_TIMESLOT0; + + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, + ts_mask, mask); + if (err) + return err; + + err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, slot_width * 1000); + if (err) + return err; + + err = cpcap_voice_call(cpcap, dai, voice_call); + if (err) + return err; + + return 0; +} + static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_component *component = dai->component; @@ -1391,6 +1513,7 @@ static const struct snd_soc_dai_ops cpcap_dai_voice_ops = { .hw_params = cpcap_voice_hw_params, .set_sysclk = cpcap_voice_set_dai_sysclk, .set_fmt = cpcap_voice_set_dai_fmt, + .set_tdm_slot = cpcap_voice_set_tdm_slot, .digital_mute = cpcap_voice_set_mute, }; -- 2.25.0

4 13

[PATCH v2 0/2] ASoC: meson: add internal DAC support
by Jerome Brunet 19 Feb '20

19 Feb '20

This patchset adds support for the internal audio DAC provided on the Amlogic gxl, g12a and sm1 SoC families. On each of these SoC families, there is glue between this codec on the audio provider. The architecture is similar to the one used for the synopsys hdmi codec on these SoCs Changes since v1 [0]: * Change some kcontrol names * Move DAC sources to DAPM [0]: https://lore.kernel.org/r/20200219133646.1035506-1-jbrunet@baylibre.com/ Jerome Brunet (2): ASoC: meson: add t9015 internal codec binding documentation ASoC: meson: add t9015 internal DAC driver .../bindings/sound/amlogic,t9015.yaml | 58 +++ sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 2 + sound/soc/meson/t9015.c | 333 ++++++++++++++++++ 4 files changed, 401 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml create mode 100644 sound/soc/meson/t9015.c -- 2.24.1

2 4

[alsa-devel] ALSA 1.0.29 release
by Jaroslav Kysela 19 Feb '20

19 Feb '20

Hello all, all userspace packages (except alsa-oss) were released today. You may download them from the ALSA website http://www.alsa-project.org or directly from the FTP server ftp://ftp.alsa-project.org/pub . Full list of changes: http://www.alsa-project.org/main/index.php/Changes_v1.0.28_v1.0.29 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

2 16

[alsa-devel] [PATCH 0/9] add ASoC components for AHUB
by Sameer Pujar 19 Feb '20

19 Feb '20

Overview ======== The Audio Hub (AHUB) is part of the Audio Processing Engine (APE) which comprises a collection of hardware accelerators for audio pre-processing and post-processing. It also includes a programmable full crossbar for routing audio data across these accelerators. This series exposes some of these below mentioned HW devices as ASoC components for Tegra platforms from Tegra210 onwards. * ADMAIF : The interface between ADMA and AHUB * XBAR : Crossbar for routing audio samples across various modules * I2S : Inter-IC Sound Controller * DMIC : Digital Microphone * DSPK : Digital Speaker Following is the summary of current series. 1. Add YAML DT binding documentation for above mentioned modules. 2. ACIF programming is same for Tegra generations and hence it is moved to a common file. 3. Add ASoC driver components for each of the above modules. 4. Add DT entries for above components for Tegra210, Tegra186 and Tegra194. 5. Enable these components for Jetson-Tx1, Jetson-Tx2 and Jetson-Xavier. Machine driver series will be sent separately. Sameer Pujar (9): dt-bindings: sound: tegra: add DT binding for AHUB ASoC: tegra: add support for CIF programming ASoC: tegra: add Tegra210 based DMIC driver ASoC: tegra: add Tegra210 based I2S driver ASoC: tegra: add Tegra210 based AHUB driver ASoC: tegra: add Tegra186 based DSPK driver ASoC: tegra: add Tegra210 based ADMAIF driver arm64: tegra: add AHUB components for few Tegra chips arm64: tegra: enable AHUB modules for few Tegra chips .../bindings/sound/nvidia,tegra186-dspk.yaml | 105 +++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 165 ++++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 130 +++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 105 +++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 112 +++ arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 48 ++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 231 ++++- arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 36 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 239 +++++- arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 40 + arch/arm64/boot/dts/nvidia/tegra210.dtsi | 145 ++++ sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 12 + sound/soc/tegra/tegra186_dspk.c | 516 +++++++++++ sound/soc/tegra/tegra186_dspk.h | 73 ++ sound/soc/tegra/tegra210_admaif.c | 896 ++++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 164 ++++ sound/soc/tegra/tegra210_ahub.c | 667 +++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 125 +++ sound/soc/tegra/tegra210_dmic.c | 522 ++++++++++++ sound/soc/tegra/tegra210_dmic.h | 85 ++ sound/soc/tegra/tegra210_i2s.c | 941 +++++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 132 +++ sound/soc/tegra/tegra30_ahub.c | 94 +- sound/soc/tegra/tegra30_ahub.h | 129 --- sound/soc/tegra/tegra30_i2s.c | 35 +- sound/soc/tegra/tegra30_i2s.h | 7 - sound/soc/tegra/tegra_cif.c | 34 + sound/soc/tegra/tegra_cif.h | 50 ++ sound/soc/tegra/tegra_pcm.c | 224 ++++- sound/soc/tegra/tegra_pcm.h | 23 +- 31 files changed, 5897 insertions(+), 244 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.c create mode 100644 sound/soc/tegra/tegra_cif.h -- 2.7.4

3 42

[PATCH 0/2] ASoC: meson: add internal DAC support
by Jerome Brunet 19 Feb '20

19 Feb '20

This patchset adds support for the internal audio DAC provided on the Amlogic gxl, g12a and sm1 SoC families. On each of these SoC families, there is glue between this codec on the audio provider. The architecture is similar to the one used for the synopsys hdmi codec on these SoCs Jerome Brunet (2): ASoC: meson: add t9015 internal codec binding documentation ASoC: meson: add t9015 internal DAC driver .../bindings/sound/amlogic,t9015.yaml | 58 ++++ sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 2 + sound/soc/meson/t9015.c | 320 ++++++++++++++++++ 4 files changed, 388 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml create mode 100644 sound/soc/meson/t9015.c -- 2.24.1

2 5

[alsa-devel] [PATCH v2] ASoC: dt-bindings: stm32: convert sai to json-schema
by Olivier Moysan 19 Feb '20

19 Feb '20

Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - use pattern for compatible of child nodes - rework dmas and clocks properties - add "additionalProperties" --- .../bindings/sound/st,stm32-sai.txt | 107 ---------- .../bindings/sound/st,stm32-sai.yaml | 191 ++++++++++++++++++ 2 files changed, 191 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt deleted file mode 100644 index 944743dd9212..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ /dev/null @@ -1,107 +0,0 @@ -STMicroelectronics STM32 Serial Audio Interface (SAI). - -The SAI interface (Serial Audio Interface) offers a wide set of audio protocols -as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. -The SAI contains two independent audio sub-blocks. Each sub-block has -its own clock generator and I/O lines controller. - -Required properties: - - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" - - reg: Base address and size of SAI common register set. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "pclk" "x8k" and "x11k" - "pclk": Clock which feeds the peripheral bus interface. - Mandatory for "st,stm32h7-sai" compatible. - Not used for "st,stm32f4-sai" compatible. - "x8k": SAI parent clock for sampling rates multiple of 8kHz. - "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. - - interrupts: cpu DAI interrupt line shared by SAI sub-blocks - -Optional properties: - - resets: Reference to a reset controller asserting the SAI - -SAI subnodes: -Two subnodes corresponding to SAI sub-block instances A et B can be defined. -Subnode can be omitted for unsused sub-block. - -SAI subnodes required properties: - - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" - for SAI sub-block A or B respectively. - - reg: Base address and size of SAI sub-block register set. - - clocks: Must contain one phandle and clock specifier pair - for sai_ck which feeds the internal clock generator. - If the SAI shares a master clock, with another SAI set as MCLK - clock provider, SAI provider phandle must be specified here. - - clock-names: Must contain "sai_ck". - Must also contain "MCLK", if SAI shares a master clock, - with a SAI set as MCLK clock provider. - - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: identifier string for each DMA request line - "tx": if sai sub-block is configured as playback DAI - "rx": if sai sub-block is configured as capture DAI - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -SAI subnodes Optional properties: - - st,sync: specify synchronization mode. - By default SAI sub-block is in asynchronous mode. - This property sets SAI sub-block as slave of another SAI sub-block. - Must contain the phandle and index of the sai sub-block providing - the synchronization. - - st,iec60958: support S/PDIF IEC6958 protocol for playback - IEC60958 protocol is not available for capture. - By default, custom protocol is assumed, meaning that protocol is - configured according to protocol defined in related DAI link node, - such as i2s, left justified, right justified, dsp and pdm protocols. - Note: ac97 protocol is not supported by SAI driver - - #clock-cells: should be 0. This property must be present if the SAI device - is a master clock provider, according to clocks bindings, described in - Documentation/devicetree/bindings/clock/clock-bindings.txt. - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&sai1b_port>; -}; - -sai1: sai1@40015800 { - compatible = "st,stm32h7-sai"; - #address-cells = <1>; - #size-cells = <1>; - ranges = <0 0x40015800 0x400>; - reg = <0x40015800 0x4>; - clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "x8k", "x11k"; - interrupts = <87>; - - sai1a: audio-controller@40015804 { - compatible = "st,stm32-sai-sub-a"; - reg = <0x4 0x1C>; - clocks = <&rcc SAI1_CK>; - clock-names = "sai_ck"; - dmas = <&dmamux1 1 87 0x400 0x0>; - dma-names = "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1a>; - - sai1b_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; - }; -}; - -audio-codec { - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000000..51dd2b3bdeb1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,191 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + maxItems: 2 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + clocks: + items: + - description: pclk feeds the peripheral bus interface. + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: x8k + - const: x11k + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + type: object + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + pattern: "st,stm32-sai-sub-[ab]" + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + maxItems: 2 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + maxItems: 2 + + dmas: + maxItems: 1 + + dma-names: + description: | + rx: SAI sub-block is configured as a capture DAI. + tx: SAI sub-block is configured as a playback DAI. + enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + allOf: + - $ref: /schemas/types.yaml#definitions/phandle-array + - maxItems: 1 + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + allOf: + - $ref: /schemas/types.yaml#definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - dmas + - dma-names + + additionalProperties: false + +allOf: + - if: + properties: + compatible: + contains: + const: st,stm32f4-sai + + - then: + properties: + clocks: + items: + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: x8k + - const: x11k + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai1: sai@4400a000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400a000 0x400>; + reg = <0x4400a000 0x4>, <0x4400a3f0 0x10>; + interrupts = <GIC_SPI 87 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&rcc SAI1>, <&rcc PLL1_Q>, <&rcc PLL2_P>; + clock-names = "pclk", "x8k", "x11k"; + resets = <&rcc SAI1_R>; + + sai1a: audio-controller@4400a004 { + compatible = "st,stm32-sai-sub-a"; + #sound-dai-cells = <0>; + reg = <0x4 0x1c>; + clocks = <&rcc SAI1_K>; + clock-names = "sai_ck"; + dmas = <&dmamux1 87 0x400 0x01>; + dma-names = "tx"; + }; + }; + +... -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: da7219: check SRM lock in trigger callback
by Brent Lu 19 Feb '20

19 Feb '20

Intel sst firmware turns on BCLK/WCLK in START Ioctl call which timing is later than the DAPM SUPPLY event handler da7219_dai_event is called (in PREPARED state). Therefore, the SRM lock check always fail. Moving the check to trigger callback could ensure the SRM is locked before DSP starts to process data and avoid possisble noise. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/codecs/da7219.c | 68 +++++++++++++++++++++++++++++++---------------- 1 file changed, 45 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index f83a6ea..0fb5ea5 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -794,9 +794,7 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); struct clk *bclk = da7219->dai_clks[DA7219_DAI_BCLK_IDX]; - u8 pll_ctrl, pll_status; - int i = 0, ret; - bool srm_lock = false; + int ret; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -820,26 +818,6 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, /* PC synchronised to DAI */ snd_soc_component_update_bits(component, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, 0); - - /* Slave mode, if SRM not enabled no need for status checks */ - pll_ctrl = snd_soc_component_read32(component, DA7219_PLL_CTRL); - if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) - return 0; - - /* Check SRM has locked */ - do { - pll_status = snd_soc_component_read32(component, DA7219_PLL_SRM_STS); - if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { - srm_lock = true; - } else { - ++i; - msleep(50); - } - } while ((i < DA7219_SRM_CHECK_RETRIES) && (!srm_lock)); - - if (!srm_lock) - dev_warn(component->dev, "SRM failed to lock\n"); - return 0; case SND_SOC_DAPM_POST_PMD: /* PC free-running */ @@ -1658,12 +1636,56 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, return 0; } +static int da7219_set_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + u8 pll_ctrl, pll_status; + int i = 0; + bool srm_lock = false; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Slave mode, if SRM not enabled no need for status checks */ + pll_ctrl = snd_soc_component_read32(component, DA7219_PLL_CTRL); + if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) + return 0; + + /* Check SRM has locked */ + do { + pll_status = snd_soc_component_read32(component, + DA7219_PLL_SRM_STS); + if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { + srm_lock = true; + } else { + ++i; + msleep(50); + } + } while ((i < DA7219_SRM_CHECK_RETRIES) && (!srm_lock)); + + if (!srm_lock) + dev_warn(component->dev, "SRM failed to lock\n"); + + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + default: + break; + } + + return 0; +} + static const struct snd_soc_dai_ops da7219_dai_ops = { .hw_params = da7219_hw_params, .set_sysclk = da7219_set_dai_sysclk, .set_pll = da7219_set_dai_pll, .set_fmt = da7219_set_dai_fmt, .set_tdm_slot = da7219_set_dai_tdm_slot, + .trigger = da7219_set_dai_trigger, }; #define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ -- 2.7.4

8 20

[PATCH] ALSA: hda/realtek - Fix a regression for mute led on Lenovo Carbon X1
by Hui Wang 19 Feb '20

19 Feb '20

Need to chain the THINKPAD_ACPI, otherwise the mute led will not work. Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- 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 477589e7ec1d..31bee0512334 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6684,6 +6684,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, -- 2.17.1

2 1

ALSA raw MIDI API - client hangup and amidi exits midiC3D0 failed: Device or resource busy
by GitHub issues - opened 19 Feb '20

19 Feb '20

alsa-project/alsa-lib issue #25 was opened from MusicMaker: While developing a client using the raw MIDI interface, bumped into an issue that went away after rebooting Ubuntu 19.10 - 64 bit. Using a MIDISPORT 4x4 one of the 4 ports could not be opened. The client (written in C) call hang and the client thus went unresponsive. No timeout, no error, no nothing. CTRL-BREAK the client and retrying many times gave the exact same result, no other apps were using the port. The other 3 ports on the interface and any other MIDI USB device connected could be opened and used without any problem. When opening the port with amidi, an error was reported, it seems amidi did not gracefully exit as well. amidi -receive -p hw:3,0,1 ALSA lib rawmidi_hw.c:233:(snd_rawmidi_hw_open) open /dev/snd/midiC3D0 failed: Device or resource busy cannot open port "hw:3,0,1": Device or resource busy Looking at the rawmidi_hw.c code there is a GOTO and SYSERR statement. Not sure if those caused this . if (fd < 0) { snd_ctl_close(ctl); SYSERR("open %s failed", filename); return -errno; . After reboot the problem went away. The problem could have been caused while fixing some by coding issue. Maybe someone with more understanding of the code can spot an issue in the code. While many applications use the raw MIDI interface. believe Bitwig and most others use the sequencer API to open ports. Just wanted to log this just in case anyone else experiences this Issue URL : https://github.com/alsa-project/alsa-lib/issues/25 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA raw MIDI API - client hangup and amidi exits midiC3D0 failed: Device or resource busy
by happy musicmaker 19 Feb '20

19 Feb '20

While developing a client using the raw MIDI interface, bumped into an issue that went away after rebooting Ubuntu 19.10 - 64 bit. Using a MIDISPORT 4x4 one of the 4 ports could not be opened. The client (written in C) call hang and the client thus went unresponsive. No timeout, no error, no nothing. CTRL-BREAK the client and retrying many times gave the exact same result, no other apps were using the port. The other 3 ports on the interface and any other MIDI USB device connected could be opened and used without any problem. When opening the port with amidi, an error was reported, it seems amidi did not gracefully exit as well. amidi -receive -p hw:3,0,1 ALSA lib rawmidi_hw.c:233:(snd_rawmidi_hw_open) open /dev/snd/midiC3D0 failed: Device or resource busy cannot open port "hw:3,0,1": Device or resource busy Looking at the rawmidi_hw.c code there is a GOTO and SYSERR statement. Not sure if those caused this . if (fd < 0) { snd_ctl_close(ctl); SYSERR("open %s failed", filename); return -errno; . After reboot the problem went away. The problem could have been caused while fixing some by coding issue. Maybe someone with more understanding of the code can spot an issue in the code. While many applications use the raw MIDI interface. believe Bitwig and most others use the sequencer API to open ports. Just wanted to log this just in case anyone else experiences this Thanks.

1 0

Applied "ASoC: tas2562: Return invalid for when bitwidth is invalid" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: tas2562: Return invalid for when bitwidth is invalid has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 310006cab991fa895cc3a42c53609778b0d50d46 Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Tue, 18 Feb 2020 12:52:52 -0600 Subject: [PATCH] ASoC: tas2562: Return invalid for when bitwidth is invalid If the bitwidth passed in to the set_bitwidth function is not supported then return an error. Fixes: 29b74236bd57 ("ASoC: tas2562: Introduce the TAS2562 amplifier") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20200218185252.26290-1-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas2562.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 729acd874c48..5b4803a76719 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -215,7 +215,8 @@ static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) break; default: - dev_info(tas2562->dev, "Not supported params format\n"); + dev_info(tas2562->dev, "Unsupported bitwidth format\n"); + return -EINVAL; } ret = snd_soc_component_update_bits(tas2562->component, -- 2.20.1

1 0

[PATCH] ASoC: soc-pcm: fix regression in soc_new_pcm()
by Stephan Gerhold 19 Feb '20

19 Feb '20

Commit af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") swapped the SNDRV_PCM_STREAM_* parameter in the snd_soc_dai_stream_valid(cpu_dai, ...) checks. But that works only for codec2codec links. For normal links it breaks registration of playback/capture-only PCM devices. E.g. on qcom/apq8016_sbc there is usually one playback-only and one capture-only PCM device, but they disappeared after the commit. The codec2codec case was added in commit a342031cdd08 ("ASoC: create pcm for codec2codec links as well") as an extra check (e.g. `playback = playback && cpu_playback->channels_min`). We should be able to simplify the code by checking directly for the correct stream type in the loop. This also fixes the regression because we check for PLAYBACK for both codec and cpu dai again when codec2codec is not used. Cc: Sameer Pujar <spujar(a)nvidia.com> Cc: Jerome Brunet <jbrunet(a)baylibre.com> Fixes: af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Original report of the regression: https://lore.kernel.org/alsa-devel/20200217144120.GA243254@gerhold.net/ I'm still quite confused about the crash that was fixed by the commit that introduced the regression... If anything, the original code added in commit a342031cdd08 ("ASoC: create pcm for codec2codec links as well") should have been too "restrictive" since it checked for both PLAYBACK and CAPTURE for the cpu dai in the codec2codec case... Audio works fine gain on apq8016_sbc with this patch, but I don't have any device with codec2codec to test this. --- sound/soc/soc-pcm.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ff1b7c7078e5..cfa24d214a9c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2888,22 +2888,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = rtd->dai_link->dpcm_capture; } else { /* Adapt stream for codec2codec links */ - struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ? - &cpu_dai->driver->playback : &cpu_dai->driver->capture; - struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ? - &cpu_dai->driver->capture : &cpu_dai->driver->playback; + int cpu_capture = rtd->dai_link->params ? + SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int cpu_playback = rtd->dai_link->params ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; for_each_rtd_codec_dai(rtd, i, codec_dai) { if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + snd_soc_dai_stream_valid(cpu_dai, cpu_playback)) playback = 1; if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + snd_soc_dai_stream_valid(cpu_dai, cpu_capture)) capture = 1; } - - capture = capture && cpu_capture->channels_min; - playback = playback && cpu_playback->channels_min; } if (rtd->dai_link->playback_only) { -- 2.25.0

3 2

Applied "ALSA: core: Expand DMA buffer information" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ALSA: core: Expand DMA buffer information has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 386dd54b3a2eedb91aa6e465e7c3a57db04f3960 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:16 +0100 Subject: [PATCH] ALSA: core: Expand DMA buffer information Update DMA buffer definition for snd_compr_runtime so it is represented similarly as in snd_pcm_runtime. While at it, modify snd_compr_set_runtime_buffer to account for newly added members. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Acked-by: Vinod Koul <vkoul(a)kernel.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/compress_driver.h | 35 ++++++++++++++++++++++++--------- 1 file changed, 26 insertions(+), 9 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index bc88d6f964da..00f633c0c3ba 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -23,7 +23,6 @@ struct snd_compr_ops; * struct snd_compr_runtime: runtime stream description * @state: stream state * @ops: pointer to DSP callbacks - * @dma_buffer_p: runtime dma buffer pointer * @buffer: pointer to kernel buffer, valid only when not in mmap mode or * DSP doesn't implement copy * @buffer_size: size of the above buffer @@ -34,11 +33,14 @@ struct snd_compr_ops; * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep * @private_data: driver private data pointer + * @dma_area: virtual buffer address + * @dma_addr: physical buffer address (not accessible from main CPU) + * @dma_bytes: size of DMA area + * @dma_buffer_p: runtime dma buffer pointer */ struct snd_compr_runtime { snd_pcm_state_t state; struct snd_compr_ops *ops; - struct snd_dma_buffer *dma_buffer_p; void *buffer; u64 buffer_size; u32 fragment_size; @@ -47,6 +49,11 @@ struct snd_compr_runtime { u64 total_bytes_transferred; wait_queue_head_t sleep; void *private_data; + + unsigned char *dma_area; + dma_addr_t dma_addr; + size_t dma_bytes; + struct snd_dma_buffer *dma_buffer_p; }; /** @@ -180,19 +187,29 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) /** * snd_compr_set_runtime_buffer - Set the Compress runtime buffer - * @substream: compress substream to set + * @stream: compress stream to set * @bufp: the buffer information, NULL to clear * * Copy the buffer information to runtime buffer when @bufp is non-NULL. * Otherwise it clears the current buffer information. */ -static inline void snd_compr_set_runtime_buffer( - struct snd_compr_stream *substream, - struct snd_dma_buffer *bufp) +static inline void +snd_compr_set_runtime_buffer(struct snd_compr_stream *stream, + struct snd_dma_buffer *bufp) { - struct snd_compr_runtime *runtime = substream->runtime; - - runtime->dma_buffer_p = bufp; + struct snd_compr_runtime *runtime = stream->runtime; + + if (bufp) { + runtime->dma_buffer_p = bufp; + runtime->dma_area = bufp->area; + runtime->dma_addr = bufp->addr; + runtime->dma_bytes = bufp->bytes; + } else { + runtime->dma_buffer_p = NULL; + runtime->dma_area = NULL; + runtime->dma_addr = 0; + runtime->dma_bytes = 0; + } } int snd_compr_stop_error(struct snd_compr_stream *stream, -- 2.20.1

1 0

Applied "ALSA: core: Implement compress page allocation and free routines" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ALSA: core: Implement compress page allocation and free routines has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b9759ef2fd1acb0d3f3dce7991c44a4c5e9e68a3 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:17 +0100 Subject: [PATCH] ALSA: core: Implement compress page allocation and free routines Add simple malloc and free methods for memory management for compress streams. Based on snd_pcm_lib_malloc_pages and snd_pcm_lib_free_pages implementation. Signed-off-by: Divya Prakash <divya1.prakash(a)intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Acked-by: Vinod Koul <vkoul(a)kernel.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/compress_driver.h | 5 ++++ sound/core/compress_offload.c | 42 +++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 00f633c0c3ba..6ce8effa0b12 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data + * @dma_buffer: allocated buffer if any */ struct snd_compr_stream { const char *name; @@ -78,6 +79,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; void *private_data; + struct snd_dma_buffer dma_buffer; }; /** @@ -212,6 +214,9 @@ snd_compr_set_runtime_buffer(struct snd_compr_stream *stream, } } +int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size); +int snd_compr_free_pages(struct snd_compr_stream *stream); + int snd_compr_stop_error(struct snd_compr_stream *stream, snd_pcm_state_t state); diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 9de1c9a0173e..509290f2efa8 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -488,6 +488,48 @@ snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg) } #endif /* !COMPR_CODEC_CAPS_OVERFLOW */ +int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size) +{ + struct snd_dma_buffer *dmab; + int ret; + + if (snd_BUG_ON(!(stream) || !(stream)->runtime)) + return -EINVAL; + dmab = kzalloc(sizeof(*dmab), GFP_KERNEL); + if (!dmab) + return -ENOMEM; + dmab->dev = stream->dma_buffer.dev; + ret = snd_dma_alloc_pages(dmab->dev.type, dmab->dev.dev, size, dmab); + if (ret < 0) { + kfree(dmab); + return ret; + } + + snd_compr_set_runtime_buffer(stream, dmab); + stream->runtime->dma_bytes = size; + return 1; +} +EXPORT_SYMBOL(snd_compr_malloc_pages); + +int snd_compr_free_pages(struct snd_compr_stream *stream) +{ + struct snd_compr_runtime *runtime = stream->runtime; + + if (snd_BUG_ON(!(stream) || !(stream)->runtime)) + return -EINVAL; + if (runtime->dma_area == NULL) + return 0; + if (runtime->dma_buffer_p != &stream->dma_buffer) { + /* It's a newly allocated buffer. Release it now. */ + snd_dma_free_pages(runtime->dma_buffer_p); + kfree(runtime->dma_buffer_p); + } + + snd_compr_set_runtime_buffer(stream, NULL); + return 0; +} +EXPORT_SYMBOL(snd_compr_free_pages); + /* revisit this with snd_pcm_preallocate_xxx */ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, struct snd_compr_params *params) -- 2.20.1

1 0

[PATCH v5 0/9] ASoC: SOF: Data probing
by Cezary Rojewski 19 Feb '20

19 Feb '20

This set of patches achieves few goals in order to enable data probing feature for audio DSP. First, provide new and alter existing interfaces (page allocation, runtime flow adjustments) to make them compress friendly. Finally, the end goal which are the probe APIs and usage itself. Probes can be treated as endpoints which allow for data extraction from or injection to target module - a great ally when debugging problematic audio issues such as distortions, glitches or gaps. Compress streams are a weapon of choice here to provide a lightweight implementation. While all available IPCs have been defined, current implementation covers extraction only, with injection scheduled for a later date. Initial review and development of probes can be found under: https://github.com/thesofproject/linux/pull/1276 with the hda-compress-enable set of patches being separated and reviewed on: https://github.com/thesofproject/linux/pull/1571 Tested on CML-U with rt5682 i2s board. Cahnges in v5: - Removed SND_SOC_SOF_HDA_LINK requirement for hda probes. Instead of sound/hda/ext, sof hda operations are used - In consequence of above, patches: ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams have been dropped. - In order to reuse pcm utility functions for hda compress, new patch has been added: ASoC: SOF: Intel: Expose SDnFMT helpers - Addressed u64/ u32 concerns from Pierre when calculating bytes transferred during runtime Changes in v4: - No changes to ALSA core and hda patches - 'probe' source added in patch "ASoC: SOF: Implement Probe IPC API" now only compiled if SND_SOC_SOF_DEBUG_PROBES is enabled. Said kconfig initial declaration has been moved to that patch too - 'extractor_stream_tag' field now decorated with _DEBUG_PROBES to account for the change above - Debugfs probe handlers now check if extractor stream is running before continuing. To achieve this, 'extractor_stream_tag' is both initialized and set to SOF_PROBE_INVALID_NODE_ID on sof_probe_compr_free(). Suggestion provided by Pierre - Reverted SND_SOC_SOF_HDA_LINK requirement-removal change for HDA probes. Code requires SND_HDA_EXT_CORE in order to function on Intel platforms - Relocated probe-compress related code from /sof/pcm.c directly to /sof/compress.c as requested by Daniel Changes in v3: - No changes to ALSA core and hda patches - Addressed sparse and all doc related warnings as reported by Pierre - Moved _DEBUG_PROBES config outside of _DEVELOPER_SUPPORT block - Renamed 'extractor' field to 'extractor_stream_tag' as requested - Relocated 'extractor_stream_tag' declaration from patch 07 to 08 Changes in v2: - No changes to ALSA core and hda patches - Removed "ASoC: Intel: sof_rt5682: Add compress probe DAI links" from the patchset list as requested by Pierre - updated copyright header for newly added files (dates) - probes over HDA no longer require SND_SOC_SOF_HDA_LINK enabled - renamed debugfs probe functions as requested by Pierre - probe IPC API has been updated to align with newest SOF FW & probe debug app (struct_size macro has been enlisted to make the size calculations transparent). This targets only "ASoC: SOF: Implement Probe IPC API" patch Cezary Rojewski (9): ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: SOF: Intel: Account for compress streams when servicing IRQs ASoC: SOF: Implement Probe IPC API ASoC: SOF: Generic probe compress operations ASoC: SOF: Intel: Expose SDnFMT helpers ASoC: SOF: Intel: Probe compress operations ASoC: SOF: Provide probe debugfs support ASoC: SOF: Intel: Add Probe compress CPU DAIs include/sound/compress_driver.h | 40 +++- include/sound/hdaudio.h | 2 + include/sound/sof/header.h | 11 ++ sound/core/compress_offload.c | 42 +++++ sound/soc/sof/Kconfig | 9 + sound/soc/sof/Makefile | 1 + sound/soc/sof/compress.c | 146 +++++++++++++++ sound/soc/sof/compress.h | 31 ++++ sound/soc/sof/core.c | 6 + sound/soc/sof/debug.c | 226 +++++++++++++++++++++++ sound/soc/sof/intel/Kconfig | 9 + sound/soc/sof/intel/Makefile | 1 + sound/soc/sof/intel/apl.c | 9 + sound/soc/sof/intel/cnl.c | 9 + sound/soc/sof/intel/hda-compress.c | 114 ++++++++++++ sound/soc/sof/intel/hda-dai.c | 28 +++ sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/intel/hda-pcm.c | 8 +- sound/soc/sof/intel/hda-stream.c | 25 ++- sound/soc/sof/intel/hda.h | 32 ++++ sound/soc/sof/ops.h | 43 +++++ sound/soc/sof/pcm.c | 7 + sound/soc/sof/probe.c | 286 +++++++++++++++++++++++++++++ sound/soc/sof/probe.h | 85 +++++++++ sound/soc/sof/sof-priv.h | 25 +++ 25 files changed, 1183 insertions(+), 16 deletions(-) create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h create mode 100644 sound/soc/sof/intel/hda-compress.c create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h -- 2.17.1

3 13

Applied "ASoC: SOF: Implement Probe IPC API" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: SOF: Implement Probe IPC API has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f3b433e4699fa358ce5b7bd7688bebe36068c199 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:19 +0100 Subject: [PATCH] ASoC: SOF: Implement Probe IPC API Add all required types and methods to support each and every request that driver could sent to firmware. Probe is one of SOF firmware features which allows for data extraction and injection directly from or to DMA stream. Exposes eight IPCs: - addition and removal of injection DMAs - addition and removal of probe points - info retrieval of injection DMAs and probe points - probe initialization and cleanup Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/sof/header.h | 11 ++ sound/soc/sof/Kconfig | 8 + sound/soc/sof/Makefile | 1 + sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/probe.c | 286 ++++++++++++++++++++++++++++++++++ sound/soc/sof/probe.h | 85 ++++++++++ 6 files changed, 394 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index bf3edd9c08b4..b79479575cc8 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -51,6 +51,7 @@ #define SOF_IPC_GLB_TRACE_MSG SOF_GLB_TYPE(0x9U) #define SOF_IPC_GLB_GDB_DEBUG SOF_GLB_TYPE(0xAU) #define SOF_IPC_GLB_TEST_MSG SOF_GLB_TYPE(0xBU) +#define SOF_IPC_GLB_PROBE SOF_GLB_TYPE(0xCU) /* * DSP Command Message Types @@ -102,6 +103,16 @@ #define SOF_IPC_STREAM_VORBIS_PARAMS SOF_CMD_TYPE(0x010) #define SOF_IPC_STREAM_VORBIS_FREE SOF_CMD_TYPE(0x011) +/* probe */ +#define SOF_IPC_PROBE_INIT SOF_CMD_TYPE(0x001) +#define SOF_IPC_PROBE_DEINIT SOF_CMD_TYPE(0x002) +#define SOF_IPC_PROBE_DMA_ADD SOF_CMD_TYPE(0x003) +#define SOF_IPC_PROBE_DMA_INFO SOF_CMD_TYPE(0x004) +#define SOF_IPC_PROBE_DMA_REMOVE SOF_CMD_TYPE(0x005) +#define SOF_IPC_PROBE_POINT_ADD SOF_CMD_TYPE(0x006) +#define SOF_IPC_PROBE_POINT_INFO SOF_CMD_TYPE(0x007) +#define SOF_IPC_PROBE_POINT_REMOVE SOF_CMD_TYPE(0x008) + /* trace */ #define SOF_IPC_TRACE_DMA_PARAMS SOF_CMD_TYPE(0x001) #define SOF_IPC_TRACE_DMA_POSITION SOF_CMD_TYPE(0x002) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 827b0ec92522..65c3cfbcb812 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -41,6 +41,14 @@ config SND_SOC_SOF_OF required to enable i.MX8 devices. Say Y if you need this option. If unsure select "N". +config SND_SOC_SOF_DEBUG_PROBES + bool "SOF enable data probing" + help + This option enables the data probing feature that can be used to + gather data directly from specific points of the audio pipeline. + Say Y if you want to enable probes. + If unsure, select "N". + config SND_SOC_SOF_DEVELOPER_SUPPORT bool "SOF developer options support" depends on EXPERT diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 0a8bc72c28a5..18d7cab9046e 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -2,6 +2,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o utils.o sof-audio.o +snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o snd-sof-pci-objs := sof-pci-dev.o snd-sof-acpi-objs := sof-acpi-dev.o diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 1837f66e361f..922052883b0a 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -106,7 +106,9 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) ret = reply.error; } else { /* reply correct size ? */ - if (reply.hdr.size != msg->reply_size) { + if (reply.hdr.size != msg->reply_size && + /* getter payload is never known upfront */ + !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", msg->reply_size, reply.hdr.size); ret = -EINVAL; diff --git a/sound/soc/sof/probe.c b/sound/soc/sof/probe.c new file mode 100644 index 000000000000..2b2f3dcfc7e9 --- /dev/null +++ b/sound/soc/sof/probe.c @@ -0,0 +1,286 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski <cezary.rojewski(a)intel.com> +// + +#include "sof-priv.h" +#include "probe.h" + +/** + * sof_ipc_probe_init - initialize data probing + * @sdev: SOF sound device + * @stream_tag: Extractor stream tag + * @buffer_size: DMA buffer size to set for extractor + * + * Host chooses whether extraction is supported or not by providing + * valid stream tag to DSP. Once specified, stream described by that + * tag will be tied to DSP for extraction for the entire lifetime of + * probe. + * + * Probing is initialized only once and each INIT request must be + * matched by DEINIT call. + */ +int sof_ipc_probe_init(struct snd_sof_dev *sdev, + u32 stream_tag, size_t buffer_size) +{ + struct sof_ipc_probe_dma_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, dma, 1); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_INIT; + msg->num_elems = 1; + msg->dma[0].stream_tag = stream_tag; + msg->dma[0].dma_buffer_size = buffer_size; + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_init); + +/** + * sof_ipc_probe_deinit - cleanup after data probing + * @sdev: SOF sound device + * + * Host sends DEINIT request to free previously initialized probe + * on DSP side once it is no longer needed. DEINIT only when there + * are no probes connected and with all injectors detached. + */ +int sof_ipc_probe_deinit(struct snd_sof_dev *sdev) +{ + struct sof_ipc_cmd_hdr msg; + struct sof_ipc_reply reply; + + msg.size = sizeof(msg); + msg.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DEINIT; + + return sof_ipc_tx_message(sdev->ipc, msg.cmd, &msg, msg.size, + &reply, sizeof(reply)); +} +EXPORT_SYMBOL(sof_ipc_probe_deinit); + +static int sof_ipc_probe_info(struct snd_sof_dev *sdev, unsigned int cmd, + void **params, size_t *num_params) +{ + struct sof_ipc_probe_info_params msg = {{{0}}}; + struct sof_ipc_probe_info_params *reply; + size_t bytes; + int ret; + + *params = NULL; + *num_params = 0; + + reply = kzalloc(SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); + if (!reply) + return -ENOMEM; + msg.rhdr.hdr.size = sizeof(msg); + msg.rhdr.hdr.cmd = SOF_IPC_GLB_PROBE | cmd; + + ret = sof_ipc_tx_message(sdev->ipc, msg.rhdr.hdr.cmd, &msg, + msg.rhdr.hdr.size, reply, SOF_IPC_MSG_MAX_SIZE); + if (ret < 0 || reply->rhdr.error < 0) + goto exit; + + if (!reply->num_elems) + goto exit; + + bytes = reply->num_elems * sizeof(reply->dma[0]); + *params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL); + if (!*params) { + ret = -ENOMEM; + goto exit; + } + *num_params = msg.num_elems; + +exit: + kfree(reply); + return ret; +} + +/** + * sof_ipc_probe_dma_info - retrieve list of active injection dmas + * @sdev: SOF sound device + * @dma: Returned list of active dmas + * @num_dma: Returned count of active dmas + * + * Host sends DMA_INFO request to obtain list of injection dmas it + * can use to transfer data over with. + * + * Note that list contains only injection dmas as there is only one + * extractor (dma) and it is always assigned on probing init. + * DSP knows exactly where data from extraction probes is going to, + * which is not the case for injection where multiple streams + * could be engaged. + */ +int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev, + struct sof_probe_dma **dma, size_t *num_dma) +{ + return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_DMA_INFO, + (void **)dma, num_dma); +} +EXPORT_SYMBOL(sof_ipc_probe_dma_info); + +/** + * sof_ipc_probe_dma_add - attach to specified dmas + * @sdev: SOF sound device + * @dma: List of streams (dmas) to attach to + * @num_dma: Number of elements in @dma + * + * Contrary to extraction, injection streams are never assigned + * on init. Before attempting any data injection, host is responsible + * for specifying streams which will be later used to transfer data + * to connected probe points. + */ +int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev, + struct sof_probe_dma *dma, size_t num_dma) +{ + struct sof_ipc_probe_dma_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, dma, num_dma); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_dma; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_ADD; + memcpy(&msg->dma[0], dma, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_dma_add); + +/** + * sof_ipc_probe_dma_remove - detach from specified dmas + * @sdev: SOF sound device + * @stream_tag: List of stream tags to detach from + * @num_stream_tag: Number of elements in @stream_tag + * + * Host sends DMA_REMOVE request to free previously attached stream + * from being occupied for injection. Each detach operation should + * match equivalent DMA_ADD. Detach only when all probes tied to + * given stream have been disconnected. + */ +int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev, + unsigned int *stream_tag, size_t num_stream_tag) +{ + struct sof_ipc_probe_dma_remove_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, stream_tag, num_stream_tag); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_stream_tag; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_REMOVE; + memcpy(&msg->stream_tag[0], stream_tag, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_dma_remove); + +/** + * sof_ipc_probe_points_info - retrieve list of active probe points + * @sdev: SOF sound device + * @desc: Returned list of active probes + * @num_desc: Returned count of active probes + * + * Host sends PROBE_POINT_INFO request to obtain list of active probe + * points, valid for disconnection when given probe is no longer + * required. + */ +int sof_ipc_probe_points_info(struct snd_sof_dev *sdev, + struct sof_probe_point_desc **desc, size_t *num_desc) +{ + return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_POINT_INFO, + (void **)desc, num_desc); +} +EXPORT_SYMBOL(sof_ipc_probe_points_info); + +/** + * sof_ipc_probe_points_add - connect specified probes + * @sdev: SOF sound device + * @desc: List of probe points to connect + * @num_desc: Number of elements in @desc + * + * Dynamically connects to provided set of endpoints. Immediately + * after connection is established, host must be prepared to + * transfer data from or to target stream given the probing purpose. + * + * Each probe point should be removed using PROBE_POINT_REMOVE + * request when no longer needed. + */ +int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, + struct sof_probe_point_desc *desc, size_t num_desc) +{ + struct sof_ipc_probe_point_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, desc, num_desc); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_desc; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_ADD; + memcpy(&msg->desc[0], desc, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_points_add); + +/** + * sof_ipc_probe_points_remove - disconnect specified probes + * @sdev: SOF sound device + * @buffer_id: List of probe points to disconnect + * @num_buffer_id: Number of elements in @desc + * + * Removes previously connected probes from list of active probe + * points and frees all resources on DSP side. + */ +int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, + unsigned int *buffer_id, size_t num_buffer_id) +{ + struct sof_ipc_probe_point_remove_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, buffer_id, num_buffer_id); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_buffer_id; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_REMOVE; + memcpy(&msg->buffer_id[0], buffer_id, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_points_remove); diff --git a/sound/soc/sof/probe.h b/sound/soc/sof/probe.h new file mode 100644 index 000000000000..45daa5552834 --- /dev/null +++ b/sound/soc/sof/probe.h @@ -0,0 +1,85 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2019-2020 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski <cezary.rojewski(a)intel.com> + */ + +#ifndef __SOF_PROBE_H +#define __SOF_PROBE_H + +#include <sound/sof/header.h> + +struct snd_sof_dev; + +#define SOF_PROBE_INVALID_NODE_ID UINT_MAX + +struct sof_probe_dma { + unsigned int stream_tag; + unsigned int dma_buffer_size; +} __packed; + +enum sof_connection_purpose { + SOF_CONNECTION_PURPOSE_EXTRACT = 1, + SOF_CONNECTION_PURPOSE_INJECT, +}; + +struct sof_probe_point_desc { + unsigned int buffer_id; + unsigned int purpose; + unsigned int stream_tag; +} __packed; + +struct sof_ipc_probe_dma_add_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + struct sof_probe_dma dma[0]; +} __packed; + +struct sof_ipc_probe_info_params { + struct sof_ipc_reply rhdr; + unsigned int num_elems; + union { + struct sof_probe_dma dma[0]; + struct sof_probe_point_desc desc[0]; + }; +} __packed; + +struct sof_ipc_probe_dma_remove_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + unsigned int stream_tag[0]; +} __packed; + +struct sof_ipc_probe_point_add_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + struct sof_probe_point_desc desc[0]; +} __packed; + +struct sof_ipc_probe_point_remove_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + unsigned int buffer_id[0]; +} __packed; + +int sof_ipc_probe_init(struct snd_sof_dev *sdev, + u32 stream_tag, size_t buffer_size); +int sof_ipc_probe_deinit(struct snd_sof_dev *sdev); +int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev, + struct sof_probe_dma **dma, size_t *num_dma); +int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev, + struct sof_probe_dma *dma, size_t num_dma); +int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev, + unsigned int *stream_tag, size_t num_stream_tag); +int sof_ipc_probe_points_info(struct snd_sof_dev *sdev, + struct sof_probe_point_desc **desc, size_t *num_desc); +int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, + struct sof_probe_point_desc *desc, size_t num_desc); +int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, + unsigned int *buffer_id, size_t num_buffer_id); + +#endif -- 2.20.1

1 0

Applied "ASoC: SOF: Generic probe compress operations" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: SOF: Generic probe compress operations has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e145e9af231adff081e0e16e1dacfb6e3c4e968f Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:20 +0100 Subject: [PATCH] ASoC: SOF: Generic probe compress operations Define system-agnostic probe compress flow which serves as a base for actual, hardware-dependent implementations. As per firmware spec, maximum of one extraction stream is allowed, while for injection, there can be plenty. Apart from probe_pointer, all probe compress operations are mandatory. Copy operation is defined as unified as its flow should be shared across all SOF systems. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/Kconfig | 1 + sound/soc/sof/Makefile | 2 +- sound/soc/sof/compress.c | 141 +++++++++++++++++++++++++++++++++++++++ sound/soc/sof/compress.h | 29 ++++++++ sound/soc/sof/core.c | 6 ++ sound/soc/sof/ops.h | 43 ++++++++++++ sound/soc/sof/sof-priv.h | 25 +++++++ 7 files changed, 246 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 65c3cfbcb812..4dda4b62509f 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_SOF_OF config SND_SOC_SOF_DEBUG_PROBES bool "SOF enable data probing" + select SND_SOC_COMPRESS help This option enables the data probing feature that can be used to gather data directly from specific points of the audio pipeline. diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 18d7cab9046e..8eca2f85c90e 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -2,7 +2,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o utils.o sof-audio.o -snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o +snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o compress.o snd-sof-pci-objs := sof-pci-dev.o snd-sof-acpi-objs := sof-acpi-dev.o diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c new file mode 100644 index 000000000000..e87cc81a0599 --- /dev/null +++ b/sound/soc/sof/compress.c @@ -0,0 +1,141 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski <cezary.rojewski(a)intel.com> +// + +#include <sound/soc.h> +#include "compress.h" +#include "ops.h" +#include "probe.h" + +int sof_probe_compr_open(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + int ret; + + ret = snd_sof_probe_compr_assign(sdev, cstream, dai); + if (ret < 0) { + dev_err(dai->dev, "Failed to assign probe stream: %d\n", ret); + return ret; + } + + sdev->extractor_stream_tag = ret; + return 0; +} +EXPORT_SYMBOL(sof_probe_compr_open); + +int sof_probe_compr_free(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + struct sof_probe_point_desc *desc; + size_t num_desc; + int i, ret; + + /* disconnect all probe points */ + ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc); + if (ret < 0) { + dev_err(dai->dev, "Failed to get probe points: %d\n", ret); + goto exit; + } + + for (i = 0; i < num_desc; i++) + sof_ipc_probe_points_remove(sdev, &desc[i].buffer_id, 1); + kfree(desc); + +exit: + ret = sof_ipc_probe_deinit(sdev); + if (ret < 0) + dev_err(dai->dev, "Failed to deinit probe: %d\n", ret); + + sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID; + snd_compr_free_pages(cstream); + + return snd_sof_probe_compr_free(sdev, cstream, dai); +} +EXPORT_SYMBOL(sof_probe_compr_free); + +int sof_probe_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + int ret; + + cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG; + cstream->dma_buffer.dev.dev = sdev->dev; + ret = snd_compr_malloc_pages(cstream, rtd->buffer_size); + if (ret < 0) + return ret; + + ret = snd_sof_probe_compr_set_params(sdev, cstream, params, dai); + if (ret < 0) + return ret; + + ret = sof_ipc_probe_init(sdev, sdev->extractor_stream_tag, + rtd->dma_bytes); + if (ret < 0) { + dev_err(dai->dev, "Failed to init probe: %d\n", ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL(sof_probe_compr_set_params); + +int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + + return snd_sof_probe_compr_trigger(sdev, cstream, cmd, dai); +} +EXPORT_SYMBOL(sof_probe_compr_trigger); + +int sof_probe_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + + return snd_sof_probe_compr_pointer(sdev, cstream, tstamp, dai); +} +EXPORT_SYMBOL(sof_probe_compr_pointer); + +int sof_probe_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + unsigned int offset, n; + void *ptr; + int ret; + + if (count > rtd->buffer_size) + count = rtd->buffer_size; + + div_u64_rem(rtd->total_bytes_transferred, rtd->buffer_size, &offset); + ptr = rtd->dma_area + offset; + n = rtd->buffer_size - offset; + + if (count < n) { + ret = copy_to_user(buf, ptr, count); + } else { + ret = copy_to_user(buf, ptr, n); + ret += copy_to_user(buf + n, rtd->dma_area, count - n); + } + + if (ret) + return count - ret; + return count; +} +EXPORT_SYMBOL(sof_probe_compr_copy); diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h new file mode 100644 index 000000000000..dccc9e008f81 --- /dev/null +++ b/sound/soc/sof/compress.h @@ -0,0 +1,29 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2019-2020 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski <cezary.rojewski(a)intel.com> + */ + +#ifndef __SOF_COMPRESS_H +#define __SOF_COMPRESS_H + +#include <sound/compress_driver.h> + +int sof_probe_compr_open(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int sof_probe_compr_free(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int sof_probe_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai); +int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); +int sof_probe_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai); +int sof_probe_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count); + +#endif diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 1d07450aff77..91acfae7935c 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -14,6 +14,9 @@ #include <sound/sof.h> #include "sof-priv.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "probe.h" +#endif /* see SOF_DBG_ flags */ int sof_core_debug; @@ -292,6 +295,9 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) sdev->pdata = plat_data; sdev->first_boot = true; sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID; +#endif dev_set_drvdata(dev, sdev); /* check all mandatory ops */ diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 7f532bcc8e9d..a771500ac442 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -393,6 +393,49 @@ snd_sof_pcm_platform_pointer(struct snd_sof_dev *sdev, return 0; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +static inline int +snd_sof_probe_compr_assign(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_assign(sdev, cstream, dai); +} + +static inline int +snd_sof_probe_compr_free(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_free(sdev, cstream, dai); +} + +static inline int +snd_sof_probe_compr_set_params(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_set_params(sdev, cstream, params, dai); +} + +static inline int +snd_sof_probe_compr_trigger(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_trigger(sdev, cstream, cmd, dai); +} + +static inline int +snd_sof_probe_compr_pointer(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai) +{ + if (sof_ops(sdev) && sof_ops(sdev)->probe_pointer) + return sof_ops(sdev)->probe_pointer(sdev, cstream, tstamp, dai); + + return 0; +} +#endif + /* machine driver */ static inline int snd_sof_machine_register(struct snd_sof_dev *sdev, void *pdata) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 00084471d0de..5d16f668d16a 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -170,6 +170,27 @@ struct snd_sof_dsp_ops { snd_pcm_uframes_t (*pcm_pointer)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + /* Except for probe_pointer, all probe ops are mandatory */ + int (*probe_assign)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_free)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_set_params)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_trigger)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_pointer)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, + struct snd_soc_dai *dai); /* optional */ +#endif + /* host read DSP stream data */ void (*ipc_msg_data)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, @@ -405,6 +426,10 @@ struct snd_sof_dev { wait_queue_head_t waitq; int code_loading; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + unsigned int extractor_stream_tag; +#endif + /* DMA for Trace */ struct snd_dma_buffer dmatb; struct snd_dma_buffer dmatp; -- 2.20.1

