September 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/5] ASoC: Fixes for MT8195 SOF support
by AngeloGioacchino Del Regno 07 Sep '22

07 Sep '22

This series fixes Sound Open Firmware support for MT8195 by making sure that the sound card driver is actually able to probe and IPC can finally happen. It is now possible to get DSP support for audio. Tested on MT8195 Tomato - Acer Chromebook Spin 513 CP513-2H (Pipewire). AngeloGioacchino Del Regno (5): ASoC: mediatek: mt8195-mt6359: Properly register sound card for SOF ASoC: SOF: mediatek: mt8195: Import namespace SND_SOC_SOF_MTK_COMMON ASoC: SOF: mediatek: mt8195: Add mailbox generic callbacks for IPC ASoC: SOF: mediatek: mt8195: Add generic pcm_{open,close} callbacks ASoC: SOF: mediatek: mt8195: Add devicetree support to select topologies sound/soc/mediatek/mt8195/mt8195-mt6359.c | 6 +++++ sound/soc/sof/mediatek/mt8195/mt8195.c | 32 ++++++++++++++++------- 2 files changed, 29 insertions(+), 9 deletions(-) -- 2.37.2

2 6

[PATCH] ASoC: ts3a227e: add parameters to control debounce times
by Astrid Rost 07 Sep '22

07 Sep '22

Add devicetree parameters to control the insertion, release and press debounce times. Signed-off-by: Astrid Rost <astrid.rost(a)axis.com> --- sound/soc/codecs/ts3a227e.c | 48 +++++++++++++++++++++++++++++++------ 1 file changed, 41 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/ts3a227e.c b/sound/soc/codecs/ts3a227e.c index d8ab0810fceb..fea274a0a91b 100644 --- a/sound/soc/codecs/ts3a227e.c +++ b/sound/soc/codecs/ts3a227e.c @@ -78,12 +78,20 @@ static const int ts3a227e_buttons[] = { #define ADC_COMPLETE_INT_DISABLE 0x04 #define INTB_DISABLE 0x08 +/* TS3A227E_REG_SETTING_1 0x4 */ +#define DEBOUNCE_INSERTION_SETTING_SFT (0) +#define DEBOUNCE_INSERTION_SETTING_MASK (0x7 << DEBOUNCE_PRESS_SETTING_SFT) + /* TS3A227E_REG_SETTING_2 0x05 */ #define KP_ENABLE 0x04 /* TS3A227E_REG_SETTING_3 0x06 */ -#define MICBIAS_SETTING_SFT (3) +#define MICBIAS_SETTING_SFT 3 #define MICBIAS_SETTING_MASK (0x7 << MICBIAS_SETTING_SFT) +#define DEBOUNCE_RELEASE_SETTING_SFT 2 +#define DEBOUNCE_RELEASE_SETTING_MASK (0x1 << DEBOUNCE_RELEASE_SETTING_SFT) +#define DEBOUNCE_PRESS_SETTING_SFT 0 +#define DEBOUNCE_PRESS_SETTING_MASK (0x3 << DEBOUNCE_PRESS_SETTING_SFT) /* TS3A227E_REG_ACCESSORY_STATUS 0x0b */ #define TYPE_3_POLE 0x01 @@ -136,7 +144,7 @@ static bool ts3a227e_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { case TS3A227E_REG_INTERRUPT ... TS3A227E_REG_INTERRUPT_DISABLE: - case TS3A227E_REG_SETTING_2: + case TS3A227E_REG_SETTING_1 ... TS3A227E_REG_SETTING_2: case TS3A227E_REG_SWITCH_STATUS_1 ... TS3A227E_REG_ADC_OUTPUT: return true; default: @@ -269,16 +277,42 @@ static const struct regmap_config ts3a227e_regmap_config = { static int ts3a227e_parse_device_property(struct ts3a227e *ts3a227e, struct device *dev) { - u32 micbias; + u32 value; + u32 setting3_value = 0; + u32 setting3_mask = 0; int err; - err = device_property_read_u32(dev, "ti,micbias", &micbias); + err = device_property_read_u32(dev, "ti,micbias", &value); if (!err) { - regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_3, - MICBIAS_SETTING_MASK, - (micbias & 0x07) << MICBIAS_SETTING_SFT); + setting3_mask = MICBIAS_SETTING_MASK; + setting3_value = + (value << MICBIAS_SETTING_SFT) & MICBIAS_SETTING_MASK; + } + + err = device_property_read_u32(dev, "ti,debounce-release", &value); + if (!err) { + setting3_mask |= DEBOUNCE_RELEASE_SETTING_MASK; + setting3_value |= (value << DEBOUNCE_RELEASE_SETTING_SFT) & + DEBOUNCE_RELEASE_SETTING_MASK; } + err = device_property_read_u32(dev, "ti,debounce-press", &value); + if (!err) { + setting3_mask |= DEBOUNCE_PRESS_SETTING_MASK; + setting3_value |= (value << DEBOUNCE_PRESS_SETTING_SFT) & + DEBOUNCE_PRESS_SETTING_MASK; + } + + if (setting3_mask) + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_3, + setting3_mask, setting3_value); + + err = device_property_read_u32(dev, "ti,debounce-insertion", &value); + if (!err) + regmap_update_bits(ts3a227e->regmap, TS3A227E_REG_SETTING_1, + DEBOUNCE_INSERTION_SETTING_MASK, (value << DEBOUNCE_INSERTION_SETTING_SFT) + & DEBOUNCE_INSERTION_SETTING_MASK); + return 0; } -- 2.20.1

