diff options
| -rw-r--r-- | sound/soc/codecs/hdmi-codec.c | 22 | ||||
| -rw-r--r-- | sound/soc/dwc/dwc-i2s.c | 13 | ||||
| -rw-r--r-- | sound/soc/fsl/fsl_asrc_dma.c | 15 | ||||
| -rw-r--r-- | sound/soc/intel/avs/path.c | 72 | ||||
| -rw-r--r-- | sound/soc/intel/avs/path.h | 5 | ||||
| -rw-r--r-- | sound/soc/intel/avs/pcm.c | 52 | ||||
| -rw-r--r-- | sound/soc/intel/boards/sof_sdw.c | 1 | ||||
| -rw-r--r-- | sound/soc/qcom/lpass.h | 3 |
8 files changed, 169 insertions, 14 deletions
diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 17019b1d680b..bc01ff65bd6f 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -842,12 +842,28 @@ static void print_eld_info(struct snd_info_entry *entry, static int hdmi_dai_proc_new(struct hdmi_codec_priv *hcp, struct snd_soc_dai *dai) { + struct snd_soc_component *component = dai->component; + struct snd_soc_card *card = component->card; + struct snd_soc_dai *d; + struct snd_soc_pcm_runtime *rtd; struct snd_info_entry *entry; char name[32]; - int err; + int err, i, id = 0; - snprintf(name, sizeof(name), "eld#%d", dai->id); - err = snd_card_proc_new(dai->component->card->snd_card, name, &entry); + /* + * To avoid duplicate proc entry, find its rtd and use rtd->id + * instead of dai->id + */ + for_each_card_rtds(card, rtd) { + for_each_rtd_dais(rtd, i, d) + if (d == dai) { + id = rtd->id; + goto found; + } + } +found: + snprintf(name, sizeof(name), "eld#%d", id); + err = snd_card_proc_new(card->snd_card, name, &entry); if (err < 0) return err; diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 4c4171bb3fbb..28001e9857d9 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -199,12 +199,10 @@ static void i2s_start(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 1); - /* I2S needs to enable IRQ to make a handshake with DMAC on the JH7110 SoC */ - if (dev->use_pio || dev->is_jh7110) - i2s_enable_irqs(dev, substream->stream, config->chan_nr); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_enable_dma(dev, substream->stream); + i2s_enable_irqs(dev, substream->stream, config->chan_nr); i2s_write_reg(dev->i2s_base, CER, 1); } @@ -218,11 +216,12 @@ static void i2s_stop(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 0); - if (dev->use_pio || dev->is_jh7110) - i2s_disable_irqs(dev, substream->stream, 8); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_disable_dma(dev, substream->stream); + i2s_disable_irqs(dev, substream->stream, 8); + + if (!dev->active) { i2s_write_reg(dev->i2s_base, CER, 0); i2s_write_reg(dev->i2s_base, IER, 0); diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index f501f47242fb..1bba48318e2d 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -156,11 +156,24 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, for_each_dpcm_be(rtd, stream, dpcm) { struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *substream_be; - struct snd_soc_dai *dai = snd_soc_rtd_to_cpu(be, 0); + struct snd_soc_dai *dai_cpu = snd_soc_rtd_to_cpu(be, 0); + struct snd_soc_dai *dai_codec = snd_soc_rtd_to_codec(be, 0); + struct snd_soc_dai *dai; if (dpcm->fe != rtd) continue; + /* + * With audio graph card, original cpu dai is changed to codec + * device in backend, so if cpu dai is dummy device in backend, + * get the codec dai device, which is the real hardware device + * connected. + */ + if (!snd_soc_dai_is_dummy(dai_cpu)) + dai = dai_cpu; + else + dai = dai_codec; + substream_be = snd_soc_dpcm_get_substream(be, stream); dma_params_be = snd_soc_dai_get_dma_data(dai, substream_be); dev_be = dai->dev; diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index ef0c1d125d66..cafb8c6198be 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -115,6 +115,78 @@ avs_path_find_variant(struct avs_dev *adev, return NULL; } +static struct acpi_nhlt_config * +avs_nhlt_config_or_default(struct avs_dev *adev, struct avs_tplg_module *t); + +int avs_path_set_constraint(struct avs_dev *adev, struct avs_tplg_path_template *template, + struct snd_pcm_hw_constraint_list *rate_list, + struct snd_pcm_hw_constraint_list *channels_list, + struct snd_pcm_hw_constraint_list *sample_bits_list) +{ + struct avs_tplg_path *path_template; + unsigned int *rlist, *clist, *slist; + size_t i; + + i = 0; + list_for_each_entry(path_template, &template->path_list, node) + i++; + + rlist = kcalloc(i, sizeof(rlist), GFP_KERNEL); + clist = kcalloc(i, sizeof(clist), GFP_KERNEL); + slist = kcalloc(i, sizeof(slist), GFP_KERNEL); + + i = 0; + list_for_each_entry(path_template, &template->path_list, node) { + struct avs_tplg_pipeline *pipeline_template; + + list_for_each_entry(pipeline_template, &path_template->ppl_list, node) { + struct avs_tplg_module *module_template; + + list_for_each_entry(module_template, &pipeline_template->mod_list, node) { + const guid_t *type = &module_template->cfg_ext->type; + struct acpi_nhlt_config *blob; + + if (!guid_equal(type, &AVS_COPIER_MOD_UUID) && + !guid_equal(type, &AVS_WOVHOSTM_MOD_UUID)) + continue; + + switch (module_template->cfg_ext->copier.dma_type) { + case AVS_DMA_DMIC_LINK_INPUT: + case AVS_DMA_I2S_LINK_OUTPUT: + case AVS_DMA_I2S_LINK_INPUT: + break; + default: + continue; + } + + blob = avs_nhlt_config_or_default(adev, module_template); + if (IS_ERR(blob)) + continue; + + rlist[i] = path_template->fe_fmt->sampling_freq; + clist[i] = path_template->fe_fmt->num_channels; + slist[i] = path_template->fe_fmt->bit_depth; + i++; + } + } + } + + if (i) { + rate_list->count = i; + rate_list->list = rlist; + channels_list->count = i; + channels_list->list = clist; + sample_bits_list->count = i; + sample_bits_list->list = slist; + } else { + kfree(rlist); + kfree(clist); + kfree(slist); + } + + return i; +} + static void avs_init_node_id(union avs_connector_node_id *node_id, struct avs_tplg_modcfg_ext *te, u32 dma_id) { diff --git a/sound/soc/intel/avs/path.h b/sound/soc/intel/avs/path.h index 7ed7e94e0a56..c65ed84aa853 100644 --- a/sound/soc/intel/avs/path.h +++ b/sound/soc/intel/avs/path.h @@ -69,6 +69,11 @@ int avs_path_reset(struct avs_path *path); int avs_path_pause(struct avs_path *path); int avs_path_run(struct avs_path *path, int trigger); +int avs_path_set_constraint(struct avs_dev *adev, struct avs_tplg_path_template *template, + struct snd_pcm_hw_constraint_list *rate_list, + struct snd_pcm_hw_constraint_list *channels_list, + struct snd_pcm_hw_constraint_list *sample_bits_list); + int avs_peakvol_set_volume(struct avs_dev *adev, struct avs_path_module *mod, struct soc_mixer_control *mc, long *input); int avs_peakvol_set_mute(struct avs_dev *adev, struct avs_path_module *mod, diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index dac463390da1..d83ef504643b 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -31,6 +31,10 @@ struct avs_dma_data { struct hdac_ext_stream *host_stream; }; + struct snd_pcm_hw_constraint_list rate_list; + struct snd_pcm_hw_constraint_list channels_list; + struct snd_pcm_hw_constraint_list sample_bits_list; + struct work_struct period_elapsed_work; struct snd_pcm_substream *substream; }; @@ -74,6 +78,45 @@ void avs_period_elapsed(struct snd_pcm_substream *substream) schedule_work(&data->period_elapsed_work); } +static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule); +static int avs_hw_constraints_init(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hw_constraint_list *r, *c, *s; + struct avs_tplg_path_template *template; + struct avs_dma_data *data; + int ret; + + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + data = snd_soc_dai_get_dma_data(dai, substream); + r = &(data->rate_list); + c = &(data->channels_list); + s = &(data->sample_bits_list); + + template = avs_dai_find_path_template(dai, !rtd->dai_link->no_pcm, substream->stream); + ret = avs_path_set_constraint(data->adev, template, r, c, s); + if (ret <= 0) + return ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, r); + if (ret < 0) + return ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, c); + if (ret < 0) + return ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, s); + if (ret < 0) + return ret; + + return 0; +} + static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); @@ -101,7 +144,7 @@ static int avs_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_d if (rtd->dai_link->ignore_suspend) adev->num_lp_paths++; - return 0; + return avs_hw_constraints_init(substream, dai); } static void avs_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) @@ -114,6 +157,10 @@ static void avs_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc if (rtd->dai_link->ignore_suspend) data->adev->num_lp_paths--; + kfree(data->rate_list.list); + kfree(data->channels_list.list); + kfree(data->sample_bits_list.list); + snd_soc_dai_set_dma_data(dai, substream, NULL); kfree(data); } @@ -927,7 +974,8 @@ static int avs_component_probe(struct snd_soc_component *component) else mach->tplg_filename = devm_kasprintf(adev->dev, GFP_KERNEL, "hda-generic-tplg.bin"); - + if (!mach->tplg_filename) + return -ENOMEM; filename = kasprintf(GFP_KERNEL, "%s/%s", component->driver->topology_name_prefix, mach->tplg_filename); if (!filename) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 90dafa810b2e..095d08b3fc82 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -764,6 +764,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { static const struct snd_pci_quirk sof_sdw_ssid_quirk_table[] = { SND_PCI_QUIRK(0x1043, 0x1e13, "ASUS Zenbook S14", SOC_SDW_CODEC_MIC), + SND_PCI_QUIRK(0x1043, 0x1f43, "ASUS Zenbook S16", SOC_SDW_CODEC_MIC), {} }; diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index 27a2bf9a6613..de3ec6f594c1 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -13,10 +13,11 @@ #include <linux/platform_device.h> #include <linux/regmap.h> #include <dt-bindings/sound/qcom,lpass.h> +#include <dt-bindings/sound/qcom,q6afe.h> #include "lpass-hdmi.h" #define LPASS_AHBIX_CLOCK_FREQUENCY 131072000 -#define LPASS_MAX_PORTS (LPASS_CDC_DMA_VA_TX8 + 1) +#define LPASS_MAX_PORTS (DISPLAY_PORT_RX_7 + 1) #define LPASS_MAX_MI2S_PORTS (8) #define LPASS_MAX_DMA_CHANNELS (8) #define LPASS_MAX_HDMI_DMA_CHANNELS (4) |