[PATCH v2 00/14] ASoC: Intel: avs: Machine boards and HDA codec support
This series focuses on populating boards/ subdirectory with supported configurations by the avs-driver. Note: it is independent of recently provided "Driver code and PCM operations" series [1], that is, code found here should not collide with it.
Series starts with a small change that adds a helper to sound pcm header, allowing for retrieving string naming a direction without the need of substream pointer. Said helper is used by codec driver code that follows it but I believe it's generic and helpful enough that it can be called an independent addition to the sound core.
Code for generic HD-Audio codec driver follows. It is a ASoC wrapper for existing HD-Audio codec code found in sound/pci/hda/. There is basically no custom logic involved up to the point that driver follows HDA_DEV_LEGACY convention, rather than the HDA_DEV_ASOC one. Commit message for the given patch iterates on this and explains crucial parts of the implementation.
From there on is a range of boards appended. All of them follow the same
scheme:
- define avs_create_dai_link() so DAI-LINKs can be created dynamically, based on the link_mask (I2S) or the number of entries in the ->pcm_list_head list (HDA) - define avs_create_dapm_routes() so DAPM routes can be created dynamically, same rules as above apply - define probe() function that creates new ASoC card, assign all required operations and resources along with calling the two above
Changes in v2: - 'link_mask' usage replaced with 'i2s_link_mask' as requested by Pierre - 'ssp_test' board renamed to 'i2s_test' to match naming convention used for other i2s machine boards - enriched commit message and Kconfig for the 'HD-Audio codec driver' patch as requested by Kai
[1]: https://lore.kernel.org/all/20220426172346.3508411-1-cezary.rojewski@intel.c...
Amadeusz Sławiński (1): ASoC: Intel: avs: Add max98373 machine board
Cezary Rojewski (13): ALSA: Add snd_pcm_direction_name() helper ASoC: codecs: Add HD-Audio codec driver ASoC: Intel: avs: Add HDAudio machine board ASoC: Intel: avs: Add DMIC machine board ASoC: Intel: avs: Add I2S-test machine board ASoC: Intel: avs: Add rt274 machine board ASoC: Intel: avs: Add rt286 machine board ASoC: Intel: avs: Add rt298 machine board ASoC: Intel: avs: Add rt5682 machine board ASoC: Intel: avs: Add nau8825 machine board ASoC: Intel: avs: Add ssm4567 machine board ASoC: Intel: avs: Add max98357a machine board ASoC: Intel: avs: Add da7219 machine board
include/sound/pcm.h | 19 +- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hda-dai.c | 102 +++++++ sound/soc/codecs/hda.c | 395 +++++++++++++++++++++++++ sound/soc/codecs/hda.h | 19 ++ sound/soc/intel/Kconfig | 3 + sound/soc/intel/avs/Makefile | 3 + sound/soc/intel/avs/boards/Kconfig | 121 ++++++++ sound/soc/intel/avs/boards/Makefile | 27 ++ sound/soc/intel/avs/boards/da7219.c | 282 ++++++++++++++++++ sound/soc/intel/avs/boards/dmic.c | 93 ++++++ sound/soc/intel/avs/boards/hdaudio.c | 294 ++++++++++++++++++ sound/soc/intel/avs/boards/i2s_test.c | 180 +++++++++++ sound/soc/intel/avs/boards/max98357a.c | 154 ++++++++++ sound/soc/intel/avs/boards/max98373.c | 239 +++++++++++++++ sound/soc/intel/avs/boards/nau8825.c | 353 ++++++++++++++++++++++ sound/soc/intel/avs/boards/rt274.c | 310 +++++++++++++++++++ sound/soc/intel/avs/boards/rt286.c | 281 ++++++++++++++++++ sound/soc/intel/avs/boards/rt298.c | 281 ++++++++++++++++++ sound/soc/intel/avs/boards/rt5682.c | 340 +++++++++++++++++++++ sound/soc/intel/avs/boards/ssm4567.c | 271 +++++++++++++++++ 22 files changed, 3775 insertions(+), 4 deletions(-) create mode 100644 sound/soc/codecs/hda-dai.c create mode 100644 sound/soc/codecs/hda.c create mode 100644 sound/soc/codecs/hda.h create mode 100644 sound/soc/intel/avs/boards/Kconfig create mode 100644 sound/soc/intel/avs/boards/Makefile create mode 100644 sound/soc/intel/avs/boards/da7219.c create mode 100644 sound/soc/intel/avs/boards/dmic.c create mode 100644 sound/soc/intel/avs/boards/hdaudio.c create mode 100644 sound/soc/intel/avs/boards/i2s_test.c create mode 100644 sound/soc/intel/avs/boards/max98357a.c create mode 100644 sound/soc/intel/avs/boards/max98373.c create mode 100644 sound/soc/intel/avs/boards/nau8825.c create mode 100644 sound/soc/intel/avs/boards/rt274.c create mode 100644 sound/soc/intel/avs/boards/rt286.c create mode 100644 sound/soc/intel/avs/boards/rt298.c create mode 100644 sound/soc/intel/avs/boards/rt5682.c create mode 100644 sound/soc/intel/avs/boards/ssm4567.c
Allow for retrieving string naming a direction of a stream without the need of substream pointer.
Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- include/sound/pcm.h | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-)
diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6b99310b5b88..26523cfe428d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1392,6 +1392,20 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
const char *snd_pcm_format_name(snd_pcm_format_t format);
+/** + * snd_pcm_direction_name - Get a string naming the direction of a stream + * @direction: Stream's direction, one of SNDRV_PCM_STREAM_XXX + * + * Returns a string naming the direction of the stream. + */ +static inline const char *snd_pcm_direction_name(int direction) +{ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + /** * snd_pcm_stream_str - Get a string naming the direction of a stream * @substream: the pcm substream instance @@ -1400,10 +1414,7 @@ const char *snd_pcm_format_name(snd_pcm_format_t format); */ static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - return "Playback"; - else - return "Capture"; + return snd_pcm_direction_name(substream->stream); }
/*
Add generic ASoC equivalent of ALSA HD-Audio codec. This codec is designed to follow HDA_DEV_LEGACY convention. Driver wrapps existing hda_codec.c handlers to prevent code duplication within the newly added code. Number of DAIs created is dependent on capabilities exposed by the codec itself. Because of this, single solution can be applied to support every single HD-Audio codec type.
At the same time, through the ASoC topology, platform drivers may limit the number of endpoints available to the userspace as codec driver exposes BE DAIs only.
Both hda_codec_probe() and hda_codec_remove() declare their expectations on device's usage_count and suspended-status. This is to catch any unexpected behavior as PM-related code for HD-Audio has been changing quite a bit throughout the years.
In order for codec DAI list to reflect its actual PCM capabilities, PCMs need to be built and that can only happen once codec device is constructed. To do that, a valid component->card->snd_card pointer is needed. Said pointer will be provided by the framework once all card components are accounted for and their probing can begin. Usage of "binder" BE DAI solves the problem - codec can be listed as one of HD-Audio card components without declaring any actual BE DAIs statically.
Relation with hdac_hda:
Addition of parallel solution is motivated by behavioral differences between hdac_hda.c and its legacy equivalent found in sound/pci/hda e.g.: lack of dynamic, based on codec capabilities, resource allocation and high cost of removing such differences on actively used targets. Major goal of codec driver presented here is to follow HD-Audio legacy behavior in 1:1 fashion by becoming a wrapper. Doing so increases code coverage of the legacy code and reduces the maintenance cost for both solutions.
Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hda-dai.c | 102 ++++++++++ sound/soc/codecs/hda.c | 395 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hda.h | 19 ++ 5 files changed, 528 insertions(+) create mode 100644 sound/soc/codecs/hda-dai.c create mode 100644 sound/soc/codecs/hda.c create mode 100644 sound/soc/codecs/hda.h
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b106e5517090..4b1819d84d83 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -937,6 +937,16 @@ config SND_SOC_HDAC_HDA tristate select SND_HDA
+config SND_SOC_HDA + tristate "HD-Audio codec driver" + select SND_HDA_EXT_CORE + select SND_HDA + help + This enables HD-Audio codec support in ASoC subsystem. Compared + to SND_SOC_HDAC_HDA, driver's behavior is identical to HD-Audio + legacy solution - including the dynamic resource allocation + based on actual codec capabilities. + config SND_SOC_ICS43432 tristate "ICS43423 and compatible i2s microphones"
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 28dc4edfd01f..d32026ae326f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -106,6 +106,7 @@ snd-soc-es8328-spi-objs := es8328-spi.o snd-soc-gtm601-objs := gtm601.o snd-soc-hdac-hdmi-objs := hdac_hdmi.o snd-soc-hdac-hda-objs := hdac_hda.o +snd-soc-hda-codec-objs := hda.o hda-dai.o snd-soc-ics43432-objs := ics43432.o snd-soc-inno-rk3036-objs := inno_rk3036.o snd-soc-isabelle-objs := isabelle.o @@ -458,6 +459,7 @@ obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o obj-$(CONFIG_SND_SOC_GTM601) += snd-soc-gtm601.o obj-$(CONFIG_SND_SOC_HDAC_HDMI) += snd-soc-hdac-hdmi.o obj-$(CONFIG_SND_SOC_HDAC_HDA) += snd-soc-hdac-hda.o +obj-$(CONFIG_SND_SOC_HDA) += snd-soc-hda-codec.o obj-$(CONFIG_SND_SOC_ICS43432) += snd-soc-ics43432.o obj-$(CONFIG_SND_SOC_INNO_RK3036) += snd-soc-inno-rk3036.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/hda-dai.c b/sound/soc/codecs/hda-dai.c new file mode 100644 index 000000000000..5371ff086261 --- /dev/null +++ b/sound/soc/codecs/hda-dai.c @@ -0,0 +1,102 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski cezary.rojewski@intel.com +// + +#include <sound/soc.h> +#include <sound/hda_codec.h> +#include "hda.h" + +static int hda_codec_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct hda_pcm_stream *stream_info; + struct hda_codec *codec; + struct hda_pcm *pcm; + int ret; + + codec = dev_to_hda_codec(dai->dev); + stream_info = snd_soc_dai_get_dma_data(dai, substream); + pcm = container_of(stream_info, struct hda_pcm, stream[substream->stream]); + + dev_dbg(dai->dev, "open stream codec: %08x, info: %p, pcm: %p %s substream: %p\n", + codec->core.vendor_id, stream_info, pcm, pcm->name, substream); + + snd_hda_codec_pcm_get(pcm); + + ret = stream_info->ops.open(stream_info, codec, substream); + if (ret < 0) { + dev_err(dai->dev, "codec open failed: %d\n", ret); + snd_hda_codec_pcm_put(pcm); + return ret; + } + + return 0; +} + +static void hda_codec_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct hda_pcm_stream *stream_info; + struct hda_codec *codec; + struct hda_pcm *pcm; + int ret; + + codec = dev_to_hda_codec(dai->dev); + stream_info = snd_soc_dai_get_dma_data(dai, substream); + pcm = container_of(stream_info, struct hda_pcm, stream[substream->stream]); + + dev_dbg(dai->dev, "close stream codec: %08x, info: %p, pcm: %p %s substream: %p\n", + codec->core.vendor_id, stream_info, pcm, pcm->name, substream); + + ret = stream_info->ops.close(stream_info, codec, substream); + if (ret < 0) + dev_err(dai->dev, "codec close failed: %d\n", ret); + + snd_hda_codec_pcm_put(pcm); +} + +static int hda_codec_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct hda_pcm_stream *stream_info; + struct hda_codec *codec; + + codec = dev_to_hda_codec(dai->dev); + stream_info = snd_soc_dai_get_dma_data(dai, substream); + + snd_hda_codec_cleanup(codec, stream_info, substream); + + return 0; +} + +static int hda_codec_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct hda_pcm_stream *stream_info; + struct hdac_stream *stream; + struct hda_codec *codec; + unsigned int format; + int ret; + + codec = dev_to_hda_codec(dai->dev); + stream = substream->runtime->private_data; + stream_info = snd_soc_dai_get_dma_data(dai, substream); + format = snd_hdac_calc_stream_format(runtime->rate, runtime->channels, runtime->format, + runtime->sample_bits, 0); + + ret = snd_hda_codec_prepare(codec, stream_info, stream->stream_tag, format, substream); + if (ret < 0) { + dev_err(dai->dev, "codec prepare failed: %d\n", ret); + return ret; + } + + return 0; +} + +const struct snd_soc_dai_ops snd_soc_hda_codec_dai_ops = { + .startup = hda_codec_dai_startup, + .shutdown = hda_codec_dai_shutdown, + .hw_free = hda_codec_dai_hw_free, + .prepare = hda_codec_dai_prepare, +}; +EXPORT_SYMBOL_GPL(snd_soc_hda_codec_dai_ops); diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c new file mode 100644 index 000000000000..edcb8bc6806b --- /dev/null +++ b/sound/soc/codecs/hda.c @@ -0,0 +1,395 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski cezary.rojewski@intel.com +// + +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> +#include <sound/hdaudio_ext.h> +#include <sound/hda_i915.h> +#include <sound/hda_codec.h> +#include "hda.h" + +static int hda_codec_create_dais(struct hda_codec *codec, int pcm_count, + struct snd_soc_dai_driver **drivers) +{ + struct device *dev = &codec->core.dev; + struct snd_soc_dai_driver *drvs; + struct hda_pcm *pcm; + int i; + + drvs = devm_kcalloc(dev, pcm_count, sizeof(*drvs), GFP_KERNEL); + if (!drvs) + return -ENOMEM; + + pcm = list_first_entry(&codec->pcm_list_head, struct hda_pcm, list); + + for (i = 0; i < pcm_count; i++, pcm = list_next_entry(pcm, list)) { + struct snd_soc_pcm_stream *stream; + int dir; + + dev_info(dev, "creating for %s %d\n", pcm->name, i); + drvs[i].id = i; + drvs[i].name = pcm->name; + drvs[i].ops = &snd_soc_hda_codec_dai_ops; + + dir = SNDRV_PCM_STREAM_PLAYBACK; + stream = &drvs[i].playback; + if (!pcm->stream[dir].substreams) { + dev_info(dev, "skipping playback dai for %s\n", pcm->name); + goto capture_dais; + } + + stream->stream_name = + devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = pcm->stream[dir].channels_min; + stream->channels_max = pcm->stream[dir].channels_max; + stream->rates = pcm->stream[dir].rates; + stream->formats = pcm->stream[dir].formats; + stream->sig_bits = pcm->stream[dir].maxbps; + +capture_dais: + dir = SNDRV_PCM_STREAM_CAPTURE; + stream = &drvs[i].capture; + if (!pcm->stream[dir].substreams) { + dev_info(dev, "skipping capture dai for %s\n", pcm->name); + continue; + } + + stream->stream_name = + devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = pcm->stream[dir].channels_min; + stream->channels_max = pcm->stream[dir].channels_max; + stream->rates = pcm->stream[dir].rates; + stream->formats = pcm->stream[dir].formats; + stream->sig_bits = pcm->stream[dir].maxbps; + } + + *drivers = drvs; + return 0; +} + +static int hda_codec_register_dais(struct hda_codec *codec, struct snd_soc_component *component) +{ + struct snd_soc_dai_driver *drvs = NULL; + struct snd_soc_dapm_context *dapm; + struct hda_pcm *pcm; + int ret, pcm_count = 0; + + if (list_empty(&codec->pcm_list_head)) + return -EINVAL; + list_for_each_entry(pcm, &codec->pcm_list_head, list) + pcm_count++; + + ret = hda_codec_create_dais(codec, pcm_count, &drvs); + if (ret < 0) + return ret; + + dapm = snd_soc_component_get_dapm(component); + + list_for_each_entry(pcm, &codec->pcm_list_head, list) { + struct snd_soc_dai *dai; + + dai = snd_soc_register_dai(component, drvs, false); + if (!dai) { + dev_err(component->dev, "register dai for %s failed\n", pcm->name); + return -EINVAL; + } + + ret = snd_soc_dapm_new_dai_widgets(dapm, dai); + if (ret < 0) { + dev_err(component->dev, "create widgets failed: %d\n", ret); + snd_soc_unregister_dai(dai); + return ret; + } + + snd_soc_dai_init_dma_data(dai, &pcm->stream[0], &pcm->stream[1]); + drvs++; + } + + return 0; +} + +static void hda_codec_unregister_dais(struct hda_codec *codec, + struct snd_soc_component *component) +{ + struct snd_soc_dai *dai, *save; + struct hda_pcm *pcm; + + for_each_component_dais_safe(component, dai, save) { + list_for_each_entry(pcm, &codec->pcm_list_head, list) { + if (strcmp(dai->driver->name, pcm->name)) + continue; + + if (dai->playback_widget) + snd_soc_dapm_free_widget(dai->playback_widget); + if (dai->capture_widget) + snd_soc_dapm_free_widget(dai->capture_widget); + snd_soc_unregister_dai(dai); + break; + } + } +} + +int hda_codec_probe_complete(struct hda_codec *codec) +{ + struct hdac_device *hdev = &codec->core; + struct hdac_bus *bus = hdev->bus; + int ret; + + ret = snd_hda_codec_build_controls(codec); + if (ret < 0) { + dev_err(&hdev->dev, "unable to create controls %d\n", ret); + goto out; + } + + /* Bus suspended codecs as it does not manage their pm */ + pm_runtime_set_active(&hdev->dev); + /* rpm was forbidden in snd_hda_codec_device_new() */ + snd_hda_codec_set_power_save(codec, 2000); + snd_hda_codec_register(codec); +out: + /* Complement pm_runtime_get_sync(bus) in probe */ + pm_runtime_mark_last_busy(bus->dev); + pm_runtime_put_autosuspend(bus->dev); + + return ret; +} +EXPORT_SYMBOL_GPL(hda_codec_probe_complete); + +/* Expects codec with usage_count=1 and status=suspended */ +static int hda_codec_probe(struct snd_soc_component *component) +{ + struct hda_codec *codec = dev_to_hda_codec(component->dev); + struct hdac_device *hdev = &codec->core; + struct hdac_bus *bus = hdev->bus; + struct hdac_ext_link *hlink; + hda_codec_patch_t patch; + int ret; + +#ifdef CONFIG_PM + WARN_ON(atomic_read(&hdev->dev.power.usage_count) != 1 || + !pm_runtime_status_suspended(&hdev->dev)); +#endif + + hlink = snd_hdac_ext_bus_link_at(bus, hdev->addr); + if (!hlink) { + dev_err(&hdev->dev, "hdac link not found\n"); + return -EIO; + } + + pm_runtime_get_sync(bus->dev); + if (hda_codec_is_display(codec)) + snd_hdac_display_power(bus, hdev->addr, true); + snd_hdac_ext_bus_link_get(bus, hlink); + + ret = snd_hda_codec_device_new(codec->bus, component->card->snd_card, hdev->addr, codec, + false); + if (ret < 0) { + dev_err(&hdev->dev, "create hda codec failed: %d\n", ret); + goto device_new_err; + } + + ret = snd_hda_codec_set_name(codec, codec->preset->name); + if (ret < 0) { + dev_err(&hdev->dev, "name failed %s\n", codec->preset->name); + goto err; + } + + ret = snd_hdac_regmap_init(&codec->core); + if (ret < 0) { + dev_err(&hdev->dev, "regmap init failed\n"); + goto err; + } + + patch = (hda_codec_patch_t)codec->preset->driver_data; + if (!patch) { + dev_err(&hdev->dev, "no patch specified?