June 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: SOF: ipc4/intel: Add missing mutex_unlock()
by Jyri Sarha 12 Jun '23

12 Jun '23

There was a missing mutex_unlock() in sof_ipc4_widget_free() use_chain_dma if-branch that caused a static analysis error. The branch should not be used in a normal working configuration and if its used its an indication of a bad topology. Add missing mutex_unlock() and a warning print if the if-branch is taken, and another warning print to a symmetric place in sof_ipc4_widget_setup(). Fixes: ca5ce0caa67fa9 ("ASoC: SOF: ipc4/intel: Add support for chained DMA") Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Link: https://lore.kernel.org/r/202303222050.dCw0fPCW-lkp@intel.com/ Signed-off-by: Jyri Sarha <jyri.sarha(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f1e1aed94da4..12775fcb6b54 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1989,8 +1989,11 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget case snd_soc_dapm_scheduler: pipeline = swidget->private; - if (pipeline->use_chain_dma) + if (pipeline->use_chain_dma) { + dev_warn(sdev->dev, "use_chain_dma set for schduler %s", + swidget->widget->name); return 0; + } dev_dbg(sdev->dev, "pipeline: %d memory pages: %d\n", swidget->pipeline_id, pipeline->mem_usage); @@ -2145,8 +2148,12 @@ static int sof_ipc4_widget_free(struct snd_sof_dev *sdev, struct snd_sof_widget struct sof_ipc4_msg msg = {{ 0 }}; u32 header; - if (pipeline->use_chain_dma) + if (pipeline->use_chain_dma) { + dev_warn(sdev->dev, "use_chain_dma set for schduler %s", + swidget->widget->name); + mutex_unlock(&ipc4_data->pipeline_state_mutex); return 0; + } header = SOF_IPC4_GLB_PIPE_INSTANCE_ID(swidget->instance_id); header |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_GLB_DELETE_PIPELINE); -- 2.34.1

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[PATCH] ALSA: pcm: fix wait_time calculations
by Oswald Buddenhagen 12 Jun '23

12 Jun '23

... in wait_for_avail() and snd_pcm_drain(). t was calculated in seconds, so it would be pretty much always zero, to be subsequently de-facto ignored due to being max(t, 10)'d. And then it (i.e., 10) would be treated as secs, which doesn't seem right. However, fixing it to properly calculate msecs would potentially cause timeouts when using twice the period size for the default timeout (which seems reasonable to me), so instead use the buffer size plus 10 percent to be on the safe side ... but that still seems insufficient, presumably because the hardware typically needs a moment to fire up. To compensate for this, we up the minimal timeout to 100ms, which is still two orders of magnitude less than the bogus minimum. substream->wait_time was also misinterpreted as jiffies, despite being documented as being in msecs. Only the soc/sof driver sets it - to 500, which looks very much like msecs were intended. Speaking of which, shouldn't snd_pcm_drain() also use substream-> wait_time? As a drive-by, make the debug messages on timeout less confusing. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/core/pcm_lib.c | 11 +++++------ sound/core/pcm_native.c | 8 ++++---- 2 files changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 02fd65993e7e..af1eb136feb0 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1878,15 +1878,14 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (substream->wait_time) { wait_time = substream->wait_time; } else { - wait_time = 10; + wait_time = 100; if (runtime->rate) { - long t = runtime->period_size * 2 / - runtime->rate; + long t = runtime->buffer_size * 1100 / runtime->rate; wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); } + wait_time = msecs_to_jiffies(wait_time); } for (;;) { @@ -1934,8 +1933,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, } if (!tout) { pcm_dbg(substream->pcm, - "%s write error (DMA or IRQ trouble?)\n", - is_playback ? "playback" : "capture"); + "%s timeout (DMA or IRQ trouble?)\n", + is_playback ? "playback write" : "capture read"); err = -EIO; break; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 331380c2438b..94185267a7b9 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2159,12 +2159,12 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) tout = MAX_SCHEDULE_TIMEOUT; else { - tout = 10; + tout = 100; if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; + long t = runtime->buffer_size * 1100 / runtime->rate; tout = max(t, tout); } - tout = msecs_to_jiffies(tout * 1000); + tout = msecs_to_jiffies(tout); } tout = schedule_timeout(tout); @@ -2187,7 +2187,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, result = -ESTRPIPE; else { dev_dbg(substream->pcm->card->dev, - "playback drain error (DMA or IRQ trouble?)\n"); + "playback drain timeout (DMA or IRQ trouble?)\n"); snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); result = -EIO; } -- 2.40.0.152.g15d061e6df

