diff options
Diffstat (limited to 'sound/soc/apple/macaudio.c')
| -rw-r--r-- | sound/soc/apple/macaudio.c | 1024 |
1 files changed, 1024 insertions, 0 deletions
diff --git a/sound/soc/apple/macaudio.c b/sound/soc/apple/macaudio.c new file mode 100644 index 000000000000..b1b74903db7d --- /dev/null +++ b/sound/soc/apple/macaudio.c @@ -0,0 +1,1024 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * ASoC machine driver for Apple Silicon Macs + * + * Copyright (C) The Asahi Linux Contributors + * + * Based on sound/soc/qcom/{sc7180.c|common.c} + * Copyright (c) 2018, Linaro Limited. + * Copyright (c) 2020, The Linux Foundation. All rights reserved. + * + * + * The platform driver has independent frontend and backend DAIs with the + * option of routing backends to any of the frontends. The platform + * driver configures the routing based on DPCM couplings in ASoC runtime + * structures, which in turn are determined from DAPM paths by ASoC. But the + * platform driver doesn't supply relevant DAPM paths and leaves that up for + * the machine driver to fill in. The filled-in virtual topology can be + * anything as long as any backend isn't connected to more than one frontend + * at any given time. (The limitation is due to the unsupported case of + * reparenting of live BEs.) + */ + +#define DEBUG + +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/simple_card_utils.h> +#include <sound/soc.h> +#include <sound/soc-jack.h> +#include <uapi/linux/input-event-codes.h> + +#define DRIVER_NAME "snd-soc-macaudio" + +/* + * CPU side is bit and frame clock provider + * I2S has both clocks inverted + */ +#define MACAUDIO_DAI_FMT (SND_SOC_DAIFMT_I2S | \ + SND_SOC_DAIFMT_CBC_CFC | \ + SND_SOC_DAIFMT_GATED | \ + SND_SOC_DAIFMT_IB_IF) +#define MACAUDIO_JACK_MASK (SND_JACK_HEADSET | SND_JACK_HEADPHONE) +#define MACAUDIO_SLOTWIDTH 32 + +struct macaudio_snd_data { + struct snd_soc_card card; + struct snd_soc_jack jack; + int jack_plugin_state; + + bool has_speakers; + unsigned int max_channels; + + struct macaudio_link_props { + /* frontend props */ + unsigned int bclk_ratio; + + /* backend props */ + bool is_speakers; + bool is_headphones; + unsigned int tdm_mask; + } *link_props; + + unsigned int speaker_nchans_array[2]; + struct snd_pcm_hw_constraint_list speaker_nchans_list; +}; + +static bool please_blow_up_my_speakers; +module_param(please_blow_up_my_speakers, bool, 0644); +MODULE_PARM_DESC(please_blow_up_my_speakers, "Allow unsafe or untested operating configurations"); + +SND_SOC_DAILINK_DEFS(primary, + DAILINK_COMP_ARRAY(COMP_CPU("mca-pcm-0")), // CPU + DAILINK_COMP_ARRAY(COMP_DUMMY()), // CODEC + DAILINK_COMP_ARRAY(COMP_EMPTY())); // platform (filled at runtime) + +SND_SOC_DAILINK_DEFS(secondary, + DAILINK_COMP_ARRAY(COMP_CPU("mca-pcm-1")), // CPU + DAILINK_COMP_ARRAY(COMP_DUMMY()), // CODEC + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +static struct snd_soc_dai_link macaudio_fe_links[] = { + { + .name = "Primary", + .stream_name = "Primary", + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_format = 1, + .dai_fmt = MACAUDIO_DAI_FMT, + SND_SOC_DAILINK_REG(primary), + }, + { + .name = "Secondary", + .stream_name = "Secondary", + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_format = 1, + .dai_fmt = MACAUDIO_DAI_FMT, + SND_SOC_DAILINK_REG(secondary), + }, +}; + +static struct macaudio_link_props macaudio_fe_link_props[] = { + { + /* + * Primary FE + * + * The bclk ratio at 64 for the primary frontend is important + * to ensure that the headphones codec's idea of left and right + * in a stereo stream over I2S fits in nicely with everyone else's. + * (This is until the headphones codec's driver supports + * set_tdm_slot.) + * + * The low bclk ratio precludes transmitting more than two + * channels over I2S, but that's okay since there is the