1 0

Applied "ASoC: SOF: Provide probe debugfs support" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: SOF: Provide probe debugfs support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 394695f410c1cd3208906451ba5420f45c420a51 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:23 +0100 Subject: [PATCH] ASoC: SOF: Provide probe debugfs support Define debugfs subdirectory delegated for IPC communication with DSP. Input format: uint,uint,(...) which are later translated into DWORDS sequence and further into instances of struct of interest given the IPC type. For Extractor probes, following have been enabled: - PROBE_POINT_ADD (echo <..> probe_points) - PROBE_POINT_REMOVE (echo <..> probe_points_remove) - PROBE_POINT_INFO (cat probe_points) Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/debug.c | 226 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 226 insertions(+) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index d2b3b99d3a20..b5c0d6cf72cc 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -17,6 +17,221 @@ #include "sof-priv.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "probe.h" + +/** + * strsplit_u32 - Split string into sequence of u32 tokens + * @buf: String to split into tokens. + * @delim: String containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + */ +static int +strsplit_u32(char **buf, const char *delim, u32 **tkns, size_t *num_tkns) +{ + char *s; + u32 *data, *tmp; + size_t count = 0; + size_t cap = 32; + int ret = 0; + + *tkns = NULL; + *num_tkns = 0; + data = kcalloc(cap, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + while ((s = strsep(buf, delim)) != NULL) { + ret = kstrtouint(s, 0, data + count); + if (ret) + goto exit; + if (++count >= cap) { + cap *= 2; + tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL); + if (!tmp) { + ret = -ENOMEM; + goto exit; + } + data = tmp; + } + } + + if (!count) + goto exit; + *tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL); + if (*tkns == NULL) { + ret = -ENOMEM; + goto exit; + } + *num_tkns = count; + +exit: + kfree(data); + return ret; +} + +static int tokenize_input(const char __user *from, size_t count, + loff_t *ppos, u32 **tkns, size_t *num_tkns) +{ + char *buf; + int ret; + + buf = kmalloc(count + 1, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = simple_write_to_buffer(buf, count, ppos, from, count); + if (ret != count) { + ret = ret >= 0 ? -EIO : ret; + goto exit; + } + + buf[count] = '\0'; + ret = strsplit_u32((char **)&buf, ",", tkns, num_tkns); +exit: + kfree(buf); + return ret; +} + +static ssize_t probe_points_read(struct file *file, + char __user *to, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + struct sof_probe_point_desc *desc; + size_t num_desc, len = 0; + char *buf; + int i, ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + buf = kzalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc); + if (ret < 0) + goto exit; + + for (i = 0; i < num_desc; i++) { + ret = snprintf(buf + len, PAGE_SIZE - len, + "Id: %#010x Purpose: %d Node id: %#x\n", + desc[i].buffer_id, desc[i].purpose, desc[i].stream_tag); + if (ret < 0) + goto free_desc; + len += ret; + } + + ret = simple_read_from_buffer(to, count, ppos, buf, len); +free_desc: + kfree(desc); +exit: + kfree(buf); + return ret; +} + +static ssize_t probe_points_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + struct sof_probe_point_desc *desc; + size_t num_tkns, bytes; + u32 *tkns; + int ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + ret = tokenize_input(from, count, ppos, &tkns, &num_tkns); + if (ret < 0) + return ret; + bytes = sizeof(*tkns) * num_tkns; + if (!num_tkns || (bytes % sizeof(*desc))) { + ret = -EINVAL; + goto exit; + } + + desc = (struct sof_probe_point_desc *)tkns; + ret = sof_ipc_probe_points_add(sdev, + desc, bytes / sizeof(*desc)); + if (!ret) + ret = count; +exit: + kfree(tkns); + return ret; +} + +static const struct file_operations probe_points_fops = { + .open = simple_open, + .read = probe_points_read, + .write = probe_points_write, + .llseek = default_llseek, +}; + +static ssize_t probe_points_remove_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + size_t num_tkns; + u32 *tkns; + int ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + ret = tokenize_input(from, count, ppos, &tkns, &num_tkns); + if (ret < 0) + return ret; + if (!num_tkns) { + ret = -EINVAL; + goto exit; + } + + ret = sof_ipc_probe_points_remove(sdev, tkns, num_tkns); + if (!ret) + ret = count; +exit: + kfree(tkns); + return ret; +} + +static const struct file_operations probe_points_remove_fops = { + .open = simple_open, + .write = probe_points_remove_write, + .llseek = default_llseek, +}; + +static int snd_sof_debugfs_probe_item(struct snd_sof_dev *sdev, + const char *name, mode_t mode, + const struct file_operations *fops) +{ + struct snd_sof_dfsentry *dfse; + + dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL); + if (!dfse) + return -ENOMEM; + + dfse->type = SOF_DFSENTRY_TYPE_BUF; + dfse->sdev = sdev; + + debugfs_create_file(name, mode, sdev->debugfs_root, dfse, fops); + /* add to dfsentry list */ + list_add(&dfse->list, &sdev->dfsentry_list); + + return 0; +} +#endif + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) #define MAX_IPC_FLOOD_DURATION_MS 1000 #define MAX_IPC_FLOOD_COUNT 10000 @@ -436,6 +651,17 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev) return err; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + err = snd_sof_debugfs_probe_item(sdev, "probe_points", + 0644, &probe_points_fops); + if (err < 0) + return err; + err = snd_sof_debugfs_probe_item(sdev, "probe_points_remove", + 0200, &probe_points_remove_fops); + if (err < 0) + return err; +#endif + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) /* create read-write ipc_flood_count debugfs entry */ err = snd_sof_debugfs_buf_item(sdev, NULL, 0, -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: Add Probe compress CPU DAIs" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: SOF: Intel: Add Probe compress CPU DAIs has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 70368106467cd8c420176bf3ab0acc797f6584bf Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:24 +0100 Subject: [PATCH] ASoC: SOF: Intel: Add Probe compress CPU DAIs Declare extraction CPU DAI as well as sof_probe_compr_ops. FE DAIs can link against these new CPU DAI to create new compress devices. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/compress.c | 5 +++++ sound/soc/sof/compress.h | 2 ++ sound/soc/sof/intel/hda-dai.c | 28 ++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 6 ++++++ sound/soc/sof/pcm.c | 7 +++++++ 5 files changed, 48 insertions(+) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index e87cc81a0599..7354dc6a49cf 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -13,6 +13,11 @@ #include "ops.h" #include "probe.h" +struct snd_compr_ops sof_probe_compressed_ops = { + .copy = sof_probe_compr_copy, +}; +EXPORT_SYMBOL(sof_probe_compressed_ops); + int sof_probe_compr_open(struct snd_compr_stream *cstream, struct snd_soc_dai *dai) { diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h index dccc9e008f81..800f163603e1 100644 --- a/sound/soc/sof/compress.h +++ b/sound/soc/sof/compress.h @@ -13,6 +13,8 @@ #include <sound/compress_driver.h> +extern struct snd_compr_ops sof_probe_compressed_ops; + int sof_probe_compr_open(struct snd_compr_stream *cstream, struct snd_soc_dai *dai); int sof_probe_compr_free(struct snd_compr_stream *cstream, diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 9c6e3f990ee3..ed5e7d2c0d43 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -399,6 +399,19 @@ static const struct snd_soc_dai_ops hda_link_dai_ops = { .trigger = hda_link_pcm_trigger, .prepare = hda_link_pcm_prepare, }; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +#include "../compress.h" + +static struct snd_soc_cdai_ops sof_probe_compr_ops = { + .startup = sof_probe_compr_open, + .shutdown = sof_probe_compr_free, + .set_params = sof_probe_compr_set_params, + .trigger = sof_probe_compr_trigger, + .pointer = sof_probe_compr_pointer, +}; + +#endif #endif /* @@ -460,5 +473,20 @@ struct snd_soc_dai_driver skl_dai[] = { .name = "Alt Analog CPU DAI", .ops = &hda_link_dai_ops, }, +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +{ + .name = "Probe Extraction CPU DAI", + .compress_new = snd_soc_new_compress, + .cops = &sof_probe_compr_ops, + .capture = { + .stream_name = "Probe Extraction", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + }, +}, +#endif #endif }; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index ca44ecb76534..537c0a930a15 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -349,7 +349,13 @@ /* Number of DAIs */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +#define SOF_SKL_NUM_DAIS 16 +#else #define SOF_SKL_NUM_DAIS 15 +#endif + #else #define SOF_SKL_NUM_DAIS 8 #endif diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index db3df02c7398..b239bbff4b5c 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -16,6 +16,9 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "compress.h" +#endif /* Create DMA buffer page table for DSP */ static int create_page_table(struct snd_soc_component *component, @@ -787,6 +790,10 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev) #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS) pd->compr_ops = &sof_compressed_ops; +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + /* override cops when probe support is enabled */ + pd->compr_ops = &sof_probe_compressed_ops; #endif pd->pcm_construct = sof_pcm_new; pd->ignore_machine = drv_name; -- 2.20.1

1 0

[PATCH v2 1/2] ASoC: rt5682: Add CCF usage for providing I2S clks
by derek.fang＠realtek.com 19 Feb '20

19 Feb '20

From: Derek Fang <derek.fang(a)realtek.com> There is a need to use RT5682 as DAI clock master for other codecs within a platform, which means that the DAI clocks are required to remain, regardless of whether the RT5682 is actually running playback/capture. The RT5682 CCF basic functions are implemented almost by the existing internal functions and asoc apis. It needs a clk provider (rt5682 mclk) to generate the bclk and wclk outputs. The RT5682 CCF supports and restricts as below: 1. Fmt of DAI-AIF1 must be configured to master before using CCF. 2. Only accept a 48MHz clk as the clk provider. 3. Only provide a 48kHz wclk and a set of multiples of wclk as bclk. There are some temporary limitations in this patch until a better implementation. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- include/sound/rt5682.h | 8 + sound/soc/codecs/rt5682.c | 407 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5682.h | 4 +- 3 files changed, 415 insertions(+), 4 deletions(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index bc2c317..6bf0e35 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -24,6 +24,12 @@ enum rt5682_jd_src { RT5682_JD1, }; +enum rt5682_dai_clks { + RT5682_DAI_WCLK_IDX, + RT5682_DAI_BCLK_IDX, + RT5682_DAI_NUM_CLKS, +}; + struct rt5682_platform_data { int ldo1_en; /* GPIO for LDO1_EN */ @@ -32,6 +38,8 @@ struct rt5682_platform_data { enum rt5682_dmic1_clk_pin dmic1_clk_pin; enum rt5682_jd_src jd_src; unsigned int btndet_delay; + + const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; #endif diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 9fbb386..6774813 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -27,6 +27,9 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> #include <sound/rt5682.h> #include "rl6231.h" @@ -45,6 +48,8 @@ static const struct rt5682_platform_data i2s_default_platform_data = { .dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3, .jd_src = RT5682_JD1, .btndet_delay = 16, + .dai_clk_names[RT5682_DAI_WCLK_IDX] = "rt5682-dai-wclk", + .dai_clk_names[RT5682_DAI_BCLK_IDX] = "rt5682-dai-bclk", }; struct rt5682_priv { @@ -58,6 +63,13 @@ struct rt5682_priv { struct mutex calibrate_mutex; bool is_sdw; +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; + struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS]; + struct clk *dai_clks[RT5682_DAI_NUM_CLKS]; + struct clk *mclk; +#endif + int sysclk; int sysclk_src; int lrck[RT5682_AIFS]; @@ -921,6 +933,7 @@ static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = &component->dapm; unsigned int val, count; if (jack_insert) { @@ -963,8 +976,13 @@ static int rt5682_headset_detect(struct snd_soc_component *component, rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); + else + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -1633,6 +1651,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -2459,12 +2478,380 @@ static int rt5682_set_bias_level(struct snd_soc_component *component, return 0; } +#ifdef CONFIG_COMMON_CLK +#define CLK_PLL2_FIN 48000000 +#define CLK_PLL2_FOUT 24576000 +#define CLK_48 48000 + +static bool rt5682_clk_check(struct rt5682_priv *rt5682) +{ + if (!rt5682->master[RT5682_AIF1]) { + dev_err(rt5682->component->dev, "sysclk/dai not set correctly\n"); + return false; + } + return true; +} + +static int rt5682_wclk_prepare(struct clk_hw *hw) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, RT5682_PWR_MB); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + + return 0; +} + +static void rt5682_wclk_unprepare(struct clk_hw *hw) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682_clk_check(rt5682)) + return; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + if (!rt5682->jack_type) + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); +} + +static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + + if (!rt5682_clk_check(rt5682)) + return 0; + /* + * Only accept to set wclk rate to 48kHz temporarily. + */ + return CLK_48; +} + +static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + /* + * Only accept to set wclk rate to 48kHz temporarily. + */ + return CLK_48; +} + +static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct clk *parent_clk; + const char * const clk_name = __clk_get_name(hw->clk); + int pre_div; + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + /* + * Whether the wclk's parent clk (mclk) exists or not, please ensure + * it is fixed or set to 48MHz before setting wclk rate. It's a + * temporary limitation. Only accept 48MHz clk as the clk provider. + * + * It will set the codec anyway by assuming mclk is 48MHz. + */ + parent_clk = clk_get_parent(hw->clk); + if (!parent_clk) + dev_warn(component->dev, + "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n", + CLK_PLL2_FIN); + + if (parent_rate != CLK_PLL2_FIN) + dev_warn(component->dev, "clk %s only support %d Hz input\n", + clk_name, CLK_PLL2_FIN); + + /* + * It's a temporary limitation. Only accept to set wclk rate to 48kHz. + * It will force wclk to 48kHz even it's not. + */ + if (rate != CLK_48) { + dev_warn(component->dev, "clk %s only support %d Hz output\n", + clk_name, CLK_48); + rate = CLK_48; + } + + /* + * To achieve the rate conversion from 48MHz to 48kHz, PLL2 is needed. + */ + rt5682_set_component_pll(component, RT5682_PLL2, RT5682_PLL2_S_MCLK, + CLK_PLL2_FIN, CLK_PLL2_FOUT); + + rt5682_set_component_sysclk(component, RT5682_SCLK_S_PLL2, 0, + CLK_PLL2_FOUT, SND_SOC_CLOCK_IN); + + pre_div = rl6231_get_clk_info(rt5682->sysclk, rate); + + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1, + RT5682_I2S_M_DIV_MASK | RT5682_I2S_CLK_SRC_MASK, + pre_div << RT5682_I2S_M_DIV_SFT | + (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT); + + return 0; +} + +static unsigned long rt5682_bclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + unsigned int bclks_per_wclk; + + snd_soc_component_read(component, RT5682_TDM_TCON_CTRL, + &bclks_per_wclk); + + switch (bclks_per_wclk & RT5682_TDM_BCLK_MS1_MASK) { + case RT5682_TDM_BCLK_MS1_256: + return parent_rate * 256; + case RT5682_TDM_BCLK_MS1_128: + return parent_rate * 128; + case RT5682_TDM_BCLK_MS1_64: + return parent_rate * 64; + case RT5682_TDM_BCLK_MS1_32: + return parent_rate * 32; + default: + return 0; + } +} + +static unsigned long rt5682_bclk_get_factor(unsigned long rate, + unsigned long parent_rate) +{ + unsigned long factor; + + factor = rate / parent_rate; + if (factor < 64) + return 32; + else if (factor < 128) + return 64; + else if (factor < 256) + return 128; + else + return 256; +} + +static long rt5682_bclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + unsigned long factor; + + if (!*parent_rate || !rt5682_clk_check(rt5682)) + return -EINVAL; + + /* + * BCLK rates are set as a multiplier of WCLK in HW. + * We don't allow changing the parent WCLK. We just do + * some rounding down based on the parent WCLK rate + * and find the appropriate multiplier of BCLK to + * get the rounded down BCLK value. + */ + factor = rt5682_bclk_get_factor(rate, *parent_rate); + + return *parent_rate * factor; +} + +static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dai *dai = NULL; + unsigned long factor; + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + factor = rt5682_bclk_get_factor(rate, parent_rate); + + for_each_component_dais(component, dai) + if (dai->id == RT5682_AIF1) + break; + if (!dai) { + dev_err(component->dev, "dai %d not found in component\n", + RT5682_AIF1); + return -ENODEV; + } + + return rt5682_set_bclk1_ratio(dai, factor); +} + +static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = { + [RT5682_DAI_WCLK_IDX] = { + .prepare = rt5682_wclk_prepare, + .unprepare = rt5682_wclk_unprepare, + .recalc_rate = rt5682_wclk_recalc_rate, + .round_rate = rt5682_wclk_round_rate, + .set_rate = rt5682_wclk_set_rate, + }, + [RT5682_DAI_BCLK_IDX] = { + .recalc_rate = rt5682_bclk_recalc_rate, + .round_rate = rt5682_bclk_round_rate, + .set_rate = rt5682_bclk_set_rate, + }, +}; + +static int rt5682_register_dai_clks(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct rt5682_platform_data *pdata = &rt5682->pdata; + struct clk_init_data init; + struct clk *dai_clk; + struct clk_lookup *dai_clk_lookup; + struct clk_hw *dai_clk_hw; + const char *parent_name; + int i, ret; + + for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) { + dai_clk_hw = &rt5682->dai_clks_hw[i]; + + switch (i) { + case RT5682_DAI_WCLK_IDX: + /* Make MCLK the parent of WCLK */ + if (rt5682->mclk) { + parent_name = __clk_get_name(rt5682->mclk); + init.parent_names = &parent_name; + init.num_parents = 1; + } else { + init.parent_names = NULL; + init.num_parents = 0; + } + break; + case RT5682_DAI_BCLK_IDX: + /* Make WCLK the parent of BCLK */ + parent_name = __clk_get_name( + rt5682->dai_clks[RT5682_DAI_WCLK_IDX]); + init.parent_names = &parent_name; + init.num_parents = 1; + break; + default: + dev_err(dev, "Invalid clock index\n"); + ret = -EINVAL; + goto err; + } + + init.name = pdata->dai_clk_names[i]; + init.ops = &rt5682_dai_clk_ops[i]; + init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE; + dai_clk_hw->init = &init; + + dai_clk = devm_clk_register(dev, dai_clk_hw); + if (IS_ERR(dai_clk)) { + dev_warn(dev, "Failed to register %s: %ld\n", + init.name, PTR_ERR(dai_clk)); + ret = PTR_ERR(dai_clk); + goto err; + } + rt5682->dai_clks[i] = dai_clk; + + if (dev->of_node) { + devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, + dai_clk_hw); + } else { + dai_clk_lookup = clkdev_create(dai_clk, init.name, + "%s", dev_name(dev)); + if (!dai_clk_lookup) { + ret = -ENOMEM; + goto err; + } else { + rt5682->dai_clks_lookup[i] = dai_clk_lookup; + } + } + } + + return 0; + +err: + do { + if (rt5682->dai_clks_lookup[i]) + clkdev_drop(rt5682->dai_clks_lookup[i]); + } while (i-- > 0); + + return ret; +} +#endif /* CONFIG_COMMON_CLK */ + static int rt5682_probe(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); +#ifdef CONFIG_COMMON_CLK + int ret; +#endif rt5682->component = component; +#ifdef CONFIG_COMMON_CLK + /* Check if MCLK provided */ + rt5682->mclk = devm_clk_get(component->dev, "mclk"); + if (IS_ERR(rt5682->mclk)) { + if (PTR_ERR(rt5682->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682->mclk); + return ret; + } + rt5682->mclk = NULL; + } + + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(component); + if (ret) + return ret; + + /* Initial setup for CCF */ + rt5682->lrck[RT5682_AIF1] = CLK_48; +#endif + return 0; } @@ -2472,6 +2859,15 @@ static void rt5682_remove(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); +#ifdef CONFIG_COMMON_CLK + int i; + + for (i = RT5682_DAI_NUM_CLKS - 1; i >= 0; --i) { + if (rt5682->dai_clks_lookup[i]) + clkdev_drop(rt5682->dai_clks_lookup[i]); + } +#endif + rt5682_reset(rt5682); } @@ -2606,6 +3002,13 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); + if (device_property_read_string_array(dev, "clock-output-names", + rt5682->pdata.dai_clk_names, + RT5682_DAI_NUM_CLKS) < 0) + dev_warn(dev, "Using default DAI clk names: %s, %s\n", + rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX], + rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]); + return 0; } diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 465c99b..f82126a 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -841,8 +841,8 @@ #define RT5682_TDM_M_LP_INV (0x1 << 1) #define RT5682_TDM_MS_MASK (0x1 << 0) #define RT5682_TDM_MS_SFT 0 -#define RT5682_TDM_MS_M (0x0 << 0) -#define RT5682_TDM_MS_S (0x1 << 0) +#define RT5682_TDM_MS_S (0x0 << 0) +#define RT5682_TDM_MS_M (0x1 << 0) /* Global Clock Control (0x0080) */ #define RT5682_SCLK_SRC_MASK (0x7 << 13) -- 2.7.4

2 2

Applied "ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8b59e642d05f0ae9800b057350c063fe7debd6bc Mon Sep 17 00:00:00 2001 From: Derek Fang <derek.fang(a)realtek.com> Date: Tue, 18 Feb 2020 21:51:52 +0800 Subject: [PATCH] ASoC: rt5682: Add DAI clock binding info for WCLK/BCLK CCF usage This patch describes that rt5682 can expose WCLK and BCLK clocks and how to use. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> Link: https://lore.kernel.org/r/1582033912-6841-2-git-send-email-derek.fang@realt… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/rt5682.txt | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 30e927a28369..ac98151d29e4 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -32,6 +32,12 @@ Optional properties: The delay time is realtek,btndet-delay value multiple of 8.192 ms. If absent, the default is 16. +- #clock-cells : Should be set to '<1>', wclk and bclk sources provided. +- clock-output-names : Name given for DAI clocks output. + +- clocks : phandle and clock specifier for codec MCLK. +- clock-names : Clock name string for 'clocks' attribute, should be "mclk". + Pins on the device (for linking into audio routes) for RT5682: * DMIC L1 @@ -53,4 +59,10 @@ rt5682 { realtek,dmic1-clk-pin = <1>; realtek,jd-src = <1>; realtek,btndet-delay = <16>; + + #clock-cells = <1>; + clock-output-names = "rt5682-dai-wclk", "rt5682-dai-bclk"; + + clocks = <&osc>; + clock-names = "mclk"; }; -- 2.20.1

1 0

Applied "ASoC: soc-pcm: move dai_get_widget()" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: move dai_get_widget() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d9303690f753dfdae51304fc89f4b04c0549a9f7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:27:39 +0900 Subject: [PATCH] ASoC: soc-pcm: move dai_get_widget() This patch moves dai_get_widget() to top side. This is prepare for cleanup soc-pcm.c Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87h7zpbouu.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 65a3856be250..23e36f4f965c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -82,6 +82,15 @@ static int soc_rtd_trigger(struct snd_soc_pcm_runtime *rtd, return 0; } +static inline +struct snd_soc_dapm_widget *dai_get_widget(struct snd_soc_dai *dai, int stream) +{ + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + return dai->playback_widget; + else + return dai->capture_widget; +} + static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, int stream, int action) { @@ -1287,15 +1296,6 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, return NULL; } -static inline struct snd_soc_dapm_widget * - dai_get_widget(struct snd_soc_dai *dai, int stream) -{ - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - return dai->playback_widget; - else - return dai->capture_widget; -} - static int widget_in_list(struct snd_soc_dapm_widget_list *list, struct snd_soc_dapm_widget *widget) { -- 2.20.1

1 0

Applied "ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be()" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 93597fae552a35d27cd1f399ffab6a6862cf9dc3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:27:43 +0900 Subject: [PATCH] ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() dpcm_get_be() has very duplicate code. dpcm_get_be() { ... if (stream == SNDRV_PCM_STREAM_PLAYBACK) { (1) /* code for Playback */ } else { (2) /* code for Capture */ } } The difference between Playback (1) and Capture (2) code is pointer only (= "playback_widget" or "caputre_widget"). OTOH, now we already has dai_get_widget() for it. This means we can merge (1) and (2). This patch do it and remove duplicated code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87ftf9bouq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 41 ++++++++++++----------------------------- 1 file changed, 12 insertions(+), 29 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 23e36f4f965c..b708db972310 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1246,47 +1246,30 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_dapm_widget *widget, int stream) { struct snd_soc_pcm_runtime *be; + struct snd_soc_dapm_widget *w; struct snd_soc_dai *dai; int i; dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - for_each_card_rtds(card, be) { + for_each_card_rtds(card, be) { - if (!be->dai_link->no_pcm) - continue; + if (!be->dai_link->no_pcm) + continue; - dev_dbg(card->dev, "ASoC: try BE : %s\n", - be->cpu_dai->playback_widget ? - be->cpu_dai->playback_widget->name : "(not set)"); + w = dai_get_widget(be->cpu_dai, stream); - if (be->cpu_dai->playback_widget == widget) - return be; + dev_dbg(card->dev, "ASoC: try BE : %s\n", + w ? w->name : "(not set)"); - for_each_rtd_codec_dai(be, i, dai) { - if (dai->playback_widget == widget) - return be; - } - } - } else { - - for_each_card_rtds(card, be) { - - if (!be->dai_link->no_pcm) - continue; + if (w == widget) + return be; - dev_dbg(card->dev, "ASoC: try BE %s\n", - be->cpu_dai->capture_widget ? - be->cpu_dai->capture_widget->name : "(not set)"); + for_each_rtd_codec_dai(be, i, dai) { + w = dai_get_widget(dai, stream); - if (be->cpu_dai->capture_widget == widget) + if (w == widget) return be; - - for_each_rtd_codec_dai(be, i, dai) { - if (dai->capture_widget == widget) - return be; - } } } -- 2.20.1

1 0

Applied "ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be()" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c2cd821603c216a6a7242b2b4c1a093051e26aaf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:27:48 +0900 Subject: [PATCH] ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() dpcm_end_walk_at_be() has very duplicate code. dpcm_end_walk_at_be() { ... if (stream == SNDRV_PCM_STREAM_PLAYBACK) { (1) /* code for Playback */ } else { (2) /* code for Capture */ } } The difference between Playback (1) and Capture (2) code is pointer only (= "playback_widget" or "caputre_widget"). OTOH, now we already has dai_get_widget() for it. This means we can merge (1) and (2). This patch do it and remove duplicated code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87eeutboul.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 37 ++++++++++++++++--------------------- 1 file changed, 16 insertions(+), 21 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b708db972310..7d4419ae63f6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1297,34 +1297,29 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, { struct snd_soc_card *card = widget->dapm->card; struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dapm_widget *w; struct snd_soc_dai *dai; + int stream; int i; - if (dir == SND_SOC_DAPM_DIR_OUT) { - for_each_card_rtds(card, rtd) { - if (!rtd->dai_link->no_pcm) - continue; + /* adjust dir to stream */ + if (dir == SND_SOC_DAPM_DIR_OUT) + stream = SNDRV_PCM_STREAM_PLAYBACK; + else + stream = SNDRV_PCM_STREAM_CAPTURE; - if (rtd->cpu_dai->playback_widget == widget) - return true; + for_each_card_rtds(card, rtd) { + if (!rtd->dai_link->no_pcm) + continue; - for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->playback_widget == widget) - return true; - } - } - } else { /* SND_SOC_DAPM_DIR_IN */ - for_each_card_rtds(card, rtd) { - if (!rtd->dai_link->no_pcm) - continue; + w = dai_get_widget(rtd->cpu_dai, stream); + if (w == widget) + return true; - if (rtd->cpu_dai->capture_widget == widget) + for_each_rtd_codec_dai(rtd, i, dai) { + w = dai_get_widget(dai, stream); + if (w == widget) return true; - - for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->capture_widget == widget) - return true; - } } } -- 2.20.1

1 0

Applied "ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 027a483871832044fa0cb8e9df208cca5230ae91 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:27:53 +0900 Subject: [PATCH] ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() dpcm_end_walk_at_be() and dpcm_get_be() are almost same code. This patch uses dpcm_get_be() from dpcm_end_walk_at_be(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87d0adbouh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 20 +++----------------- 1 file changed, 3 insertions(+), 17 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7d4419ae63f6..1d48be24bfaa 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1297,10 +1297,7 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, { struct snd_soc_card *card = widget->dapm->card; struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dapm_widget *w; - struct snd_soc_dai *dai; int stream; - int i; /* adjust dir to stream */ if (dir == SND_SOC_DAPM_DIR_OUT) @@ -1308,20 +1305,9 @@ static bool dpcm_end_walk_at_be(struct snd_soc_dapm_widget *widget, else stream = SNDRV_PCM_STREAM_CAPTURE; - for_each_card_rtds(card, rtd) { - if (!rtd->dai_link->no_pcm) - continue; - - w = dai_get_widget(rtd->cpu_dai, stream); - if (w == widget) - return true; - - for_each_rtd_codec_dai(rtd, i, dai) { - w = dai_get_widget(dai, stream); - if (w == widget) - return true; - } - } + rtd = dpcm_get_be(card, widget, stream); + if (rtd) + return true; return false; } -- 2.20.1

1 0

Applied "ASoC: soc-pcm: remove soc_dpcm_be_digital_mute()" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c9645d2a952b7925b6708b24242cd5ed04975648 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:27:57 +0900 Subject: [PATCH] ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() No one is using soc_dpcm_be_digital_mute(). If it exists only by assumption that "it may be necessary someday", let's remove it now. Otherwise code maintenance will be difficult. We can revive it when we really needed it. Let's remove it, so far. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87blpxbouc.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-dpcm.h | 1 - sound/soc/soc-pcm.c | 27 --------------------------- 2 files changed, 28 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index b654ebfc8766..665516387671 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -141,7 +141,6 @@ void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream, enum snd_soc_dpcm_state state); /* internal use only */ -int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_runtime_update(struct snd_soc_card *); #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1d48be24bfaa..b8ea4d892031 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2719,33 +2719,6 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) mutex_unlock(&card->mutex); return ret; } -int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) -{ - struct snd_soc_dpcm *dpcm; - struct snd_soc_dai *dai; - - for_each_dpcm_be(fe, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { - - struct snd_soc_pcm_runtime *be = dpcm->be; - int i; - - if (be->dai_link->ignore_suspend) - continue; - - for_each_rtd_codec_dai(be, i, dai) { - struct snd_soc_dai_driver *drv = dai->driver; - - dev_dbg(be->dev, "ASoC: BE digital mute %s\n", - be->dai_link->name); - - if (drv->ops && drv->ops->digital_mute && - dai->playback_active) - drv->ops->digital_mute(dai, mute); - } - } - - return 0; -} static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) { -- 2.20.1

1 0

Applied "ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state()" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 289a7e64f8583aaa45847c7fa3b7fabf8d48fd6b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:04 +0900 Subject: [PATCH] ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() No one is using snd_soc_dpcm_be_get/set_state(). If it exists only by assumption that "it may be necessary someday", let's remove it now. Otherwise code maintenance will be difficult. We can revive it when we really needed it. Let's remove it, so far. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87a75hbou7.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-dpcm.h | 8 -------- sound/soc/soc-pcm.c | 16 ---------------- 2 files changed, 24 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 665516387671..3e7819d2a6aa 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -132,14 +132,6 @@ int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, struct snd_pcm_substream * snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream); -/* get the BE runtime state */ -enum snd_soc_dpcm_state - snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream); - -/* set the BE runtime state */ -void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream, - enum snd_soc_dpcm_state state); - /* internal use only */ int soc_dpcm_runtime_update(struct snd_soc_card *); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b8ea4d892031..bd4e4f86f5b2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2954,22 +2954,6 @@ struct snd_pcm_substream * } EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream); -/* get the BE runtime state */ -enum snd_soc_dpcm_state - snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream) -{ - return be->dpcm[stream].state; -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_get_state); - -/* set the BE runtime state */ -void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, - int stream, enum snd_soc_dpcm_state state) -{ - be->dpcm[stream].state = state; -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_set_state); - /* * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE * are not running, paused or suspended for the specified stream direction. -- 2.20.1

1 0

Applied "ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 085d22be035db245c44714cf879a73eae06c9f6b Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:07 +0900 Subject: [PATCH] ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code Below functions are doing very similar things, the difference is used state only. snd_soc_dpcm_can_be_free_stop() snd_soc_dpcm_can_be_params() This patch adds common snd_soc_dpcm_check_state(), and use it from snd_soc_dpcm_can_be_free_stop() / snd_soc_dpcm_can_be_params(). It can reduce duplicate code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/878sl1bou2.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 70 ++++++++++++++++++++++----------------------- 1 file changed, 35 insertions(+), 35 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index bd4e4f86f5b2..d77a2c22a04f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2954,17 +2954,17 @@ struct snd_pcm_substream * } EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream); -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) +static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, + int stream, + const enum snd_soc_dpcm_state *states, + int num_states) { struct snd_soc_dpcm *dpcm; int state; int ret = 1; unsigned long flags; + int i; spin_lock_irqsave(&fe->card->dpcm_lock, flags); for_each_dpcm_fe(be, stream, dpcm) { @@ -2973,18 +2973,34 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, continue; state = dpcm->fe->dpcm[stream].state; - if (state == SND_SOC_DPCM_STATE_START || - state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND) { - ret = 0; - break; + for (i = 0; i < num_states; i++) { + if (state == states[i]) { + ret = 0; + break; + } } } spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); - /* it's safe to free/stop this BE DAI */ + /* it's safe to do this BE DAI */ return ret; } + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* @@ -2994,30 +3010,14 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { - struct snd_soc_dpcm *dpcm; - int state; - int ret = 1; - unsigned long flags; - - spin_lock_irqsave(&fe->card->dpcm_lock, flags); - for_each_dpcm_fe(be, stream, dpcm) { - - if (dpcm->fe == fe) - continue; + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + SND_SOC_DPCM_STATE_PREPARE, + }; - state = dpcm->fe->dpcm[stream].state; - if (state == SND_SOC_DPCM_STATE_START || - state == SND_SOC_DPCM_STATE_PAUSED || - state == SND_SOC_DPCM_STATE_SUSPEND || - state == SND_SOC_DPCM_STATE_PREPARE) { - ret = 0; - break; - } - } - spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); - - /* it's safe to change hw_params */ - return ret; + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); -- 2.20.1

1 0

Applied "ASoC: soc-pcm: use goto and remove multi return" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: use goto and remove multi return has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From cae06eb92557f0a073835380e57abee5f8173d73 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:11 +0900 Subject: [PATCH] ASoC: soc-pcm: use goto and remove multi return When we use some kind of lock, we need to do unlock. In that time, multi unlock/return is not good implementation. This patch add label and use goto at dpcm_fe_dai_open() to reduce such code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/877e0lboty.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index d77a2c22a04f..5a79a830ee18 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2733,8 +2733,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) ret = dpcm_path_get(fe, stream, &list); if (ret < 0) { - mutex_unlock(&fe->card->mutex); - return ret; + goto open_end; } else if (ret == 0) { dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); @@ -2755,6 +2754,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) dpcm_clear_pending_state(fe, stream); dpcm_path_put(&list); +open_end: mutex_unlock(&fe->card->mutex); return ret; } -- 2.20.1

1 0

Applied "ASoC: soc-pcm: merge playback/cature_active into stream_active" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: merge playback/cature_active into stream_active has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0f6011fd79a2fb92cb80177fd6bdc8aac3a3cd93 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:15 +0900 Subject: [PATCH] ASoC: soc-pcm: merge playback/cature_active into stream_active DAI has playback_active and capture_active to care usage count. OTOH, we have SNDRV_PCM_STREAM_PLAYBACK/CAPTURE. But because of this kind of implementation mismatch, ALSA SoC has many verbose code. To solve this issue, this patch merge playback_active/capture_active into stream_active[2]; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/875zg5botu.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-dai.h | 3 +-- sound/soc/codecs/cs4271.c | 4 ++-- sound/soc/dwc/dwc-i2s.c | 4 ++-- sound/soc/soc-core.c | 17 +++++++++-------- sound/soc/soc-pcm.c | 25 ++++++++++++------------- 5 files changed, 26 insertions(+), 27 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 04c23ac0dfff..7481e468be39 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -322,8 +322,7 @@ struct snd_soc_dai { struct snd_soc_dai_driver *driver; /* DAI runtime info */ - unsigned int capture_active; /* stream usage count */ - unsigned int playback_active; /* stream usage count */ + unsigned int stream_active[SNDRV_PCM_STREAM_LAST + 1]; /* usage count */ unsigned int active; diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 04b86a51e055..62f412d6f9f2 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -356,9 +356,9 @@ static int cs4271_hw_params(struct snd_pcm_substream *substream, */ if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - !dai->capture_active) || + !dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) || (substream->stream == SNDRV_PCM_STREAM_CAPTURE && - !dai->playback_active)) { + !dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK])) { ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN, CS4271_MODE2_PDN); diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 7eeca2150b2d..a8bff6f08a69 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -427,9 +427,9 @@ static int dw_i2s_resume(struct snd_soc_component *component) clk_enable(dev->clk); for_each_component_dais(component, dai) { - if (dai->playback_active) + if (dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]) dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK); - if (dai->capture_active) + if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6a58a8f6e3c4..f0ae1a7d7e09 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -365,19 +365,20 @@ EXPORT_SYMBOL_GPL(snd_soc_get_pcm_runtime); void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *codec_dai = rtd->codec_dai; + int playback = SNDRV_PCM_STREAM_PLAYBACK; mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, - codec_dai->playback_active ? "active" : "inactive", + codec_dai->stream_active[playback] ? "active" : "inactive", rtd->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ if (rtd->pop_wait == 1) { rtd->pop_wait = 0; - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + snd_soc_dapm_stream_event(rtd, playback, SND_SOC_DAPM_STREAM_STOP); } @@ -514,6 +515,7 @@ int snd_soc_suspend(struct device *dev) struct snd_soc_card *card = dev_get_drvdata(dev); struct snd_soc_component *component; struct snd_soc_pcm_runtime *rtd; + int playback = SNDRV_PCM_STREAM_PLAYBACK; int i; /* If the card is not initialized yet there is nothing to do */ @@ -537,9 +539,8 @@ int snd_soc_suspend(struct device *dev) continue; for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->playback_active) - snd_soc_dai_digital_mute(dai, 1, - SNDRV_PCM_STREAM_PLAYBACK); + if (dai->stream_active[playback]) + snd_soc_dai_digital_mute(dai, 1, playback); } } @@ -680,14 +681,14 @@ static void soc_resume_deferred(struct work_struct *work) /* unmute any active DACs */ for_each_card_rtds(card, rtd) { struct snd_soc_dai *dai; + int playback = SNDRV_PCM_STREAM_PLAYBACK; if (rtd->dai_link->ignore_suspend) continue; for_each_rtd_codec_dai(rtd, i, dai) { - if (dai->playback_active) - snd_soc_dai_digital_mute(dai, 0, - SNDRV_PCM_STREAM_PLAYBACK); + if (dai->stream_active[playback]) + snd_soc_dai_digital_mute(dai, 0, playback); } } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5a79a830ee18..6fd69574ca31 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -100,15 +100,9 @@ static void snd_soc_runtime_action(struct snd_soc_pcm_runtime *rtd, lockdep_assert_held(&rtd->card->pcm_mutex); - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active += action; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->playback_active += action; - } else { - cpu_dai->capture_active += action; - for_each_rtd_codec_dai(rtd, i, codec_dai) - codec_dai->capture_active += action; - } + cpu_dai->stream_active[stream] += action; + for_each_rtd_codec_dai(rtd, i, codec_dai) + codec_dai->stream_active[stream] += action; cpu_dai->active += action; cpu_dai->component->active += action; @@ -967,8 +961,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* apply codec digital mute */ for_each_rtd_codec_dai(rtd, i, codec_dai) { - if ((playback && codec_dai->playback_active == 1) || - (!playback && codec_dai->capture_active == 1)) + int playback_active = codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]; + int capture_active = codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]; + + if ((playback && playback_active == 1) || + (!playback && capture_active == 1)) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); } @@ -2634,7 +2631,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) goto capture; /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->playback_active || !fe->codec_dai->playback_active) + if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] || + !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]) goto capture; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); @@ -2665,7 +2663,8 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) return 0; /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->capture_active || !fe->codec_dai->capture_active) + if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] || + !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) return 0; paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); -- 2.20.1

1 0

Applied "ALSA: pcm.h: add for_each_pcm_streams()" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ALSA: pcm.h: add for_each_pcm_streams() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3193abd26b515ccac65e1c323533cb7f53d06176 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:19 +0900 Subject: [PATCH] ALSA: pcm.h: add for_each_pcm_streams() ALSA code has SNDRV_PCM_STREAM_PLAYBACK/CAPTURE everywhere. Having for_each_xxxx macro is useful. This patch adds for_each_pcm_streams() for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/874kvpbotq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/pcm.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index f657ff08f317..2628246b76fa 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -644,6 +644,11 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, #define snd_pcm_group_for_each_entry(s, substream) \ list_for_each_entry(s, &substream->group->substreams, link_list) +#define for_each_pcm_streams(stream) \ + for (stream = SNDRV_PCM_STREAM_PLAYBACK; \ + stream <= SNDRV_PCM_STREAM_LAST; \ + stream++) + /** * snd_pcm_running - Check whether the substream is in a running state * @substream: substream to check -- 2.20.1

1 0

Applied "ASoC: soc-core: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-core: use for_each_pcm_streams() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d74c2a156b710e9ad81193a60e037430f8894c0c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:25 +0900 Subject: [PATCH] ASoC: soc-core: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/8736b9botk.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 31 +++++++++++++------------------ 1 file changed, 13 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f0ae1a7d7e09..30c17fde14ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -432,6 +432,7 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_component *component; struct device *dev; int ret; + int stream; /* * for rtd->dev @@ -466,10 +467,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( rtd->dev = dev; INIT_LIST_HEAD(&rtd->list); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].be_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_PLAYBACK].fe_clients); - INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); + for_each_pcm_streams(stream) { + INIT_LIST_HEAD(&rtd->dpcm[stream].be_clients); + INIT_LIST_HEAD(&rtd->dpcm[stream].fe_clients); + } dev_set_drvdata(dev, rtd); INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); @@ -559,17 +560,14 @@ int snd_soc_suspend(struct device *dev) snd_soc_flush_all_delayed_work(card); for_each_card_rtds(card, rtd) { + int stream; if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_SUSPEND); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_SUSPEND); + for_each_pcm_streams(stream) + snd_soc_dapm_stream_event(rtd, stream, + SND_SOC_DAPM_STREAM_SUSPEND); } /* Recheck all endpoints too, their state is affected by suspend */ @@ -665,17 +663,14 @@ static void soc_resume_deferred(struct work_struct *work) } for_each_card_rtds(card, rtd) { + int stream; if (rtd->dai_link->ignore_suspend) continue; - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_RESUME); - - snd_soc_dapm_stream_event(rtd, - SNDRV_PCM_STREAM_CAPTURE, - SND_SOC_DAPM_STREAM_RESUME); + for_each_pcm_streams(stream) + snd_soc_dapm_stream_event(rtd, stream, + SND_SOC_DAPM_STREAM_RESUME); } /* unmute any active DACs */ -- 2.20.1

