diff options
Diffstat (limited to 'sound')
45 files changed, 474 insertions, 270 deletions
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 83b79edfa52d..439a358ecfe9 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -215,7 +215,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { return remap_pfn_range(area, area->vm_start, - dmab->addr >> PAGE_SHIFT, + page_to_pfn(virt_to_page(dmab->area)), area->vm_end - area->vm_start, area->vm_page_prot); } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 14e32825c339..71323d807dbf 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -246,12 +246,21 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - if (substream->ops->mmap || - (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV && - substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC)) + if (substream->ops->mmap || substream->ops->page) return true; - return dma_can_mmap(substream->dma_buffer.dev.dev); + switch (substream->dma_buffer.dev.type) { + case SNDRV_DMA_TYPE_UNKNOWN: + /* 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; + default: + return dma_can_mmap(substream->dma_buffer.dev.dev); + } } static int constrain_mask_params(struct snd_pcm_substream *substream, @@ -3063,9 +3072,14 @@ static int snd_pcm_ioctl_sync_ptr_compat(struct snd_pcm_substream *substream, boundary = 0x7fffffff; snd_pcm_stream_lock_irq(substream); /* FIXME: we should consider the boundary for the sync from app */ - if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) - control->appl_ptr = scontrol.appl_ptr; - else + if (!(sflags & SNDRV_PCM_SYNC_PTR_APPL)) { + err = pcm_lib_apply_appl_ptr(substream, + scontrol.appl_ptr); + if (err < 0) { + snd_pcm_stream_unlock_irq(substream); + return err; + } + } else scontrol.appl_ptr = control->appl_ptr % boundary; if (!(sflags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN)) control->avail_min = scontrol.avail_min; @@ -3664,6 +3678,8 @@ static vm_fault_t snd_pcm_mmap_data_fault(struct vm_fault *vmf) return VM_FAULT_SIGBUS; if (substream->ops->page) page = substream->ops->page(substream, offset); + else if (!snd_pcm_get_dma_buf(substream)) + page = virt_to_page(runtime->dma_area + offset); else page = snd_sgbuf_get_page(snd_pcm_get_dma_buf(substream), offset); if (!page) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index b9c2ce2b8d5a..84d78630463e 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -514,10 +514,11 @@ static int check_and_subscribe_port(struct snd_seq_client *client, return err; } -static void delete_and_unsubscribe_port(struct snd_seq_client *client, - struct snd_seq_client_port *port, - struct snd_seq_subscribers *subs, - bool is_src, bool ack) +/* called with grp->list_mutex held */ +static void __delete_and_unsubscribe_port(struct snd_seq_client *client, + struct snd_seq_client_port *port, + struct snd_seq_subscribers *subs, + bool is_src, bool ack) { struct snd_seq_port_subs_info *grp; struct list_head *list; @@ -525,7 +526,6 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, grp = is_src ? &port->c_src : &port->c_dest; list = is_src ? &subs->src_list : &subs->dest_list; - down_write(&grp->list_mutex); write_lock_irq(&grp->list_lock); empty = list_empty(list); if (!empty) @@ -535,6 +535,18 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, if (!empty) unsubscribe_port(client, port, grp, &subs->info, ack); +} + +static void delete_and_unsubscribe_port(struct snd_seq_client *client, + struct snd_seq_client_port *port, + struct snd_seq_subscribers *subs, + bool is_src, bool ack) +{ + struct snd_seq_port_subs_info *grp; + + grp = is_src ? &port->c_src : &port->c_dest; + down_write(&grp->list_mutex); + __delete_and_unsubscribe_port(client, port, subs, is_src, ack); up_write(&grp->list_mutex); } @@ -590,27 +602,30 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, struct snd_seq_client_port *dest_port, struct snd_seq_port_subscribe *info) { - struct snd_seq_port_subs_info *src = &src_port->c_src; + struct snd_seq_port_subs_info *dest = &dest_port->c_dest; struct snd_seq_subscribers *subs; int err = -ENOENT; - down_write(&src->list_mutex); + /* always start from deleting the dest port for avoiding concurrent + * deletions + */ + down_write(&dest->list_mutex); /* look for the connection */ - list_for_each_entry(subs, &src->list_head, src_list) { + list_for_each_entry(subs, &dest->list_head, dest_list) { if (match_subs_info(info, &subs->info)) { - atomic_dec(&subs->ref_count); /* mark as not ready */ + __delete_and_unsubscribe_port(dest_client, dest_port, + subs, false, + connector->number != dest_client->number); err = 0; break; } } - up_write(&src->list_mutex); + up_write(&dest->list_mutex); if (err < 0) return err; delete_and_unsubscribe_port(src_client, src_port, subs, true, connector->number != src_client->number); - delete_and_unsubscribe_port(dest_client, dest_port, subs, false, - connector->number != dest_client->number); kfree(subs); return 0; } diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index d8be146793ee..c9d0ba353463 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -319,6 +319,10 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, .device = 0x4b55, }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b58, + }, #endif /* Alder Lake */ diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 5bbe6695689d..7ad8c5f7b664 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -816,6 +816,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7); + spin_unlock_irqrestore(&p->chip->mixer_lock, flags); spin_lock(&p->chip->reg_lock); set_mode_register(p->chip, 0xc0); /* c0 = STOP */ @@ -855,6 +856,7 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel spin_unlock(&p->chip->reg_lock); /* restore PCM volume */ + spin_lock_irqsave(&p->chip->mixer_lock, flags); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR); spin_unlock_irqrestore(&p->chip->mixer_lock, flags); @@ -880,6 +882,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p) mixR = snd_sbmixer_read(p->chip, SB_DSP4_PCM_DEV + 1); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL & 0x7); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR & 0x7); + spin_unlock_irqrestore(&p->chip->mixer_lock, flags); spin_lock(&p->chip->reg_lock); if (p->running & SNDRV_SB_CSP_ST_QSOUND) { @@ -894,6 +897,7 @@ static int snd_sb_csp_stop(struct snd_sb_csp * p) spin_unlock(&p->chip->reg_lock); /* restore PCM volume */ + spin_lock_irqsave(&p->chip->mixer_lock, flags); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV, mixL); snd_sbmixer_write(p->chip, SB_DSP4_PCM_DEV + 1, mixR); spin_unlock_irqrestore(&p->chip->mixer_lock, flags); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 4b2cc8cb55c4..e143e69d8184 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1940,6 +1940,8 @@ static int hdmi_add_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) static const struct snd_pci_quirk force_connect_list[] = { SND_PCI_QUIRK(0x103c, 0x870f, "HP", 1), SND_PCI_QUIRK(0x103c, 0x871a, "HP", 1), + SND_PCI_QUIRK(0x1462, 0xec94, "MS-7C94", 1), + SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", 1), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1389cfd5e0db..a065260d0d20 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8274,9 +8274,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), @@ -8429,6 +8431,7 @@ 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), @@ -8463,6 +8466,7 @@ 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), @@ -8626,6 +8630,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), 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 84e3906abd4f..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), }, @@ -576,6 +581,7 @@ static struct snd_soc_dai_link cz_dai_5682_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_rt5682_init, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_rt5682_play_ops, SND_SOC_DAILINK_REG(designware1, rt5682, platform), }, @@ -585,6 +591,7 @@ static struct snd_soc_dai_link cz_dai_5682_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_rt5682_cap_ops, SND_SOC_DAILINK_REG(designware2, rt5682, platform), }, @@ -594,6 +601,7 @@ static struct snd_soc_dai_link cz_dai_5682_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_rt5682_max_play_ops, SND_SOC_DAILINK_REG(designware3, mx, platform), }, @@ -604,6 +612,7 @@ static struct snd_soc_dai_link cz_dai_5682_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_rt5682_dmic0_cap_ops, SND_SOC_DAILINK_REG(designware3, adau, platform), }, @@ -614,6 +623,7 @@ static struct snd_soc_dai_link cz_dai_5682_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_rt5682_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 7ebae3f09435..db16071205ba 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1325,7 +1325,7 @@ config SND_SOC_SSM2305 high-efficiency mono Class-D audio power amplifiers. config SND_SOC_SSM2518 - tristate + tristate "Analog Devices SSM2518 Class-D Amplifier" depends on I2C config SND_SOC_SSM2602 @@ -1557,7 +1557,9 @@ config SND_SOC_WCD934X Qualcomm SoCs like SDM845. 