On Asymmetric multiprocessor, there is Cortex-A core and Cortex-M core,
Linux is running on A core, RTOS is running on M core.
The audio hardware device can be controlled by Cortex-M device,
So audio playback/capture can be handled by M core.
Rpmsg is the interface for sending and receiving msg to and from M
core, that we can create a virtual sound on Cortex-A core side.
A core will tell the Cortex-M core sound format/rate/channel,
where is the data buffer, what is the period size, when to start,
when to stop and when suspend or resume happen, each of this behavior
there is defined rpmsg command.
Especially we designed the low power audio case, that is to
allocate a large buffer and fill the data, then Cortex-A core can go
to sleep mode, Cortex-M core continue to play the sound, when the
buffer is consumed, Cortex-M core will trigger the Cortex-A core to
wakeup to fill data.
changes in v4:
- remove the sound card node, merge the property to cpu dai node
according to Rob's comments.
- sound card device will be registered by cpu dai driver.
- Fix do_div issue reported by kernel test robot
changes in v3:
- add local refcount for clk enablement in hw_params()
- update the document according Rob's comments
changes in v2:
- update codes and comments according to Mark's comments
Shengjiu Wang (6):
ASoC: soc-component: Add snd_soc_pcm_component_ack
ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg
ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg cpu dai driver
ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel
ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg
ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg
.../devicetree/bindings/sound/fsl,rpmsg.yaml | 118 +++
include/sound/soc-component.h | 3 +
sound/soc/fsl/Kconfig | 28 +
sound/soc/fsl/Makefile | 6 +
sound/soc/fsl/fsl_rpmsg.c | 283 ++++++
sound/soc/fsl/fsl_rpmsg.h | 42 +
sound/soc/fsl/imx-audio-rpmsg.c | 151 +++
sound/soc/fsl/imx-pcm-rpmsg.c | 919 ++++++++++++++++++
sound/soc/fsl/imx-pcm-rpmsg.h | 512 ++++++++++
sound/soc/fsl/imx-rpmsg.c | 127 +++
sound/soc/soc-component.c | 14 +
sound/soc/soc-pcm.c | 2 +
12 files changed, 2205 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
create mode 100644 sound/soc/fsl/fsl_rpmsg.c
create mode 100644 sound/soc/fsl/fsl_rpmsg.h
create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c
create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c
create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h
create mode 100644 sound/soc/fsl/imx-rpmsg.c
--
2.27.0
Add snd_soc_pcm_component_ack back, which can be used to get an
updated buffer pointer in the platform driver.
On Asymmetric multiprocessor, this pointer can be sent to Cortex-M
core for audio processing.
Signed-off-by: Shengjiu Wang <[email protected]>
---
include/sound/soc-component.h | 3 +++
sound/soc/soc-component.c | 14 ++++++++++++++
sound/soc/soc-pcm.c | 2 ++
3 files changed, 19 insertions(+)
diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 7dc75b39287f..722cfab28d29 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -146,6 +146,8 @@ struct snd_soc_component_driver {
int (*mmap)(struct snd_soc_component *component,
struct snd_pcm_substream *substream,
struct vm_area_struct *vma);
+ int (*ack)(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream);
const struct snd_compress_ops *compress_ops;
@@ -498,5 +500,6 @@ int snd_soc_pcm_component_pm_runtime_get(struct snd_soc_pcm_runtime *rtd,
void *stream);
void snd_soc_pcm_component_pm_runtime_put(struct snd_soc_pcm_runtime *rtd,
void *stream, int rollback);
+int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream);
#endif /* __SOC_COMPONENT_H */
diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c
index 8415e9bd2932..3a5e84e16a87 100644
--- a/sound/soc/soc-component.c
+++ b/sound/soc/soc-component.c
@@ -1212,3 +1212,17 @@ void snd_soc_pcm_component_pm_runtime_put(struct snd_soc_pcm_runtime *rtd,
soc_component_mark_pop(component, stream, pm);
}
}
+
+int snd_soc_pcm_component_ack(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct snd_soc_component *component;
+ int i;
+
+ /* FIXME: use 1st pointer */
+ for_each_rtd_components(rtd, i, component)
+ if (component->driver->ack)
+ return component->driver->ack(component, substream);
+
+ return 0;
+}
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index ba8ffbf8a5d3..e75b404a9f36 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2826,6 +2826,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
rtd->ops.page = snd_soc_pcm_component_page;
if (drv->mmap)
rtd->ops.mmap = snd_soc_pcm_component_mmap;
+ if (drv->ack)
+ rtd->ops.ack = snd_soc_pcm_component_ack;
}
if (playback)
--
2.27.0
This is a cpu dai driver for rpmsg audio use case,
which is mainly used for getting the user's configuration
from devicetree and configure the clocks which is used by
Cortex-M core.