\n"); + ret = -EINVAL; + goto err; + } + + ret = patch(codec); + if (ret < 0) { + dev_err(&hdev->dev, "patch failed %d\n", ret); + goto err; + } + + /* configure codec for 1:1 PCM:DAI mapping */ + codec->mst_no_extra_pcms = 1; + + ret = snd_hda_codec_parse_pcms(codec); + if (ret < 0) { + dev_err(&hdev->dev, "unable to map pcms to dai %d\n", ret); + goto parse_pcms_err; + } + + ret = hda_codec_register_dais(codec, component); + if (ret < 0) { + dev_err(&hdev->dev, "update dais failed: %d\n", ret); + goto parse_pcms_err; + } + + if (!hda_codec_is_display(codec)) { + ret = hda_codec_probe_complete(codec); + if (ret < 0) + goto complete_err; + } + + codec->core.lazy_cache = true; + + return 0; + +complete_err: + hda_codec_unregister_dais(codec, component); +parse_pcms_err: + if (codec->patch_ops.free) + codec->patch_ops.free(codec); +err: + snd_hda_codec_cleanup_for_unbind(codec); +device_new_err: + if (hda_codec_is_display(codec)) + snd_hdac_display_power(bus, hdev->addr, false); + + snd_hdac_ext_bus_link_put(bus, hlink); + + pm_runtime_mark_last_busy(bus->dev); + pm_runtime_put_autosuspend(bus->dev); + return ret; +} + +/* Leaves codec with usage_count=1 and status=suspended */ +static void hda_codec_remove(struct snd_soc_component *component) +{ + struct hda_codec *codec = dev_to_hda_codec(component->dev); + struct hdac_device *hdev = &codec->core; + struct hdac_bus *bus = hdev->bus; + struct hdac_ext_link *hlink; + bool was_registered = codec->registered; + + /* Don't allow any more runtime suspends */ + pm_runtime_forbid(&hdev->dev); + + hda_codec_unregister_dais(codec, component); + + if (codec->patch_ops.free) + codec->patch_ops.free(codec); + + snd_hda_codec_cleanup_for_unbind(codec); + pm_runtime_put_noidle(&hdev->dev); + /* snd_hdac_device_exit() is only called on bus remove */ + pm_runtime_set_suspended(&hdev->dev); + + if (hda_codec_is_display(codec)) + snd_hdac_display_power(bus, hdev->addr, false); + + hlink = snd_hdac_ext_bus_link_at(bus, hdev->addr); + if (hlink) + snd_hdac_ext_bus_link_put(bus, hlink); + /* + * HDMI card's hda_codec_probe_complete() (see late_probe()) may + * not be called due to early error, leaving bus uc unbalanced + */ + if (!was_registered) { + pm_runtime_mark_last_busy(bus->dev); + pm_runtime_put_autosuspend(bus->dev); + } + +#ifdef CONFIG_PM + WARN_ON(atomic_read(&hdev->dev.power.usage_count) != 1 || + !pm_runtime_status_suspended(&hdev->dev)); +#endif +} + +static const struct snd_soc_dapm_route hda_dapm_routes[] = { + {"AIF1TX", NULL, "Codec Input Pin1"}, + {"AIF2TX", NULL, "Codec Input Pin2"}, + {"AIF3TX", NULL, "Codec Input Pin3"}, + + {"Codec Output Pin1", NULL, "AIF1RX"}, + {"Codec Output Pin2", NULL, "AIF2RX"}, + {"Codec Output Pin3", NULL, "AIF3RX"}, +}; + +static const struct snd_soc_dapm_widget hda_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "Analog Codec Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "Digital Codec Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "Alt Analog Codec Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Analog Codec Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "Digital Codec Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "Alt Analog Codec Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Input Pins */ + SND_SOC_DAPM_INPUT("Codec Input Pin1"), + SND_SOC_DAPM_INPUT("Codec Input Pin2"), + SND_SOC_DAPM_INPUT("Codec Input Pin3"), + + /* Output Pins */ + SND_SOC_DAPM_OUTPUT("Codec Output Pin1"), + SND_SOC_DAPM_OUTPUT("Codec Output Pin2"), + SND_SOC_DAPM_OUTPUT("Codec Output Pin3"), +}; + +static struct snd_soc_dai_driver card_binder_dai = { + .id = -1, + .name = "codec-probing-DAI", +}; + +static int hda_hdev_attach(struct hdac_device *hdev) +{ + struct hda_codec *codec = dev_to_hda_codec(&hdev->dev); + struct snd_soc_component_driver *comp_drv; + + comp_drv = devm_kzalloc(&hdev->dev, sizeof(*comp_drv), GFP_KERNEL); + if (!comp_drv) + return -ENOMEM; + + /* + * It's save to rely on dev_name() rather than a copy as component + * driver's lifetime is directly tied to hda codec one + */ + comp_drv->name = dev_name(&hdev->dev); + comp_drv->probe = hda_codec_probe; + comp_drv->remove = hda_codec_remove; + comp_drv->idle_bias_on = false; + if (!hda_codec_is_display(codec)) { + comp_drv->dapm_widgets = hda_dapm_widgets; + comp_drv->num_dapm_widgets = ARRAY_SIZE(hda_dapm_widgets); + comp_drv->dapm_routes = hda_dapm_routes; + comp_drv->num_dapm_routes = ARRAY_SIZE(hda_dapm_routes); + } + + return snd_soc_register_component(&hdev->dev, comp_drv, &card_binder_dai, 1); +} + +static int hda_hdev_detach(struct hdac_device *hdev) +{ + struct hda_codec *codec = dev_to_hda_codec(&hdev->dev); + + if (codec->registered) + cancel_delayed_work_sync(&codec->jackpoll_work); + + snd_soc_unregister_component(&hdev->dev); + + return 0; +} + +const struct hdac_ext_bus_ops soc_hda_ext_bus_ops = { + .hdev_attach = hda_hdev_attach, + .hdev_detach = hda_hdev_detach, +}; +EXPORT_SYMBOL_GPL(soc_hda_ext_bus_ops); + +MODULE_DESCRIPTION("HD-Audio codec driver"); +MODULE_AUTHOR("Cezary Rojewski cezary.rojewski@intel.com"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/hda.h b/sound/soc/codecs/hda.h new file mode 100644 index 000000000000..78a2be4945b1 --- /dev/null +++ b/sound/soc/codecs/hda.h @@ -0,0 +1,19 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright(c) 2021-2022 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski cezary.rojewski@intel.com + */ + +#ifndef SND_SOC_CODECS_HDA_H +#define SND_SOC_CODECS_HDA_H + +#define hda_codec_is_display(codec) \ + ((((codec)->core.vendor_id >> 16) & 0xFFFF) == 0x8086) + +extern const struct snd_soc_dai_ops snd_soc_hda_codec_dai_ops; + +extern const struct hdac_ext_bus_ops soc_hda_ext_bus_ops; +int hda_codec_probe_complete(struct hda_codec *codec); + +#endif
Hi Cezary,
On Wed, 11 May 2022, Cezary Rojewski wrote:
Add generic ASoC equivalent of ALSA HD-Audio codec. This codec is designed to follow HDA_DEV_LEGACY convention. Driver wrapps existing hda_codec.c handlers to prevent code duplication within the newly added code. Number of DAIs created is dependent on capabilities exposed by the codec itself. Because of this, single solution can be applied to support every single HD-Audio codec type.
At the same time, through the ASoC topology, platform drivers may limit the number of endpoints available to the userspace as codec driver exposes BE DAIs only.
Both hda_codec_probe() and hda_codec_remove() declare their expectations on device's usage_count and suspended-status. This is to catch any unexpected behavior as PM-related code for HD-Audio has been changing quite a bit throughout the years.
In order for codec DAI list to reflect its actual PCM capabilities, PCMs need to be built and that can only happen once codec device is constructed. To do that, a valid component->card->snd_card pointer is needed. Said pointer will be provided by the framework once all card components are accounted for and their probing can begin. Usage of "binder" BE DAI solves the problem - codec can be listed as one of HD-Audio card components without declaring any actual BE DAIs statically.
Relation with hdac_hda:
Addition of parallel solution is motivated by behavioral differences between hdac_hda.c and its legacy equivalent found in sound/pci/hda e.g.: lack of dynamic, based on codec capabilities, resource allocation and high cost of removing such differences on actively used targets. Major goal of codec driver presented here is to follow HD-Audio legacy behavior in 1:1 fashion by becoming a wrapper. Doing so increases code coverage of the legacy code and reduces the maintenance cost for both solutions.
Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com
Thanks for your patch, which is now commit b5df2a7dca1cc6c6 ("ASoC: codecs: Add HD-Audio codec driver") in sound-asoc/for-next.
--- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -937,6 +937,16 @@ config SND_SOC_HDAC_HDA tristate select SND_HDA
+config SND_SOC_HDA
- tristate "HD-Audio codec driver"
- select SND_HDA_EXT_CORE
- select SND_HDA
I am wondering if this needs a platform dependency? Or perhaps this symbol should be made invisible, as it is selected by SND_SOC_INTEL_AVS_MACH_HDAUDIO? Are there any other users?
Thanks!
- help
This enables HD-Audio codec support in ASoC subsystem. Compared
to SND_SOC_HDAC_HDA, driver's behavior is identical to HD-Audio
legacy solution - including the dynamic resource allocation
based on actual codec capabilities.
config SND_SOC_ICS43432 tristate "ICS43423 and compatible i2s microphones"
Gr{oetje,eeting}s,
Geert
-- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert@linux-m68k.org
In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds
On 2022-07-05 12:20 PM, Geert Uytterhoeven wrote:
Hi Cezary,
On Wed, 11 May 2022, Cezary Rojewski wrote:
...
Relation with hdac_hda:
Addition of parallel solution is motivated by behavioral differences between hdac_hda.c and its legacy equivalent found in sound/pci/hda e.g.: lack of dynamic, based on codec capabilities, resource allocation and high cost of removing such differences on actively used targets. Major goal of codec driver presented here is to follow HD-Audio legacy behavior in 1:1 fashion by becoming a wrapper. Doing so increases code coverage of the legacy code and reduces the maintenance cost for both solutions.
Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com
Thanks for your patch, which is now commit b5df2a7dca1cc6c6 ("ASoC: codecs: Add HD-Audio codec driver") in sound-asoc/for-next.
--- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -937,6 +937,16 @@ config SND_SOC_HDAC_HDA tristate select SND_HDA
+config SND_SOC_HDA + tristate "HD-Audio codec driver" + select SND_HDA_EXT_CORE + select SND_HDA
I am wondering if this needs a platform dependency? Or perhaps this symbol should be made invisible, as it is selected by SND_SOC_INTEL_AVS_MACH_HDAUDIO? Are there any other users?
Your feedback is much appreciated!
So, this codec-driver - HDAudio codec driver to be strict - is _generic_ i.e. works with any HDAudio codec type, be it Cirrus, Maxim, Conexant or anything else.
It makes heavy use of HDAudio library found in sound/hda/ and re-uses a lot of non-DSP HDAudio related code found in sound/pci/hda. It's basically a wrapper for logic present in ALSA (sound/) so ASoC (sound/soc/) users can have first class HDAudio codec support without us duplicating the entire implementation.
Right now the only user is avs-driver found in sound/soc/intel/avs. By design, any HDAudio bus driver in ASoC framework can make use of it though. Because of that, we decided not to couple avs-driver and this HDAudio codec driver tightly.
Gr{oetje,eeting}s,
This made my day :) it's wonderful!
Connect AVS driver with ASoC HDAudio codec with help of this machine board. Similarly to its platform and codec components, DAI links and routes are being created dynamically so single board can be used across all HDAudio codec types.