1 0

Applied "ASoC: soc-pcm: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-pcm: use for_each_pcm_streams() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 7083f877ea66e106f90e9a1a0dabb19ebbacc4e6 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:28 +0900 Subject: [PATCH] ASoC: soc-pcm: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/871rqtboth.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 97 ++++++++++++++++----------------------------- 1 file changed, 35 insertions(+), 62 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6fd69574ca31..63f67eb7c077 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2612,6 +2612,7 @@ static int dpcm_run_old_update(struct snd_soc_pcm_runtime *fe, int stream) static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) { struct snd_soc_dapm_widget_list *list; + int stream; int count, paths; if (!fe->dai_link->dynamic) @@ -2625,69 +2626,42 @@ static int soc_dpcm_fe_runtime_update(struct snd_soc_pcm_runtime *fe, int new) dev_dbg(fe->dev, "ASoC: DPCM %s runtime update for FE %s\n", new ? "new" : "old", fe->dai_link->name); - /* skip if FE doesn't have playback capability */ - if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK) || - !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_PLAYBACK)) - goto capture; - - /* skip if FE isn't currently playing */ - if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] || - !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]) - goto capture; - - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "playback"); - return paths; - } - - /* update any playback paths */ - count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_PLAYBACK, &list, new); - if (count) { - if (new) - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_PLAYBACK); - else - dpcm_run_old_update(fe, SNDRV_PCM_STREAM_PLAYBACK); + for_each_pcm_streams(stream) { - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_PLAYBACK); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_PLAYBACK); - } - - dpcm_path_put(&list); + /* skip if FE doesn't have playback/capture capability */ + if (!snd_soc_dai_stream_valid(fe->cpu_dai, stream) || + !snd_soc_dai_stream_valid(fe->codec_dai, stream)) + continue; -capture: - /* skip if FE doesn't have capture capability */ - if (!snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE) || - !snd_soc_dai_stream_valid(fe->codec_dai, SNDRV_PCM_STREAM_CAPTURE)) - return 0; + /* skip if FE isn't currently playing/capturing */ + if (!fe->cpu_dai->stream_active[stream] || + !fe->codec_dai->stream_active[stream]) + continue; - /* skip if FE isn't currently capturing */ - if (!fe->cpu_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE] || - !fe->codec_dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) - return 0; + paths = dpcm_path_get(fe, stream, &list); + if (paths < 0) { + dev_warn(fe->dev, "ASoC: %s no valid %s path\n", + fe->dai_link->name, + stream == SNDRV_PCM_STREAM_PLAYBACK ? + "playback" : "capture"); + return paths; + } - paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); - if (paths < 0) { - dev_warn(fe->dev, "ASoC: %s no valid %s path\n", - fe->dai_link->name, "capture"); - return paths; - } + /* update any playback/capture paths */ + count = dpcm_process_paths(fe, stream, &list, new); + if (count) { + if (new) + dpcm_run_new_update(fe, stream); + else + dpcm_run_old_update(fe, stream); - /* update any old capture paths */ - count = dpcm_process_paths(fe, SNDRV_PCM_STREAM_CAPTURE, &list, new); - if (count) { - if (new) - dpcm_run_new_update(fe, SNDRV_PCM_STREAM_CAPTURE); - else - dpcm_run_old_update(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_clear_pending_state(fe, stream); + dpcm_be_disconnect(fe, stream); + } - dpcm_clear_pending_state(fe, SNDRV_PCM_STREAM_CAPTURE); - dpcm_be_disconnect(fe, SNDRV_PCM_STREAM_CAPTURE); + dpcm_path_put(&list); } - dpcm_path_put(&list); - return 0; } @@ -3114,19 +3088,18 @@ static ssize_t dpcm_state_read_file(struct file *file, char __user *user_buf, { struct snd_soc_pcm_runtime *fe = file->private_data; ssize_t out_count = PAGE_SIZE, offset = 0, ret = 0; + int stream; char *buf; buf = kmalloc(out_count, GFP_KERNEL); if (!buf) return -ENOMEM; - if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) - offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_PLAYBACK, - buf + offset, out_count - offset); - - if (snd_soc_dai_stream_valid(fe->cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) - offset += dpcm_show_state(fe, SNDRV_PCM_STREAM_CAPTURE, - buf + offset, out_count - offset); + for_each_pcm_streams(stream) + if (snd_soc_dai_stream_valid(fe->cpu_dai, stream)) + offset += dpcm_show_state(fe, stream, + buf + offset, + out_count - offset); ret = simple_read_from_buffer(user_buf, count, ppos, buf, offset); -- 2.20.1

1 0

Applied "ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ee10fbe1cdf7cb4ae62f5e23ccd771e696b8f404 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:32 +0900 Subject: [PATCH] ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87zhdhaa8x.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 2cc25651661c..d6b4831e8aec 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -237,7 +237,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, max_buffer_size = SIZE_MAX; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + for_each_pcm_streams(i) { substream = rtd->pcm->streams[i].substream; if (!substream) continue; @@ -371,8 +371,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, dev = config->dma_dev; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; - i++) { + for_each_pcm_streams(i) { if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) name = "rx-tx"; else @@ -401,8 +400,7 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) { unsigned int i; - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; - i++) { + for_each_pcm_streams(i) { if (!pcm->chan[i]) continue; dma_release_channel(pcm->chan[i]); -- 2.20.1

1 0

Applied "ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0a170be9631ea8335e494f3c5f7ab720287023a2 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:36 +0900 Subject: [PATCH] ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87y2t1aa8t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index a8bff6f08a69..515f88456dbd 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -422,15 +422,15 @@ static int dw_i2s_resume(struct snd_soc_component *component) { struct dw_i2s_dev *dev = snd_soc_component_get_drvdata(component); struct snd_soc_dai *dai; + int stream; if (dev->capability & DW_I2S_MASTER) clk_enable(dev->clk); for_each_component_dais(component, dai) { - if (dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK]) - dw_i2s_config(dev, SNDRV_PCM_STREAM_PLAYBACK); - if (dai->stream_active[SNDRV_PCM_STREAM_CAPTURE]) - dw_i2s_config(dev, SNDRV_PCM_STREAM_CAPTURE); + for_each_pcm_streams(stream) + if (dai->stream_active[stream]) + dw_i2s_config(dev, stream); } return 0; -- 2.20.1

1 0

Applied "ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fa7b2a1fcb92906a284b0824b45866a7b8afb599 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:40 +0900 Subject: [PATCH] ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87wo8laa8p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_asrc_dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index ece130f59d15..44e5924be870 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -400,7 +400,7 @@ static int fsl_asrc_dma_pcm_new(struct snd_soc_component *component, return ret; } - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + for_each_pcm_streams(i) { substream = pcm->streams[i].substream; if (!substream) continue; @@ -428,7 +428,7 @@ static void fsl_asrc_dma_pcm_free(struct snd_soc_component *component, struct snd_pcm_substream *substream; int i; - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_LAST; i++) { + for_each_pcm_streams(i) { substream = pcm->streams[i].substream; if (!substream) continue; -- 2.20.1

1 0

Applied "ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4c260c3f19bd16e6b11841aad1162f5a105ed24e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:44 +0900 Subject: [PATCH] ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87v9o5aa8m.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/lpass-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index b05091c283b7..5d1bc5757169 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -529,7 +529,7 @@ static void lpass_platform_pcm_free(struct snd_soc_component *component, struct snd_pcm_substream *substream; int i; - for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) { + for_each_pcm_streams(i) { substream = pcm->streams[i].substream; if (substream) { snd_dma_free_pages(&substream->dma_buffer); -- 2.20.1

1 0

Applied "ASoC: sof: sof-audio: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: sof: sof-audio: use for_each_pcm_streams() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 525c4107da8c0a86aa3548dc6e1d0014749e95f7 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:47 +0900 Subject: [PATCH] ASoC: sof: sof-audio: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87tv3paa8i.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/sof-audio.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 75f2ef2bd94b..fc4ed2a8a914 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -23,7 +23,7 @@ bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev) int dir; list_for_each_entry(spcm, &sdev->pcm_list, list) { - for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + for_each_pcm_streams(dir) { substream = spcm->stream[dir].substream; if (!substream || !substream->runtime) continue; @@ -71,7 +71,7 @@ int sof_set_hw_params_upon_resume(struct device *dev) * have been suspended. */ list_for_each_entry(spcm, &sdev->pcm_list, list) { - for (dir = 0; dir <= SNDRV_PCM_STREAM_CAPTURE; dir++) { + for_each_pcm_streams(dir) { /* * do not reset hw_params upon resume for streams that * were kept running during suspend @@ -319,16 +319,11 @@ struct snd_sof_pcm *snd_sof_find_spcm_comp(struct snd_soc_component *scomp, int dir; list_for_each_entry(spcm, &sdev->pcm_list, list) { - dir = SNDRV_PCM_STREAM_PLAYBACK; - if (spcm->stream[dir].comp_id == comp_id) { - *direction = dir; - return spcm; - } - - dir = SNDRV_PCM_STREAM_CAPTURE; - if (spcm->stream[dir].comp_id == comp_id) { - *direction = dir; - return spcm; + for_each_pcm_streams(dir) { + if (spcm->stream[dir].comp_id == comp_id) { + *direction = dir; + return spcm; + } } } -- 2.20.1

1 0

Applied "ALSA: usx2y: use for_each_pcm_streams() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ALSA: usx2y: use for_each_pcm_streams() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ffd11d1e7ad4602d5d1c2b4517ad316f7587d4d9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 17 Feb 2020 17:28:51 +0900 Subject: [PATCH] ALSA: usx2y: use for_each_pcm_streams() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87sgj9aa8e.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/usb/usx2y/usbusx2yaudio.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 772f6f3ccbb1..37d290fe9d43 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -906,11 +906,12 @@ static const struct snd_pcm_ops snd_usX2Y_pcm_ops = */ static void usX2Y_audio_stream_free(struct snd_usX2Y_substream **usX2Y_substream) { - kfree(usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]); - usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK] = NULL; + int stream; - kfree(usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]); - usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE] = NULL; + for_each_pcm_streams(stream) { + kfree(usX2Y_substream[stream]); + usX2Y_substream[stream] = NULL; + } } static void snd_usX2Y_pcm_private_free(struct snd_pcm *pcm) -- 2.20.1

1 0

[PATCH v2 0/2] INtroduce the TLV320ADCx140 codec family
by Dan Murphy 18 Feb '20

18 Feb '20

Hello Introducing the Texas Instruments TLV320ADCx140 quad channel audio ADC. This device supports 4 analog inputs, 8 digital inputs or a combination of analog and digital microphones. TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140 Dan Murphy (2): dt-bindings: sound: Add TLV320ADCx140 dt bindings ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family .../bindings/sound/tlv320adcx140.yaml | 83 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adcx140.c | 851 ++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 130 +++ 5 files changed, 1075 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h -- 2.25.0

2 6

[alsa-devel] [PATCH RFC] ASoC: core: ensure component names are unique
by Jerome Brunet 18 Feb '20

18 Feb '20

Make sure each ASoC component is registered with a unique name. The component is derived from the device name. If a device registers more than one component, the component names will be the same. This usually brings up a warning about the debugfs directory creation of the component since directory already exists. In such case, start numbering the component of the device so the names don't collide anymore. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- * Here is an example of /sys/kernel/debug/asoc/components before the change: hdmi-audio-codec.3.auto c1105400.audio-controller c1105400.audio-controller c1105400.audio-controller c8832000.audio-controller analog-amplifier snd-soc-dummy snd-soc-dummy * and after: hdmi-audio-codec.3.auto c1105400.audio-controller-2 c1105400.audio-controller-1 c1105400.audio-controller c8832000.audio-controller analog-amplifier snd-soc-dummy-1 snd-soc-dummy I wanted to keep the name as they were for the devices registering only one component but we could also start the numbering from the first component instead of the second. sound/soc/soc-core.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 03b87427faa7..6a58a8f6e3c4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2446,6 +2446,33 @@ static int snd_soc_register_dais(struct snd_soc_component *component, return ret; } +static char *snd_soc_component_unique_name(struct device *dev, + struct snd_soc_component *component) +{ + struct snd_soc_component *pos; + int count = 0; + char *name, *unique; + + name = fmt_single_name(dev, &component->id); + if (!name) + return name; + + /* Count the number of components registred by the device */ + for_each_component(pos) { + if (dev == pos->dev) + count++; + } + + /* Keep naming as it is for the 1st component */ + if (!count) + return name; + + unique = devm_kasprintf(dev, GFP_KERNEL, "%s-%d", name, count); + devm_kfree(dev, name); + + return unique; +} + static int snd_soc_component_initialize(struct snd_soc_component *component, const struct snd_soc_component_driver *driver, struct device *dev) { @@ -2454,7 +2481,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->card_list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); + component->name = snd_soc_component_unique_name(dev, component); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.24.1

5 11

[RESEND PATCH] ASoC: tas2562: Return invalid for when bitwidth is invalid
by Dan Murphy 18 Feb '20

18 Feb '20

If the bitwidth passed in to the set_bitwidth function is not supported then return an error. Fixes: 29b74236bd57 ("ASoC: tas2562: Introduce the TAS2562 amplifier") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 729acd874c48..5b4803a76719 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -215,7 +215,8 @@ static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) break; default: - dev_info(tas2562->dev, "Not supported params format\n"); + dev_info(tas2562->dev, "Unsupported bitwidth format\n"); + return -EINVAL; } ret = snd_soc_component_update_bits(tas2562->component, -- 2.25.0

1 0

[PATCH] ASoC: tas2562: Return invalid for when bitwidth is invalid
by Dan Murphy 18 Feb '20

18 Feb '20

If the bitwidth passed in to the set_bitwidth function is not supported then return an error. 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z=SaGP7KbI}*Z(Za@V3a;5Q)k6DEa?D<jYH5(wCq^dnU3P`QEmCHdz~;Zzea&OZ`8h zfaadmm-|Ta|JfK8{*&`l>dA4uAG#zW_2qt3(;R?EJ4}-OmHLv>d#hDH``%Mws>phz z0jVeXVPpuC^{e9ju2kr!MEj+@Dl>x8`#p&W@ocHom+vpKK$^2sIrpRt(OO|pzQ53Y tM#ClBFXJc8Z`pQav#H`*GfQVTtx47^F<M^Z>CdN)DWp>-$^Pc+|2IHPD@*_Y literal 0 HcmV?d00001 diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 729acd874c48..5b4803a76719 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -215,7 +215,8 @@ static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) break; default: - dev_info(tas2562->dev, "Not supported params format\n"); + dev_info(tas2562->dev, "Unsupported bitwidth format\n"); + return -EINVAL; } ret = snd_soc_component_update_bits(tas2562->component, -- 2.25.0

2 2

[asoc:for-5.7 21/76] wcd9335.c:undefined reference to `slim_stream_prepare'
by kbuild test robot 18 Feb '20

18 Feb '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.7 head: 3cd23f021e2e5f3350125abcb39f12430df87d06 commit: ea00d95200d02ece71f5814d41b14f2eb16d598b [21/76] ASoC: Use imply for SND_SOC_ALL_CODECS config: x86_64-randconfig-s2-20200217 (attached as .config) compiler: gcc-7 (Debian 7.5.0-5) 7.5.0 reproduce: git checkout ea00d95200d02ece71f5814d41b14f2eb16d598b # save the attached .config to linux build tree make ARCH=x86_64 If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): ld: sound/soc/codecs/ab8500-codec.o: in function `ab8500_codec_read_reg': ab8500-codec.c:(.text+0x479): undefined reference to `abx500_get_register_interruptible' ld: sound/soc/codecs/ab8500-codec.o: in function `ab8500_codec_probe': ab8500-codec.c:(.text+0xdf1): undefined reference to `abx500_get_register_interruptible' ld: ab8500-codec.c:(.text+0xe18): undefined reference to `abx500_set_register_interruptible' ld: sound/soc/codecs/ab8500-codec.o: in function `ab8500_codec_write_reg': ab8500-codec.c:(.text+0x4b1): undefined reference to `abx500_set_register_interruptible' ld: sound/soc/codecs/mc13783.o: in function `mc13783_probe': mc13783.c:(.text+0x24c): undefined reference to `mc13xxx_reg_write' ld: mc13783.c:(.text+0x25e): undefined reference to `mc13xxx_reg_write' ld: mc13783.c:(.text+0x270): undefined reference to `mc13xxx_reg_write' ld: mc13783.c:(.text+0x282): undefined reference to `mc13xxx_reg_write' ld: mc13783.c:(.text+0x294): undefined reference to `mc13xxx_reg_write' ld: sound/soc/codecs/mc13783.o:mc13783.c:(.text+0x2a6): more undefined references to `mc13xxx_reg_write' follow ld: sound/soc/codecs/mc13783.o: in function `mc13783_probe': mc13783.c:(.text+0x2c5): undefined reference to `mc13xxx_reg_rmw' ld: mc13783.c:(.text+0x2e4): undefined reference to `mc13xxx_reg_rmw' ld: sound/soc/codecs/mc13783.o: in function `mc13783_remove': mc13783.c:(.text+0x212): undefined reference to `mc13xxx_reg_rmw' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_trigger': >> wcd9335.c:(.text+0x323): undefined reference to `slim_stream_prepare' >> ld: wcd9335.c:(.text+0x32f): undefined reference to `slim_stream_enable' >> ld: wcd9335.c:(.text+0x345): undefined reference to `slim_stream_unprepare' >> ld: wcd9335.c:(.text+0x351): undefined reference to `slim_stream_disable' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_slim_status': >> wcd9335.c:(.text+0x1970): undefined reference to `of_slim_get_device' >> ld: wcd9335.c:(.text+0x1995): undefined reference to `slim_get_logical_addr' >> ld: wcd9335.c:(.text+0x19b2): undefined reference to `__regmap_init_slimbus' ld: wcd9335.c:(.text+0x19f3): undefined reference to `__regmap_init_slimbus' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_hw_params': >> wcd9335.c:(.text+0x348f): undefined reference to `slim_stream_allocate' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_slim_driver_init': >> wcd9335.c:(.init.text+0x3c): undefined reference to `__slim_driver_register' ld: sound/soc/codecs/wcd9335.o: in function `wcd9335_slim_driver_exit': >> wcd9335.c:(.exit.text+0x8): undefined reference to `slim_driver_unregister' ld: sound/soc/codecs/wcd934x.o: in function `wcd934x_trigger': >> wcd934x.c:(.text+0x30d): undefined reference to `slim_stream_prepare' >> ld: wcd934x.c:(.text+0x319): undefined reference to `slim_stream_enable' >> ld: wcd934x.c:(.text+0x32f): undefined reference to `slim_stream_unprepare' >> ld: wcd934x.c:(.text+0x33b): undefined reference to `slim_stream_disable' ld: sound/soc/codecs/wcd934x.o: in function `wcd934x_hw_params': >> wcd934x.c:(.text+0x2620): undefined reference to `slim_stream_allocate' ld: sound/soc/codecs/wcd934x.o: in function `wcd934x_codec_probe': >> wcd934x.c:(.text+0x27ee): undefined reference to `of_slim_get_device' >> ld: wcd934x.c:(.text+0x2818): undefined reference to `slim_get_logical_addr' >> ld: wcd934x.c:(.text+0x2836): undefined reference to `__regmap_init_slimbus' --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

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[alsa-devel] [PATCH v4 00/18] ASoC: soc-pcm cleanup step3
by Kuninori Morimoto 18 Feb '20

18 Feb '20

Hi Mark These are v4 of soc-pcm cleanup step3. Main change from v3 is that - tidyup Subject on [10/18] patch - added Takashi's Reviewed-by / Acked-by on [10/18], [18/18] Kuninori Morimoto (18): 1) ASoC: soc-pcm: move dai_get_widget() 2) ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() 3) ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() 4) ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() 5) ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() 6) ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() 7) ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code 8) ASoC: soc-pcm: use goto and remove multi return 9) ASoC: soc-pcm: merge playback/cature_active into stream_active 10) ASoC: soc.h: add for_each_pcm_streams() 11) ASoC: soc-core: use for_each_pcm_streams() macro 12) ASoC: soc-pcm: use for_each_pcm_streams() macro 13) ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro 14) ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro 15) ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro 16) ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro 17) ASoC: sof: sof-audio: use for_each_pcm_streams() macro 18) ALSA: usx2y: use for_each_pcm_streams() macro include/sound/pcm.h | 5 + include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 - sound/soc/codecs/cs4271.c | 4 +- sound/soc/dwc/dwc-i2s.c | 8 +- sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/soc-core.c | 48 ++-- sound/soc/soc-generic-dmaengine-pcm.c | 8 +- sound/soc/soc-pcm.c | 327 +++++++++----------------- sound/soc/sof/sof-audio.c | 19 +- sound/usb/usx2y/usbusx2yaudio.c | 9 +- 12 files changed, 162 insertions(+), 284 deletions(-) -- 2.17.1

2 19

[alsa-devel] [PATCH v4 00/11] ASoC: SOF: Data probing
by Cezary Rojewski 18 Feb '20

18 Feb '20

This set of patches achieves few goals in order to enable data probing feature for audio DSP. First, provide new and alter existing interfaces (page allocation, runtime flow adjustments) to make them compress friendly. For HDA part, work has been done to account for compress streams when servicing IRQs, setting up BDLs and assigning DMAs. Finally, the end goal which are the probe APIs and usage itself. Probes can be treated as endpoints which allow for data extraction from or injection to target module - a great ally when debugging problematic audio issues such as distortions, glitches or gaps. Compress streams are a weapon of choice here to provide a lightweight implementation. While all available IPCs have been defined, current implementation covers extraction only, with injection scheduled for a later date. Initial review and development of probes can be found under: https://github.com/thesofproject/linux/pull/1276 with the hda-compress-enable set of patches being separated and reviewed on: https://github.com/thesofproject/linux/pull/1571 Tested on CML-U with rt5682 i2s board. Changes in v4: - No changes to ALSA core and hda patches - 'probe' source added in patch "ASoC: SOF: Implement Probe IPC API" now only compiled if SND_SOC_SOF_DEBUG_PROBES is enabled. Said kconfig initial declaration has been moved to that patch too - 'extractor_stream_tag' field now decorated with _DEBUG_PROBES to account for the change above - Debugfs probe handlers now check if extractor stream is running before continuing. To achieve this, 'extractor_stream_tag' is both initialized and set to SOF_PROBE_INVALID_NODE_ID on sof_probe_compr_free(). Suggestion provided by Pierre - Reverted SND_SOC_SOF_HDA_LINK requirement-removal change for HDA probes. Code requires SND_HDA_EXT_CORE in order to function on Intel platforms - Relocated probe-compress related code from /sof/pcm.c directly to /sof/compress.c as requested by Daniel Changes in v3: - No changes to ALSA core and hda patches - Addressed sparse and all doc related warnings as reported by Pierre - Moved _DEBUG_PROBES config outside of _DEVELOPER_SUPPORT block - Renamed 'extractor' field to 'extractor_stream_tag' as requested - Relocated 'extractor_stream_tag' declaration from patch 07 to 08 Changes in v2: - No changes to ALSA core and hda patches - Removed "ASoC: Intel: sof_rt5682: Add compress probe DAI links" from the patchset list as requested by Pierre - updated copyright header for newly added files (dates) - probes over HDA no longer require SND_SOC_SOF_HDA_LINK enabled - renamed debugfs probe functions as requested by Pierre - probe IPC API has been updated to align with newest SOF FW & probe debug app (struct_size macro has been enlisted to make the size calculations transparent). This targets only "ASoC: SOF: Implement Probe IPC API" patch Cezary Rojewski (11): ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: SOF: Intel: Account for compress streams when servicing IRQs ASoC: SOF: Implement Probe IPC API ASoC: SOF: Generic probe compress operations ASoC: SOF: Intel: Probe compress operations ASoC: SOF: Provide probe debugfs support ASoC: SOF: Intel: Add Probe compress CPU DAIs include/sound/compress_driver.h | 40 +++- include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 2 + include/sound/sof/header.h | 11 ++ sound/core/compress_offload.c | 42 +++++ sound/hda/ext/hdac_ext_stream.c | 49 ++++- sound/hda/hdac_controller.c | 4 +- sound/hda/hdac_stream.c | 52 ++++-- sound/soc/sof/Kconfig | 9 + sound/soc/sof/Makefile | 1 + sound/soc/sof/compress.c | 146 +++++++++++++++ sound/soc/sof/compress.h | 31 ++++ sound/soc/sof/core.c | 6 + sound/soc/sof/debug.c | 226 +++++++++++++++++++++++ sound/soc/sof/intel/Kconfig | 10 + sound/soc/sof/intel/Makefile | 1 + sound/soc/sof/intel/apl.c | 9 + sound/soc/sof/intel/cnl.c | 9 + sound/soc/sof/intel/hda-compress.c | 132 +++++++++++++ sound/soc/sof/intel/hda-dai.c | 28 +++ sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/intel/hda-stream.c | 26 ++- sound/soc/sof/intel/hda.h | 30 +++ sound/soc/sof/ops.h | 43 +++++ sound/soc/sof/pcm.c | 7 + sound/soc/sof/probe.c | 286 +++++++++++++++++++++++++++++ sound/soc/sof/probe.h | 85 +++++++++ sound/soc/sof/sof-priv.h | 25 +++ 28 files changed, 1277 insertions(+), 39 deletions(-) create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h create mode 100644 sound/soc/sof/intel/hda-compress.c create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h -- 2.17.1

2 15

Questions about event implementation in ALSA Sequencer
by Takashi Sakamoto 18 Feb '20

18 Feb '20

Hi, I'm working for alsa-gobject[1] so that it supports API for ALSA Sequencer. At present I've mostly done it just with direct dispatch support[2] (thus transmission via queue is for my later work). Then I have some questions about the design of event in ALSA Sequencer. I'd like to ask for some advices (mainly Iwai-san, perhaps). 1. use case of SNDRV_SEQ_EVENT_LENGTH_VARUSR in 'struct snd_seq_event' In my understanding, the flag is used for a case that sender transmits the value of pointer itself and its length to the receiver in the shape of 'struct snd_seq_ev_ext'. I assume that two use cases. If the sender and receiver are in the same process, the event is a request for the receiver to operate data in the same VMA. If the sender and receiver are in different processes, the event is a request for pointer-based calculation between the peer. If the above understanding is correct, it's hard to represent this type of event for g-i interface because g-i is the object-based framework. Any raw pointer without explicit type is hard to be exposed to g-i applications as long as I know, and it's going to be unsupported, perhaps. 2. event data type corresponding to event type Some event types are expected to specific data type; e.g. SNDRV_SEQ_EVENT_NOTE is for 'struct snd_seq_ev_note' and SNDRV_SEQ_EVENT_CONTROLLER is for 'struct snd_seq_ev_ctrl'. However there're some types for any data type; e.g. SNDRV_SEQ_EVENT_ECHO or SNDRV_SEQ_EVENT_USR0. For the kind of types, the event structure has no information about which data type should be used and user applications voluntarily decide the data type. Therefore the sender and receiver should share a kind of protocol in advance. This means that userspace applications require API to select data type independent of event type itself. 3. quote event and specific event types. Two event types are reserved for 'struct snd_seq_ev_quote'; i.e. SNDRV_SEQ_EVENT_KERNEL_ERROR and SNDRV_SEQ_EVENT_KERNEL_QUOTE(obsoleted?). However, the quote structure is exposed to userspace itself. Furthermore, as of v5.5 kernel, there's no in-kernel code to check the quote data from/to user client. Is it better to produce API so that userspace application can transfer the quote event? [1] https://github.com/alsa-project/alsa-gobject [2] https://github.com/takaswie/alsa-gobject/tree/topic/seq Regards Takashi Sakamoto

2 2

Re: linux-next: Tree for Feb 18 (sound/soc/codecs/wm5110.c)
by Randy Dunlap 18 Feb '20

18 Feb '20

On 2/17/20 8:28 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20200217: > on x86_64: WARNING: unmet direct dependencies detected for SND_SOC_WM5110 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_WM5110 [=n] Selected by [y]: - SND_SOC_BELLS [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_SAMSUNG [=y] && MFD_ARIZONA [=y] && I2C [=y] && SPI_MASTER [=y] && (MACH_WLF_CRAGG_6410 || COMPILE_TEST [=y]) - SND_SOC_SAMSUNG_TM2_WM5110 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_SAMSUNG [=y] && MFD_ARIZONA [=y] && I2C [=y] && SPI_MASTER [=y] && (GPIOLIB [=y] || COMPILE_TEST [=y]) Full randconfig file is attached. -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

3 4

[GIT PULL] ASoC fixes for v5.6
by Mark Brown 18 Feb '20

18 Feb '20

The following changes since commit 13426feaf46c48fcddb591e89d35120fcc90527f: ASoC: wcd934x: Add missing COMMON_CLK dependency to SND_SOC_ALL_CODECS (2020-02-05 10:05:19 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.6-rc2 for you to fetch changes up to 96781fd941b39e1f78098009344ebcd7af861c67: ASoC: sun8i-codec: Fix setting DAI data format (2020-02-17 21:58:41 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.6 A few fixes sent in since the merge window, none of them with global impact but all important for the users they affect. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: atmel: fix atmel_ssc_set_audio link failure Kai Vehmanen (3): ALSA: hda: do not override bus codec_mask in link_get() ASoC: SOF: Intel: hda: fix ordering bug in resume flow ASoC: SOF: Intel: hda: move i915 init earlier Oleksandr Suvorov (1): ASoC: fsl_sai: Fix exiting path on probing failure Ravulapati Vishnu vardhan rao (2): ASoC: amd: Buffer Size instead of MAX Buffer ASoC: amd: ACP needs to be powered off in BIOS. Samuel Holland (2): ASoC: codec2codec: avoid invalid/double-free of pcm runtime ASoC: sun8i-codec: Fix setting DAI data format Tzung-Bi Shih (3): ASoC: max98090: revert invalid fix for handling SHDN ASoC: dapm: remove snd_soc_dapm_put_enum_double_locked ASoC: hdmi-codec: set plugged_cb to NULL when component removing include/sound/soc-dapm.h | 2 - sound/hda/ext/hdac_ext_controller.c | 9 +- sound/soc/amd/raven/acp3x-i2s.c | 8 + sound/soc/amd/raven/acp3x-pcm-dma.c | 7 +- sound/soc/amd/raven/pci-acp3x.c | 23 -- sound/soc/atmel/Kconfig | 4 +- sound/soc/atmel/Makefile | 10 +- sound/soc/codecs/hdmi-codec.c | 10 + sound/soc/codecs/max98090.c | 434 ++++++++++-------------------------- sound/soc/codecs/max98090.h | 3 +- sound/soc/fsl/fsl_sai.c | 22 +- sound/soc/soc-dapm.c | 57 ++--- sound/soc/sof/intel/hda-codec.c | 12 +- sound/soc/sof/intel/hda-dsp.c | 11 + sound/soc/sof/intel/hda.c | 19 +- sound/soc/sunxi/sun8i-codec.c | 3 +- 16 files changed, 219 insertions(+), 415 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: SOF: intel: Fix probe error without i915 graphics
by Takashi Iwai 18 Feb '20

18 Feb '20

When the driver is configured with CONFIG_SND_SOC_SOF_HDA, the SOF driver tries to bind with i915 audio component. But there is also a system without Intel graphics. On such a system, snd_hdac_i915_init() returns -ENODEV error and it leads to the whole probe error of SOF driver. For avoiding this spurious probe error, this patch changes snd_hdac_i915_init() to return -ENOENT for non-existing i915 graphics case. Then the caller can check the error code and handle as no fatal error. In SOF side, a few changes have been added to hda_codec_i915_init() and hda_codec_i915_exit() for skipping the i915 init and exit calls on such a system. BugLink: https://bugzilla.opensuse.org/show_bug.cgi?id=1163677 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206085 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- This is an attempt to fix the regression for HP OMEN 17, which showed the probe error as: [ 6.278811] sof-audio-pci 0000:00:1f.3: error: init i915 and HDMI codec failed [ 6.278907] sof-audio-pci 0000:00:1f.3: error: failed to probe DSP -19 Although the i915 binding error is gone by the patch, it still fails to probe: [ 6.298259] sof-audio-pci 0000:00:1f.3: error: no matching ASoC machine driver found - aborting probe [ 6.298299] sof-audio-pci 0000:00:1f.3: error: failed to get machine info -19 [ 6.298668] sof-audio-pci 0000:00:1f.3: error: sof_probe_work failed err: -19 I'm submitting this fix as is for now since the i915 error itself gets fixed, and the rest seems to be another unsolved problem. Takashi sound/hda/hdac_i915.c | 4 +++- sound/soc/sof/intel/hda-codec.c | 5 +++++ 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 3c2db3816029..a451df3efc8a 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -116,6 +116,8 @@ static const struct drm_audio_component_audio_ops i915_init_ops = { * with i915 graphics driver. * * Returns zero for success or a negative error code. + * -ENOENT indicates that i915 graphics doesn't exist. + * -ENODEV means the binding with i915 graphics failed. */ int snd_hdac_i915_init(struct hdac_bus *bus) { @@ -123,7 +125,7 @@ int snd_hdac_i915_init(struct hdac_bus *bus) int err; if (!i915_gfx_present()) - return -ENODEV; + return -ENOENT; init_completion(&bind_complete); diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 9106ab8dac6f..7202355a701e 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -186,6 +186,8 @@ int hda_codec_i915_init(struct snd_sof_dev *sdev) /* i915 exposes a HDA codec for HDMI audio */ ret = snd_hdac_i915_init(bus); + if (ret == -ENOENT) + return 0; /* no i915 graphics present on the system */ if (ret < 0) return ret; @@ -200,6 +202,9 @@ int hda_codec_i915_exit(struct snd_sof_dev *sdev) struct hdac_bus *bus = sof_to_bus(sdev); int ret; + if (!bus->audio_component) /* not bound with i915 */ + return 0; + hda_codec_i915_display_power(sdev, false); ret = snd_hdac_i915_exit(bus); -- 2.16.4

2 8

[PATCH] ALSA: hda: Use scnprintf() for printing texts for sysfs/procfs
by Takashi Iwai 18 Feb '20

18 Feb '20

Some code in HD-audio driver calls snprintf() in a loop and still expects that the return value were actually written size, while snprintf() returns the expected would-be length instead. When the given buffer limit were small, this leads to a buffer overflow. Use scnprintf() for addressing those issues. It returns the actually written size unlike snprintf(). Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/hda/hdmi_chmap.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_eld.c | 2 +- sound/pci/hda/hda_sysfs.c | 4 ++-- 4 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index 5fd6d575e123..aad5c4bf4d34 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -250,7 +250,7 @@ void snd_hdac_print_channel_allocation(int spk_alloc, char *buf, int buflen) for (i = 0, j = 0; i < ARRAY_SIZE(cea_speaker_allocation_names); i++) { if (spk_alloc & (1 << i)) - j += snprintf(buf + j, buflen - j, " %s", + j += scnprintf(buf + j, buflen - j, " %s", cea_speaker_allocation_names[i]); } buf[j] = '\0'; /* necessary when j == 0 */ diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5dc42f932739..53e7732ef752 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4022,7 +4022,7 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen) for (i = 0, j = 0; i < ARRAY_SIZE(bits); i++) if (pcm & (AC_SUPPCM_BITS_8 << i)) - j += snprintf(buf + j, buflen - j, " %d", bits[i]); + j += scnprintf(buf + j, buflen - j, " %d", bits[i]); buf[j] = '\0'; /* necessary when j == 0 */ } diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index bb46c89b7f63..136477ed46ae 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -360,7 +360,7 @@ static void hdmi_print_pcm_rates(int pcm, char *buf, int buflen) for (i = 0, j = 0; i < ARRAY_SIZE(alsa_rates); i++) if (pcm & (1 << i)) - j += snprintf(buf + j, buflen - j, " %d", + j += scnprintf(buf + j, buflen - j, " %d", alsa_rates[i]); buf[j] = '\0'; /* necessary when j == 0 */ diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c index 0607ed5d1959..eb8ec109d7ad 100644 --- a/sound/pci/hda/hda_sysfs.c +++ b/sound/pci/hda/hda_sysfs.c @@ -222,7 +222,7 @@ static ssize_t init_verbs_show(struct device *dev, int i, len = 0; mutex_lock(&codec->user_mutex); snd_array_for_each(&codec->init_verbs, i, v) { - len += snprintf(buf + len, PAGE_SIZE - len, + len += scnprintf(buf + len, PAGE_SIZE - len, "0x%02x 0x%03x 0x%04x\n", v->nid, v->verb, v->param); } @@ -272,7 +272,7 @@ static ssize_t hints_show(struct device *dev, int i, len = 0; mutex_lock(&codec->user_mutex); snd_array_for_each(&codec->hints, i, hint) { - len += snprintf(buf + len, PAGE_SIZE - len, + len += scnprintf(buf + len, PAGE_SIZE - len, "%s = %s\n", hint->key, hint->val); } mutex_unlock(&codec->user_mutex); -- 2.16.4

1 0

[alsa-devel] [PATCH v3 0/5] soundwire: intel: add DAI callbacks
by Pierre-Louis Bossart 18 Feb '20

18 Feb '20

The existing mainline code is missing most of the DAI callbacks needed for a functional implementation, and the existing ones need to be modified to provide the relevant information to ASoC/SOF drivers. As suggested by Vinod, these patches are shared first - with the risk that they are separated from the actual DAI enablement, so reviewers might wonder why they are needed in the first place. For reference, the complete set of 90+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 Changes since v2: Add missing kfree for stream name (feedback from Vinod) changes since v1: Fix string allocation (only feedback from Vinod) Pierre-Louis Bossart (2): soundwire: intel: rename res field as link_res soundwire: intel: free all resources on hw_free() Rander Wang (3): soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback drivers/soundwire/intel.c | 198 ++++++++++++++++++++++++++++++++++---- 1 file changed, 178 insertions(+), 20 deletions(-) -- 2.20.1

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[PATCH] ALSA: hda/realtek - Apply quirk for yet another MSI laptop
by Takashi Iwai 18 Feb '20

18 Feb '20

MSI GP65 laptop with SSID 1462:1293 requires the same quirk as other MSI models. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204159 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 82485e06dde1..477589e7ec1d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2449,6 +2449,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), -- 2.16.4

1 0

Re: [alsa-devel] [PATCH 1/2] ASoC: rt5682: Add CCF usage for providing I2S clks
by Mark Brown 18 Feb '20

18 Feb '20

On Mon, Feb 17, 2020 at 10:59:55AM +0000, Derek [方德義] wrote: > > Subject: Re: [PATCH 1/2] ASoC: rt5682: Add CCF usage for providing I2S clks > > > + if (parent_rate != CLK_PLL2_FIN) > > > + dev_warn(component->dev, "clk %s only support %d Hz input\n", > > > + clk_name, CLK_PLL2_FIN); > > > + if (rate != CLK_48) { > > > + dev_warn(component->dev, "clk %s only support %d Hz output\n", > > > + clk_name, CLK_48); > > > + rate = CLK_48; > > > + } > > Are these genuine restrictions of the hardware or is this just being > > hard coded in the driver? > It's hard coded for an application case. > Generate a 48kHz clk with a parent 48 MHz clk. I see... obviously this is really not good quality of implementation although it doesn't actually break any external interfaces so I guess we could take it. I would at least want to see some clear comments in both the code and the changelog explaining that this limitation is just a temporary hack until a better implementation is done though.

2 1

[alsa-devel] [PATCH v2 0/2] ASoC hdmi-codec: fix HDMI jack reporting
by Tzung-Bi Shih 17 Feb '20

17 Feb '20

With proper unbinding audio components and inserting the HDMI external display, kernel crashes as the following messages: Unable to handle kernel NULL pointer dereference at virtual address ... [snip] Call trace: plugged_cb+0x1c/0x74 mtk_hdmi_update_plugged_status+0x48/0x6c hdmi_conn_detect+0x1c/0x28 drm_helper_probe_detect+0x110/0x170 drm_helper_probe_single_connector_modes+0xd4/0x608 drm_mode_getconnector+0x1e8/0x418 The 1st patch fixes the crash by notifying hdmi-codec's consumers to not report jack status anymore when component removing. The 2nd patch fixes race condition in mediatek/mtk_hdmi.c. Changes from v1: (https://patchwork.kernel.org/patch/11379979/) - added the 1st patch - use mutex to protect plugged_cb and codec_dev in 2nd patch Tzung-Bi Shih (2): ASoC: hdmi-codec: set plugged_cb to NULL when component removing drm/mediatek: fix race condition for HDMI jack status reporting drivers/gpu/drm/mediatek/mtk_hdmi.c | 11 ++++++++++- sound/soc/codecs/hdmi-codec.c | 10 ++++++++++ 2 files changed, 20 insertions(+), 1 deletion(-) -- 2.25.0.265.gbab2e86ba0-goog

3 7

[alsa-devel] [PATCH] ASoC: amd: ACP needs to be powered off in BIOS.
by Ravulapati Vishnu vardhan rao 17 Feb '20

17 Feb '20

Removed this logic because It is BIOS which needs to power off the ACP power domian through ACP_PGFSM_CTRL register when you De-initialize ACP Engine. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 23 ----------------------- 1 file changed, 23 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 65330bb..da60e2e 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -45,23 +45,6 @@ static int acp3x_power_on(void __iomem *acp3x_base) return -ETIMEDOUT; } -static int acp3x_power_off(void __iomem *acp3x_base) -{ - u32 val; - int timeout; - - rv_writel(ACP_PGFSM_CNTL_POWER_OFF_MASK, - acp3x_base + mmACP_PGFSM_CONTROL); - timeout = 0; - while (++timeout < 500) { - val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); - if ((val & ACP_PGFSM_STATUS_MASK) == ACP_POWERED_OFF) - return 0; - udelay(1); - } - return -ETIMEDOUT; -} - static int acp3x_reset(void __iomem *acp3x_base) { u32 val; @@ -115,12 +98,6 @@ static int acp3x_deinit(void __iomem *acp3x_base) pr_err("ACP3x reset failed\n"); return ret; } - /* power off */ - ret = acp3x_power_off(acp3x_base); - if (ret) { - pr_err("ACP3x power off failed\n"); - return ret; - } return 0; } -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: rt1015: fix typo for bypass boost control
by jack.yu＠realtek.com 17 Feb '20

17 Feb '20

From: Jack Yu <jack.yu(a)realtek.com> Fix typo for "Bypass Boost" control. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 6d490e2dbc25..d300b417dd50 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -444,7 +444,7 @@ static int rt1015_boost_mode_put(struct snd_kcontrol *kcontrol, return 0; } -static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol, +static int rt1015_bypass_boost_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -457,7 +457,7 @@ static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol, return 0; } -static int rt5518_bypass_boost_put(struct snd_kcontrol *kcontrol, +static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -497,7 +497,7 @@ static const struct snd_kcontrol_new rt1015_snd_controls[] = { rt1015_boost_mode_get, rt1015_boost_mode_put), SOC_ENUM("Mono LR Select", rt1015_mono_lr_sel), SOC_SINGLE_EXT("Bypass Boost", SND_SOC_NOPM, 0, 1, 0, - rt5518_bypass_boost_get, rt5518_bypass_boost_put), + rt1015_bypass_boost_get, rt1015_bypass_boost_put), }; static int rt1015_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, -- 2.25.0

2 1

[alsa-devel] [PATCH] ASoC: meson: aiu: simplify component addition
by Jerome Brunet 17 Feb '20

17 Feb '20

Now that the component name is unique within ASoC, there is no need to hack the debugfs prefix to add more than one ASoC component to a linux device. Remove the unnecessary function and use snd_soc_register_component() directly. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/aiu-acodec-ctrl.c | 7 +++---- sound/soc/meson/aiu-codec-ctrl.c | 7 +++---- sound/soc/meson/aiu.c | 20 -------------------- sound/soc/meson/aiu.h | 8 -------- 4 files changed, 6 insertions(+), 36 deletions(-) diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c index b8e88b1a4fc8..7078197e0cc5 100644 --- a/sound/soc/meson/aiu-acodec-ctrl.c +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -197,8 +197,7 @@ static const struct snd_soc_component_driver aiu_acodec_ctrl_component = { int aiu_acodec_ctrl_register_component(struct device *dev) { - return aiu_add_component(dev, &aiu_acodec_ctrl_component, - aiu_acodec_ctrl_dai_drv, - ARRAY_SIZE(aiu_acodec_ctrl_dai_drv), - "acodec"); + return snd_soc_register_component(dev, &aiu_acodec_ctrl_component, + aiu_acodec_ctrl_dai_drv, + ARRAY_SIZE(aiu_acodec_ctrl_dai_drv)); } diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c index 8646a953e3b3..4b773d3e8b07 100644 --- a/sound/soc/meson/aiu-codec-ctrl.c +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -144,9 +144,8 @@ static const struct snd_soc_component_driver aiu_hdmi_ctrl_component = { int aiu_hdmi_ctrl_register_component(struct device *dev) { - return aiu_add_component(dev, &aiu_hdmi_ctrl_component, - aiu_hdmi_ctrl_dai_drv, - ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv), - "hdmi"); + return snd_soc_register_component(dev, &aiu_hdmi_ctrl_component, + aiu_hdmi_ctrl_dai_drv, + ARRAY_SIZE(aiu_hdmi_ctrl_dai_drv)); } diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 34b40b8b8299..d3e2d40e9562 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -71,26 +71,6 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, return 0; } -int aiu_add_component(struct device *dev, - const struct snd_soc_component_driver *component_driver, - struct snd_soc_dai_driver *dai_drv, - int num_dai, - const char *debugfs_prefix) -{ - struct snd_soc_component *component; - - component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL); - if (!component) - return -ENOMEM; - -#ifdef CONFIG_DEBUG_FS - component->debugfs_prefix = debugfs_prefix; -#endif - - return snd_soc_add_component(dev, component, component_driver, - dai_drv, num_dai); -} - static int aiu_cpu_of_xlate_dai_name(struct snd_soc_component *component, struct of_phandle_args *args, const char **dai_name) diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index 097c26de7b7c..06a968c55728 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -11,9 +11,7 @@ struct clk; struct clk_bulk_data; struct device; struct of_phandle_args; -struct snd_soc_component_driver; struct snd_soc_dai; -struct snd_soc_dai_driver; struct snd_soc_dai_ops; enum aiu_clk_ids { @@ -45,12 +43,6 @@ int aiu_of_xlate_dai_name(struct snd_soc_component *component, const char **dai_name, unsigned int component_id); -int aiu_add_component(struct device *dev, - const struct snd_soc_component_driver *component_driver, - struct snd_soc_dai_driver *dai_drv, - int num_dai, - const char *debugfs_prefix); - int aiu_hdmi_ctrl_register_component(struct device *dev); int aiu_acodec_ctrl_register_component(struct device *dev); -- 2.24.1

2 1

[alsa-devel] [RFC] ASoC: soc-pcm: crash in snd_soc_dapm_new_dai
by Sameer Pujar 17 Feb '20

17 Feb '20

Crash happens in snd_soc_dapm_new_dai() when substream->private_data access is made and substream is NULL here. This is seen for DAIs where only playback or capture stream is defined. This seems to be happening for codec2codec DAI link. Both playback and capture are 0 during soc_new_pcm(). This is probably happening because cpu_dai and codec_dai are both validated either for SNDRV_PCM_STREAM_PLAYBACK or SNDRV_PCM_STREAM_CAPTURE. Shouldn't be playback = 1 when, - playback stream is available for codec_dai AND - capture stream is available for cpu_dai and vice-versa for capture = 1? Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 74d340d..5aa9c0b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2855,10 +2855,10 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) for_each_rtd_codec_dai(rtd, i, codec_dai) { if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) playback = 1; if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) capture = 1; } -- 2.7.4

3 7

[alsa-devel] [PATCH] ALSA: hda/realtek - Apply quirk for MSI GP63, too
by Takashi Iwai 17 Feb '20

17 Feb '20

The same quirk that was applied to MSI GL73 is needed for MSI GP63, too. Adding the entry with the SSID 1462:1228. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206503 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 6c8cb4ce517e..82485e06dde1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2447,6 +2447,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte EP45-DS3/Z87X-UD3H", ALC889_FIXUP_FRONT_HP_NO_PRESENCE), SND_PCI_QUIRK(0x1458, 0xa0b8, "Gigabyte AZ370-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), + SND_PCI_QUIRK(0x1462, 0x1228, "MSI-GP63", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), -- 2.16.4

1 0

[alsa-devel] [PATCH 0/2] ALSA: usb-audio: UAC2 effect unit parsing
by Takashi Iwai 17 Feb '20

17 Feb '20

Hi, UAC2 Effect Unit needs different parsing from Processing Unit and co, and this patch set implements the partial support for that. As it's the fix on top of the latest USB-audio fix, I'd like to apply through sound git tree. So please review and ack if USB header change is OK. thanks, Takashi === Takashi Iwai (2): usb: audio-v2: Add uac2_effect_unit_descriptor definition ALSA: usb-audio: Parse source ID of UAC2 effect unit include/linux/usb/audio-v2.h | 12 ++++++++++++ sound/usb/mixer.c | 6 ++++++ 2 files changed, 18 insertions(+) -- 2.16.4