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" @@ -1813,11 +1815,6 @@ config SND_SOC_ZL38060 which consists of a Digital Signal Processor (DSP), several Digital Audio Interfaces (DAIs), analog outputs, and a block of 14 GPIOs. -config SND_SOC_ZX_AUD96P22 - tristate "ZTE ZX AUD96P22 CODEC" - depends on I2C - select REGMAP_I2C - # Amp config SND_SOC_LM4857 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index de8b83dd2c76..7bb38c370842 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -583,7 +583,10 @@ 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/rt5631.c b/sound/soc/codecs/rt5631.c index 3000bc128b5b..38356ea2bd6e 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1695,6 +1695,8 @@ static const struct regmap_config rt5631_regmap_config = { .reg_defaults = rt5631_reg, .num_reg_defaults = ARRAY_SIZE(rt5631_reg), .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, }; static int rt5631_i2c_probe(struct i2c_client *i2c, diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e4c91571abae..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) @@ -973,10 +974,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); - if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 51870d50f419..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; } @@ -1604,6 +1614,8 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, ret); return ret; } + regcache_cache_only(aic31xx->regmap, true); + aic31xx->dev = &i2c->dev; aic31xx->irq = i2c->irq; diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 81952984613d..2513922a0292 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c63b717040ed..2e9175b37dc9 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -250,8 +250,8 @@ static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0); static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0); /* -12dB min, 0.5dB steps */ static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0); - -static DECLARE_TLV_DB_LINEAR(tlv_spk_vol, TLV_DB_GAIN_MUTE, 0); +/* -6dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_tas_driver_gain, -5850, 50, 0); static DECLARE_TLV_DB_SCALE(tlv_amp_vol, 0, 600, 1); static const char * const lo_cm_text[] = { @@ -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; @@ -1063,21 +1082,20 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { }; static const struct snd_kcontrol_new aic32x4_tas2505_snd_controls[] = { - SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL, - AIC32X4_LDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm), + SOC_SINGLE_S8_TLV("PCM Playback Volume", + AIC32X4_LDACVOL, -0x7f, 0x30, tlv_pcm), SOC_ENUM("DAC Playback PowerTune Switch", l_ptm_enum), - SOC_DOUBLE_R_S_TLV("HP Driver Playback Volume", AIC32X4_HPLGAIN, - AIC32X4_HPLGAIN, 0, -0x6, 0x1d, 5, 0, - tlv_driver_gain), - SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, - AIC32X4_HPLGAIN, 6, 0x01, 1), - SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), + SOC_SINGLE_TLV("HP Driver Gain Volume", + AIC32X4_HPLGAIN, 0, 0x74, 1, tlv_tas_driver_gain), + SOC_SINGLE("HP DAC Playback Switch", AIC32X4_HPLGAIN, 6, 1, 1), - SOC_SINGLE_RANGE_TLV("Speaker Driver Playback Volume", TAS2505_SPKVOL1, - 0, 0, 117, 1, tlv_spk_vol), - SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", TAS2505_SPKVOL2, - 4, 5, 0, tlv_amp_vol), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2505_SPKVOL1, 0, 0x74, 1, tlv_tas_driver_gain), + SOC_SINGLE_TLV("Speaker Amplifier Playback Volume", + TAS2505_SPKVOL2, 4, 5, 0, tlv_amp_vol), + + SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0), }; static const struct snd_kcontrol_new hp_output_mixer_controls[] = { diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 78b76eceff8f..2fcc97370be2 