Signed-off-by: Shengjiu Wang <[email protected]>
---
sound/soc/fsl/Kconfig | 7 +
sound/soc/fsl/Makefile | 2 +
sound/soc/fsl/fsl_rpmsg.c | 283 ++++++++++++++++++++++++++++++++++++++
sound/soc/fsl/fsl_rpmsg.h | 42 ++++++
4 files changed, 334 insertions(+)
create mode 100644 sound/soc/fsl/fsl_rpmsg.c
create mode 100644 sound/soc/fsl/fsl_rpmsg.h
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index d7f30036d434..f833856f6de0 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -115,6 +115,13 @@ config SND_SOC_FSL_AUD2HTX
config SND_SOC_FSL_UTILS
tristate
+config SND_SOC_FSL_RPMSG
+ tristate "NXP Audio Base On RPMSG support"
+ help
+ Say Y if you want to add rpmsg audio support for the Freescale CPUs.
+ This option is only useful for out-of-tree drivers since
+ in-tree drivers select it automatically.
+
config SND_SOC_IMX_PCM_DMA
tristate
select SND_SOC_GENERIC_DMAENGINE_PCM
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 8c5fa8a859c0..b63802f345cc 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -27,6 +27,7 @@ snd-soc-fsl-mqs-objs := fsl_mqs.o
snd-soc-fsl-easrc-objs := fsl_easrc.o
snd-soc-fsl-xcvr-objs := fsl_xcvr.o
snd-soc-fsl-aud2htx-objs := fsl_aud2htx.o
+snd-soc-fsl-rpmsg-objs := fsl_rpmsg.o
obj-$(CONFIG_SND_SOC_FSL_AUDMIX) += snd-soc-fsl-audmix.o
obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o
@@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_FSL_EASRC) += snd-soc-fsl-easrc.o
obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o
obj-$(CONFIG_SND_SOC_FSL_XCVR) += snd-soc-fsl-xcvr.o
obj-$(CONFIG_SND_SOC_FSL_AUD2HTX) += snd-soc-fsl-aud2htx.o
+obj-$(CONFIG_SND_SOC_FSL_RPMSG) += snd-soc-fsl-rpmsg.o
# MPC5200 Platform Support
obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c
new file mode 100644
index 000000000000..35b3ee376338
--- /dev/null
+++ b/sound/soc/fsl/fsl_rpmsg.c
@@ -0,0 +1,283 @@
+// SPDX-License-Identifier: GPL-2.0+
+// Copyright 2018-2021 NXP
+
+#include <linux/clk.h>
+#include <linux/clk-provider.h>
+#include <linux/delay.h>
+#include <linux/dmaengine.h>
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/of_address.h>
+#include <linux/pm_runtime.h>
+#include <linux/rpmsg.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+
+#include "fsl_rpmsg.h"
+#include "imx-pcm.h"
+
+#define FSL_RPMSG_RATES (SNDRV_PCM_RATE_8000 | \
+ SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_48000)
+#define FSL_RPMSG_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+
+static const unsigned int fsl_rpmsg_rates[] = {
+ 8000, 11025, 16000, 22050, 44100,
+ 32000, 48000, 96000, 88200, 176400, 192000,
+ 352800, 384000, 705600, 768000, 1411200, 2822400,
+};
+
+static const struct snd_pcm_hw_constraint_list fsl_rpmsg_rate_constraints = {
+ .count = ARRAY_SIZE(fsl_rpmsg_rates),
+ .list = fsl_rpmsg_rates,
+};
+
+static int fsl_rpmsg_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct fsl_rpmsg *rpmsg = snd_soc_dai_get_drvdata(dai);
+ struct clk *p = rpmsg->mclk, *pll = 0, *npll = 0;
+ u64 rate = params_rate(params);
+ int ret = 0;
+
+ /* Get current pll parent */
+ while (p && rpmsg->pll8k && rpmsg->pll11k) {
+ struct clk *pp = clk_get_parent(p);