secondary + * FE for speaker arrays anyway. + */ + .bclk_ratio = 64, + }, + { + /* + * Secondary FE + * + * Here we want frames plenty long to be able to drive all + * those fancy speaker arrays. + */ + .bclk_ratio = 256, + } +}; + +static int macaudio_copy_link(struct device *dev, struct snd_soc_dai_link *target, + struct snd_soc_dai_link *source) +{ + memcpy(target, source, sizeof(struct snd_soc_dai_link)); + + target->cpus = devm_kmemdup(dev, target->cpus, + sizeof(*target->cpus) * target->num_cpus, + GFP_KERNEL); + target->codecs = devm_kmemdup(dev, target->codecs, + sizeof(*target->codecs) * target->num_codecs, + GFP_KERNEL); + target->platforms = devm_kmemdup(dev, target->platforms, + sizeof(*target->platforms) * target->num_platforms, + GFP_KERNEL); + + if (!target->cpus || !target->codecs || !target->platforms) + return -ENOMEM; + + return 0; +} + +static int macaudio_parse_of_component(struct device_node *node, int index, + struct snd_soc_dai_link_component *comp) +{ + struct of_phandle_args args; + int ret; + + ret = of_parse_phandle_with_args(node, "sound-dai", "#sound-dai-cells", + index, &args); + if (ret) + return ret; + comp->of_node = args.np; + return snd_soc_get_dai_name(&args, &comp->dai_name); +} + +/* + * Parse one DPCM backend from the devicetree. This means taking one + * of the CPU DAIs and combining it with one or more CODEC DAIs. + */ +static int macaudio_parse_of_be_dai_link(struct macaudio_snd_data *ma, + struct snd_soc_dai_link *link, + int be_index, int ncodecs_per_be, + struct device_node *cpu, + struct device_node *codec) +{ + struct snd_soc_dai_link_component *comp; + struct device *dev = ma->card.dev; + int codec_base = be_index * ncodecs_per_be; + int ret, i; + + link->no_pcm = 1; + link->dpcm_playback = 1; + link->dpcm_capture = 1; + + link->dai_fmt = MACAUDIO_DAI_FMT; + + link->num_codecs = ncodecs_per_be; + link->codecs = devm_kcalloc(dev, ncodecs_per_be, + sizeof(*comp), GFP_KERNEL); + link->num_cpus = 1; + link->cpus = devm_kzalloc(dev, sizeof(*comp), GFP_KERNEL); + + if (!link->codecs || !link->cpus) + return -ENOMEM; + + link->num_platforms = 0; + + for_each_link_codecs(link, i, comp) { + ret = macaudio_parse_of_component(codec, codec_base + i, comp); + if (ret) + return ret; + } + + ret = macaudio_parse_of_component(cpu, be_index, link->cpus); + if (ret) + return ret; + + link->name = link->cpus[0].dai_name; + + return 0; +} + +static int macaudio_parse_of(struct macaudio_snd_data *ma) +{ + struct device_node *codec = NULL; + struct device_node *cpu = NULL; + struct device_node *np = NULL; + struct device_node *platform = NULL; + struct snd_soc_dai_link *link = NULL; + struct snd_soc_card *card = &ma->card; + struct device *dev = card->dev; + struct macaudio_link_props *link_props; + int ret, num_links, i; + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret) { + dev_err(dev, "Error parsing card name: %d\n", ret); + return ret; + } + + /* Populate links, start with the fixed number of FE links */ + num_links = ARRAY_SIZE(macaudio_fe_links); + + /* Now add together the (dynamic) number of BE links */ + for_each_available_child_of_node(dev->of_node, np) { + int num_cpus; + + cpu = of_get_child_by_name(np, "cpu"); + if (!cpu) { + dev_err(dev, "missing CPU DAI node at %pOF\n", np); + ret = -EINVAL; + goto err_free; + } + + num_cpus = of_count_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells"); + + if (num_cpus <= 0) { + dev_err(card->dev, "missing sound-dai property at %pOF\n", cpu); + ret = -EINVAL; + goto err_free; + } + of_node_put(cpu); + cpu = NULL; + + /* Each CPU specified counts as one BE link */ + num_links += num_cpus; + } + + /* Allocate the DAI link array */ + card->dai_link = devm_kcalloc(dev, num_links, sizeof(*link), GFP_KERNEL); + ma->link_props = devm_kcalloc(dev, num_links, sizeof(*ma->link_props), GFP_KERNEL); + if (!card->dai_link || !ma->link_props) + return -ENOMEM; + + card->num_links = num_links; + link = card->dai_link; + link_props = ma->link_props; + + for (i = 0; i < ARRAY_SIZE(macaudio_fe_links); i++) { + ret = macaudio_copy_link(dev, link, &macaudio_fe_links[i]); + if (ret) + goto err_free; + + memcpy(link_props, &macaudio_fe_link_props[i], sizeof(struct macaudio_link_props)); + link++; link_props++; + } + + for (i = 0; i < num_links; i++) + card->dai_link[i].id = i; + + /* Fill in the BEs */ + for_each_available_child_of_node(dev->of_node, np) { + const char *link_name; + bool speakers; + int be_index, num_codecs, num_bes, ncodecs_per_cpu, nchannels; + unsigned int left_mask, right_mask; + + ret = of_property_read_string(np, "link-name", &link_name); + if (ret) { + dev_err(card->dev, "missing link name\n"); + goto err_free; + } + + speakers = !strcmp(link_name, "Speaker") + || !strcmp(link_name, "Speakers"); + if (speakers) + ma->has_speakers = 1; + + cpu = of_get_child_by_name(np, "cpu"); + codec = of_get_child_by_name(np, "codec"); + + if (!codec || !cpu) { + dev_err(dev, "missing DAI specifications for '%s'\n", link_name); + ret = -EINVAL; + goto err_free; + } + + num_bes = of_count_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells"); + if (num_bes <= 0) { + dev_err(card->dev, "missing sound-dai property at %pOF\n", cpu); + ret = -EINVAL; + goto err_free; + } + + num_codecs = of_count_phandle_with_args(codec, "sound-dai", + "#sound-dai-cells"); + if (num_codecs <= 0) { + dev_err(card->dev, "missing sound-dai property at %pOF\n", codec); + ret = -EINVAL; + goto err_free; + } + + if (num_codecs % num_bes != 0) { + dev_err(card->dev, "bad combination of CODEC (%d) and CPU (%d) number at %pOF\n", + num_codecs, num_bes, np); + ret = -EINVAL; + goto err_free; + } + + /* + * Now parse the cpu/codec lists into a number of DPCM backend links. + * In each link there will be one DAI from the cpu list paired with + * an evenly distributed number of DAIs from the codec list. (As is + * the binding semantics.) + */ + ncodecs_per_cpu = num_codecs / num_bes; + nchannels = num_codecs * (speakers ? 1 : 2); + + /* Save the max number of channels on the platform */ + if (nchannels > ma->max_channels) + ma->max_channels = nchannels; + + /* + * If there is a single speaker, assign two channels to it, because + * it can do downmix. + */ + if (nchannels < 2) + nchannels = 2; + + left_mask = 0; + for (i = 0; i < nchannels; i += 2) + left_mask = left_mask << 2 | 1; + right_mask = left_mask << 1; + + for (be_index = 0; be_index < num_bes; be_index++) { + ret = macaudio_parse_of_be_dai_link(ma, link, be_index, + ncodecs_per_cpu, cpu, codec); + if (ret) + goto err_free; + + link_props->is_speakers = speakers; + link_props->is_headphones = !speakers; + + if (num_bes == 2) + /* This sound peripheral is split between left and right BE */ + link_props->tdm_mask = be_index ? right_mask : left_mask; + else + /* One BE covers all of the peripheral */ + link_props->tdm_mask = left_mask | right_mask; + + /* Steal platform OF reference for use in FE links later */ + platform = link->cpus->of_node; + + link++; link_props++; + } + + of_node_put(codec); + of_node_put(cpu); + cpu = codec = NULL; + } + + for (i = 0; i < ARRAY_SIZE(macaudio_fe_links); i++) + card->dai_link[i].platforms->of_node = platform; + + return 0; + +err_free: + of_node_put(codec); + of_node_put(cpu); + of_node_put(np); + + if (!card->dai_link) + return ret; + + for (i = 0; i < num_links; i++) { + /* + * TODO: If we don't go through this path are the references + * freed inside ASoC? + */ + snd_soc_of_put_dai_link_codecs(&card->dai_link[i]); + snd_soc_of_put_dai_link_cpus(&card->dai_link[i]); + } + + return ret; +} + +static int