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[PATCH v2 1/2] ASoC: dt-bindings: qcom,wsa8840: Add WSA884x family of speakers
by Krzysztof Kozlowski 12 Jun '23

12 Jun '23

Add binding for WSA8840/WSA8845/WSA8845H smart speaker amplifiers used in Qualcomm QRD8550 board with SM8550 SoC. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Correct compatible (sdw version 1 -> 2). Cc: Patrick Lai <quic_plai(a)quicinc.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../bindings/sound/qcom,wsa8840.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml new file mode 100644 index 000000000000..e6723c9e312a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa8840.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8840/WSA8845/WSA8845H smart speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +description: + WSA884X is a family of Qualcomm Aqstic smart speaker amplifiers using + SoundWire digital audio interface. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw20217020400 + + reg: + maxItems: 1 + + powerdown-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vdd-1p8-supply: true + vdd-io-supply: true + +required: + - compatible + - reg + - powerdown-gpios + - '#sound-dai-cells' + - vdd-1p8-supply + - vdd-io-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire-controller { + #address-cells = <2>; + #size-cells = <0>; + + speaker@0,1 { + compatible = "sdw20217020400"; + reg = <0 1>; + pinctrl-names = "default"; + pinctrl-0 = <&spkr_2_sd_n_active>; + powerdown-gpios = <&lpass_tlmm 18 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + vdd-1p8-supply = <&vreg_l15b_1p8>; + vdd-io-supply = <&vreg_l3g_1p2>; + }; + }; -- 2.34.1

2 4

[PATCH] ALSA: usb-audio: Fix broken resume due to UAC3 power state
by Takashi Iwai 12 Jun '23

12 Jun '23

As reported in the bugzilla below, the PM resume of a UAC3 device may fail due to the incomplete power state change, stuck at D1. The reason is that the driver expects the full D0 power state change only at hw_params, while the normal PCM resume procedure doesn't call hw_params. For fixing the bug, we add the same power state update to D0 at the prepare callback, which is certainly called by the resume procedure. Note that, with this change, the power state change in the hw_params becomes almost redundant, since snd_usb_hw_params() doesn't touch the parameters (at least it tires so). But dropping it is still a bit risky (e.g. we have the media-driver binding), so I leave the D0 power state change in snd_usb_hw_params() as is for now. Fixes: a0a4959eb4e9 ("ALSA: usb-audio: Operate UAC3 Power Domains in PCM callbacks") Cc: <stable(a)vger.kernel.org> Link: https://bugzilla.kernel.org/show_bug.cgi?id=217539 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index eec5232f9fb2..08bf535ed163 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -650,6 +650,10 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0); + if (ret < 0) + goto unlock; + again: if (subs->sync_endpoint) { ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); -- 2.35.3

1 0

[PATCH] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Takashi Iwai 12 Jun '23

12 Jun '23

Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 07efb38f58ac..f2940b29595f 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -172,6 +173,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strscpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -322,15 +324,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -340,14 +344,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -372,13 +376,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -393,10 +401,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -415,6 +422,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.35.3

1 0

[PATCH 0/2] ALSA: firewire: use 'GPL' string for module license
by Takashi Sakamoto 12 Jun '23

12 Jun '23

Hi, Nowadays, the use of "GPL" string in MODULE_LICENSE macro seems to obsolete "GPL v2" string. This series includes two patches to use it for modules in ALSA firewire stack contributed by two developers. This change does not pose any issue for modules contributed by Takashi Sakamoto, the author of this series, while it is advisable to have careful consideration for modules contributed by Clemens Ladisch, since the code related to license will be modified by a developer who is not the copyright holder. A commit bf7fbeeae6db ("module: Cure the MODULE_LICENSE "GPL" vs. "GPL v2" bogosity") adds detail documentation about the macro itself and clarifies the difference between "GPL" and "GPL v2". The macro is designed to provide sufficient information to kernel module loader and user space tools. Both strings have the same meanings, but the latter is legacy. In conclusion, any change made to the usage of the macro neither result in significant alterations to license nor copyright. Takashi Sakamoto (2): ALSA: firewire: use 'GPL' string for module license contributed by Takashi Sakamoto ALSA: firewire: use 'GPL' string for module license contributed by Clemens Ladisch sound/firewire/bebob/bebob.c | 2 +- sound/firewire/dice/dice.c | 2 +- sound/firewire/digi00x/digi00x.c | 2 +- sound/firewire/fireface/ff.c | 2 +- sound/firewire/fireworks/fireworks.c | 2 +- sound/firewire/isight.c | 2 +- sound/firewire/lib.c | 2 +- sound/firewire/motu/motu.c | 2 +- sound/firewire/oxfw/oxfw.c | 2 +- sound/firewire/tascam/tascam.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) -- 2.39.2