+
+ if (clk_is_match(pp, rpmsg->pll8k) ||
+ clk_is_match(pp, rpmsg->pll11k)) {
+ pll = pp;
+ break;
+ }
+ p = pp;
+ }
+
+ /* Switch to another pll parent if needed. */
+ if (pll) {
+ npll = (do_div(rate, 8000) ? rpmsg->pll11k : rpmsg->pll8k);
+ if (!clk_is_match(pll, npll)) {
+ ret = clk_set_parent(p, npll);
+ if (ret < 0)
+ dev_warn(dai->dev, "failed to set parent %s: %d\n",
+ __clk_get_name(npll), ret);
+ }
+ }
+
+ if (!(rpmsg->mclk_streams & BIT(substream->stream))) {
+ ret = clk_prepare_enable(rpmsg->mclk);
+ if (ret) {
+ dev_err(dai->dev, "failed to enable mclk: %d\n", ret);
+ return ret;
+ }
+
+ rpmsg->mclk_streams |= BIT(substream->stream);
+ }
+
+ return ret;
+}
+
+static int fsl_rpmsg_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct fsl_rpmsg *rpmsg = snd_soc_dai_get_drvdata(dai);
+
+ if (rpmsg->mclk_streams & BIT(substream->stream)) {
+ clk_disable_unprepare(rpmsg->mclk);
+ rpmsg->mclk_streams &= ~BIT(substream->stream);
+ }
+
+ return 0;
+}
+
+static int fsl_rpmsg_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ int ret;
+
+ ret = snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &fsl_rpmsg_rate_constraints);
+
+ return ret;
+}
+
+static const struct snd_soc_dai_ops fsl_rpmsg_dai_ops = {
+ .startup = fsl_rpmsg_startup,
+ .hw_params = fsl_rpmsg_hw_params,
+ .hw_free = fsl_rpmsg_hw_free,
+};
+
+static struct snd_soc_dai_driver fsl_rpmsg_dai = {
+ .playback = {
+ .stream_name = "CPU-Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_KNOT,
+ .formats = FSL_RPMSG_FORMATS,
+ },
+ .capture = {
+ .stream_name = "CPU-Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_KNOT,
+ .formats = FSL_RPMSG_FORMATS,
+ },
+ .symmetric_rate = 1,
+ .symmetric_channels = 1,
+ .symmetric_sample_bits = 1,
+ .ops = &fsl_rpmsg_dai_ops,
+};
+
+static const struct snd_soc_component_driver fsl_component = {
+ .name = "fsl-rpmsg",
+};
+
+static const struct of_device_id fsl_rpmsg_ids[] = {
+ { .compatible = "fsl,imx7ulp-rpmsg"},
+ { .compatible = "fsl,imx8mm-rpmsg"},
+ { .compatible = "fsl,imx8mn-rpmsg"},
+ { .compatible = "fsl,imx8mp-rpmsg"},
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids);
+
+static int fsl_rpmsg_probe(struct platform_device *pdev)
+{
+ struct device_node *np = pdev->dev.of_node;
+ struct fsl_rpmsg *rpmsg;
+ int ret;
+
+ rpmsg = devm_kzalloc(&pdev->dev, sizeof(struct fsl_rpmsg), GFP_KERNEL);
+ if (!rpmsg)
+ return -ENOMEM;
+
+ ret = of_property_read_u32(np, "fsl,audioindex", &rpmsg->audioindex);
+ if (ret)
+ rpmsg->audioindex = 0;
+
+ if (of_property_read_u32(np, "fsl,buffer-size", &rpmsg->buffer_size))
+ rpmsg->buffer_size = IMX_DEFAULT_DMABUF_SIZE;
+
+ if (of_property_read_bool(pdev->dev.of_node, "fsl,enable-lpa"))
+ rpmsg->enable_lpa = 1;
+
+ ret = of_property_read_u32(np, "fsl,version", &rpmsg->version);
+ if (ret)
+ rpmsg->version = API_VERSION_V2;
+
+ /*Get the optional clocks */
+ rpmsg->ipg = devm_clk_get(&pdev->dev, "ipg");
+ if (IS_ERR(rpmsg->ipg))
+ rpmsg->ipg = NULL;
+
+ rpmsg->mclk = devm_clk_get(&pdev->dev, "mclk");
+ if (IS_ERR(rpmsg->mclk))
+ rpmsg->mclk = NULL;
+
+ rpmsg->dma = devm_clk_get(&pdev->dev, "dma");
