(I am having trouble delivering mail to linux.intel.com, so I reply to the list
and CC at least...)
> On 6. 6. 2022, at 22:02, Pierre-Louis Bossart <pierre-louis.bossart@xxxxxxxxxxxxxxx> wrote:
>
>
>> + * Virtual FE/BE Playback Topology
>> + * -------------------------------
>> + *
>> + * 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 is 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 a particular 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.)
>> + *
>> + * The DAPM routing that this machine-level driver makes up has two use-cases
>> + * in mind:
>> + *
>> + * - Using a single PCM for playback such that it conditionally sinks to either
>> + * speakers or headphones based on the plug-in state of the headphones jack.
>> + * All the while making the switch transparent to userspace. This has the
>> + * drawback of requiring a sample stream suited for both speakers and
>> + * headphones, which is hard to come by on machines where tailored DSP for
>> + * speakers in userspace is desirable or required.
>> + *
>> + * - Driving the headphones and speakers from distinct PCMs, having userspace
>> + * bridge the difference and apply different signal processing to the two.
>> + *
>> + * In the end the topology supplied by this driver looks like this:
>> + *
>> + * PCMs (frontends) I2S Port Groups (backends)
>> + * ──────────────── ──────────────────────────
>> + *
>> + * ┌──────────┐ ┌───────────────► ┌─────┐ ┌──────────┐
>> + * │ Primary ├───────┤ │ Mux │ ──► │ Speakers │
>> + * └──────────┘ │ ┌──────────► └─────┘ └──────────┘
>> + * ┌─── │ ───┘ ▲
>> + * ┌──────────┐ │ │ │
>> + * │Secondary ├──┘ │ ┌────────────┴┐
>> + * └──────────┘ ├────►│Plug-in Demux│
>> + * │ └────────────┬┘
>> + * │ │
>> + * │ ▼
>> + * │ ┌─────┐ ┌──────────┐
>> + * └───────────────► │ Mux │ ──► │Headphones│
>> + * └─────┘ └──────────┘
>> + */
>
> In Patch2, the 'clusters' are described as front-ends, with I2S as
> back-ends. Here the PCMs are described as front-ends, but there's no
> mention of clusters. Either one of the two descriptions is outdated, or
> there's something missing to help reconcile the two pieces of information?
Both descriptions should be in sync. Maybe I don’t know the proper
terminology. In both cases the frontend is meant to be the actual I2S
transceiver unit, and backend the I2S port on the SoC’s periphery,
which can be routed to any of transceiver units. (Multiple ports can
be routed to the same unit, which means the ports will have the same
clocks and data line -- that's a configuration we need to support to
drive some of the speaker arrays, hence the backend/frontend
distinction).
Maybe I am using 'PCM' in a confusing way here? What I meant is a
subdevice that’s visible from userspace, because I have seen it used
that way in ALSA codebase.
>> +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_kcalloc(dev, target->num_cpus,
>> + sizeof(*target->cpus), GFP_KERNEL);
>> + target->codecs = devm_kcalloc(dev, target->num_codecs,
>> + sizeof(*target->codecs), GFP_KERNEL);
>> + target->platforms = devm_kcalloc(dev, target->num_platforms,
>> + sizeof(*target->platforms), GFP_KERNEL);
>> +
>> + if (!target->cpus || !target->codecs || !target->platforms)
>> + return -ENOMEM;
>> +
>> + memcpy(target->cpus, source->cpus, sizeof(*target->cpus) * target->num_cpus);
>> + memcpy(target->codecs, source->codecs, sizeof(*target->codecs) * target->num_codecs);
>> + memcpy(target->platforms, source->platforms, sizeof(*target->platforms) * target->num_platforms);
>
>
> use devm_kmemdup?
Looks like what I am looking for.
>> +
>> + return 0;
>> +}
>
>> +static int macaudio_get_runtime_mclk_fs(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].mclk_fs;
>> +
>> + for_each_dpcm_fe(rtd, substream->stream, dpcm) {
>> + int fe_id = dpcm->fe->dai_link->id;
>> +
>> + return ma->link_props[fe_id].mclk_fs;
>> + }
>
> I am not sure what the concept of mclk would mean for a front-end? This
> is typically very I2S-specific, i.e. a back-end property, no?
Right, that’s a result of the confusion from above. Hope I cleared it up
somehow. The frontend already decides the clocks and data serialization,
hence mclk/fs is a frontend-prop here.
>> +
>> + 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 mclk_fs = macaudio_get_runtime_mclk_fs(substream);
>> + int i;
>> +
>> + if (mclk_fs) {
>> + struct snd_soc_dai *dai;
>> + int mclk = params_rate(params) * mclk_fs;
>> +
>> + for_each_rtd_codec_dais(rtd, i, dai)
>> + snd_soc_dai_set_sysclk(dai, 0, mclk, SND_SOC_CLOCK_IN);
>> +
>> + snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, SND_SOC_CLOCK_OUT);
>> + }
>> +
>> + return 0;
>> +}
>> +
>> +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 mclk_fs = macaudio_get_runtime_mclk_fs(substream);
>> + int i;
>> +
>> + if (mclk_fs) {
>> + 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 = {
>> + .shutdown = macaudio_dpcm_shutdown,
>> + .hw_params = macaudio_dpcm_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->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 -EOPNOTSUPP (see the comment
>> + * around mclk_fs value for primary FE).
>> + */
>> + if (ret == -EOPNOTSUPP && 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);
>> + }
>
> this is weird, set_jack() is invoked by the ASoC core. You shouldn't
> need to do this?
That’s interesting. Where would it be invoked? How does ASoC know which codec
it attaches to?
>> +
>> + 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);
>> + }
>
> same, why is this needed?