1 0

[PATCH] ASoC: simple-card-utils: switch to using gpiod API
by Dmitry Torokhov 07 Sep '22

07 Sep '22

This patch switches the driver away from legacy gpio/of_gpio API to gpiod API, and removes use of of_get_named_gpio_flags() which I want to make private to gpiolib. Signed-off-by: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> --- sound/soc/generic/simple-card-utils.c | 28 ++++++++++++++------------- 1 file changed, 15 insertions(+), 13 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 1b201dd09259..bef16833c487 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -9,7 +9,6 @@ #include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> -#include <linux/of_gpio.h> #include <linux/of_graph.h> #include <sound/jack.h> #include <sound/pcm_params.h> @@ -729,12 +728,12 @@ int asoc_simple_init_jack(struct snd_soc_card *card, char *pin) { struct device *dev = card->dev; - enum of_gpio_flags flags; + struct gpio_desc *desc; char prop[128]; char *pin_name; char *gpio_name; int mask; - int det; + int error; if (!prefix) prefix = ""; @@ -742,36 +741,39 @@ int asoc_simple_init_jack(struct snd_soc_card *card, sjack->gpio.gpio = -ENOENT; if (is_hp) { - snprintf(prop, sizeof(prop), "%shp-det-gpio", prefix); + snprintf(prop, sizeof(prop), "%shp-det", prefix); pin_name = pin ? pin : "Headphones"; gpio_name = "Headphone detection"; mask = SND_JACK_HEADPHONE; } else { - snprintf(prop, sizeof(prop), "%smic-det-gpio", prefix); + snprintf(prop, sizeof(prop), "%smic-det", prefix); pin_name = pin ? pin : "Mic Jack"; gpio_name = "Mic detection"; mask = SND_JACK_MICROPHONE; } - det = of_get_named_gpio_flags(dev->of_node, prop, 0, &flags); - if (det == -EPROBE_DEFER) - return -EPROBE_DEFER; + desc = gpiod_get_optional(dev, prop, GPIOD_IN); + error = PTR_ERR_OR_ZERO(desc); + if (error) + return error; + + if (desc) { + error = gpiod_set_consumer_name(desc, gpio_name); + if (error) + return error; - if (gpio_is_valid(det)) { sjack->pin.pin = pin_name; sjack->pin.mask = mask; sjack->gpio.name = gpio_name; sjack->gpio.report = mask; - sjack->gpio.gpio = det; - sjack->gpio.invert = !!(flags & OF_GPIO_ACTIVE_LOW); + sjack->gpio.desc = desc; sjack->gpio.debounce_time = 150; snd_soc_card_jack_new_pins(card, pin_name, mask, &sjack->jack, &sjack->pin, 1); - snd_soc_jack_add_gpios(&sjack->jack, 1, - &sjack->gpio); + snd_soc_jack_add_gpios(&sjack->jack, 1, &sjack->gpio); } return 0; -- 2.37.2.789.g6183377224-goog -- Dmitry

3 2

[PATCH] ASoC: dt-bindings: qcom,q6core: remove binding
by Krzysztof Kozlowski 07 Sep '22