macaudio_get_runtime_bclk_ratio(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dpcm *dpcm; + + /* + * If this is a FE, look it up in link_props directly. + * If this is a BE, look it up in the respective FE. + */ + if (!rtd->dai_link->no_pcm) + return ma->link_props[rtd->dai_link->id].bclk_ratio; + + for_each_dpcm_fe(rtd, substream->stream, dpcm) { + int fe_id = dpcm->fe->dai_link->id; + + return ma->link_props[fe_id].bclk_ratio; + } + + return 0; +} + +static int macaudio_dpcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int bclk_ratio = macaudio_get_runtime_bclk_ratio(substream); + int i; + + if (bclk_ratio) { + struct snd_soc_dai *dai; + int mclk = params_rate(params) * bclk_ratio; + + for_each_rtd_codec_dais(rtd, i, dai) { + snd_soc_dai_set_sysclk(dai, 0, mclk, SND_SOC_CLOCK_IN); + snd_soc_dai_set_bclk_ratio(dai, bclk_ratio); + } + + snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, SND_SOC_CLOCK_OUT); + snd_soc_dai_set_bclk_ratio(cpu_dai, bclk_ratio); + } + + return 0; +} + +static int macaudio_fe_startup(struct snd_pcm_substream *substream) +{ + + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(rtd->card); + int ret; + + /* The FEs must never have more channels than the hardware */ + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, 0, ma->max_channels); + + if (ret < 0) { + dev_err(rtd->dev, "Failed to constrain FE %d! %d", rtd->dai_link->id, ret); + return ret; + } + + return 0; +} + +static int macaudio_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_pcm_runtime *be; + struct snd_soc_dpcm *dpcm; + + be = NULL; + for_each_dpcm_be(rtd, substream->stream, dpcm) { + be = dpcm->be; + break; + } + + if (!be) { + dev_err(rtd->dev, "opening PCM device '%s' with no audio route configured (bad settings applied to the sound card)\n", + rtd->dai_link->name); + return -EINVAL; + } + + return macaudio_dpcm_hw_params(substream, params); +} + + +static void macaudio_dpcm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_dai *dai; + int bclk_ratio = macaudio_get_runtime_bclk_ratio(substream); + int i; + + if (bclk_ratio) { + for_each_rtd_codec_dais(rtd, i, dai) + snd_soc_dai_set_sysclk(dai, 0, 0, SND_SOC_CLOCK_IN); + + snd_soc_dai_set_sysclk(cpu_dai, 0, 0, SND_SOC_CLOCK_OUT); + } +} + +static const struct snd_soc_ops macaudio_fe_ops = { + .startup = macaudio_fe_startup, + .shutdown = macaudio_dpcm_shutdown, + .hw_params = macaudio_fe_hw_params, +}; + +static const struct snd_soc_ops macaudio_be_ops = { + .shutdown = macaudio_dpcm_shutdown, + .hw_params = macaudio_dpcm_hw_params, +}; + +static int macaudio_be_assign_tdm(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id]; + struct snd_soc_dai *dai; + unsigned int mask; + int nslots, ret, i; + + if (!props->tdm_mask) + return 0; + + mask = props->tdm_mask; + nslots = __fls(mask) + 1; + + if (rtd->dai_link->num_codecs == 1) { + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(rtd, 0), mask, + 0, nslots, MACAUDIO_SLOTWIDTH); + + /* + * Headphones get a pass on -ENOTSUPP (see the comment + * around bclk_ratio value for primary FE). + */ + if (ret == -ENOTSUPP && props->is_headphones) + return 0; + + return ret; + } + + for_each_rtd_codec_dais(rtd, i, dai) { + int slot = __ffs(mask); + + mask &= ~(1 << slot); + ret = snd_soc_dai_set_tdm_slot(dai, 1 << slot, 0, nslots, + MACAUDIO_SLOTWIDTH); + if (ret) + return ret; + } + + return 0; +} + +static int macaudio_be_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id]; + struct snd_soc_dai *dai; + int i, ret; + + ret = macaudio_be_assign_tdm(rtd); + if (ret < 0) + return ret; + + if (props->is_headphones) { + for_each_rtd_codec_dais(rtd, i, dai) + snd_soc_component_set_jack(dai->component, &ma->jack, NULL); + } + + return 0; +} + +static void macaudio_be_exit(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id]; + struct snd_soc_dai *dai; + int i; + + if (props->is_headphones) { + for_each_rtd_codec_dais(rtd, i, dai) + snd_soc_component_set_jack(dai->component, NULL, NULL); + } +} + +static int macaudio_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id]; + int nslots = props->bclk_ratio / MACAUDIO_SLOTWIDTH; + + return snd_soc_dai_set_tdm_slot(asoc_rtd_to_cpu(rtd, 0), (1 << nslots) - 1, + (1 << nslots) - 1, nslots, MACAUDIO_SLOTWIDTH); +} + +static struct snd_soc_jack_pin macaudio_jack_pins[] = { + { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int macaudio_probe(struct snd_soc_card *card) +{ + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + int ret; + + dev_dbg(card->dev, "%s!\n", __func__); + + ret = snd_soc_card_jack_new_pins(card, "Headphone Jack", + SND_JACK_HEADSET | SND_JACK_HEADPHONE, + &ma->jack, macaudio_jack_pins, + ARRAY_SIZE(macaudio_jack_pins)); + if (ret < 0) { + dev_err(card->dev, "jack creation failed: %d\n", ret); + return ret; + } + + return ret; +} + +static int macaudio_add_backend_dai_route(struct snd_soc_card *card, struct snd_soc_dai *dai, + bool is_speakers) +{ + struct snd_soc_dapm_route routes[2]; + struct snd_soc_dapm_route *r; + int nroutes = 0; + int ret; + + memset(routes, 0, sizeof(routes)); + + dev_dbg(card->dev, "adding routes for '%s'\n", dai->name); + + r = &routes[nroutes++]; + if (is_speakers) + r->source = "Speaker Playback"; + else + r->source = "Headphone Playback"; + r->sink = dai->playback_widget->name; + + /* If headphone jack, add capture path */ + if (!is_speakers) { + r = &routes[nroutes++]; + r->source = dai->capture_widget->name; + r->sink = "Headset Capture"; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, routes, nroutes); + if (ret) + dev_err(card->dev, "failed adding dynamic DAPM routes for %s\n", + dai->name); + return ret; +} + +static int macaudio_add_pin_routes(struct snd_soc_card *card, struct snd_soc_component *component, + bool is_speakers) +{ + struct snd_soc_dapm_route routes[2]; + struct snd_soc_dapm_route *r; + int nroutes = 0; + char buf[32]; + int ret; + + memset(routes, 0, sizeof(routes)); + + /* Connect the far ends of CODECs to pins */ + if (is_speakers) { + r = &routes[nroutes++]; + r->source = "OUT"; + if (component->name_prefix) { + snprintf(buf, sizeof(buf) - 1, "%s OUT", component->name_prefix); + r->source = buf; + } + r->sink = "Speaker"; + } else { + r = &routes[nroutes++]; + r->source = "Jack HP"; + r->sink = "Headphone"; + r = &routes[nroutes++]; + r->source = "Headset Mic"; + r->sink = "Jack HS"; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, routes, nroutes); + if (ret) + dev_err(card->dev, "failed adding dynamic DAPM routes for %s\n", + component->name); + return ret; +} + +static int macaudio_late_probe(struct snd_soc_card *card) +{ + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *dai; + int ret, i; + + /* Add the dynamic DAPM routes */ + for_each_card_rtds(card, rtd) { + struct macaudio_link_props *props = &ma->link_props[rtd->dai_link->id]; + + if (!rtd->dai_link->no_pcm) + continue; + + for_each_rtd_cpu_dais(rtd, i, dai) { + ret = macaudio_add_backend_dai_route(card, dai, props->is_speakers); + + if (ret) + return ret; + } + + for_each_rtd_codec_dais(rtd, i, dai) { + ret = macaudio_add_pin_routes(card, dai->component, + props->is_speakers); + + if (ret) + return ret; + } + } + + return 0; +} + +#define CHECK(call, pattern, value) \ + { \ + int ret = call(card, pattern, value); \ + if (ret < 1 && !please_blow_up_my_speakers) { \ + dev_err(card->dev, "%s on '%s': %d\n", #call, pattern, ret); \ + return ret; \ + } \ + dev_dbg(card->dev, "%s on '%s': %d hits\n", #call, pattern, ret); \ + } + + +static int macaudio_j274_fixup_controls(struct snd_soc_card *card) +{ + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + + if (ma->has_speakers) { + CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 