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[PATCH v2 00/11] ASoC: qcom: audioreach: add compress offload support
by Srinivas Kandagatla 12 Jun '23

12 Jun '23

This patchset adds compressed offload support to Qualcomm audioreach drivers. Currently it supports AAC, MP3 and FALC along with gapless. Tested this on SM8450 and sc7280. thanks, srini Changes since v1: - removed lots of code duplication - moved ALSA patch out of this series. Mohammad Rafi Shaik (4): ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup callback ASoC: q6dsp: q6apm: add end of stream events ASoC: q6dsp: audioreach: Add support to set compress format params ASoC: q6dsp: audioreach: Add gapless feature support Srinivas Kandagatla (7): ASoC: q6dsp: audioreach: add helper function to set u32 param ASoC: q6dsp: audioreach: Add placeholder decoder for compress playback ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress DAI and codec caps get callbacks ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI callbacks ASoC: q6dsp: q6apm-dai: Add mmap and copy compress DAI callbacks sound/soc/qcom/qdsp6/audioreach.c | 248 ++++++++++------- sound/soc/qcom/qdsp6/audioreach.h | 51 ++++ sound/soc/qcom/qdsp6/q6apm-dai.c | 445 ++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6apm.c | 68 +++++ sound/soc/qcom/qdsp6/q6apm.h | 6 + sound/soc/qcom/sc7280.c | 23 +- 6 files changed, 745 insertions(+), 96 deletions(-) -- 2.21.0

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[ PATCH v2 2/3] ASoC: loongson: Add Loongson ASoC Sound Card Support
by YingKun Meng 12 Jun '23