Card makes use of "binder" BE DAI Link so HDAudio codec driver can be listed as one of its components. This allows for BE DAIs to be created dynamically, based on HDAudio codec capabilities.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/Kconfig | 3 + sound/soc/intel/avs/Makefile | 3 + sound/soc/intel/avs/boards/Kconfig | 15 ++ sound/soc/intel/avs/boards/Makefile | 5 + sound/soc/intel/avs/boards/hdaudio.c | 294 +++++++++++++++++++++++++++ 5 files changed, 320 insertions(+) create mode 100644 sound/soc/intel/avs/boards/Kconfig create mode 100644 sound/soc/intel/avs/boards/Makefile create mode 100644 sound/soc/intel/avs/boards/hdaudio.c
diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 7c85d1bb9c12..e5107a3ce16a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -226,5 +226,8 @@ config SND_SOC_INTEL_AVS capabilities. This includes Skylake, Kabylake, Amberlake and Apollolake.
+# Machine board drivers +source "sound/soc/intel/avs/boards/Kconfig" + # ASoC codec drivers source "sound/soc/intel/boards/Kconfig" diff --git a/sound/soc/intel/avs/Makefile b/sound/soc/intel/avs/Makefile index b6b93ae80304..919212825f21 100644 --- a/sound/soc/intel/avs/Makefile +++ b/sound/soc/intel/avs/Makefile @@ -10,3 +10,6 @@ snd-soc-avs-objs += trace.o CFLAGS_trace.o := -I$(src)
obj-$(CONFIG_SND_SOC_INTEL_AVS) += snd-soc-avs.o + +# Machine support +obj-$(CONFIG_SND_SOC) += boards/ diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig new file mode 100644 index 000000000000..de62c0437f6e --- /dev/null +++ b/sound/soc/intel/avs/boards/Kconfig @@ -0,0 +1,15 @@ +# SPDX-License-Identifier: GPL-2.0-only +menu "Intel AVS Machine drivers" + depends on SND_SOC_INTEL_AVS + +comment "Available DSP configurations" + +config SND_SOC_INTEL_AVS_MACH_HDAUDIO + tristate "HD-Audio generic board" + select SND_SOC_HDA + help + This adds support for AVS with HDAudio codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + +endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile new file mode 100644 index 000000000000..e5281148e5d4 --- /dev/null +++ b/sound/soc/intel/avs/boards/Makefile @@ -0,0 +1,5 @@ +# SPDX-License-Identifier: GPL-2.0-only + +snd-soc-avs-hdaudio-objs := hdaudio.o + +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o diff --git a/sound/soc/intel/avs/boards/hdaudio.c b/sound/soc/intel/avs/boards/hdaudio.c new file mode 100644 index 000000000000..d2fc41d39448 --- /dev/null +++ b/sound/soc/intel/avs/boards/hdaudio.c @@ -0,0 +1,294 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/platform_device.h> +#include <sound/hda_codec.h> +#include <sound/hda_i915.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/hda.h" + +static int avs_create_dai_links(struct device *dev, struct hda_codec *codec, int pcm_count, + const char *platform_name, struct snd_soc_dai_link **links) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + struct hda_pcm *pcm; + const char *cname = dev_name(&codec->core.dev); + int i; + + dl = devm_kcalloc(dev, pcm_count, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + pcm = list_first_entry(&codec->pcm_list_head, struct hda_pcm, list); + + for (i = 0; i < pcm_count; i++, pcm = list_next_entry(pcm, list)) { + dl[i].name = devm_kasprintf(dev, GFP_KERNEL, "%s link%d", cname, i); + if (!dl[i].name) + return -ENOMEM; + + dl[i].id = i; + dl[i].nonatomic = 1; + dl[i].no_pcm = 1; + dl[i].dpcm_playback = 1; + dl[i].dpcm_capture = 1; + dl[i].platforms = platform; + dl[i].num_platforms = 1; + + dl[i].codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + dl[i].cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + if (!dl[i].codecs || !dl[i].cpus) + return -ENOMEM; + + dl[i].cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "%s-cpu%d", cname, i); + if (!dl[i].cpus->dai_name) + return -ENOMEM; + + dl[i].codecs->name = devm_kstrdup(dev, cname, GFP_KERNEL); + dl[i].codecs->dai_name = pcm->name; + dl[i].num_codecs = 1; + dl[i].num_cpus = 1; + } + + *links = dl; + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, struct hda_codec *codec, int pcm_count, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + struct hda_pcm *pcm; + const char *cname = dev_name(&codec->core.dev); + int i, n = 0; + + /* at max twice the number of pcms */ + dr = devm_kcalloc(dev, pcm_count * 2, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + pcm = list_first_entry(&codec->pcm_list_head, struct hda_pcm, list); + + for (i = 0; i < pcm_count; i++, pcm = list_next_entry(pcm, list)) { + struct hda_pcm_stream *stream; + int dir; + + dir = SNDRV_PCM_STREAM_PLAYBACK; + stream = &pcm->stream[dir]; + if (!stream->substreams) + goto capture_routes; + + dr[n].sink = devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + dr[n].source = devm_kasprintf(dev, GFP_KERNEL, "%s-cpu%d Tx", cname, i); + if (!dr[n].sink || !dr[n].source) + return -ENOMEM; + n++; + +capture_routes: + dir = SNDRV_PCM_STREAM_CAPTURE; + stream = &pcm->stream[dir]; + if (!stream->substreams) + continue; + + dr[n].sink = devm_kasprintf(dev, GFP_KERNEL, "%s-cpu%d Rx", cname, i); + dr[n].source = devm_kasprintf(dev, GFP_KERNEL, "%s %s", pcm->name, + snd_pcm_direction_name(dir)); + if (!dr[n].sink || !dr[n].source) + return -ENOMEM; + n++; + } + + *routes = dr; + *num_routes = n; + return 0; +} + +/* Should be aligned with SectionPCM's name from topology */ +#define FEDAI_NAME_PREFIX "HDMI" + +static struct snd_pcm * +avs_card_hdmi_pcm_at(struct snd_soc_card *card, int hdmi_idx) +{ + struct snd_soc_pcm_runtime *rtd; + int dir = SNDRV_PCM_STREAM_PLAYBACK; + + for_each_card_rtds(card, rtd) { + struct snd_pcm *spcm; + int ret, n; + + spcm = rtd->pcm ? rtd->pcm->streams[dir].pcm : NULL; + if (!spcm || !strstr(spcm->id, FEDAI_NAME_PREFIX)) + continue; + + ret = sscanf(spcm->id, FEDAI_NAME_PREFIX "%d", &n); + if (ret != 1) + continue; + if (n == hdmi_idx) + return rtd->pcm; + } + + return NULL; +} + +static int avs_card_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); + struct hda_codec *codec = mach->pdata; + struct hda_pcm *hpcm; + /* Topology pcm indexing is 1-based */ + int i = 1; + + list_for_each_entry(hpcm, &codec->pcm_list_head, list) { + struct snd_pcm *spcm; + + spcm = avs_card_hdmi_pcm_at(card, i); + if (spcm) { + hpcm->pcm = spcm; + hpcm->device = spcm->device; + dev_info(card->dev, "%s: mapping HDMI converter %d to PCM %d (%p)\n", + __func__, i, hpcm->device, spcm); + } else { + hpcm->pcm = NULL; + hpcm->device = SNDRV_PCM_INVALID_DEVICE; + dev_warn(card->dev, "%s: no PCM in topology for HDMI converter %d\n", + __func__, i); + } + i++; + } + + return hda_codec_probe_complete(codec); +} + +static int avs_probing_link_init(struct snd_soc_pcm_runtime *rtm) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_acpi_mach *mach; + struct snd_soc_dai_link *links = NULL; + struct snd_soc_card *card = rtm->card; + struct hda_codec *codec; + struct hda_pcm *pcm; + int ret, n, pcm_count = 0; + + mach = dev_get_platdata(card->dev); + codec = mach->pdata; + + if (list_empty(&codec->pcm_list_head)) + return -EINVAL; + list_for_each_entry(pcm, &codec->pcm_list_head, list) + pcm_count++; + + ret = avs_create_dai_links(card->dev, codec, pcm_count, mach->mach_params.platform, &links); + if (ret < 0) { + dev_err(card->dev, "create links failed: %d\n", ret); + return ret; + } + + for (n = 0; n < pcm_count; n++) { + ret = snd_soc_add_pcm_runtime(card, &links[n]); + if (ret < 0) { + dev_err(card->dev, "add links failed: %d\n", ret); + return ret; + } + } + + ret = avs_create_dapm_routes(card->dev, codec, pcm_count, &routes, &n); + if (ret < 0) { + dev_err(card->dev, "create routes failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, routes, n); + if (ret < 0) { + dev_err(card->dev, "add routes failed: %d\n", ret); + return ret; + } + + return 0; +} + +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); + +static struct snd_soc_dai_link probing_link = { + .name = "probing-LINK", + .id = -1, + .nonatomic = 1, + .no_pcm = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .cpus = dummy, + .num_cpus = ARRAY_SIZE(dummy), + .init = avs_probing_link_init, +}; + +static int avs_hdaudio_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *binder; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + struct hda_codec *codec; + + mach = dev_get_platdata(dev); + codec = mach->pdata; + + /* codec may be unloaded before card's probe() fires */ + if (!device_is_registered(&codec->core.dev)) + return -ENODEV; + + binder = devm_kmemdup(dev, &probing_link, sizeof(probing_link), GFP_KERNEL); + if (!binder) + return -ENOMEM; + + binder->platforms = devm_kzalloc(dev, sizeof(*binder->platforms), GFP_KERNEL); + binder->codecs = devm_kzalloc(dev, sizeof(*binder->codecs), GFP_KERNEL); + if (!binder->platforms || !binder->codecs) + return -ENOMEM; + + binder->codecs->name = devm_kstrdup(dev, dev_name(&codec->core.dev), GFP_KERNEL); + if (!binder->codecs->name) + return -ENOMEM; + + binder->platforms->name = mach->mach_params.platform; + binder->num_platforms = 1; + binder->codecs->dai_name = "codec-probing-DAI"; + binder->num_codecs = 1; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = binder->codecs->name; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = binder; + card->num_links = 1; + card->fully_routed = true; + if (hda_codec_is_display(codec)) + card->late_probe = avs_card_late_probe; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_hdaudio_driver = { + .probe = avs_hdaudio_probe, + .driver = { + .name = "avs_hdaudio", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_hdaudio_driver) + +MODULE_DESCRIPTION("Intel HD-Audio machine driver"); +MODULE_AUTHOR("Cezary Rojewski cezary.rojewski@intel.com"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_hdaudio");
Support AVS-DMIC configuration by implementing board connecting AVS platform component with DMIC codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 8 +++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/dmic.c | 93 +++++++++++++++++++++++++++++ 3 files changed, 103 insertions(+) create mode 100644 sound/soc/intel/avs/boards/dmic.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index de62c0437f6e..1d4597fa9814 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -4,6 +4,14 @@ menu "Intel AVS Machine drivers"
comment "Available DSP configurations"
+config SND_SOC_INTEL_AVS_MACH_DMIC + tristate "DMIC generic board" + select SND_SOC_DMIC + help + This adds support for AVS with Digital Mic array configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_HDAUDIO tristate "HD-Audio generic board" select SND_SOC_HDA diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index e5281148e5d4..2ff35d4d97d8 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -1,5 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only
+snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o
+obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o diff --git a/sound/soc/intel/avs/boards/dmic.c b/sound/soc/intel/avs/boards/dmic.c new file mode 100644 index 000000000000..90a921638572 --- /dev/null +++ b/sound/soc/intel/avs/boards/dmic.c @@ -0,0 +1,93 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/device.h> +#include <linux/module.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> + +SND_SOC_DAILINK_DEF(dmic_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); +SND_SOC_DAILINK_DEF(dmic_wov_pin, DAILINK_COMP_ARRAY(COMP_CPU("DMIC WoV Pin"))); +SND_SOC_DAILINK_DEF(dmic_codec, DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); +/* Name overridden on probe */ +SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM(""))); + +static struct snd_soc_dai_link card_dai_links[] = { + /* Back ends */ + { + .name = "DMIC", + .id = 0, + .dpcm_capture = 1, + .nonatomic = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), + }, + { + .name = "DMIC WoV", + .id = 1, + .dpcm_capture = 1, + .nonatomic = 1, + .no_pcm = 1, + .ignore_suspend = 1, + SND_SOC_DAILINK_REG(dmic_wov_pin, dmic_codec, platform), + }, +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route card_routes[] = { + {"DMic", NULL, "SoC DMIC"}, + {"DMIC Rx", NULL, "Capture"}, + {"DMIC WoV Rx", NULL, "Capture"}, +}; + +static int avs_dmic_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + int ret; + + mach = dev_get_platdata(dev); + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_dmic"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = card_dai_links; + card->num_links = ARRAY_SIZE(card_dai_links); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = card_routes; + card->num_dapm_routes = ARRAY_SIZE(card_routes); + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, mach->mach_params.platform); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_dmic_driver = { + .probe = avs_dmic_probe, + .driver = { + .name = "avs_dmic", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_dmic_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_dmic");
Allow for testing audio streaming over I2S interface through SSP loopback. No actual codec is needed here as board is intended for SSP loopback scenarios only. One playback and one capture endpoint is exposed per SSP port.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 6 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/i2s_test.c | 180 ++++++++++++++++++++++++++ 3 files changed, 188 insertions(+) create mode 100644 sound/soc/intel/avs/boards/i2s_test.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 1d4597fa9814..5b89fcb5f07f 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -20,4 +20,10 @@ config SND_SOC_INTEL_AVS_MACH_HDAUDIO Say Y or m if you have such a device. This is a recommended option. If unsure select "N".
+config SND_SOC_INTEL_AVS_MACH_I2S_TEST + tristate "I2S test board" + help + This adds support for I2S test-board which can be used to verify + transfer over I2S interface with SSP loopback scenarios. + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 2ff35d4d97d8..fa1a279106be 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -2,6 +2,8 @@
snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o +snd-soc-avs-i2s-test-objs := i2s_test.o
obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o diff --git a/sound/soc/intel/avs/boards/i2s_test.c b/sound/soc/intel/avs/boards/i2s_test.c new file mode 100644 index 000000000000..461b651cd331 --- /dev/null +++ b/sound/soc/intel/avs/boards/i2s_test.c @@ -0,0 +1,180 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/module.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "snd-soc-dummy"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "snd-soc-dummy-dai"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_dr = 2; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + dr[0].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%dpb", ssp_port); + dr[0].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[0].sink || !dr[0].source) + return -ENOMEM; + + dr[1].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[1].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%dcp", ssp_port); + if (!dr[1].sink || !dr[1].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_create_dapm_widgets(struct device *dev, int ssp_port, + struct snd_soc_dapm_widget **widgets, int *num_widgets) +{ + struct snd_soc_dapm_widget *dw; + const int num_dw = 2; + + dw = devm_kcalloc(dev, num_dw, sizeof(*dw), GFP_KERNEL); + if (!dw) + return -ENOMEM; + + dw[0].id = snd_soc_dapm_hp; + dw[0].reg = SND_SOC_NOPM; + dw[0].name = devm_kasprintf(dev, GFP_KERNEL, "ssp%dpb", ssp_port); + if (!dw[0].name) + return -ENOMEM; + + dw[1].id = snd_soc_dapm_mic; + dw[1].reg = SND_SOC_NOPM; + dw[1].name = devm_kasprintf(dev, GFP_KERNEL, "ssp%dcp", ssp_port); + if (!dw[1].name) + return -ENOMEM; + + *widgets = dw; + *num_widgets = num_dw; + + return 0; +} + +static int avs_i2s_test_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_widget *widgets; + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, num_widgets; + int ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = devm_kasprintf(dev, GFP_KERNEL, "ssp%ld-loopback", ssp_port); + if (!card->name) + return -ENOMEM; + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d\n", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d\n", ret); + return ret; + } + + ret = avs_create_dapm_widgets(dev, ssp_port, &widgets, &num_widgets); + if (ret) { + dev_err(dev, "Failed to create dapm widgets: %d\n", ret); + return ret; + } + + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->dapm_widgets = widgets; + card->num_dapm_widgets = num_widgets; + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_i2s_test_driver = { + .probe = avs_i2s_test_probe, + .driver = { + .name = "avs_i2s_test", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_i2s_test_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_i2s_test");
To support AVS-rt274 configuration add machine board connecting AVS platform component driver with rt274 codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 10 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt274.c | 310 ++++++++++++++++++++++++++++ 3 files changed, 322 insertions(+) create mode 100644 sound/soc/intel/avs/boards/rt274.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 5b89fcb5f07f..9058919c99a7 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -26,4 +26,14 @@ config SND_SOC_INTEL_AVS_MACH_I2S_TEST This adds support for I2S test-board which can be used to verify transfer over I2S interface with SSP loopback scenarios.
+config SND_SOC_INTEL_AVS_MACH_RT274 + tristate "rt274 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT274 + help + This adds support for ASoC machine driver with RT274 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index fa1a279106be..e94f04d00ffc 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -3,7 +3,9 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o +snd-soc-avs-rt274-objs := rt274.o
obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o diff --git a/sound/soc/intel/avs/boards/rt274.c b/sound/soc/intel/avs/boards/rt274.c new file mode 100644 index 000000000000..afef5a3ca60b --- /dev/null +++ b/sound/soc/intel/avs/boards/rt274.c @@ -0,0 +1,310 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/rt274.h" + +#define AVS_RT274_FREQ_OUT 24000000 +#define AVS_RT274_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static int +avs_rt274_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, AVS_RT274_FREQ_OUT, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + int ratio = 100; + + snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + AVS_RT274_BE_FIXUP_RATE * ratio, AVS_RT274_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, "failed to enable PLL2: %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, avs_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"MIC", NULL, "Platform Clock"}, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_rt274_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET, jack, pins, num_pins); + if (ret) + return ret; + + snd_soc_component_set_jack(component, jack, NULL); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(card->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int avs_rt274_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = AVS_RT274_BE_FIXUP_RATE; + channels->min = channels->max = 2; + + /* set SSPN to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT34C2:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt274-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt274_codec_init; + dl->be_hw_params_fixup = avs_rt274_be_fixup; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt274_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt274"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt274_driver = { + .probe = avs_rt274_probe, + .driver = { + .name = "avs_rt274", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt274_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt274");
To support AVS-rt286 configuration add machine board connecting AVS platform component driver with rt286 codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 10 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt286.c | 281 ++++++++++++++++++++++++++++ 3 files changed, 293 insertions(+) create mode 100644 sound/soc/intel/avs/boards/rt286.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 9058919c99a7..707e9e96746d 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -36,4 +36,14 @@ config SND_SOC_INTEL_AVS_MACH_RT274 Say Y or m if you have such a device. This is a recommended option. If unsure select "N".
+config SND_SOC_INTEL_AVS_MACH_RT286 + tristate "rt286 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT286 + help + This adds support for ASoC machine driver with RT286 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index e94f04d00ffc..7ea4ad38c7df 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -4,8 +4,10 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o snd-soc-avs-rt274-objs := rt274.o +snd-soc-avs-rt286-objs := rt286.o
obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o diff --git a/sound/soc/intel/avs/boards/rt286.c b/sound/soc/intel/avs/boards/rt286.c new file mode 100644 index 000000000000..e51d4e181274 --- /dev/null +++ b/sound/soc/intel/avs/boards/rt286.c @@ -0,0 +1,281 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/rt286.h" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detect */ + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC1", NULL, "Mic Jack"}, + + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_rt286_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, + pins, num_pins); + if (ret) + return ret; + + snd_soc_component_set_jack(component, jack, NULL); + + return 0; +} + +static int avs_rt286_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int +avs_rt286_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(runtime->dev, "Set codec sysclk failed: %d\n", ret); + + return ret; +} + +static const struct snd_soc_ops avs_rt286_ops = { + .hw_params = avs_rt286_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343A:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt286-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt286_codec_init; + dl->be_hw_params_fixup = avs_rt286_be_fixup; + dl->ops = &avs_rt286_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt286_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt286"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt286_driver = { + .probe = avs_rt286_probe, + .driver = { + .name = "avs_rt286", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt286_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt286");
To support AVS-rt298 configuration add machine board connecting AVS platform component driver with rt298 codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 10 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt298.c | 281 ++++++++++++++++++++++++++++ 3 files changed, 293 insertions(+) create mode 100644 sound/soc/intel/avs/boards/rt298.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 707e9e96746d..b4dc2b02097d 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -46,4 +46,14 @@ config SND_SOC_INTEL_AVS_MACH_RT286 Say Y or m if you have such a device. This is a recommended option. If unsure select "N".