07 Sep '22

qcom,q6core is already described in soc/qcom/qcom,apr.yaml. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../devicetree/bindings/sound/qcom,q6core.txt | 21 ------------------- 1 file changed, 21 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6core.txt diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.txt b/Documentation/devicetree/bindings/sound/qcom,q6core.txt deleted file mode 100644 index 5cd4cc9b1fde..000000000000 --- a/Documentation/devicetree/bindings/sound/qcom,q6core.txt +++ /dev/null @@ -1,21 +0,0 @@ -Qualcomm ADSP Core service binding - -Q6CORE is one of the APR audio service on Q6DSP. -Please refer to qcom,apr.txt for details of the common apr service bindings -used by the apr service device. - -- but must contain the following property: - -- compatible: - Usage: required - Value type: <stringlist> - Definition: must be "qcom,q6core-v<MAJOR-NUMBER>.<MINOR-NUMBER>". - Or "qcom,q6core" where the version number can be queried - from DSP. - example "qcom,q6core-v2.0" - -= EXAMPLE -apr-service@3 { - compatible = "qcom,q6core"; - reg = <APR_SVC_ADSP_CORE>; -}; -- 2.34.1

2 1

[PATCH 0/4] ASoC: soc-dapm.c: random cleanup
by Kuninori Morimoto 07 Sep '22

07 Sep '22

Hi Mark These are cleanup patches for soc-dapm.c. Each patches are not related, very random cleanup. Kuninori Morimoto (4): ASoC: soc-dapm.c: don't use kzalloc() for param on snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: don't use WARN_ON() at snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: tidyup snd_soc_dai_link_event_pre_pmu() ASoC: soc-dapm.c: fixup snd_soc_dapm_new_control_unlocked() error handling sound/soc/soc-dapm.c | 77 +++++++++++++++++++------------------------- 1 file changed, 33 insertions(+), 44 deletions(-) -- 2.25.1

2 8

utils: cplay: Add support for ID3v2 tag skip
by GitHub pull_request - edited 07 Sep '22

07 Sep '22

alsa-project/tinycompress pull request #15 was edited from LaurentiuM1234: This patch will allow playing MP3 files containing ID3v2 tag at the beginning of the file. This is done simply by parsing the ID3v2 header, extracting the header size and seeking to that position (+10 bytes from the header size which is not included in the 'header size' field of the ID3v2 header) Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/15 Patch URL : https://github.com/alsa-project/tinycompress/pull/15.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

utils: cplay: Add support for ID3v2 header skip
by GitHub pull_request - opened 07 Sep '22

07 Sep '22

alsa-project/tinycompress pull request #15 was opened from LaurentiuM1234: This patch will allow playing MP3 files containing ID3v2 tag at the beginning of the file. This is done simply by parsing the ID3v2 header, extracting the header size and seeking to that position (+10 bytes from the header size which is not included in the 'header size' field of the ID3v2 header) Signed-off-by: Laurentiu Mihalcea <laurentiu.mihalcea(a)nxp.com> Request URL : https://github.com/alsa-project/tinycompress/pull/15 Patch URL : https://github.com/alsa-project/tinycompress/pull/15.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH] ALSA: usb-audio: Clear fixed clock rate at closing EP
by Takashi Iwai 07 Sep '22

07 Sep '22

The recent commit c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") tries to manage the clock rate shared by several endpoints. This was intended for avoiding the unmatched rate by a different endpoint, but unfortunately, it introduced a regression for PulseAudio and pipewire, too; those applications try to probe the multiple possible rates (44.1k and 48kHz) and setting up the normal rate fails but only the last rate is applied. The cause is that the last sample rate is still left to the clock reference even after closing the endpoint, and this value is still used at the next open. It happens only when applications set up via PCM prepare but don't start/stop the stream; the rate is reset when the stream is stopped, but it's not cleared at close. This patch addresses the issue above, simply by clearing the rate set in the clock reference at the last close of each endpoint. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Reported-by: Jason A. Donenfeld <Jason(a)zx2c4.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/ Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index a42f2ce19455..8c8f9a851f89 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -925,6 +925,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, endpoint_set_interface(chip, ep, false); if (!--ep->opened) { + if (ep->clock_ref && !atomic_read(&ep->clock_ref->locked)) + ep->clock_ref->rate = 0; ep->iface = 0; ep->altsetting = 0; ep->cur_audiofmt = NULL; -- 2.35.3

2 1

[PATCH] ALSA: usb-audio: Don't refcount multiple accesses on the single clock
by Jason A. Donenfeld 07 Sep '22

