June 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: max98390: Fix incorrect printf qualifier
by Takashi Iwai 04 Jun '20

04 Jun '20

This patch addresses a compile warning: sound/soc/codecs/max98390.c:781:3: warning: format ‘%ld’ expects argument of type ‘long int’, but argument 4 has type ‘size_t {aka const unsigned int}’ [-Wformat=] Fixes: a6e3f4f34cdb ("ASoC: max98390: Added Amplifier Driver") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index b9ce44dda886..be7cd0aeb6a6 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -778,7 +778,7 @@ static int max98390_dsm_init(struct snd_soc_component *component) } dev_dbg(component->dev, - "max98390: param fw size %ld\n", + "max98390: param fw size %zd\n", fw->size); dsm_param = (char *)fw->data; dsm_param += MAX98390_DSM_PAYLOAD_OFFSET; -- 2.16.4

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[PATCH] ASoC: fsl-asoc-card: Defer probe when fail to find codec device
by Shengjiu Wang 04 Jun '20

04 Jun '20

Defer probe when fail to find codec device, because the codec device maybe probed later than machine driver. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index cf4feb835743..00be73900888 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -581,7 +581,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { dev_err(&pdev->dev, "failed to find codec device\n"); - ret = -EINVAL; + ret = -EPROBE_DEFER; goto asrc_fail; } -- 2.21.0

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Re: [PATCH] ASoC: max98390: fix build warning detected by -Wformat
by Mark Brown 04 Jun '20

04 Jun '20

On Thu, Jun 04, 2020 at 04:49:34PM +0800, Xiaoming Ni wrote: > Fix build warning: > sound/soc/codecs/max98390.c:781:3: warning: format '%ld' expects > argument of type 'long int', but argument 4 has type 'size_t {aka > const unsigned int}' [-Wformat=] Thanks but this was already fixed.

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alsa-gobject: ctl: fulfill documentation for ALSACtl
by GitHub pull_request - opened 04 Jun '20

04 Jun '20

alsa-project/alsa-gobject pull request #22 was opened from takaswie: This patch fulfills alsactl documentation for GObject-derived objects and boxed objects. Request URL : https://github.com/alsa-project/alsa-gobject/pull/22 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/22.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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alsa-project alsa-ucm : Audio endpoint is not switching when external mic connected
by Mukunda, Vijendar 04 Jun '20

04 Jun '20

We recently up streamed ACP PDM driver for Renoir APU. Our ACP IP has an internal PDM Decoder block where DMIC is directly connected. There are no KControls defined for PDM Driver. Our platform supports on board DMIC and South bridge Azalia (HD Audio ) endpoint . By default, DMIC endpoint is used as audio input device. When Headset Mic jack connected, it's not switching to External Mic. How does audio endpoint will switch in gnome when external mic is connected? >From our understanding, UCM conf file for HDA driver is also required for configuring mixer controls for HD Audio capture device. Similarly when External Mic is removed , it has to switch to internal dmic . Need help in resolving this issue. Thanks, Vijendar

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UCM file required for Multiple sound card scenario for Renoir APU
by GitHub issues - opened 04 Jun '20

04 Jun '20

alsa-project/alsa-ucm-conf issue #30 was opened from VijendarReddy: We recently upstreamed PDM Driver for Renoir APU. Renoir APU platform has Onboard DMIC + Southbridge AZ (HD Audio Endpoint) support. When External Mic connected, Device switching should happen from Internal DMIC to external Mic. Need UCM file to support Multiple sound cards for Renoir APU. Alsa info link: https://pastebin.ubuntu.com/p/4kprhDZYbg/ Created Ticket based on discussion on the below-mentioned thread. https://www.alsa-project.org/pipermail/alsa-devel/2020-June/168527.html Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/30 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 00/24] ASoC: soc-component: collect component functions
by Kuninori Morimoto 04 Jun '20