+config SND_SOC_INTEL_AVS_MACH_RT298 + tristate "rt298 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT298 + help + This adds support for ASoC machine driver with RT298 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 7ea4ad38c7df..0fd664694c8c 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -5,9 +5,11 @@ snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o +snd-soc-avs-rt298-objs := rt298.o
obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o diff --git a/sound/soc/intel/avs/boards/rt298.c b/sound/soc/intel/avs/boards/rt298.c new file mode 100644 index 000000000000..b28d36872dcb --- /dev/null +++ b/sound/soc/intel/avs/boards/rt298.c @@ -0,0 +1,281 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/rt298.h" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detect */ + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC1", NULL, "Mic Jack"}, + + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_rt298_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, jack, + pins, num_pins); + if (ret) + return ret; + + snd_soc_component_set_jack(component, jack, NULL); + + return 0; +} + +static int avs_rt298_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int +avs_rt298_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL, 19200000, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "Set codec sysclk failed: %d\n", ret); + + return ret; +} + +static const struct snd_soc_ops avs_rt298_ops = { + .hw_params = avs_rt298_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343A:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt298-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_rt298_codec_init; + dl->be_hw_params_fixup = avs_rt298_be_fixup; + dl->ops = &avs_rt298_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt298_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt298"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt298_driver = { + .probe = avs_rt298_probe, + .driver = { + .name = "avs_rt298", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt298_driver); + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt298");
To support AVS-rt5682 configuration add machine board connecting AVS platform component driver with rt5682 codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 10 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/rt5682.c | 340 ++++++++++++++++++++++++++++ 3 files changed, 352 insertions(+) create mode 100644 sound/soc/intel/avs/boards/rt5682.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index b4dc2b02097d..767eae57be57 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -56,4 +56,14 @@ config SND_SOC_INTEL_AVS_MACH_RT298 Say Y or m if you have such a device. This is a recommended option. If unsure select "N".
+config SND_SOC_INTEL_AVS_MACH_RT5682 + tristate "rt5682 in I2S mode" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_RT5682_I2C + help + This adds support for ASoC machine driver with RT5682 I2S audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 0fd664694c8c..fd49bd2a5876 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-avs-i2s-test-objs := i2s_test.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o snd-soc-avs-rt298-objs := rt298.o +snd-soc-avs-rt5682-objs := rt5682.o
obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o @@ -13,3 +14,4 @@ obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT5682) += snd-soc-avs-rt5682.o diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c new file mode 100644 index 000000000000..01f9b9f0c12b --- /dev/null +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -0,0 +1,340 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/clk.h> +#include <linux/dmi.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/rt5682.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../common/soc-intel-quirks.h" +#include "../../../codecs/rt5682.h" + +#define AVS_RT5682_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) +#define AVS_RT5682_SSP_CODEC_MASK (GENMASK(2, 0)) +#define AVS_RT5682_MCLK_EN BIT(3) +#define AVS_RT5682_MCLK_24MHZ BIT(4) + +/* Default: MCLK on, MCLK 19.2M, SSP0 */ +static unsigned long avs_rt5682_quirk = AVS_RT5682_MCLK_EN | AVS_RT5682_SSP_CODEC(0); + +static int avs_rt5682_quirk_cb(const struct dmi_system_id *id) +{ + avs_rt5682_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id avs_rt5682_quirk_table[] = { + { + .callback = avs_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "WhiskeyLake Client"), + }, + .driver_data = (void *)(AVS_RT5682_MCLK_EN | + AVS_RT5682_MCLK_24MHZ | + AVS_RT5682_SSP_CODEC(1)), + }, + { + .callback = avs_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), + }, + .driver_data = (void *)(AVS_RT5682_MCLK_EN | + AVS_RT5682_SSP_CODEC(0)), + }, + {} +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detect */ + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* other jacks */ + { "IN1P", NULL, "Headset Mic" }, +}; + +static int avs_rt5682_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int ret; + + jack = snd_soc_card_get_drvdata(card); + + /* Need to enable ASRC function for 24MHz mclk rate */ + if ((avs_rt5682_quirk & AVS_RT5682_MCLK_EN) && + (avs_rt5682_quirk & AVS_RT5682_MCLK_24MHZ)) { + rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | + RT5682_AD_STEREO1_FILTER, RT5682_CLK_SEL_I2S1_ASRC); + } + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, jack); + if (ret) { + dev_err(card->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + ret = snd_soc_component_set_jack(component, jack, NULL); + if (ret) { + dev_err(card->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return 0; +}; + +static int +avs_rt5682_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + int clk_id, clk_freq; + int pll_out, ret; + + if (avs_rt5682_quirk & AVS_RT5682_MCLK_EN) { + clk_id = RT5682_PLL1_S_MCLK; + if (avs_rt5682_quirk & AVS_RT5682_MCLK_24MHZ) + clk_freq = 24000000; + else + clk_freq = 19200000; + } else { + clk_id = RT5682_PLL1_S_BCLK1; + clk_freq = params_rate(params) * 50; + } + + pll_out = params_rate(params) * 512; + + ret = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, pll_out); + if (ret < 0) + dev_err(runtime->dev, "snd_soc_dai_set_pll err = %d\n", ret); + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(runtime->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* slot_width should equal or large than data length, set them be the same */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, params_width(params)); + if (ret < 0) { + dev_err(runtime->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops avs_rt5682_ops = { + .hw_params = avs_rt5682_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-10EC5682:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "rt5682-aif1"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->init = avs_rt5682_codec_init; + dl->ops = &avs_rt5682_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "AIF1 Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_rt5682_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + if (pdev->id_entry && pdev->id_entry->driver_data) + avs_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data; + + dmi_check_system(avs_rt5682_quirk_table); + dev_dbg(dev, "avs_rt5682_quirk = %lx\n", avs_rt5682_quirk); + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_rt5682"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_rt5682_driver = { + .probe = avs_rt5682_probe, + .driver = { + .name = "avs_rt5682", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_rt5682_driver) + +MODULE_AUTHOR("Cezary Rojewski cezary.rojewski@intel.com"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_rt5682");
To support AVS-nau8825 configuration add machine board connecting AVS platform component driver with nau8825 codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 11 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/nau8825.c | 353 +++++++++++++++++++++++++++ 3 files changed, 366 insertions(+) create mode 100644 sound/soc/intel/avs/boards/nau8825.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 767eae57be57..6bf8fa1924a2 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -26,6 +26,17 @@ config SND_SOC_INTEL_AVS_MACH_I2S_TEST This adds support for I2S test-board which can be used to verify transfer over I2S interface with SSP loopback scenarios.
+config SND_SOC_INTEL_AVS_MACH_NAU8825 + tristate "nau8825 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_NAU8825 + help + This adds support for ASoC machine driver with NAU8825 I2S audio codec. + It is meant to be used with AVS driver. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_RT274 tristate "rt274 in I2S mode" depends on I2C diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index fd49bd2a5876..9ac14b269f56 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -3,6 +3,7 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o +snd-soc-avs-nau8825-objs := nau8825.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o snd-soc-avs-rt298-objs := rt298.o @@ -11,6 +12,7 @@ snd-soc-avs-rt5682-objs := rt5682.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_NAU8825) += snd-soc-avs-nau8825.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c new file mode 100644 index 000000000000..f76909e9f990 --- /dev/null +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -0,0 +1,353 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/nau8825.h" + +#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" + +static int +avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EINVAL; + } + + if (!SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "set sysclk err = %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, avs_nau8825_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + { "MIC", NULL, "Headset Mic" }, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static int avs_nau8825_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_jack_pin *pins; + struct snd_soc_jack *jack; + struct snd_soc_card *card = runtime->card; + int num_pins, ret; + + jack = snd_soc_card_get_drvdata(card); + num_pins = ARRAY_SIZE(card_headset_pins); + + pins = devm_kmemdup(card->dev, card_headset_pins, sizeof(*pins) * num_pins, GFP_KERNEL); + if (!pins) + return -ENOMEM; + + /* + * 4 buttons here map to the google Reference headset. + * The use of these buttons can be decided by the user space. + */ + ret = snd_soc_card_jack_new_pins(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, + jack, pins, num_pins); + if (ret) + return ret; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + //snd_soc_component_set_jack(component, jack, NULL); + // TODO: Fix nau8825 codec to use .set_jack, like everyone else + nau8825_enable_jack_detect(component, jack); + + return 0; +} + +static int +avs_nau8825_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int avs_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtm, 0); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + + case SNDRV_PCM_TRIGGER_RESUME: + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } + + return ret; +} + + +static const struct snd_soc_ops avs_nau8825_ops = { + .trigger = avs_nau8825_trigger, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-10508825:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, SKL_NUVOTON_CODEC_DAI); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_nau8825_codec_init; + dl->be_hw_params_fixup = avs_nau8825_be_fixup; + dl->ops = &avs_nau8825_ops; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EINVAL; + } + + if (codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] && + codec_dai->playback_widget->active) + snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, SND_SOC_CLOCK_IN); + + return avs_card_set_jack(card, jack); +} + +static int avs_nau8825_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_nau8825"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_nau8825_driver = { + .probe = avs_nau8825_probe, + .driver = { + .name = "avs_nau8825", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_nau8825_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_nau8825");
To support AVS-ssm4567 configuration add machine board connecting AVS platform component driver with ssm4567 codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 11 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/ssm4567.c | 271 +++++++++++++++++++++++++++ 3 files changed, 284 insertions(+) create mode 100644 sound/soc/intel/avs/boards/ssm4567.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 6bf8fa1924a2..7020e7bf196e 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -77,4 +77,15 @@ config SND_SOC_INTEL_AVS_MACH_RT5682 Say Y or m if you have such a device. This is a recommended option. If unsure select "N".