12 Jun '23

From: Yingkun Meng <mengyingkun(a)loongson.cn> The Loongson ASoC Sound Card is a general ASoC DAI Link driver that can be used for Loongson CPU DAI drivers and external CODECs. The driver supports the use of ACPI table to describe device resources. On loongson 7axxx platforms, the audio device is an ACPI device. Signed-off-by: Yingkun Meng <mengyingkun(a)loongson.cn> --- sound/soc/loongson/Kconfig | 10 ++ sound/soc/loongson/Makefile | 4 + sound/soc/loongson/loongson_card.c | 237 +++++++++++++++++++++++++++++ 3 files changed, 251 insertions(+) create mode 100644 sound/soc/loongson/loongson_card.c diff --git a/sound/soc/loongson/Kconfig b/sound/soc/loongson/Kconfig index 4478ac91e402..c175f9de19a8 100644 --- a/sound/soc/loongson/Kconfig +++ b/sound/soc/loongson/Kconfig @@ -13,4 +13,14 @@ config SND_SOC_LOONGSON_I2S_PCI The controller is found in loongson bridge chips or SoCs, and work as a PCI device. +config SND_SOC_LOONGSON_CARD + tristate "Loongson Sound Card Driver" + select SND_SOC_LOONGSON_I2S_PCI + help + Say Y or M if you want to add support for SoC audio using + loongson I2S controller. + + The driver add support for ALSA SoC Audio support using + loongson I2S controller. + endmenu diff --git a/sound/soc/loongson/Makefile b/sound/soc/loongson/Makefile index cfd0de1b1b22..a173a0fe17fe 100644 --- a/sound/soc/loongson/Makefile +++ b/sound/soc/loongson/Makefile @@ -2,3 +2,7 @@ #Platform Support snd-soc-loongson-i2s-pci-objs := loongson_i2s_pci.o loongson_i2s.o obj-$(CONFIG_SND_SOC_LOONGSON_I2S_PCI) += snd-soc-loongson-i2s-pci.o + +#Machine Support +snd-soc-loongson-card-objs := loongson_card.o +obj-$(CONFIG_SND_SOC_LOONGSON_CARD) += snd-soc-loongson-card.o diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c new file mode 100644 index 000000000000..7f0ce8957a4f --- /dev/null +++ b/sound/soc/loongson/loongson_card.c @@ -0,0 +1,237 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Loongson ASoC Audio driver + * + * Copyright (C) 2022 Loongson Technology Corporation Limited + */ + +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <linux/acpi.h> +#include <linux/pci.h> +#include <sound/pcm_params.h> + +static char codec_name[SND_ACPI_I2C_ID_LEN]; + +struct loongson_card_data { + struct snd_soc_card snd_card; + unsigned int mclk_fs; +}; + +static int loongson_card_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 *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct loongson_card_data *ls_card = snd_soc_card_get_drvdata(rtd->card); + int ret, mclk; + + if (ls_card->mclk_fs) { + mclk = ls_card->mclk_fs * params_rate(params); + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "cpu_dai clock not set\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "codec_dai clock not set\n"); + return ret; + } + } + return 0; +} + +static const struct snd_soc_ops loongson_ops = { + .hw_params = loongson_card_hw_params, +}; + +SND_SOC_DAILINK_DEFS(analog, + DAILINK_COMP_ARRAY(COMP_CPU("loongson-i2s")), + DAILINK_COMP_ARRAY(COMP_EMPTY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +static struct snd_soc_dai_link loongson_dai_links[] = { + { + .name = "HiFi PAIF TX", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBC_CFC, + SND_SOC_DAILINK_REG(analog), + .ops = &loongson_ops, + }, + { + .name = "HiFi PAIF RX", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBC_CFC, + SND_SOC_DAILINK_REG(analog), + .ops = &loongson_ops, + }, +}; + +static int loongson_card_parse_acpi(struct loongson_card_data *data) +{ + struct snd_soc_card *card = &data->snd_card; + struct fwnode_handle *fwnode = card->dev->fwnode; + struct fwnode_reference_args args; + const char *codec_dai_name; + struct acpi_device *adev; + struct device *phy_dev; + int ret, i; + + /* fixup platform name based on reference node */ + memset(&args, 0, sizeof(args)); + ret = acpi_node_get_property_reference(fwnode, "cpu", 0, &args); + if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + dev_err(card->dev, "No matching phy in ACPI table\n"); + return ret; + } + adev = to_acpi_device_node(args.fwnode); + phy_dev = acpi_get_first_physical_node(adev); + if (!phy_dev) + return -EPROBE_DEFER; + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].platforms->name = dev_name(phy_dev); + + /* fixup codec name based on reference node */ + memset(&args, 0, sizeof(args)); + ret = acpi_node_get_property_reference(fwnode, "codec", 0, &args); + if (ACPI_FAILURE(ret) || !is_acpi_device_node(args.fwnode)) { + dev_err(card->dev, "No matching phy in ACPI table\n"); + return ret; + } + adev = to_acpi_device_node(args.fwnode); + snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->name = codec_name; + + device_property_read_string(card->dev, "codec-dai-name", + &codec_dai_name); + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->dai_name = codec_dai_name; + + return 0; +} + +static int loongson_card_parse_of(struct loongson_card_data *data) +{ + const char *cpu_dai_name, *codec_dai_name; + struct device_node *cpu, *codec; + struct snd_soc_card *card = &data->snd_card; + struct device *dev = card->dev; + struct of_phandle_args args; + int ret, i; + + cpu = of_get_child_by_name(dev->of_node, "cpu"); + if (!cpu) { + dev_err(dev, "platform property missing or invalid\n"); + return -EINVAL; + } + codec = of_get_child_by_name(dev->of_node, "codec"); + if (!codec) { + dev_err(dev, "audio-codec property missing or invalid\n"); + ret = -EINVAL; + goto err; + } + + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(dev, "codec node missing #sound-dai-cells\n"); + goto err; + } + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].cpus->of_node = args.np; + + ret = of_parse_phandle_with_args(codec, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(dev, "codec node missing #sound-dai-cells\n"); + goto err; + } + for (i = 0; i < card->num_links; i++) + loongson_dai_links[i].codecs->of_node = args.np; + + snd_soc_of_get_dai_name(cpu, &cpu_dai_name); + snd_soc_of_get_dai_name(codec, &codec_dai_name); + for (i = 0; i < card->num_links; i++) { + loongson_dai_links[i].cpus->dai_name = cpu_dai_name; + loongson_dai_links[i].codecs->dai_name = codec_dai_name; + } + of_node_put(cpu); + of_node_put(codec); + + return 0; + +err: + of_node_put(cpu); + of_node_put(codec); + return ret; +} + +static int loongson_asoc_card_probe(struct platform_device *pdev) +{ + struct loongson_card_data *ls_priv; + struct snd_soc_card *card; + int ret; + + ls_priv = devm_kzalloc(&pdev->dev, sizeof(*ls_priv), GFP_KERNEL); + if (!ls_priv) + return -ENOMEM; + + card = &ls_priv->snd_card; + + card->dev = &pdev->dev; + card->owner = THIS_MODULE; + card->dai_link = loongson_dai_links; + card->num_links = ARRAY_SIZE(loongson_dai_links); + snd_soc_card_set_drvdata(card, ls_priv); + + ret = device_property_read_string(&pdev->dev, "model", &card->name); + if (ret) { + dev_err(&pdev->dev, "Error parsing card name: %d\n", ret); + return ret; + } + ret = device_property_read_u32(&pdev->dev, "mclk-fs", &ls_priv->mclk_fs); + if (ret) { + dev_err(&pdev->dev, "Error parsing mclk-fs: %d\n", ret); + return ret; + } + + if (has_acpi_companion(&pdev->dev)) + ret = loongson_card_parse_acpi(ls_priv); + else + ret = loongson_card_parse_of(ls_priv); + if (ret < 0) + return ret; + + ret = devm_snd_soc_register_card(&pdev->dev, card); + + return ret; +} + +static const struct of_device_id loongson_asoc_dt_ids[] = { + { .compatible = "loongson,ls-audio-card" }, + { /* sentinel */ }, +}; +MODULE_DEVICE_TABLE(of, loongson_asoc_dt_ids); + +static struct platform_driver loongson_audio_driver = { + .probe = loongson_asoc_card_probe, + .driver = { + .name = "loongson-asoc-card", + .pm = &snd_soc_pm_ops, + .of_match_table = of_match_ptr(loongson_asoc_dt_ids), + }, +}; +module_platform_driver(loongson_audio_driver); + +MODULE_DESCRIPTION("Loongson ASoc Sound Card driver"); +MODULE_AUTHOR("Loongson Technology Corporation Limited"); +MODULE_LICENSE("GPL"); -- 2.33.0