04 Jun '20

Hi Mark We have soc-component.c now, but still many component related functions are implemented many place. This patch-set collect these into soc-component.c. Kuninori Morimoto (24): ASoC: soc-component: add soc_component_pin() and share code ASoC: soc-component: move snd_soc_component_xxx_regmap() to soc-component ASoC: soc-component: move snd_soc_component_initialize() to soc-component.c ASoC: soc-component: add soc_component_err() ASoC: soc-component: add snd_soc_pcm_component_prepare() ASoC: soc-component: add snd_soc_pcm_component_hw_params() ASoC: soc-component: add snd_soc_pcm_component_hw_free() ASoC: soc-component: add snd_soc_pcm_component_trigger() ASoC: soc-component: add snd_soc_component_compr_open() ASoC: soc-component: add snd_soc_component_compr_free() ASoC: soc-component: add snd_soc_component_compr_trigger() ASoC: soc-component: add snd_soc_component_compr_set_params() ASoC: soc-component: add snd_soc_component_compr_get_params() ASoC: soc-component: add snd_soc_component_compr_get_caps() ASoC: soc-component: add snd_soc_component_compr_get_codec_caps() ASoC: soc-component: add snd_soc_component_compr_ack() ASoC: soc-component: add snd_soc_component_compr_pointer() ASoC: soc-component: add snd_soc_component_compr_copy() ASoC: soc-component: add snd_soc_component_compr_set_metadata() ASoC: soc-component: add snd_soc_component_compr_get_metadata() ASoC: soc-component: add snd_soc_component_init() ASoC: soc-component: merge soc-io.c into soc-component.c ASoC: soc-component: merge soc_pcm_trigger_start/stop() ASoC: soc-component: tidyup Copyright include/sound/soc-component.h | 51 +- sound/soc/Makefile | 2 +- sound/soc/soc-component.c | 912 ++++++++++++++++++++++++++-------- sound/soc/soc-compress.c | 223 ++------- sound/soc/soc-core.c | 102 +--- sound/soc/soc-io.c | 202 -------- sound/soc/soc-pcm.c | 115 ++--- 7 files changed, 830 insertions(+), 777 deletions(-) delete mode 100644 sound/soc/soc-io.c -- 2.17.1

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[PATCH 1/2] ALSA: usb-audio: Fix inconsistent card PM state after resume
by Takashi Iwai 04 Jun '20

04 Jun '20

When a USB-audio interface gets runtime-suspended via auto-pm feature, the driver suspends all functionality and increment chip->num_suspended_intf. Later on, when the system gets suspended to S3, the driver increments chip->num_suspended_intf again, skips the device changes, and sets the card power state to SNDRV_CTL_POWER_D3hot. In return, when the system gets resumed from S3, the resume callback decrements chip->num_suspended_intf. Since this refcount is still not zero (it's been runtime-suspended), the whole resume is skipped. But there is a small pitfall here. The problem is that the driver doesn't restore the card power state after this resume call, leaving it as SNDRV_CTL_POWER_D3hot. So, even after the system resume finishes, the card instance still appears as if it were system-suspended, and this confuses many ioctl accesses that are blocked unexpectedly. In details, we have two issues behind the scene: one is that the card power state is changed only when the refcount becomes zero, and another is that the prior auto-suspend check is kept in a boolean flag. Although the latter problem is almost negligible since the auto-pm feature is imposed only on the primary interface, but this can be a potential problem on the devices with multiple interfaces. This patch addresses those issues by the following: - Replace chip->autosuspended boolean flag with chip->system_suspend counter - At the first system-suspend, chip->num_suspended_intf is recorded to chip->system_suspend - At system-resume, the card power state is restored when the chip->num_suspended_intf refcount reaches to chip->system_suspend, i.e. the state returns to the auto-suspended Also, the patch fixes yet another hidden problem by the code refactoring along with the fixes above: namely, when some resume procedure failed, the driver left chip->num_suspended_intf that was already decreased, and it might lead to the refcount unbalance. In the new code, the refcount decrement is done after the whole resume procedure, and the problem is avoided as well. Fixes: 0662292aec05 ("ALSA: usb-audio: Handle normal and auto-suspend equally") Reported-and-tested-by: Macpaul Lin <macpaul.lin(a)mediatek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 19 ++++++++++++------- sound/usb/usbaudio.h | 2 +- 2 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index fd6fd1726ea0..359f7a04be1c 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -843,9 +843,6 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) if (chip == (void *)-1L) return 0; - chip->autosuspended = !!PMSG_IS_AUTO(message); - if (!chip->autosuspended) - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); if (!chip->num_suspended_intf++) { list_for_each_entry(as, &chip->pcm_list, list) { snd_usb_pcm_suspend(as); @@ -858,6 +855,11 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) snd_usb_mixer_suspend(mixer); } + if (!PMSG_IS_AUTO(message) && !chip->system_suspend) { + snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + chip->system_suspend = chip->num_suspended_intf; + } + return 0; } @@ -871,10 +873,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) if (chip == (void *)-1L) return 0; - if (--chip->num_suspended_intf) - return 0; atomic_inc(&chip->active); /* avoid autopm */ + if (chip->num_suspended_intf > 1) + goto out; list_for_each_entry(as, &chip->pcm_list, list) { err = snd_usb_pcm_resume(as); @@ -896,9 +898,12 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) snd_usbmidi_resume(p); } - if (!chip->autosuspended) + out: + if (chip->num_suspended_intf == chip->system_suspend) { snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); - chip->autosuspended = 0; + chip->system_suspend = 0; + } + chip->num_suspended_intf--; err_out: atomic_dec(&chip->active); /* allow autopm after this point */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1c892c7f14d7..e0ebfb25fbd5 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -26,7 +26,7 @@ struct snd_usb_audio { struct usb_interface *pm_intf; u32 usb_id; struct mutex mutex; - unsigned int autosuspended:1; + unsigned int system_suspend; atomic_t active; atomic_t shutdown; atomic_t usage_count; -- 2.25.0