2 5

[alsa-devel] DMARC & gmail & cirrus.com
by Jaroslav Kysela 17 Feb '20

17 Feb '20

Hi all, today, all gmail users were unsubscribed from the mailing list, because we have users who sends e-mails from domains with the DMARC policy reject (like cirrus.com). The mailman counts those bounces and unsubscribe users who do not receive those e-mails. The nice explanation is here: https://www.linuxchix.org/content/mailing-list-changes I can enable the "Replace the From: header address" in the mailman config to "Munge From" like in the above example, but it will mean that From: will be altered (see the explanation). Another option is to disable the bounce check in mailman, but it will cause that the "dead" subscribers are not detected anymore. Any opinions on this? Jaroslav SMTP error: (host alt1.gmail-smtp-in.l.google.com[173.194.73.26] said: 550-5.7.26 Unauthenticated email from cirrus.com is not accepted due to domain's 550-5.7.26 DMARC policy. Please contact the administrator of cirrus.com domain 550-5.7.26 if this was a legitimate mail. Please visit 550-5.7.26 https://support.google.com/mail/answer/2451690 to learn about the 450 4.7.26 DMARC initiative. m14si10079981ljg.84 - gsmtp (in reply to end of DATA command)) DMARC settings for cirrus.com: v=DMARC1; p=reject; fo=1;rua=mailto:dmarc_rua@emaildefense.proofpoint.com;ruf=mailto:dmarc_ruf@emaildefense.proofpoint.com -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH v3 00/18] ASoC: soc-pcm cleanup step3
by Kuninori Morimoto 17 Feb '20

17 Feb '20

Hi Mark These are v3 of soc-pcm cleanup step3. Main change from v2 is that last patch was splited into each files (= [10/18] to [18/18]). And all patches have Reviewed-by tag from Ranjani Kuninori Morimoto (18): 1) ASoC: soc-pcm: move dai_get_widget() 2) ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() 3) ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() 4) ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() 5) ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() 6) ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() 7) ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code 8) ASoC: soc-pcm: use goto and remove multi return 9) ASoC: soc-pcm: merge playback/cature_active into stream_active 10) ASoC: soc.h: add for_each_pcm_streams() 11) ASoC: soc-core: use for_each_pcm_streams() macro 12) ASoC: soc-pcm: use for_each_pcm_streams() macro 13) ASoC: soc-generic-dmaengine-pcm: use for_each_pcm_streams() macro 14) ASoC: dwc: dwc-i2s: use for_each_pcm_streams() macro 15) ASoC: fsl: fsl_asrc_dma: use for_each_pcm_streams() macro 16) ASoC: qcom: lpass-platform: use for_each_pcm_streams() macro 17) ASoC: sof: sof-audio: use for_each_pcm_streams() macro 18) ALSA: usx2y: use for_each_pcm_streams() macro include/sound/pcm.h | 5 + include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 - sound/soc/codecs/cs4271.c | 4 +- sound/soc/dwc/dwc-i2s.c | 8 +- sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/soc-core.c | 48 ++-- sound/soc/soc-generic-dmaengine-pcm.c | 8 +- sound/soc/soc-pcm.c | 327 +++++++++----------------- sound/soc/sof/sof-audio.c | 19 +- sound/usb/usx2y/usbusx2yaudio.c | 9 +- 12 files changed, 162 insertions(+), 284 deletions(-) -- 2.17.1

2 21

[alsa-devel] [PATCH] drm/mediatek: fix race condition for HDMI jack status reporting
by Tzung-Bi Shih 17 Feb '20

17 Feb '20

hdmi_conn_detect and mtk_hdmi_audio_hook_plugged_cb would be called by different threads. Imaging the following calling sequence: Thread A Thread B -------------------------------------------------------------------- mtk_hdmi_audio_hook_plugged_cb() mtk_cec_hpd_high() -> disconnected hdmi_conn_detect() mtk_cec_hpd_high() -> connected plugged_cb(connected) plugged_cb(disconnected) The latest disconnected is false reported. Makes mtk_cec_hpd_high and plugged_cb atomic to fix. plugged_cb and codec_dev are also in danger of race condition. Instead of using mutex to protect them: - Checks NULLs first. - Uses WRITE_ONCE() to prevent store tearing (i.e. write to plugged_cb after codec_dev). - Uses codec_dev as a signal to report HDMI jack status. Fixes: 5d3c64477392 ("drm/mediatek: support HDMI jack status reporting") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Previous discussion: https://patchwork.kernel.org/patch/11367625/ Previous attempt: https://patchwork.kernel.org/patch/11378413/ drivers/gpu/drm/mediatek/mtk_hdmi.c | 18 ++++++++++++++---- 1 file changed, 14 insertions(+), 4 deletions(-) diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c index 03aeb73005ef..b1e5d0c538fa 100644 --- a/drivers/gpu/drm/mediatek/mtk_hdmi.c +++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c @@ -12,6 +12,7 @@ #include <linux/io.h> #include <linux/kernel.h> #include <linux/mfd/syscon.h> +#include <linux/mutex.h> #include <linux/of_platform.h> #include <linux/of.h> #include <linux/of_gpio.h> @@ -171,6 +172,7 @@ struct mtk_hdmi { bool enabled; hdmi_codec_plugged_cb plugged_cb; struct device *codec_dev; + struct mutex update_plugged_status_lock; }; static inline struct mtk_hdmi *hdmi_ctx_from_bridge(struct drm_bridge *b) @@ -1199,10 +1201,13 @@ static void mtk_hdmi_clk_disable_audio(struct mtk_hdmi *hdmi) static enum drm_connector_status mtk_hdmi_update_plugged_status(struct mtk_hdmi *hdmi) { - bool connected = mtk_cec_hpd_high(hdmi->cec_dev); + bool connected; - if (hdmi->plugged_cb && hdmi->codec_dev) + mutex_lock(&hdmi->update_plugged_status_lock); + connected = mtk_cec_hpd_high(hdmi->cec_dev); + if (hdmi->codec_dev) hdmi->plugged_cb(hdmi->codec_dev, connected); + mutex_unlock(&hdmi->update_plugged_status_lock); return connected ? connector_status_connected : connector_status_disconnected; @@ -1669,8 +1674,12 @@ static int mtk_hdmi_audio_hook_plugged_cb(struct device *dev, void *data, { struct mtk_hdmi *hdmi = data; - hdmi->plugged_cb = fn; - hdmi->codec_dev = codec_dev; + if (!fn || !codec_dev) + return -EINVAL; + + /* Use WRITE_ONCE() to prevent store tearing. */ + WRITE_ONCE(hdmi->plugged_cb, fn); + WRITE_ONCE(hdmi->codec_dev, codec_dev); mtk_hdmi_update_plugged_status(hdmi); return 0; @@ -1729,6 +1738,7 @@ static int mtk_drm_hdmi_probe(struct platform_device *pdev) return ret; } + mutex_init(&hdmi->update_plugged_status_lock); platform_set_drvdata(pdev, hdmi); ret = mtk_hdmi_output_init(hdmi); -- 2.25.0.225.g125e21ebc7-goog

2 5

[alsa-devel] [PATCH v2 00/10] ASoC: soc-pcm cleanup step3
by Kuninori Morimoto 17 Feb '20

17 Feb '20

Hi Mark These are v2 of soc-pcm cleanup step3. I removed unneeded Multi-Codec patch. And re-ordered / merged my local tree and add [09/10] patch to make [10/10] more effective. Kuninori Morimoto (10): 1) ASoC: soc-pcm: move dai_get_widget() 2) ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() 3) ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() 4) ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() 5) ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() 6) ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() 7) ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code 8) ASoC: soc-pcm: use goto and remove multi return 9) ASoC: soc-pcm: merge playback/cature_active into stream_active 10) ASoC: soc.h: add for_each_pcm_streams() include/sound/pcm.h | 5 + include/sound/soc-dai.h | 3 +- include/sound/soc-dpcm.h | 9 - sound/soc/codecs/cs4271.c | 4 +- sound/soc/dwc/dwc-i2s.c | 8 +- sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/soc-core.c | 48 ++-- sound/soc/soc-generic-dmaengine-pcm.c | 8 +- sound/soc/soc-pcm.c | 327 +++++++++----------------- sound/soc/sof/sof-audio.c | 19 +- sound/usb/usx2y/usbusx2yaudio.c | 9 +- 12 files changed, 162 insertions(+), 284 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v2 0/3] simple-audio-card codec2codec support
by Samuel Holland 17 Feb '20

17 Feb '20

We are currently using simple-audio-card on the Allwinner A64 SoC. The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the CPU, the modem, and Bluetooth. Adding support for the secondary AIFs requires adding codec2codec DAI links. Since the modem and bt-sco codec DAI drivers only have one set of possible PCM parameters (namely, 8kHz mono S16LE), there's no real need for a machine driver to specify the DAI link configuration. The parameters for these "simple" DAI links can be chosen automatically. This series adds codec2codec DAI link support to simple-audio-card. Codec to codec links are automatically detected when DAIs in the link belong to codec components. I tried to reuse as much code as possible, so the first two patches refactor a couple of helper functions to be more generic. The last patch adds the new feature and its documentation. Samuel Holland (3): ALSA: pcm: Add a non-runtime version of snd_pcm_limit_hw_rates ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec to codec DAI links Documentation/sound/soc/codec-to-codec.rst | 9 +++- include/sound/pcm.h | 9 +++- include/sound/soc.h | 3 ++ sound/core/pcm_misc.c | 18 +++---- sound/soc/generic/simple-card-utils.c | 50 ++++++++++++++++++++ sound/soc/soc-pcm.c | 55 +++++++++++++++------- 6 files changed, 116 insertions(+), 28 deletions(-) -- 2.24.1

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Re: [alsa-devel] Duet Firewire
by Takashi Sakamoto 16 Feb '20

16 Feb '20

Hi, (C.C.ed to alsa-devel) On Sun, Feb 16, 2020 at 03:52:27AM +0000, roberto lopez torres wrote: > Hi, i just got a Duet Firewire, and i'm trying to make it work with 5.3.0-29-gen > eric #31-Ubuntu > > I have inputs and outputs from the interface but can't config anything > at the moment, and i got a noise every 16 seconds on headphones > monitoring. > > I'm pretty new using linux. The ALSA support for Apogee Duet FireWire starts at Linux kernel v4.21. As you can see, the support is just for packet streaming, thus it's just possible to transmit/receive PCM frames and MIDI messages. However, the device is designed to run just depending on its internal clock (e.g. 47994 frames/sec), thus at present it generates periodical noise since the ALSA driver is designed to transmit just the same number of PCM frames as sampling rate (e.g. 48000 frames/sec). The support of any control is expected to be implemented in userspace application. This is the reason that you can see no elements in ALSA control application (e.g. alsamixer). However, the protocol to control the device is not clear yet. As long as I know, it's complicated and requires the batch of codes for the application. This is the reason that ALSA in-kernel driver doesn't have the code. Regards Takashi Sakamoto

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[alsa-devel] Best way to interface with equalizer
by Manusha Wijekoon 16 Feb '20

16 Feb '20

Hello I want to control ALSA equalizer from a mobile phone GUI. Essentially I would like to build a small GUI that runs on a phone to allow a user to change equalizer visually. I want this to be very smooth and stable. Few options that I can think of are: 1. Have an agent program running in the linux machine which controls alsa by sending various commands through linux shell. 2. Agent program running in the linux machine directly interfaces to ALSA through its C interface. 3. Is there a control interface provided by ALSA, for example REST like service? 3 seem to be the best option as it involves less work. 1 looks fragile to me. What are your recommendations? Thank you S

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[alsa-devel] [PATCH v3] ALSA: usb-audio: Add support for Presonus Studio 1810c
by mickflemm＠gmail.com 15 Feb '20

15 Feb '20

From: Nick Kossifidis <mickflemm(a)gmail.com> This patch adds support for Presonus Studio 1810c, a usb interface that's UAC2 compliant with a few quirks and a few extra hw-specific controls. I've tested all 3 altsettings and the added switch controls and they work as expected. More infos on the card: https://www.presonus.com/products/Studio-1810c Note that this work is based on packet inspection with usbmon. I just wanted to get this card to work for using it on our open-source radio station: https://github.com/UoC-Radio v2 address issues reported by Takashi: * Properly get/set enum type controls * Prevent race condition on switch_get/set * Various control naming changes * Various coding style fixes v3 improve readability of sample rate filtering and some other minor changes. Signed-off-by: Nick Kossifidis <mickflemm(a)gmail.com> --- sound/usb/Makefile | 1 + sound/usb/format.c | 37 +++ sound/usb/mixer_quirks.c | 5 + sound/usb/mixer_s1810c.c | 595 +++++++++++++++++++++++++++++++++++++++ sound/usb/mixer_s1810c.h | 7 + sound/usb/quirks.c | 36 +++ 6 files changed, 681 insertions(+) create mode 100644 sound/usb/mixer_s1810c.c create mode 100644 sound/usb/mixer_s1810c.h diff --git a/sound/usb/Makefile b/sound/usb/Makefile index 78edd7d2f..56031026b 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -13,6 +13,7 @@ snd-usb-audio-objs := card.o \ mixer_scarlett.o \ mixer_scarlett_gen2.o \ mixer_us16x08.o \ + mixer_s1810c.o \ pcm.o \ power.o \ proc.o \ diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db7130..462187953 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -226,6 +226,36 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof return 0; } + +/* + * Presonus Studio 1810c supports a limited set of sampling + * rates per altsetting but reports the full set each time. + * If we don't filter out the unsupported rates and attempt + * to configure the card, it will hang refusing to do any + * further audio I/O until a hard reset is performed. + * + * The list of supported rates per altsetting (set of available + * I/O channels) is described in the owner's manual, section 2.2. + */ +static bool s1810c_valid_sample_rate(struct audioformat *fp, + unsigned int rate) +{ + switch (fp->altsetting) { + case 1: + /* All ADAT ports available */ + return rate <= 48000; + case 2: + /* Half of ADAT ports available */ + return (rate == 88200 || rate == 96000); + case 3: + /* Analog I/O only (no S/PDIF nor ADAT) */ + return rate >= 176400; + default: + return false; + } + return false; +} + /* * Helper function to walk the array of sample rate triplets reported by * the device. The problem is that we need to parse whole array first to @@ -262,6 +292,12 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, } for (rate = min; rate <= max; rate += res) { + + /* Filter out invalid rates on Presonus Studio 1810c */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c) && + !s1810c_valid_sample_rate(fp, rate)) + goto skip_rate; + if (fp->rate_table) fp->rate_table[nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) @@ -276,6 +312,7 @@ static int parse_uac2_sample_rate_range(struct snd_usb_audio *chip, break; } +skip_rate: /* avoid endless loop */ if (res == 0) break; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 39e27ae6c..71e705fab 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -34,6 +34,7 @@ #include "mixer_scarlett.h" #include "mixer_scarlett_gen2.h" #include "mixer_us16x08.h" +#include "mixer_s1810c.h" #include "helper.h" struct std_mono_table { @@ -2277,6 +2278,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fd4): /* RME */ err = snd_rme_controls_create(mixer); break; + + case USB_ID(0x0194f, 0x010c): /* Presonus Studio 1810c */ + err = snd_sc1810_init_mixer(mixer); + break; } return err; diff --git a/sound/usb/mixer_s1810c.c b/sound/usb/mixer_s1810c.c new file mode 100644 index 000000000..8e2a05f80 --- /dev/null +++ b/sound/usb/mixer_s1810c.c @@ -0,0 +1,595 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm(a)gmail.com> + * + * Based on reverse engineering of the communication protocol + * between the windows driver / Univeral Control (UC) program + * and the device, through usbmon. + * + * For now this bypasses the mixer, with all channels split, + * so that the software can mix with greater flexibility. + * It also adds controls for the 4 buttons on the front of + * the device. + */ + +#include <linux/usb.h> +#include <linux/usb/audio-v2.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> + +#include "usbaudio.h" +#include "mixer.h" +#include "mixer_quirks.h" +#include "helper.h" +#include "mixer_s1810c.h" + +#define SC1810C_CMD_REQ 160 +#define SC1810C_CMD_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT) +#define SC1810C_CMD_F1 0x50617269 +#define SC1810C_CMD_F2 0x14 + +/* + * DISCLAIMER: These are just guesses based on the + * dumps I got. + * + * It seems like a selects between + * device (0), mixer (0x64) and output (0x65) + * + * For mixer (0x64): + * * b selects an input channel (see below). + * * c selects an output channel pair (see below). + * * d selects left (0) or right (1) of that pair. + * * e 0-> disconnect, 0x01000000-> connect, + * 0x0109-> used for stereo-linking channels, + * e is also used for setting volume levels + * in which case b is also set so I guess + * this way it is possible to set the volume + * level from the specified input to the + * specified output. + * + * IN Channels: + * 0 - 7 Mic/Inst/Line (Analog inputs) + * 8 - 9 S/PDIF + * 10 - 17 ADAT + * 18 - 35 DAW (Inputs from the host) + * + * OUT Channels (pairs): + * 0 -> Main out + * 1 -> Line1/2 + * 2 -> Line3/4 + * 3 -> S/PDIF + * 4 -> ADAT? + * + * For device (0): + * * b and c are not used, at least not on the + * dumps I got. + * * d sets the control id to be modified + * (see below). + * * e sets the setting for that control. + * (so for the switches I was interested + * in it's 0/1) + * + * For output (0x65): + * * b is the output channel (see above). + * * c is zero. + * * e I guess the same as with mixer except 0x0109 + * which I didn't see in my dumps. + * + * The two fixed fields have the same values for + * mixer and output but a different set for device. + */ +struct s1810c_ctl_packet { + u32 a; + u32 b; + u32 fixed1; + u32 fixed2; + u32 c; + u32 d; + u32 e; +}; + +#define SC1810C_CTL_LINE_SW 0 +#define SC1810C_CTL_MUTE_SW 1 +#define SC1810C_CTL_AB_SW 3 +#define SC1810C_CTL_48V_SW 4 + +#define SC1810C_SET_STATE_REQ 161 +#define SC1810C_SET_STATE_REQTYPE SC1810C_CMD_REQTYPE +#define SC1810C_SET_STATE_F1 0x64656D73 +#define SC1810C_SET_STATE_F2 0xF4 + +#define SC1810C_GET_STATE_REQ 162 +#define SC1810C_GET_STATE_REQTYPE \ + (USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN) +#define SC1810C_GET_STATE_F1 SC1810C_SET_STATE_F1 +#define SC1810C_GET_STATE_F2 SC1810C_SET_STATE_F2 + +#define SC1810C_STATE_F1_IDX 2 +#define SC1810C_STATE_F2_IDX 3 + +/* + * This packet includes mixer volumes and + * various other fields, it's an extended + * version of ctl_packet, with a and b + * being zero and different f1/f2. + */ +struct s1810c_state_packet { + u32 fields[63]; +}; + +#define SC1810C_STATE_48V_SW 58 +#define SC1810C_STATE_LINE_SW 59 +#define SC1810C_STATE_MUTE_SW 60 +#define SC1810C_STATE_AB_SW 62 + +struct s1810_mixer_state { + uint16_t seqnum; + struct mutex usb_mutex; + struct mutex data_mutex; +}; + +static int +snd_s1810c_send_ctl_packet(struct usb_device *dev, u32 a, + u32 b, u32 c, u32 d, u32 e) +{ + struct s1810c_ctl_packet pkt = { 0 }; + int ret = 0; + + pkt.fixed1 = SC1810C_CMD_F1; + pkt.fixed2 = SC1810C_CMD_F2; + + pkt.a = a; + pkt.b = b; + pkt.c = c; + pkt.d = d; + /* + * Value for settings 0/1 for this + * output channel is always 0 (probably because + * there is no ADAT output on 1810c) + */ + pkt.e = (c == 4) ? 0 : e; + + ret = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + SC1810C_CMD_REQ, + SC1810C_CMD_REQTYPE, 0, 0, &pkt, sizeof(pkt)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send ctl packet\n"); + return ret; + } + return 0; +} + +/* + * When opening Universal Control the program periodicaly + * sends and receives state packets for syncinc state between + * the device and the host. + * + * Note that if we send only the request to get data back we'll + * get an error, we need to first send an empty state packet and + * then ask to receive a filled. Their seqnumbers must also match. + */ +static int +snd_sc1810c_get_status_field(struct usb_device *dev, + u32 *field, int field_idx, uint16_t *seqnum) +{ + struct s1810c_state_packet pkt_out = { 0 }; + struct s1810c_state_packet pkt_in = { 0 }; + int ret = 0; + + pkt_out.fields[SC1810C_STATE_F1_IDX] = SC1810C_SET_STATE_F1; + pkt_out.fields[SC1810C_STATE_F2_IDX] = SC1810C_SET_STATE_F2; + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_SET_STATE_REQ, + SC1810C_SET_STATE_REQTYPE, + (*seqnum), 0, &pkt_out, sizeof(pkt_out)); + if (ret < 0) { + dev_warn(&dev->dev, "could not send state packet (%d)\n", ret); + return ret; + } + + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + SC1810C_GET_STATE_REQ, + SC1810C_GET_STATE_REQTYPE, + (*seqnum), 0, &pkt_in, sizeof(pkt_in)); + if (ret < 0) { + dev_warn(&dev->dev, "could not get state field %u (%d)\n", + field_idx, ret); + return ret; + } + + (*field) = pkt_in.fields[field_idx]; + (*seqnum)++; + return 0; +} + +/* + * This is what I got when bypassing the mixer with + * all channels split. I'm not 100% sure of what's going + * on, I could probably clean this up based on my observations + * but I prefer to keep the same behavior as the windows driver. + */ +static int snd_s1810c_init_mixer_maps(struct snd_usb_audio *chip) +{ + u32 a, b, c, e, n, off; + struct usb_device *dev = chip->dev; + + /* Set initial volume levels ? */ + a = 0x64; + e = 0xbc; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off, c = 0; b < 18 + off; b++) { + /* This channel to all outputs ? */ + for (c = 0; c <= 8; c++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + /* This channel to main output (again) */ + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + /* + * I noticed on UC that DAW channels have different + * initial volumes, so this makes sense. + */ + e = 0xb53bf0; + } + + /* Connect analog outputs ? */ + a = 0x65; + e = 0x01000000; + for (b = 1; b < 3; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 0, 0, 1, e); + + /* Set initial volume levels for S/PDIF mappings ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF output ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Connect all outputs (again) ? */ + a = 0x65; + e = 0x01000000; + for (b = 0; b < 4; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, 0, 1, e); + } + + /* Basic routing to get sound out of the device */ + a = 0x64; + e = 0x01000000; + for (c = 0; c < 4; c++) { + for (b = 0; b < 36; b++) { + if ((c == 0 && b == 18) || /* DAW1/2 -> Main */ + (c == 1 && b == 20) || /* DAW3/4 -> Line3/4 */ + (c == 2 && b == 22) || /* DAW4/5 -> Line5/6 */ + (c == 3 && b == 24)) { /* DAW5/6 -> S/PDIF */ + /* Left */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + b++; + /* Right */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } else { + /* Leave the rest disconnected */ + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, 0); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, 0); + } + } + } + + /* Set initial volume levels for S/PDIF (again) ? */ + a = 0x64; + e = 0xbc; + c = 3; + for (n = 0; n < 2; n++) { + off = n * 18; + for (b = off; b < 18 + off; b++) { + snd_s1810c_send_ctl_packet(dev, a, b, c, 0, e); + snd_s1810c_send_ctl_packet(dev, a, b, c, 1, e); + } + e = 0xb53bf0; + } + + /* Connect S/PDIF outputs (again) ? */ + a = 0x65; + e = 0x01000000; + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + /* Again ? */ + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 0, e); + snd_s1810c_send_ctl_packet(dev, a, 3, 0, 1, e); + + return 0; +} + +/* + * Sync state with the device and retrieve the requested field, + * whose index is specified in (kctl->private_value & 0xFF), + * from the received fields array. + */ +static int +snd_s1810c_get_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl, u32 *state) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 field = 0; + u32 ctl_idx = (u32) (kctl->private_value & 0xFF); + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_sc1810c_get_status_field(chip->dev, &field, + ctl_idx, &private->seqnum); + if (ret < 0) + goto unlock; + + *state = field; + unlock: + mutex_unlock(&private->usb_mutex); + return ret ? ret : 0; +} + +/* + * Send a control packet to the device for the control id + * specified in (kctl->private_value >> 8) with value + * specified in (kctl->private_value >> 16). + */ +static int +snd_s1810c_set_switch_state(struct usb_mixer_interface *mixer, + struct snd_kcontrol *kctl) +{ + struct snd_usb_audio *chip = mixer->chip; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_id = (pval >> 8) & 0xFF; + u32 ctl_val = (pval >> 16) & 0x1; + int ret = 0; + + mutex_lock(&private->usb_mutex); + ret = snd_s1810c_send_ctl_packet(chip->dev, 0, 0, 0, ctl_id, ctl_val); + mutex_unlock(&private->usb_mutex); + return ret; +} + +/* Generic get/set/init functions for switch controls */ + +static int +snd_s1810c_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 state = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &state); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + ctl_elem->value.enumerated.item[0] = (int)state; + break; + default: + ctl_elem->value.integer.value[0] = (long)state; + } + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? ret : 0; +} + +static int +snd_s1810c_switch_set(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ctl_elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + struct s1810_mixer_state *private = mixer->private_data; + u32 pval = (u32) kctl->private_value; + u32 ctl_idx = pval & 0xFF; + u32 curval = 0; + u32 newval = 0; + int ret = 0; + + mutex_lock(&private->data_mutex); + ret = snd_s1810c_get_switch_state(mixer, kctl, &curval); + if (ret < 0) + goto unlock; + + switch (ctl_idx) { + case SC1810C_STATE_LINE_SW: + case SC1810C_STATE_AB_SW: + newval = (u32) ctl_elem->value.enumerated.item[0]; + break; + default: + newval = (u32) ctl_elem->value.integer.value[0]; + } + + if (curval == newval) + goto unlock; + + kctl->private_value &= ~(0x1 << 16); + kctl->private_value |= (unsigned int)(newval & 0x1) << 16; + ret = snd_s1810c_set_switch_state(mixer, kctl); + + unlock: + mutex_unlock(&private->data_mutex); + return (ret < 0) ? 0 : 1; +} + +static int +snd_s1810c_switch_init(struct usb_mixer_interface *mixer, + const struct snd_kcontrol_new *new_kctl) +{ + struct snd_kcontrol *kctl; + struct usb_mixer_elem_info *elem; + + elem = kzalloc(sizeof(struct usb_mixer_elem_info), GFP_KERNEL); + if (!elem) + return -ENOMEM; + + elem->head.mixer = mixer; + elem->control = 0; + elem->head.id = 0; + elem->channels = 1; + + kctl = snd_ctl_new1(new_kctl, elem); + if (!kctl) { + kfree(elem); + return -ENOMEM; + } + kctl->private_free = snd_usb_mixer_elem_free; + + return snd_usb_mixer_add_control(&elem->head, kctl); +} + +static int +snd_s1810c_line_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Preamp On (Mic/Inst)", + "Preamp Off (Line in)" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_line_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line 1/2 Source Type", + .info = snd_s1810c_line_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_LINE_SW | SC1810C_CTL_LINE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_mute_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mute Main Out Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_MUTE_SW | SC1810C_CTL_MUTE_SW << 8) +}; + +static const struct snd_kcontrol_new snd_s1810c_48v_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "48V Phantom Power On Mic Inputs Switch", + .info = snd_ctl_boolean_mono_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_48V_SW | SC1810C_CTL_48V_SW << 8) +}; + +static int +snd_s1810c_ab_sw_info(struct snd_kcontrol *kctl, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "1/2", + "3/4" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static const struct snd_kcontrol_new snd_s1810c_ab_sw = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone 1 Source Route", + .info = snd_s1810c_ab_sw_info, + .get = snd_s1810c_switch_get, + .put = snd_s1810c_switch_set, + .private_value = (SC1810C_STATE_AB_SW | SC1810C_CTL_AB_SW << 8) +}; + +static void snd_sc1810_mixer_state_free(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = mixer->private_data; + kfree(private); + mixer->private_data = NULL; +} + +/* Entry point, called from mixer_quirks.c */ +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer) +{ + struct s1810_mixer_state *private = NULL; + struct snd_usb_audio *chip = mixer->chip; + struct usb_device *dev = chip->dev; + int ret = 0; + + /* Run this only once */ + if (!list_empty(&chip->mixer_list)) + return 0; + + dev_info(&dev->dev, + "Presonus Studio 1810c, device_setup: %u\n", chip->setup); + if (chip->setup == 1) + dev_info(&dev->dev, "(8out/18in @ 48KHz)\n"); + else if (chip->setup == 2) + dev_info(&dev->dev, "(6out/8in @ 192KHz)\n"); + else + dev_info(&dev->dev, "(8out/14in @ 96KHz)\n"); + + ret = snd_s1810c_init_mixer_maps(chip); + if (ret < 0) + return ret; + + private = kzalloc(sizeof(struct s1810_mixer_state), GFP_KERNEL); + if (!private) + return -ENOMEM; + + mutex_init(&private->usb_mutex); + mutex_init(&private->data_mutex); + + mixer->private_data = private; + mixer->private_free = snd_sc1810_mixer_state_free; + + private->seqnum = 1; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_line_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_mute_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_48v_sw); + if (ret < 0) + return ret; + + ret = snd_s1810c_switch_init(mixer, &snd_s1810c_ab_sw); + if (ret < 0) + return ret; + return ret; +} diff --git a/sound/usb/mixer_s1810c.h b/sound/usb/mixer_s1810c.h new file mode 100644 index 000000000..a79a3743c --- /dev/null +++ b/sound/usb/mixer_s1810c.h @@ -0,0 +1,7 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Presonus Studio 1810c driver for ALSA + * Copyright (C) 2019 Nick Kossifidis <mickflemm(a)gmail.com> + */ + +int snd_sc1810_init_mixer(struct usb_mixer_interface *mixer); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a81c20664..9bbdea882 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1227,6 +1227,38 @@ static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } +static int s1810c_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + /* + * Altno settings: + * + * Playback (Interface 1): + * 1: 6 Analog + 2 S/PDIF + * 2: 6 Analog + 2 S/PDIF + * 3: 6 Analog + * + * Capture (Interface 2): + * 1: 8 Analog + 2 S/PDIF + 8 ADAT + * 2: 8 Analog + 2 S/PDIF + 4 ADAT + * 3: 8 Analog + */ + + /* + * I'll leave 2 as the default one and + * use device_setup to switch to the + * other two. + */ + if ((chip->setup == 0 || chip->setup > 2) && altno != 2) + return 1; + else if (chip->setup == 1 && altno != 1) + return 1; + else if (chip->setup == 2 && altno != 3) + return 1; + + return 0; +} + int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, int iface, int altno) @@ -1240,6 +1272,10 @@ int snd_usb_apply_interface_quirk(struct snd_usb_audio *chip, /* fasttrackpro usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2012)) return fasttrackpro_skip_setting_quirk(chip, iface, altno); + /* presonus studio 1810c: skip altsets incompatible with device_setup */ + if (chip->usb_id == USB_ID(0x0194f, 0x010c)) + return s1810c_skip_setting_quirk(chip, iface, altno); + return 0; } -- 2.23.0

2 1

[alsa-devel] [PATCH 0/4] simple-audio-card codec2codec support
by Samuel Holland 15 Feb '20

15 Feb '20

We are currently using simple-audio-card on the Allwinner A64 SoC. The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the CPU, the modem, and Bluetooth. Adding support for the secondary AIFs requires adding codec2codec DAI links. Since the modem and bt-sco codec DAI drivers only have one set of possible PCM parameters (namely, 8kHz mono S16LE), there's no real need for a machine driver to specify the DAI link configuration. The parameters for these "simple" DAI links can be chosen automatically. This series adds a single "codec-to-codec" property to the simple-audio-card binding, which does exactly what it says. It works out rather nicely without making the device tree binding too complicated. The first patch fixes a bug I found while implementing this feature. I tried to reuse as much code as possible, so the middle two patches refactor a couple of helper functions to be more generic. Finally, the last patch adds the new feature and its documentation. Samuel Holland (4): ASoC: codec2codec: avoid invalid/double-free of pcm runtime ALSA: pcm: Make snd_pcm_limit_hw_rates take hw directly ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec-to-codec dai_links .../devicetree/bindings/sound/simple-card.txt | 1 + Documentation/sound/soc/codec-to-codec.rst | 9 ++- .../drm/bridge/synopsys/dw-hdmi-ahb-audio.c | 2 +- include/sound/pcm.h | 2 +- include/sound/simple_card_utils.h | 1 + include/sound/soc.h | 3 + sound/arm/aaci.c | 2 +- sound/arm/pxa2xx-ac97.c | 2 +- sound/core/pcm_misc.c | 14 ++--- sound/firewire/dice/dice-pcm.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 2 +- sound/firewire/motu/motu-pcm.c | 2 +- sound/firewire/tascam/tascam-pcm.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/cs5535audio/cs5535audio_pcm.c | 4 +- sound/pci/hda/hda_controller.c | 4 +- sound/pci/intel8x0.c | 2 +- sound/pci/sis7019.c | 2 +- sound/pci/via82xx.c | 4 +- sound/soc/generic/simple-card-utils.c | 39 +++++++++++++ sound/soc/generic/simple-card.c | 12 ++++ sound/soc/soc-dapm.c | 3 - sound/soc/soc-pcm.c | 57 +++++++++++++------ sound/usb/caiaq/audio.c | 4 +- 25 files changed, 130 insertions(+), 49 deletions(-) -- 2.24.1

4 12

[alsa-devel] [RFC PATCH 0/2] soundwire: add master_device/driver support
by Pierre-Louis Bossart 15 Feb '20

15 Feb '20

The SoundWire master representation needs to evolve to take into account: a) a May 2019 recommendation from Greg KH to remove platform devices b) the need on Intel platforms to support hardware startup only once the power rail dependencies are settled. The SoundWire master is not an independent IP at all. c) the need to deal with external wakes handled by the PCI subsystem in specific low-power modes In case it wasn't clear, the SoundWire subsystem is currently unusable with v5.5 on devices hitting the shelves very soon (race conditions, power management, suspend/resume, etc). v5.6 will only provide interface changes and no functional improvement. We've circled on the same concepts since September 2019, and I hope this direction is now aligned with the suggestions from Vinod Koul and that we can target v5.7 as the 'SoundWire just works(tm)' version. This series is provided as an RFC since it depends on patches already for review. Pierre-Louis Bossart (2): soundwire: bus_type: add master_device/driver support soundwire: intel: transition to sdw_master_device/driver support drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus_type.c | 141 +++++++++++++- drivers/soundwire/intel.c | 96 ++++++---- drivers/soundwire/intel.h | 8 +- drivers/soundwire/intel_init.c | 283 ++++++++++++++++++++++------- drivers/soundwire/master.c | 119 ++++++++++++ drivers/soundwire/slave.c | 7 +- include/linux/soundwire/sdw.h | 83 +++++++++ include/linux/soundwire/sdw_type.h | 36 +++- 9 files changed, 662 insertions(+), 113 deletions(-) create mode 100644 drivers/soundwire/master.c -- 2.20.1

2 5

[alsa-devel] [PATCH 0/3] ASoC: max98090: revert invalid fixes
by Tzung-Bi Shih 14 Feb '20

14 Feb '20

A critical side-effect was observed along with the commit 62d5ae4cafb7 ("ASoC: max98090: save and restore SHDN when changing sensitive registers"). When keep playing something, (e.g. sox -n -t s16 -r 48000 -c 2 - synth sine 440 vol 0.1 | \ aplay -D hw:0,0 -f S16_LE -r 48000 -c 2) the recorded sound has chance to break (clipping). (e.g. while [ 1 ]; do arecord -q -f S16_LE -d 2 -r 48000 -c 2 test.raw \ sox -t s16 -r 48000 -c 2 test.raw -n stat \ ... done) The series reverts all related commits. I feel really sorry for the whole mess. Tzung-Bi Shih (3): ASoC: max98090: revert invalid fix for handling SHDN ASoC: dapm: remove snd_soc_dapm_put_enum_double_locked ASoC: core: remove SND_SOC_BYTES_E include/sound/soc-dapm.h | 2 - include/sound/soc.h | 6 - sound/soc/codecs/max98090.c | 434 ++++++++++-------------------------- sound/soc/codecs/max98090.h | 3 +- sound/soc/soc-dapm.c | 54 ++--- 5 files changed, 137 insertions(+), 362 deletions(-) -- 2.25.0.265.gbab2e86ba0-goog

2 6

[alsa-devel] [PATCH 0/5] ASoC: meson: aiu: fixups
by Jerome Brunet 14 Feb '20

14 Feb '20

This patchset fixes some mistakes which slipped through my initial submission. Most of these issues have been reported by coccinelle and sparse. Jerome Brunet (5): ASoC: meson: aiu: remove unused encoder structure ASoC: meson: aiu: fix clk bulk size allocation ASoC: meson: aiu: fix irq registration ASoC: meson: aiu: fix acodec dai input name init ASoC: meson: codec-glue: fix pcm format cast warning sound/soc/meson/aiu-acodec-ctrl.c | 1 - sound/soc/meson/aiu-encoder-i2s.c | 7 ------- sound/soc/meson/aiu.c | 10 +++------- sound/soc/meson/aiu.h | 2 +- sound/soc/meson/meson-codec-glue.c | 2 +- 5 files changed, 5 insertions(+), 17 deletions(-) -- 2.24.1

2 10

[alsa-devel] alsa-ucm-conf via portage on arm
by Curtis Malainey 14 Feb '20

14 Feb '20

Hello Takashi and Jaroslav, There appears to be a upstream package issue for alsa-ucm-conf in that portage seems to have it restricted to x86 and amd64 platforms. Given the nature of the ucms I figure this is an error and was hoping you know who to speak to in order to resolve this. If this isn't an error what can I do to help get the repo to state in which it is supported on arm/arm64? See https://gitweb.gentoo.org/repo/gentoo.git/commit/media-libs/alsa-ucm-conf/a… Curtis

2 2

[alsa-devel] [PATCH] ALSA: pcm: oss: Unlock mutex temporarily for sleeping at read/write
by Takashi Iwai 14 Feb '20

14 Feb '20

ALSA PCM OSS layer calls the generic __snd_pcm_lib_xfer() helper for the actual transfer of the audio data. The xfer helper may sleep long for waiting for the enough space becoming empty for read/write, and it does unlock/relock for the substream lock. This works fine, so far, but a slight problem specific to OSS layer is that OSS layer wraps yet more mutex (runtime->oss.params_lock) over __snd_pcm_lib_xfer() call; so this mutex is still locked during a possible long sleep, and it prevents the whole ioctl and other actions applied to the given stream. This patch adds the temporarily unlock and relock of the mutex around __snd_pcm_lib_xfer() call in the OSS layer to be more friendly to the concurrent accesses. The long mutex protection itself shouldn't be a real issue for the normal systems, and its influence appears only on strange things like fuzzers. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 707eb2a9d50c..930def8201f4 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1217,8 +1217,10 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const if (ret < 0) break; } + mutex_unlock(&runtime->oss.params_lock); ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true, frames, in_kernel); + mutex_lock(&runtime->oss.params_lock); if (ret != -EPIPE && ret != -ESTRPIPE) break; /* test, if we can't store new data, because the stream */ @@ -1254,8 +1256,10 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p ret = snd_pcm_oss_capture_position_fixup(substream, &delay); if (ret < 0) break; + mutex_unlock(&runtime->oss.params_lock); ret = __snd_pcm_lib_xfer(substream, (void *)ptr, true, frames, in_kernel); + mutex_lock(&runtime->oss.params_lock); if (ret == -EPIPE) { if (runtime->status->state == SNDRV_PCM_STATE_DRAINING) { ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); -- 2.16.4

1 0

[alsa-devel] [PATCH AUTOSEL 4.4 079/100] ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
by Sasha Levin 14 Feb '20

14 Feb '20

From: Peter Große <pegro(a)friiks.de> [ Upstream commit ef7d84caa5928b40b1c93a26dbe5a3f12737c6ab ] Lenovo Thinkpad T420s uses the same codec as T420, so apply the same quirk to enable audio output on a docking station. Signed-off-by: Peter Große <pegro(a)friiks.de> Link: https://lore.kernel.org/r/20200122180106.9351-1-pegro@friiks.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 05e745e2f4271..3150ddfbdb25e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -866,6 +866,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21d2, "Lenovo T420s", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.4 066/100] ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m
by Sasha Levin 14 Feb '20

14 Feb '20

From: Chen Zhou <chenzhou10(a)huawei.com> [ Upstream commit 8fea78029f5e6ed734ae1957bef23cfda1af4354 ] If CONFIG_SND_ATMEL_SOC_DMA=m, build error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register' Function atmel_pcm_dma_platform_register is defined under CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Chen Zhou <chenzhou10(a)huawei.com> Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 2d30464b81cef..d7b471c69f4fb 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -24,6 +24,8 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC_DMA tristate + select SND_ATMEL_SOC_DMA + select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC tristate -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.4 056/100] ALSA: sh: Fix compile warning wrt const
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f1dd4795b1523fbca7ab4344dd5a8bb439cc770d ] A long-standing compile warning was seen during build test: sound/sh/aica.c: In function 'load_aica_firmware': sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Link: https://lore.kernel.org/r/20200105144823.29547-69-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/aica.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index ad3d9ae380349..dd601b39f69ef 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -120,10 +120,10 @@ static void spu_memset(u32 toi, u32 what, int length) } /* spu_memload - write to SPU address space */ -static void spu_memload(u32 toi, void *from, int length) +static void spu_memload(u32 toi, const void *from, int length) { unsigned long flags; - u32 *froml = from; + const u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; u32 val; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.4 042/100] ALSA: usx2y: Adjust indentation in snd_usX2Y_hwdep_dsp_status
by Sasha Levin 14 Feb '20

14 Feb '20

From: Nathan Chancellor <natechancellor(a)gmail.com> [ Upstream commit df4654bd6e42125d9b85ce3a26eaca2935290b98 ] Clang warns: ../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation; statement is not part of the previous 'if' [-Wmisleading-indentation] info->version = USX2Y_DRIVER_VERSION; ^ ../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here if (us428->chip_status & USX2Y_STAT_CHIP_INIT) ^ 1 warning generated. This warning occurs because there is a space before the tab on this line. Remove it so that the indentation is consistent with the Linux kernel coding style and clang no longer warns. This was introduced before the beginning of git history so no fixes tag. Link: https://github.com/ClangBuiltLinux/linux/issues/831 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usX2Yhwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 0b34dbc8f3020..7dcb33d3886bc 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -132,7 +132,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code if (us428->chip_status & USX2Y_STAT_CHIP_INIT) info->chip_ready = 1; - info->version = USX2Y_DRIVER_VERSION; + info->version = USX2Y_DRIVER_VERSION; return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 113/141] ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
by Sasha Levin 14 Feb '20

14 Feb '20

From: Peter Große <pegro(a)friiks.de> [ Upstream commit ef7d84caa5928b40b1c93a26dbe5a3f12737c6ab ] Lenovo Thinkpad T420s uses the same codec as T420, so apply the same quirk to enable audio output on a docking station. Signed-off-by: Peter Große <pegro(a)friiks.de> Link: https://lore.kernel.org/r/20200122180106.9351-1-pegro@friiks.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8557b94e462cb..1e99500dbb6c8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -853,6 +853,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21d2, "Lenovo T420s", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 090/141] ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m
by Sasha Levin 14 Feb '20

14 Feb '20

From: Chen Zhou <chenzhou10(a)huawei.com> [ Upstream commit 8fea78029f5e6ed734ae1957bef23cfda1af4354 ] If CONFIG_SND_ATMEL_SOC_DMA=m, build error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register' Function atmel_pcm_dma_platform_register is defined under CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Chen Zhou <chenzhou10(a)huawei.com> Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 22aec9a1e9a49..838d03a138ca2 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -25,6 +25,8 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC_DMA tristate + select SND_ATMEL_SOC_DMA + select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC tristate -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 078/141] ALSA: sh: Fix compile warning wrt const
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f1dd4795b1523fbca7ab4344dd5a8bb439cc770d ] A long-standing compile warning was seen during build test: sound/sh/aica.c: In function 'load_aica_firmware': sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Link: https://lore.kernel.org/r/20200105144823.29547-69-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/aica.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index fbbc252795599..2a127feb8e293 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -117,10 +117,10 @@ static void spu_memset(u32 toi, u32 what, int length) } /* spu_memload - write to SPU address space */ -static void spu_memload(u32 toi, void *from, int length) +static void spu_memload(u32 toi, const void *from, int length) { unsigned long flags; - u32 *froml = from; + const u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; u32 val; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.9 059/141] ALSA: usx2y: Adjust indentation in snd_usX2Y_hwdep_dsp_status
by Sasha Levin 14 Feb '20

14 Feb '20

From: Nathan Chancellor <natechancellor(a)gmail.com> [ Upstream commit df4654bd6e42125d9b85ce3a26eaca2935290b98 ] Clang warns: ../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation; statement is not part of the previous 'if' [-Wmisleading-indentation] info->version = USX2Y_DRIVER_VERSION; ^ ../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here if (us428->chip_status & USX2Y_STAT_CHIP_INIT) ^ 1 warning generated. This warning occurs because there is a space before the tab on this line. Remove it so that the indentation is consistent with the Linux kernel coding style and clang no longer warns. This was introduced before the beginning of git history so no fixes tag. Link: https://github.com/ClangBuiltLinux/linux/issues/831 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usX2Yhwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 0b34dbc8f3020..7dcb33d3886bc 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -132,7 +132,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code if (us428->chip_status & USX2Y_STAT_CHIP_INIT) info->chip_ready = 1; - info->version = USX2Y_DRIVER_VERSION; + info->version = USX2Y_DRIVER_VERSION; return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 154/186] ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
by Sasha Levin 14 Feb '20

14 Feb '20

From: Peter Große <pegro(a)friiks.de> [ Upstream commit ef7d84caa5928b40b1c93a26dbe5a3f12737c6ab ] Lenovo Thinkpad T420s uses the same codec as T420, so apply the same quirk to enable audio output on a docking station. Signed-off-by: Peter Große <pegro(a)friiks.de> Link: https://lore.kernel.org/r/20200122180106.9351-1-pegro@friiks.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 382b6d2ed8036..9cc9304ff21a7 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -969,6 +969,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21d2, "Lenovo T420s", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 149/186] ALSA: hda/hdmi - add retry logic to parse_intel_hdmi()
by Sasha Levin 14 Feb '20

14 Feb '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 2928fa0a97ebb9549cb877fdc99aed9b95438c3a ] The initial snd_hda_get_sub_node() can fail on certain devices (e.g. some Chromebook models using Intel GLK). The failure rate is very low, but as this is is part of the probe process, end-user impact is high. In observed cases, related hardware status registers have expected values, but the node query still fails. Retrying the node query does seem to help, so fix the problem by adding retry logic to the query. This does not impact non-Intel platforms. BugLink: https://github.com/thesofproject/linux/issues/1642 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200120160117.29130-4-kai.vehmanen@linux.intel.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f214055972150..12913368c2314 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2574,9 +2574,12 @@ static int alloc_intel_hdmi(struct hda_codec *codec) /* parse and post-process for Intel codecs */ static int parse_intel_hdmi(struct hda_codec *codec) { - int err; + int err, retries = 3; + + do { + err = hdmi_parse_codec(codec); + } while (err < 0 && retries--); - err = hdmi_parse_codec(codec); if (err < 0) { generic_spec_free(codec); return err; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 117/186] ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m
by Sasha Levin 14 Feb '20

