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[PATCH] ASoC: mediatek: mt8188: add constraints for PCM
by Trevor Wu 04 Jul '23

04 Jul '23

For certain projects, only a limited number of parameters are necessary. Therefore, it may be beneficial to add constraints that restrict the capacity of the PCM. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 145 +++++++++++++++++++++- 1 file changed, 144 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index ac69c23e0da1..0e4e1b8c99e2 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -30,6 +30,7 @@ #define TEST_MISO_DONE_2 BIT(29) #define NAU8825_HS_PRESENT BIT(0) +#define DPCM_FE_CONSTRAINT BIT(16) /* * Maxim MAX98390 @@ -452,6 +453,143 @@ enum { DAI_LINK_UL_SRC_BE, }; +static int mt8188_playback_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 48000 + }; + static const unsigned int channels[] = { + 2 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct mt8188_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mt8188_card_data *data = (struct mt8188_card_data *)priv->private_data; + int ret; + + if (!(data->quirk & DPCM_FE_CONSTRAINT)) + return 0; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list channel failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops mt8188_playback_ops = { + .startup = mt8188_playback_startup, +}; + +static int mt8188_capture_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 48000 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct mt8188_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mt8188_card_data *data = (struct mt8188_card_data *)priv->private_data; + int ret; + + if (!(data->quirk & DPCM_FE_CONSTRAINT)) + return 0; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops mt8188_capture_ops = { + .startup = mt8188_capture_startup, +}; + +static int mt8188_hdmi_dptx_startup(struct snd_pcm_substream *substream) +{ + static const unsigned int rates[] = { + 48000 + }; + static const unsigned int channels[] = { + 2, 4, 6, 8 + }; + static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, + }; + static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, + }; + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct mt8188_mt6359_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct mt8188_card_data *data = (struct mt8188_card_data *)priv->private_data; + int ret; + + if (!(data->quirk & DPCM_FE_CONSTRAINT)) + return 0; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list rate failed\n"); + return ret; + } + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + if (ret < 0) { + dev_err(rtd->dev, "hw_constraint_list channel failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops mt8188_hdmi_dptx_playback_ops = { + .startup = mt8188_hdmi_dptx_startup, +}; + static int mt8188_dptx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -723,6 +861,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8188_playback_ops, SND_SOC_DAILINK_REG(playback2), }, [DAI_LINK_DL3_FE] = { @@ -734,6 +873,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8188_playback_ops, SND_SOC_DAILINK_REG(playback3), }, [DAI_LINK_DL6_FE] = { @@ -778,6 +918,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .ops = &mt8188_hdmi_dptx_playback_ops, SND_SOC_DAILINK_REG(playback10), }, [DAI_LINK_DL11_FE] = { @@ -833,6 +974,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8188_capture_ops, SND_SOC_DAILINK_REG(capture4), }, [DAI_LINK_UL5_FE] = { @@ -844,6 +986,7 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, + .ops = &mt8188_capture_ops, SND_SOC_DAILINK_REG(capture5), }, [DAI_LINK_UL6_FE] = { @@ -1122,7 +1265,7 @@ static struct mt8188_card_data mt8188_evb_card = { static struct mt8188_card_data mt8188_nau8825_card = { .name = "mt8188_nau8825", - .quirk = NAU8825_HS_PRESENT, + .quirk = NAU8825_HS_PRESENT | DPCM_FE_CONSTRAINT, }; static const struct of_device_id mt8188_mt6359_dt_match[] = { -- 2.18.0

