dpcm_end_walk_at_be() stops the graph walk when first BE is found for the given FE component. In a component model we may want to connect multiple DAIs from different components. A new flag is introduced in 'snd_soc_card', which when set allows DAI/component chaining. Later PCM operations can be called for all these listed components for a valid DAPM path.

Signed-off-by: Sameer Pujar spujar@nvidia.com --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 3 ++- 2 files changed, 3 insertions(+), 1 deletion(-)

diff --git a/include/sound/soc.h b/include/sound/soc.h index 3b038c5..9b69d70 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1084,6 +1084,7 @@ struct snd_soc_card { unsigned int fully_routed:1; unsigned int disable_route_checks:1; unsigned int probed:1; + unsigned int component_chaining:1;

void *drvdata; }; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 09e8d70..25904b8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1283,7 +1283,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe,

/* get number of valid DAI paths and their widgets */ paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, - dpcm_end_walk_at_be); + fe->card->component_chaining ? + NULL : dpcm_end_walk_at_be);

dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback");

PCM devices are created for FE dai links with 'no-pcm' flag as '0'. Such DAI links have CPU component which implement either pcm_construct() or pcm_new() at component or dai level respectively. Based on this, current patch exposes a helper function to identify such components and populate 'no_pcm' flag for DPCM DAI link.

This helps to have BE<->BE component links where PCM devices need not be created for CPU component involved in such links.

Signed-off-by: Sameer Pujar spujar@nvidia.com Cc: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- sound/soc/generic/audio-graph-card.c | 25 +++++++++++++++++++++++++ 1 file changed, 25 insertions(+)

diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 1e20562..9b06841 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -111,6 +111,17 @@ static int graph_get_dai_id(struct device_node *ep) return id; }

+static bool soc_component_is_pcm(struct snd_soc_dai_link_component *dlc) +{ + struct snd_soc_dai *dai = snd_soc_find_dai_with_mutex(dlc); + + if (dai && (dai->component->driver->pcm_construct || + dai->driver->pcm_new)) + return true; + + return false; +} + static int asoc_simple_parse_dai(struct device_node *ep, struct snd_soc_dai_link_component *dlc, int *is_single_link) @@ -205,6 +216,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, int dup_codec) { struct device *dev = simple_priv_to_dev(priv); + struct snd_soc_card *card = simple_priv_to_card(priv); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); struct device_node *top = dev->of_node; @@ -259,6 +271,19 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (ret < 0) goto out_put_node;

+ /* + * In BE<->BE connections it is not required to create + * PCM devices at CPU end of the dai link and thus 'no_pcm' + * flag needs to be set. It is useful when there are many + * BE components and some of these have to be connected to + * form a valid audio path. + * + * For example: FE <-> BE1 <-> BE2 <-> ... <-> BEn where + * there are 'n' BE components in the path. + */ + if (card->component_chaining && !soc_component_is_pcm(cpus)) + dai_link->no_pcm = 1; + /* card->num_links includes Codec */ asoc_simple_canonicalize_cpu(dai_link, is_single_links); } else {

Add new members in struct 'asoc_simple_priv'. Idea is to leverage simple or graph card driver as much as possible and vendor can maintain a thin driver to control the behavior by populating these newly exposed members.

Following are the members added in 'asoc_simple_priv':

- 'ops' struct: In some cases SoC vendor drivers may want to implement 'snd_soc_ops' callbacks differently. In such cases custom callbacks would be used.

- 'force_dpcm' flag: Right now simple or graph card drivers detect DAI links as DPCM links if:

* The dpcm_selectable is set AND * Codec is connected to multiple CPU endpoints or aconvert property is used for rate/channels.

So there is no way to directly specify usage of DPCM alone. So a flag is exposed to mark all links as DPCM. Vendor driver can set this if required.

- 'dpcm_selectable': Currently simple or audio graph drivers provide a way to enable this for specific compatibles. However vendor driver may want to define some additional info. Thus expose this variable where vendor drivers can set this if required.