14 Feb '20

From: Chen Zhou <chenzhou10(a)huawei.com> [ Upstream commit 8fea78029f5e6ed734ae1957bef23cfda1af4354 ] If CONFIG_SND_ATMEL_SOC_DMA=m, build error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register' Function atmel_pcm_dma_platform_register is defined under CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Chen Zhou <chenzhou10(a)huawei.com> Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 4a56f3dfba513..23887613b5c39 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -25,6 +25,8 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC_DMA tristate + select SND_ATMEL_SOC_DMA + select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC tristate -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 099/186] ALSA: sh: Fix compile warning wrt const
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f1dd4795b1523fbca7ab4344dd5a8bb439cc770d ] A long-standing compile warning was seen during build test: sound/sh/aica.c: In function 'load_aica_firmware': sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Link: https://lore.kernel.org/r/20200105144823.29547-69-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/aica.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index fdc680ae8aa09..d9acf551a8985 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -117,10 +117,10 @@ static void spu_memset(u32 toi, u32 what, int length) } /* spu_memload - write to SPU address space */ -static void spu_memload(u32 toi, void *from, int length) +static void spu_memload(u32 toi, const void *from, int length) { unsigned long flags; - u32 *froml = from; + const u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; u32 val; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 098/186] ALSA: sh: Fix unused variable warnings
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5da116f164ce265e397b8f59af5c39e4a61d61a5 ] Remove unused variables that are left over after the conversion of new PCM ops: sound/sh/sh_dac_audio.c:166:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:186:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:205:26: warning: unused variable 'runtime' Fixes: 1cc2f8ba0b3e ("ALSA: sh: Convert to the new PCM ops") Link: https://lore.kernel.org/r/20200104110057.13875-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/sh_dac_audio.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index 834b2574786f5..6251b5e1b64a2 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -190,7 +190,6 @@ static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; if (copy_from_user_toio(chip->data_buffer + pos, src, count)) return -EFAULT; @@ -210,7 +209,6 @@ static int snd_sh_dac_pcm_copy_kernel(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memcpy_toio(chip->data_buffer + pos, src, count); chip->buffer_end = chip->data_buffer + pos + count; @@ -229,7 +227,6 @@ static int snd_sh_dac_pcm_silence(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memset_io(chip->data_buffer + pos, 0, count); chip->buffer_end = chip->data_buffer + pos + count; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 076/186] ALSA: usx2y: Adjust indentation in snd_usX2Y_hwdep_dsp_status
by Sasha Levin 14 Feb '20

14 Feb '20

From: Nathan Chancellor <natechancellor(a)gmail.com> [ Upstream commit df4654bd6e42125d9b85ce3a26eaca2935290b98 ] Clang warns: ../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation; statement is not part of the previous 'if' [-Wmisleading-indentation] info->version = USX2Y_DRIVER_VERSION; ^ ../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here if (us428->chip_status & USX2Y_STAT_CHIP_INIT) ^ 1 warning generated. This warning occurs because there is a space before the tab on this line. Remove it so that the indentation is consistent with the Linux kernel coding style and clang no longer warns. This was introduced before the beginning of git history so no fixes tag. Link: https://github.com/ClangBuiltLinux/linux/issues/831 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usX2Yhwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index f4b3cda412fcc..e75271e731b2d 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -131,7 +131,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code if (us428->chip_status & USX2Y_STAT_CHIP_INIT) info->chip_ready = 1; - info->version = USX2Y_DRIVER_VERSION; + info->version = USX2Y_DRIVER_VERSION; return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.14 011/186] ALSA: ctl: allow TLV read operation for callback type of element in locked case
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d61fe22c2ae42d9fd76c34ef4224064cca4b04b0 ] A design of ALSA control core allows applications to execute three operations for TLV feature; read, write and command. Furthermore, it allows driver developers to process the operations by two ways; allocated array or callback function. In the former, read operation is just allowed, thus developers uses the latter when device driver supports variety of models or the target model is expected to dynamically change information stored in TLV container. The core also allows applications to lock any element so that the other applications can't perform write operation to the element for element value and TLV information. When the element is locked, write and command operation for TLV information are prohibited as well as element value. Any read operation should be allowed in the case. At present, when an element has callback function for TLV information, TLV read operation returns EPERM if the element is locked. On the other hand, the read operation is success when an element has allocated array for TLV information. In both cases, read operation is success for element value expectedly. This commit fixes the bug. This change can be backported to v4.14 kernel or later. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20191223093347.15279-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/control.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 36571cd49be33..a0ce22164957c 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1467,8 +1467,9 @@ static int call_tlv_handler(struct snd_ctl_file *file, int op_flag, if (kctl->tlv.c == NULL) return -ENXIO; - /* When locked, this is unavailable. */ - if (vd->owner != NULL && vd->owner != file) + /* Write and command operations are not allowed for locked element. */ + if (op_flag != SNDRV_CTL_TLV_OP_READ && + vd->owner != NULL && vd->owner != file) return -EPERM; return kctl->tlv.c(kctl, op_flag, size, buf); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 207/252] ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
by Sasha Levin 14 Feb '20

14 Feb '20

From: Peter Große <pegro(a)friiks.de> [ Upstream commit ef7d84caa5928b40b1c93a26dbe5a3f12737c6ab ] Lenovo Thinkpad T420s uses the same codec as T420, so apply the same quirk to enable audio output on a docking station. Signed-off-by: Peter Große <pegro(a)friiks.de> Link: https://lore.kernel.org/r/20200122180106.9351-1-pegro@friiks.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5500dd437b44d..78bb96263bc27 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -935,6 +935,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21d2, "Lenovo T420s", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 200/252] ALSA: hda/hdmi - add retry logic to parse_intel_hdmi()
by Sasha Levin 14 Feb '20

14 Feb '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 2928fa0a97ebb9549cb877fdc99aed9b95438c3a ] The initial snd_hda_get_sub_node() can fail on certain devices (e.g. some Chromebook models using Intel GLK). The failure rate is very low, but as this is is part of the probe process, end-user impact is high. In observed cases, related hardware status registers have expected values, but the node query still fails. Retrying the node query does seem to help, so fix the problem by adding retry logic to the query. This does not impact non-Intel platforms. BugLink: https://github.com/thesofproject/linux/issues/1642 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200120160117.29130-4-kai.vehmanen@linux.intel.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c827a2a89cc3d..c67fadd5aae53 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2604,9 +2604,12 @@ static int alloc_intel_hdmi(struct hda_codec *codec) /* parse and post-process for Intel codecs */ static int parse_intel_hdmi(struct hda_codec *codec) { - int err; + int err, retries = 3; + + do { + err = hdmi_parse_codec(codec); + } while (err < 0 && retries--); - err = hdmi_parse_codec(codec); if (err < 0) { generic_spec_free(codec); return err; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 155/252] ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m
by Sasha Levin 14 Feb '20

14 Feb '20

From: Chen Zhou <chenzhou10(a)huawei.com> [ Upstream commit 8fea78029f5e6ed734ae1957bef23cfda1af4354 ] If CONFIG_SND_ATMEL_SOC_DMA=m, build error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register' Function atmel_pcm_dma_platform_register is defined under CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Chen Zhou <chenzhou10(a)huawei.com> Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 64b784e96f844..dad778e5884bd 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -25,6 +25,8 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC_DMA tristate + select SND_ATMEL_SOC_DMA + select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC tristate -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 130/252] ALSA: sh: Fix compile warning wrt const
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f1dd4795b1523fbca7ab4344dd5a8bb439cc770d ] A long-standing compile warning was seen during build test: sound/sh/aica.c: In function 'load_aica_firmware': sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Link: https://lore.kernel.org/r/20200105144823.29547-69-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/aica.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index ad3f71358486a..69ac44b335602 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -117,10 +117,10 @@ static void spu_memset(u32 toi, u32 what, int length) } /* spu_memload - write to SPU address space */ -static void spu_memload(u32 toi, void *from, int length) +static void spu_memload(u32 toi, const void *from, int length) { unsigned long flags; - u32 *froml = from; + const u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; u32 val; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 128/252] ALSA: sh: Fix unused variable warnings
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5da116f164ce265e397b8f59af5c39e4a61d61a5 ] Remove unused variables that are left over after the conversion of new PCM ops: sound/sh/sh_dac_audio.c:166:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:186:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:205:26: warning: unused variable 'runtime' Fixes: 1cc2f8ba0b3e ("ALSA: sh: Convert to the new PCM ops") Link: https://lore.kernel.org/r/20200104110057.13875-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/sh_dac_audio.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index 834b2574786f5..6251b5e1b64a2 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -190,7 +190,6 @@ static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; if (copy_from_user_toio(chip->data_buffer + pos, src, count)) return -EFAULT; @@ -210,7 +209,6 @@ static int snd_sh_dac_pcm_copy_kernel(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memcpy_toio(chip->data_buffer + pos, src, count); chip->buffer_end = chip->data_buffer + pos + count; @@ -229,7 +227,6 @@ static int snd_sh_dac_pcm_silence(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memset_io(chip->data_buffer + pos, 0, count); chip->buffer_end = chip->data_buffer + pos + count; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 101/252] ALSA: usx2y: Adjust indentation in snd_usX2Y_hwdep_dsp_status
by Sasha Levin 14 Feb '20

14 Feb '20

From: Nathan Chancellor <natechancellor(a)gmail.com> [ Upstream commit df4654bd6e42125d9b85ce3a26eaca2935290b98 ] Clang warns: ../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation; statement is not part of the previous 'if' [-Wmisleading-indentation] info->version = USX2Y_DRIVER_VERSION; ^ ../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here if (us428->chip_status & USX2Y_STAT_CHIP_INIT) ^ 1 warning generated. This warning occurs because there is a space before the tab on this line. Remove it so that the indentation is consistent with the Linux kernel coding style and clang no longer warns. This was introduced before the beginning of git history so no fixes tag. Link: https://github.com/ClangBuiltLinux/linux/issues/831 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usX2Yhwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index c1dd9a7b48df6..36b3459703640 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -131,7 +131,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code if (us428->chip_status & USX2Y_STAT_CHIP_INIT) info->chip_ready = 1; - info->version = USX2Y_DRIVER_VERSION; + info->version = USX2Y_DRIVER_VERSION; return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 4.19 011/252] ALSA: ctl: allow TLV read operation for callback type of element in locked case
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d61fe22c2ae42d9fd76c34ef4224064cca4b04b0 ] A design of ALSA control core allows applications to execute three operations for TLV feature; read, write and command. Furthermore, it allows driver developers to process the operations by two ways; allocated array or callback function. In the former, read operation is just allowed, thus developers uses the latter when device driver supports variety of models or the target model is expected to dynamically change information stored in TLV container. The core also allows applications to lock any element so that the other applications can't perform write operation to the element for element value and TLV information. When the element is locked, write and command operation for TLV information are prohibited as well as element value. Any read operation should be allowed in the case. At present, when an element has callback function for TLV information, TLV read operation returns EPERM if the element is locked. On the other hand, the read operation is success when an element has allocated array for TLV information. In both cases, read operation is success for element value expectedly. This commit fixes the bug. This change can be backported to v4.14 kernel or later. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20191223093347.15279-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/control.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 649d3217590ed..d1312f14d78fb 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1468,8 +1468,9 @@ static int call_tlv_handler(struct snd_ctl_file *file, int op_flag, if (kctl->tlv.c == NULL) return -ENXIO; - /* When locked, this is unavailable. */ - if (vd->owner != NULL && vd->owner != file) + /* Write and command operations are not allowed for locked element. */ + if (op_flag != SNDRV_CTL_TLV_OP_READ && + vd->owner != NULL && vd->owner != file) return -EPERM; return kctl->tlv.c(kctl, op_flag, size, buf); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 404/459] ALSA: usb-audio: add quirks for Line6 Helix devices fw>=2.82
by Sasha Levin 14 Feb '20

14 Feb '20

From: Nicola Lunghi <nick83ola(a)gmail.com> [ Upstream commit b81cbf7abfc94878a3c6f0789f2185ee55b1cc21 ] With firmware 2.82 Line6 changed the usb id of some of the Helix devices but the quirks is still needed. Add it to the quirk list for line6 helix family of devices. Thanks to Jens for pointing out the missing ids. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> Link: https://lore.kernel.org/r/20200125150917.5040-1-nick83ola@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/format.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db71305f63..53922f73467f4 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -296,6 +296,9 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0E41, 0x4242): /* Line6 Helix Rack */ case USB_ID(0x0E41, 0x4244): /* Line6 Helix LT */ case USB_ID(0x0E41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0E41, 0x4248): /* Line6 Helix >= fw 2.82 */ + case USB_ID(0x0E41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ + case USB_ID(0x0E41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ /* supported rates: 48Khz */ kfree(fp->rate_table); fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 377/459] ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
by Sasha Levin 14 Feb '20

14 Feb '20

From: Peter Große <pegro(a)friiks.de> [ Upstream commit ef7d84caa5928b40b1c93a26dbe5a3f12737c6ab ] Lenovo Thinkpad T420s uses the same codec as T420, so apply the same quirk to enable audio output on a docking station. Signed-off-by: Peter Große <pegro(a)friiks.de> Link: https://lore.kernel.org/r/20200122180106.9351-1-pegro@friiks.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 90aa0f400a57d..1e20e85e9b466 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -922,6 +922,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21d2, "Lenovo T420s", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 366/459] ALSA: hda/hdmi - add retry logic to parse_intel_hdmi()
by Sasha Levin 14 Feb '20

14 Feb '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 2928fa0a97ebb9549cb877fdc99aed9b95438c3a ] The initial snd_hda_get_sub_node() can fail on certain devices (e.g. some Chromebook models using Intel GLK). The failure rate is very low, but as this is is part of the probe process, end-user impact is high. In observed cases, related hardware status registers have expected values, but the node query still fails. Retrying the node query does seem to help, so fix the problem by adding retry logic to the query. This does not impact non-Intel platforms. BugLink: https://github.com/thesofproject/linux/issues/1642 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200120160117.29130-4-kai.vehmanen@linux.intel.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8ac805a634f4e..307ca1f036762 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2794,9 +2794,12 @@ static int alloc_intel_hdmi(struct hda_codec *codec) /* parse and post-process for Intel codecs */ static int parse_intel_hdmi(struct hda_codec *codec) { - int err; + int err, retries = 3; + + do { + err = hdmi_parse_codec(codec); + } while (err < 0 && retries--); - err = hdmi_parse_codec(codec); if (err < 0) { generic_spec_free(codec); return err; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 349/459] ASoC: SOF: Intel: hda: Fix SKL dai count
by Sasha Levin 14 Feb '20

14 Feb '20

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit a6947c9d86bcfd61b758b5693eba58defe7fd2ae ] With fourth pin added for iDisp for skl_dai, update SOF_SKL_DAI_NUM to account for the change. Without this, dais from the bottom of the list are skipped. In current state that's the case for 'Alt Analog CPU DAI'. Fixes: ac42b142cd76 ("ASoC: SOF: Intel: hda: Add iDisp4 DAI") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200113114054.9716-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 23e430d3e0568..4be53ef2eab6e 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -336,7 +336,7 @@ /* Number of DAIs */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) -#define SOF_SKL_NUM_DAIS 14 +#define SOF_SKL_NUM_DAIS 15 #else #define SOF_SKL_NUM_DAIS 8 #endif -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 331/459] ALSA: usb-audio: add implicit fb quirk for MOTU M Series
by Sasha Levin 14 Feb '20

14 Feb '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit c249177944b650816069f6c49b769baaa94339dc ] This fixes crackling sound during playback. Further note: MOTU is known for reusing Product IDs for different devices or different generations of the device (e.g. MicroBook I/II/IIc shares a single Product ID). This patch was only tested with M4 audio interface, but the same Product ID is also used by M2. Hope it will work for M2 as well. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200115151358.56672-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index fa24bd491cf6a..ad8f38380aa3e 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + ep = 0x81; + ifnum = 2; + goto add_sync_ep_from_ifnum; case USB_ID(0x0582, 0x01d8): /* BOSS Katana */ /* BOSS Katana amplifiers do not need quirks */ return 0; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 323/459] ALSA: usb-audio: unlock on error in probe
by Sasha Levin 14 Feb '20

14 Feb '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit a3afa29942b84b4e2548beacccc3a68b8d77e3dc ] We need to unlock before we returning on this error path. Fixes: 73ac9f5e5b43 ("ALSA: usb-audio: Add boot quirk for MOTU M Series") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200115174604.rhanfgy4j3uc65cx@kili.mountain Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index e6a618a239948..54f9ce38471e6 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -599,7 +599,7 @@ static int usb_audio_probe(struct usb_interface *intf, if (! chip) { err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); if (err < 0) - return err; + goto __error; /* it's a fresh one. * now look for an empty slot and create a new card instance -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 303/459] ASoC: SOF: Intel: hda-dai: fix compilation warning in pcm_prepare
by Sasha Levin 14 Feb '20

14 Feb '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit d873997192ddcacb5333575502be2f91ea4b47b8 ] Fix GCC warning with W=1, previous cleanup did not remove unnecessary variable. sound/soc/sof/intel/hda-dai.c: In function ‘hda_link_pcm_prepare’: sound/soc/sof/intel/hda-dai.c:265:31: warning: variable ‘hda_stream’ set but not used [-Wunused-but-set-variable] 265 | struct sof_intel_hda_stream *hda_stream; | ^~~~~~~~~~ Fixes: a3ebccb52efdf ("ASoC: SOF: Intel: hda: reset link DMA state in prepare") Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200113205620.27285-1-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-dai.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 896d21984b735..1923b0c36bcef 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -261,14 +261,11 @@ static int hda_link_pcm_prepare(struct snd_pcm_substream *substream, { struct hdac_ext_stream *link_dev = snd_soc_dai_get_dma_data(dai, substream); - struct sof_intel_hda_stream *hda_stream; struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(dai->component); struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); int stream = substream->stream; - hda_stream = hstream_to_sof_hda_stream(link_dev); - if (link_dev->link_prepared) return 0; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 292/459] ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m
by Sasha Levin 14 Feb '20

14 Feb '20

From: Chen Zhou <chenzhou10(a)huawei.com> [ Upstream commit 8fea78029f5e6ed734ae1957bef23cfda1af4354 ] If CONFIG_SND_ATMEL_SOC_DMA=m, build error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register' Function atmel_pcm_dma_platform_register is defined under CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Chen Zhou <chenzhou10(a)huawei.com> Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c229ed829..d1dc8e6366dcb 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -19,6 +19,8 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC tristate + select SND_ATMEL_SOC_DMA + select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC_PDC tristate "SoC PCM DAI support for AT91 SSC controller using PDC" -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 291/459] ALSA: usb-audio: Add boot quirk for MOTU M Series
by Sasha Levin 14 Feb '20

14 Feb '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit 73ac9f5e5b43a5dbadb61f27dae7a971f7ec0d22 ] Add delay to make sure that audio urbs are not sent too early. Otherwise the device hangs. Windows driver makes ~2s delay, so use about the same time delay value. snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which is an overkill. Thus a quirk that is called only once is implemented. Also send two vendor-specific control messages before and after the delay. This behaviour is blindly copied from the Windows driver. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200112102358.18085-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.c | 4 ++++ sound/usb/quirks.c | 38 ++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 5 +++++ 3 files changed, 47 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index db91dc76cc915..e6a618a239948 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -597,6 +597,10 @@ static int usb_audio_probe(struct usb_interface *intf, } } if (! chip) { + err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); + if (err < 0) + return err; + /* it's a fresh one. * now look for an empty slot and create a new card instance */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 82184036437b5..0464df785f2ee 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1113,6 +1113,31 @@ static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) return err; } +static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) +{ + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x0, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + msleep(2000); + + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x20, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + return 0; +} + /* * Setup quirks */ @@ -1297,6 +1322,19 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return 0; } +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int id) +{ + switch (id) { + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + return snd_usb_motu_m_series_boot_quirk(dev); + } + + return 0; +} + /* * check if the device uses big-endian samples */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index a80e0ddd07364..df0355843a4c1 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -20,6 +20,11 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, const struct snd_usb_audio_quirk *quirk, unsigned int usb_id); +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id); + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 250/459] ASoC: Intel: sof_rt5682: Ignore the speaker amp when there isn't one.
by Sasha Levin 14 Feb '20

14 Feb '20

From: Sam McNally <sammc(a)chromium.org> [ Upstream commit d4b74e218a8d0d6cf58e546627ab9d4d4f2645ab ] Some members of the Google_Hatch family include a rt5682 jack codec, but no speaker amplifier. This uses the same driver (sof_rt5682) as a combination of rt5682 jack codec and max98357a speaker amplifier. Within the sof_rt5682 driver, these cases are not currently distinguishable, relying on a DMI quirk to decide the configuration. This causes an incorrect configuration when only the rt5682 is present on a Google_Hatch device. For CML, the jack codec is used as the primary key when matching, with a possible speaker amplifier described in quirk_data. The two cases of interest are the second and third 10EC5682 entries in snd_soc_acpi_intel_cml_machines[]. The second entry matches the combination of rt5682 and max98357a, resulting in the quirk_data field in the snd_soc_acpi_mach being non-null, pointing at max98357a_spk_codecs, the snd_soc_acpi_codecs for the matched speaker amplifier. The third entry matches just the rt5682, resulting in a null quirk_data. The sof_rt5682 driver's DMI data matching identifies that a speaker amplifier is present for all Google_Hatch family devices. Detect cases where there is no speaker amplifier by checking for a null quirk_data in the snd_soc_acpi_mach and remove the speaker amplifier bit in that case. Signed-off-by: Sam McNally <sammc(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200103124921.v3.1.Ib87c4a7fbb3fc818ea12198e291b… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 9441ddfeea5e6..06b7d6c6c9a04 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -594,6 +594,14 @@ static int sof_audio_probe(struct platform_device *pdev) dmi_check_system(sof_rt5682_quirk_table); + mach = (&pdev->dev)->platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -654,7 +662,6 @@ static int sof_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_rt5682.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682, -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 246/459] ALSA: sh: Fix compile warning wrt const
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f1dd4795b1523fbca7ab4344dd5a8bb439cc770d ] A long-standing compile warning was seen during build test: sound/sh/aica.c: In function 'load_aica_firmware': sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Link: https://lore.kernel.org/r/20200105144823.29547-69-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/aica.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 52e9cfb4f8197..8421b2f9c9f38 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -101,10 +101,10 @@ static void spu_memset(u32 toi, u32 what, int length) } /* spu_memload - write to SPU address space */ -static void spu_memload(u32 toi, void *from, int length) +static void spu_memload(u32 toi, const void *from, int length) { unsigned long flags; - u32 *froml = from; + const u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; u32 val; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 245/459] ALSA: hda/realtek - Apply mic mute LED quirk for Dell E7xx laptops, too
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5fab5829674c279839a7408ab30c71c6dfe726b9 ] Dell E7xx laptops have also mic mute LED that is driven by the dell-laptop platform driver. Bind it with the capture control as already done for other models. A caveat is that the fixup hook for the mic mute LED has to be applied at last, otherwise it results in the invalid override of the callback. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205529 Link: https://lore.kernel.org/r/20200105081119.21396-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 68832f52c1ad2..e8d5f6befa1f3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5848,6 +5848,7 @@ enum { ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, ALC288_FIXUP_DISABLE_AAMIX, + ALC292_FIXUP_DELL_E7X_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, @@ -6543,12 +6544,19 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE }, - [ALC292_FIXUP_DELL_E7X] = { + [ALC292_FIXUP_DELL_E7X_AAMIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_xps13, .chained = true, .chain_id = ALC292_FIXUP_DISABLE_AAMIX }, + [ALC292_FIXUP_DELL_E7X] = { + .type = HDA_FIXUP_FUNC, + .v.func = snd_hda_gen_fixup_micmute_led, + /* micmute fixup must be applied at last */ + .chained_before = true, + .chain_id = ALC292_FIXUP_DELL_E7X_AAMIX, + }, [ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.4 242/459] ALSA: sh: Fix unused variable warnings
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5da116f164ce265e397b8f59af5c39e4a61d61a5 ] Remove unused variables that are left over after the conversion of new PCM ops: sound/sh/sh_dac_audio.c:166:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:186:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:205:26: warning: unused variable 'runtime' Fixes: 1cc2f8ba0b3e ("ALSA: sh: Convert to the new PCM ops") Link: https://lore.kernel.org/r/20200104110057.13875-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/sh_dac_audio.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index ed877a138965d..7c46494466ff1 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -175,7 +175,6 @@ static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; if (copy_from_user_toio(chip->data_buffer + pos, src, count)) return -EFAULT; @@ -195,7 +194,6 @@ static int snd_sh_dac_pcm_copy_kernel(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memcpy_toio(chip->data_buffer + pos, src, count); chip->buffer_end = chip->data_buffer + pos + count; @@ -214,7 +212,6 @@ static int snd_sh_dac_pcm_silence(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memset_io(chip->data_buffer + pos, 0, count); chip->buffer_end = chip->data_buffer + pos + count; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 238/459] ASoC: soc-topology: fix endianness issues
by Sasha Levin 14 Feb '20

14 Feb '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 72bbeda0222bcd382ee33b3aff71346074410c21 ] Sparse complains about a series of easy warnings, fix. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200102195952.9465-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-topology.c | 42 +++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index fef01e1dd15c5..d00203ef8305f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -604,9 +604,11 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, ext_ops = tplg->bytes_ext_ops; num_ops = tplg->bytes_ext_ops_count; for (i = 0; i < num_ops; i++) { - if (!sbe->put && ext_ops[i].id == be->ext_ops.put) + if (!sbe->put && + ext_ops[i].id == le32_to_cpu(be->ext_ops.put)) sbe->put = ext_ops[i].put; - if (!sbe->get && ext_ops[i].id == be->ext_ops.get) + if (!sbe->get && + ext_ops[i].id == le32_to_cpu(be->ext_ops.get)) sbe->get = ext_ops[i].get; } @@ -621,11 +623,11 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, num_ops = tplg->io_ops_count; for (i = 0; i < num_ops; i++) { - if (k->put == NULL && ops[i].id == hdr->ops.put) + if (k->put == NULL && ops[i].id == le32_to_cpu(hdr->ops.put)) k->put = ops[i].put; - if (k->get == NULL && ops[i].id == hdr->ops.get) + if (k->get == NULL && ops[i].id == le32_to_cpu(hdr->ops.get)) k->get = ops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && ops[i].id == le32_to_cpu(hdr->ops.info)) k->info = ops[i].info; } @@ -638,11 +640,11 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, num_ops = ARRAY_SIZE(io_ops); for (i = 0; i < num_ops; i++) { - if (k->put == NULL && ops[i].id == hdr->ops.put) + if (k->put == NULL && ops[i].id == le32_to_cpu(hdr->ops.put)) k->put = ops[i].put; - if (k->get == NULL && ops[i].id == hdr->ops.get) + if (k->get == NULL && ops[i].id == le32_to_cpu(hdr->ops.get)) k->get = ops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && ops[i].id == le32_to_cpu(hdr->ops.info)) k->info = ops[i].info; } @@ -931,7 +933,7 @@ static int soc_tplg_denum_create_texts(struct soc_enum *se, if (se->dobj.control.dtexts == NULL) return -ENOMEM; - for (i = 0; i < ec->items; i++) { + for (i = 0; i < le32_to_cpu(ec->items); i++) { if (strnlen(ec->texts[i], SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == SNDRV_CTL_ELEM_ID_NAME_MAXLEN) { @@ -1325,7 +1327,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( if (kc[i].name == NULL) goto err_sm; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = mc->hdr.access; + kc[i].access = le32_to_cpu(mc->hdr.access); /* we only support FL/FR channel mapping atm */ sm->reg = tplc_chan_get_reg(tplg, mc->channel, @@ -1337,10 +1339,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( sm->rshift = tplc_chan_get_shift(tplg, mc->channel, SNDRV_CHMAP_FR); - sm->max = mc->max; - sm->min = mc->min; - sm->invert = mc->invert; - sm->platform_max = mc->platform_max; + sm->max = le32_to_cpu(mc->max); + sm->min = le32_to_cpu(mc->min); + sm->invert = le32_to_cpu(mc->invert); + sm->platform_max = le32_to_cpu(mc->platform_max); sm->dobj.index = tplg->index; INIT_LIST_HEAD(&sm->dobj.list); @@ -1401,7 +1403,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( goto err_se; tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); + le32_to_cpu(ec->priv.size)); dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", ec->hdr.name); @@ -1411,7 +1413,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( if (kc[i].name == NULL) goto err_se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = ec->hdr.access; + kc[i].access = le32_to_cpu(ec->hdr.access); /* we only support FL/FR channel mapping atm */ se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); @@ -1420,8 +1422,8 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( se->shift_r = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FR); - se->items = ec->items; - se->mask = ec->mask; + se->items = le32_to_cpu(ec->items); + se->mask = le32_to_cpu(ec->mask); se->dobj.index = tplg->index; switch (le32_to_cpu(ec->hdr.ops.info)) { @@ -1523,9 +1525,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( if (kc[i].name == NULL) goto err_sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = be->hdr.access; + kc[i].access = le32_to_cpu(be->hdr.access); - sbe->max = be->max; + sbe->max = le32_to_cpu(be->max); INIT_LIST_HEAD(&sbe->dobj.list); /* map standard io handlers and check for external handlers */ -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 225/459] ASoC: Intel: kbl_da7219_max98357a: remove unused variable 'constraints_16000' and 'ch_mono'
by Sasha Levin 14 Feb '20

14 Feb '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit c5614fb8e3d13be7bba79f71b798468a3a6224f7 ] sound/soc/intel/boards/kbl_da7219_max98357a.c:343:48: warning: constraints_16000 defined but not used [-Wunused-const-variable=] sound/soc/intel/boards/kbl_da7219_max98357a.c:348:27: warning: ch_mono defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20191224140237.36732-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb69..0d55319a0773c 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -336,19 +336,6 @@ static struct snd_soc_ops kabylake_dmic_ops = { .startup = kabylake_dmic_startup, }; -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 194/459] ALSA: usx2y: Adjust indentation in snd_usX2Y_hwdep_dsp_status
by Sasha Levin 14 Feb '20

14 Feb '20

From: Nathan Chancellor <natechancellor(a)gmail.com> [ Upstream commit df4654bd6e42125d9b85ce3a26eaca2935290b98 ] Clang warns: ../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation; statement is not part of the previous 'if' [-Wmisleading-indentation] info->version = USX2Y_DRIVER_VERSION; ^ ../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here if (us428->chip_status & USX2Y_STAT_CHIP_INIT) ^ 1 warning generated. This warning occurs because there is a space before the tab on this line. Remove it so that the indentation is consistent with the Linux kernel coding style and clang no longer warns. This was introduced before the beginning of git history so no fixes tag. Link: https://github.com/ClangBuiltLinux/linux/issues/831 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usX2Yhwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index d1caa8ed9e681..9985fc139487b 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -119,7 +119,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code if (us428->chip_status & USX2Y_STAT_CHIP_INIT) info->chip_ready = 1; - info->version = USX2Y_DRIVER_VERSION; + info->version = USX2Y_DRIVER_VERSION; return 0; } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 152/459] ASoC: intel: sof_rt5682: Add support for tgl-max98357a-rt5682
by Sasha Levin 14 Feb '20

14 Feb '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> [ Upstream commit 6605f0ca3af3b964635287ec7c9dadc812b78eb0 ] This patch adds the driver data and updates quirk info for tgl with max98357a speaker amp and ALC5682 headset codec. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191126143205.21987-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 377ff17dedb98..9441ddfeea5e6 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -589,6 +589,9 @@ static int sof_audio_probe(struct platform_device *pdev) if (!ctx) return -ENOMEM; + if (pdev->id_entry && pdev->id_entry->driver_data) + sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data; + dmi_check_system(sof_rt5682_quirk_table); if (soc_intel_is_byt() || soc_intel_is_cht()) { @@ -680,6 +683,21 @@ static int sof_rt5682_remove(struct platform_device *pdev) return 0; } +static const struct platform_device_id board_ids[] = { + { + .name = "sof_rt5682", + }, + { + .name = "tgl_max98357a_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { } +}; + static struct platform_driver sof_audio = { .probe = sof_audio_probe, .remove = sof_rt5682_remove, @@ -687,6 +705,7 @@ static struct platform_driver sof_audio = { .name = "sof_rt5682", .pm = &snd_soc_pm_ops, }, + .id_table = board_ids, }; module_platform_driver(sof_audio) @@ -696,3 +715,4 @@ MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); +MODULE_ALIAS("platform:tgl_max98357a_rt5682"); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 151/459] ASoC: intel: sof_rt5682: Add quirk for number of HDMI DAI's
by Sasha Levin 14 Feb '20

14 Feb '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> [ Upstream commit c68e07970eca79106b0c35b88a12298569590081 ] TGL supports one more HDMI DAI than previous models. So add quirk support for number of HDMI DAI's. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191126143205.21987-2-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 4f6e58c3954a2..377ff17dedb98 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -34,6 +34,10 @@ #define SOF_RT5682_SSP_AMP(quirk) \ (((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK) #define SOF_RT5682_MCLK_BYTCHT_EN BIT(9) +#define SOF_RT5682_NUM_HDMIDEV_SHIFT 10 +#define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) +#define SOF_RT5682_NUM_HDMIDEV(quirk) \ + ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -585,6 +589,8 @@ static int sof_audio_probe(struct platform_device *pdev) if (!ctx) return -ENOMEM; + dmi_check_system(sof_rt5682_quirk_table); + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -595,11 +601,13 @@ static int sof_audio_probe(struct platform_device *pdev) SOF_RT5682_SSP_CODEC(2); } else { dmic_be_num = 2; - hdmi_num = 3; + hdmi_num = (sof_rt5682_quirk & SOF_RT5682_NUM_HDMIDEV_MASK) >> + SOF_RT5682_NUM_HDMIDEV_SHIFT; + /* default number of HDMI DAI's */ + if (!hdmi_num) + hdmi_num = 3; } - dmi_check_system(sof_rt5682_quirk_table); - /* need to get main clock from pmc */ if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.4 027/459] ALSA: ctl: allow TLV read operation for callback type of element in locked case
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d61fe22c2ae42d9fd76c34ef4224064cca4b04b0 ] A design of ALSA control core allows applications to execute three operations for TLV feature; read, write and command. Furthermore, it allows driver developers to process the operations by two ways; allocated array or callback function. In the former, read operation is just allowed, thus developers uses the latter when device driver supports variety of models or the target model is expected to dynamically change information stored in TLV container. The core also allows applications to lock any element so that the other applications can't perform write operation to the element for element value and TLV information. When the element is locked, write and command operation for TLV information are prohibited as well as element value. Any read operation should be allowed in the case. At present, when an element has callback function for TLV information, TLV read operation returns EPERM if the element is locked. On the other hand, the read operation is success when an element has allocated array for TLV information. In both cases, read operation is success for element value expectedly. This commit fixes the bug. This change can be backported to v4.14 kernel or later. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20191223093347.15279-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/control.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 7a4d8690ce41f..08ca7666e84cf 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1430,8 +1430,9 @@ static int call_tlv_handler(struct snd_ctl_file *file, int op_flag, if (kctl->tlv.c == NULL) return -ENXIO; - /* When locked, this is unavailable. */ - if (vd->owner != NULL && vd->owner != file) + /* Write and command operations are not allowed for locked element. */ + if (op_flag != SNDRV_CTL_TLV_OP_READ && + vd->owner != NULL && vd->owner != file) return -EPERM; return kctl->tlv.c(kctl, op_flag, size, buf); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 489/542] ASoC: Intel: consistent HDMI codec probing code
by Sasha Levin 14 Feb '20

14 Feb '20

From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> [ Upstream commit 98ff5c262f27aafee077a4c096f71a8566e9e948 ] Multiple Intel ASoC machine drivers repeat the same pattern in their .late_probe() methods: they first check whether the common HDMI codec driver is used, if not, they proceed by linking the legacy HDMI driver to each HDMI port. While doing that they use some inconsistent code: 1. after the loop they check, whether the list contained at least one element and if not, they return an error. However, the earlier code to use the common HDMI driver uses the first element of the same list without checking. To fix this we move the check to the top of the function. 2. some of those .late_probe() implementations execute code, only needed for the common HDMI driver, before checking, whether the driver is used. Move the code to after the check. 3. Some of those functions also perform a redundant initialisation of the "err" variable. This patch fixes those issues. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200124213625.30186-8-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 14 +++++++------- sound/soc/intel/boards/bxt_rt298.c | 14 +++++++------- sound/soc/intel/boards/cml_rt1011_rt5682.c | 13 +++++++------ sound/soc/intel/boards/glk_rt5682_max98357a.c | 16 ++++++++-------- sound/soc/intel/boards/sof_rt5682.c | 15 ++++++++------- 5 files changed, 37 insertions(+), 35 deletions(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 5873abb46441c..749b1c4f1ceec 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -617,12 +617,15 @@ static int bxt_card_late_probe(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, broxton_map, ARRAY_SIZE(broxton_map)); - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct bxt_hdmi_pcm, - head); - component = pcm->codec_dai->component; + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; - if (ctx->common_hdmi_codec_drv) + if (ctx->common_hdmi_codec_drv) { + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct bxt_hdmi_pcm, + head); + component = pcm->codec_dai->component; return hda_dsp_hdmi_build_controls(card, component); + } list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; @@ -643,9 +646,6 @@ static int bxt_card_late_probe(struct snd_soc_card *card) i++; } - if (!component) - return -EINVAL; - return hdac_hdmi_jack_port_init(component, &card->dapm); } diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index eabf9d8468ae5..becfc4fc1aff3 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -529,12 +529,15 @@ static int bxt_card_late_probe(struct snd_soc_card *card) int err, i = 0; char jack_name[NAME_SIZE]; - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct bxt_hdmi_pcm, - head); - component = pcm->codec_dai->component; + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; - if (ctx->common_hdmi_codec_drv) + if (ctx->common_hdmi_codec_drv) { + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct bxt_hdmi_pcm, + head); + component = pcm->codec_dai->component; return hda_dsp_hdmi_build_controls(card, component); + } list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; @@ -555,9 +558,6 @@ static int bxt_card_late_probe(struct snd_soc_card *card) i++; } - if (!component) - return -EINVAL; - return hdac_hdmi_jack_port_init(component, &card->dapm); } diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index 5f1bf6d3800c6..a54636f77c8e6 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -241,12 +241,15 @@ static int sof_card_late_probe(struct snd_soc_card *card) struct hdmi_pcm *pcm; int ret, i = 0; - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm, - head); - component = pcm->codec_dai->component; + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; - if (ctx->common_hdmi_codec_drv) + if (ctx->common_hdmi_codec_drv) { + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm, + head); + component = pcm->codec_dai->component; return hda_dsp_hdmi_build_controls(card, component); + } list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; @@ -265,8 +268,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) i++; } - if (!component) - return -EINVAL; return hdac_hdmi_jack_port_init(component, &card->dapm); } diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index b36264d1d1cd3..94c6bdfab63bb 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -544,15 +544,18 @@ static int glk_card_late_probe(struct snd_soc_card *card) struct snd_soc_component *component = NULL; char jack_name[NAME_SIZE]; struct glk_hdmi_pcm *pcm; - int err = 0; + int err; int i = 0; - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct glk_hdmi_pcm, - head); - component = pcm->codec_dai->component; + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; - if (ctx->common_hdmi_codec_drv) + if (ctx->common_hdmi_codec_drv) { + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct glk_hdmi_pcm, + head); + component = pcm->codec_dai->component; return hda_dsp_hdmi_build_controls(card, component); + } list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; @@ -573,9 +576,6 @@ static int glk_card_late_probe(struct snd_soc_card *card) i++; } - if (!component) - return -EINVAL; - return hdac_hdmi_jack_port_init(component, &card->dapm); } diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 8a13231dee15d..5d878873a8e08 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -273,19 +273,22 @@ static int sof_card_late_probe(struct snd_soc_card *card) struct snd_soc_component *component = NULL; char jack_name[NAME_SIZE]; struct sof_hdmi_pcm *pcm; - int err = 0; + int err; int i = 0; /* HDMI is not supported by SOF on Baytrail/CherryTrail */ if (is_legacy_cpu) return 0; - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, - head); - component = pcm->codec_dai->component; + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; - if (ctx->common_hdmi_codec_drv) + if (ctx->common_hdmi_codec_drv) { + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, + head); + component = pcm->codec_dai->component; return hda_dsp_hdmi_build_controls(card, component); + } list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; @@ -305,8 +308,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) i++; } - if (!component) - return -EINVAL; return hdac_hdmi_jack_port_init(component, &card->dapm); } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 470/542] ALSA: usb-audio: add quirks for Line6 Helix devices fw>=2.82
by Sasha Levin 14 Feb '20

14 Feb '20

From: Nicola Lunghi <nick83ola(a)gmail.com> [ Upstream commit b81cbf7abfc94878a3c6f0789f2185ee55b1cc21 ] With firmware 2.82 Line6 changed the usb id of some of the Helix devices but the quirks is still needed. Add it to the quirk list for line6 helix family of devices. Thanks to Jens for pointing out the missing ids. Signed-off-by: Nicola Lunghi <nick83ola(a)gmail.com> Link: https://lore.kernel.org/r/20200125150917.5040-1-nick83ola@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/format.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/format.c b/sound/usb/format.c index d79db71305f63..53922f73467f4 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -296,6 +296,9 @@ static int line6_parse_audio_format_rates_quirk(struct snd_usb_audio *chip, case USB_ID(0x0E41, 0x4242): /* Line6 Helix Rack */ case USB_ID(0x0E41, 0x4244): /* Line6 Helix LT */ case USB_ID(0x0E41, 0x4246): /* Line6 HX-Stomp */ + case USB_ID(0x0E41, 0x4248): /* Line6 Helix >= fw 2.82 */ + case USB_ID(0x0E41, 0x4249): /* Line6 Helix Rack >= fw 2.82 */ + case USB_ID(0x0E41, 0x424a): /* Line6 Helix LT >= fw 2.82 */ /* supported rates: 48Khz */ kfree(fp->rate_table); fp->rate_table = kmalloc(sizeof(int), GFP_KERNEL); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 439/542] ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
by Sasha Levin 14 Feb '20

14 Feb '20

From: Peter Große <pegro(a)friiks.de> [ Upstream commit ef7d84caa5928b40b1c93a26dbe5a3f12737c6ab ] Lenovo Thinkpad T420s uses the same codec as T420, so apply the same quirk to enable audio output on a docking station. Signed-off-by: Peter Große <pegro(a)friiks.de> Link: https://lore.kernel.org/r/20200122180106.9351-1-pegro@friiks.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 90aa0f400a57d..1e20e85e9b466 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -922,6 +922,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21d2, "Lenovo T420s", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 427/542] ASoC: soc-generic-dmaengine-pcm: Fix error handling
by Sasha Levin 14 Feb '20

14 Feb '20

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 130128098a4e5ce9a0dfbdf9a7e27a43579901fd ] Remove the return value checking, that is to align with the code before adding snd_dmaengine_pcm_refine_runtime_hwparams function. Otherwise it causes a regression on the HiKey board: [ 17.721424] hi6210_i2s f7118000.i2s: ASoC: can't open component f7118000.i2s: -6 Fixes: e957204e732b ("ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Reported-by: John Stultz <john.stultz(a)linaro.org> Link: https://lore.kernel.org/r/1579505286-32085-1-git-send-email-shengjiu.wang@n… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-generic-dmaengine-pcm.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index a428ff393ea26..2b5f3b1b062bc 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -117,7 +117,6 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, struct dma_chan *chan = pcm->chan[substream->stream]; struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_hardware hw; - int ret; if (pcm->config && pcm->config->pcm_hardware) return snd_soc_set_runtime_hwparams(substream, @@ -138,12 +137,15 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) hw.info |= SNDRV_PCM_INFO_BATCH; - ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, - dma_data, - &hw, - chan); - if (ret) - return ret; + /** + * FIXME: Remove the return value check to align with the code + * before adding snd_dmaengine_pcm_refine_runtime_hwparams + * function. + */ + snd_dmaengine_pcm_refine_runtime_hwparams(substream, + dma_data, + &hw, + chan); return snd_soc_set_runtime_hwparams(substream, &hw); } -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 426/542] ALSA: hda/hdmi - add retry logic to parse_intel_hdmi()
by Sasha Levin 14 Feb '20

14 Feb '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 2928fa0a97ebb9549cb877fdc99aed9b95438c3a ] The initial snd_hda_get_sub_node() can fail on certain devices (e.g. some Chromebook models using Intel GLK). The failure rate is very low, but as this is is part of the probe process, end-user impact is high. In observed cases, related hardware status registers have expected values, but the node query still fails. Retrying the node query does seem to help, so fix the problem by adding retry logic to the query. This does not impact non-Intel platforms. BugLink: https://github.com/thesofproject/linux/issues/1642 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200120160117.29130-4-kai.vehmanen@linux.intel.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_hdmi.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bde50414029d9..4f195c7d966a9 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2862,9 +2862,12 @@ static int alloc_intel_hdmi(struct hda_codec *codec) /* parse and post-process for Intel codecs */ static int parse_intel_hdmi(struct hda_codec *codec) { - int err; + int err, retries = 3; + + do { + err = hdmi_parse_codec(codec); + } while (err < 0 && retries--); - err = hdmi_parse_codec(codec); if (err < 0) { generic_spec_free(codec); return err; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 405/542] ASoC: wm_adsp: Correct cache handling of new kernel control API
by Sasha Levin 14 Feb '20