+ if (IS_ERR(rpmsg->dma))
+ rpmsg->dma = NULL;
+
+ rpmsg->pll8k = devm_clk_get(&pdev->dev, "pll8k");
+ if (IS_ERR(rpmsg->pll8k))
+ rpmsg->pll8k = NULL;
+
+ rpmsg->pll11k = devm_clk_get(&pdev->dev, "pll11k");
+ if (IS_ERR(rpmsg->pll11k))
+ rpmsg->pll11k = NULL;
+
+ platform_set_drvdata(pdev, rpmsg);
+ pm_runtime_enable(&pdev->dev);
+
+ ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
+ &fsl_rpmsg_dai, 1);
+ if (ret)
+ return ret;
+
+ rpmsg->card_pdev = platform_device_register_data(&pdev->dev,
+ "imx-audio-rpmsg",
+ PLATFORM_DEVID_NONE,
+ NULL,
+ 0);
+ if (IS_ERR(rpmsg->card_pdev)) {
+ dev_err(&pdev->dev, "failed to register rpmsg card\n");
+ ret = PTR_ERR(rpmsg->card_pdev);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int fsl_rpmsg_remove(struct platform_device *pdev)
+{
+ struct fsl_rpmsg *rpmsg = platform_get_drvdata(pdev);
+
+ if (rpmsg->card_pdev)
+ platform_device_unregister(rpmsg->card_pdev);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int fsl_rpmsg_runtime_resume(struct device *dev)
+{
+ struct fsl_rpmsg *rpmsg = dev_get_drvdata(dev);
+ int ret;
+
+ ret = clk_prepare_enable(rpmsg->ipg);
+ if (ret) {
+ dev_err(dev, "failed to enable ipg clock: %d\n", ret);
+ goto ipg_err;
+ }
+
+ ret = clk_prepare_enable(rpmsg->dma);
+ if (ret) {
+ dev_err(dev, "Failed to enable dma clock %d\n", ret);
+ goto dma_err;
+ }
+
+ return 0;
+
+dma_err:
+ clk_disable_unprepare(rpmsg->ipg);
+ipg_err:
+ return ret;
+}
+
+static int fsl_rpmsg_runtime_suspend(struct device *dev)
+{
+ struct fsl_rpmsg *rpmsg = dev_get_drvdata(dev);
+
+ clk_disable_unprepare(rpmsg->dma);
+ clk_disable_unprepare(rpmsg->ipg);
+
+ return 0;
+}
+#endif
+
+static const struct dev_pm_ops fsl_rpmsg_pm_ops = {
+ SET_RUNTIME_PM_OPS(fsl_rpmsg_runtime_suspend,
+ fsl_rpmsg_runtime_resume,
+ NULL)
+ SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend,
+ pm_runtime_force_resume)
+};
+
+static struct platform_driver fsl_rpmsg_driver = {
+ .probe = fsl_rpmsg_probe,
+ .remove = fsl_rpmsg_remove,
+ .driver = {
+ .name = "fsl_rpmsg",
+ .pm = &fsl_rpmsg_pm_ops,
+ .of_match_table = fsl_rpmsg_ids,
+ },
+};
+module_platform_driver(fsl_rpmsg_driver);
+
+MODULE_DESCRIPTION("Freescale SoC Audio PRMSG CPU Interface");
+MODULE_AUTHOR("Shengjiu Wang <[email protected]>");
+MODULE_ALIAS("platform:fsl_rpmsg");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_rpmsg.h b/sound/soc/fsl/fsl_rpmsg.h
new file mode 100644
index 000000000000..c6dd3dde3293
--- /dev/null
+++ b/sound/soc/fsl/fsl_rpmsg.h
@@ -0,0 +1,42 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * Copyright 2017-2021 NXP
+ */
+
+#ifndef __FSL_RPMSG_H
+#define __FSL_RPMSG_H
+
+#define API_VERSION_V1 1
+#define API_VERSION_V2 2
+
+/*
+ * struct fsl_rpmsg - rpmsg private data
+ *
+ * @ipg: ipg clock for cpu dai (SAI)
+ * @mclk: master clock for cpu dai (SAI)
+ * @dma: clock for dma device
+ * @pll8k: parent clock for multiple of 8kHz frequency
+ * @pll11k: parent clock for multiple of 11kHz frequency
+ * @card_pdev: Platform_device pointer to register a sound card
+ * @mclk_streams: Active streams that are using baudclk
+ * @force_lpa: force enable low power audio routine if condition satisfy