4 5

[PATCH v1 1/3] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 03 Jul '23

03 Jul '23

Integrate tas2781 configs for Lenovo Laptops. All of the tas2781s in the laptop will be aggregated as one speaker. The code support realtek as the primary codec. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v1: - remove white space at the end of the line in alc_fixup_headset_mode_alc255_no_hp_mic - Add tiwai(a)suse.de into Cc list - remove useless index - combine ALC287_FIXUP_TAS2781_I2C_2 and ALC287_FIXUP_TAS2781_I2C_4 together as ALC287_FIXUP_TAS2781_I2C, The code view all the tas2781s in the laptop as one instance. - delete the white space at the end of the line in alc_fixup_headset_mode_alc255_no_hp_mic - fix Laptop 0x17aa38be get the wrong fixup_id and enter into the wrong entry. --- sound/pci/hda/patch_realtek.c | 114 ++++++++++++++++++++++++++++++++-- 1 file changed, 110 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index afe8253f9a4f..2e5354a5decf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5883,7 +5883,7 @@ static void alc_fixup_headset_mode_alc255_no_hp_mic(struct hda_codec *codec, struct alc_spec *spec = codec->spec; spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; alc255_set_default_jack_type(codec); - } + } else alc_fixup_headset_mode(codec, fix, action); } @@ -6705,7 +6705,7 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } -struct cs35l41_dev_name { +struct scodec_dev_name { const char *bus; const char *hid; int index; @@ -6714,7 +6714,7 @@ struct cs35l41_dev_name { /* match the device name in a slightly relaxed manner */ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) { - struct cs35l41_dev_name *p = data; + struct scodec_dev_name *p = data; const char *d = dev_name(dev); int n = strlen(p->bus); char tmp[32]; @@ -6730,12 +6730,32 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +static int comp_match_tas2781_dev_name(struct device *dev, + void *data) +{ + struct scodec_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); + + return !strcmp(d + n, tmp); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct cs35l41_dev_name *rec; + struct scodec_dev_name *rec; int ret, i; switch (action) { @@ -6763,6 +6783,41 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } } +static void tas2781_generic_fixup(struct hda_codec *cdc, int action, + const char *bus, const char *hid) +{ + struct device *dev = hda_codec_dev(cdc); + struct alc_spec *spec = cdc->spec; + struct scodec_dev_name *rec; + int ret; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); + if (!rec) + return; + rec->bus = bus; + rec->hid = hid; + rec->index = 0; + spec->comps[0].codec = cdc; + component_match_add(dev, &spec->match, + comp_match_tas2781_dev_name, rec); + ret = component_master_add_with_match(dev, &comp_master_ops, + spec->match); + if (ret) + codec_err(cdc, + "Fail to register component aggregator %d\n", + ret); + else + spec->gen.pcm_playback_hook = + comp_generic_playback_hook; + break; + case HDA_FIXUP_ACT_FREE: + component_master_del(dev, &comp_master_ops); + break; + } +} + static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); @@ -6790,6 +6845,12 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); } +static void tas2781_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7210,6 +7271,7 @@ enum { ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, + ALC287_FIXUP_TAS2781_I2C, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9255,6 +9317,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC287_FIXUP_TAS2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9813,6 +9881,33 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", + ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", + ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -10728,6 +10823,17 @@ static int patch_alc269(struct hda_codec *codec) codec->fixup_id = HDA_FIXUP_ID_NOT_SET; } + /* FIXME: Laptop 0x17aa38be will get the wrong fixup_id and + * enter into the wrong entry. + * Correct the wrong entry. + */ + if (codec->fixup_id == ALC287_FIXUP_YOGA7_14ITL_SPEAKERS && + codec->core.vendor_id == 0x10ec0287 && + codec->core.subsystem_id == 0x17aa38be) { + codec_dbg(codec, "Clear wrong fixup for 17aa38be\n"); + codec->fixup_id = ALC287_FIXUP_TAS2781_I2C; + } + snd_hda_pick_pin_fixup(codec, alc269_pin_fixup_tbl, alc269_fixups, true); snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, -- 2.34.1

2 4

[PATCH] ASoC: codecs: wcd938x: fix soundwire initialisation race
by Johan Hovold 03 Jul '23

