diff options
Diffstat (limited to 'sound')
40 files changed, 845 insertions, 149 deletions
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 09c0e2a6489c..71323d807dbf 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -251,7 +251,10 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) switch (substream->dma_buffer.dev.type) { case SNDRV_DMA_TYPE_UNKNOWN: - return false; + /* we can't know the device, so just assume that the driver does + * everything right + */ + return true; case SNDRV_DMA_TYPE_CONTINUOUS: case SNDRV_DMA_TYPE_VMALLOC: return true; diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 0ef242fdd3bc..fff18b5d4e05 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -153,7 +153,7 @@ static int init_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream) struct cmp_connection *conn; enum cmp_direction c_dir; enum amdtp_stream_direction s_dir; - unsigned int flags = CIP_UNAWARE_SYT; + unsigned int flags = 0; int err; if (!(oxfw->quirks & SND_OXFW_QUIRK_BLOCKING_TRANSMISSION)) @@ -161,6 +161,13 @@ static int init_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream) else flags |= CIP_BLOCKING; + // OXFW 970/971 has no function to generate playback timing according to the sequence + // of value in syt field, thus the packet should include NO_INFO value in the field. + // However, some models just ignore data blocks in packet with NO_INFO for audio data + // processing. + if (!(oxfw->quirks & SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET)) + flags |= CIP_UNAWARE_SYT; + if (stream == &oxfw->tx_stream) { conn = &oxfw->out_conn; c_dir = CMP_OUTPUT; diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 84971d78d152..cb5b5e3a481b 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -159,8 +159,10 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id return snd_oxfw_scs1x_add(oxfw); } - if (entry->vendor_id == OUI_APOGEE && entry->model_id == MODEL_DUET_FW) - oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + if (entry->vendor_id == OUI_APOGEE && entry->model_id == MODEL_DUET_FW) { + oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION | + SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET; + } /* * TASCAM FireOne has physical control and requires a pair of additional diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index ee47abcb0c90..c13034f6c2ca 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -42,6 +42,11 @@ enum snd_oxfw_quirk { SND_OXFW_QUIRK_BLOCKING_TRANSMISSION = 0x04, // Stanton SCS1.d and SCS1.m support unique transaction. SND_OXFW_QUIRK_SCS_TRANSACTION = 0x08, + // Apogee Duet FireWire ignores data blocks in packet with NO_INFO for audio data + // processing, while output level meter moves. Any value in syt field of packet takes + // the device to process audio data even if the value is invalid in a point of + // IEC 61883-1/6. + SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET = 0x10, }; /* This is an arbitrary number for convinience. */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index e97d00585e8e..481d8f8d3396 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3460,7 +3460,7 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, struct hda_gen_spec *spec = codec->spec; const struct hda_input_mux *imux; struct nid_path *path; - int i, adc_idx, err = 0; + int i, adc_idx, ret, err = 0; imux = &spec->input_mux; adc_idx = kcontrol->id.index; @@ -3470,9 +3470,13 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, if (!path || !path->ctls[type]) continue; kcontrol->private_value = path->ctls[type]; - err = func(kcontrol, ucontrol); - if (err < 0) + ret = func(kcontrol, ucontrol); + if (ret < 0) { + err = ret; break; + } + if (ret > 0) + err = 1; } mutex_unlock(&codec->control_mutex); if (err >= 0 && spec->cap_sync_hook) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0322b289505e..0062c18b646a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -883,10 +883,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } -static void azx_shutdown_chip(struct azx *chip) +static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset) { azx_stop_chip(chip); - azx_enter_link_reset(chip); + if (!skip_link_reset) + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); display_power(chip, false); } @@ -895,6 +896,11 @@ static void azx_shutdown_chip(struct azx *chip) static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); +static void azx_shutdown_chip(struct azx *chip) +{ + __azx_shutdown_chip(chip, false); +} + static void azx_add_card_list(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -2385,7 +2391,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - azx_shutdown_chip(chip); + __azx_shutdown_chip(chip, true); } /* PCI IDs */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 21c521596c9d..7ad689f991e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6658,6 +6658,7 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, + ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, }; static const struct hda_fixup alc269_fixups[] = { @@ -8242,6 +8243,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_XIAOMI_HEADSET_MIC }, + [ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8332,6 +8339,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), + SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -8431,13 +8439,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), - SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), @@ -8465,6 +8474,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), + SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a5c1a2c4eae4..773a136161f1 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1041,6 +1041,7 @@ static const struct hda_fixup via_fixups[] = { }; static const struct snd_pci_quirk vt2002p_fixups[] = { + SND_PCI_QUIRK(0x1043, 0x13f7, "Asus B23E", VIA_FIXUP_POWER_SAVE), SND_PCI_QUIRK(0x1043, 0x1487, "Asus G75", VIA_FIXUP_ASUS_G75), SND_PCI_QUIRK(0x1043, 0x8532, "Asus X202E", VIA_FIXUP_INTMIC_BOOST), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo", VIA_FIXUP_POWER_SAVE), diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 