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3317,13 +3317,6 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component) (WCD938X_DIGITAL_INTR_LEVEL_0 + i), 0); } - ret = wcd938x_irq_init(wcd938x, component->dev); - if (ret) { - dev_err(component->dev, "%s: IRQ init failed: %d\n", - __func__, ret); - return ret; - } - wcd938x->hphr_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, WCD938X_IRQ_HPHR_PDM_WD_INT); wcd938x->hphl_pdm_wd_int = regmap_irq_get_virq(wcd938x->irq_chip, @@ -3553,7 +3546,6 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_PB] = dev_get_drvdata(wcd938x->rxdev); wcd938x->sdw_priv[AIF1_PB]->wcd938x = wcd938x; - wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; wcd938x->txdev = wcd938x_sdw_device_get(wcd938x->txnode); if (!wcd938x->txdev) { @@ -3562,7 +3554,6 @@ static int wcd938x_bind(struct device *dev) } wcd938x->sdw_priv[AIF1_CAP] = dev_get_drvdata(wcd938x->txdev); wcd938x->sdw_priv[AIF1_CAP]->wcd938x = wcd938x; - wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; wcd938x->tx_sdw_dev = dev_to_sdw_dev(wcd938x->txdev); if (!wcd938x->tx_sdw_dev) { dev_err(dev, "could not get txslave with matching of dev\n"); @@ -3595,6 +3586,15 @@ static int wcd938x_bind(struct device *dev) return PTR_ERR(wcd938x->regmap); } + ret = wcd938x_irq_init(wcd938x, dev); + if (ret) { + dev_err(dev, "%s: IRQ init failed: %d\n", __func__, ret); + return ret; + } + + wcd938x->sdw_priv[AIF1_PB]->slave_irq = wcd938x->virq; + wcd938x->sdw_priv[AIF1_CAP]->slave_irq = wcd938x->virq; + ret = wcd938x_set_micbias_data(wcd938x); if (ret < 0) { dev_err(dev, "%s: bad micbias pdata\n", __func__); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 37aa020f23f6..fe15cbc7bcaf 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -282,6 +282,7 @@ /* * HALO_CCM_CORE_CONTROL */ +#define HALO_CORE_RESET 0x00000200 #define HALO_CORE_EN 0x00000001 /* @@ -746,7 +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, @@ -1213,7 +1213,7 @@ static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, mutex_lock(&ctl->dsp->pwr_lock); - ret = wm_coeff_read_ctrl_raw(ctl, ctl->cache, size); + ret = wm_coeff_read_ctrl(ctl, ctl->cache, size); if (!ret && copy_to_user(bytes, ctl->cache, size)) ret = -EFAULT; @@ -3333,7 +3333,8 @@ static int wm_halo_start_core(struct wm_adsp *dsp) { return regmap_update_bits(dsp->regmap, dsp->base + HALO_CCM_CORE_CONTROL, - HALO_CORE_EN, HALO_CORE_EN); + HALO_CORE_RESET | HALO_CORE_EN, + HALO_CORE_RESET | HALO_CORE_EN); } static void wm_halo_stop_core(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..5db2f4865bbb 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 = substream->runtime->dma_addr; 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/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_max98373.c b/sound/soc/intel/boards/sof_sdw_max98373.c index 0e7ed906b341..25daef910aee 100644 --- a/sound/soc/intel/boards/sof_sdw_max98373.c +++ b/sound/soc/intel/boards/sof_sdw_max98373.c @@ -55,43 +55,68 @@ static int spk_init(struct snd_soc_pcm_runtime *rtd) return ret; } -static int max98373_sdw_trigger(struct snd_pcm_substream *substream, int cmd) +static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int ret; + int j; - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - /* enable max98373 first */ - ret = max_98373_trigger(substream, cmd); - if (ret < 0) - break; - - ret = sdw_trigger(substream, cmd); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = sdw_trigger(substream, cmd); - if (ret < 0) - break; - - ret = max_98373_trigger(substream, cmd); - break; - default: - ret = -EINVAL; - break; + /* set spk pin by playback only */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + for_each_rtd_codec_dais(rtd, j, codec_dai) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(cpu_dai->component); + char pin_name[16]; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + if (enable) + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + else + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + + if (!ret) + snd_soc_dapm_sync(dapm); } - return ret; + return 0; +} + +static int mx8373_sdw_prepare(struct snd_pcm_substream *substream) +{ + int ret = 0; + + /* according to soc_pcm_prepare dai link prepare is called first */ + ret = sdw_prepare(substream); + if (ret < 0) + return ret; + + return mx8373_enable_spk_pin(substream, true); +} + +static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream) +{ + int ret = 0; + + /* according to soc_pcm_hw_free dai link free is called first */ + ret = sdw_hw_free(substream); + if (ret < 0) + return ret; + + return mx8373_enable_spk_pin(substream, false); } static const struct snd_soc_ops max_98373_sdw_ops = { .startup = sdw_startup, - .prepare = sdw_prepare, - .trigger = max98373_sdw_trigger, - .hw_free = sdw_hw_free, + .prepare = mx8373_sdw_prepare, + .trigger = sdw_trigger, + .hw_free = mx8373_sdw_hw_free, .shutdown = sdw_shutdown, }; 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/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c index 2b758a18c2ea..5b8a274419ed 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c @@ -341,6 +341,7 @@ static int set_mtkaif_rx(struct mtk_base_afe *afe) case MT8183_MTKAIF_PROTOCOL_1: regmap_write(afe->regmap, AFE_AUD_PAD_TOP, 0x31); regmap_write(afe->regmap, AFE_ADDA_MTKAIF_CFG0, 0x0); + break; default: break; } 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/soc-pcm.c b/sound/soc/soc-pcm.c index 46513bb97904..d1c570ca21ea 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1015,6 +1015,7 @@ out: static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); int ret = -EINVAL, _ret = 0; int rollback = 0; @@ -1055,14 +1056,23 @@ start_err: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); - if (ret < 0) - break; + if (rtd->dai_link->stop_dma_first) { + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; - ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); - if (ret < 0) - break; + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } else { + ret = snd_soc_pcm_dai_trigger(substream, cmd, rollback); + if (ret < 0) + break; + ret = snd_soc_pcm_component_trigger(substream, cmd, rollback); + if (ret < 0) + break; + } ret = snd_soc_link_trigger(substream, cmd, rollback); break; } 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/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index a00262184efa..d04ce84fe7cc 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -89,6 +89,7 @@ static const struct sof_dev_desc adls_desc = { static const struct sof_dev_desc adl_desc = { .machines = snd_soc_acpi_intel_adl_machines, .alt_machines = snd_soc_acpi_intel_adl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 573374b89b10..d3276b4595af 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -213,19 +213,19 @@ snd_pcm_uframes_t tegra_pcm_pointer(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(tegra_pcm_pointer); -static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, +static int tegra_pcm_preallocate_dma_buffer(struct device *dev, struct snd_pcm *pcm, int stream, size_t size) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - buf->area = dma_alloc_wc(pcm->card->dev, size, &buf->addr, GFP_KERNEL); + buf->area = dma_alloc_wc(dev, size, &buf->addr, GFP_KERNEL); if (!buf->area) return -ENOMEM; buf->private_data = NULL; buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; + buf->dev.dev = dev; buf->bytes = size; return 0; @@ -244,31 +244,28 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) if (!buf->area) return; - dma_free_wc(pcm->card->dev, buf->bytes, buf->area, buf->addr); + dma_free_wc(buf->dev.dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; } -static int tegra_pcm_dma_allocate(struct snd_soc_pcm_runtime *rtd, +static int tegra_pcm_dma_allocate(struct device *dev, struct snd_soc_pcm_runtime *rtd, size_t size) { - struct snd_card *card = rtd->card->snd_card; struct snd_pcm *pcm = rtd->pcm; int ret; - ret = dma_set_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + ret = dma_set_mask_and_coherent(dev, DMA_BIT_MASK(32)); if (ret < 0) return ret; if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK, size); + ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_PLAYBACK, size); if (ret) goto err; } if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = tegra_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE, size); + ret = tegra_pcm_preallocate_dma_buffer(dev, pcm, SNDRV_PCM_STREAM_CAPTURE, size); if (ret) goto err_free_play; } @@ -284,7 +281,16 @@ err: int tegra_pcm_construct(struct snd_soc_component *component, struct snd_soc_pcm_runtime *rtd) { - return tegra_pcm_dma_allocate(rtd, tegra_pcm_hardware.buffer_bytes_max); + struct device *dev = component->dev; + + /* + * Fallback for backwards-compatibility with older device trees that + * have the iommus property in the virtual, top-level "sound" node. + */ + if (!of_get_property(dev->of_node, "iommus", NULL)) + dev = rtd->card->snd_card->dev; + + return tegra_pcm_dma_allocate(dev, rtd, tegra_pcm_hardware.buffer_bytes_max); } EXPORT_SYMBOL_GPL(tegra_pcm_construct); diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index a7c0484d44ec..265bbc5a2f96 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv, return ret; } - if (priv->hsdiv_rates[domain->parent_clk_id] != scki) { + if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) { dev_dbg(priv->dev, "%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n", audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI", @@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream) j721e_rule_rate, &priv->rate_range, SNDRV_PCM_HW_PARAM_RATE, -1); - mutex_unlock(&priv->mutex); if (ret) - return ret; + goto out; /* Reset TDM slots to 32 */ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; } - return 0; + if (ret == -ENOTSUPP) + ret = 0; +out: + if (ret) + domain->active--; + mutex_unlock(&priv->mutex); + + return ret; } static int j721e_audio_hw_params(struct snd_pcm_substream *substream, 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); diff --git a/sound/usb/card.c b/sound/usb/card.c index 2f6a62416c05..a1f8c3a026f5 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -907,7 +907,7 @@ static void usb_audio_disconnect(struct usb_interface *intf) } } - if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND) + if (chip->quirk_type == QUIRK_SETUP_DISABLE_AUTOSUSPEND) usb_enable_autosuspend(interface_to_usbdev(intf)); chip->num_interfaces--; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 52de52288e10..14456f61539e 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -324,6 +324,12 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, sources[ret - 1], visited, validate); if (ret > 0) { + /* + * For Samsung USBC Headset (AKG), setting clock selector again + * will result in incorrect default clock setting problems + */ + if (chip->usb_id == USB_ID(0x04e8, 0xa051)) + return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); if (err < 0) return err; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 30b3e128e28d..9b713b4a5ec4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1816,6 +1816,15 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer, strlcat(name, " - Output Jack", name_size); } +/* get connector value to "wake up" the USB audio */ +static int connector_mixer_resume(struct usb_mixer_elem_list *list) +{ + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); + + get_connector_value(cval, NULL, NULL); + return 0; +} + /* Build a mixer control for a UAC connector control (jack-detect) */ static void build_connector_control(struct usb_mixer_interface *mixer, const struct usbmix_name_map *imap, @@ -1833,6 +1842,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer, if (!cval) return; snd_usb_mixer_elem_init_std(&cval->head, mixer, term->id); + + /* set up a specific resume callback */ + cval->head.resume = connector_mixer_resume; + /* * UAC2: The first byte from reading the UAC2_TE_CONNECTOR control returns the * number of channels connected. @@ -3295,7 +3308,15 @@ static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, { struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); static const char * const val_types[] = { - "BOOLEAN", "INV_BOOLEAN", "S8", "U8", "S16", "U16", "S32", "U32", + [USB_MIXER_BOOLEAN] = "BOOLEAN", + [USB_MIXER_INV_BOOLEAN] = "INV_BOOLEAN", + [USB_MIXER_S8] = "S8", + [USB_MIXER_U8] = "U8", + [USB_MIXER_S16] = "S16", + [USB_MIXER_U16] = "U16", + [USB_MIXER_S32] = "S32", + [USB_MIXER_U32] = "U32", + [USB_MIXER_BESPOKEN] = "BESPOKEN", }; snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " "channels=%i, type=\"%s\"\n", cval->head.id, @@ -3634,23 +3655,15 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_resume(struct usb_mixer_elem_list *list) -{ - struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); - - /* get connector value to "wake up" the USB audio */ - if (cval->val_type == USB_MIXER_BOOLEAN && cval->channels == 1) - get_connector_value(cval, NULL, NULL); - - return 0; -} - static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) { - int err = default_mixer_resume(list); + int err; - if (err < 0) - return err; + if (list->resume) { + err = list->resume(list); + if (err < 0) + return err; + } return restore_mixer_value(list); } @@ -3689,7 +3702,7 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = default_mixer_resume; + list->resume = NULL; list->reset_resume = default_mixer_reset_resume; #endif } diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index f9d698a37153..3d5848d5481b 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -228,7 +228,7 @@ enum { }; static const char *const scarlett2_dim_mute_names[SCARLETT2_DIM_MUTE_COUNT] = { - "Mute", "Dim" + "Mute Playback Switch", "Dim Playback Switch" }; /* Description of each hardware port type: @@ -1856,9 +1856,15 @@ static int scarlett2_mute_ctl_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *elem = kctl->private_data; - struct scarlett2_data *private = elem->head.mixer->private_data; + struct usb_mixer_interface *mixer = elem->head.mixer; + struct scarlett2_data *private = mixer->private_data; int index = line_out_remap(private, elem->control); + mutex_lock(&private->data_mutex); + if (private->vol_updated) + scarlett2_update_volumes(mixer); + mutex_unlock(&private->data_mutex); + ucontrol->value.integer.value[0] = private->mute_switch[index]; return 0; } @@ -1955,10 +1961,12 @@ static void scarlett2_vol_ctl_set_writable(struct usb_mixer_interface *mixer, ~SNDRV_CTL_ELEM_ACCESS_WRITE; } - /* Notify of write bit change */ - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + /* Notify of write bit and possible value change */ + snd_ctl_notify(card, + SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &private->vol_ctls[index]->id); - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + snd_ctl_notify(card, + SNDRV_CTL_EVENT_MASK_VALUE | SNDRV_CTL_EVENT_MASK_INFO, &private->mute_ctls[index]->id); } @@ -2530,14 +2538,18 @@ static int scarlett2_add_direct_monitor_ctl(struct usb_mixer_interface *mixer) { struct scarlett2_data *private = mixer->private_data; const struct scarlett2_device_info *info = private->info; + const char *s; if (!info->direct_monitor) return 0; + s = info->direct_monitor == 1 + ? "Direct Monitor Playback Switch" + : "Direct Monitor Playback Enum"; + return scarlett2_add_new_ctl( mixer, &scarlett2_direct_monitor_ctl[info->direct_monitor - 1], - 0, 1, "Direct Monitor Playback Switch", - &private->direct_monitor_ctl); + 0, 1, s, &private->direct_monitor_ctl); } /*** Speaker Switching Control ***/ @@ -2589,7 +2601,9 @@ static int scarlett2_speaker_switch_enable(struct usb_mixer_interface *mixer) /* disable the line out SW/HW switch */ scarlett2_sw_hw_ctl_ro(private, i); - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, + snd_ctl_notify(card, + SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &private->sw_hw_ctls[i]->id); } @@ -2913,7 +2927,7 @@ static int scarlett2_dim_mute_ctl_put(struct snd_kcontrol *kctl, if (private->vol_sw_hw_switch[line_index]) { private->mute_switch[line_index] = val; snd_ctl_notify(mixer->chip->card, - SNDRV_CTL_EVENT_MASK_INFO, + SNDRV_CTL_EVENT_MASK_VALUE, &private->mute_ctls[i]->id); } } @@ -3455,7 +3469,7 @@ static int scarlett2_add_msd_ctl(struct usb_mixer_interface *mixer) /* Add MSD control */ return scarlett2_add_new_ctl(mixer, &scarlett2_msd_ctl, - 0, 1, "MSD Mode", NULL); + 0, 1, "MSD Mode Switch", NULL); } /*** Cleanup/Suspend Callbacks ***/ diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8b8bee3c3dd6..326d1b0ea5e6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1897,6 +1897,10 @@ static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */ REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ + REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ + REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ + REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ + REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ { 0 } /* terminator */ }; |