14); // 20 set by macOS, this is 3 dB below + } + + return 0; +} + +static int macaudio_j313_fixup_controls(struct snd_soc_card *card) { + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + + if (ma->has_speakers) { + if (!please_blow_up_my_speakers) { + dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n"); + return -EINVAL; + } + + CHECK(snd_soc_set_enum_kctl, "* ASI1 Sel", "Left"); + CHECK(snd_soc_deactivate_kctl, "* ASI1 Sel", 0); + + /* !!! This is copied from j274, not obtained by looking at + * what macOS sets. + */ + CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 14); + + /* + * Since we don't set the right slots yet to avoid + * driver conflict on the I2S bus sending ISENSE/VSENSE + * samples from the codecs back to us, disable the + * controls. + */ + CHECK(snd_soc_deactivate_kctl, "* VSENSE Switch", 0); + CHECK(snd_soc_deactivate_kctl, "* ISENSE Switch", 0); + } + + return 0; +} + +static int macaudio_j314_fixup_controls(struct snd_soc_card *card) +{ + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + + if (ma->has_speakers) { + if (!please_blow_up_my_speakers) { + dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n"); + return -EINVAL; + } + + CHECK(snd_soc_set_enum_kctl, "* ASI1 Sel", "Left"); + CHECK(snd_soc_deactivate_kctl, "* ASI1 Sel", 0); + CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 9); // 15 set by macOS, this is 3 dB below + CHECK(snd_soc_set_enum_kctl, "* Tweeter HPF Corner Frequency", "800 Hz"); + CHECK(snd_soc_deactivate_kctl, "* Tweeter HPF Corner Frequency", 0); + + /* + * The speaker amps suffer from spurious overcurrent + * events on their unmute, so enable autoretry. + */ + CHECK(snd_soc_set_enum_kctl, "* OCE Handling", "Retry"); + CHECK(snd_soc_deactivate_kctl, "* OCE Handling", 0); + + /* + * Since we don't set the right slots yet to avoid + * driver conflict on the I2S bus sending ISENSE/VSENSE + * samples from the codecs back to us, disable the + * controls. + */ + CHECK(snd_soc_deactivate_kctl, "* VSENSE Switch", 0); + CHECK(snd_soc_deactivate_kctl, "* ISENSE Switch", 0); + } + + return 0; +} + +static int macaudio_j375_fixup_controls(struct snd_soc_card *card) +{ + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + + if (ma->has_speakers) { + if (!please_blow_up_my_speakers) { + dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n"); + return -EINVAL; + } + + CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 14); // 20 set by macOS, this is 3 dB below + } + + return 0; +} + +static int macaudio_j493_fixup_controls(struct snd_soc_card *card) +{ + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + + if (ma->has_speakers) { + if (!please_blow_up_my_speakers) { + dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n"); + return -EINVAL; + } + + CHECK(snd_soc_limit_volume, "* Amp Gain Volume", 9); // 15 set by macOS, this is 3 dB below + } + + return 0; +} + +static int macaudio_fallback_fixup_controls(struct snd_soc_card *card) +{ + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card); + + if (ma->has_speakers && !please_blow_up_my_speakers) { + dev_err(card->dev, "driver can't assure safety on this model, refusing probe\n"); + return -EINVAL; + } + + return 0; +} + +#undef CHECK + +static const char * const macaudio_spk_mux_texts[] = { + "Primary", + "Secondary" +}; + +SOC_ENUM_SINGLE_VIRT_DECL(macaudio_spk_mux_enum, macaudio_spk_mux_texts); + +static const struct snd_kcontrol_new macaudio_spk_mux = + SOC_DAPM_ENUM("Speaker Playback Mux", macaudio_spk_mux_enum); + +static const char * const macaudio_hp_mux_texts[] = { + "Primary", + "Secondary" +}; + +SOC_ENUM_SINGLE_VIRT_DECL(macaudio_hp_mux_enum, macaudio_hp_mux_texts); + +static const struct snd_kcontrol_new macaudio_hp_mux = + SOC_DAPM_ENUM("Headphones Playback Mux", macaudio_hp_mux_enum); + +static