14 Feb '20

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 73ecf1a673d3502dff1445f06675aba65ff20ce7 ] The recently added API that exposes firmware mixer controls to the kernel is missing cache handling and all writes bypass the cache, this obviously causes the cache to get out of sync with the hardware. Factor out the cache handling into two new helper functions and call those from both the normal ALSA control handlers and the new kernel API. Fixes: eb65ccdb0836 ("ASoC: wm_adsp: Expose mixer control API") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200114161841.451-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 98 ++++++++++++++++++++------------------ 1 file changed, 51 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 2a9b610f6d435..d3d32b501acae 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1030,8 +1030,8 @@ static int wm_coeff_write_acked_control(struct wm_coeff_ctl *ctl, return -ETIMEDOUT; } -static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, - const void *buf, size_t len) +static int wm_coeff_write_ctrl_raw(struct wm_coeff_ctl *ctl, + const void *buf, size_t len) { struct wm_adsp *dsp = ctl->dsp; void *scratch; @@ -1061,6 +1061,23 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, return 0; } +static int wm_coeff_write_ctrl(struct wm_coeff_ctl *ctl, + const void *buf, size_t len) +{ + int ret = 0; + + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) + ret = -EPERM; + else if (buf != ctl->cache) + memcpy(ctl->cache, buf, len); + + ctl->set = 1; + if (ctl->enabled && ctl->dsp->running) + ret = wm_coeff_write_ctrl_raw(ctl, buf, len); + + return ret; +} + static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { @@ -1071,16 +1088,7 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, int ret = 0; mutex_lock(&ctl->dsp->pwr_lock); - - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - ret = -EPERM; - else - memcpy(ctl->cache, p, ctl->len); - - ctl->set = 1; - if (ctl->enabled && ctl->dsp->running) - ret = wm_coeff_write_control(ctl, p, ctl->len); - + ret = wm_coeff_write_ctrl(ctl, p, ctl->len); mutex_unlock(&ctl->dsp->pwr_lock); return ret; @@ -1096,15 +1104,10 @@ static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - if (copy_from_user(ctl->cache, bytes, size)) { + if (copy_from_user(ctl->cache, bytes, size)) ret = -EFAULT; - } else { - ctl->set = 1; - if (ctl->enabled && ctl->dsp->running) - ret = wm_coeff_write_control(ctl, ctl->cache, size); - else if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - ret = -EPERM; - } + else + ret = wm_coeff_write_ctrl(ctl, ctl->cache, size); mutex_unlock(&ctl->dsp->pwr_lock); @@ -1135,8 +1138,8 @@ static int wm_coeff_put_acked(struct snd_kcontrol *kctl, return ret; } -static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, - void *buf, size_t len) +static int wm_coeff_read_ctrl_raw(struct wm_coeff_ctl *ctl, + void *buf, size_t len) { struct wm_adsp *dsp = ctl->dsp; void *scratch; @@ -1166,29 +1169,37 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, return 0; } -static int wm_coeff_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int wm_coeff_read_ctrl(struct wm_coeff_ctl *ctl, void *buf, size_t len) { - struct soc_bytes_ext *bytes_ext = - (struct soc_bytes_ext *)kctl->private_value; - struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); - char *p = ucontrol->value.bytes.data; int ret = 0; - mutex_lock(&ctl->dsp->pwr_lock); - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { if (ctl->enabled && ctl->dsp->running) - ret = wm_coeff_read_control(ctl, p, ctl->len); + return wm_coeff_read_ctrl_raw(ctl, buf, len); else - ret = -EPERM; + return -EPERM; } else { if (!ctl->flags && ctl->enabled && ctl->dsp->running) - ret = wm_coeff_read_control(ctl, ctl->cache, ctl->len); + ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, ctl->len); - memcpy(p, ctl->cache, ctl->len); + if (buf != ctl->cache) + memcpy(buf, ctl->cache, len); } + return ret; +} + +static int wm_coeff_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + char *p = ucontrol->value.bytes.data; + int ret; + + mutex_lock(&ctl->dsp->pwr_lock); + ret = wm_coeff_read_ctrl(ctl, p, ctl->len); mutex_unlock(&ctl->dsp->pwr_lock); return ret; @@ -1204,15 +1215,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { - if (ctl->enabled && ctl->dsp->running) - ret = wm_coeff_read_control(ctl, ctl->cache, size); - else - ret = -EPERM; - } else { - if (!ctl->flags && ctl->enabled && ctl->dsp->running) - ret = wm_coeff_read_control(ctl, ctl->cache, size); - } + ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size); if (!ret && copy_to_user(bytes, ctl->cache, size)) ret = -EFAULT; @@ -1340,7 +1343,7 @@ static int wm_coeff_init_control_caches(struct wm_adsp *dsp) * created so we don't need to do anything. */ if (!ctl->flags || (ctl->flags & WMFW_CTL_FLAG_READABLE)) { - ret = wm_coeff_read_control(ctl, ctl->cache, ctl->len); + ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, ctl->len); if (ret < 0) return ret; } @@ -1358,7 +1361,8 @@ static int wm_coeff_sync_controls(struct wm_adsp *dsp) if (!ctl->enabled) continue; if (ctl->set && !(ctl->flags & WMFW_CTL_FLAG_VOLATILE)) { - ret = wm_coeff_write_control(ctl, ctl->cache, ctl->len); + ret = wm_coeff_write_ctrl_raw(ctl, ctl->cache, + ctl->len); if (ret < 0) return ret; } @@ -2048,7 +2052,7 @@ int wm_adsp_write_ctl(struct wm_adsp *dsp, const char *name, int type, if (len > ctl->len) return -EINVAL; - ret = wm_coeff_write_control(ctl, buf, len); + ret = wm_coeff_write_ctrl(ctl, buf, len); kcontrol = snd_soc_card_get_kcontrol(dsp->component->card, ctl->name); snd_ctl_notify(dsp->component->card->snd_card, @@ -2070,7 +2074,7 @@ int wm_adsp_read_ctl(struct wm_adsp *dsp, const char *name, int type, if (len > ctl->len) return -EINVAL; - return wm_coeff_read_control(ctl, buf, len); + return wm_coeff_read_ctrl(ctl, buf, len); } EXPORT_SYMBOL_GPL(wm_adsp_read_ctl); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 381/542] ALSA: usb-audio: add implicit fb quirk for MOTU M Series
by Sasha Levin 14 Feb '20

14 Feb '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit c249177944b650816069f6c49b769baaa94339dc ] This fixes crackling sound during playback. Further note: MOTU is known for reusing Product IDs for different devices or different generations of the device (e.g. MicroBook I/II/IIc shares a single Product ID). This patch was only tested with M4 audio interface, but the same Product ID is also used by M2. Hope it will work for M2 as well. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200115151358.56672-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0e4eab96e23e0..c9e1609296dff 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -348,6 +348,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x84; ifnum = 0; goto add_sync_ep_from_ifnum; + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + ep = 0x81; + ifnum = 2; + goto add_sync_ep_from_ifnum; case USB_ID(0x0582, 0x01d8): /* BOSS Katana */ /* BOSS Katana amplifiers do not need quirks */ return 0; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 371/542] ALSA: usb-audio: unlock on error in probe
by Sasha Levin 14 Feb '20

14 Feb '20

From: Dan Carpenter <dan.carpenter(a)oracle.com> [ Upstream commit a3afa29942b84b4e2548beacccc3a68b8d77e3dc ] We need to unlock before we returning on this error path. Fixes: 73ac9f5e5b43 ("ALSA: usb-audio: Add boot quirk for MOTU M Series") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Link: https://lore.kernel.org/r/20200115174604.rhanfgy4j3uc65cx@kili.mountain Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2f582ac7cf789..827fb0bc8b561 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -602,7 +602,7 @@ static int usb_audio_probe(struct usb_interface *intf, if (! chip) { err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); if (err < 0) - return err; + goto __error; /* it's a fresh one. * now look for an empty slot and create a new card instance -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 348/542] ASoC: SOF: Intel: hda-dai: fix compilation warning in pcm_prepare
by Sasha Levin 14 Feb '20

14 Feb '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit d873997192ddcacb5333575502be2f91ea4b47b8 ] Fix GCC warning with W=1, previous cleanup did not remove unnecessary variable. sound/soc/sof/intel/hda-dai.c: In function ‘hda_link_pcm_prepare’: sound/soc/sof/intel/hda-dai.c:265:31: warning: variable ‘hda_stream’ set but not used [-Wunused-but-set-variable] 265 | struct sof_intel_hda_stream *hda_stream; | ^~~~~~~~~~ Fixes: a3ebccb52efdf ("ASoC: SOF: Intel: hda: reset link DMA state in prepare") Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200113205620.27285-1-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda-dai.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 896d21984b735..1923b0c36bcef 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -261,14 +261,11 @@ static int hda_link_pcm_prepare(struct snd_pcm_substream *substream, { struct hdac_ext_stream *link_dev = snd_soc_dai_get_dma_data(dai, substream); - struct sof_intel_hda_stream *hda_stream; struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(dai->component); struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); int stream = substream->stream; - hda_stream = hstream_to_sof_hda_stream(link_dev); - if (link_dev->link_prepared) return 0; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 336/542] ASoC: atmel: fix build error with CONFIG_SND_ATMEL_SOC_DMA=m
by Sasha Levin 14 Feb '20

14 Feb '20

From: Chen Zhou <chenzhou10(a)huawei.com> [ Upstream commit 8fea78029f5e6ed734ae1957bef23cfda1af4354 ] If CONFIG_SND_ATMEL_SOC_DMA=m, build error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register' Function atmel_pcm_dma_platform_register is defined under CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Chen Zhou <chenzhou10(a)huawei.com> Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c229ed829..d1dc8e6366dcb 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -19,6 +19,8 @@ config SND_ATMEL_SOC_DMA config SND_ATMEL_SOC_SSC tristate + select SND_ATMEL_SOC_DMA + select SND_ATMEL_SOC_PDC config SND_ATMEL_SOC_SSC_PDC tristate "SoC PCM DAI support for AT91 SSC controller using PDC" -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 335/542] ASoC: txx9: Remove unused rtd variable
by Sasha Levin 14 Feb '20

14 Feb '20

From: Paul Burton <paulburton(a)kernel.org> [ Upstream commit ec0f6a4c4a987aa20b2e77e0db2ae555276e45e6 ] Commit a857e073ffc6 ("ASoC: txx9: txx9aclc: remove snd_pcm_ops") removed the last use of the rtd variable but didn't remove its definition, leading to the following warning/error for MIPS rbtx49xx_defconfig builds: sound/soc/txx9/txx9aclc.c: In function 'txx9aclc_pcm_hw_params': sound/soc/txx9/txx9aclc.c:54:30: error: unused variable 'rtd' [-Werror=unused-variable] struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); ^~~ Resolve this by removing the unused variable. Signed-off-by: Paul Burton <paulburton(a)kernel.org> Fixes: a857e073ffc6 ("ASoC: txx9: txx9aclc: remove snd_pcm_ops") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20200109191422.334516-1-paulburton@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/txx9/txx9aclc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 33c78d33e5a1d..9a55926ebf07b 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -51,7 +51,6 @@ static int txx9aclc_pcm_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct txx9aclc_dmadata *dmadata = runtime->private_data; int ret; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 333/542] ALSA: usb-audio: Add boot quirk for MOTU M Series
by Sasha Levin 14 Feb '20

14 Feb '20

From: Alexander Tsoy <alexander(a)tsoy.me> [ Upstream commit 73ac9f5e5b43a5dbadb61f27dae7a971f7ec0d22 ] Add delay to make sure that audio urbs are not sent too early. Otherwise the device hangs. Windows driver makes ~2s delay, so use about the same time delay value. snd_usb_apply_boot_quirk() is called 3 times for my MOTU M4, which is an overkill. Thus a quirk that is called only once is implemented. Also send two vendor-specific control messages before and after the delay. This behaviour is blindly copied from the Windows driver. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> Link: https://lore.kernel.org/r/20200112102358.18085-1-alexander@tsoy.me Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.c | 4 ++++ sound/usb/quirks.c | 38 ++++++++++++++++++++++++++++++++++++++ sound/usb/quirks.h | 5 +++++ 3 files changed, 47 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9f743ebae615d..2f582ac7cf789 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -600,6 +600,10 @@ static int usb_audio_probe(struct usb_interface *intf, } } if (! chip) { + err = snd_usb_apply_boot_quirk_once(dev, intf, quirk, id); + if (err < 0) + return err; + /* it's a fresh one. * now look for an empty slot and create a new card instance */ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 82184036437b5..0464df785f2ee 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1113,6 +1113,31 @@ static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) return err; } +static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) +{ + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x0, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + msleep(2000); + + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + 1, USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0x20, 0, NULL, 0, 1000); + + if (ret < 0) + return ret; + + return 0; +} + /* * Setup quirks */ @@ -1297,6 +1322,19 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return 0; } +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int id) +{ + switch (id) { + case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ + return snd_usb_motu_m_series_boot_quirk(dev); + } + + return 0; +} + /* * check if the device uses big-endian samples */ diff --git a/sound/usb/quirks.h b/sound/usb/quirks.h index a80e0ddd07364..df0355843a4c1 100644 --- a/sound/usb/quirks.h +++ b/sound/usb/quirks.h @@ -20,6 +20,11 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, const struct snd_usb_audio_quirk *quirk, unsigned int usb_id); +int snd_usb_apply_boot_quirk_once(struct usb_device *dev, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk, + unsigned int usb_id); + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, struct audioformat *fmt); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 288/542] ASoC: Intel: sof_rt5682: Ignore the speaker amp when there isn't one.
by Sasha Levin 14 Feb '20

14 Feb '20

From: Sam McNally <sammc(a)chromium.org> [ Upstream commit d4b74e218a8d0d6cf58e546627ab9d4d4f2645ab ] Some members of the Google_Hatch family include a rt5682 jack codec, but no speaker amplifier. This uses the same driver (sof_rt5682) as a combination of rt5682 jack codec and max98357a speaker amplifier. Within the sof_rt5682 driver, these cases are not currently distinguishable, relying on a DMI quirk to decide the configuration. This causes an incorrect configuration when only the rt5682 is present on a Google_Hatch device. For CML, the jack codec is used as the primary key when matching, with a possible speaker amplifier described in quirk_data. The two cases of interest are the second and third 10EC5682 entries in snd_soc_acpi_intel_cml_machines[]. The second entry matches the combination of rt5682 and max98357a, resulting in the quirk_data field in the snd_soc_acpi_mach being non-null, pointing at max98357a_spk_codecs, the snd_soc_acpi_codecs for the matched speaker amplifier. The third entry matches just the rt5682, resulting in a null quirk_data. The sof_rt5682 driver's DMI data matching identifies that a speaker amplifier is present for all Google_Hatch family devices. Detect cases where there is no speaker amplifier by checking for a null quirk_data in the snd_soc_acpi_mach and remove the speaker amplifier bit in that case. Signed-off-by: Sam McNally <sammc(a)chromium.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200103124921.v3.1.Ib87c4a7fbb3fc818ea12198e291b… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ad8a2b4bc7092..8a13231dee15d 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -603,6 +603,14 @@ static int sof_audio_probe(struct platform_device *pdev) dmi_check_system(sof_rt5682_quirk_table); + mach = (&pdev->dev)->platform_data; + + /* A speaker amp might not be present when the quirk claims one is. + * Detect this via whether the machine driver match includes quirk_data. + */ + if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) + sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -663,7 +671,6 @@ static int sof_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); sof_audio_card_rt5682.dev = &pdev->dev; - mach = (&pdev->dev)->platform_data; /* set platform name for each dailink */ ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682, -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 284/542] ALSA: sh: Fix compile warning wrt const
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit f1dd4795b1523fbca7ab4344dd5a8bb439cc770d ] A long-standing compile warning was seen during build test: sound/sh/aica.c: In function 'load_aica_firmware': sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] Fixes: 198de43d758c ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device") Link: https://lore.kernel.org/r/20200105144823.29547-69-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/aica.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index bf1fb0d8a9306..f69072d2564c8 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -101,10 +101,10 @@ static void spu_memset(u32 toi, u32 what, int length) } /* spu_memload - write to SPU address space */ -static void spu_memload(u32 toi, void *from, int length) +static void spu_memload(u32 toi, const void *from, int length) { unsigned long flags; - u32 *froml = from; + const u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; u32 val; -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 283/542] ALSA: hda/realtek - Apply mic mute LED quirk for Dell E7xx laptops, too
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5fab5829674c279839a7408ab30c71c6dfe726b9 ] Dell E7xx laptops have also mic mute LED that is driven by the dell-laptop platform driver. Bind it with the capture control as already done for other models. A caveat is that the fixup hook for the mic mute LED has to be applied at last, otherwise it results in the invalid override of the callback. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205529 Link: https://lore.kernel.org/r/20200105081119.21396-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f2ea3528bfb1d..f952df64d5302 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5848,6 +5848,7 @@ enum { ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, ALC288_FIXUP_DISABLE_AAMIX, + ALC292_FIXUP_DELL_E7X_AAMIX, ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, @@ -6543,12 +6544,19 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC293_FIXUP_DELL1_MIC_NO_PRESENCE }, - [ALC292_FIXUP_DELL_E7X] = { + [ALC292_FIXUP_DELL_E7X_AAMIX] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_dell_xps13, .chained = true, .chain_id = ALC292_FIXUP_DISABLE_AAMIX }, + [ALC292_FIXUP_DELL_E7X] = { + .type = HDA_FIXUP_FUNC, + .v.func = snd_hda_gen_fixup_micmute_led, + /* micmute fixup must be applied at last */ + .chained_before = true, + .chain_id = ALC292_FIXUP_DELL_E7X_AAMIX, + }, [ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 280/542] ALSA: sh: Fix unused variable warnings
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 5da116f164ce265e397b8f59af5c39e4a61d61a5 ] Remove unused variables that are left over after the conversion of new PCM ops: sound/sh/sh_dac_audio.c:166:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:186:26: warning: unused variable 'runtime' sound/sh/sh_dac_audio.c:205:26: warning: unused variable 'runtime' Fixes: 1cc2f8ba0b3e ("ALSA: sh: Convert to the new PCM ops") Link: https://lore.kernel.org/r/20200104110057.13875-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/sh/sh_dac_audio.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index f9e36abc98ac7..725992937e8f1 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -175,7 +175,6 @@ static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; if (copy_from_user_toio(chip->data_buffer + pos, src, count)) return -EFAULT; @@ -195,7 +194,6 @@ static int snd_sh_dac_pcm_copy_kernel(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memcpy_toio(chip->data_buffer + pos, src, count); chip->buffer_end = chip->data_buffer + pos + count; @@ -214,7 +212,6 @@ static int snd_sh_dac_pcm_silence(struct snd_pcm_substream *substream, { /* channel is not used (interleaved data) */ struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; memset_io(chip->data_buffer + pos, 0, count); chip->buffer_end = chip->data_buffer + pos + count; -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 275/542] ASoC: soc-topology: fix endianness issues
by Sasha Levin 14 Feb '20

14 Feb '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 72bbeda0222bcd382ee33b3aff71346074410c21 ] Sparse complains about a series of easy warnings, fix. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200102195952.9465-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-topology.c | 42 +++++++++++++++++++++------------------- 1 file changed, 22 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4e1fe623c3908..0119f07cece6f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -604,9 +604,11 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, ext_ops = tplg->bytes_ext_ops; num_ops = tplg->bytes_ext_ops_count; for (i = 0; i < num_ops; i++) { - if (!sbe->put && ext_ops[i].id == be->ext_ops.put) + if (!sbe->put && + ext_ops[i].id == le32_to_cpu(be->ext_ops.put)) sbe->put = ext_ops[i].put; - if (!sbe->get && ext_ops[i].id == be->ext_ops.get) + if (!sbe->get && + ext_ops[i].id == le32_to_cpu(be->ext_ops.get)) sbe->get = ext_ops[i].get; } @@ -621,11 +623,11 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, num_ops = tplg->io_ops_count; for (i = 0; i < num_ops; i++) { - if (k->put == NULL && ops[i].id == hdr->ops.put) + if (k->put == NULL && ops[i].id == le32_to_cpu(hdr->ops.put)) k->put = ops[i].put; - if (k->get == NULL && ops[i].id == hdr->ops.get) + if (k->get == NULL && ops[i].id == le32_to_cpu(hdr->ops.get)) k->get = ops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && ops[i].id == le32_to_cpu(hdr->ops.info)) k->info = ops[i].info; } @@ -638,11 +640,11 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, num_ops = ARRAY_SIZE(io_ops); for (i = 0; i < num_ops; i++) { - if (k->put == NULL && ops[i].id == hdr->ops.put) + if (k->put == NULL && ops[i].id == le32_to_cpu(hdr->ops.put)) k->put = ops[i].put; - if (k->get == NULL && ops[i].id == hdr->ops.get) + if (k->get == NULL && ops[i].id == le32_to_cpu(hdr->ops.get)) k->get = ops[i].get; - if (k->info == NULL && ops[i].id == hdr->ops.info) + if (k->info == NULL && ops[i].id == le32_to_cpu(hdr->ops.info)) k->info = ops[i].info; } @@ -931,7 +933,7 @@ static int soc_tplg_denum_create_texts(struct soc_enum *se, if (se->dobj.control.dtexts == NULL) return -ENOMEM; - for (i = 0; i < ec->items; i++) { + for (i = 0; i < le32_to_cpu(ec->items); i++) { if (strnlen(ec->texts[i], SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == SNDRV_CTL_ELEM_ID_NAME_MAXLEN) { @@ -1325,7 +1327,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( if (kc[i].name == NULL) goto err_sm; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = mc->hdr.access; + kc[i].access = le32_to_cpu(mc->hdr.access); /* we only support FL/FR channel mapping atm */ sm->reg = tplc_chan_get_reg(tplg, mc->channel, @@ -1337,10 +1339,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( sm->rshift = tplc_chan_get_shift(tplg, mc->channel, SNDRV_CHMAP_FR); - sm->max = mc->max; - sm->min = mc->min; - sm->invert = mc->invert; - sm->platform_max = mc->platform_max; + sm->max = le32_to_cpu(mc->max); + sm->min = le32_to_cpu(mc->min); + sm->invert = le32_to_cpu(mc->invert); + sm->platform_max = le32_to_cpu(mc->platform_max); sm->dobj.index = tplg->index; INIT_LIST_HEAD(&sm->dobj.list); @@ -1401,7 +1403,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( goto err_se; tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - ec->priv.size); + le32_to_cpu(ec->priv.size)); dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", ec->hdr.name); @@ -1411,7 +1413,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( if (kc[i].name == NULL) goto err_se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = ec->hdr.access; + kc[i].access = le32_to_cpu(ec->hdr.access); /* we only support FL/FR channel mapping atm */ se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); @@ -1420,8 +1422,8 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( se->shift_r = tplc_chan_get_shift(tplg, ec->channel, SNDRV_CHMAP_FR); - se->items = ec->items; - se->mask = ec->mask; + se->items = le32_to_cpu(ec->items); + se->mask = le32_to_cpu(ec->mask); se->dobj.index = tplg->index; switch (le32_to_cpu(ec->hdr.ops.info)) { @@ -1523,9 +1525,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( if (kc[i].name == NULL) goto err_sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = be->hdr.access; + kc[i].access = le32_to_cpu(be->hdr.access); - sbe->max = be->max; + sbe->max = le32_to_cpu(be->max); INIT_LIST_HEAD(&sbe->dobj.list); /* map standard io handlers and check for external handlers */ -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 261/542] ASoC: Intel: kbl_da7219_max98357a: remove unused variable 'constraints_16000' and 'ch_mono'
by Sasha Levin 14 Feb '20

14 Feb '20

From: YueHaibing <yuehaibing(a)huawei.com> [ Upstream commit c5614fb8e3d13be7bba79f71b798468a3a6224f7 ] sound/soc/intel/boards/kbl_da7219_max98357a.c:343:48: warning: constraints_16000 defined but not used [-Wunused-const-variable=] sound/soc/intel/boards/kbl_da7219_max98357a.c:348:27: warning: ch_mono defined but not used [-Wunused-const-variable=] They are never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20191224140237.36732-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98357a.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 537a88932bb69..0d55319a0773c 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -336,19 +336,6 @@ static struct snd_soc_ops kabylake_dmic_ops = { .startup = kabylake_dmic_startup, }; -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); -- 2.20.1

1 0

[alsa-devel] [PATCH AUTOSEL 5.5 221/542] ALSA: usx2y: Adjust indentation in snd_usX2Y_hwdep_dsp_status
by Sasha Levin 14 Feb '20

14 Feb '20

From: Nathan Chancellor <natechancellor(a)gmail.com> [ Upstream commit df4654bd6e42125d9b85ce3a26eaca2935290b98 ] Clang warns: ../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation; statement is not part of the previous 'if' [-Wmisleading-indentation] info->version = USX2Y_DRIVER_VERSION; ^ ../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here if (us428->chip_status & USX2Y_STAT_CHIP_INIT) ^ 1 warning generated. This warning occurs because there is a space before the tab on this line. Remove it so that the indentation is consistent with the Linux kernel coding style and clang no longer warns. This was introduced before the beginning of git history so no fixes tag. Link: https://github.com/ClangBuiltLinux/linux/issues/831 Signed-off-by: Nathan Chancellor <natechancellor(a)gmail.com> Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usX2Yhwdep.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index d1caa8ed9e681..9985fc139487b 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -119,7 +119,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, info->num_dsps = 2; // 0: Prepad Data, 1: FPGA Code if (us428->chip_status & USX2Y_STAT_CHIP_INIT) info->chip_ready = 1; - info->version = USX2Y_DRIVER_VERSION; + info->version = USX2Y_DRIVER_VERSION; return 0; } -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 175/542] ASoC: SOF: Intel: hda: solve MSI issues by merging ipc and stream irq handlers
by Sasha Levin 14 Feb '20

14 Feb '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 7c11af9fcdc425b80f140a218d4fef9f17734bfc ] The existing code uses two handlers for a shared edge-based MSI interrupts. In corner cases, interrupts are lost, leading to IPC timeouts. Those timeouts do not appear in legacy mode. This patch merges the two handlers and threads into a single one, and simplifies the mask/unmask operations by using a single top-level mask (Global Interrupt Enable). The handler only checks for interrupt sources using the Global Interrupt Status (GIS) field, and all the actual work happens in the thread. This also enables us to remove the use of spin locks. Stream events are prioritized over IPC ones. This patch was tested with HDaudio and SoundWire platforms, and all known IPC timeout issues are solved in MSI mode. The SoundWire-specific patches will be provided in follow-up patches, where the SoundWire interrupts are handled in the same thread as IPC and stream interrupts. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191204212859.13239-1-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/apl.c | 1 - sound/soc/sof/intel/cnl.c | 5 --- sound/soc/sof/intel/hda-ipc.c | 23 +++-------- sound/soc/sof/intel/hda-stream.c | 20 ++++----- sound/soc/sof/intel/hda.c | 69 ++++++++++++++++++++++---------- sound/soc/sof/intel/hda.h | 11 ++--- 6 files changed, 70 insertions(+), 59 deletions(-) diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index 7daa8eb456c8d..6f45e14f2b2e3 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -41,7 +41,6 @@ const struct snd_sof_dsp_ops sof_apl_ops = { .block_write = sof_block_write, /* doorbell */ - .irq_handler = hda_dsp_ipc_irq_handler, .irq_thread = hda_dsp_ipc_irq_thread, /* ipc */ diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index 0e1e265f3f3b3..9bd169e2691e2 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -106,10 +106,6 @@ static irqreturn_t cnl_ipc_irq_thread(int irq, void *context) "nothing to do in IPC IRQ thread\n"); } - /* re-enable IPC interrupt */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, - HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); - return IRQ_HANDLED; } @@ -231,7 +227,6 @@ const struct snd_sof_dsp_ops sof_cnl_ops = { .block_write = sof_block_write, /* doorbell */ - .irq_handler = hda_dsp_ipc_irq_handler, .irq_thread = cnl_ipc_irq_thread, /* ipc */ diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 0fd2153c17695..1837f66e361fd 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -230,22 +230,15 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) "nothing to do in IPC IRQ thread\n"); } - /* re-enable IPC interrupt */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIC, - HDA_DSP_ADSPIC_IPC, HDA_DSP_ADSPIC_IPC); - return IRQ_HANDLED; } -/* is this IRQ for ADSP ? - we only care about IPC here */ -irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context) +/* Check if an IPC IRQ occurred */ +bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev) { - struct snd_sof_dev *sdev = context; - int ret = IRQ_NONE; + bool ret = false; u32 irq_status; - spin_lock(&sdev->hw_lock); - /* store status */ irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS); dev_vdbg(sdev->dev, "irq handler: irq_status:0x%x\n", irq_status); @@ -255,16 +248,10 @@ irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context) goto out; /* IPC message ? */ - if (irq_status & HDA_DSP_ADSPIS_IPC) { - /* disable IPC interrupt */ - snd_sof_dsp_update_bits_unlocked(sdev, HDA_DSP_BAR, - HDA_DSP_REG_ADSPIC, - HDA_DSP_ADSPIC_IPC, 0); - ret = IRQ_WAKE_THREAD; - } + if (irq_status & HDA_DSP_ADSPIS_IPC) + ret = true; out: - spin_unlock(&sdev->hw_lock); return ret; } diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 29ab432816701..927a36f92c242 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -549,22 +549,23 @@ int hda_dsp_stream_hw_free(struct snd_sof_dev *sdev, return 0; } -irqreturn_t hda_dsp_stream_interrupt(int irq, void *context) +bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev) { - struct hdac_bus *bus = context; - int ret = IRQ_WAKE_THREAD; + struct hdac_bus *bus = sof_to_bus(sdev); + bool ret = false; u32 status; - spin_lock(&bus->reg_lock); + /* The function can be called at irq thread, so use spin_lock_irq */ + spin_lock_irq(&bus->reg_lock); status = snd_hdac_chip_readl(bus, INTSTS); dev_vdbg(bus->dev, "stream irq, INTSTS status: 0x%x\n", status); - /* Register inaccessible, ignore it.*/ - if (status == 0xffffffff) - ret = IRQ_NONE; + /* if Register inaccessible, ignore it.*/ + if (status != 0xffffffff) + ret = true; - spin_unlock(&bus->reg_lock); + spin_unlock_irq(&bus->reg_lock); return ret; } @@ -602,7 +603,8 @@ static bool hda_dsp_stream_check(struct hdac_bus *bus, u32 status) irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context) { - struct hdac_bus *bus = context; + struct snd_sof_dev *sdev = context; + struct hdac_bus *bus = sof_to_bus(sdev); #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) u32 rirb_status; #endif diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index fb17b87b684bf..82ecadda886c6 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -499,6 +499,49 @@ static const struct sof_intel_dsp_desc return chip_info; } +static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + + /* + * Get global interrupt status. It includes all hardware interrupt + * sources in the Intel HD Audio controller. + */ + if (snd_sof_dsp_read(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTSTS) & + SOF_HDA_INTSTS_GIS) { + + /* disable GIE interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + SOF_HDA_INTCTL, + SOF_HDA_INT_GLOBAL_EN, + 0); + + return IRQ_WAKE_THREAD; + } + + return IRQ_NONE; +} + +static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) +{ + struct snd_sof_dev *sdev = context; + + /* deal with streams and controller first */ + if (hda_dsp_check_stream_irq(sdev)) + hda_dsp_stream_threaded_handler(irq, sdev); + + if (hda_dsp_check_ipc_irq(sdev)) + sof_ops(sdev)->irq_thread(irq, sdev); + + /* enable GIE interrupt */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + SOF_HDA_INTCTL, + SOF_HDA_INT_GLOBAL_EN, + SOF_HDA_INT_GLOBAL_EN); + + return IRQ_HANDLED; +} + int hda_dsp_probe(struct snd_sof_dev *sdev) { struct pci_dev *pci = to_pci_dev(sdev->dev); @@ -603,9 +646,7 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) */ if (hda_use_msi && pci_alloc_irq_vectors(pci, 1, 1, PCI_IRQ_MSI) > 0) { dev_info(sdev->dev, "use msi interrupt mode\n"); - hdev->irq = pci_irq_vector(pci, 0); - /* ipc irq number is the same of hda irq */ - sdev->ipc_irq = hdev->irq; + sdev->ipc_irq = pci_irq_vector(pci, 0); /* initialised to "false" by kzalloc() */ sdev->msi_enabled = true; } @@ -616,28 +657,17 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) * in IO-APIC mode, hda->irq and ipc_irq are using the same * irq number of pci->irq */ - hdev->irq = pci->irq; sdev->ipc_irq = pci->irq; } - dev_dbg(sdev->dev, "using HDA IRQ %d\n", hdev->irq); - ret = request_threaded_irq(hdev->irq, hda_dsp_stream_interrupt, - hda_dsp_stream_threaded_handler, - IRQF_SHARED, "AudioHDA", bus); - if (ret < 0) { - dev_err(sdev->dev, "error: failed to register HDA IRQ %d\n", - hdev->irq); - goto free_irq_vector; - } - dev_dbg(sdev->dev, "using IPC IRQ %d\n", sdev->ipc_irq); - ret = request_threaded_irq(sdev->ipc_irq, hda_dsp_ipc_irq_handler, - sof_ops(sdev)->irq_thread, IRQF_SHARED, - "AudioDSP", sdev); + ret = request_threaded_irq(sdev->ipc_irq, hda_dsp_interrupt_handler, + hda_dsp_interrupt_thread, + IRQF_SHARED, "AudioDSP", sdev); if (ret < 0) { dev_err(sdev->dev, "error: failed to register IPC IRQ %d\n", sdev->ipc_irq); - goto free_hda_irq; + goto free_irq_vector; } pci_set_master(pci); @@ -668,8 +698,6 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) free_ipc_irq: free_irq(sdev->ipc_irq, sdev); -free_hda_irq: - free_irq(hdev->irq, bus); free_irq_vector: if (sdev->msi_enabled) pci_free_irq_vectors(pci); @@ -715,7 +743,6 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) SOF_HDA_PPCTL_GPROCEN, 0); free_irq(sdev->ipc_irq, sdev); - free_irq(hda->irq, bus); if (sdev->msi_enabled) pci_free_irq_vectors(pci); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 18d7e72bf9b72..63df888dddb6c 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -43,11 +43,14 @@ /* SOF_HDA_GCTL register bist */ #define SOF_HDA_GCTL_RESET BIT(0) -/* SOF_HDA_INCTL and SOF_HDA_INTSTS regs */ +/* SOF_HDA_INCTL regs */ #define SOF_HDA_INT_GLOBAL_EN BIT(31) #define SOF_HDA_INT_CTRL_EN BIT(30) #define SOF_HDA_INT_ALL_STREAM 0xff +/* SOF_HDA_INTSTS regs */ +#define SOF_HDA_INTSTS_GIS BIT(31) + #define SOF_HDA_MAX_CAPS 10 #define SOF_HDA_CAP_ID_OFF 16 #define SOF_HDA_CAP_ID_MASK GENMASK(SOF_HDA_CAP_ID_OFF + 11,\ @@ -406,8 +409,6 @@ struct sof_intel_hda_dev { /* the maximum number of streams (playback + capture) supported */ u32 stream_max; - int irq; - /* PM related */ bool l1_support_changed;/* during suspend, is L1SEN changed or not */ @@ -511,11 +512,12 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, struct snd_pcm_hw_params *params); int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, struct hdac_ext_stream *stream, int cmd); -irqreturn_t hda_dsp_stream_interrupt(int irq, void *context); irqreturn_t hda_dsp_stream_threaded_handler(int irq, void *context); int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct hdac_stream *stream); +bool hda_dsp_check_ipc_irq(struct snd_sof_dev *sdev); +bool hda_dsp_check_stream_irq(struct snd_sof_dev *sdev); struct hdac_ext_stream * hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction); @@ -540,7 +542,6 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev); int hda_dsp_ipc_get_mailbox_offset(struct snd_sof_dev *sdev); int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id); -irqreturn_t hda_dsp_ipc_irq_handler(int irq, void *context); irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context); int hda_dsp_ipc_cmd_done(struct snd_sof_dev *sdev, int dir); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 174/542] ASoC: intel: sof_rt5682: Add support for tgl-max98357a-rt5682
by Sasha Levin 14 Feb '20

14 Feb '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> [ Upstream commit 6605f0ca3af3b964635287ec7c9dadc812b78eb0 ] This patch adds the driver data and updates quirk info for tgl with max98357a speaker amp and ALC5682 headset codec. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191126143205.21987-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 57adadacbf436..ad8a2b4bc7092 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -598,6 +598,9 @@ static int sof_audio_probe(struct platform_device *pdev) if (!ctx) return -ENOMEM; + if (pdev->id_entry && pdev->id_entry->driver_data) + sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data; + dmi_check_system(sof_rt5682_quirk_table); if (soc_intel_is_byt() || soc_intel_is_cht()) { @@ -691,6 +694,21 @@ static int sof_rt5682_remove(struct platform_device *pdev) return 0; } +static const struct platform_device_id board_ids[] = { + { + .name = "sof_rt5682", + }, + { + .name = "tgl_max98357a_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { } +}; + static struct platform_driver sof_audio = { .probe = sof_audio_probe, .remove = sof_rt5682_remove, @@ -698,6 +716,7 @@ static struct platform_driver sof_audio = { .name = "sof_rt5682", .pm = &snd_soc_pm_ops, }, + .id_table = board_ids, }; module_platform_driver(sof_audio) @@ -707,3 +726,4 @@ MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:sof_rt5682"); +MODULE_ALIAS("platform:tgl_max98357a_rt5682"); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 173/542] ASoC: intel: sof_rt5682: Add quirk for number of HDMI DAI's
by Sasha Levin 14 Feb '20

14 Feb '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> [ Upstream commit c68e07970eca79106b0c35b88a12298569590081 ] TGL supports one more HDMI DAI than previous models. So add quirk support for number of HDMI DAI's. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20191126143205.21987-2-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 751b8ea6ae1f5..57adadacbf436 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -35,6 +35,10 @@ #define SOF_RT5682_SSP_AMP(quirk) \ (((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK) #define SOF_RT5682_MCLK_BYTCHT_EN BIT(9) +#define SOF_RT5682_NUM_HDMIDEV_SHIFT 10 +#define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10)) +#define SOF_RT5682_NUM_HDMIDEV(quirk) \ + ((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK) /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -594,6 +598,8 @@ static int sof_audio_probe(struct platform_device *pdev) if (!ctx) return -ENOMEM; + dmi_check_system(sof_rt5682_quirk_table); + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -604,11 +610,13 @@ static int sof_audio_probe(struct platform_device *pdev) SOF_RT5682_SSP_CODEC(2); } else { dmic_be_num = 2; - hdmi_num = 3; + hdmi_num = (sof_rt5682_quirk & SOF_RT5682_NUM_HDMIDEV_MASK) >> + SOF_RT5682_NUM_HDMIDEV_SHIFT; + /* default number of HDMI DAI's */ + if (!hdmi_num) + hdmi_num = 3; } - dmi_check_system(sof_rt5682_quirk_table); - /* need to get main clock from pmc */ if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) { ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); -- 2.20.1

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[alsa-devel] [PATCH AUTOSEL 5.5 033/542] ALSA: ctl: allow TLV read operation for callback type of element in locked case
by Sasha Levin 14 Feb '20

14 Feb '20

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit d61fe22c2ae42d9fd76c34ef4224064cca4b04b0 ] A design of ALSA control core allows applications to execute three operations for TLV feature; read, write and command. Furthermore, it allows driver developers to process the operations by two ways; allocated array or callback function. In the former, read operation is just allowed, thus developers uses the latter when device driver supports variety of models or the target model is expected to dynamically change information stored in TLV container. The core also allows applications to lock any element so that the other applications can't perform write operation to the element for element value and TLV information. When the element is locked, write and command operation for TLV information are prohibited as well as element value. Any read operation should be allowed in the case. At present, when an element has callback function for TLV information, TLV read operation returns EPERM if the element is locked. On the other hand, the read operation is success when an element has allocated array for TLV information. In both cases, read operation is success for element value expectedly. This commit fixes the bug. This change can be backported to v4.14 kernel or later. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20191223093347.15279-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/control.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 7a4d8690ce41f..08ca7666e84cf 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1430,8 +1430,9 @@ static int call_tlv_handler(struct snd_ctl_file *file, int op_flag, if (kctl->tlv.c == NULL) return -ENXIO; - /* When locked, this is unavailable. */ - if (vd->owner != NULL && vd->owner != file) + /* Write and command operations are not allowed for locked element. */ + if (op_flag != SNDRV_CTL_TLV_OP_READ && + vd->owner != NULL && vd->owner != file) return -EPERM; return kctl->tlv.c(kctl, op_flag, size, buf); -- 2.20.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Don't create a mixer element with bogus volume range
by Takashi Iwai 14 Feb '20

14 Feb '20

Some USB-audio descriptors provide a bogus volume range (e.g. volume min and max are identical), which confuses user-space. This patch makes the driver skipping such a control element. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206221 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 81b2db0edd5f..906fd38ce647 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1674,6 +1674,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, /* get min/max values */ get_min_max_with_quirks(cval, 0, kctl); + /* skip a bogus volume range */ + if (cval->max <= cval->min) { + usb_audio_dbg(mixer->chip, + "[%d] FU [%s] skipped due to invalid volume\n", + cval->head.id, kctl->id.name); + snd_ctl_free_one(kctl); + return; + } + + if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); if (cval->dBmin < cval->dBmax || !cval->initialized) { -- 2.16.4

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[alsa-devel] [PATCH 0/9] ASoC: meson: gx: add audio output support
by Jerome Brunet 14 Feb '20

14 Feb '20

This patchset adds support for the i2s and spdif audio outputs of the amlogic GX SoC family, such the S905, S905X/D, S912 and S805X. These SoCs are used by a fair amount of boards actively maintained upstream. This was tested on: * amlogic s912 q200 * libretech s805x-ac (frite) * libretech s905x-cc (potato) * libretech s905d-pc (tartiflette) This could also possibly support meson8 32bits SoCs but I have not tested it myself and it could require some further tweaks. The audio subsystem found on these SoCs has now been dropped in the newer designs. All recent SoCs families (like g12a and sm1) derive from the AXG audio architecture. Jerome Brunet (9): ASoC: core: allow a dt node to provide several components ASoC: meson: g12a: extract codec-to-codec utils ASoC: meson: aiu: add audio output dt-bindings ASoC: meson: aiu: add i2s and spdif support ASoC: meson: aiu: add hdmi codec control support ASoC: meson: aiu: add internal dac codec control support ASoC: meson: axg: extract sound card utils ASoC: meson: gx: add sound card dt-binding documentation ASoC: meson: gx: add sound card support .../bindings/sound/amlogic,aiu.yaml | 111 +++++ .../bindings/sound/amlogic,gx-sound-card.yaml | 113 +++++ include/dt-bindings/sound/meson-aiu.h | 18 + sound/soc/meson/Kconfig | 24 ++ sound/soc/meson/Makefile | 15 + sound/soc/meson/aiu-acodec-ctrl.c | 205 +++++++++ sound/soc/meson/aiu-codec-ctrl.c | 152 +++++++ sound/soc/meson/aiu-encoder-i2s.c | 324 ++++++++++++++ sound/soc/meson/aiu-encoder-spdif.c | 209 +++++++++ sound/soc/meson/aiu-fifo-i2s.c | 153 +++++++ sound/soc/meson/aiu-fifo-spdif.c | 186 ++++++++ sound/soc/meson/aiu-fifo.c | 223 ++++++++++ sound/soc/meson/aiu-fifo.h | 50 +++ sound/soc/meson/aiu.c | 390 +++++++++++++++++ sound/soc/meson/aiu.h | 91 ++++ sound/soc/meson/axg-card.c | 403 ++---------------- sound/soc/meson/g12a-tohdmitx.c | 219 ++-------- sound/soc/meson/gx-card.c | 141 ++++++ sound/soc/meson/meson-card-utils.c | 385 +++++++++++++++++ sound/soc/meson/meson-card.h | 55 +++ sound/soc/meson/meson-codec-glue.c | 149 +++++++ sound/soc/meson/meson-codec-glue.h | 32 ++ sound/soc/soc-core.c | 8 + 23 files changed, 3104 insertions(+), 552 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,aiu.yaml create mode 100644 Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml create mode 100644 include/dt-bindings/sound/meson-aiu.h create mode 100644 sound/soc/meson/aiu-acodec-ctrl.c create mode 100644 sound/soc/meson/aiu-codec-ctrl.c create mode 100644 sound/soc/meson/aiu-encoder-i2s.c create mode 100644 sound/soc/meson/aiu-encoder-spdif.c create mode 100644 sound/soc/meson/aiu-fifo-i2s.c create mode 100644 sound/soc/meson/aiu-fifo-spdif.c create mode 100644 sound/soc/meson/aiu-fifo.c create mode 100644 sound/soc/meson/aiu-fifo.h create mode 100644 sound/soc/meson/aiu.c create mode 100644 sound/soc/meson/aiu.h create mode 100644 sound/soc/meson/gx-card.c create mode 100644 sound/soc/meson/meson-card-utils.c create mode 100644 sound/soc/meson/meson-card.h create mode 100644 sound/soc/meson/meson-codec-glue.c create mode 100644 sound/soc/meson/meson-codec-glue.h -- 2.24.1

2 24

[alsa-devel] [PATCH 1/2] ASoC: rt5682: Add CCF usage for providing I2S clks
by derek.fang＠realtek.com 14 Feb '20

