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

[PATCH] ALSA: usb-audio: Skip probe of UA-101 devices
by Takashi Iwai 08 Apr '21

08 Apr '21

UA-101 device and co are supported by another driver, snd-ua101, but the USB audio class driver (snd-usb-audio) catches all and this resulted in the lack of functionality like missing MIDI devices. This patch introduces a sort of deny-listing for those devices to just return -ENODEV at probe in snd-usb-audio driver, so that it falls back to the probe by snd-ua101. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=212477 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 2 ++ sound/usb/quirks-table.h | 21 +++++++++++++++++++++ sound/usb/usbaudio.h | 1 + 3 files changed, 24 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index b6f4c0848e66..a482287a2d7f 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -715,6 +715,8 @@ static int usb_audio_probe(struct usb_interface *intf, quirk = get_alias_quirk(dev, id); if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) return -ENXIO; + if (quirk && quirk->ifnum == QUIRK_NODEV_INTERFACE) + return -ENODEV; err = snd_usb_apply_boot_quirk(dev, intf, quirk, id); if (err < 0) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 1165a5ac60f2..1f555d7b7752 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1674,6 +1674,27 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, + +/* UA101 and co are supported by another driver */ +{ + USB_DEVICE(0x0582, 0x0044), /* UA-1000 high speed */ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_NODEV_INTERFACE + }, +}, +{ + USB_DEVICE(0x0582, 0x007d), /* UA-101 high speed */ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_NODEV_INTERFACE + }, +}, +{ + USB_DEVICE(0x0582, 0x008d), /* UA-101 full speed */ + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_NODEV_INTERFACE + }, +}, + /* this catches most recent vendor-specific Roland devices */ { .match_flags = USB_DEVICE_ID_MATCH_VENDOR | diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 60b9dd7df6bb..7f91ec1c5646 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -75,6 +75,7 @@ struct snd_usb_audio { */ /* special values for .ifnum */ +#define QUIRK_NODEV_INTERFACE -3 /* return -ENODEV */ #define QUIRK_NO_INTERFACE -2 #define QUIRK_ANY_INTERFACE -1 -- 2.26.2

1 0

[PATCH] slimbus: qcom-ngd-ctrl: Fix runtime PM imbalance in qcom_slim_ngd_enable
by Dinghao Liu 08 Apr '21

08 Apr '21

When slim_register_controller() fails, a pairing PM usage counter increment is needed to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- drivers/slimbus/qcom-ngd-ctrl.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index c054e83ab636..99cf2ab3d862 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -1268,6 +1268,7 @@ static int qcom_slim_ngd_enable(struct qcom_slim_ngd_ctrl *ctrl, bool enable) ret = slim_register_controller(&ctrl->ctrl); if (ret) { dev_err(ctrl->dev, "error adding slim controller\n"); + pm_runtime_get_noresume(ctrl->dev); return ret; } -- 2.17.1

1 0

[PATCH] ASoC: codecs: Fix rumtime PM imbalance in tas2552_probe
by Dinghao Liu 08 Apr '21

08 Apr '21

There is a rumtime PM imbalance between the error handling path after devm_snd_soc_register_component() and all other error handling paths. Fix this by moving PM runtime decrement to the end of the function. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> --- sound/soc/codecs/tas2552.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index bd00c35116cd..52de6f7b4227 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -718,13 +718,6 @@ static int tas2552_probe(struct i2c_client *client, return ret; } - pm_runtime_set_active(&client->dev); - pm_runtime_set_autosuspend_delay(&client->dev, 1000); - pm_runtime_use_autosuspend(&client->dev); - pm_runtime_enable(&client->dev); - pm_runtime_mark_last_busy(&client->dev); - pm_runtime_put_sync_autosuspend(&client->dev); - dev_set_drvdata(&client->dev, data); ret = devm_snd_soc_register_component(&client->dev, @@ -733,6 +726,13 @@ static int tas2552_probe(struct i2c_client *client, if (ret < 0) dev_err(&client->dev, "Failed to register component: %d\n", ret); + pm_runtime_set_active(&client->dev); + pm_runtime_set_autosuspend_delay(&client->dev, 1000); + pm_runtime_use_autosuspend(&client->dev); + pm_runtime_enable(&client->dev); + pm_runtime_mark_last_busy(&client->dev); + pm_runtime_put_sync_autosuspend(&client->dev); + return ret; } -- 2.17.1