+ * @enable_lpa: enable low power audio routine according to dts setting
+ * @buffer_size: pre allocated dma buffer size
+ * @audioindex: audio instance index
+ * @version: rpmsg image version
+ */
+struct fsl_rpmsg {
+ struct clk *ipg;
+ struct clk *mclk;
+ struct clk *dma;
+ struct clk *pll8k;
+ struct clk *pll11k;
+ struct platform_device *card_pdev;
+ unsigned int mclk_streams;
+ int force_lpa;
+ int enable_lpa;
+ int buffer_size;
+ int audioindex;
+ int version;
+};
+#endif /* __FSL_RPMSG_H */
--
2.27.0
This driver is used to accept the message from rpmsg audio
channel, and if this driver is probed, it will help to register
the platform driver, the platform driver will use this
audio channel to send and receive messages to and from Cortex-M
core.
Signed-off-by: Shengjiu Wang <[email protected]>
---
sound/soc/fsl/Kconfig | 4 +
sound/soc/fsl/Makefile | 1 +
sound/soc/fsl/imx-audio-rpmsg.c | 151 ++++++++++++++++++++++++++++++++
3 files changed, 156 insertions(+)
create mode 100644 sound/soc/fsl/imx-audio-rpmsg.c
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index f833856f6de0..cc81f6b917be 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -126,6 +126,10 @@ config SND_SOC_IMX_PCM_DMA
tristate
select SND_SOC_GENERIC_DMAENGINE_PCM
+config SND_SOC_IMX_AUDIO_RPMSG
+ tristate
+ depends on RPMSG
+
config SND_SOC_IMX_AUDMUX
tristate "Digital Audio Mux module support"
help
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index b63802f345cc..f08f3cd07ff5 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -60,6 +60,7 @@ obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o
obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o
+obj-$(CONFIG_SND_SOC_IMX_AUDIO_RPMSG) += imx-audio-rpmsg.o
# i.MX Machine Support
snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
diff --git a/sound/soc/fsl/imx-audio-rpmsg.c b/sound/soc/fsl/imx-audio-rpmsg.c
new file mode 100644
index 000000000000..145edb1492b4
--- /dev/null
+++ b/sound/soc/fsl/imx-audio-rpmsg.c
@@ -0,0 +1,151 @@
+// SPDX-License-Identifier: GPL-2.0+
+// Copyright 2017-2020 NXP
+
+#include <linux/module.h>
+#include <linux/rpmsg.h>
+#include "imx-pcm-rpmsg.h"
+
+/*
+ * struct imx_audio_rpmsg: private data
+ *
+ * @rpmsg_pdev: pointer of platform device
+ */
+struct imx_audio_rpmsg {
+ struct platform_device *rpmsg_pdev;
+};
+
+static int imx_audio_rpmsg_cb(struct rpmsg_device *rpdev, void *data, int len,
+ void *priv, u32 src)
+{
+ struct imx_audio_rpmsg *rpmsg = dev_get_drvdata(&rpdev->dev);
+ struct rpmsg_info *info = platform_get_drvdata(rpmsg->rpmsg_pdev);
+ struct rpmsg_r_msg *r_msg = (struct rpmsg_r_msg *)data;
+ struct rpmsg_msg *msg;
+ unsigned long flags;
+
+ dev_dbg(&rpdev->dev, "get from%d: cmd:%d. %d\n",
+ src, r_msg->header.cmd, r_msg->param.resp);
+
+ switch (r_msg->header.type) {
+ case MSG_TYPE_C:
+ /* TYPE C is notification from M core */
+ switch (r_msg->header.cmd) {
+ case TX_PERIOD_DONE:
+ spin_lock_irqsave(&info->lock[TX], flags);
+ msg = &info->msg[TX_PERIOD_DONE + MSG_TYPE_A_NUM];
+
+ /*
+ * Low power mode: get the buffer pointer from
+ * receive msg.