14 Feb '20

From: "derek.fang" <derek.fang(a)realtek.com> There is a need to use RT5682 as DAI clock master for other codecs within a system, which means that the DAI clocks are required to remain, regardless of whether the RT5682 is actually running playback/capture. Signed-off-by: derek.fang <derek.fang(a)realtek.com> --- include/sound/rt5682.h | 8 + sound/soc/codecs/rt5682.c | 389 +++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt5682.h | 4 +- 3 files changed, 397 insertions(+), 4 deletions(-) diff --git a/include/sound/rt5682.h b/include/sound/rt5682.h index bc2c317..6bf0e35 100644 --- a/include/sound/rt5682.h +++ b/include/sound/rt5682.h @@ -24,6 +24,12 @@ enum rt5682_jd_src { RT5682_JD1, }; +enum rt5682_dai_clks { + RT5682_DAI_WCLK_IDX, + RT5682_DAI_BCLK_IDX, + RT5682_DAI_NUM_CLKS, +}; + struct rt5682_platform_data { int ldo1_en; /* GPIO for LDO1_EN */ @@ -32,6 +38,8 @@ struct rt5682_platform_data { enum rt5682_dmic1_clk_pin dmic1_clk_pin; enum rt5682_jd_src jd_src; unsigned int btndet_delay; + + const char *dai_clk_names[RT5682_DAI_NUM_CLKS]; }; #endif diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 9fbb386..197e615 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -27,6 +27,9 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> #include <sound/rt5682.h> #include "rl6231.h" @@ -45,6 +48,8 @@ static const struct rt5682_platform_data i2s_default_platform_data = { .dmic1_clk_pin = RT5682_DMIC1_CLK_GPIO3, .jd_src = RT5682_JD1, .btndet_delay = 16, + .dai_clk_names[RT5682_DAI_WCLK_IDX] = "rt5682-dai-wclk", + .dai_clk_names[RT5682_DAI_BCLK_IDX] = "rt5682-dai-bclk", }; struct rt5682_priv { @@ -58,6 +63,13 @@ struct rt5682_priv { struct mutex calibrate_mutex; bool is_sdw; +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; + struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS]; + struct clk *dai_clks[RT5682_DAI_NUM_CLKS]; + struct clk *mclk; +#endif + int sysclk; int sysclk_src; int lrck[RT5682_AIFS]; @@ -921,6 +933,7 @@ static int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = &component->dapm; unsigned int val, count; if (jack_insert) { @@ -963,8 +976,13 @@ static int rt5682_headset_detect(struct snd_soc_component *component, rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, - RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); + if (snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); + else + snd_soc_component_update_bits(component, + RT5682_PWR_ANLG_1, + RT5682_PWR_VREF2 | RT5682_PWR_MB, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, 0); @@ -1633,6 +1651,7 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, NULL, 0), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -2459,12 +2478,362 @@ static int rt5682_set_bias_level(struct snd_soc_component *component, return 0; } +#ifdef CONFIG_COMMON_CLK +#define CLK_PLL2_FIN 48000000 +#define CLK_PLL2_FOUT 24576000 +#define CLK_48 48000 + +static bool rt5682_clk_check(struct rt5682_priv *rt5682) +{ + if (!rt5682->master[RT5682_AIF1]) { + dev_err(rt5682->component->dev, "sysclk/dai not set correctly\n"); + return false; + } + return true; +} + +static int rt5682_wclk_prepare(struct clk_hw *hw) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, RT5682_PWR_MB); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2F"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2B"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + + return 0; +} + +static void rt5682_wclk_unprepare(struct clk_hw *hw) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + if (!rt5682_clk_check(rt5682)) + return; + + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); + if (!rt5682->jack_type) + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, + RT5682_PWR_MB, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "I2S1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2F"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2B"); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); +} + +static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + + if (!rt5682_clk_check(rt5682)) + return 0; + + return CLK_48; +} + +static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + return CLK_48; +} + +static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_WCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct clk *parent_clk; + const char * const clk_name = __clk_get_name(hw->clk); + int pre_div; + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + parent_clk = clk_get_parent(hw->clk); + if (!parent_clk) + dev_warn(component->dev, + "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n", + CLK_PLL2_FIN); + + if (parent_rate != CLK_PLL2_FIN) + dev_warn(component->dev, "clk %s only support %d Hz input\n", + clk_name, CLK_PLL2_FIN); + + if (rate != CLK_48) { + dev_warn(component->dev, "clk %s only support %d Hz output\n", + clk_name, CLK_48); + rate = CLK_48; + } + + rt5682_set_component_pll(component, RT5682_PLL2, RT5682_PLL2_S_MCLK, + CLK_PLL2_FIN, CLK_PLL2_FOUT); + + rt5682_set_component_sysclk(component, RT5682_SCLK_S_PLL2, 0, + CLK_PLL2_FOUT, SND_SOC_CLOCK_IN); + + pre_div = rl6231_get_clk_info(rt5682->sysclk, rate); + + snd_soc_component_update_bits(component, RT5682_ADDA_CLK_1, + RT5682_I2S_M_DIV_MASK | RT5682_I2S_CLK_SRC_MASK, + pre_div << RT5682_I2S_M_DIV_SFT | + (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT); + + return 0; +} + +static unsigned long rt5682_bclk_recalc_rate(struct clk_hw *hw, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + unsigned int bclks_per_wclk; + + snd_soc_component_read(component, RT5682_TDM_TCON_CTRL, + &bclks_per_wclk); + + switch (bclks_per_wclk & RT5682_TDM_BCLK_MS1_MASK) { + case RT5682_TDM_BCLK_MS1_256: + return parent_rate * 256; + case RT5682_TDM_BCLK_MS1_128: + return parent_rate * 128; + case RT5682_TDM_BCLK_MS1_64: + return parent_rate * 64; + case RT5682_TDM_BCLK_MS1_32: + return parent_rate * 32; + default: + return 0; + } +} + +static unsigned long rt5682_bclk_get_factor(unsigned long rate, + unsigned long parent_rate) +{ + unsigned long factor; + + factor = rate / parent_rate; + if (factor < 64) + return 32; + else if (factor < 128) + return 64; + else if (factor < 256) + return 128; + else + return 256; +} + +static long rt5682_bclk_round_rate(struct clk_hw *hw, unsigned long rate, + unsigned long *parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + unsigned long factor; + + if (!*parent_rate || !rt5682_clk_check(rt5682)) + return -EINVAL; + + /* + * BCLK rates are set as a multiplier of WCLK in HW. + * We don't allow changing the parent WCLK. We just do + * some rounding down based on the parent WCLK rate + * and find the appropriate multiplier of BCLK to + * get the rounded down BCLK value. + */ + factor = rt5682_bclk_get_factor(rate, *parent_rate); + + return *parent_rate * factor; +} + +static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate, + unsigned long parent_rate) +{ + struct rt5682_priv *rt5682 = + container_of(hw, struct rt5682_priv, + dai_clks_hw[RT5682_DAI_BCLK_IDX]); + struct snd_soc_component *component = rt5682->component; + struct snd_soc_dai *dai = NULL; + unsigned long factor; + + if (!rt5682_clk_check(rt5682)) + return -EINVAL; + + factor = rt5682_bclk_get_factor(rate, parent_rate); + + for_each_component_dais(component, dai) + if (dai->id == RT5682_AIF1) + break; + if (!dai) { + dev_err(component->dev, "dai %d not found in component\n", + RT5682_AIF1); + return -ENODEV; + } + + return rt5682_set_bclk1_ratio(dai, factor); +} + +static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = { + [RT5682_DAI_WCLK_IDX] = { + .prepare = rt5682_wclk_prepare, + .unprepare = rt5682_wclk_unprepare, + .recalc_rate = rt5682_wclk_recalc_rate, + .round_rate = rt5682_wclk_round_rate, + .set_rate = rt5682_wclk_set_rate, + }, + [RT5682_DAI_BCLK_IDX] = { + .recalc_rate = rt5682_bclk_recalc_rate, + .round_rate = rt5682_bclk_round_rate, + .set_rate = rt5682_bclk_set_rate, + }, +}; + +static int rt5682_register_dai_clks(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct rt5682_platform_data *pdata = &rt5682->pdata; + struct clk_init_data init; + struct clk *dai_clk; + struct clk_lookup *dai_clk_lookup; + struct clk_hw *dai_clk_hw; + const char *parent_name; + int i, ret; + + for (i = 0; i < RT5682_DAI_NUM_CLKS; ++i) { + dai_clk_hw = &rt5682->dai_clks_hw[i]; + + switch (i) { + case RT5682_DAI_WCLK_IDX: + /* Make MCLK the parent of WCLK */ + if (rt5682->mclk) { + parent_name = __clk_get_name(rt5682->mclk); + init.parent_names = &parent_name; + init.num_parents = 1; + } else { + init.parent_names = NULL; + init.num_parents = 0; + } + break; + case RT5682_DAI_BCLK_IDX: + /* Make WCLK the parent of BCLK */ + parent_name = __clk_get_name( + rt5682->dai_clks[RT5682_DAI_WCLK_IDX]); + init.parent_names = &parent_name; + init.num_parents = 1; + break; + default: + dev_err(dev, "Invalid clock index\n"); + ret = -EINVAL; + goto err; + } + + init.name = pdata->dai_clk_names[i]; + init.ops = &rt5682_dai_clk_ops[i]; + init.flags = CLK_GET_RATE_NOCACHE | CLK_SET_RATE_GATE; + dai_clk_hw->init = &init; + + dai_clk = devm_clk_register(dev, dai_clk_hw); + if (IS_ERR(dai_clk)) { + dev_warn(dev, "Failed to register %s: %ld\n", + init.name, PTR_ERR(dai_clk)); + ret = PTR_ERR(dai_clk); + goto err; + } + rt5682->dai_clks[i] = dai_clk; + + if (dev->of_node) { + devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, + dai_clk_hw); + } else { + dai_clk_lookup = clkdev_create(dai_clk, init.name, + "%s", dev_name(dev)); + if (!dai_clk_lookup) { + ret = -ENOMEM; + goto err; + } else { + rt5682->dai_clks_lookup[i] = dai_clk_lookup; + } + } + } + + return 0; + +err: + do { + if (rt5682->dai_clks_lookup[i]) + clkdev_drop(rt5682->dai_clks_lookup[i]); + } while (i-- > 0); + + return ret; +} +#endif /* CONFIG_COMMON_CLK */ + static int rt5682_probe(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); +#ifdef CONFIG_COMMON_CLK + int ret; +#endif rt5682->component = component; +#ifdef CONFIG_COMMON_CLK + /* Check if MCLK provided */ + rt5682->mclk = devm_clk_get(component->dev, "mclk"); + if (IS_ERR(rt5682->mclk)) { + if (PTR_ERR(rt5682->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682->mclk); + return ret; + } + rt5682->mclk = NULL; + } + + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(component); + if (ret) + return ret; + + /* Initial setup for CCF */ + rt5682->lrck[RT5682_AIF1] = CLK_48; +#endif + return 0; } @@ -2472,6 +2841,15 @@ static void rt5682_remove(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); +#ifdef CONFIG_COMMON_CLK + int i; + + for (i = RT5682_DAI_NUM_CLKS - 1; i >= 0; --i) { + if (rt5682->dai_clks_lookup[i]) + clkdev_drop(rt5682->dai_clks_lookup[i]); + } +#endif + rt5682_reset(rt5682); } @@ -2606,6 +2984,13 @@ static int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev) rt5682->pdata.ldo1_en = of_get_named_gpio(dev->of_node, "realtek,ldo1-en-gpios", 0); + if (device_property_read_string_array(dev, "clock-output-names", + rt5682->pdata.dai_clk_names, + RT5682_DAI_NUM_CLKS) < 0) + dev_warn(dev, "Using default DAI clk names: %s, %s\n", + rt5682->pdata.dai_clk_names[RT5682_DAI_WCLK_IDX], + rt5682->pdata.dai_clk_names[RT5682_DAI_BCLK_IDX]); + return 0; } diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 465c99b..f82126a 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -841,8 +841,8 @@ #define RT5682_TDM_M_LP_INV (0x1 << 1) #define RT5682_TDM_MS_MASK (0x1 << 0) #define RT5682_TDM_MS_SFT 0 -#define RT5682_TDM_MS_M (0x0 << 0) -#define RT5682_TDM_MS_S (0x1 << 0) +#define RT5682_TDM_MS_S (0x0 << 0) +#define RT5682_TDM_MS_M (0x1 << 0) /* Global Clock Control (0x0080) */ #define RT5682_SCLK_SRC_MASK (0x7 << 13) -- 2.7.4

2 2

Re: [alsa-devel] [PATCH] ASoC: rt1015: fix typo for bypass boost control
by Mark Brown 14 Feb '20

14 Feb '20

On Wed, Feb 12, 2020 at 01:59:49AM +0000, Jack Yu wrote: > Fix typo for "Bypass Boost" control. This doesn't apply against current code, please check and resend. Also this is more of a cut'n'paste error than a typo.

2 2

[alsa-devel] [PATCH 0/3] ALSA: KCSAN fixes
by Takashi Iwai 14 Feb '20

14 Feb '20

Hi, recently syzbot reports repeatedly about the concurrent accesses with KCSAN. Those should be mostly harmless but still safer to be papered over, so here are fixes. There are two fixes for ALSA sequencer core, one about applying the spinlock for the queue bitmap flag accesses, and another about the current time retrievals. The last patch is about rawmidi and changes simply bit fields to bools. I didn't mark them Cc-to-stable because there are no reproducer and the patterns are basically safe -- although it's fine to backport these patches to stable branches. Takashi === Takashi Iwai (3): ALSA: seq: Avoid concurrent access to queue flags ALSA: seq: Fix concurrent access to queue current tick/time ALSA: rawmidi: Avoid bit fields for state flags include/sound/rawmidi.h | 6 +++--- sound/core/seq/seq_clientmgr.c | 4 ++-- sound/core/seq/seq_queue.c | 29 ++++++++++++++++++++++------- sound/core/seq/seq_timer.c | 13 ++++++++++--- sound/core/seq/seq_timer.h | 3 ++- 5 files changed, 39 insertions(+), 16 deletions(-) -- 2.16.4

1 3

[alsa-devel] [PATCH -next] ASoC: SOF: imx8: Fix randbuild error
by YueHaibing 14 Feb '20

14 Feb '20

when do randconfig like this: CONFIG_SND_SOC_SOF_IMX8_SUPPORT=y CONFIG_SND_SOC_SOF_IMX8=y CONFIG_SND_SOC_SOF_OF=y CONFIG_IMX_DSP=m CONFIG_IMX_SCU=y there is a link error: sound/soc/sof/imx/imx8.o: In function 'imx8_send_msg': imx8.c:(.text+0x380): undefined reference to 'imx_dsp_ring_doorbell' Select IMX_DSP in SND_SOC_SOF_IMX8_SUPPORT to fix this Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: f9ad75468453 ("ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index bae4f7b..81274906 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -14,7 +14,7 @@ if SND_SOC_SOF_IMX_TOPLEVEL config SND_SOC_SOF_IMX8_SUPPORT bool "SOF support for i.MX8" depends on IMX_SCU - depends on IMX_DSP + select IMX_DSP help This adds support for Sound Open Firmware for NXP i.MX8 platforms Say Y if you have such a device. -- 2.7.4

4 9

[alsa-devel] [asoc:for-5.7 65/70] sound/soc/meson/aiu.c:186:31-37: ERROR: application of sizeof to pointer
by kbuild test robot 14 Feb '20

14 Feb '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.7 head: e37a0c313a0f8ba0b8de9c30db98fbc77bd8d446 commit: 6ae9ca9ce986bffe71fd0fbf9595de8500891b52 [65/70] ASoC: meson: aiu: add i2s and spdif support If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> coccinelle warnings: (new ones prefixed by >>) >> sound/soc/meson/aiu.c:186:31-37: ERROR: application of sizeof to pointer -- >> sound/soc/meson/aiu.c:298:2-9: line 298 is redundant because platform_get_irq() already prints an error sound/soc/meson/aiu.c:304:2-9: line 304 is redundant because platform_get_irq() already prints an error -- >> sound/soc/meson/aiu.c:297:5-17: WARNING: Unsigned expression compared with zero: aiu -> i2s . irq < 0 >> sound/soc/meson/aiu.c:303:5-19: WARNING: Unsigned expression compared with zero: aiu -> spdif . irq < 0 vim +186 sound/soc/meson/aiu.c 177 178 static int aiu_clk_bulk_get(struct device *dev, 179 const char * const *ids, 180 unsigned int num, 181 struct aiu_interface *interface) 182 { 183 struct clk_bulk_data *clks; 184 int i, ret; 185 > 186 clks = devm_kcalloc(dev, num, sizeof(clks), GFP_KERNEL); 187 if (!clks) 188 return -ENOMEM; 189 190 for (i = 0; i < num; i++) 191 clks[i].id = ids[i]; 192 193 ret = devm_clk_bulk_get(dev, num, clks); 194 if (ret < 0) 195 return ret; 196 197 interface->clks = clks; 198 interface->clk_num = num; 199 return 0; 200 } 201 202 static const char * const aiu_i2s_ids[] = { 203 [PCLK] = "i2s_pclk", 204 [AOCLK] = "i2s_aoclk", 205 [MCLK] = "i2s_mclk", 206 [MIXER] = "i2s_mixer", 207 }; 208 209 static const char * const aiu_spdif_ids[] = { 210 [PCLK] = "spdif_pclk", 211 [AOCLK] = "spdif_aoclk", 212 [MCLK] = "spdif_mclk_sel" 213 }; 214 215 static int aiu_clk_get(struct device *dev) 216 { 217 struct aiu *aiu = dev_get_drvdata(dev); 218 int ret; 219 220 aiu->pclk = devm_clk_get(dev, "pclk"); 221 if (IS_ERR(aiu->pclk)) { 222 if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER) 223 dev_err(dev, "Can't get the aiu pclk\n"); 224 return PTR_ERR(aiu->pclk); 225 } 226 227 aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk"); 228 if (IS_ERR(aiu->spdif_mclk)) { 229 if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER) 230 dev_err(dev, "Can't get the aiu spdif master clock\n"); 231 return PTR_ERR(aiu->spdif_mclk); 232 } 233 234 ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids), 235 &aiu->i2s); 236 if (ret) { 237 if (ret != -EPROBE_DEFER) 238 dev_err(dev, "Can't get the i2s clocks\n"); 239 return ret; 240 } 241 242 ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids), 243 &aiu->spdif); 244 if (ret) { 245 if (ret != -EPROBE_DEFER) 246 dev_err(dev, "Can't get the spdif clocks\n"); 247 return ret; 248 } 249 250 ret = clk_prepare_enable(aiu->pclk); 251 if (ret) { 252 dev_err(dev, "peripheral clock enable failed\n"); 253 return ret; 254 } 255 256 ret = devm_add_action_or_reset(dev, 257 (void(*)(void *))clk_disable_unprepare, 258 aiu->pclk); 259 if (ret) 260 dev_err(dev, "failed to add reset action on pclk"); 261 262 return ret; 263 } 264 265 static int aiu_probe(struct platform_device *pdev) 266 { 267 struct device *dev = &pdev->dev; 268 void __iomem *regs; 269 struct regmap *map; 270 struct aiu *aiu; 271 int ret; 272 273 aiu = devm_kzalloc(dev, sizeof(*aiu), GFP_KERNEL); 274 if (!aiu) 275 return -ENOMEM; 276 platform_set_drvdata(pdev, aiu); 277 278 ret = device_reset(dev); 279 if (ret) { 280 if (ret != -EPROBE_DEFER) 281 dev_err(dev, "Failed to reset device\n"); 282 return ret; 283 } 284 285 regs = devm_platform_ioremap_resource(pdev, 0); 286 if (IS_ERR(regs)) 287 return PTR_ERR(regs); 288 289 map = devm_regmap_init_mmio(dev, regs, &aiu_regmap_cfg); 290 if (IS_ERR(map)) { 291 dev_err(dev, "failed to init regmap: %ld\n", 292 PTR_ERR(map)); 293 return PTR_ERR(map); 294 } 295 296 aiu->i2s.irq = platform_get_irq_byname(pdev, "i2s"); > 297 if (aiu->i2s.irq < 0) { > 298 dev_err(dev, "Can't get i2s irq\n"); 299 return aiu->i2s.irq; 300 } 301 302 aiu->spdif.irq = platform_get_irq_byname(pdev, "spdif"); > 303 if (aiu->spdif.irq < 0) { 304 dev_err(dev, "Can't get spdif irq\n"); 305 return aiu->spdif.irq; 306 } 307 308 ret = aiu_clk_get(dev); 309 if (ret) 310 return ret; 311 312 /* Register the cpu component of the aiu */ 313 ret = snd_soc_register_component(dev, &aiu_cpu_component, 314 aiu_cpu_dai_drv, 315 ARRAY_SIZE(aiu_cpu_dai_drv)); 316 if (ret) 317 dev_err(dev, "Failed to register cpu component\n"); 318 319 return ret; 320 } 321 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[alsa-devel] [PATCH 00/10] ASoC: soc-pcm cleanup step3
by Kuninori Morimoto 14 Feb '20

14 Feb '20

Hi Mark These are step3 of soc-pcm cleanup. Basically it do nothing for behavior, but cleanup code. But, only [09/10] patch exchanges behavior (= care Multi Codec). It is one challenge. Kuninori Morimoto (10): 1) ASoC: soc-pcm: move dai_get_widget() 2) ASoC: soc-pcm: use dai_get_widget() at dpcm_get_be() 3) ASoC: soc-pcm: use dai_get_widget() at dpcm_end_walk_at_be() 4) ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be() 5) ASoC: soc-pcm: remove soc_dpcm_be_digital_mute() 6) ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() 7) ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code 8) ASoC: soc-pcm: use goto and remove multi return 9) ASoC: soc-pcm: care Multi Codec at soc_dpcm_fe_runtime_update() 10) ASoC: soc.h: add for_each_pcm_stream() include/sound/soc-dpcm.h | 9 -- include/sound/soc.h | 5 + sound/soc/fsl/fsl_asrc_dma.c | 4 +- sound/soc/soc-core.c | 31 ++--- sound/soc/soc-pcm.c | 237 +++++++++++++---------------------- 5 files changed, 105 insertions(+), 181 deletions(-) -- 2.17.1 Thank you for your help !! Best regards --- Kuninori Morimoto

4 20

[alsa-devel] [PATCH] ASoC: dt-bindings: renesas, rsnd: switch to yaml base Documentation
by Kuninori Morimoto 14 Feb '20

14 Feb '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. It is still keeping detail explanations at .txt Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../bindings/sound/renesas,rsnd.txt | 518 ----------------- .../bindings/sound/renesas,rsnd.yaml | 528 ++++++++++++++++++ 2 files changed, 528 insertions(+), 518 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 797fd035434c..b731f16aea84 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -253,521 +253,3 @@ This is example of TDM 6ch. Driver can automatically switches TDM <-> stereo mode in this case. see "Example: simple sound card for TDM" - -============================================= -Required properties: -============================================= - -- compatible : "renesas,rcar_sound-<soctype>", fallbacks - "renesas,rcar_sound-gen1" if generation1, and - "renesas,rcar_sound-gen2" if generation2 (or RZ/G1) - "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) - Examples with soctypes are: - - "renesas,rcar_sound-r8a7743" (RZ/G1M) - - "renesas,rcar_sound-r8a7744" (RZ/G1N) - - "renesas,rcar_sound-r8a7745" (RZ/G1E) - - "renesas,rcar_sound-r8a77470" (RZ/G1C) - - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - - "renesas,rcar_sound-r8a7778" (R-Car M1A) - - "renesas,rcar_sound-r8a7779" (R-Car H1) - - "renesas,rcar_sound-r8a7790" (R-Car H2) - - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - - "renesas,rcar_sound-r8a7793" (R-Car M2-N) - - "renesas,rcar_sound-r8a7794" (R-Car E2) - - "renesas,rcar_sound-r8a7795" (R-Car H3) - - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - - "renesas,rcar_sound-r8a77990" (R-Car E3) - - "renesas,rcar_sound-r8a77995" (R-Car D3) -- reg : Should contain the register physical address. - required register is - SRU/ADG/SSI if generation1 - SRU/ADG/SSIU/SSI/AUDIO-DMAC-periperi if generation2/generation3 - Select extended AUDIO-DMAC-periperi address if SoC has it, - otherwise select normal AUDIO-DMAC-periperi address. -- reg-names : Should contain the register names. - scu/adg/ssi if generation1 - scu/adg/ssiu/ssi/audmapp if generation2/generation3 -- rcar_sound,ssi : Should contain SSI feature. - The number of SSI subnode should be same as HW. - see below for detail. -- rcar_sound,ssiu : Should contain SSIU feature. - The number of SSIU subnode should be same as HW. - see below for detail. -- rcar_sound,src : Should contain SRC feature. - The number of SRC subnode should be same as HW. - see below for detail. -- rcar_sound,ctu : Should contain CTU feature. - The number of CTU subnode should be same as HW. - see below for detail. -- rcar_sound,mix : Should contain MIX feature. - The number of MIX subnode should be same as HW. - see below for detail. -- rcar_sound,dvc : Should contain DVC feature. - The number of DVC subnode should be same as HW. - see below for detail. -- rcar_sound,dai : DAI contents. - The number of DAI subnode should be same as HW. - see below for detail. -- #sound-dai-cells : it must be 0 if your system is using single DAI - it must be 1 if your system is using multi DAI -- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. -- clock-names : List of necessary clock names. - "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", - "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" - -Optional properties: -- #clock-cells : it must be 0 if your system has audio_clkout - it must be 1 if your system has audio_clkout0/1/2/3 -- clock-frequency : for all audio_clkout0/1/2/3 -- clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn - is asynchronizes with lr-clock. -- resets : References to SSI resets. -- reset-names : List of valid reset names. - "ssi-all", "ssi.X" - -SSI subnode properties: -- interrupts : Should contain SSI interrupt for PIO transfer -- shared-pin : if shared clock pin -- pio-transfer : use PIO transfer mode -- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case -- dma : Should contain Audio DMAC entry -- dma-names : SSI case "rx" (=playback), "tx" (=capture) - Deprecated: see SSIU subnode properties - SSIU case "rxu" (=playback), "txu" (=capture) - -SSIU subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -SRC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -DVC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "tx" (=playback/capture) - -DAI subnode properties: -- playback : list of playback modules -- capture : list of capture modules - - -============================================= -Example: -============================================= - -rcar_sound: sound@ec500000 { - #sound-dai-cells = <1>; - compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; - reg = <0 0xec500000 0 0x1000>, /* SCU */ - <0 0xec5a0000 0 0x100>, /* ADG */ - <0 0xec540000 0 0x1000>, /* SSIU */ - <0 0xec541000 0 0x1280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ - reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; - - clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, - <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, - <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, - <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, - <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, - <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, - <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, - <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, - <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, - <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, - <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, - <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, - <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; - clock-names = "ssi-all", - "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", - "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", - "src.9", "src.8", "src.7", "src.6", "src.5", - "src.4", "src.3", "src.2", "src.1", "src.0", - "dvc.0", "dvc.1", - "clk_a", "clk_b", "clk_c", "clk_i"; - - rcar_sound,dvc { - dvc0: dvc-0 { - dmas = <&audma0 0xbc>; - dma-names = "tx"; - }; - dvc1: dvc-1 { - dmas = <&audma0 0xbe>; - dma-names = "tx"; - }; - }; - - rcar_sound,mix { - mix0: mix-0 { }; - mix1: mix-1 { }; - }; - - rcar_sound,ctu { - ctu00: ctu-0 { }; - ctu01: ctu-1 { }; - ctu02: ctu-2 { }; - ctu03: ctu-3 { }; - ctu10: ctu-4 { }; - ctu11: ctu-5 { }; - ctu12: ctu-6 { }; - ctu13: ctu-7 { }; - }; - - rcar_sound,src { - src0: src-0 { - interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x85>, <&audma1 0x9a>; - dma-names = "rx", "tx"; - }; - src1: src-1 { - interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x87>, <&audma1 0x9c>; - dma-names = "rx", "tx"; - }; - src2: src-2 { - interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x89>, <&audma1 0x9e>; - dma-names = "rx", "tx"; - }; - src3: src-3 { - interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8b>, <&audma1 0xa0>; - dma-names = "rx", "tx"; - }; - src4: src-4 { - interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8d>, <&audma1 0xb0>; - dma-names = "rx", "tx"; - }; - src5: src-5 { - interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8f>, <&audma1 0xb2>; - dma-names = "rx", "tx"; - }; - src6: src-6 { - interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x91>, <&audma1 0xb4>; - dma-names = "rx", "tx"; - }; - src7: src-7 { - interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x93>, <&audma1 0xb6>; - dma-names = "rx", "tx"; - }; - src8: src-8 { - interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x95>, <&audma1 0xb8>; - dma-names = "rx", "tx"; - }; - src9: src-9 { - interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x97>, <&audma1 0xba>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssiu { - ssiu00: ssiu-0 { - dmas = <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx"; - }; - ssiu01: ssiu-1 { - dmas = <&audma0 0x35>, <&audma1 0x36>; - dma-names = "rx", "tx"; - }; - - ... - - ssiu95: ssiu-49 { - dmas = <&audma0 0xA5>, <&audma1 0xA6>; - dma-names = "rx", "tx"; - }; - ssiu96: ssiu-50 { - dmas = <&audma0 0xA7>, <&audma1 0xA8>; - dma-names = "rx", "tx"; - }; - ssiu97: ssiu-51 { - dmas = <&audma0 0xA9>, <&audma1 0xAA>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssi { - ssi0: ssi-0 { - interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>; - dma-names = "rx", "tx"; - }; - ssi1: ssi-1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>; - dma-names = "rx", "tx"; - }; - - ... - - ssi8: ssi-8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>; - dma-names = "rx", "tx"; - }; - ssi9: ssi-9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,dai { - dai0 { - playback = <&ssi5 &src5>; - capture = <&ssi6>; - }; - dai1 { - playback = <&ssi3>; - }; - dai2 { - capture = <&ssi4>; - }; - dai3 { - playback = <&ssi7>; - }; - dai4 { - capture = <&ssi8>; - }; - }; -}; - -============================================= -Example: simple sound card -============================================= - - rsnd_ak4643: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; - }; - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &src2 &dvc0>; - capture = <&ssi1 &src3 &dvc1>; - }; - }; -}; - -&ssi1 { - shared-pin; -}; - -============================================= -Example: simple sound card for Asynchronous mode -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * SRC Asynchronous mode setting - * Playback: - * All input data will be converted to 48kHz - * Capture: - * Inputed 48kHz data will be converted to - * system specified Hz - */ - simple-audio-card,convert-rate = <48000>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for channel convert -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * CTU setting - * All input data will be converted to 2ch - * as output data - */ - simple-audio-card,convert-channels = <2>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for MIXer -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - simple-audio-card,cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - simple-audio-card,cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - simple-audio-card,codec { - ... - }; -}; - -&rcar_sound { - ... - rcar_sound,dai { - dai0 { - playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; - }; - dai1 { - playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; - }; - }; -}; - -============================================= -Example: simple sound card for TDM -============================================= - -rsnd_tdm: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - dai-tdm-slot-num = <6>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&xxx>; - }; -}; - -============================================= -Example: simple sound card for TDM Split -============================================= - -sound_card: sound { - compatible = "audio-graph-scu-card"; - prefix = "xxxx"; - routing = "xxxx Playback", "DAI0 Playback", - "xxxx Playback", "DAI1 Playback", - "xxxx Playback", "DAI2 Playback", - "xxxx Playback", "DAI3 Playback"; - convert-channels = <8>; /* TDM Split */ - - dais = <&rsnd_port0 /* playback ch1/ch2 */ - &rsnd_port1 /* playback ch3/ch4 */ - &rsnd_port2 /* playback ch5/ch6 */ - &rsnd_port3 /* playback ch7/ch8 */ - >; -}; - -audio-codec { - ... - port { - codec_0: endpoint@1 { - remote-endpoint = <&rsnd_ep0>; - }; - codec_1: endpoint@2 { - remote-endpoint = <&rsnd_ep1>; - }; - codec_2: endpoint@3 { - remote-endpoint = <&rsnd_ep2>; - }; - codec_3: endpoint@4 { - remote-endpoint = <&rsnd_ep3>; - }; - }; -}; - -&rcar_sound { - ... - ports { - rsnd_port0: port@0 { - rsnd_ep0: endpoint { - remote-endpoint = <&codec_0>; - ... - playback = <&ssiu30 &ssi3>; - }; - }; - rsnd_port1: port@1 { - rsnd_ep1: endpoint { - remote-endpoint = <&codec_1>; - ... - playback = <&ssiu31 &ssi3>; - }; - }; - rsnd_port2: port@2 { - rsnd_ep2: endpoint { - remote-endpoint = <&codec_2>; - ... - playback = <&ssiu32 &ssi3>; - }; - }; - rsnd_port3: port@3 { - rsnd_ep3: endpoint { - remote-endpoint = <&codec_3>; - ... - playback = <&ssiu33 &ssi3>; - }; - }; - }; -}; - -============================================= -Example: simple sound card for Multi channel -============================================= - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml new file mode 100644 index 000000000000..11afb9ee8321 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,528 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,rsnd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas R-Car Sound Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for Gen1 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7778 # R-Car M1A + - renesas,rcar_sound-r8a7779 # R-Car H1 + - enum: + - renesas,rcar_sound-gen1 + # for Gen2 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7743 # RZ/G1M + - renesas,rcar_sound-r8a7744 # RZ/G1N + - renesas,rcar_sound-r8a7745 # RZ/G1E + - renesas,rcar_sound-r8a77470 # RZ/G1C + - renesas,rcar_sound-r8a7790 # R-Car H2 + - renesas,rcar_sound-r8a7791 # R-Car M2-W + - renesas,rcar_sound-r8a7793 # R-Car M2-N + - renesas,rcar_sound-r8a7794 # R-Car E2 + - enum: + - renesas,rcar_sound-gen2 + # for Gen3 SoC + - items: + - enum: + - renesas,rcar_sound-r8a774a1 # RZ/G2M + - renesas,rcar_sound-r8a774b1 # RZ/G2N + - renesas,rcar_sound-r8a774c0 # RZ/G2E + - renesas,rcar_sound-r8a7795 # R-Car H3 + - renesas,rcar_sound-r8a7796 # R-Car M3-W + - renesas,rcar_sound-r8a77965 # R-Car M3-N + - renesas,rcar_sound-r8a77990 # R-Car E3 + - renesas,rcar_sound-r8a77995 # R-Car D3 + - enum: + - renesas,rcar_sound-gen3 + # for Generic + - items: + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + + reg: + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + + reg-names: + description: | + Should contain the register names. + scu/adg/ssi if generation1 + scu/adg/ssiu/ssi/audmapp if generation2/generation3 + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + allOf: + - items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + + "#sound-dai-cells": + description: | + it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAIs + enum: [0, 1] + + "#clock-cells": + description: | + it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 + enum: [0, 1] + + clock-frequency: + description: for audio_clkout0/1/2/3 + $ref: /schemas/types.yaml#/definitions/uint32-array + + clkout-lr-asynchronous: + description: audio_clkoutn is asynchronizes with lr-clock. + $ref: /schemas/types.yaml#/definitions/flag + + power-domains: true + + resets: + description: References to SSI resets + $ref: /schemas/types.yaml#/definitions/phandle-array + + reset-names: + description: List of valid reset names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + + clocks: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. + + clock-names: + description: List of necessary clock names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + - src.9 + - src.8 + - src.7 + - src.6 + - src.5 + - src.4 + - src.3 + - src.2 + - src.1 + - src.0 + - mix.1 + - mix.0 + - ctu.1 + - ctu.0 + - dvc.1 + - dvc.0 + - clk_a + - clk_b + - clk_c + - clk_i + + # For OF-graph + port: + description: OF-Graph subnode + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + endpoint: + type: object + properties: + remote-endpoint: + $ref: /schemas/types.yaml#/definitions/phandle-array + dai-format: + $ref: "simple-card.yaml#/definitions/format" + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + required: + - remote-endpoint + + # For multi OF-graph + ports: + description: multi OF-Graph subnode + type: object + patternProperties: + "port(@.*)?": + $ref: "#properties/port" + +patternProperties: + "^rcar_sound,dvc$": + description: DVC subnode. + type: object + patternProperties: + "dvc-.": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + const: "tx" + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,mix$": + description: MIX subnode. + type: object + patternProperties: + "mix-.": + type: object + additionalProperties: false + + "^rcar_sound,ctu$": + description: CTU subnode. + type: object + patternProperties: + "ctu-.": + type: object + additionalProperties: false + + "^rcar_sound,src$": + description: SRC subnode. + type: object + patternProperties: + "src-.": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + if: + properties: + status: + const: "ok" + then: + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssiu$": + description: SSIU subnode. + type: object + patternProperties: + "ssiu-.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssi$": + description: SSI subnode. + type: object + patternProperties: + "ssi-.": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + - txu # if no ssiu node + - rxu # if no ssiu node + + shared-pin: + description: shared clock pin + $ref: /schemas/types.yaml#/definitions/flag + pio-transfer: + description: PIO transfer mode + $ref: /schemas/types.yaml#/definitions/flag + no-busif: + description: BUSIF is not used when [mem -> SSI] via DMA case + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + # For DAI base + "^rcar_sound,dai$": + description: DAI subnode. + type: object + patternProperties: + "dai.*": + type: object + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + anyOf: + - required: + - playback + - required: + - capture + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2"; + reg = <0 0xec500000 0 0x1000>, /* SCU */ + <0 0xec5a0000 0 0x100>, /* ADG */ + <0 0xec540000 0 0x1000>, /* SSIU */ + <0 0xec541000 0 0x1280>, /* SSI */ + <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks 1005>, /* SSI-ALL */ + <&mstp10_clks 1006>, <&mstp10_clks 1007>, /* SSI9, SSI8 */ + <&mstp10_clks 1008>, <&mstp10_clks 1009>, /* SSI7, SSI6 */ + <&mstp10_clks 1010>, <&mstp10_clks 1011>, /* SSI5, SSI4 */ + <&mstp10_clks 1012>, <&mstp10_clks 1013>, /* SSI3, SSI2 */ + <&mstp10_clks 1014>, <&mstp10_clks 1015>, /* SSI1, SSI0 */ + <&mstp10_clks 1022>, <&mstp10_clks 1023>, /* SRC9, SRC8 */ + <&mstp10_clks 1024>, <&mstp10_clks 1025>, /* SRC7, SRC6 */ + <&mstp10_clks 1026>, <&mstp10_clks 1027>, /* SRC5, SRC4 */ + <&mstp10_clks 1028>, <&mstp10_clks 1029>, /* SRC3, SRC2 */ + <&mstp10_clks 1030>, <&mstp10_clks 1031>, /* SRC1, SRC0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* DVC0, DVC1 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* CLKA, CLKB */ + <&mstp10_clks 1019>, <&mstp10_clks 1018>; /* CLKC, CLKI */ + + clock-names = "ssi-all", + "ssi.9", "ssi.8", + "ssi.7", "ssi.6", + "ssi.5", "ssi.4", + "ssi.3", "ssi.2", + "ssi.1", "ssi.0", + "src.9", "src.8", + "src.7", "src.6", + "src.5", "src.4", + "src.3", "src.2", + "src.1", "src.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", + "clk_c", "clk_i"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + status = "disabled"; + }; + src1: src-1 { + interrupts = <0 353 0>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + /* skip after src-2 */ + }; + + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + /* skip after ssiu-2 */ + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <0 370 1>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <0 371 1>; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; + }; + /* skip other ssi-2 */ + }; + + /* DAI base */ + rcar_sound,dai { + dai0 { + playback = <&ssi5 &src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; + + /* OF-Graph */ + port { + rsnd_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + + /* Multi OF-Graph */ + ports { + #address-cells = <1>; + #size-cells = <0>; + rsnd_port0: port@0 { + reg = <0>; + rsnd_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint1>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + rsnd_port1: port@1 { + reg = <1>; + rsnd_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint2>; + + dai-format = "i2s"; + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + }; + }; + + + /* For OF-Graph */ + codec { + ports { + port@0 { + codec_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + port@1 { + codec_endpoint1: endpoint { + remote-endpoint = <&rsnd_endpoint0>; + }; + }; + port@2 { + codec_endpoint2: endpoint { + remote-endpoint = <&rsnd_endpoint1>; + }; + }; + }; + }; -- 2.17.1

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Re: [alsa-devel] linux-next: Tree for Feb 13 (sound/soc/codecs/snd-soc-ab8500-codec)
by Randy Dunlap 13 Feb '20

13 Feb '20

On 2/12/20 7:52 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20200212: > on x86_64: # CONFIG_ABX500_CORE is not set [this is in MFD] ERROR: "abx500_set_register_interruptible" [sound/soc/codecs/snd-soc-ab8500-codec.ko] undefined! ERROR: "abx500_get_register_interruptible" [sound/soc/codecs/snd-soc-ab8500-codec.ko] undefined! Full randconfig file is attached. -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

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[alsa-devel] [PATCH 0/6] ASoC: use for_each_rtd_codec_dai() macro
by Kuninori Morimoto 13 Feb '20

13 Feb '20

Hi Mark Some drivers aren't using for_each_rtd_codec_dai() macro. This patch-set switches to use it. Kuninori Morimoto (6): ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro drivers/soundwire/qcom.c | 7 ++++--- sound/soc/intel/boards/cml_rt1011_rt5682.c | 3 +-- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- .../mediatek/mt8183/mt8183-da7219-max98357.c | 8 ++++---- sound/soc/qcom/apq8016_sbc.c | 7 +++---- sound/soc/qcom/sdm845.c | 20 +++++++++---------- 6 files changed, 25 insertions(+), 28 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ALSA: pcm: Fix memory leak at closing a stream without hw_free
by Takashi Iwai 13 Feb '20

13 Feb '20

ALSA PCM core recently introduced a new managed PCM buffer allocation mode that does allocate / free automatically at hw_params and hw_free. However, it overlooked the code path directly calling hw_free PCM ops at releasing the PCM substream, and it may result in a memory leak as spotted by syzkaller when no buffer preallocation is used (e.g. vmalloc buffer). This patch papers over it with a slight refactoring. The hw_free ops call and relevant tasks are unified in a new helper function, and call it from both places. Fixes: 0dba808eae26 ("ALSA: pcm: Introduce managed buffer allocation mode") Reported-by: syzbot+30edd0f34bfcdc548ac4(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 24 +++++++++++++++--------- 1 file changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index bb23f5066654..4ac42ee1238c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -786,10 +786,22 @@ static int snd_pcm_hw_params_user(struct snd_pcm_substream *substream, return err; } +static int do_hw_free(struct snd_pcm_substream *substream) +{ + int result = 0; + + snd_pcm_sync_stop(substream); + if (substream->ops->hw_free) + result = substream->ops->hw_free(substream); + if (substream->managed_buffer_alloc) + snd_pcm_lib_free_pages(substream); + return result; +} + static int snd_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; - int result = 0; + int result; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; @@ -806,11 +818,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream) snd_pcm_stream_unlock_irq(substream); if (atomic_read(&substream->mmap_count)) return -EBADFD; - snd_pcm_sync_stop(substream); - if (substream->ops->hw_free) - result = substream->ops->hw_free(substream); - if (substream->managed_buffer_alloc) - snd_pcm_lib_free_pages(substream); + result = do_hw_free(substream); snd_pcm_set_state(substream, SNDRV_PCM_STATE_OPEN); pm_qos_remove_request(&substream->latency_pm_qos_req); return result; @@ -2529,9 +2537,7 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream) snd_pcm_drop(substream); if (substream->hw_opened) { - if (substream->ops->hw_free && - substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) - substream->ops->hw_free(substream); + do_hw_free(substream); substream->ops->close(substream); substream->hw_opened = 0; } -- 2.16.4

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[alsa-devel] [PATCH v2 0/5] soundwire: intel: add DAI callbacks
by Pierre-Louis Bossart 13 Feb '20

13 Feb '20

The existing mainline code is missing most of the DAI callbacks needed for a functional implementation, and the existing ones need to be modified to provide the relevant information to SOF drivers. As suggested by Vinod, these patches are shared first - with the risk that they are separated from the actual DAI enablement, so reviewers might wonder why they are needed in the first place. For reference, the complete set of 90+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 changes since v1: Fix string allocation (only feedback from Vinod) Pierre-Louis Bossart (2): soundwire: intel: rename res field as link_res soundwire: intel: free all resources on hw_free() Rander Wang (3): soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback drivers/soundwire/intel.c | 197 ++++++++++++++++++++++++++++++++++---- 1 file changed, 177 insertions(+), 20 deletions(-) -- 2.20.1

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[alsa-devel] [RESEND PATCH 0/4] ASoC: mediatek: mt8183-da7219: pick fixes from mt8183-mt6358
by Tzung-Bi Shih 13 Feb '20

13 Feb '20

Cherry-picks fixes from mt8183-mt6358-ts3a227-max98357 machine driver. The 1st patch from commit 1df1e5457cd1 ("ASoC: Mediatek: MT8183: change supported formats of DL2 and UL1"). The 2nd patch from commit 6191cbde5ff0 ("ASoC: mediatek: mt8183: switch tdm pins gpio function when playback on or off"). The 3rd patch from commit 8e58c521bcb8 ("ASoC: mediatek: mt8183: tdm hw support tdm out and 8ch i2s out"). The 4th patch squashes the following: - commit 63ba8e4d04ab ("ASoC: mediatek: mt8183: move card registration to the end of probe") - commit 95d779b4b8e7 ("ASoC: mediatek: mt8183: move headset jack to card-specific storage") - commit e5b3ae3e8c47 ("ASoC: mediatek: mt8183: fix space issues") Changes from previous version: (https://mailman.alsa-project.org/pipermail/alsa-devel/2020-February/162364.…) - None, only rebase and resend. Tzung-Bi Shih (4): ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1 ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out ASoC: mediatek: mt8183-da7219: apply some refactors .../mediatek/mt8183/mt8183-da7219-max98357.c | 164 ++++++++++++++++-- 1 file changed, 147 insertions(+), 17 deletions(-) -- 2.25.0.225.g125e21ebc7-goog

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[alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1" to the asoc tree
by Mark Brown 13 Feb '20

13 Feb '20

The patch ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1 has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ec7ba9e1500b0af5bf30b4e56bfaaf3a88850bbf Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih <tzungbi(a)google.com> Date: Thu, 13 Feb 2020 11:27:25 +0800 Subject: [PATCH] ASoC: mediatek: mt8183-da7219: change supported formats of DL2 and UL1 DL2 and UL1 are for BTSCO. Provides only 16-bit, mono, 8kHz and 16kHz to userspace. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20200213112003.1.Ie5aedb9d34ebfc7f05ceb382bfe346c… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 42 +++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 1626541cc0d6..b52ffed882a7 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -116,6 +116,46 @@ static int mt8183_i2s_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } +static int +mt8183_da7219_max98357_bt_sco_startup( + struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 8000, 16000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const unsigned int channels[] = { + 1, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); + runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + + return 0; +} + +static const struct snd_soc_ops mt8183_da7219_max98357_bt_sco_ops = { + .startup = mt8183_da7219_max98357_bt_sco_startup, +}; + /* FE */ SND_SOC_DAILINK_DEFS(playback1, DAILINK_COMP_ARRAY(COMP_CPU("DL1")), @@ -222,6 +262,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8183_da7219_max98357_bt_sco_ops, SND_SOC_DAILINK_REG(playback2), }, { @@ -240,6 +281,7 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { SND_SOC_DPCM_TRIGGER_PRE}, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8183_da7219_max98357_bt_sco_ops, SND_SOC_DAILINK_REG(capture1), }, { -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed" to the asoc tree
by Mark Brown 13 Feb '20

13 Feb '20

The patch ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8726ee6148fe24e2b29d4a961ad95c4ff8025d1d Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih <tzungbi(a)google.com> Date: Thu, 13 Feb 2020 11:27:26 +0800 Subject: [PATCH] ASoC: mediatek: mt8183-da7219: pull TDM GPIO pins down when probed 1. Switch TDM GPIO pins according to playback on or off. 2. Pull TDM GPIO pins down when probed to avoid current leakage. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20200213112003.2.I1d568b0c99742c6e755d051aadfd52e… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 104 +++++++++++++++++- 1 file changed, 98 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index b52ffed882a7..d7685916a5cb 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -18,6 +18,22 @@ static struct snd_soc_jack headset_jack; +enum PINCTRL_PIN_STATE { + PIN_STATE_DEFAULT = 0, + PIN_TDM_OUT_ON, + PIN_TDM_OUT_OFF, + PIN_STATE_MAX +}; + +static const char * const mt8183_pin_str[PIN_STATE_MAX] = { + "default", "aud_tdm_out_on", "aud_tdm_out_off", +}; + +struct mt8183_da7219_max98357_priv { + struct pinctrl *pinctrl; + struct pinctrl_state *pin_states[PIN_STATE_MAX]; +}; + static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -244,6 +260,47 @@ SND_SOC_DAILINK_DEFS(tdm, DAILINK_COMP_ARRAY(COMP_DUMMY()), DAILINK_COMP_ARRAY(COMP_EMPTY())); +static int mt8183_da7219_tdm_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_ON])) + return PTR_ERR(priv->pin_states[PIN_TDM_OUT_ON]); + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_ON]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + + return ret; +} + +static void mt8183_da7219_tdm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(rtd->card); + int ret; + + if (IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) + return; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); +} + +static struct snd_soc_ops mt8183_da7219_tdm_ops = { + .startup = mt8183_da7219_tdm_startup, + .shutdown = mt8183_da7219_tdm_shutdown, +}; + static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { /* FE */ { @@ -395,6 +452,8 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ignore_suspend = 1, + .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, + .ops = &mt8183_da7219_tdm_ops, SND_SOC_DAILINK_REG(tdm), }, }; @@ -470,7 +529,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) struct snd_soc_card *card = &mt8183_da7219_max98357_card; struct device_node *platform_node; struct snd_soc_dai_link *dai_link; - struct pinctrl *default_pins; + struct mt8183_da7219_max98357_priv *priv; int ret, i; card->dev = &pdev->dev; @@ -504,12 +563,45 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return ret; } - default_pins = - devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); - if (IS_ERR(default_pins)) { - dev_err(&pdev->dev, "%s set pins failed\n", + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + snd_soc_card_set_drvdata(card, priv); + + priv->pinctrl = devm_pinctrl_get(&pdev->dev); + if (IS_ERR(priv->pinctrl)) { + dev_err(&pdev->dev, "%s devm_pinctrl_get failed\n", __func__); - return PTR_ERR(default_pins); + return PTR_ERR(priv->pinctrl); + } + + for (i = 0; i < PIN_STATE_MAX; i++) { + priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl, + mt8183_pin_str[i]); + if (IS_ERR(priv->pin_states[i])) { + ret = PTR_ERR(priv->pin_states[i]); + dev_info(&pdev->dev, "%s Can't find pin state %s %d\n", + __func__, mt8183_pin_str[i], ret); + } + } + + if (!IS_ERR(priv->pin_states[PIN_TDM_OUT_OFF])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_TDM_OUT_OFF]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); + } + + if (!IS_ERR(priv->pin_states[PIN_STATE_DEFAULT])) { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_DEFAULT]); + if (ret) + dev_info(&pdev->dev, + "%s failed to select state %d\n", + __func__, ret); } return ret; -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out" to the asoc tree
by Mark Brown 13 Feb '20