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[PATCH] ASoC: max98390: Fix potential crash during param fw loading
by Steve Lee 04 Jun '20

04 Jun '20

malformed firmware file can cause out-of-bound access and crash during dsm_param bin loading. - add MIN/MAX param size to avoid out-of-bound access. - read start addr and size of param and check bound. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 23 ++++++++++++++++++----- sound/soc/codecs/max98390.h | 3 ++- 2 files changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index be7cd0aeb6a6..8c7db568fe64 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -754,6 +754,7 @@ static struct snd_soc_dai_driver max98390_dai[] = { static int max98390_dsm_init(struct snd_soc_component *component) { int ret; + int param_size, param_start_addr; char filename[128]; const char *vendor, *product; struct max98390_priv *max98390 = @@ -780,14 +781,27 @@ static int max98390_dsm_init(struct snd_soc_component *component) dev_dbg(component->dev, "max98390: param fw size %zd\n", fw->size); + if (fw->size < MAX98390_DSM_PARAM_MIN_SIZE) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } dsm_param = (char *)fw->data; + param_start_addr = (dsm_param[0] & 0xff) | (dsm_param[1] & 0xff) << 8; + param_size = (dsm_param[2] & 0xff) | (dsm_param[3] & 0xff) << 8; + if (param_size > MAX98390_DSM_PARAM_MAX_SIZE || + param_start_addr < DSM_STBASS_HPF_B0_BYTE0) { + dev_err(component->dev, + "param fw is invalid.\n"); + goto err_alloc; + } dsm_param += MAX98390_DSM_PAYLOAD_OFFSET; - regmap_bulk_write(max98390->regmap, DSM_EQ_BQ1_B0_BYTE0, - dsm_param, - fw->size - MAX98390_DSM_PAYLOAD_OFFSET); - release_firmware(fw); + regmap_bulk_write(max98390->regmap, param_start_addr, + dsm_param, param_size); regmap_write(max98390->regmap, MAX98390_R23E1_DSP_GLOBAL_EN, 0x01); +err_alloc: + release_firmware(fw); err: return ret; } @@ -847,7 +861,6 @@ static int max98390_probe(struct snd_soc_component *component) /* Dsm Setting */ regmap_write(max98390->regmap, DSM_VOL_CTRL, 0x94); - regmap_write(max98390->regmap, DSMIG_EN, 0x19); regmap_write(max98390->regmap, MAX98390_R203A_AMP_EN, 0x80); if (max98390->ref_rdc_value) { regmap_write(max98390->regmap, DSM_TPROT_RECIP_RDC_ROOM_BYTE0, diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index f59cb114d957..5f444e7779b0 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -650,7 +650,8 @@ /* DSM register offset */ #define MAX98390_DSM_PAYLOAD_OFFSET 16 -#define MAX98390_DSM_PAYLOAD_OFFSET_2 495 +#define MAX98390_DSM_PARAM_MAX_SIZE 770 +#define MAX98390_DSM_PARAM_MIN_SIZE 670 struct max98390_priv { struct regmap *regmap; -- 2.17.1