8a13462e1a63..5dcf77af07af 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_COMPRESS config SND_SOC_TOPOLOGY bool + select SND_DYNAMIC_MINORS config SND_SOC_TOPOLOGY_KUNIT_TEST tristate "KUnit tests for SoC topology" diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 9449fb40a956..3c60c5f96dcb 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -525,6 +525,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_da7219_play_ops, SND_SOC_DAILINK_REG(designware1, dlgs, platform), }, @@ -534,6 +535,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_da7219_cap_ops, SND_SOC_DAILINK_REG(designware2, dlgs, platform), }, @@ -543,6 +545,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_max_play_ops, SND_SOC_DAILINK_REG(designware3, mx, platform), }, @@ -553,6 +556,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_dmic0_cap_ops, SND_SOC_DAILINK_REG(designware3, adau, platform), }, @@ -563,6 +567,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_dmic1_cap_ops, SND_SOC_DAILINK_REG(designware2, adau, platform), }, diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 143155a840ac..cc1ce6f22caa 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -969,7 +969,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component, acp_set_sram_bank_state(rtd->acp_mmio, 0, true); /* Save for runtime private data */ - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = runtime->dma_addr; rtd->order = get_order(size); /* Fill the page table entries in ACP SRAM */ diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 8148b0d22e88..597d7c4b2a6b 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -286,7 +286,7 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, pr_err("pinfo failed\n"); } size = params_buffer_bytes(params); - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = substream->runtime->dma_addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); config_acp3x_dma(rtd, substream->stream); return 0; diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index bd20622b0933..0391c28dd078 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -242,7 +242,7 @@ static int acp_pdm_dma_hw_params(struct snd_soc_component *component, return -EINVAL; size = params_buffer_bytes(params); period_bytes = params_period_bytes(params); - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = substream->runtime->dma_addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); config_acp_dma(rtd, substream->stream); init_pdm_ring_buffer(MEM_WINDOW_START, size, period_bytes, diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 19438da5dfa5..7b8040e812a1 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -382,6 +382,8 @@ static const struct dev_pm_ops rn_acp_pm = { .runtime_resume = snd_rn_acp_resume, .suspend = snd_rn_acp_suspend, .resume = snd_rn_acp_resume, + .restore = snd_rn_acp_resume, + .poweroff = snd_rn_acp_suspend, }; static void snd_rn_acp_remove(struct pci_dev *pci) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a3b784ed4f70..9ff1600ca823 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -187,6 +187,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT715_SDCA_SDW imply SND_SOC_RT1308_SDW imply SND_SOC_RT1316_SDW + imply SND_SOC_SDW_MOCKUP imply SND_SOC_SGTL5000 imply SND_SOC_SI476X imply SND_SOC_SIMPLE_AMPLIFIER @@ -1287,6 +1288,23 @@ config SND_SOC_RT715_SDCA_SDW select REGMAP_SOUNDWIRE select REGMAP_SOUNDWIRE_MBQ +config SND_SOC_SDW_MOCKUP + tristate "SoundWire mockup codec" + depends on EXPERT + depends on SOUNDWIRE + help + This option enables a SoundWire mockup codec that does not drive the + bus, take part in the command/command protocol or generate data on a + Source port. + This option is only intended to be used for tests on a device + with a connector, in combination with a bus analyzer, or to test new + topologies that differ from the actual hardware layout. + This mockup device could be totally virtual but could also be a + real physical one with one key restriction: it is not allowed by the + SoundWire specification to be configured via a sideband mechanism and + generate audio data for capture. However, nothing prevents such a + peripheral device from snooping the bus. + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" @@ -1559,6 +1577,7 @@ config SND_SOC_WCD934X config SND_SOC_WCD938X depends on SND_SOC_WCD938X_SDW tristate + depends on SOUNDWIRE || !SOUNDWIRE config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index de8b83dd2c76..8dcea2c4604a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -203,6 +203,7 @@ snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-rt711-sdca-objs := rt711-sdca.o rt711-sdca-sdw.o snd-soc-rt715-objs := rt715.o rt715-sdw.o snd-soc-rt715-sdca-objs := rt715-sdca.o rt715-sdca-sdw.o +snd-soc-sdw-mockup-objs := sdw-mockup.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -530,6 +531,7 @@ obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_RT711_SDCA_SDW) += snd-soc-rt711-sdca.o obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o obj-$(CONFIG_SND_SOC_RT715_SDCA_SDW) += snd-soc-rt715-sdca.o +obj-$(CONFIG_SND_SOC_SDW_MOCKUP) += snd-soc-sdw-mockup.