13 Feb '20

The patch ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 195a6431710543ace009ff8e32dcf1e087506199 Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih <tzungbi(a)google.com> Date: Thu, 13 Feb 2020 11:27:27 +0800 Subject: [PATCH] ASoC: mediatek: mt8183-da7219: support TDM out and 8ch I2S out Supports TDM out and 8ch I2S out. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20200213112003.3.I30f0b8c87d5ec2a0e5f1b0fabf0a8cc… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index d7685916a5cb..c7f766f24e44 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -450,6 +450,9 @@ static struct snd_soc_dai_link mt8183_da7219_max98357_dai_links[] = { { .name = "TDM", .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_IF | + SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, .ignore_suspend = 1, .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, -- 2.20.1

1 0

[alsa-devel] Applied "ASoC: mediatek: mt8183-da7219: apply some refactors" to the asoc tree
by Mark Brown 13 Feb '20

13 Feb '20

The patch ASoC: mediatek: mt8183-da7219: apply some refactors has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ff0035e4c22371a29e4e0d4a07cdce5726fe50aa Mon Sep 17 00:00:00 2001 From: Tzung-Bi Shih <tzungbi(a)google.com> Date: Thu, 13 Feb 2020 11:27:28 +0800 Subject: [PATCH] ASoC: mediatek: mt8183-da7219: apply some refactors 1. Moves headset jack to card-specific storage. 2. Removes trailing blank line. 3. Moves card registration to the end of probe. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20200213112003.4.Ia542007f51d3de753a9e0a83135ee07… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../mediatek/mt8183/mt8183-da7219-max98357.c | 19 ++++++------------- 1 file changed, 6 insertions(+), 13 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index c7f766f24e44..c0c85972cfb7 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -16,8 +16,6 @@ #include "../../codecs/da7219-aad.h" #include "../../codecs/da7219.h" -static struct snd_soc_jack headset_jack; - enum PINCTRL_PIN_STATE { PIN_STATE_DEFAULT = 0, PIN_TDM_OUT_ON, @@ -32,6 +30,7 @@ static const char * const mt8183_pin_str[PIN_STATE_MAX] = { struct mt8183_da7219_max98357_priv { struct pinctrl *pinctrl; struct pinctrl_state *pin_states[PIN_STATE_MAX]; + struct snd_soc_jack headset_jack; }; static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, @@ -510,6 +509,8 @@ static int mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) { int ret; + struct mt8183_da7219_max98357_priv *priv = + snd_soc_card_get_drvdata(component->card); /* Enable Headset and 4 Buttons Jack detection */ ret = snd_soc_card_jack_new(&mt8183_da7219_max98357_card, @@ -517,12 +518,12 @@ mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, - &headset_jack, + &priv->headset_jack, NULL, 0); if (ret) return ret; - da7219_aad_jack_det(component, &headset_jack); + da7219_aad_jack_det(component, &priv->headset_jack); return ret; } @@ -559,13 +560,6 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) return -EINVAL; } - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) { - dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", - __func__, ret); - return ret; - } - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) return -ENOMEM; @@ -607,7 +601,7 @@ static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) __func__, ret); } - return ret; + return devm_snd_soc_register_card(&pdev->dev, card); } #ifdef CONFIG_OF @@ -634,4 +628,3 @@ MODULE_DESCRIPTION("MT8183-DA7219-MAX98357 ALSA SoC machine driver"); MODULE_AUTHOR("Shunli Wang <shunli.wang(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("mt8183_da7219_max98357 soc card"); - -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: rt5682: Enable PLL2 function
by derek.fang＠realtek.com 13 Feb '20

13 Feb '20

From: "derek.fang" <derek.fang(a)realtek.com> Enable RT5682 PLL2 function to implement the more complex frequency conversion. Signed-off-by: derek.fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682.c | 211 ++++++++++++++++++++++++++++++++++++---------- sound/soc/codecs/rt5682.h | 39 ++++++++- 2 files changed, 203 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 82a6366..9fbb386 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -64,9 +64,9 @@ struct rt5682_priv { int bclk[RT5682_AIFS]; int master[RT5682_AIFS]; - int pll_src; - int pll_in; - int pll_out; + int pll_src[RT5682_PLLS]; + int pll_in[RT5682_PLLS]; + int pll_out[RT5682_PLLS]; int jack_type; }; @@ -75,6 +75,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, {RT5682_I2C_CTRL, 0x000f}, + {RT5682_PLL2_INTERNAL, 0x8266}, }; static const struct reg_default rt5682_reg[] = { @@ -222,7 +223,7 @@ static const struct reg_default rt5682_reg[] = { {0x0148, 0x0000}, {0x0149, 0x0000}, {0x0150, 0x79a1}, - {0x0151, 0x0000}, + {0x0156, 0xaaaa}, {0x0160, 0x4ec0}, {0x0161, 0x0080}, {0x0162, 0x0200}, @@ -928,10 +929,10 @@ static int rt5682_headset_detect(struct snd_soc_component *component, RT5682_PWR_VREF2 | RT5682_PWR_MB, RT5682_PWR_VREF2 | RT5682_PWR_MB); snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, 0); usleep_range(15000, 20000); snd_soc_component_update_bits(component, - RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); + RT5682_PWR_ANLG_1, RT5682_PWR_FV2, RT5682_PWR_FV2); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, RT5682_PWR_CBJ, RT5682_PWR_CBJ); @@ -1298,6 +1299,21 @@ static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, return 0; } +static int is_sys_clk_from_pll2(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5682_GLB_CLK); + val &= RT5682_SCLK_SRC_MASK; + if (val == RT5682_SCLK_SRC_PLL2) + return 1; + else + return 0; +} + static int is_using_asrc(struct snd_soc_dapm_widget *w, struct snd_soc_dapm_widget *sink) { @@ -1612,9 +1628,11 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL2B", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2B_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL2F", RT5682_PWR_ANLG_3, RT5682_PWR_PLL2F_BIT, - 0, NULL, 0), + 0, set_filter_clk, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_SUPPLY("Vref1", RT5682_PWR_ANLG_1, RT5682_PWR_VREF1_BIT, 0, rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5682_PWR_ANLG_1, RT5682_PWR_VREF2_BIT, 0, + NULL, 0), /* ASRC */ SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5682_PLL_TRACK_1, @@ -1796,7 +1814,11 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { /*PLL*/ {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"ADC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2}, + {"ADC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2}, {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL2B", is_sys_clk_from_pll2}, + {"DAC Stereo1 Filter", NULL, "PLL2F", is_sys_clk_from_pll2}, /*ASRC*/ {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, @@ -2053,8 +2075,10 @@ static int rt5682_hw_params(struct snd_pcm_substream *substream, RT5682_I2S1_DL_MASK, len_1); if (rt5682->master[RT5682_AIF1]) { snd_soc_component_update_bits(component, - RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK, - pre_div << RT5682_I2S_M_DIV_SFT); + RT5682_ADDA_CLK_1, RT5682_I2S_M_DIV_MASK | + RT5682_I2S_CLK_SRC_MASK, + pre_div << RT5682_I2S_M_DIV_SFT | + (rt5682->sysclk_src) << RT5682_I2S_CLK_SRC_SFT); } if (params_channels(params) == 1) /* mono mode */ snd_soc_component_update_bits(component, @@ -2227,61 +2251,157 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, unsigned int freq_out) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - struct rl6231_pll_code pll_code; + struct rl6231_pll_code pll_code, pll2f_code, pll2b_code; + unsigned int pll2_fout1; int ret; - if (source == rt5682->pll_src && freq_in == rt5682->pll_in && - freq_out == rt5682->pll_out) + if (source == rt5682->pll_src[pll_id] && + freq_in == rt5682->pll_in[pll_id] && + freq_out == rt5682->pll_out[pll_id]) return 0; if (!freq_in || !freq_out) { dev_dbg(component->dev, "PLL disabled\n"); - rt5682->pll_in = 0; - rt5682->pll_out = 0; + rt5682->pll_in[pll_id] = 0; + rt5682->pll_out[pll_id] = 0; snd_soc_component_update_bits(component, RT5682_GLB_CLK, RT5682_SCLK_SRC_MASK, RT5682_SCLK_SRC_MCLK); return 0; } - switch (source) { - case RT5682_PLL1_S_MCLK: - snd_soc_component_update_bits(component, RT5682_GLB_CLK, - RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_MCLK); - break; - case RT5682_PLL1_S_BCLK1: - snd_soc_component_update_bits(component, RT5682_GLB_CLK, - RT5682_PLL1_SRC_MASK, RT5682_PLL1_SRC_BCLK1); - break; - default: - dev_err(component->dev, "Unknown PLL Source %d\n", source); - return -EINVAL; - } + if (pll_id == RT5682_PLL2) { + switch (source) { + case RT5682_PLL2_S_MCLK: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL2_SRC_MASK, + RT5682_PLL2_SRC_MCLK); + break; + default: + dev_err(component->dev, "Unknown PLL2 Source %d\n", + source); + return -EINVAL; + } - ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); - if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); - return ret; + /** + * PLL2 concatenates 2 PLL units. + * We suggest the Fout of the front PLL is 3.84MHz. + */ + pll2_fout1 = 3840000; + ret = rl6231_pll_calc(freq_in, pll2_fout1, &pll2f_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + freq_in); + return ret; + } + dev_dbg(component->dev, "PLL2F: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n", + freq_in, pll2_fout1, + pll2f_code.m_bp, + (pll2f_code.m_bp ? 0 : pll2f_code.m_code), + pll2f_code.n_code, pll2f_code.k_code); + + ret = rl6231_pll_calc(pll2_fout1, freq_out, &pll2b_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + pll2_fout1); + return ret; + } + dev_dbg(component->dev, "PLL2B: fin=%d fout=%d bypass=%d m=%d n=%d k=%d\n", + pll2_fout1, freq_out, + pll2b_code.m_bp, + (pll2b_code.m_bp ? 0 : pll2b_code.m_code), + pll2b_code.n_code, pll2b_code.k_code); + + snd_soc_component_write(component, RT5682_PLL2_CTRL_1, + pll2f_code.k_code << RT5682_PLL2F_K_SFT | + pll2b_code.k_code << RT5682_PLL2B_K_SFT | + pll2b_code.m_code); + snd_soc_component_write(component, RT5682_PLL2_CTRL_2, + pll2f_code.m_code << RT5682_PLL2F_M_SFT | + pll2b_code.n_code); + snd_soc_component_write(component, RT5682_PLL2_CTRL_3, + pll2f_code.n_code << RT5682_PLL2F_N_SFT); + snd_soc_component_update_bits(component, RT5682_PLL2_CTRL_4, + RT5682_PLL2B_M_BP_MASK | RT5682_PLL2F_M_BP_MASK | 0xf, + (pll2b_code.m_bp ? 1 : 0) << RT5682_PLL2B_M_BP_SFT | + (pll2f_code.m_bp ? 1 : 0) << RT5682_PLL2F_M_BP_SFT | + 0xf); + } else { + switch (source) { + case RT5682_PLL1_S_MCLK: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK, + RT5682_PLL1_SRC_MCLK); + break; + case RT5682_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, + RT5682_GLB_CLK, RT5682_PLL1_SRC_MASK, + RT5682_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL1 Source %d\n", + source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", + freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5682_PLL_CTRL_1, + pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5682_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | + pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); } - dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", - pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), - pll_code.n_code, pll_code.k_code); + rt5682->pll_in[pll_id] = freq_in; + rt5682->pll_out[pll_id] = freq_out; + rt5682->pll_src[pll_id] = source; + + return 0; +} + +static int rt5682_set_bclk1_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); - snd_soc_component_write(component, RT5682_PLL_CTRL_1, - pll_code.n_code << RT5682_PLL_N_SFT | pll_code.k_code); - snd_soc_component_write(component, RT5682_PLL_CTRL_2, - (pll_code.m_bp ? 0 : pll_code.m_code) << RT5682_PLL_M_SFT | - pll_code.m_bp << RT5682_PLL_M_BP_SFT | RT5682_PLL_RST); + rt5682->bclk[dai->id] = ratio; - rt5682->pll_in = freq_in; - rt5682->pll_out = freq_out; - rt5682->pll_src = source; + switch (ratio) { + case 256: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_256); + break; + case 128: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_128); + break; + case 64: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5682_TDM_TCON_CTRL, + RT5682_TDM_BCLK_MS1_MASK, RT5682_TDM_BCLK_MS1_32); + break; + default: + dev_err(dai->dev, "Invalid bclk1 ratio %d\n", ratio); + return -EINVAL; + } return 0; } -static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +static int rt5682_set_bclk2_ratio(struct snd_soc_dai *dai, unsigned int ratio) { struct snd_soc_component *component = dai->component; struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); @@ -2300,7 +2420,7 @@ static int rt5682_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) RT5682_I2S2_BCLK_MS2_32); break; default: - dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + dev_err(dai->dev, "Invalid bclk2 ratio %d\n", ratio); return -EINVAL; } @@ -2389,12 +2509,13 @@ static const struct snd_soc_dai_ops rt5682_aif1_dai_ops = { .hw_params = rt5682_hw_params, .set_fmt = rt5682_set_dai_fmt, .set_tdm_slot = rt5682_set_tdm_slot, + .set_bclk_ratio = rt5682_set_bclk1_ratio, }; static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { .hw_params = rt5682_hw_params, .set_fmt = rt5682_set_dai_fmt, - .set_bclk_ratio = rt5682_set_bclk_ratio, + .set_bclk_ratio = rt5682_set_bclk2_ratio, }; static struct snd_soc_dai_driver rt5682_dai[] = { diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index 4d3a8c4..465c99b 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -177,7 +177,7 @@ #define RT5682_TEST_MODE_CTRL_4 0x0148 #define RT5682_TEST_MODE_CTRL_5 0x0149 #define RT5682_PLL1_INTERNAL 0x0150 -#define RT5682_PLL2_INTERNAL 0x0151 +#define RT5682_PLL2_INTERNAL 0x0156 #define RT5682_STO_NG2_CTRL_1 0x0160 #define RT5682_STO_NG2_CTRL_2 0x0161 #define RT5682_STO_NG2_CTRL_3 0x0162 @@ -738,7 +738,7 @@ #define RT5682_ADC_OSR_D_24 (0x7 << 12) #define RT5682_ADC_OSR_D_32 (0x8 << 12) #define RT5682_ADC_OSR_D_48 (0x9 << 12) -#define RT5682_I2S_M_DIV_MASK (0xf << 12) +#define RT5682_I2S_M_DIV_MASK (0xf << 8) #define RT5682_I2S_M_DIV_SFT 8 #define RT5682_I2S_M_D_1 (0x0 << 8) #define RT5682_I2S_M_D_2 (0x1 << 8) @@ -820,6 +820,12 @@ #define RT5682_TDM_DF_PCM_B (0x3 << 11) #define RT5682_TDM_DF_PCM_A_N (0x6 << 11) #define RT5682_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5682_TDM_BCLK_MS1_MASK (0x3 << 9) +#define RT5682_TDM_BCLK_MS1_SFT 9 +#define RT5682_TDM_BCLK_MS1_32 (0x0 << 9) +#define RT5682_TDM_BCLK_MS1_64 (0x1 << 9) +#define RT5682_TDM_BCLK_MS1_128 (0x2 << 9) +#define RT5682_TDM_BCLK_MS1_256 (0x3 << 9) #define RT5682_TDM_CL_MASK (0x3 << 4) #define RT5682_TDM_CL_16 (0x0 << 4) #define RT5682_TDM_CL_20 (0x1 << 4) @@ -1049,6 +1055,28 @@ #define RT5682_PWR_CLK1M_PD (0x0 << 8) #define RT5682_PWR_CLK1M_PU (0x1 << 8) +/* PLL2 M/N/K Code Control 1 (0x009b) */ +#define RT5682_PLL2F_K_MASK (0x1f << 8) +#define RT5682_PLL2F_K_SFT 8 +#define RT5682_PLL2B_K_MASK (0xf << 4) +#define RT5682_PLL2B_K_SFT 4 +#define RT5682_PLL2B_M_MASK (0xf << 0) + +/* PLL2 M/N/K Code Control 2 (0x009c) */ +#define RT5682_PLL2F_M_MASK (0x3f << 8) +#define RT5682_PLL2F_M_SFT 8 +#define RT5682_PLL2B_N_MASK (0x3f << 0) + +/* PLL2 M/N/K Code Control 2 (0x009d) */ +#define RT5682_PLL2F_N_MASK (0x7f << 8) +#define RT5682_PLL2F_N_SFT 8 + +/* PLL2 M/N/K Code Control 2 (0x009e) */ +#define RT5682_PLL2B_M_BP_MASK (0x1 << 11) +#define RT5682_PLL2B_M_BP_SFT 11 +#define RT5682_PLL2F_M_BP_MASK (0x1 << 7) +#define RT5682_PLL2F_M_BP_SFT 7 + /* RC Clock Control (0x009f) */ #define RT5682_POW_IRQ (0x1 << 15) #define RT5682_POW_JDH (0x1 << 14) @@ -1315,6 +1343,13 @@ enum { RT5682_PLL1_S_MCLK, RT5682_PLL1_S_BCLK1, RT5682_PLL1_S_RCCLK, + RT5682_PLL2_S_MCLK, +}; + +enum { + RT5682_PLL1, + RT5682_PLL2, + RT5682_PLLS, }; enum { -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: doc: fix snd_hda_intel driver name
by Radoslaw Smigielski 13 Feb '20

13 Feb '20

Update driver name snd-hda-intel to proper, existing driver name snd_hda_intel in Documentation/sound/hd-audio/notes.rst. Signed-off-by: Radoslaw Smigielski <radoslaw.smigielski(a)interia.pl> --- Documentation/sound/hd-audio/notes.rst | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index 0f3109d9abc8..56ccc15f4d26 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -17,12 +17,12 @@ methods for the HD-audio hardware. The HD-audio component consists of two parts: the controller chip and the codec chips on the HD-audio bus. Linux provides a single driver -for all controllers, snd-hda-intel. Although the driver name contains +for all controllers, snd_hda_intel. Although the driver name contains a word of a well-known hardware vendor, it's not specific to it but for all controller chips by other companies. Since the HD-audio controllers are supposed to be compatible, the single snd-hda-driver should work in most cases. But, not surprisingly, there are known -bugs and issues specific to each controller type. The snd-hda-intel +bugs and issues specific to each controller type. The snd_hda_intel driver has a bunch of workarounds for these as described below. A controller may have multiple codecs. Usually you have one audio @@ -31,7 +31,7 @@ multiple audio codecs, e.g. for analog and digital outputs, and the driver might not work properly because of conflict of mixer elements. This should be fixed in future if such hardware really exists. -The snd-hda-intel driver has several different codec parsers depending +The snd_hda_intel driver has several different codec parsers depending on the codec. It has a generic parser as a fallback, but this functionality is fairly limited until now. Instead of the generic parser, usually the codec-specific parser (coded in patch_*.c) is used @@ -226,7 +226,7 @@ the external amplifier bits. Thus a headphone output has a slightly better chance. Before making a bug report, double-check whether the mixer is set up -correctly. The recent version of snd-hda-intel driver provides mostly +correctly. The recent version of snd_hda_intel driver provides mostly "Master" volume control as well as "Front" volume (where Front indicates the front-channels). In addition, there can be individual "Headphone" and "Speaker" controls. @@ -596,7 +596,7 @@ For example, if you have two cards, one for an on-board analog and one for an HDMI video board, you may pass patch option like below: :: - options snd-hda-intel patch=on-board-patch,hdmi-patch + options snd_hda_intel patch=on-board-patch,hdmi-patch Power-Saving -- 2.24.1

2 3

[alsa-devel] [PATCH v2 0/5] soundwire: stream: fix state machines and transitions
by Pierre-Louis Bossart 13 Feb '20

13 Feb '20

The existing stream support works fine with simple cases, but does not map well with ALSA transitions for underflows/resume where prepare() can be called multiple times. Concurrency with multiple devices per links or multiple streams enabled on the same link also needs to be fixed. These patches are the result of hours of validation on the Intel side and should benefit other implementations since there is nothing hardware-specific. The Intel-specific changes being reviewed do depend on those stream changes though to be functional. Changes since v1: Removed spurious code block change flagged by Vinod No change (replies provided in v1 thread) Github link issue is public, no reason to remove it Bandwidth computation on ALSA prepare/start (for resume cases) handled internally in stream layer. Kept emacs comment formatting. No additional code/test for concurrent streams (not supported due to locking) Bard Liao (1): soundwire: stream: only prepare stream when it is configured. Pierre-Louis Bossart (2): soundwire: stream: update state machine and add state checks soundwire: stream: do not update parameters during DISABLED-PREPARED transition Rander Wang (2): soundwire: stream: fix support for multiple Slaves on the same link soundwire: stream: don't program ports when a stream that has not been prepared Documentation/driver-api/soundwire/stream.rst | 61 +++++++++---- drivers/soundwire/stream.c | 90 ++++++++++++++++--- 2 files changed, 124 insertions(+), 27 deletions(-) -- 2.20.1

2 7

[alsa-devel] [PATCH] drm/mediatek: fix race condition of plugged_cb and codec_dev
by Tzung-Bi Shih 13 Feb '20

13 Feb '20

hdmi_conn_detect and mtk_hdmi_audio_hook_plugged_cb woule be called by different threads. plugged_cb and codec_dev are in danger of race condition. Fixes by: - Checks NULLs first. - Uses WRITE_ONCE() to prevent store tearing (i.e. write to plugged_cb after codec_dev). - Uses codec_dev as a signal to report HDMI jack status. The patch intends to not use mutex to protect the shared variables. In runtime, mtk_hdmi_audio_hook_plugged_cb would be called only once if no one rebinds the sound card. In contrast, hdmi_conn_detect could be called multiple times. Uses a mutex within hdmi_conn_detect calling path is a waste (no lock contention). Fixes: 5d3c64477392 ("drm/mediatek: support HDMI jack status reporting") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- drivers/gpu/drm/mediatek/mtk_hdmi.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/drivers/gpu/drm/mediatek/mtk_hdmi.c b/drivers/gpu/drm/mediatek/mtk_hdmi.c index 03aeb73005ef..746567c91bb7 100644 --- a/drivers/gpu/drm/mediatek/mtk_hdmi.c +++ b/drivers/gpu/drm/mediatek/mtk_hdmi.c @@ -1201,7 +1201,7 @@ mtk_hdmi_update_plugged_status(struct mtk_hdmi *hdmi) { bool connected = mtk_cec_hpd_high(hdmi->cec_dev); - if (hdmi->plugged_cb && hdmi->codec_dev) + if (hdmi->codec_dev) hdmi->plugged_cb(hdmi->codec_dev, connected); return connected ? @@ -1669,8 +1669,12 @@ static int mtk_hdmi_audio_hook_plugged_cb(struct device *dev, void *data, { struct mtk_hdmi *hdmi = data; - hdmi->plugged_cb = fn; - hdmi->codec_dev = codec_dev; + if (!fn || !codec_dev) + return -EINVAL; + + /* Use WRITE_ONCE() to prevent store tearing. */ + WRITE_ONCE(hdmi->plugged_cb, fn); + WRITE_ONCE(hdmi->codec_dev, codec_dev); mtk_hdmi_update_plugged_status(hdmi); return 0; -- 2.25.0.225.g125e21ebc7-goog

1 1

[alsa-devel] [PATCH v2 0/3] ASoC: mediatek: mt8173-rt5650: HDMI jack reporting
by Tzung-Bi Shih 13 Feb '20

13 Feb '20

The series supports HDMI jack reporting to mt8173-rt5650. The 1st patch is less related. It stops running in probe() if failed to register audio driver. The 2nd patch supports jack reporting in DRM driver. The 3rd patch supports jack reporting in mt8173-rt5650 ASoC machine driver. Changes from v1: - refactor the 2nd patch to simplify the code Tzung-Bi Shih (3): drm/mediatek: exit earlier if failed to register audio driver drm/mediatek: support HDMI jack status reporting ASoC: mediatek: mt8173-rt5650: support HDMI jack reporting drivers/gpu/drm/mediatek/mtk_hdmi.c | 45 ++++++++++++++++++++--- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 17 ++++++++- 2 files changed, 55 insertions(+), 7 deletions(-) -- 2.25.0.341.g760bfbb309-goog

3 11

[alsa-devel] [PATCH] ALSA: usb-audio: Add clock validity quirk for Denon MC7000/MCX8000
by Alexander Tsoy 13 Feb '20

13 Feb '20

It should be safe to ignore clock validity check result if the following conditions are met: - only one single sample rate is supported; - the terminal is directly connected to the clock source; - the clock type is internal. This is to deal with some Denon DJ controllers that always reports that clock is invalid. Tested-by: Tobias Oszlanyi <toszlanyi(a)yahoo.de> Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/clock.c | 91 ++++++++++++++++++++++++++++++++-------------- sound/usb/clock.h | 4 +- sound/usb/format.c | 3 +- 3 files changed, 66 insertions(+), 32 deletions(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 018b1ecb5404..a48313dfa967 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -151,8 +151,34 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return ret; } +/* + * Assume the clock is valid if clock source supports only one single sample + * rate, the terminal is connected directly to it (there is no clock selector) + * and clock type is internal. This is to deal with some Denon DJ controllers + * that always reports that clock is invalid. + */ +static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, + struct audioformat *fmt, + int source_id) +{ + if (fmt->protocol == UAC_VERSION_2) { + struct uac_clock_source_descriptor *cs_desc = + snd_usb_find_clock_source(chip->ctrl_intf, source_id); + + if (!cs_desc) + return false; + + return (fmt->nr_rates == 1 && + (fmt->clock & 0xff) == cs_desc->bClockID && + (cs_desc->bmAttributes & 0x3) != + UAC_CLOCK_SOURCE_TYPE_EXT); + } + + return false; +} + static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, - int protocol, + struct audioformat *fmt, int source_id) { int err; @@ -160,7 +186,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; u32 bmControls; - if (protocol == UAC_VERSION_3) { + if (fmt->protocol == UAC_VERSION_3) { struct uac3_clock_source_descriptor *cs_desc = snd_usb_find_clock_source_v3(chip->ctrl_intf, source_id); @@ -194,10 +220,14 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, return false; } - return data ? true : false; + if (data) + return true; + else + return uac_clock_source_is_valid_quirk(chip, fmt, source_id); } -static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, +static int __uac_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, int entity_id, unsigned long *visited, bool validate) { struct uac_clock_source_descriptor *source; @@ -217,7 +247,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, source = snd_usb_find_clock_source(chip->ctrl_intf, entity_id); if (source) { entity_id = source->bClockID; - if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_2, + if (validate && !uac_clock_source_is_valid(chip, fmt, entity_id)) { usb_audio_err(chip, "clock source %d is not valid, cannot use\n", @@ -248,8 +278,9 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, } cur = ret; - ret = __uac_clock_find_source(chip, selector->baCSourceID[ret - 1], - visited, validate); + ret = __uac_clock_find_source(chip, fmt, + selector->baCSourceID[ret - 1], + visited, validate); if (!validate || ret > 0 || !chip->autoclock) return ret; @@ -260,8 +291,9 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, if (i == cur) continue; - ret = __uac_clock_find_source(chip, selector->baCSourceID[i - 1], - visited, true); + ret = __uac_clock_find_source(chip, fmt, + selector->baCSourceID[i - 1], + visited, true); if (ret < 0) continue; @@ -281,14 +313,16 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, int entity_id, /* FIXME: multipliers only act as pass-thru element for now */ multiplier = snd_usb_find_clock_multiplier(chip->ctrl_intf, entity_id); if (multiplier) - return __uac_clock_find_source(chip, multiplier->bCSourceID, - visited, validate); + return __uac_clock_find_source(chip, fmt, + multiplier->bCSourceID, + visited, validate); return -EINVAL; } -static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, - unsigned long *visited, bool validate) +static int __uac3_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, int entity_id, + unsigned long *visited, bool validate) { struct uac3_clock_source_descriptor *source; struct uac3_clock_selector_descriptor *selector; @@ -307,7 +341,7 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, source = snd_usb_find_clock_source_v3(chip->ctrl_intf, entity_id); if (source) { entity_id = source->bClockID; - if (validate && !uac_clock_source_is_valid(chip, UAC_VERSION_3, + if (validate && !uac_clock_source_is_valid(chip, fmt, entity_id)) { usb_audio_err(chip, "clock source %d is not valid, cannot use\n", @@ -338,7 +372,8 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, } cur = ret; - ret = __uac3_clock_find_source(chip, selector->baCSourceID[ret - 1], + ret = __uac3_clock_find_source(chip, fmt, + selector->baCSourceID[ret - 1], visited, validate); if (!validate || ret > 0 || !chip->autoclock) return ret; @@ -350,8 +385,9 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, if (i == cur) continue; - ret = __uac3_clock_find_source(chip, selector->baCSourceID[i - 1], - visited, true); + ret = __uac3_clock_find_source(chip, fmt, + selector->baCSourceID[i - 1], + visited, true); if (ret < 0) continue; @@ -372,7 +408,8 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, multiplier = snd_usb_find_clock_multiplier_v3(chip->ctrl_intf, entity_id); if (multiplier) - return __uac3_clock_find_source(chip, multiplier->bCSourceID, + return __uac3_clock_find_source(chip, fmt, + multiplier->bCSourceID, visited, validate); return -EINVAL; @@ -389,18 +426,18 @@ static int __uac3_clock_find_source(struct snd_usb_audio *chip, int entity_id, * * Returns the clock source UnitID (>=0) on success, or an error. */ -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, - int entity_id, bool validate) +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, bool validate) { DECLARE_BITMAP(visited, 256); memset(visited, 0, sizeof(visited)); - switch (protocol) { + switch (fmt->protocol) { case UAC_VERSION_2: - return __uac_clock_find_source(chip, entity_id, visited, + return __uac_clock_find_source(chip, fmt, fmt->clock, visited, validate); case UAC_VERSION_3: - return __uac3_clock_find_source(chip, entity_id, visited, + return __uac3_clock_find_source(chip, fmt, fmt->clock, visited, validate); default: return -EINVAL; @@ -501,8 +538,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * automatic clock selection if the current clock is not * valid. */ - clock = snd_usb_clock_find_source(chip, fmt->protocol, - fmt->clock, true); + clock = snd_usb_clock_find_source(chip, fmt, true); if (clock < 0) { /* We did not find a valid clock, but that might be * because the current sample rate does not match an @@ -510,8 +546,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, * and we will do another validation after setting the * rate. */ - clock = snd_usb_clock_find_source(chip, fmt->protocol, - fmt->clock, false); + clock = snd_usb_clock_find_source(chip, fmt, false); if (clock < 0) return clock; } @@ -577,7 +612,7 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, validation: /* validate clock after rate change */ - if (!uac_clock_source_is_valid(chip, fmt->protocol, clock)) + if (!uac_clock_source_is_valid(chip, fmt, clock)) return -ENXIO; return 0; } diff --git a/sound/usb/clock.h b/sound/usb/clock.h index 076e31b79ee0..68df0fbe09d0 100644 --- a/sound/usb/clock.h +++ b/sound/usb/clock.h @@ -6,7 +6,7 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, struct usb_host_interface *alts, struct audioformat *fmt, int rate); -int snd_usb_clock_find_source(struct snd_usb_audio *chip, int protocol, - int entity_id, bool validate); +int snd_usb_clock_find_source(struct snd_usb_audio *chip, + struct audioformat *fmt, bool validate); #endif /* __USBAUDIO_CLOCK_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 9260136e4c9b..a77d6b1c403d 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -325,8 +325,7 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; unsigned char tmp[2], *data; int nr_triplets, data_size, ret = 0, ret_l6; - int clock = snd_usb_clock_find_source(chip, fp->protocol, - fp->clock, false); + int clock = snd_usb_clock_find_source(chip, fp, false); if (clock < 0) { dev_err(&dev->dev, -- 2.24.1

2 1

Re: [alsa-devel] Welcome to the "Alsa-devel" mailing list
by Srinivas Pulukuru 13 Feb '20

13 Feb '20

This is my first post to this mailing list, so apologize for any inconvenience if this is a wrong place to post such issues. If this is a wrong place please forward it to the right mailing list or let me know and I'll repost it there. ****************************** I are trying to create a 16 channel shared TDM frame that is shared between 8 different playback sources, to be sent out to the default hw:0 card. I used dshare plugin for this purpose, please see the attached asound.conf. I tried using the share plugin but the performance is really bad and is not practical for our use given the number of channels that we are using. when starting the playback the poll function called from snd_pcm_wait_nocheck in pcm.c hangs trying to access the /dev/snd/timer and never returns. It is able to access the first couple of times then hangs after that. ==> snd_pcm_write_areas ==> snd_pcm_wait_nocheck ==> poll() I see that the snd_pcm_direct_t->spcm->hw.ptr that tracks the frames in time is stuck and is not getting incremented in this case. I have seen several old posts regarding this issue and not sure which version has this been fixed or it still an issue. I have tried using dmix plugin too with the same result. My system doesn't have a sound card at the back end of the processor(IMX8M). I just need a shared buffer passed to the SAI interface that gets sent out of the SAI datalines (I2S) as a TDM frame. The codec drivers have been replaced with null terminated codecs as there at no DAC's connected to the processor. The SAI I2S datalines are connected to a FPGA that demultiplexes and processes the TDM frame to be sent out to multiple audio outputs. I am using alsa 1.1.2 and kernel imx8m 4.14.98 . I have tried to move to later version of alsa that I was able to build. I went till 1.1.7 but I still get the same issue. I have posted more details in the following link in case you need more details including the asound.conf and the batch file that runs 8 different pcm files to 8 different virtual sources connected to the same HW card. https://unix.stackexchange.com/questions/567295/alsa-16-channel-audio-playb… ******************************** Thanks, Srinivas Srinivas Pulukuru | Sr. Firmware Engineer Phone: 800-237-2041 Email: spulukuru(a)crestron.com Crestron Plano 7250 Dallas Parkway, Suite 600, Plano, Texas 75024 -----Original Message----- From: Alsa-devel <alsa-devel-bounces(a)alsa-project.org> On Behalf Of alsa-devel-request(a)alsa-project.org Sent: Wednesday, February 12, 2020 11:29 PM To: Srinivas Pulukuru <spulukuru(a)crestron.com> Subject: Welcome to the "Alsa-devel" mailing list Welcome to the Alsa-devel(a)alsa-project.org mailing list! To post to this list, send your message to: alsa-devel(a)alsa-project.org General information about the mailing list is at: https://urldefense.proofpoint.com/v2/url?u=https-3A__mailman.alsa-2Dproject… If you ever want to unsubscribe or change your options (eg, switch to or from digest mode, change your password, etc.), visit your subscription page at: https://urldefense.proofpoint.com/v2/url?u=https-3A__mailman.alsa-2Dproject… You can also make such adjustments via email by sending a message to: Alsa-devel-request(a)alsa-project.org with the word `help' in the subject or body (don't include the quotes), and you will get back a message with instructions. You must know your password to change your options (including changing the password, itself) or to unsubscribe without confirmation. It is: psr9611p Normally, Mailman will remind you of your alsa-project.org mailing list passwords once every month, although you can disable this if you prefer. This reminder will also include instructions on how to unsubscribe or change your account options. There is also a button on your options page that will email your current password to you.

1 0

[alsa-devel] [PATCH] ASoC: dt-bindings: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 13 Feb '20

13 Feb '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/simple-card.txt | 351 -------------- .../bindings/sound/simple-card.yaml | 459 ++++++++++++++++++ 2 files changed, 459 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6e37b..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 000000000000..7c9ced323787 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,459 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + $ref: /schemas/types.yaml#/definitions/phandle-array + + bitclock-master: + description: Indicates dai-link bit clock master + $ref: /schemas/types.yaml#/definitions/phandle-array + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + clocks: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/phandle-array + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": true + "#size-cells": true + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + + simple-audio-card,widgets: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + simple-audio-card,routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + sink, the second being the source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + "^simple-audio-card,cpu(@.*)?": + description: CPU node + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@.*)?": + description: Codec node + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" +# use common *-gpio for +# simple-audio-card,hp-det-gpio: +# simple-audio-card,mic-det-gpio: + + "^simple-audio-card,dai-link(@.*)?": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + # use common *-gpio for + # hp-det-gpio: + # mic-det-gpio: + + patternProperties: + "^cpu(@.*)?": + $ref: "#/definitions/dai" + "^codec(@.*)?": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.17.1

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[alsa-devel] USB Type-C monitor flashes once when play a video file after unplug and re-plug the monitor
by AJ_Cheng＠compal.com 13 Feb '20

13 Feb '20

Hi Sirs, Here is chromebook SW team from Compal. As the mail title, we hit issue that the external monitor will flash once when play video after hot pluging. We can reproduce not only on chromebook but also ubuntu 16.04. There has higher failure rate with Dell Solomon dock and Dell S2718D monitor. We found adding the delay in "sound/pci/hda/patch_hdmi.c " can fix this issue.(as the attachment) May need your help to review and advice. Thanks. Here is the issue number in gitlab for more detail. https://gitlab.freedesktop.org/drm/intel/issues/318 AJ Cheng NID/NID1 e-mail: AJ_Cheng(a)compal.com<mailto:AJ_Cheng@compal.com> Tel: +886-2-8797-8599 ext. 17561 Mobile : +886-932827829 COMPAL Electronics, Inc.

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[alsa-devel] [PATCH 0/3] Add new module driver for new ASRC
by Shengjiu Wang 13 Feb '20

13 Feb '20

Add new module driver for new ASRC in i.MX815/865 Shengjiu Wang (3): ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI and platform drivers .../devicetree/bindings/sound/fsl,easrc.txt | 57 + sound/soc/fsl/fsl_asrc.h | 11 +- sound/soc/fsl/fsl_asrc_common.h | 22 + sound/soc/fsl/fsl_easrc.c | 2265 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 668 +++++ sound/soc/fsl/fsl_easrc_dma.c | 440 ++++ 6 files changed, 3453 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.txt create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h create mode 100644 sound/soc/fsl/fsl_easrc_dma.c -- 2.21.0

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[alsa-devel] [PATCH] ASoC: sh: fsi: Restore devm_ioremap() alignment
by Geert Uytterhoeven 13 Feb '20

13 Feb '20

The alignment of the continuation of the devm_ioremap() call in fsi_probe() was broken. Join the lines, as all parameters can fit on a single line. Fixes: 4bdc0d676a643140 ("remove ioremap_nocache and devm_ioremap_nocache") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/sh/fsi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 4b35ef402604ecd8..5ef4221be6c3965c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1938,8 +1938,7 @@ static int fsi_probe(struct platform_device *pdev) if (!master) return -ENOMEM; - master->base = devm_ioremap(&pdev->dev, - res->start, resource_size(res)); + master->base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); if (!master->base) { dev_err(&pdev->dev, "Unable to ioremap FSI registers.\n"); return -ENXIO; -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: wm0010: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 13 Feb '20

13 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertenly introduced[3] to the codebase from now on. This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index abd2defe7530..d7a882a6adf2 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -46,7 +46,7 @@ struct dfw_binrec { u8 command; u32 length:24; u32 address; - uint8_t data[0]; + uint8_t data[]; } __packed; struct dfw_inforec { -- 2.25.0

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[alsa-devel] [PATCH 0/2] ASoC: mediatek: mt8183-da7219: add separate speaker switch
by Tzung-Bi Shih 13 Feb '20

13 Feb '20

Per previous discussion[1], the patch moves control of SD_MODE from PCM trigger to DAPM (i.e. set SD_MODE earilier if playback). Possible drawback: may generate pop noise when BCLK=on but LRCLK=off. As the datasheet[2] mentioned: > Do not remove LRCLK while BCLK is present. > Removing LRCLK while BCLK is present can cause unexpected output behavior > including a large DC output voltage. That is also the original purpose of sdmode-delay: to make sure SD_MODE is the last one to be set. [1]: https://patchwork.kernel.org/patch/11363953/ [2]: https://datasheets.maximintegrated.com/en/ds/MAX98357A-MAX98357B.pdf Tzung-Bi Shih (2): ASoC: max98357a: move control of SD_MODE to DAPM ASoC: mediatek: mt8183-da7219: add speaker switch sound/soc/codecs/max98357a.c | 36 +++++++++---------- .../mediatek/mt8183/mt8183-da7219-max98357.c | 19 ++++++++++ 2 files changed, 36 insertions(+), 19 deletions(-) -- 2.25.0.225.g125e21ebc7-goog

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[alsa-devel] [PATCH] dt-bindings: Convert the binding file google, cros-ec-codec.txt to yaml format.
by Dafna Hirschfeld 13 Feb '20

13 Feb '20

This was tested and verified with: make dt_binding_check DT_SCHEMA_FILES=Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml Signed-off-by: Dafna Hirschfeld <dafna.hirschfeld(a)collabora.com> --- .../bindings/sound/google,cros-ec-codec.txt | 44 ------------- .../bindings/sound/google,cros-ec-codec.yaml | 62 +++++++++++++++++++ 2 files changed, 62 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt deleted file mode 100644 index 8ca52dcc5572..000000000000 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.txt +++ /dev/null @@ -1,44 +0,0 @@ -Audio codec controlled by ChromeOS EC - -Google's ChromeOS EC codec is a digital mic codec provided by the -Embedded Controller (EC) and is controlled via a host-command interface. - -An EC codec node should only be found as a sub-node of the EC node (see -Documentation/devicetree/bindings/mfd/cros-ec.txt). - -Required properties: -- compatible: Must contain "google,cros-ec-codec" -- #sound-dai-cells: Should be 1. The cell specifies number of DAIs. - -Optional properties: -- reg: Pysical base address and length of shared memory region from EC. - It contains 3 unsigned 32-bit integer. The first 2 integers - combine to become an unsigned 64-bit physical address. The last - one integer is length of the shared memory. -- memory-region: Shared memory region to EC. A "shared-dma-pool". See - ../reserved-memory/reserved-memory.txt for details. - -Example: - -{ - ... - - reserved_mem: reserved_mem { - compatible = "shared-dma-pool"; - reg = <0 0x52800000 0 0x100000>; - no-map; - }; -} - -cros-ec@0 { - compatible = "google,cros-ec-spi"; - - ... - - cros_ec_codec: ec-codec { - compatible = "google,cros-ec-codec"; - #sound-dai-cells = <1>; - reg = <0x0 0x10500000 0x80000>; - memory-region = <&reserved_mem>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml new file mode 100644 index 000000000000..94a85d0cbf43 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -0,0 +1,62 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/google,cros-ec-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Audio codec controlled by ChromeOS EC + +maintainers: + - Cheng-Yi Chiang <cychiang(a)chromium.org> + +description: | + Google's ChromeOS EC codec is a digital mic codec provided by the + Embedded Controller (EC) and is controlled via a host-command interface. + An EC codec node should only be found as a sub-node of the EC node (see + Documentation/devicetree/bindings/mfd/cros-ec.txt). + +properties: + compatible: + const: google,cros-ec-codec + + "#sound-dai-cells": + const: 1 + + reg: + items: + - description: | + Physical base address and length of shared memory region from EC. + It contains 3 unsigned 32-bit integer. The first 2 integers + combine to become an unsigned 64-bit physical address. + The last one integer is the length of the shared memory. + + memory-region: + $ref: '/schemas/types.yaml#/definitions/phandle' + description: | + Shared memory region to EC. A "shared-dma-pool". + See ../reserved-memory/reserved-memory.txt for details. + +required: + - compatible + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + reserved_mem: reserved_mem { + compatible = "shared-dma-pool"; + reg = <0 0x52800000 0 0x100000>; + no-map; + }; + cros-ec@0 { + compatible = "google,cros-ec-spi"; + #address-cells = <2>; + #size-cells = <1>; + cros_ec_codec: ec-codec { + compatible = "google,cros-ec-codec"; + #sound-dai-cells = <1>; + reg = <0x0 0x10500000 0x80000>; + memory-region = <&reserved_mem>; + }; + }; -- 2.17.1

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[alsa-devel] [PATCH 0/3] ASoC: Fix SND_SOC_ALL_CODECS imply fallout
by Geert Uytterhoeven 13 Feb '20

13 Feb '20

Hi Mark, As expected[*], commit ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS") introduced several build failures, due to missing dependencies. This patch series fixes several of them. There are probably more. Do we want to pursue this, or just revert instead? Thanks! [*] https://lore.kernel.org/alsa-devel/20200207091351.18133-1-geert@linux-m68k.… Geert Uytterhoeven (3): ASoC: Fix SND_SOC_ALL_CODECS imply SPI fallout ASoC: Fix SND_SOC_ALL_CODECS imply I2C fallout ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout sound/soc/codecs/Kconfig | 88 +++++++++++++++++++++++++++++++++++++--- 1 file changed, 82 insertions(+), 6 deletions(-) -- 2.17.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[alsa-devel] [patch] tas571x: use symbolic register offsets
by Hermann Lauer 12 Feb '20

12 Feb '20

Hello, the appended patch converts the numeric register offsets in tas5711_reg_defaults to the defined constants TAS571X_* Tested with a tas5713 chip for which the driver works, too. Thanks, greetings Hermann -- Administration/Zentrale Dienste, Interdiziplinaeres Zentrum fuer wissenschaftliches Rechnen der Universitaet Heidelberg IWR; INF 205; 69120 Heidelberg; Tel: (06221)54-14405 Fax: -14427 Email: Hermann.Lauer(a)iwr.uni-heidelberg.de

1 0

[alsa-devel] [PATCH v3 6/6] ASoC: qdsp6: dt-bindings: Add q6afe pcm dt binding documentation
by Adam Serbinski 12 Feb '20

12 Feb '20

This patch adds documentation of bindings required for PCM ports on AFE. Signed-off-by: Adam Serbinski <adam(a)serbinski.com> CC: Andy Gross <agross(a)kernel.org> CC: Mark Rutland <mark.rutland(a)arm.com> CC: Liam Girdwood <lgirdwood(a)gmail.com> CC: Patrick Lai <plai(a)codeaurora.org> CC: Banajit Goswami <bgoswami(a)codeaurora.org> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.com> CC: alsa-devel(a)alsa-project.org CC: linux-arm-msm(a)vger.kernel.org CC: devicetree(a)vger.kernel.org CC: linux-kernel(a)vger.kernel.org --- .../devicetree/bindings/sound/qcom,q6afe.txt | 42 +++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt index d74888b9f1bb..6b1b17d31a2a 100644 --- a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -51,6 +51,24 @@ configuration of each dai. Must contain the following properties. Definition: Must be list of serial data lines used by this dai. should be one or more of the 0-3 sd lines. + - qcom,pcm-quantype + Usage: required for pcm interface + Value type: <u32> + Definition: PCM quantization type + 0 - ALAW, no padding + 1 - MULAW, no padding + 2 - Linear, no padding + 3 - ALAW, padding + 4 - MULAW, padding + 5 - Linear, padding + + - qcom,pcm-slot-mapping + Usage: required for pcm interface + Value type: <prop-encoded-array> + Definition: Slot mapping for audio channels. Array size is the number + of slots, minimum 1, maximum 4. The value is 0 for no mapping + to the slot, or the channel number from 1 to 32. + - qcom,tdm-sync-mode: Usage: required for tdm interface Value type: <prop-encoded-array> @@ -174,5 +192,29 @@ q6afe@4 { reg = <23>; qcom,sd-lines = <1>; }; + + pri-pcm-rx@105 { + reg = <105>; + qcom,pcm-quantype = <2>; + qcom,pcm-slot-mapping = <1>; + }; + + pri-pcm-tx@106 { + reg = <106>; + qcom,pcm-quantype = <2>; + qcom,pcm-slot-mapping = <1>; + }; + + quat-pcm-rx@111 { + reg = <111>; + qcom,pcm-quantype = <5>; + qcom,pcm-slot-mapping = <0 0 1>; + }; + + quat-pcm-tx@112 { + reg = <112>; + qcom,pcm-quantype = <5>; + qcom,pcm-slot-mapping = <0 0 1>; + }; }; }; -- 2.21.1

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