03 Jul '23

Make sure that the soundwire device used for register accesses has been enumerated and initialised before trying to read the codec variant during component probe. This specifically avoids interpreting (a masked and shifted) -EBUSY errno as the variant: wcd938x_codec audio-codec: ASoC: error at soc_component_read_no_lock on audio-codec for register: [0x000034b0] -16 in case the soundwire device has not yet been initialised, which in turn prevents some headphone controls from being registered. Fixes: 8d78602aa87a ("ASoC: codecs: wcd938x: add basic driver") Cc: stable(a)vger.kernel.org # 5.14 Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Reported-by: Steev Klimaszewski <steev(a)kali.org> Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- sound/soc/codecs/wcd938x.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index e3ae4fb2c4db..4571588fad62 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3080,9 +3080,18 @@ static int wcd938x_irq_init(struct wcd938x_priv *wcd, struct device *dev) static int wcd938x_soc_codec_probe(struct snd_soc_component *component) { struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + struct sdw_slave *tx_sdw_dev = wcd938x->tx_sdw_dev; struct device *dev = component->dev; + unsigned long time_left; int ret, i; + time_left = wait_for_completion_timeout(&tx_sdw_dev->initialization_complete, + msecs_to_jiffies(2000)); + if (!time_left) { + dev_err(dev, "soundwire device init timeout\n"); + return -ETIMEDOUT; + } + snd_soc_component_init_regmap(component, wcd938x->regmap); ret = pm_runtime_resume_and_get(dev); -- 2.39.3

3 2

[PATCH 0/5] ASoC: codecs: wcd938x/wcd934x: loglevel fix and cleanups
by Johan Hovold 03 Jul '23

03 Jul '23

When investigating a race in the wcd938x driver I noticed that the MBHC impedance measurements where printed at error loglevel which is clearly wrong. Fix that, and clean up the logging somewhat by using dev_printk() and addressing some style issues. Included are also two patches that drop the bogus inline keywords from the functions involved. Johan Johan Hovold (5): ASoC: codecs: wcd938x: fix mbhc impedance loglevel ASoC: codecs: wcd938x: drop inline keywords ASoC: codecs: wcd938x: use dev_printk() for impedance logging ASoC: codecs: wcd934x: demote impedance printk loglevel ASoC: codecs: wcd934x: drop inline keywords sound/soc/codecs/wcd934x.c | 8 ++++---- sound/soc/codecs/wcd938x.c | 17 +++++++++-------- 2 files changed, 13 insertions(+), 12 deletions(-) -- 2.39.3

3 9

[PATCH] ASoC: qdsp6: q6apm: use dai link pcm id as pcm device number
by Srinivas Kandagatla 03 Jul '23

03 Jul '23

For some reason we ended up with a setup without this flag. This resulted in inconsistent sound card devices numbers which are also not starting as expected at dai_link->id. (Ex: MultiMedia1 pcm ended up with device number 4 instead of 0) With this patch patch now the MultiMedia1 PCM ends up with device number 0 as expected. Fixes: 9b4fe0f1cd79 ("ASoC: qdsp6: audioreach: add q6apm-dai support") Cc: <Stable(a)vger.kernel.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm-dai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 5eb0b864c740..c90db6daabbd 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -840,6 +840,7 @@ static const struct snd_soc_component_driver q6apm_fe_dai_component = { .pointer = q6apm_dai_pointer, .trigger = q6apm_dai_trigger, .compress_ops = &q6apm_dai_compress_ops, + .use_dai_pcm_id = true, }; static int q6apm_dai_probe(struct platform_device *pdev) -- 2.21.0

5 17

[PATCH] ALSA: pcm: Fix potential data race at PCM memory allocation helpers
by Takashi Iwai 03 Jul '23