- 'data': A void pointer member is provided. This would be useful when vendor driver wants to define its own structure for internal usage and still wants to rely on most of the other members from 'asoc_simple_priv'.

Subsequent patches in the series illustrate usage for above.

Signed-off-by: Sameer Pujar spujar@nvidia.com Cc: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/simple_card_utils.h | 4 ++++ 1 file changed, 4 insertions(+)

diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 86a1e95..9825308 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -56,6 +56,10 @@ struct asoc_simple_priv { struct asoc_simple_dai *dais; struct snd_soc_codec_conf *codec_conf; struct gpio_desc *pa_gpio; + const struct snd_soc_ops *ops; + unsigned int force_dpcm:1; + uintptr_t dpcm_selectable; + void *data; }; #define simple_priv_to_card(priv) (&(priv)->snd_card) #define simple_priv_to_props(priv, i) ((priv)->dai_props + (i))

This commit exposes following functions which can be used by a sound card driver based on audio graph driver. Idea is vendors can have a thin driver and re-use common stuff from audio graph driver.

- graph_card_probe() - graph_get_dais_count() - graph_parse_of()

In doing so a new header file is added for audio graph which can be included by vendor drivers.

Signed-off-by: Sameer Pujar spujar@nvidia.com Cc: Kuninori Morimoto kuninori.morimoto.gx@renesas.com --- include/sound/graph_card.h | 19 +++++++++++++++++++ sound/soc/generic/audio-graph-card.c | 11 +++++++---- 2 files changed, 26 insertions(+), 4 deletions(-) create mode 100644 include/sound/graph_card.h

diff --git a/include/sound/graph_card.h b/include/sound/graph_card.h new file mode 100644 index 0000000..0372c1c --- /dev/null +++ b/include/sound/graph_card.h @@ -0,0 +1,19 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ASoC audio graph card support + * + */ + +#ifndef __GRAPH_CARD_H +#define __GRAPH_CARD_H + +#include <sound/simple_card_utils.h> + +int graph_card_probe(struct snd_soc_card *card); + +void graph_get_dais_count(struct asoc_simple_priv *priv, + struct link_info *li); + +int graph_parse_of(struct asoc_simple_priv *priv); + +#endif /* __GRAPH_CARD_H */ diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index fb34e34..ae234bf 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -529,7 +529,7 @@ static int graph_for_each_link(struct asoc_simple_priv *priv, return 0; }

-static int graph_parse_of(struct asoc_simple_priv *priv) +int graph_parse_of(struct asoc_simple_priv *priv) { struct snd_soc_card *card = simple_priv_to_card(priv); struct link_info li; @@ -566,6 +566,7 @@ static int graph_parse_of(struct asoc_simple_priv *priv)

return asoc_simple_parse_card_name(card, NULL); } +EXPORT_SYMBOL_GPL(graph_parse_of);

static int graph_count_noml(struct asoc_simple_priv *priv, struct device_node *cpu_ep, @@ -604,8 +605,8 @@ static int graph_count_dpcm(struct asoc_simple_priv *priv, return 0; }

-static void graph_get_dais_count(struct asoc_simple_priv *priv, - struct link_info *li) +void graph_get_dais_count(struct asoc_simple_priv *priv, + struct link_info *li) { struct device *dev = simple_priv_to_dev(priv);

@@ -661,8 +662,9 @@ static void graph_get_dais_count(struct asoc_simple_priv *priv, dev_dbg(dev, "link %d, dais %d, ccnf %d\n", li->link, li->dais, li->conf); } +EXPORT_SYMBOL_GPL(graph_get_dais_count);

-static int graph_card_probe(struct snd_soc_card *card) +int graph_card_probe(struct snd_soc_card *card) { struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); int ret; @@ -677,6 +679,7 @@ static int graph_card_probe(struct snd_soc_card *card)

return 0; } +EXPORT_SYMBOL_GPL(graph_card_probe);

static int graph_probe(struct platform_device *pdev) {

Add Tegra audio machine driver which is based on generic audio graph card driver. It re-uses most of the common stuff from audio graph driver and uses the same DT binding. Required Tegra specific customizations are done in the driver.