07 Sep '22

This reverts commit 03a8b0df757f1beb21ba1626e23ca7412e48b525. This reverts commit c11117b634f4f832c4420d3cf41c44227f140ce1. Pipewire and PulseAudio start devices with 44.1khz before changing them to 48khz (or something different). By locking the rate, daemons are unable to enumerate possible rates, and so they never change them to a more optimal rate. This revert patch should allow 48khz audio again. Fixes: c11117b634f4 ("ALSA: usb-audio: Refcount multiple accesses on the single clock") Cc: stable(a)vger.kernel.org Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Niklāvs Koļesņikovs <89q1r14hd(a)relay.firefox.com> Cc: Wim Taymans <wtaymans(a)redhat.com> Link: https://lore.kernel.org/all/YxXIWv8dYmg1tnXP@zx2c4.com/ Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2620 Signed-off-by: Jason A. Donenfeld <Jason(a)zx2c4.com> --- sound/usb/card.c | 1 - sound/usb/card.h | 3 +- sound/usb/endpoint.c | 90 +++++--------------------------------------- sound/usb/usbaudio.h | 1 - 4 files changed, 10 insertions(+), 85 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index d356743de2ff..12bca2e85505 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -643,7 +643,6 @@ static int snd_usb_audio_create(struct usb_interface *intf, INIT_LIST_HEAD(&chip->pcm_list); INIT_LIST_HEAD(&chip->ep_list); INIT_LIST_HEAD(&chip->iface_ref_list); - INIT_LIST_HEAD(&chip->clock_ref_list); INIT_LIST_HEAD(&chip->midi_list); INIT_LIST_HEAD(&chip->mixer_list); diff --git a/sound/usb/card.h b/sound/usb/card.h index ca75f2206170..87f042d06ce0 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -44,7 +44,6 @@ struct audioformat { struct snd_usb_substream; struct snd_usb_iface_ref; -struct snd_usb_clock_ref; struct snd_usb_endpoint; struct snd_usb_power_domain; @@ -63,7 +62,6 @@ struct snd_urb_ctx { struct snd_usb_endpoint { struct snd_usb_audio *chip; struct snd_usb_iface_ref *iface_ref; - struct snd_usb_clock_ref *clock_ref; int opened; /* open refcount; protect with chip->mutex */ atomic_t running; /* running status */ @@ -140,6 +138,7 @@ struct snd_usb_endpoint { unsigned int cur_period_frames; unsigned int cur_period_bytes; unsigned int cur_buffer_periods; + unsigned char cur_clock; spinlock_t lock; struct list_head list; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0d7b73bf7945..0f5e9224a799 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -35,14 +35,6 @@ struct snd_usb_iface_ref { struct list_head list; }; -/* clock refcounting */ -struct snd_usb_clock_ref { - unsigned char clock; - atomic_t locked; - int rate; - struct list_head list; -}; - /* * snd_usb_endpoint is a model that abstracts everything related to an * USB endpoint and its streaming. @@ -599,25 +591,6 @@ iface_ref_find(struct snd_usb_audio *chip, int iface) return ip; } -/* Similarly, a refcount object for clock */ -static struct snd_usb_clock_ref * -clock_ref_find(struct snd_usb_audio *chip, int clock) -{ - struct snd_usb_clock_ref *ref; - - list_for_each_entry(ref, &chip->clock_ref_list, list) - if (ref->clock == clock) - return ref; - - ref = kzalloc(sizeof(*ref), GFP_KERNEL); - if (!ref) - return NULL; - ref->clock = clock; - atomic_set(&ref->locked, 0); - list_add_tail(&ref->list, &chip->clock_ref_list); - return ref; -} - /* * Get the existing endpoint object corresponding EP * Returns NULL if not present. @@ -795,14 +768,6 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, goto unlock; } - if (fp->protocol != UAC_VERSION_1) { - ep->clock_ref = clock_ref_find(chip, fp->clock); - if (!ep->clock_ref) { - ep = NULL; - goto unlock; - } - } - ep->cur_audiofmt = fp; ep->cur_channels = fp->channels; ep->cur_rate = params_rate(params); @@ -812,6 +777,7 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, ep->cur_period_frames = params_period_size(params); ep->cur_period_bytes = ep->cur_period_frames * ep->cur_frame_bytes; ep->cur_buffer_periods = params_periods(params); + ep->cur_clock = fp->clock; if (ep->type == SND_USB_ENDPOINT_TYPE_SYNC) endpoint_set_syncinterval(chip, ep); @@ -928,8 +894,8 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, ep->altsetting = 0; ep->cur_audiofmt = NULL; ep->cur_rate = 0; + ep->cur_clock = 0; ep->iface_ref = NULL; - ep->clock_ref = NULL; usb_audio_dbg(chip, "EP 0x%x closed\n", ep->ep_num); } mutex_unlock(&chip->mutex); @@ -941,8 +907,6 @@ void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep) ep->need_setup = true; if (ep->iface_ref) ep->iface_ref->need_setup = true; - if (ep->clock_ref) - ep->clock_ref->rate = 0; } /* @@ -1350,33 +1314,6 @@ static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, return 0; } -static int init_sample_rate(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) -{ - struct snd_usb_clock_ref *clock = ep->clock_ref; - int err; - - if (clock) { - if (atomic_read(&clock->locked)) - return 0; - if (clock->rate == ep->cur_rate) - return 0; - if (clock->rate && clock->rate != ep->cur_rate) { - usb_audio_dbg(chip, "Mismatched sample rate %d vs %d for EP 0x%x\n", - clock->rate, ep->cur_rate, ep->ep_num); - return -EINVAL; - } - } - - err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); - if (err < 0) - return err; - - if (clock) - clock->rate = ep->cur_rate; - return 0; -} - /* * snd_usb_endpoint_configure: Configure the endpoint * @@ -1406,7 +1343,8 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, * to update at each EP configuration */ if (ep->cur_audiofmt->protocol == UAC_VERSION_1) { - err = init_sample_rate(chip, ep); + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, + ep->cur_rate); if (err < 0) goto unlock; } @@ -1436,7 +1374,7 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = init_sample_rate(chip, ep); + err = snd_usb_init_sample_rate(chip, ep->cur_audiofmt, ep->cur_rate); if (err < 0) goto unlock; @@ -1469,15 +1407,15 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, /* get the current rate set to the given clock by any endpoint */ int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock) { - struct snd_usb_clock_ref *ref; + struct snd_usb_endpoint *ep; int rate = 0; if (!clock) return 0; mutex_lock(&chip->mutex); - list_for_each_entry(ref, &chip->clock_ref_list, list) { - if (ref->clock == clock) { - rate = ref->rate; + list_for_each_entry(ep, &chip->ep_list, list) { + if (ep->cur_clock == clock && ep->cur_rate) { + rate = ep->cur_rate; break; } } @@ -1518,9 +1456,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) if (atomic_inc_return(&ep->running) != 1) return 0; - if (ep->clock_ref) - atomic_inc(&ep->clock_ref->locked); - ep->active_mask = 0; ep->unlink_mask = 0; ep->phase = 0; @@ -1630,9 +1565,6 @@ void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending) if (ep->sync_source) WRITE_ONCE(ep->sync_source->sync_sink, NULL); stop_urbs(ep, false, keep_pending); - if (ep->clock_ref) - if (!atomic_dec_return(&ep->clock_ref->locked)) - ep->clock_ref->rate = 0; } } @@ -1659,16 +1591,12 @@ void snd_usb_endpoint_free_all(struct snd_usb_audio *chip) { struct snd_usb_endpoint *ep, *en; struct snd_usb_iface_ref *ip, *in; - struct snd_usb_clock_ref *cp, *cn; list_for_each_entry_safe(ep, en, &chip->ep_list, list) kfree(ep); list_for_each_entry_safe(ip, in, &chip->iface_ref_list, list) kfree(ip); - - list_for_each_entry_safe(cp, cn, &chip->clock_ref_list, list) - kfree(cp); } /* diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ffbb4b0d09a0..044cd7ab27cb 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -45,7 +45,6 @@ struct snd_usb_audio { struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ struct list_head iface_ref_list; /* list of interface refcounts */ - struct list_head clock_ref_list; /* list of clock refcounts */ int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ -- 2.37.3