const struct snd_soc_dapm_widget macaudio_snd_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SPK("Speaker (Static)", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + + SND_SOC_DAPM_MUX("Speaker Playback Mux", SND_SOC_NOPM, 0, 0, &macaudio_spk_mux), + SND_SOC_DAPM_MUX("Headphone Playback Mux", SND_SOC_NOPM, 0, 0, &macaudio_hp_mux), + + SND_SOC_DAPM_AIF_OUT("Speaker Playback", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("Headphone Playback", NULL, 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_IN("Headset Capture", NULL, 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_kcontrol_new macaudio_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_route macaudio_dapm_routes[] = { + /* Playback paths */ + { "Speaker Playback Mux", "Primary", "PCM0 TX" }, + { "Speaker Playback Mux", "Secondary", "PCM1 TX" }, + { "Speaker Playback", NULL, "Speaker Playback Mux"}, + + { "Headphone Playback Mux", "Primary", "PCM0 TX" }, + { "Headphone Playback Mux", "Secondary", "PCM1 TX" }, + { "Headphone Playback", NULL, "Headphone Playback Mux"}, + /* + * Additional paths (to specific I2S ports) are added dynamically. + */ + + /* Capture paths */ + { "PCM0 RX", NULL, "Headset Capture" }, +}; + +static const struct of_device_id macaudio_snd_device_id[] = { + { .compatible = "apple,j274-macaudio", .data = macaudio_j274_fixup_controls }, + { .compatible = "apple,j313-macaudio", .data = macaudio_j313_fixup_controls }, + { .compatible = "apple,j314-macaudio", .data = macaudio_j314_fixup_controls }, + { .compatible = "apple,j375-macaudio", .data = macaudio_j375_fixup_controls }, + { .compatible = "apple,j413-macaudio", .data = macaudio_j314_fixup_controls }, + { .compatible = "apple,j493-macaudio", .data = macaudio_j493_fixup_controls }, + { .compatible = "apple,macaudio"}, + { } +}; +MODULE_DEVICE_TABLE(of, macaudio_snd_device_id); + +static int macaudio_snd_platform_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct macaudio_snd_data *data; + struct device *dev = &pdev->dev; + struct snd_soc_dai_link *link; + const struct of_device_id *of_id; + int ret; + int i; + + of_id = of_match_device(macaudio_snd_device_id, dev); + if (!of_id) + return -EINVAL; + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + card = &data->card; + snd_soc_card_set_drvdata(card, data); + + card->owner = THIS_MODULE; + card->driver_name = "macaudio"; + card->dev = dev; + card->dapm_widgets = macaudio_snd_widgets; + card->num_dapm_widgets = ARRAY_SIZE(macaudio_snd_widgets); + card->dapm_routes = macaudio_dapm_routes; + card->num_dapm_routes = ARRAY_SIZE(macaudio_dapm_routes); + card->controls = macaudio_controls; + card->num_controls = ARRAY_SIZE(macaudio_controls); + card->probe = macaudio_probe; + card->late_probe = macaudio_late_probe; + card->component_chaining = true; + card->fully_routed = true; + + if (of_id->data) + card->fixup_controls = of_id->data; + else + card->fixup_controls = macaudio_fallback_fixup_controls; + + ret = macaudio_parse_of(data); + if (ret) + return dev_err_probe(&pdev->dev, ret, "failed OF parsing\n"); + + for_each_card_prelinks(card, i, link) { + if (link->no_pcm) { + link->ops = &macaudio_be_ops; + link->init = macaudio_be_init; + link->exit = macaudio_be_exit; + } else { + link->ops = &macaudio_fe_ops; + link->init = macaudio_fe_init; + } + } + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver macaudio_snd_driver = { + .probe = macaudio_snd_platform_probe, + .driver = { + .name = DRIVER_NAME, + .of_match_table = macaudio_snd_device_id, + .pm = &snd_soc_pm_ops, + }, +}; +module_platform_driver(macaudio_snd_driver); + +MODULE_AUTHOR("Martin PoviĊĦer <povik+lin@cutebit.org>"); +MODULE_DESCRIPTION("Apple Silicon Macs machine-level sound driver"); +MODULE_LICENSE("GPL"); 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