+ */
+ if (r_msg->header.major == 1 &&
+ r_msg->header.minor == 2)
+ msg->r_msg.param.buffer_tail =
+ r_msg->param.buffer_tail;
+ else
+ msg->r_msg.param.buffer_tail++;
+
+ msg->r_msg.param.buffer_tail %= info->num_period[TX];
+ spin_unlock_irqrestore(&info->lock[TX], flags);
+ info->callback[TX](info->callback_param[TX]);
+ break;
+ case RX_PERIOD_DONE:
+ spin_lock_irqsave(&info->lock[RX], flags);
+ msg = &info->msg[RX_PERIOD_DONE + MSG_TYPE_A_NUM];
+
+ if (r_msg->header.major == 1 &&
+ r_msg->header.minor == 2)
+ msg->r_msg.param.buffer_tail =
+ r_msg->param.buffer_tail;
+ else
+ msg->r_msg.param.buffer_tail++;
+
+ msg->r_msg.param.buffer_tail %= info->num_period[1];
+ spin_unlock_irqrestore(&info->lock[RX], flags);
+ info->callback[RX](info->callback_param[RX]);
+ break;
+ default:
+ dev_warn(&rpdev->dev, "unknown msg command\n");
+ break;
+ }
+ break;
+ case MSG_TYPE_B:
+ /* TYPE B is response msg */
+ memcpy(&info->r_msg, r_msg, sizeof(struct rpmsg_r_msg));
+ complete(&info->cmd_complete);
+ break;
+ default:
+ dev_warn(&rpdev->dev, "unknown msg type\n");
+ break;
+ }
+
+ return 0;
+}
+
+static int imx_audio_rpmsg_probe(struct rpmsg_device *rpdev)
+{
+ struct imx_audio_rpmsg *data;
+ int ret = 0;
+
+ dev_info(&rpdev->dev, "new channel: 0x%x -> 0x%x!\n",
+ rpdev->src, rpdev->dst);
+
+ data = devm_kzalloc(&rpdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data)
+ return -ENOMEM;
+
+ dev_set_drvdata(&rpdev->dev, data);
+
+ /* Register platform driver for rpmsg routine */
+ data->rpmsg_pdev = platform_device_register_data(&rpdev->dev,
+ IMX_PCM_DRV_NAME,
+ PLATFORM_DEVID_NONE,
+ NULL, 0);
+ if (IS_ERR(data->rpmsg_pdev)) {
+ dev_err(&rpdev->dev, "failed to register rpmsg platform.\n");
+ ret = PTR_ERR(data->rpmsg_pdev);
+ }
+
+ return ret;
+}
+
+static void imx_audio_rpmsg_remove(struct rpmsg_device *rpdev)
+{
+ struct imx_audio_rpmsg *data = dev_get_drvdata(&rpdev->dev);
+
+ if (data->rpmsg_pdev)
+ platform_device_unregister(data->rpmsg_pdev);
+
+ dev_info(&rpdev->dev, "audio rpmsg driver is removed\n");
+}
+
+static struct rpmsg_device_id imx_audio_rpmsg_id_table[] = {
+ { .name = "rpmsg-audio-channel" },
+ { },
+};
+
+static struct rpmsg_driver imx_audio_rpmsg_driver = {
+ .drv.name = "imx_audio_rpmsg",
+ .drv.owner = THIS_MODULE,
+ .id_table = imx_audio_rpmsg_id_table,
+ .probe = imx_audio_rpmsg_probe,
+ .callback = imx_audio_rpmsg_cb,
+ .remove = imx_audio_rpmsg_remove,
+};
+
+static int __init imx_audio_rpmsg_init(void)
+{
+ return register_rpmsg_driver(&imx_audio_rpmsg_driver);
+}
+
+static void __exit imx_audio_rpmsg_exit(void)
+{
+ unregister_rpmsg_driver(&imx_audio_rpmsg_driver);
+}
+module_init(imx_audio_rpmsg_init);
+module_exit(imx_audio_rpmsg_exit);
+
+MODULE_DESCRIPTION("Freescale SoC Audio RPMSG interface");
+MODULE_AUTHOR("Shengjiu Wang <[email protected]>");
+MODULE_ALIAS("platform:imx_audio_rpmsg");
+MODULE_LICENSE("GPL v2");
--
2.27.0
Platform driver based on rpmsg is the interface for sending and
receiving rpmsg to and from M core. It will tell the Cortex-M core
sound format/rate/channel, where is the data buffer, where is
the period size, when to start, when to stop and when suspend
or resume happen, each this behavior there is defined rpmsg
command.