2 7

[PATCH] ALSA: hda/ca0132: fix potential memory leak in dspxfr_image()
by Jianglei Nie 07 Sep '22

07 Sep '22

dspxfr_image() allocates DSP ports for the download stream with dsp_allocate_ports_format(). When gets some error, the allocated DSP ports are not released, which will lead to a memory leak. We can fix it by releasing DSP ports with dsp_free_ports() when getting some error. Signed-off-by: Jianglei Nie <niejianglei2021(a)163.com> --- sound/pci/hda/patch_ca0132.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 208933792787..6b8f45e14075 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -3455,6 +3455,7 @@ static int dspxfr_image(struct hda_codec *codec, &port_map_mask); if (status < 0) { codec_dbg(codec, "alloc ports fail\n"); + dsp_free_ports(codec); goto exit; } @@ -3463,6 +3464,7 @@ static int dspxfr_image(struct hda_codec *codec, WIDGET_CHIP_CTRL, stream_id, 0, &response); if (status < 0) { codec_dbg(codec, "set stream chan fail\n"); + dsp_free_ports(codec); goto exit; } @@ -3470,6 +3472,7 @@ static int dspxfr_image(struct hda_codec *codec, if (!is_valid(fls_data)) { codec_dbg(codec, "FLS check fail\n"); status = -EINVAL; + dsp_free_ports(codec); goto exit; } status = dspxfr_one_seg(codec, fls_data, reloc, -- 2.25.1

2 1