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[PATCH alsa-lib 0/3] ALSA-lib updates for UMP 1.1
by Takashi Iwai 12 Jun '23

12 Jun '23

Hi, this is a patch set for updating alsa-lib to catch up the changes for MIDI 2.0 UMP v1.1 features. The basic functionality remains, while the uapi header is updated a few API functions for the new features are added. Takashi === Takashi Iwai (3): uapi: Update rawmidi API to 2.0.4 ump: Add UMP 1.1 features seq: Add UMP 1.1 features include/seq.h | 3 ++ include/sound/uapi/asequencer.h | 5 ++- include/sound/uapi/asound.h | 21 ++++++++-- include/ump.h | 22 +++++++++++ include/ump_msg.h | 67 +++++++++++++++++++++++++++++++ src/Versions.in | 2 + src/rawmidi/ump.c | 70 +++++++++++++++++++++++++++++++++ src/seq/seq.c | 32 +++++++++++++++ 8 files changed, 218 insertions(+), 4 deletions(-) -- 2.35.3

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[PATCH] ASoC: mediatek: mt8173: Fix error path
by Ricardo Ribalda Delgado 12 Jun '23

12 Jun '23

After reordering the irq probe, the error path was not properly done. Lets fix it. Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Cc: stable(a)kernel.org Fixes: 4cbb264d4e91 ("ASoC: mediatek: mt8173: Enable IRQ when pdata is ready") Signed-off-by: Ricardo Ribalda Delgado <ribalda(a)chromium.org> --- 4cbb264d4e91 ("ASoC: mediatek: mt8173: Enable IRQ when pdata is ready") introduced a bug on the probe error path. Lets fix it. --- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index f93c2ec8beb7..f2fdcc74069a 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1070,6 +1070,10 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) afe->dev = &pdev->dev; + irq_id = platform_get_irq(pdev, 0); + if (irq_id <= 0) + return irq_id < 0 ? irq_id : -ENXIO; + afe->base_addr = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(afe->base_addr)) return PTR_ERR(afe->base_addr); @@ -1175,14 +1179,11 @@ static int mt8173_afe_pcm_dev_probe(struct platform_device *pdev) if (ret) goto err_cleanup_components; - irq_id = platform_get_irq(pdev, 0); - if (irq_id <= 0) - return irq_id < 0 ? irq_id : -ENXIO; ret = devm_request_irq(afe->dev, irq_id, mt8173_afe_irq_handler, 0, "Afe_ISR_Handle", (void *)afe); if (ret) { dev_err(afe->dev, "could not request_irq\n"); - goto err_pm_disable; + goto err_cleanup_components; } dev_info(&pdev->dev, "MT8173 AFE driver initialized.\n"); --- base-commit: 858fd168a95c5b9669aac8db6c14a9aeab446375 change-id: 20230612-mt8173-fixup-bc602a96dbad Best regards, -- Ricardo Ribalda Delgado <ribalda(a)chromium.org>

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