03 Jul '23

The PCM memory allocation helpers have a sanity check against too many buffer allocations. However, the check is performed without a proper lock and the allocation isn't serialized; this allows user to allocate more memories than predefined max size. Practically seen, this isn't really a big problem, as it's more or less some "soft limit" as a sanity check, and it's not possible to allocate unlimitedly. But it's still better to address this for more consistent behavior. The patch covers the size check in do_alloc_pages() with the card->memory_mutex, and increases the allocated size there for preventing the further overflow. When the actual allocation fails, the size is decreased accordingly. Reported-by: BassCheck <bass(a)buaa.edu.cn> Reported-by: Tuo Li <islituo(a)gmail.com> Link: https://lore.kernel.org/r/CADm8Tek6t0WedK+3Y6rbE5YEt19tML8BUL45N2ji4ZAz1KcN… Reviewed-by: Jaroslav Kysela <perex(a)perex.cz> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_memory.c | 44 +++++++++++++++++++++++++++++++++-------- 1 file changed, 36 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 7bde7fb64011..a0b951471699 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -31,15 +31,41 @@ static unsigned long max_alloc_per_card = 32UL * 1024UL * 1024UL; module_param(max_alloc_per_card, ulong, 0644); MODULE_PARM_DESC(max_alloc_per_card, "Max total allocation bytes per card."); +static void __update_allocated_size(struct snd_card *card, ssize_t bytes) +{ + card->total_pcm_alloc_bytes += bytes; +} + +static void update_allocated_size(struct snd_card *card, ssize_t bytes) +{ + mutex_lock(&card->memory_mutex); + __update_allocated_size(card, bytes); + mutex_unlock(&card->memory_mutex); +} + +static void decrease_allocated_size(struct snd_card *card, size_t bytes) +{ + mutex_lock(&card->memory_mutex); + WARN_ON(card->total_pcm_alloc_bytes < bytes); + __update_allocated_size(card, -(ssize_t)bytes); + mutex_unlock(&card->memory_mutex); +} + static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, int str, size_t size, struct snd_dma_buffer *dmab) { enum dma_data_direction dir; int err; + /* check and reserve the requested size */ + mutex_lock(&card->memory_mutex); if (max_alloc_per_card && - card->total_pcm_alloc_bytes + size > max_alloc_per_card) + card->total_pcm_alloc_bytes + size > max_alloc_per_card) { + mutex_unlock(&card->memory_mutex); return -ENOMEM; + } + __update_allocated_size(card, size); + mutex_unlock(&card->memory_mutex); if (str == SNDRV_PCM_STREAM_PLAYBACK) dir = DMA_TO_DEVICE; @@ -47,9 +73,14 @@ static int do_alloc_pages(struct snd_card *card, int type, struct device *dev, dir = DMA_FROM_DEVICE; err = snd_dma_alloc_dir_pages(type, dev, dir, size, dmab); if (!err) { - mutex_lock(&card->memory_mutex); - card->total_pcm_alloc_bytes += dmab->bytes; - mutex_unlock(&card->memory_mutex); + /* the actual allocation size might be bigger than requested, + * and we need to correct the account + */ + if (dmab->bytes != size) + update_allocated_size(card, dmab->bytes - size); + } else { + /* take back on allocation failure */ + decrease_allocated_size(card, size); } return err; } @@ -58,10 +89,7 @@ static void do_free_pages(struct snd_card *card, struct snd_dma_buffer *dmab) { if (!dmab->area) return; - mutex_lock(&card->memory_mutex); - WARN_ON(card->total_pcm_alloc_bytes < dmab->bytes); - card->total_pcm_alloc_bytes -= dmab->bytes; - mutex_unlock(&card->memory_mutex); + decrease_allocated_size(card, dmab->bytes); snd_dma_free_pages(dmab); dmab->area = NULL; } -- 2.35.3

1 0

[PATCH 5/5] ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 03 Jul '23

03 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> During ClockStop Mode0, peripheral interrupts are disabled. When system level resume is invoked, Peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt722-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index cc57e4e27805..e9103ffb3f50 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -463,8 +463,16 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev) if (!rt722->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt722->disable_irq == true) { + mutex_lock(&rt722->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt722->disable_irq = false; + mutex_unlock(&rt722->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT722_PROBE_TIMEOUT)); -- 2.40.1

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[PATCH 4/5] ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 03 Jul '23

03 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> During ClockStop Mode0, peripheral interrupts are disabled. When system level resume is invoked, Peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index ad06267b0ea0..6bc50396a0f6 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -438,8 +438,16 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) if (!rt712->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt712->disable_irq == true) { + mutex_lock(&rt712->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt712->disable_irq = false; + mutex_unlock(&rt712->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); -- 2.40.1

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[PATCH 3/5] ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 03 Jul '23

03 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> During ClockStop Mode0, peripheral interrupts are disabled. When system level resume is invoked, Peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 119e1f9605d7..23f23f714b39 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -438,8 +438,16 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

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[PATCH 2/5] ASoC: rt711: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 03 Jul '23

03 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> During ClockStop Mode0, peripheral interrupts are disabled. When system level resume is invoked, Peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 87dafcb4545d..33dced388f9e 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -538,8 +538,15 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.40.1

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