3 2

Implicit feedback on BOSS GT-1, the saga continues...
by Mike Oliphant 08 Apr '21

08 Apr '21

I had thought that the recent implicit feedback changes were fully working on the BOSS GT-1, but it turns out that I just hadn't tested well enough. Audio playback and capture works, but with periodic dropouts. I get the exact same behavior as I did with the quirk to completely disable implicit feedback. Without the implicit feedback, you get dropouts from clock drift - how bad probably varies from card to card. On mine it is every second or so. If I switch playback feedback for the GT-1 to generic by doing "IMPLICIT_FB_GENERIC_DEV(0x0582, 0x01d6)", I get the previous old behavior, which is that playback completely fails to start. With generic playback feedback, and using my previous patch to endpoint.c to avoid playback waiting on capture mentioned here: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-January/161951.h… playback and capture work perfectly for me.

3 12

[PATCH] sound:ppc: fix spelling typo of values
by caizhichao 08 Apr '21

08 Apr '21

From: caizhichao <caizhichao(a)yulong.com> vaules -> values Signed-off-by: caizhichao <caizhichao(a)yulong.com> --- sound/ppc/snd_ps3_reg.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ppc/snd_ps3_reg.h b/sound/ppc/snd_ps3_reg.h index 566a318..e2212b7 100644 --- a/sound/ppc/snd_ps3_reg.h +++ b/sound/ppc/snd_ps3_reg.h @@ -308,7 +308,7 @@ each interrupt in this register. Writing 1b to a field containing 1b clears field and de-asserts interrupt. Writing 0b to a field has no effect. -Field vaules are the following: +Field values are the following: 0 - Interrupt hasn't occurred. 1 - Interrupt has occurred. -- 1.9.1

2 1

[PATCH] ASoC: simple-card: fix possible uninitialized single_cpu local variable
by Krzysztof Kozlowski 07 Apr '21

07 Apr '21

The 'single_cpu' local variable is assigned by asoc_simple_parse_dai() and later used in a asoc_simple_canonicalize_cpu() call, assuming the entire function did not exit on errors. However the first function returns 0 if passed device_node is NULL, thus leaving the variable uninitialized and reporting success. Addresses-Coverity: Uninitialized scalar variable Fixes: 8f7f298a3337 ("ASoC: simple-card-utils: separate asoc_simple_card_parse_dai()") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)canonical.com> --- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/simple-card.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index ee1d924d68e5..76036ea377a9 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -367,7 +367,7 @@ static int graph_dai_link_of(struct asoc_simple_priv *priv, struct device_node *top = dev->of_node; struct asoc_simple_dai *cpu_dai; struct asoc_simple_dai *codec_dai; - int ret, single_cpu; + int ret, single_cpu = 0; dev_dbg(dev, "link_of (%pOF)\n", cpu_ep); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 41aa40765a8d..9a05f44fc3a9 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -249,7 +249,7 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, struct device_node *plat = NULL; char prop[128]; char *prefix = ""; - int ret, single_cpu; + int ret, single_cpu = 0; cpu = np; node = of_get_parent(np); -- 2.25.1

3 2

[PATCH] ASoC: meson: axg-fifo: add NO_PERIOD_WAKEUP support
by Jerome Brunet 07 Apr '21

07 Apr '21

On the AXG family, the fifo irq is not necessary for the HW to operate. It is just used to notify that a period has elapsed. If userpace does not care for these wakeups (such as pipewire), we are just wasting CPU cycles. Add support for NO_PERIOD_WAKEUP and disable irq when they are no needed. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-fifo.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index b2e867113226..b9af2d513e09 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -27,8 +27,8 @@ static struct snd_pcm_hardware axg_fifo_hw = { SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), - + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), .formats = AXG_FIFO_FORMATS, .rate_min = 5512, .rate_max = 192000, @@ -113,7 +113,7 @@ int axg_fifo_pcm_hw_params(struct snd_soc_component *component, { struct snd_pcm_runtime *runtime = ss->runtime; struct axg_fifo *fifo = axg_fifo_data(ss); - unsigned int burst_num, period, threshold; + unsigned int burst_num, period, threshold, irq_en; dma_addr_t end_ptr; period = params_period_bytes(params); @@ -142,10 +142,11 @@ int axg_fifo_pcm_hw_params(struct snd_soc_component *component, regmap_field_write(fifo->field_threshold, threshold ? threshold - 1 : 0); - /* Enable block count irq */ + /* Enable irq if necessary */ + irq_en = runtime->no_period_wakeup ? 0 : FIFO_INT_COUNT_REPEAT; regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT), - CTRL0_INT_EN(FIFO_INT_COUNT_REPEAT)); + CTRL0_INT_EN(irq_en)); return 0; } -- 2.30.2