+config SND_SOC_INTEL_AVS_MACH_SSM4567 + tristate "ssm4567 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_SSM4567 + help + This adds support for ASoC machine driver with SSM4567 I2S audio codec. + It is meant to be used with AVS driver. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + endmenu diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 9ac14b269f56..ea67fc711d9d 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -8,6 +8,7 @@ snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o snd-soc-avs-rt298-objs := rt298.o snd-soc-avs-rt5682-objs := rt5682.o +snd-soc-avs-ssm4567-objs := ssm4567.o
obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o @@ -17,3 +18,4 @@ obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT298) += snd-soc-avs-rt298.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT5682) += snd-soc-avs-rt5682.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_SSM4567) += snd-soc-avs-ssm4567.o diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c new file mode 100644 index 000000000000..9f84c8ab3447 --- /dev/null +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -0,0 +1,271 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include "../../../codecs/nau8825.h" + +#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" +#define SKL_SSM_CODEC_DAI "ssm4567-hifi" + +static struct snd_soc_codec_conf card_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF("i2c-INT343B:00"), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF("i2c-INT343B:01"), + .name_prefix = "Right", + }, +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Speaker"), + SOC_DAPM_PIN_SWITCH("Right Speaker"), +}; + +static int +platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EINVAL; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "set sysclk err = %d\n", ret); + } else { + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "set sysclk err = %d\n", ret); + } + + return ret; +} + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_SPK("Left Speaker", NULL), + SND_SOC_DAPM_SPK("Right Speaker", NULL), + SND_SOC_DAPM_SPK("DP1", NULL), + SND_SOC_DAPM_SPK("DP2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + {"Left Speaker", NULL, "Left OUT"}, + {"Right Speaker", NULL, "Right OUT"}, +}; + +static int avs_ssm4567_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + + /* Slot 1 for left */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(runtime, 0), 0x01, 0x01, 2, 48); + if (ret < 0) + return ret; + + /* Slot 2 for right */ + ret = snd_soc_dai_set_tdm_slot(asoc_rtd_to_codec(runtime, 1), 0x02, 0x02, 2, 48); + if (ret < 0) + return ret; + + return 0; +} + +static int +avs_ssm4567_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs) * 2, GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs[0].name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343B:00"); + dl->codecs[0].dai_name = devm_kasprintf(dev, GFP_KERNEL, "ssm4567-hifi"); + dl->codecs[1].name = devm_kasprintf(dev, GFP_KERNEL, "i2c-INT343B:01"); + dl->codecs[1].dai_name = devm_kasprintf(dev, GFP_KERNEL, "ssm4567-hifi"); + if (!dl->cpus->dai_name || !dl->codecs[0].name || !dl->codecs[0].dai_name || + !dl->codecs[1].name || !dl->codecs[1].dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 2; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_ssm4567_codec_init; + dl->be_hw_params_fixup = avs_ssm4567_be_fixup; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + dl->ignore_pmdown_time = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 4; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Left Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Right Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Left Capture Sense"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Right Capture Sense"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_ssm4567_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_ssm4567-adi"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->codec_conf = card_codec_conf; + card->num_configs = ARRAY_SIZE(card_codec_conf); + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + card->disable_route_checks = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_ssm4567_driver = { + .probe = avs_ssm4567_probe, + .driver = { + .name = "avs_ssm4567", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_ssm4567_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_ssm4567");
To support AVS-max98357a configuration add machine board connecting AVS platform component driver with max98357a codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 10 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/max98357a.c | 154 +++++++++++++++++++++++++ 3 files changed, 166 insertions(+) create mode 100644 sound/soc/intel/avs/boards/max98357a.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 7020e7bf196e..28e6691270d9 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -26,6 +26,16 @@ config SND_SOC_INTEL_AVS_MACH_I2S_TEST This adds support for I2S test-board which can be used to verify transfer over I2S interface with SSP loopback scenarios.
+config SND_SOC_INTEL_AVS_MACH_MAX98357A + tristate "max98357A I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_MAX98357A + help + This adds support for AVS with MAX98357A I2S codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_NAU8825 tristate "nau8825 I2S board" depends on I2C diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index ea67fc711d9d..f7ac1151a8f7 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -3,6 +3,7 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o +snd-soc-avs-max98357a-objs := max98357a.o snd-soc-avs-nau8825-objs := nau8825.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o @@ -13,6 +14,7 @@ snd-soc-avs-ssm4567-objs := ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98357A) += snd-soc-avs-max98357a.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_NAU8825) += snd-soc-avs-nau8825.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c new file mode 100644 index 000000000000..921f42caf7e0 --- /dev/null +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -0,0 +1,154 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + { "Spk", NULL, "Speaker" }, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "MX98357A:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, "HiFi"); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 1; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "HiFi Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_max98357a_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_max98357a"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_max98357a_driver = { + .probe = avs_max98357a_probe, + .driver = { + .name = "avs_max98357a", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_max98357a_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_max98357a");
From: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com
To support AVS-max98373 configuration add machine board connecting AVS platform component driver with max98373 codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 10 ++ sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/max98373.c | 239 ++++++++++++++++++++++++++ 3 files changed, 251 insertions(+) create mode 100644 sound/soc/intel/avs/boards/max98373.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index 28e6691270d9..d3be6dc1fc10 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -36,6 +36,16 @@ config SND_SOC_INTEL_AVS_MACH_MAX98357A Say Y or m if you have such a device. This is a recommended option. If unsure select "N".
+config SND_SOC_INTEL_AVS_MACH_MAX98373 + tristate "max98373 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_MAX98373 + help + This adds support for AVS with MAX98373 I2S codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_NAU8825 tristate "nau8825 I2S board" depends on I2C diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index f7ac1151a8f7..0bce31e192ce 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -4,6 +4,7 @@ snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o snd-soc-avs-max98357a-objs := max98357a.o +snd-soc-avs-max98373-objs := max98373.o snd-soc-avs-nau8825-objs := nau8825.o snd-soc-avs-rt274-objs := rt274.o snd-soc-avs-rt286-objs := rt286.o @@ -15,6 +16,7 @@ obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98357A) += snd-soc-avs-max98357a.o +obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_MAX98373) += snd-soc-avs-max98373.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_NAU8825) += snd-soc-avs-nau8825.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT274) += snd-soc-avs-rt274.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_RT286) += snd-soc-avs-rt286.o diff --git a/sound/soc/intel/avs/boards/max98373.c b/sound/soc/intel/avs/boards/max98373.c new file mode 100644 index 000000000000..0fa8f5606385 --- /dev/null +++ b/sound/soc/intel/avs/boards/max98373.c @@ -0,0 +1,239 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2022 Intel Corporation. All rights reserved. +// +// Authors: Cezary Rojewski cezary.rojewski@intel.com +// Amadeusz Slawinski amadeuszx.slawinski@linux.intel.com +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> + +#define MAX98373_DEV0_NAME "i2c-MX98373:00" +#define MAX98373_DEV1_NAME "i2c-MX98373:01" +#define MAX98373_CODEC_NAME "max98373-aif1" + +static struct snd_soc_codec_conf card_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX98373_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX98373_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +static int +avs_max98373_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int avs_max98373_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *runtime = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int ret, i; + + for_each_rtd_codec_dais(runtime, i, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX98373_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAX98373_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); + if (ret < 0) { + dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); + return ret; + } + } + } + + return 0; +} + +static const struct snd_soc_ops avs_max98373_ops = { + .hw_params = avs_max98373_hw_params, +}; + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs) * 2, GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs[0].name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_DEV0_NAME); + dl->codecs[0].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_CODEC_NAME); + dl->codecs[1].name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_DEV1_NAME); + dl->codecs[1].dai_name = devm_kasprintf(dev, GFP_KERNEL, MAX98373_CODEC_NAME); + if (!dl->cpus->dai_name || !dl->codecs[0].name || !dl->codecs[0].dai_name || + !dl->codecs[1].name || !dl->codecs[1].dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 2; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC; + dl->be_hw_params_fixup = avs_max98373_be_fixup; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + dl->ignore_pmdown_time = 1; + dl->ops = &avs_max98373_ops; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Left HiFi Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Right HiFi Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_max98373_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->name = "avs_max98373"; + card->dev = dev; + card->owner = THIS_MODULE; + card->dai_link = dai_link; + card->num_links = 1; + card->codec_conf = card_codec_conf; + card->num_configs = ARRAY_SIZE(card_codec_conf); + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_max98373_driver = { + .probe = avs_max98373_probe, + .driver = { + .name = "avs_max98373", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_max98373_driver) + +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_max98373");
To support AVS-da7219 configuration add machine board connecting AVS platform component driver with da7219 codec one.