Details on the customizations done:

- Update PLL rates at runtime: Tegra HW supports multiple sample rates (multiples of 8x and 11.025x) and both of these groups require different PLL rates. Hence there is a requirement to update this at runtime. This is achieved by providing a custom 'snd_soc_ops' and in hw_param() callback PLL rate is updated as per the sample rate.

- Internal structure 'tegra_audio_graph_data' is used to maintain clock handles of PLL.

- The 'force_dpcm' flag is set to use DPCM for all DAI links.

- The 'component_chaining' flag is set to use DPCM with component model.

Signed-off-by: Sameer Pujar spujar@nvidia.com --- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_audio_graph_card.c | 291 +++++++++++++++++++++++++++++++ 3 files changed, 302 insertions(+) create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c

diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 3d91bd3..9959605 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -118,6 +118,15 @@ config SND_SOC_TEGRA210_ADMAIF channel. Buffer size is configurable for each ADMAIIF channel. Say Y or M if you want to add support for Tegra210 ADMAIF module.

+config SND_SOC_TEGRA_AUDIO_GRAPH_CARD + tristate "Audio Graph Card based Tegra driver" + depends on SND_AUDIO_GRAPH_CARD + help + Config to enable Tegra audio machine driver based on generic + audio graph driver. It is a thin driver written to customize + few things for Tegra audio. Most of the code is re-used from + audio graph driver and the same DT bindings are used. + config SND_SOC_TEGRA_RT5640 tristate "SoC Audio support for Tegra boards using an RT5640 codec" depends on SND_SOC_TEGRA && I2C && GPIOLIB diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index 60040a0..b17dd6e 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -38,6 +38,7 @@ snd-soc-tegra-trimslice-objs := trimslice.o snd-soc-tegra-alc5632-objs := tegra_alc5632.o snd-soc-tegra-max98090-objs := tegra_max98090.o snd-soc-tegra-sgtl5000-objs := tegra_sgtl5000.o +snd-soc-tegra-audio-graph-card-objs := tegra_audio_graph_card.o

obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o obj-$(CONFIG_SND_SOC_TEGRA_RT5677) += snd-soc-tegra-rt5677.o @@ -48,3 +49,4 @@ obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o obj-$(CONFIG_SND_SOC_TEGRA_MAX98090) += snd-soc-tegra-max98090.o obj-$(CONFIG_SND_SOC_TEGRA_SGTL5000) += snd-soc-tegra-sgtl5000.o +obj-$(CONFIG_SND_SOC_TEGRA_AUDIO_GRAPH_CARD) += snd-soc-tegra-audio-graph-card.o diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c new file mode 100644 index 0000000..cfbd3f0 --- /dev/null +++ b/sound/soc/tegra/tegra_audio_graph_card.c @@ -0,0 +1,291 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// tegra_audio_graph_card.c - Audio Graph based Tegra Machine Driver +// +// Copyright (c) 2020 NVIDIA CORPORATION. All rights reserved. + +#include <linux/math64.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/platform_device.h> +#include <sound/graph_card.h> +#include <sound/pcm_params.h> + +#define MAX_PLLA_OUT0_DIV 128 + +enum srate_type { + /* + * Sample rates multiple of 8000 Hz and below are supported: + * ( 8000, 16000, 32000, 48000, 96000, 192000 Hz ) + */ + x8_RATE, + + /* + * Sample rates multiple of 11025 Hz and below are supported: + * ( 11025, 22050, 44100, 88200, 176400 Hz ) + */ + x11_RATE, + + NUM_RATE_TYPE, +}; + +struct tegra_audio_graph_data { + struct clk *clk_plla_out0; + struct clk *clk_plla; +}; + +struct tegra_audio_chip_data { + unsigned int plla_out0_rates[NUM_RATE_TYPE]; + unsigned int plla_rates[NUM_RATE_TYPE]; +}; + +/* Setup PLL clock as per the given sample rate */ +static int tegra_audio_graph_update_pll(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct device *dev = rtd->card->dev; + struct tegra_audio_graph_data *graph_data = + (struct tegra_audio_graph_data *)priv->data; + struct tegra_audio_chip_data *chip_data = + (struct tegra_audio_chip_data *)of_device_get_match_data(dev); + unsigned int srate = params_rate(params); + unsigned int plla_rate, plla_out0_rate, bclk; + int err; + + /* There is nothing to configure */ + if (!chip_data) + return 0; + + switch (srate) { + case 11025: + case 22050: + case 44100: + case 88200: + case 176400: + plla_out0_rate = chip_data->plla_out0_rates[x11_RATE]; + plla_rate = chip_data->plla_rates[x11_RATE]; + break; + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + case 192000: + plla_out0_rate = chip_data->plla_out0_rates[x8_RATE]; + plla_rate = chip_data->plla_rates[x8_RATE]; + break; + default: + dev_err(rtd->card->dev, "Unsupported sample rate %u\n", + srate); + return -EINVAL; + } + + /* + * Below is the clock relation: + * + * PLLA + * | + * |--> PLLA_OUT0 + * | + * |---> I2S modules + * | + * |---> DMIC modules + * | + * |---> DSPK modules + * + * + * Default PLLA_OUT0 rate might be too high when I/O is running + * at minimum PCM configurations. This may result in incorrect + * clock rates and glitchy audio. The maximum divider is 128 + * and any thing higher than that won't work. Thus reduce PLLA_OUT0 + * to work for lower configurations. + * + * This problem is seen for I2S only, as DMIC and DSPK minimum + * clock requirements are under allowed divider limits. + */ + bclk = srate * params_channels(params) * params_width(params); + if (div_u64(plla_out0_rate, bclk) > MAX_PLLA_OUT0_DIV) + plla_out0_rate >>= 1; + + dev_dbg(rtd->card->dev, + "Update clock rates: PLLA(= %u Hz) and PLLA_OUT0(= %u Hz)\n", + plla_rate, plla_out0_rate); + + /* Set PLLA rate */ + err = clk_set_rate(graph_data->clk_plla, plla_rate); + if (err) { + dev_err(rtd->card->dev, + "Can't set plla rate for %u, err: %d\n", + plla_rate, err); + return err; + } + + /* Set PLLA_OUT0 rate */ + err = clk_set_rate(graph_data->clk_plla_out0, plla_out0_rate); + if (err) { + dev_err(rtd->card->dev, + "Can't set plla_out0 rate %u, err: %d\n", + plla_out0_rate, err); + return err; + } + + return err; +} + +static int tegra_audio_graph_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int err; + + /* + * This gets called for each DAI link (FE or BE) when DPCM is used. + * We