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o @@ -583,7 +585,10 @@ obj-$(CONFIG_SND_SOC_WCD_MBHC) += snd-soc-wcd-mbhc.o obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o obj-$(CONFIG_SND_SOC_WCD934X) += snd-soc-wcd934x.o obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x.o -obj-$(CONFIG_SND_SOC_WCD938X_SDW) += snd-soc-wcd938x-sdw.o +ifdef CONFIG_SND_SOC_WCD938X_SDW +# avoid link failure by forcing sdw code built-in when needed +obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x-sdw.o +endif obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index eff013f295be..99c022be94a6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = { .use_single_write = true, }; -static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); +static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { @@ -425,34 +425,23 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_CF_SHIFT, cs42l42_wnf3_freq_text); -static const char * const cs42l42_wnf05_freq_text[] = { - "280Hz", "315Hz", "350Hz", "385Hz", - "420Hz", "455Hz", "490Hz", "525Hz" -}; - -static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL, - CS42L42_ADC_WNF_CF_SHIFT, - cs42l42_wnf05_freq_text); - static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, - CS42L42_ADC_NOTCH_DIS_SHIFT, true, false), + CS42L42_ADC_NOTCH_DIS_SHIFT, true, true), SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL, CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false), SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL, CS42L42_ADC_INV_SHIFT, true, false), SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL, CS42L42_ADC_DIG_BOOST_SHIFT, true, false), - SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME, - CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv), + SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv), SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_EN_SHIFT, true, false), SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_HPF_EN_SHIFT, true, false), SOC_ENUM("HPF Corner Freq", cs42l42_hpf_freq_enum), SOC_ENUM("WNF 3dB Freq", cs42l42_wnf3_freq_enum), - SOC_ENUM("WNF 05dB Freq", cs42l42_wnf05_freq_enum), /* DAC Volume and Filter Controls */ SOC_SINGLE("DACA Invert Switch", CS42L42_DAC_CTL1, @@ -471,8 +460,8 @@ static const struct snd_soc_dapm_widget cs42l42_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HP"), SND_SOC_DAPM_DAC("DAC", NULL, CS42L42_PWR_CTL1, CS42L42_HP_PDN_SHIFT, 1), SND_SOC_DAPM_MIXER("MIXER", CS42L42_PWR_CTL1, CS42L42_MIXER_PDN_SHIFT, 1, NULL, 0), - SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH1_SHIFT, 0), - SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, CS42L42_ASP_RX_DAI0_EN, CS42L42_ASP_RX0_CH2_SHIFT, 0), + SND_SOC_DAPM_AIF_IN("SDIN1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SDIN2", NULL, 1, SND_SOC_NOPM, 0, 0), /* Playback Requirements */ SND_SOC_DAPM_SUPPLY("ASP DAI0", CS42L42_PWR_CTL1, CS42L42_ASP_DAI_PDN_SHIFT, 1, NULL, 0), @@ -630,6 +619,8 @@ static int cs42l42_pll_config(struct snd_soc_component *component) for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { if (pll_ratio_table[i].sclk == clk) { + cs42l42->pll_config = i; + /* Configure the internal sample rate */ snd_soc_component_update_bits(component, CS42L42_MCLK_CTL, CS42L42_INTERNAL_FS_MASK, @@ -638,14 +629,9 @@ static int cs42l42_pll_config(struct snd_soc_component *component) (pll_ratio_table[i].mclk_int != 24000000)) << CS42L42_INTERNAL_FS_SHIFT); - /* Set the MCLK src (PLL or SCLK) and the divide - * ratio - */ + snd_soc_component_update_bits(component, CS42L42_MCLK_SRC_SEL, - CS42L42_MCLK_SRC_SEL_MASK | CS42L42_MCLKDIV_MASK, - (pll_ratio_table[i].mclk_src_sel - << CS42L42_MCLK_SRC_SEL_SHIFT) | (pll_ratio_table[i].mclk_div << CS42L42_MCLKDIV_SHIFT)); /* Set up the LRCLK */ @@ -681,15 +667,6 @@ static int cs42l42_pll_config(struct snd_soc_component *component) CS42L42_FSYNC_PULSE_WIDTH_MASK, CS42L42_FRAC1_VAL(fsync - 1) << CS42L42_FSYNC_PULSE_WIDTH_SHIFT); - snd_soc_component_update_bits(component, - CS42L42_ASP_FRM_CFG, - CS42L42_ASP_5050_MASK, - CS42L42_ASP_5050_MASK); - /* Set the frame delay to 1.0 SCLK clocks */ - snd_soc_component_update_bits(component, CS42L42_ASP_FRM_CFG, - CS42L42_ASP_FSD_MASK, - CS42L42_ASP_FSD_1_0 << - CS42L42_ASP_FSD_SHIFT); /* Set the sample rates (96k or lower) */ snd_soc_component_update_bits(component, CS42L42_FS_RATE_EN, CS42L42_FS_EN_MASK, @@ -789,7 +766,18 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - case SND_SOC_DAIFMT_LEFT_J: + /* + * 5050 mode, frame starts on falling edge of LRCLK, + * frame delayed by 1.0 SCLKs + */ + snd_soc_component_update_bits(component, + CS42L42_ASP_FRM_CFG, + CS42L42_ASP_STP_MASK | + CS42L42_ASP_5050_MASK | + CS42L42_ASP_FSD_MASK, + CS42L42_ASP_5050_MASK | + (CS42L42_ASP_FSD_1_0 << + CS42L42_ASP_FSD_SHIFT)); break; default: return -EINVAL; @@ -819,6 +807,25 @@ static int cs42l42_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } +static int cs42l42_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + + /* + * Sample rates < 44.1 kHz would produce an out-of-range SCLK with + * a standard I2S frame. If the machine driver sets SCLK it must be + * legal. + */ + if (cs42l42->sclk) + return 0; + + /* Machine driver has not set a SCLK, limit bottom end to 44.1 kHz */ + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + 44100, 192000); +} + static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -832,6 +839,10 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, cs42l42->srate = params_rate(params); cs42l42->bclk = snd_soc_params_to_bclk(params); + /* I2S frame always has 2 channels even for mono audio */ + if (channels == 1) + cs42l42->bclk *= 2; + switch(substream->stream) { case SNDRV_PCM_STREAM_CAPTURE: if (channels == 2) { @@ -855,6 +866,17 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_CH2_AP_RES, CS42L42_ASP_RX_CH_AP_MASK | CS42L42_ASP_RX_CH_RES_MASK, val); + + /* Channel B comes from the last active channel */ + snd_soc_component_update_bits(component, CS42L42_SP_RX_CH_SEL, + CS42L42_SP_RX_CHB_SEL_MASK, + (channels - 1) << CS42L42_SP_RX_CHB_SEL_SHIFT); + + /* Both LRCLK slots must be enabled */ + snd_soc_component_update_bits(component, CS42L42_ASP_RX_DAI0_EN, + CS42L42_ASP_RX0_CH_EN_MASK, + BIT(CS42L42_ASP_RX0_CH1_SHIFT) | + BIT(CS42L42_ASP_RX0_CH2_SHIFT)); break; default: break; @@ -900,13 +922,21 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) */ regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_osc_seq, ARRAY_SIZE(cs42l42_to_osc_seq)); + + /* Must disconnect PLL before stopping it */ + snd_soc_component_update_bits(component, + CS42L42_MCLK_SRC_SEL, + CS42L42_MCLK_SRC_SEL_MASK, + 0); + usleep_range(100, 200); + snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 0); } } else { if (!cs42l42->stream_use) { /* SCLK must be running before codec unmute */ - if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600)) { + if (pll_ratio_table[cs42l42->pll_config].mclk_src_sel) { snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 