>
>> +}
>> +
>> +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->mclk_fs / 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 int macaudio_jack_event(struct notifier_block *nb, unsigned long event,
>> + void *data);
>> +
>> +static struct notifier_block macaudio_jack_nb = {
>> + .notifier_call = macaudio_jack_event,
>> +};
>
> why is this needed? we have already many ways of dealing with the jack
> events (dare I say too many ways?).
Because I want to update the DAPM paths based on the jack status,
specifically I want to set macaudio_plugin_demux. I don’t know how
else it could be done.
>> +
>> +static int macaudio_probe(struct snd_soc_card *card)
>> +{
>> + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(card);
>> + int ret;
>> +
>> + ma->pin.pin = "Headphones";
>> + ma->pin.mask = SND_JACK_HEADSET | SND_JACK_HEADPHONE;
>> + ret = snd_soc_card_jack_new(card, ma->pin.pin,
>> + SND_JACK_HEADSET |
>> + SND_JACK_HEADPHONE |
>> + SND_JACK_BTN_0 | SND_JACK_BTN_1 |
>> + SND_JACK_BTN_2 | SND_JACK_BTN_3,
>> + &ma->jack, &ma->pin, 1);
>> +
>> + if (ret < 0) {
>> + dev_err(card->dev, "jack creation failed: %d\n", ret);
>> + return ret;
>> + }
>> +
>> + snd_soc_jack_notifier_register(&ma->jack, &macaudio_jack_nb);
>> +
>> + 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];
>> + int nroutes;
>> + int ret;
>
> newline?
>
>> + memset(routes, 0, sizeof(routes));
>> +
>> + dev_dbg(card->dev, "adding routes for '%s'\n", dai->name);
>> +
>> + if (is_speakers)
>> + routes[0].source = "Speakers Playback";
>> + else
>> + routes[0].source = "Headphones Playback";
>> + routes[0].sink = dai->playback_widget->name;
>> + nroutes = 1;
>> +
>> + if (!is_speakers) {
>> + routes[1].source = dai->capture_widget->name;
>> + routes[1].sink = "Headphones Capture";
>> + nroutes = 2;
>> + }
>> +
>> + 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 bool macaudio_match_kctl_name(const char *pattern, const char *name)
>> +{
>> + if (pattern[0] == '*') {
>> + int namelen, patternlen;
>> +
>> + pattern++;
>> + if (pattern[0] == ' ')
>> + pattern++;
>> +
>> + namelen = strlen(name);
>> + patternlen = strlen(pattern);
>> +
>> + if (namelen > patternlen)
>> + name += (namelen - patternlen);
>> + }
>> +
>> + return !strcmp(name, pattern);
>> +}
>> +
>> +static int macaudio_limit_volume(struct snd_soc_card *card,
>> + const char *pattern, int max)
>> +{
>> + struct snd_kcontrol *kctl;
>> + struct soc_mixer_control *mc;
>> + int found = 0;
>> +
>> + list_for_each_entry(kctl, &card->snd_card->controls, list) {
>> + if (!macaudio_match_kctl_name(pattern, kctl->id.name))
>> + continue;
>> +
>> + found++;
>> + dev_dbg(card->dev, "limiting volume on '%s'\n", kctl->id.name);
>> +
>> + /*
>> + * TODO: This doesn't decrease the volume if it's already
>> + * above the limit!
>> + */
>> + mc = (struct soc_mixer_control *)kctl->private_value;
>> + if (max <= mc->max)
>> + mc->platform_max = max;
>> +
>> + }
>> +
>> + return found;
>> +}
>> +
>> +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;
>> + }
>> + }
>> +
>> + if (!ma->mdata) {
>> + dev_err(card->dev, "driver doesn't know speaker limits for this model\n");
>> + return void_warranty ? 0 : -EINVAL;
>> + }
>> +
>> + macaudio_limit_volume(card, "* Amp Gain", ma->mdata->spk_amp_gain_max);
>> + return 0;
>> +}
>> +
>> +static const char * const macaudio_plugin_demux_texts[] = {
>> + "Speakers",
>> + "Headphones"
>> +};
>> +
>> +SOC_ENUM_SINGLE_VIRT_DECL(macaudio_plugin_demux_enum, macaudio_plugin_demux_texts);
>> +
>> +static int macaudio_plugin_demux_get(struct snd_kcontrol *kcontrol,
>> + struct snd_ctl_elem_value *ucontrol)
>> +{
>> + struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
>> + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(dapm->card);
>> +
>> + /*
>> + * TODO: Determine what locking is in order here...
>> + */
>> + ucontrol->value.enumerated.item[0] = ma->jack_plugin_state;
>> +
>> + return 0;
>> +}
>> +
>> +static int macaudio_jack_event(struct notifier_block *nb, unsigned long event,
>> + void *data)
>> +{
>> + struct snd_soc_jack *jack = data;
>> + struct macaudio_snd_data *ma = snd_soc_card_get_drvdata(jack->card);
>> +
>> + ma->jack_plugin_state = !!event;
>> +
>> + if (!ma->plugin_demux_kcontrol)
>> + return 0;
>> +
>> + snd_soc_dapm_mux_update_power(&ma->card.dapm, ma->plugin_demux_kcontrol,
>> + ma->jack_plugin_state,
>> + (struct soc_enum *) &macaudio_plugin_demux_enum, NULL);
>
> the term 'plugin' can be understood in many ways by different audio
> folks. 'plugin' is usually the term used for processing libraries (VST,
> LADSPA, etc). I think here you meant 'jack control'?
So ‘jack control’ would be understood as the jack plugged/unplugged status?
>
>> +
>> + return 0;
>> +}
>> +
>
Martin