may not want to update PLLA rate for each call. So PLLA update + * must be restricted to external I/O links (I2S, DMIC or DSPK) since + * they actually depend on it. I/O modules update their clocks in + * hw_param() of their respective component driver and PLLA rate + * update here helps them to derive appropriate rates. + * + * TODO: When more HW accelerators get added (like sample rate + * converter, volume gain controller etc., which don't really + * depend on PLLA) we need a better way to filter here. + */ + if (cpu_dai->driver->ops && rtd->dai_link->no_pcm) { + err = tegra_audio_graph_update_pll(substream, params); + if (err) + return err; + } + + return asoc_simple_hw_params(substream, params); +} + +static const struct snd_soc_ops tegra_audio_graph_ops = { + .startup = asoc_simple_startup, + .shutdown = asoc_simple_shutdown, + .hw_params = tegra_audio_graph_hw_params, +}; + +static int tegra_audio_graph_card_probe(struct snd_soc_card *card) +{ + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(card); + struct tegra_audio_graph_data *graph_data = + (struct tegra_audio_graph_data *)priv->data; + + graph_data->clk_plla = devm_clk_get(card->dev, "pll_a"); + if (IS_ERR(graph_data->clk_plla)) { + dev_err(card->dev, "Can't retrieve clk pll_a\n"); + return PTR_ERR(graph_data->clk_plla); + } + + graph_data->clk_plla_out0 = devm_clk_get(card->dev, "plla_out0"); + if (IS_ERR(graph_data->clk_plla_out0)) { + dev_err(card->dev, "Can't retrieve clk plla_out0\n"); + return PTR_ERR(graph_data->clk_plla_out0); + } + + return graph_card_probe(card); +} + +static int tegra_audio_graph_probe(struct platform_device *pdev) +{ + struct asoc_simple_priv *priv; + struct device *dev = &pdev->dev; + struct snd_soc_card *card; + struct link_info li; + int err; + + /* Allocate the private data and the DAI link array */ + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->data = (struct tegra_audio_graph_data *) + devm_kzalloc(dev, sizeof(*priv->data), GFP_KERNEL); + if (!priv->data) + return -ENOMEM; + + card = simple_priv_to_card(priv); + + card->owner = THIS_MODULE; + card->dev = dev; + card->component_chaining = true; + card->probe = tegra_audio_graph_card_probe; + + priv->ops = &tegra_audio_graph_ops; + priv->force_dpcm = 1; + + memset(&li, 0, sizeof(li)); + graph_get_dais_count(priv, &li); + if (!li.link || !li.dais) + return -EINVAL; + + err = asoc_simple_init_priv(priv, &li); + if (err < 0) + return err; + + err = graph_parse_of(priv); + if (err < 0) { + if (err != -EPROBE_DEFER) + dev_err(dev, "Parse error %d\n", err); + goto cleanup; + } + + snd_soc_card_set_drvdata(card, priv); + + asoc_simple_debug_info(priv); + + err = devm_snd_soc_register_card(dev, card); + if (err < 0) + goto cleanup; + + return 0; + +cleanup: + asoc_simple_clean_reference(card); + + return err; +} + +static const struct tegra_audio_chip_data tegra210_data = { + /* PLLA */ + .plla_rates[x8_RATE] = 368640000, + .plla_rates[x11_RATE] = 338688000, + /* PLLA_OUT0 */ + .plla_out0_rates[x8_RATE] = 49152000, + .plla_out0_rates[x11_RATE] = 45158400, +}; + +static const struct tegra_audio_chip_data tegra186_data = { + /* PLLA */ + .plla_rates[x8_RATE] = 245760000, + .plla_rates[x11_RATE] = 270950400, + /* PLLA_OUT0 */ + .plla_out0_rates[x8_RATE] = 49152000, + .plla_out0_rates[x11_RATE] = 45158400, +}; + +static const struct of_device_id graph_of_tegra_match[] = { + { .compatible = "nvidia,tegra210-audio-graph-card", + .data = &tegra210_data }, + { .compatible = "nvidia,tegra186-audio-graph-card", + .data = &tegra186_data }, + {}, +}; +MODULE_DEVICE_TABLE(of, graph_of_tegra_match); + +static struct platform_driver tegra_audio_graph_card = { + .driver = { + .name = "tegra-audio-graph-card", + .pm = &snd_soc_pm_ops, + .of_match_table = graph_of_tegra_match, + }, + .probe = tegra_audio_graph_probe, +}; +module_platform_driver(tegra_audio_graph_card); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("ASoC Tegra Audio Graph Sound Card"); +MODULE_AUTHOR("Sameer Pujar spujar@nvidia.com");