1); @@ -927,6 +957,12 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) CS42L42_PLL_LOCK_TIMEOUT_US); if (ret < 0) dev_warn(component->dev, "PLL failed to lock: %d\n", ret); + + /* PLL must be running to drive glitchless switch logic */ + snd_soc_component_update_bits(component, + CS42L42_MCLK_SRC_SEL, + CS42L42_MCLK_SRC_SEL_MASK, + CS42L42_MCLK_SRC_SEL_MASK); } /* Mark SCLK as present, turn off internal oscillator */ @@ -960,8 +996,8 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE ) - static const struct snd_soc_dai_ops cs42l42_ops = { + .startup = cs42l42_dai_startup, .hw_params = cs42l42_pcm_hw_params, .set_fmt = cs42l42_set_dai_fmt, .set_sysclk = cs42l42_set_sysclk, diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 206b3c81d3e0..8734f6828f3e 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -653,6 +653,8 @@ /* Page 0x25 Audio Port Registers */ #define CS42L42_SP_RX_CH_SEL (CS42L42_PAGE_25 + 0x01) +#define CS42L42_SP_RX_CHB_SEL_SHIFT 2 +#define CS42L42_SP_RX_CHB_SEL_MASK (3 << CS42L42_SP_RX_CHB_SEL_SHIFT) #define CS42L42_SP_RX_ISOC_CTL (CS42L42_PAGE_25 + 0x02) #define CS42L42_SP_RX_RSYNC_SHIFT 6 @@ -775,6 +777,7 @@ struct cs42l42_private { struct gpio_desc *reset_gpio; struct completion pdn_done; struct snd_soc_jack *jack; + int pll_config; int bclk; u32 sclk; u32 srate; diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15bd8335f667..db88be48c998 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -828,36 +828,6 @@ static void nau8824_int_status_clear_all(struct regmap *regmap) } } -static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_disable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_disable_pin(dapm, pin); - } -} - -static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_force_enable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_force_enable_pin(dapm, pin); - } -} - static void nau8824_eject_jack(struct nau8824 *nau8824) { struct snd_soc_dapm_context *dapm = nau8824->dapm; @@ -866,8 +836,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824) /* Clear all interruption status */ nau8824_int_status_clear_all(regmap); - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); /* Enable the insertion interruption, disable the ejection @@ -897,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - nau8824_dapm_enable_pin(nau8824, "MICBIAS"); - nau8824_dapm_enable_pin(nau8824, "SAR"); + snd_soc_dapm_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); @@ -909,8 +879,8 @@ static void nau8824_jdet_work(struct work_struct *work) if (adc_value < HEADSET_SARADC_THD) { event |= SND_JACK_HEADPHONE; - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); } else { event |= SND_JACK_HEADSET; diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index abcd6f483788..51ecaa2abcd1 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -44,6 +44,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, {RT5682_SAR_IL_CMD_3, 0x8365}, + {RT5682_SAR_IL_CMD_6, 0x0180}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) diff --git a/sound/soc/codecs/sdw-mockup.c b/sound/soc/codecs/sdw-mockup.c new file mode 100644 index 000000000000..8ea13cfa9f8e --- /dev/null +++ b/sound/soc/codecs/sdw-mockup.c @@ -0,0 +1,312 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// sdw-mockup.c -- a mockup SoundWire codec for tests where only the host +// drives the bus. +// +// Copyright(c) 2021 Intel Corporation +// +// + +#include <linux/device.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/soundwire/sdw_registers.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +struct sdw_mockup_priv { + struct sdw_slave *slave; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +static int sdw_mockup_component_probe(struct snd_soc_component *component) +{ + return 0; +} + +static void sdw_mockup_component_remove(struct snd_soc_component *component) +{ +} + +static const struct snd_soc_component_driver snd_soc_sdw_mockup_component = { + .probe = sdw_mockup_component_probe, + .remove = sdw_mockup_component_remove, +}; + +static int sdw_mockup_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void sdw_mockup_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int sdw_mockup_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct sdw_mockup_priv *sdw_mockup = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int num_channels; + int port; + int ret; + + stream = snd_soc_dai_get_dma_data(dai, substream); + if (!stream) + return -EINVAL; + + if (!sdw_mockup->slave) + return -EINVAL; + + /* SoundWire specific configuration */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + direction = SDW_DATA_DIR_TX; + port = 8; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = (1 << num_channels) - 1; + port_config.num = port; + + ret = sdw_stream_add_slave(sdw_mockup->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (ret) + dev_err(dai->dev, "Unable to configure port\n"); + + return ret; +} + +static int sdw_mockup_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct sdw_mockup_priv *sdw_mockup = snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_mockup->slave) + return -EINVAL; + + sdw_stream_remove_slave(sdw_mockup->slave, stream->sdw_stream); + return 0; +} + +static const struct snd_soc_dai_ops sdw_mockup_ops = { + .hw_params = sdw_mockup_pcm_hw_params, + .hw_free = sdw_mockup_pcm_hw_free, + .set_sdw_stream = sdw_mockup_set_sdw_stream, + .shutdown = sdw_mockup_shutdown, +}; + +static struct snd_soc_dai_driver sdw_mockup_dai[] = { + { + .name = "sdw-mockup-aif1", + .id = 1, + .playback = { + .stream_name = "DP1 Playback", + .channels_min = 1, + .channels_max = 2, + }, + .capture = { + .stream_name = "DP8 Capture", + .channels_min = 1, + .channels_max = 2, + }, + .ops = &sdw_mockup_ops, + }, +}; + +static int sdw_mockup_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + return 0; +} + +static int sdw_mockup_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = false; + + /* + * first we need to allocate memory for set bits in port lists + * the port allocation is completely