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[RESEND PATCH v1 0/6] Enable Greybus Audio codec driver
by Vaibhav Agarwal 03 Jun '20

03 Jun '20

[REQUEST] This patch series intends to "Enable Greybus Audio codec driver" existing in the staging tree. I have shared the original patch series with Greybus-Dev mailing list and as per the suggestion from Alexandre, I'm also interested to push Greybus Audio to sound soc tree. Thus, now I'm resending it to ASoc maintainers for review. Following is the top level SW design for GB Codec driver and GB Audio modules. +--------------+ +-------------->+GBA Module 1 | | +--------------+ +-----------------------+ | | | | | Greybus | +--------------+ | SND SOC | Audio +-------------->+GBA Module 2 | | Framework | Codec | +--------------+ | | Driver | | | | +-----------------------+ +--------------+ +-------------->+GBA Module N | +--------------+ Patch 5 contains the changes to provide helper APIs to link DAPM DAI widgets for the module added and remove/free component controls for the module removed dynamically. Eventually, I propose to extend snd_soc_xxx APIs for this purpose. Kindly share your inputs. [COVER LETTER] The existing GB Audio codec driver is dependent on MSM8994 Audio driver. During the development stage, this depdency was configured due to various changes involved in MSM Audio driver to enable addtional codec card and some of the changes proposed in mainline ASoC framework. However, these are not the real dependencies and some of them can be easily removed. The folowing patch series includes the changes to resolve unnecessary depedencies and make the codec driver functional with the latest kernel. Patch 1,2: Incudes jack framework related changes. Patch 3,4,5: Resolves compilation error observed with the latest kernel and also provides helper APIs required to allow dynamic addition/removal of modules. Patch 6: Finally provides config options and related Makefile changes to enable GB Codec driver. Thanks to Alexandre for raising the headsup [1] and motivating me to provide the necessary changes. [1] https://lore.kernel.org/lkml/20200507212912.599433-1-alexandre.belloni@boot… Vaibhav Agarwal (6): staging: greybus: audio: Update snd_jack FW usage as per new APIs staging: greybus: audio: Maintain jack list within GB Audio module staging: greybus: audio: Resolve compilation errors for GB codec module staging: greybus: audio: Resolve compilation error in topology parser staging: greybus: audio: Add helper APIs for dynamic audio modules staging: greybus: audio: Enable GB codec, audio module compilation. drivers/staging/greybus/Kconfig | 14 +- drivers/staging/greybus/Makefile | 6 +- drivers/staging/greybus/audio_codec.c | 187 +++++++++++++-------- drivers/staging/greybus/audio_codec.h | 12 +- drivers/staging/greybus/audio_helper.c | 197 +++++++++++++++++++++++ drivers/staging/greybus/audio_helper.h | 17 ++ drivers/staging/greybus/audio_module.c | 20 +-- drivers/staging/greybus/audio_topology.c | 130 +++++++-------- 8 files changed, 427 insertions(+), 156 deletions(-) create mode 100644 drivers/staging/greybus/audio_helper.c create mode 100644 drivers/staging/greybus/audio_helper.h base-commit: ae73e7784871ebe2c43da619b4a1e2c9ff81508d -- 2.26.2

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