Expose a header which describes DT bindings required to use audio-graph based sound card. All Tegra210 based platforms can include this header and add platform specific information. Currently, from SoC point of view, all links are exposed for ADMAIF, AHUB, I2S and DMIC components.

Signed-off-by: Sameer Pujar spujar@nvidia.com --- .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 138 +++++++++++++++++++++ 1 file changed, 138 insertions(+) create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi

diff --git a/arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi b/arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi new file mode 100644 index 0000000..6287d28c --- /dev/null +++ b/arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi @@ -0,0 +1,138 @@ +// SPDX-License-Identifier: GPL-2.0 + +/ { + tegra_sound { + status = "disabled"; + + clocks = <&tegra_car TEGRA210_CLK_PLL_A>, + <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + clock-names = "pll_a", "plla_out0"; + + assigned-clocks = <&tegra_car TEGRA210_CLK_PLL_A>, + <&tegra_car TEGRA210_CLK_PLL_A_OUT0>, + <&tegra_car TEGRA210_CLK_EXTERN1>; + assigned-clock-parents = <0>, <0>, <&tegra_car TEGRA210_CLK_PLL_A_OUT0>; + assigned-clock-rates = <368640000>, <49152000>, <12288000>; + }; +}; + +&tegra_admaif { + admaif1_port: port@0 { + admaif1_ep: endpoint { + remote-endpoint = <&xbar_admaif1_ep>; + }; + }; + admaif2_port: port@1 { + admaif2_ep: endpoint { + remote-endpoint = <&xbar_admaif2_ep>; + }; + }; + admaif3_port: port@2 { + admaif3_ep: endpoint { + remote-endpoint = <&xbar_admaif3_ep>; + }; + }; + admaif4_port: port@3 { + admaif4_ep: endpoint { + remote-endpoint = <&xbar_admaif4_ep>; + }; + }; + admaif5_port: port@4 { + admaif5_ep: endpoint { + remote-endpoint = <&xbar_admaif5_ep>; + }; + }; + admaif6_port: port@5 { + admaif6_ep: endpoint { + remote-endpoint = <&xbar_admaif6_ep>; + }; + }; + admaif7_port: port@6 { + admaif7_ep: endpoint { + remote-endpoint = <&xbar_admaif7_ep>; + }; + }; + admaif8_port: port@7 { + admaif8_ep: endpoint { + remote-endpoint = <&xbar_admaif8_ep>; + }; + }; + admaif9_port: port@8 { + admaif9_ep: endpoint { + remote-endpoint = <&xbar_admaif9_ep>; + }; + }; + admaif10_port: port@9 { + admaif10_ep: endpoint { + remote-endpoint = <&xbar_admaif10_ep>; + }; + }; +}; + +&tegra_ahub { + ports { + #address-cells = <1>; + #size-cells = <0>; + + port@0 { + reg = <0x0>; + xbar_admaif1_ep: endpoint { + remote-endpoint = <&admaif1_ep>; + }; + }; + port@1 { + reg = <0x1>; + xbar_admaif2_ep: endpoint { + remote-endpoint = <&admaif2_ep>; + }; + }; + port@2 { + reg = <0x2>; + xbar_admaif3_ep: endpoint { + remote-endpoint = <&admaif3_ep>; + }; + }; + port@3 { + reg = <0x3>; + xbar_admaif4_ep: endpoint { + remote-endpoint = <&admaif4_ep>; + }; + }; + port@4 { + reg = <0x4>; + xbar_admaif5_ep: endpoint { + remote-endpoint = <&admaif5_ep>; + }; + }; + port@5 { + reg = <0x5>; + xbar_admaif6_ep: endpoint { + remote-endpoint = <&admaif6_ep>; + }; + }; + port@6 { + reg = <0x6>; + xbar_admaif7_ep: endpoint { + remote-endpoint = <&admaif7_ep>; + }; + }; + port@7 { + reg = <0x7>; + xbar_admaif8_ep: endpoint { + remote-endpoint = <&admaif8_ep>; + }; + }; + port@8 { + reg = <0x8>; + xbar_admaif9_ep: endpoint { + remote-endpoint = <&admaif9_ep>; + }; + }; + port@9 { + reg = <0x9>; + xbar_admaif10_ep: endpoint { + remote-endpoint = <&admaif10_ep>; + }; + }; + }; +};