Signed-off-by: Amadeusz Sławiński amadeuszx.slawinski@linux.intel.com Signed-off-by: Cezary Rojewski cezary.rojewski@intel.com --- sound/soc/intel/avs/boards/Kconfig | 10 + sound/soc/intel/avs/boards/Makefile | 2 + sound/soc/intel/avs/boards/da7219.c | 282 ++++++++++++++++++++++++++++ 3 files changed, 294 insertions(+) create mode 100644 sound/soc/intel/avs/boards/da7219.c
diff --git a/sound/soc/intel/avs/boards/Kconfig b/sound/soc/intel/avs/boards/Kconfig index d3be6dc1fc10..4d68e3ef992b 100644 --- a/sound/soc/intel/avs/boards/Kconfig +++ b/sound/soc/intel/avs/boards/Kconfig @@ -4,6 +4,16 @@ menu "Intel AVS Machine drivers"
comment "Available DSP configurations"
+config SND_SOC_INTEL_AVS_MACH_DA7219 + tristate "da7219 I2S board" + depends on I2C + depends on MFD_INTEL_LPSS || COMPILE_TEST + select SND_SOC_DA7219 + help + This adds support for AVS with DA7219 I2S codec configuration. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". + config SND_SOC_INTEL_AVS_MACH_DMIC tristate "DMIC generic board" select SND_SOC_DMIC diff --git a/sound/soc/intel/avs/boards/Makefile b/sound/soc/intel/avs/boards/Makefile index 0bce31e192ce..25e8c4bb07db 100644 --- a/sound/soc/intel/avs/boards/Makefile +++ b/sound/soc/intel/avs/boards/Makefile @@ -1,5 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only
+snd-soc-avs-da7219-objs := da7219.o snd-soc-avs-dmic-objs := dmic.o snd-soc-avs-hdaudio-objs := hdaudio.o snd-soc-avs-i2s-test-objs := i2s_test.o @@ -12,6 +13,7 @@ snd-soc-avs-rt298-objs := rt298.o snd-soc-avs-rt5682-objs := rt5682.o snd-soc-avs-ssm4567-objs := ssm4567.o
+obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DA7219) += snd-soc-avs-da7219.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_DMIC) += snd-soc-avs-dmic.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_HDAUDIO) += snd-soc-avs-hdaudio.o obj-$(CONFIG_SND_SOC_INTEL_AVS_MACH_i2s_TEST) += snd-soc-avs-i2s-test.o diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c new file mode 100644 index 000000000000..02ae542ad779 --- /dev/null +++ b/sound/soc/intel/avs/boards/da7219.c @@ -0,0 +1,282 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// Copyright(c) 2021-2022 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski cezary.rojewski@intel.com +// + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-acpi.h> +#include <sound/soc-dapm.h> +#include <uapi/linux/input-event-codes.h> +#include "../../../codecs/da7219.h" +#include "../../../codecs/da7219-aad.h" + +#define DA7219_DAI_NAME "da7219-hifi" + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found. Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, + 0, DA7219_PLL_FREQ_OUT_98304); + if (ret) + dev_err(card->dev, "failed to start PLL: %d\n", ret); + } + + return ret; +} + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), +}; + +static const struct snd_soc_dapm_route card_base_routes[] = { + /* HP jack connectors - unknown if we have jack detection */ + {"Headphone Jack", NULL, "HPL"}, + {"Headphone Jack", NULL, "HPR"}, + + {"MIC", NULL, "Headset Mic"}, + + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headset Mic", NULL, "Platform Clock" }, +}; + +static int avs_da7219_codec_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_component *component = asoc_rtd_to_codec(runtime, 0)->component; + struct snd_soc_card *card = runtime->card; + struct snd_soc_jack *jack; + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(runtime, 0); + int clk_freq; + int ret; + + jack = snd_soc_card_get_drvdata(card); + clk_freq = 19200000; + + ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, clk_freq, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(card->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | + SND_JACK_BTN_1 | SND_JACK_BTN_2 | + SND_JACK_BTN_3 | SND_JACK_LINEOUT, jack); + if (ret) { + dev_err(card->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + da7219_aad_jack_det(component, jack); + + return 0; +} + +static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, + struct snd_soc_dai_link **dai_link) +{ + struct snd_soc_dai_link_component *platform; + struct snd_soc_dai_link *dl; + + dl = devm_kzalloc(dev, sizeof(*dl), GFP_KERNEL); + platform = devm_kzalloc(dev, sizeof(*platform), GFP_KERNEL); + if (!dl || !platform) + return -ENOMEM; + + platform->name = platform_name; + + dl->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-Codec", ssp_port); + dl->cpus = devm_kzalloc(dev, sizeof(*dl->cpus), GFP_KERNEL); + dl->codecs = devm_kzalloc(dev, sizeof(*dl->codecs), GFP_KERNEL); + if (!dl->name || !dl->cpus || !dl->codecs) + return -ENOMEM; + + dl->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", ssp_port); + dl->codecs->name = devm_kasprintf(dev, GFP_KERNEL, "i2c-DLGS7219:00"); + dl->codecs->dai_name = devm_kasprintf(dev, GFP_KERNEL, DA7219_DAI_NAME); + if (!dl->cpus->dai_name || !dl->codecs->name || !dl->codecs->dai_name) + return -ENOMEM; + + dl->num_cpus = 1; + dl->num_codecs = 1; + dl->platforms = platform; + dl->num_platforms = 1; + dl->id = 0; + dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->init = avs_da7219_codec_init; + dl->nonatomic = 1; + dl->no_pcm = 1; + dl->dpcm_capture = 1; + dl->dpcm_playback = 1; + + *dai_link = dl; + + return 0; +} + +static int avs_create_dapm_routes(struct device *dev, int ssp_port, + struct snd_soc_dapm_route **routes, int *num_routes) +{ + struct snd_soc_dapm_route *dr; + const int num_base = ARRAY_SIZE(card_base_routes); + const int num_dr = num_base + 2; + int idx; + + dr = devm_kcalloc(dev, num_dr, sizeof(*dr), GFP_KERNEL); + if (!dr) + return -ENOMEM; + + memcpy(dr, card_base_routes, num_base * sizeof(*dr)); + + idx = num_base; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "Playback"); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Tx", ssp_port); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + idx++; + dr[idx].sink = devm_kasprintf(dev, GFP_KERNEL, "ssp%d Rx", ssp_port); + dr[idx].source = devm_kasprintf(dev, GFP_KERNEL, "Capture"); + if (!dr[idx].sink || !dr[idx].source) + return -ENOMEM; + + *routes = dr; + *num_routes = num_dr; + + return 0; +} + +static int avs_card_set_jack(struct snd_soc_card *card, struct snd_soc_jack *jack) +{ + struct snd_soc_component *component; + + for_each_card_components(card, component) + snd_soc_component_set_jack(component, jack, NULL); + return 0; +} + +static int avs_card_remove(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_suspend_pre(struct snd_soc_card *card) +{ + return avs_card_set_jack(card, NULL); +} + +static int avs_card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); + + return avs_card_set_jack(card, jack); +} + +static int avs_da7219_probe(struct platform_device *pdev) +{ + struct snd_soc_dapm_route *routes; + struct snd_soc_dai_link *dai_link; + struct snd_soc_acpi_mach *mach; + struct snd_soc_card *card; + struct snd_soc_jack *jack; + struct device *dev = &pdev->dev; + const char *pname; + int num_routes, ssp_port, ret; + + mach = dev_get_platdata(dev); + pname = mach->mach_params.platform; + ssp_port = __ffs(mach->mach_params.i2s_link_mask); + + ret = avs_create_dai_link(dev, pname, ssp_port, &dai_link); + if (ret) { + dev_err(dev, "Failed to create dai link: %d", ret); + return ret; + } + + ret = avs_create_dapm_routes(dev, ssp_port, &routes, &num_routes); + if (ret) { + dev_err(dev, "Failed to create dapm routes: %d", ret); + return ret; + } + + jack = devm_kzalloc(dev, sizeof(*jack), GFP_KERNEL); + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!jack || !card) + return -ENOMEM; + + card->name = "avs_da7219"; + card->dev = dev; + card->owner = THIS_MODULE; + card->remove = avs_card_remove; + card->suspend_pre = avs_card_suspend_pre; + card->resume_post = avs_card_resume_post; + card->dai_link = dai_link; + card->num_links = 1; + card->controls = card_controls; + card->num_controls = ARRAY_SIZE(card_controls); + card->dapm_widgets = card_widgets; + card->num_dapm_widgets = ARRAY_SIZE(card_widgets); + card->dapm_routes = routes; + card->num_dapm_routes = num_routes; + card->fully_routed = true; + snd_soc_card_set_drvdata(card, jack); + + ret = snd_soc_fixup_dai_links_platform_name(card, pname); + if (ret) + return ret; + + return devm_snd_soc_register_card(dev, card); +} + +static struct platform_driver avs_da7219_driver = { + .probe = avs_da7219_probe, + .driver = { + .name = "avs_da7219", + .pm = &snd_soc_pm_ops, + }, +}; + +module_platform_driver(avs_da7219_driver); + +MODULE_AUTHOR("Cezary Rojewski cezary.rojewski@intel.com"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:avs_da7219");
On 2022-05-11 6:23 PM, Cezary Rojewski wrote:
This series focuses on populating boards/ subdirectory with supported configurations by the avs-driver. Note: it is independent of recently provided "Driver code and PCM operations" series [1], that is, code found here should not collide with it.
Hello,
Friendly ping as PCM series got merged but this one looks orphaned :( Patches addressing fls() issues reported by bots have been sent as separate series [1].
[1]: https://lore.kernel.org/lkml/20220525144844.1571705-1-amadeuszx.slawinski@li...
Regards, Czarek
On Thu, May 26, 2022 at 12:14:09PM +0200, Cezary Rojewski wrote:
Friendly ping as PCM series got merged but this one looks orphaned :( Patches addressing fls() issues reported by bots have been sent as separate series [1].
Please don't send content free pings and please allow a reasonable time for review. People get busy, go on holiday, attend conferences and so on so unless there is some reason for urgency (like critical bug fixes) please allow at least a couple of weeks for review. If there have been review comments then people may be waiting for those to be addressed.
Sending content free pings adds to the mail volume (if they are seen at all) which is often the problem and since they can't be reviewed directly if something has gone wrong you'll have to resend the patches anyway, so sending again is generally a better approach though there are some other maintainers who like them - if in doubt look at how patches for the subsystem are normally handled.
On Wed, 11 May 2022 18:23:49 +0200, Cezary Rojewski wrote:
This series focuses on populating boards/ subdirectory with supported configurations by the avs-driver. Note: it is independent of recently provided "Driver code and PCM operations" series [1], that is, code found here should not collide with it.
Series starts with a small change that adds a helper to sound pcm header, allowing for retrieving string naming a direction without the need of substream pointer. Said helper is used by codec driver code that follows it but I believe it's generic and helpful enough that it can be called an independent addition to the sound core.
[...]
Applied to
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next
Thanks!
[01/14] ALSA: Add snd_pcm_direction_name() helper commit: 90b12a88b710cdc80c00552dfbd589228978bffe [02/14] ASoC: codecs: Add HD-Audio codec driver commit: b5df2a7dca1cc6c66eee0005c92094855dc2028c [03/14] ASoC: Intel: avs: Add HDAudio machine board commit: 97030a43371ea29d65f332d288eb73e8f7bdb3a9 [04/14] ASoC: Intel: avs: Add DMIC machine board commit: 6575e5cae7525b07d0b5fbd7d42323363919a867 [05/14] ASoC: Intel: avs: Add I2S-test machine board commit: e39acc4cfd9250e7b8ec01897570f3009659c3d6 [06/14] ASoC: Intel: avs: Add rt274 machine board commit: e2a4cbf277c4561d01f1aafa3cfafe46bf3feec7 [07/14] ASoC: Intel: avs: Add rt286 machine board commit: 1d395ee2e19b33a1008acfc7af186f2851b63d01 [08/14] ASoC: Intel: avs: Add rt298 machine board commit: 88429ab16df4cd4a1a77d45b90ec95cf62cc22d1 [09/14] ASoC: Intel: avs: Add rt5682 machine board commit: 748102786b3ce0bf402c2dc42386cbfaab71ac39 [10/14] ASoC: Intel: avs: Add nau8825 machine board commit: 32ee40b5590081a6b38a55e4ab16b47085f93afe [11/14] ASoC: Intel: avs: Add ssm4567 machine board commit: 69ea14efe99b533652255b07a9736a9856f50ea5 [12/14] ASoC: Intel: avs: Add max98357a machine board commit: 282c8f8de72f95325225d94caef61f3cc96401da [13/14] ASoC: Intel: avs: Add max98373 machine board commit: 223a0a945821b96f4ccd9940ee975499706e1794 [14/14] ASoC: Intel: avs: Add da7219 machine board commit: 6b5b0d6f36dd45e22f1710e8bcd97f28b4ba41f5
All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying to this mail.
Thanks, Mark
participants (3)
-
Cezary Rojewski
-
Geert Uytterhoeven
-
Mark Brown