2 1

[PATCH 0/4] Adds SPI support
by Jiri Prchal 07 Apr '21

07 Apr '21

Moved I2C stuff to separated file, shered thingk to header file. Added SPI in separeted file. Jiri Prchal (4): ASoC: codecs: tlv320aic3x: move model definitions ASoC: codecs: tlv320aic3x: rename probe function ASoC: codecs: tlv320aic3x: move I2C to separated file ASoC: codecs: tlv320aic3x: add SPI support sound/soc/codecs/Kconfig | 16 ++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/tlv320aic3x-i2c.c | 70 +++++++++++++++++++ sound/soc/codecs/tlv320aic3x-spi.c | 76 ++++++++++++++++++++ sound/soc/codecs/tlv320aic3x.c | 108 ++++++++--------------------- sound/soc/codecs/tlv320aic3x.h | 12 ++++ 6 files changed, 204 insertions(+), 82 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic3x-i2c.c create mode 100644 sound/soc/codecs/tlv320aic3x-spi.c -- 2.25.1

3 8

[PATCH -next] ASoC: codecs: lpass-va-macro: constify static struct snd_soc_dai_ops
by Ye Bin 07 Apr '21

07 Apr '21

The snd_soc_dai_ops structures is only stored in the ops field of a snd_soc_dai_driver structure, so make the snd_soc_dai_ops structure const to allow the compiler to put it in read-only memory. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Ye Bin <yebin10(a)huawei.com> --- sound/soc/codecs/lpass-va-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 3d6976a3d9e4..5294c57b2cd4 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -894,7 +894,7 @@ static int va_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) return 0; } -static struct snd_soc_dai_ops va_macro_dai_ops = { +static const struct snd_soc_dai_ops va_macro_dai_ops = { .hw_params = va_macro_hw_params, .get_channel_map = va_macro_get_channel_map, .mute_stream = va_macro_digital_mute,

2 1

[PATCH] ASoC: meson: axg-frddr: set fifo depth according to the period
by Jerome Brunet 07 Apr '21

07 Apr '21

When the period is small, using all the FRDDR fifo depth increases the latency of the playback because the following device won't start pulling data until the fifo reaches the depth set. We can adjust this depth so trim it down for small periods. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-frddr.c | 26 ++++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/meson/axg-frddr.c b/sound/soc/meson/axg-frddr.c index c3ae8ac30745..8ed114de0bdf 100644 --- a/sound/soc/meson/axg-frddr.c +++ b/sound/soc/meson/axg-frddr.c @@ -11,6 +11,7 @@ #include <linux/regmap.h> #include <linux/module.h> #include <linux/of_platform.h> +#include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dai.h> @@ -46,11 +47,28 @@ static int g12a_frddr_dai_prepare(struct snd_pcm_substream *substream, return 0; } +static int axg_frddr_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + unsigned int period, depth, val; + + period = params_period_bytes(params); + + /* Trim the FIFO depth if the period is small to improve latency */ + depth = min(period, fifo->depth); + val = (depth / AXG_FIFO_BURST) - 1; + regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK, + CTRL1_FRDDR_DEPTH(val)); + + return 0; +} + static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); - unsigned int val; int ret; /* Enable pclk to access registers and clock the fifo ip */ @@ -61,11 +79,6 @@ static int axg_frddr_dai_startup(struct snd_pcm_substream *substream, /* Apply single buffer mode to the interface */ regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_FRDDR_PP_MODE, 0); - /* Use all fifo depth */ - val = (fifo->depth / AXG_FIFO_BURST) - 1; - regmap_update_bits(fifo->map, FIFO_CTRL1, CTRL1_FRDDR_DEPTH_MASK, - CTRL1_FRDDR_DEPTH(val)); - return 0; } @@ -84,6 +97,7 @@ static int axg_frddr_pcm_new(struct snd_soc_pcm_runtime *rtd, } static const struct snd_soc_dai_ops axg_frddr_ops = { + .hw_params = axg_frddr_dai_hw_params, .startup = axg_frddr_dai_startup, .shutdown = axg_frddr_dai_shutdown, }; -- 2.30.2

2 1