arbitrary: + * DP0 is not supported + * DP1 is sink + * DP8 is source + */ + prop->source_ports = BIT(8); + prop->sink_ports = BIT(1); + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), + GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + j++; + } + + prop->simple_clk_stop_capable = true; + + /* wake-up event */ + prop->wake_capable = 0; + + return 0; +} + +static int sdw_mockup_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + return 0; +} + +static int sdw_mockup_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + return 0; +} + +static const struct sdw_slave_ops sdw_mockup_slave_ops = { + .read_prop = sdw_mockup_read_prop, + .interrupt_callback = sdw_mockup_interrupt_callback, + .update_status = sdw_mockup_update_status, + .bus_config = sdw_mockup_bus_config, +}; + +static int sdw_mockup_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct device *dev = &slave->dev; + struct sdw_mockup_priv *sdw_mockup; + int ret; + + sdw_mockup = devm_kzalloc(dev, sizeof(*sdw_mockup), GFP_KERNEL); + if (!sdw_mockup) + return -ENOMEM; + + dev_set_drvdata(dev, sdw_mockup); + sdw_mockup->slave = slave; + + slave->is_mockup_device = true; + + ret = devm_snd_soc_register_component(dev, + &snd_soc_sdw_mockup_component, + sdw_mockup_dai, + ARRAY_SIZE(sdw_mockup_dai)); + + return ret; +} + +static int sdw_mockup_sdw_remove(struct sdw_slave *slave) +{ + return 0; +} + +/* + * Intel reserved parts ID with the following mapping expected: + * 0xAAAA: generic full-duplex codec + * 0xAA55: headset codec (mock-up of RT711/RT5682) - full-duplex + * 0x55AA: amplifier (mock-up of RT1308/Maxim 98373) - playback only with + * IV feedback + * 0x5555: mic codec (mock-up of RT715) - capture-only + */ +static const struct sdw_device_id sdw_mockup_id[] = { + SDW_SLAVE_ENTRY_EXT(0x0105, 0xAAAA, 0x0, 0, 0), + SDW_SLAVE_ENTRY_EXT(0x0105, 0xAA55, 0x0, 0, 0), + SDW_SLAVE_ENTRY_EXT(0x0105, 0x55AA, 0x0, 0, 0), + SDW_SLAVE_ENTRY_EXT(0x0105, 0x5555, 0x0, 0, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, sdw_mockup_id); + +static struct sdw_driver sdw_mockup_sdw_driver = { + .driver = { + .name = "sdw-mockup", + .owner = THIS_MODULE, + }, + .probe = sdw_mockup_sdw_probe, + .remove = sdw_mockup_sdw_remove, + .ops = &sdw_mockup_slave_ops, + .id_table = sdw_mockup_id, +}; +module_sdw_driver(sdw_mockup_sdw_driver); + +MODULE_DESCRIPTION("ASoC SDW mockup codec driver"); +MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index b504d63385b3..52d2c968b5c0 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -35,6 +35,9 @@ #include "tlv320aic31xx.h" +static int aic31xx_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data); + static const struct reg_default aic31xx_reg_defaults[] = { { AIC31XX_CLKMUX, 0x00 }, { AIC31XX_PLLPR, 0x11 }, @@ -1256,6 +1259,13 @@ static int aic31xx_power_on(struct snd_soc_component *component) return ret; } + /* + * The jack detection configuration is in the same register + * that is used to report jack detect status so is volatile + * and not covered by the cache sync, restore it separately. + */ + aic31xx_set_jack(component, aic31xx->jack, NULL); + return 0; } diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index dcd8aeb45cb3..2e9175b37dc9 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -682,11 +682,20 @@ static int aic32x4_set_dosr(struct snd_soc_component *component, u16 dosr) static int aic32x4_set_processing_blocks(struct snd_soc_component *component, u8 r_block, u8 p_block) { - if (r_block > 18 || p_block > 25) - return -EINVAL; + struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); + + if (aic32x4->type == AIC32X4_TYPE_TAS2505) { + if (r_block || p_block > 3) + return -EINVAL; - snd_soc_component_write(component, AIC32X4_ADCSPB, r_block); - snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + } else { /* AIC32x4 */ + if (r_block > 18 || p_block > 25) + return -EINVAL; + + snd_soc_component_write(component, AIC32X4_ADCSPB, r_block); + snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + } return 0; } @@ -695,6 +704,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, unsigned int sample_rate, unsigned int channels, unsigned int bit_depth) { + struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u8 aosr; u16 dosr; u8 adc_resource_class, dac_resource_class; @@ -721,19 +731,28 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, adc_resource_class = 6; dac_resource_class = 8; dosr_increment = 8; - aic32x4_set_processing_blocks(component, 1, 1); + if (aic32x4->type == AIC32X4_TYPE_TAS2505) + aic32x4_set_processing_blocks(component, 0, 1); + else + aic32x4_set_processing_blocks(component, 1, 1); } else if (sample_rate <= 96000) { aosr = 64; adc_resource_class = 6; dac_resource_class = 8; dosr_increment = 4; - aic32x4_set_processing_blocks(component, 1, 9); + if (aic32x4->type == AIC32X4_TYPE_TAS2505) + aic32x4_set_processing_blocks(component, 0, 1); + else + aic32x4_set_processing_blocks(component, 1, 9); } else if (sample_rate == 192000) { aosr = 32; adc_resource_class = 3; dac_resource_class = 4; dosr_increment = 2; - aic32x4_set_processing_blocks(component, 13, 19); + if (aic32x4->type == AIC32X4_TYPE_TAS2505) + aic32x4_set_processing_blocks(component, 0, 1); + else + aic32x4_set_processing_blocks(component, 13, 19); } else { dev_err(component->dev, "Sampling rate not supported\n"); return -EINVAL; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 549d98241dae..fe15cbc7bcaf 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -747,7 +747,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp) { wm_adsp_debugfs_clear(dsp); - debugfs_remove_recursive(dsp->debugfs_root); } #else static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp, diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 4124aa2fc247..905c7965f653 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -127,7 +127,7 @@ static void sst_fill_alloc_params(struct snd_pcm_substream *substream, snd_pcm_uframes_t period_size; ssize_t periodbytes; ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); - u32 buffer_addr = virt_to_phys(substream->dma_buffer.area); + u32 buffer_addr = virt_to_phys(substream->runtime->dma_area); channels = substream->runtime->channels; period_size = substream->runtime->period_size; @@ -233,7 +233,6 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); sst_fill_alloc_params(substream, &alloc_params); - substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; str_params.aparams = alloc_params; str_params.codec = SST_CODEC_TYPE_PCM; diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 7e29b0d911e2..046955bf717c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -602,6 +602,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_DMIC select SND_SOC_INTEL_HDA_DSP_COMMON select SND_SOC_INTEL_SOF_MAXIM_COMMON + select SND_SOC_SDW_MOCKUP help Add support for Intel SoundWire-based platforms connected to MAX98373, RT700, RT711, RT1308 and RT715 diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 896251d742fe..b7b3b0bf994a 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -404,7 +404,7 @@ static int audio_probe(struct platform_device *pdev) return -ENOMEM; /* By default dais[0] is configured for max98373 */ - if (!strcmp(pdev->name, "sof_da7219_max98360a")) { + if (!strcmp(pdev->name, "sof_da7219_mx98360a")) { dais[0] = (struct snd_soc_dai_link) { .name = "SSP1-Codec", .id = 0, diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1a867c73a48e..82d909ef7a97 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -328,6 +328,19 @@ static const struct snd_soc_ops sdw_ops = { .shutdown = sdw_shutdown, }; +static int sof_sdw_mic_codec_mockup_init(const struct snd_soc_acpi_link_adr *link, + struct snd_soc_dai_link *dai_links, + struct sof_sdw_codec_info *info, + bool playback) +{ + /* + * force DAI link to use same ID as RT715 and DMIC + * to reuse topologies + */ + dai_links->id = SDW_DMIC_DAI_ID; + return 0; +} + static struct sof_sdw_codec_info codec_info_list[] = { { .part_id = 0x700, @@ -410,6 +423,34 @@ static struct sof_sdw_codec_info codec_info_list[] = { .dai_name = "rt5682-sdw", .init = sof_sdw_rt5682_init, }, + { + .part_id = 0xaaaa, /* generic codec mockup */ + .version_id = 0, + .direction = {true, true}, + .dai_name = "sdw-mockup-aif1", + .init = NULL, + }, + { + .part_id = 0xaa55, /* headset codec mockup */ + .version_id = 0, + .direction = {true, true}, + .dai_name = "sdw-mockup-aif1", + .init = NULL, + }, + { + .part_id = 0x55aa, /* amplifier mockup */ + .version_id = 0, + .direction = {true, false}, + .dai_name = "sdw-mockup-aif1", + .init = NULL, + }, + { + .part_id = 0x5555, + .version_id = 0, + .direction = {false, true}, + .dai_name = "sdw-mockup-aif1", + .init = sof_sdw_mic_codec_mockup_init, + }, }; static inline int find_codec_info_part(u64 adr) diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 12a205ccdeeb..fef0b2d1de68 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -9,7 +9,8 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-cml-match.o soc-acpi-intel-icl-match.o \ soc-acpi-intel-tgl-match.o soc-acpi-intel-ehl-match.o \ soc-acpi-intel-jsl-match.o soc-acpi-intel-adl-match.o \ - soc-acpi-intel-hda-match.o + soc-acpi-intel-hda-match.o \ + soc-acpi-intel-sdw-mockup-match.o obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o snd-soc-sst-ipc.o obj-$(CONFIG_SND_SOC_ACPI_INTEL_MATCH) += snd-soc-acpi-intel-match.o diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index 39dad32564e6..94b650767e11 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -9,6 +9,7 @@ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include "../skylake/skl.h" +#include "soc-acpi-intel-sdw-mockup-match.h" static struct skl_machine_pdata cnl_pdata = { .use_tplg_pcm = true, @@ -60,6 +61,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[] = { .sof_fw_filename = "sof-cnl.ri", .sof_tplg_filename = "sof-cnl-rt5682-sdw2.tplg" }, + { + .link_mask = GENMASK(3, 0), + .links = sdw_mockup_headset_2amps_mic, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-rt711-rt1308-rt715.tplg", + }, + { + .link_mask = BIT(0) | BIT(1) | BIT(3), + .links = sdw_mockup_headset_1amp_mic, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-cnl.ri", + .sof_tplg_filename = "sof-cml-rt711-rt1308-mono-rt715.tplg", + }, {} }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_cnl_sdw_machines); diff --git a/sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.c b/sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.c new file mode 100644 index 000000000000..a3d33997736a --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.c @@ -0,0 +1,166 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// soc-acpi-intel-sdw-mockup-match.c - tables and support for SoundWire +// mockup device ACPI enumeration. +// +// Copyright (c) 2021, Intel Corporation. +// + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> +#include "soc-acpi-intel-sdw-mockup-match.h" + +static const struct snd_soc_acpi_endpoint sdw_mockup_single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_endpoint sdw_mockup_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint sdw_mockup_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1, +}; + +static const struct snd_soc_acpi_adr_device sdw_mockup_headset_0_adr[] = { + { + .adr = 0x0000000105AA5500ull, + .num_endpoints = 1, + .endpoints = &sdw_mockup_single_endpoint, + .name_prefix = "sdw_mockup_headset0" + } +}; + +static const struct snd_soc_acpi_adr_device sdw_mockup_headset_1_adr[] = { + { + .adr = 0x0001000105AA5500ull, + .num_endpoints = 1, + .endpoints = &sdw_mockup_single_endpoint, + .name_prefix = "sdw_mockup_headset1" + } +}; + +static const struct snd_soc_acpi_adr_device sdw_mockup_amp_1_adr[] = { + { + .adr = 0x000100010555AA00ull, + .num_endpoints = 1, + .endpoints = &sdw_mockup_single_endpoint, + .name_prefix = "sdw_mockup_amp1" + } +}; + +static const struct snd_soc_acpi_adr_device sdw_mockup_amp_2_adr[] = { + { + .adr = 0x000200010555AA00ull, + .num_endpoints = 1, + .endpoints = &sdw_mockup_single_endpoint, + .name_prefix = "sdw_mockup_amp2" + } +}; + +static const struct snd_soc_acpi_adr_device sdw_mockup_mic_0_adr[] = { + { + .adr = 0x0000000105555500ull, + .num_endpoints = 1, + .endpoints = &sdw_mockup_single_endpoint, + .name_prefix = "sdw_mockup_mic0" + } +}; + +static const struct snd_soc_acpi_adr_device sdw_mockup_mic_3_adr[] = { + { + .adr = 0x0003000105555500ull, + .num_endpoints = 1, + .endpoints = &sdw_mockup_single_endpoint, + .name_prefix = "sdw_mockup_mic3" + } +}; + +static const struct snd_soc_acpi_adr_device sdw_mockup_amp_1_group1_adr[] = { + { + .adr = 0x000100010555AA00ull, + .num_endpoints = 1, + .endpoints = &sdw_mockup_l_endpoint, + .name_prefix = "sdw_mockup_amp1_l" + } +}; + +static const struct snd_soc_acpi_adr_device sdw_mockup_amp_2_group1_adr[] = { + { + .adr = 0x000200010555AA00ull, + .num_endpoints = 1, + .endpoints = &sdw_mockup_r_endpoint, + .name_prefix = "sdw_mockup_amp2_r" + } +}; + +const struct snd_soc_acpi_link_adr sdw_mockup_headset_1amp_mic[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(sdw_mockup_headset_0_adr), + .adr_d = sdw_mockup_headset_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(sdw_mockup_amp_1_adr), + .adr_d = sdw_mockup_amp_1_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(sdw_mockup_mic_3_adr), + .adr_d = sdw_mockup_mic_3_adr, + }, + {} +}; + +const struct snd_soc_acpi_link_adr sdw_mockup_headset_2amps_mic[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(sdw_mockup_headset_0_adr), + .adr_d = sdw_mockup_headset_0_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(sdw_mockup_amp_1_group1_adr), + .adr_d = sdw_mockup_amp_1_group1_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(sdw_mockup_amp_2_group1_adr), + .adr_d = sdw_mockup_amp_2_group1_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(sdw_mockup_mic_3_adr), + .adr_d = sdw_mockup_mic_3_adr, + }, + {} +}; + +const struct snd_soc_acpi_link_adr sdw_mockup_mic_headset_1amp[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(sdw_mockup_headset_1_adr), + .adr_d = sdw_mockup_headset_1_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(sdw_mockup_amp_2_adr), + .adr_d = sdw_mockup_amp_2_adr, + }, + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(sdw_mockup_mic_0_adr), + .adr_d = sdw_mockup_mic_0_adr, + }, + {} +}; diff --git a/sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.h b/sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.h new file mode 100644 index 000000000000..c99eecd19e03 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.h @@ -0,0 +1,17 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * soc-acpi-intel-sdw-mockup-match.h - tables and support for SoundWire + * mockup device ACPI enumeration. + * + * Copyright (c) 2021, Intel Corporation. + * + */ + +#ifndef _SND_SOC_ACPI_INTEL_SDW_MOCKUP_MATCH +#define _SND_SOC_ACPI_INTEL_SDW_MOCKUP_MATCH + +extern const struct snd_soc_acpi_link_adr sdw_mockup_headset_1amp_mic[]; +extern const struct snd_soc_acpi_link_adr sdw_mockup_headset_2amps_mic[]; +extern const struct snd_soc_acpi_link_adr sdw_mockup_mic_headset_1amp[]; + +#endif diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 66595e3ab13f..e2488f0eaff8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -8,6 +8,7 @@ #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> +#include "soc-acpi-intel-sdw-mockup-match.h" static const struct snd_soc_acpi_codecs tgl_codecs = { .num_codecs = 1, @@ -351,6 +352,28 @@ EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_tgl_machines); /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[] = { + /* mockup tests need to be first */ + { + .link_mask = GENMASK(3, 0), + .links = sdw_mockup_headset_2amps_mic, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt711-rt1308-rt715.tplg", + }, + { + .link_mask = BIT(0) | BIT(1) | BIT(3), + .links = sdw_mockup_headset_1amp_mic, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt711-rt1308-mono-rt715.tplg", + }, + { + .link_mask = BIT(0) | BIT(1) | BIT(2), + .links = sdw_mockup_mic_headset_1amp, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-tgl.ri", + .sof_tplg_filename = "sof-tgl-rt715-rt711-rt1308-mono.tplg", + }, { .link_mask = 0x7, .links = tgl_sdw_rt711_link1_rt1308_link2_rt715_link0, diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index c2a5933bfcfc..700a18561a94 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -104,8 +104,6 @@ static int kirkwood_dma_open(struct snd_soc_component *component, int err; struct snd_pcm_runtime *runtime = substream->runtime; struct kirkwood_dma_data *priv = kirkwood_priv(substream); - const struct mbus_dram_target_info *dram; - unsigned long addr; snd_soc_set_runtime_hwparams(substream, &kirkwood_dma_snd_hw); @@ -142,20 +140,14 @@ static int kirkwood_dma_open(struct snd_soc_component *component, writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); } - dram = mv_mbus_dram_info(); - addr = substream->dma_buffer.addr; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (priv->substream_play) return -EBUSY; priv->substream_play = substream; - kirkwood_dma_conf_mbus_windows(priv->io, - KIRKWOOD_PLAYBACK_WIN, addr, dram); } else { if (priv->substream_rec) return -EBUSY; priv->substream_rec = substream; - kirkwood_dma_conf_mbus_windows(priv->io, - KIRKWOOD_RECORD_WIN, addr, dram); } return 0; @@ -182,6 +174,23 @@ static int kirkwood_dma_close(struct snd_soc_component *component, return 0; } +static int kirkwood_dma_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct kirkwood_dma_data *priv = kirkwood_priv(substream); + const struct mbus_dram_target_info *dram = mv_mbus_dram_info(); + unsigned long addr = substream->runtime->dma_addr; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + kirkwood_dma_conf_mbus_windows(priv->io, + KIRKWOOD_PLAYBACK_WIN, addr, dram); + else + kirkwood_dma_conf_mbus_windows(priv->io, + KIRKWOOD_RECORD_WIN, addr, dram); + return 0; +} + static int kirkwood_dma_prepare(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -246,6 +255,7 @@ const struct snd_soc_component_driver kirkwood_soc_component = { .name = DRV_NAME, .open = kirkwood_dma_open, .close = kirkwood_dma_close, + .hw_params = kirkwood_dma_hw_params, .prepare = kirkwood_dma_prepare, .pointer = kirkwood_dma_pointer, .pcm_construct = kirkwood_dma_new, diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 3a5e84e16a87..c8dfd0de30e4 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -148,86 +148,75 @@ int snd_soc_component_set_bias_level(struct snd_soc_component *component, return soc_component_ret(component, ret); } -static int soc_component_pin(struct snd_soc_component *component, - const char *pin, - int (*pin_func)(struct snd_soc_dapm_context *dapm, - const char *pin)) -{ - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); - char *full_name; - int ret; - - if (!component->name_prefix) { - ret = pin_func(dapm, pin); - goto end; - } - - full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin); - if (!full_name) { - ret = -ENOMEM; - goto end; - } - - ret = pin_func(dapm, full_name); - kfree(full_name); -end: - return soc_component_ret(component, ret); -} - int snd_soc_component_enable_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_enable_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_enable_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin); int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_enable_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_enable_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin_unlocked); int snd_soc_component_disable_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_disable_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_disable_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin); int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_disable_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_disable_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin_unlocked); int snd_soc_component_nc_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_nc_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_nc_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin); int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_nc_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_nc_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin_unlocked); int snd_soc_component_get_pin_status(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_get_pin_status); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_get_pin_status(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_get_pin_status); int snd_soc_component_force_enable_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_force_enable_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin); @@ -235,7 +224,9 @@ int snd_soc_component_force_enable_pin_unlocked( struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_force_enable_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin_unlocked); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 4bce89b5ea40..4447f515e8b1 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -278,6 +278,8 @@ config SND_SOC_SOF_HDA config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE tristate + select SOUNDWIRE_INTEL if SND_SOC_SOF_INTEL_SOUNDWIRE + select SND_INTEL_SOUNDWIRE_ACPI if SND_SOC_SOF_INTEL_SOUNDWIRE config SND_SOC_SOF_INTEL_SOUNDWIRE tristate "SOF support for SoundWire" @@ -285,8 +287,6 @@ config SND_SOC_SOF_INTEL_SOUNDWIRE depends on SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE depends on ACPI && SOUNDWIRE depends on !(SOUNDWIRE=m && SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE=y) - select SOUNDWIRE_INTEL - select SND_INTEL_SOUNDWIRE_ACPI help This adds support for SoundWire with Sound Open Firmware for Intel(R) platforms. diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index c91aa951df22..acfeca42604c 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -107,8 +107,8 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) } else { /* reply correct size ? */ if (reply.hdr.size != msg->reply_size && - /* getter payload is never known upfront */ - !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { + /* getter payload is never known upfront */ + ((reply.hdr.cmd & SOF_GLB_TYPE_MASK) != SOF_IPC_GLB_PROBE)) { dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", msg->reply_size, reply.hdr.size); ret = -EINVAL; diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index e1e368ff2b12..891e6e1b9121 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -187,12 +187,16 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) int hda_sdw_startup(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hdev; + struct snd_sof_pdata *pdata = sdev->pdata; hdev = sdev->pdata->hw_pdata; if (!hdev->sdw) return 0; + if (pdata->machine && !pdata->machine->mach_params.link_mask) + return 0; + return sdw_intel_startup(hdev->sdw); } @@ -1002,6 +1006,14 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) hda_mach->mach_params.dmic_num = dmic_num; pdata->machine = hda_mach; pdata->tplg_filename = tplg_filename; + + if (codec_num == 2) { + /* + * Prevent SoundWire links from starting when an external + * HDaudio codec is used + */ + hda_mach->mach_params.link_mask = 0; + } } } diff --git a/sound/soc/uniphier/aio-dma.c b/sound/soc/uniphier/aio-dma.c index 3c1628a3a1ac..3d9736e7381f 100644 --- a/sound/soc/uniphier/aio-dma.c +++ b/sound/soc/uniphier/aio-dma.c @@ -198,7 +198,7 @@ static int uniphier_aiodma_mmap(struct snd_soc_component *component, vma->vm_page_prot = pgprot_writecombine(vma->vm_page_prot); return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, + substream->runtime->dma_addr >> PAGE_SHIFT, vma->vm_end - vma->vm_start, vma->vm_page_prot); } diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 1d59fb668c77..91afea9d5de6 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -452,8 +452,8 @@ static int xlnx_formatter_pcm_hw_params(struct snd_soc_component *component, stream_data->buffer_size = size; - low = lower_32_bits(substream->dma_buffer.addr); - high = upper_32_bits(substream->dma_buffer.addr); + low = lower_32_bits(runtime->dma_addr); + high = upper_32_bits(runtime->dma_addr); writel(low, stream_data->mmio + XLNX_AUD_BUFF_ADDR_LSB); writel(high, stream_data->mmio + XLNX_AUD_BUFF_ADDR_MSB); |