Especially we designed the low power audio case, that is to
allocate a large buffer and fill the data, then Cortex-A core can go
to sleep mode, Cortex-M core continue to play the sound, when the
buffer is consumed, Cortex-M core will trigger the Cortex-A core to
wake up.
Signed-off-by: Shengjiu Wang <[email protected]>
---
sound/soc/fsl/Kconfig | 5 +
sound/soc/fsl/Makefile | 1 +
sound/soc/fsl/imx-pcm-rpmsg.c | 919 ++++++++++++++++++++++++++++++++++
sound/soc/fsl/imx-pcm-rpmsg.h | 512 +++++++++++++++++++
4 files changed, 1437 insertions(+)
create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.c
create mode 100644 sound/soc/fsl/imx-pcm-rpmsg.h
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index cc81f6b917be..3b94882aee99 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -130,6 +130,11 @@ config SND_SOC_IMX_AUDIO_RPMSG
tristate
depends on RPMSG
+config SND_SOC_IMX_PCM_RPMSG
+ tristate
+ depends on SND_SOC_IMX_AUDIO_RPMSG
+ select SND_SOC_GENERIC_DMAENGINE_PCM
+
config SND_SOC_IMX_AUDMUX
tristate "Digital Audio Mux module support"
help
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index f08f3cd07ff5..ce4f4324c3a2 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -61,6 +61,7 @@ obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o
obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o
obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o
obj-$(CONFIG_SND_SOC_IMX_AUDIO_RPMSG) += imx-audio-rpmsg.o
+obj-$(CONFIG_SND_SOC_IMX_PCM_RPMSG) += imx-pcm-rpmsg.o
# i.MX Machine Support
snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c
new file mode 100644
index 000000000000..f05d5d489560
--- /dev/null
+++ b/sound/soc/fsl/imx-pcm-rpmsg.c
@@ -0,0 +1,919 @@
+// SPDX-License-Identifier: GPL-2.0+
+// Copyright 2017-2021 NXP
+
+#include <linux/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/rpmsg.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/soc.h>
+
+#include "imx-pcm.h"
+#include "fsl_rpmsg.h"
+#include "imx-pcm-rpmsg.h"
+
+static struct snd_pcm_hardware imx_rpmsg_pcm_hardware = {
+ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
+ SNDRV_PCM_INFO_PAUSE |
+ SNDRV_PCM_INFO_RESUME,
+ .buffer_bytes_max = IMX_DEFAULT_DMABUF_SIZE,
+ .period_bytes_min = 512,
+ .period_bytes_max = 65536,
+ .periods_min = 2,
+ .periods_max = 6000,
+ .fifo_size = 0,
+};
+
+static int imx_rpmsg_pcm_send_message(struct rpmsg_msg *msg,
+ struct rpmsg_info *info)
+{
+ struct rpmsg_device *rpdev = info->rpdev;
+ int ret = 0;
+
+ mutex_lock(&info->msg_lock);
+ if (!rpdev) {
+ dev_err(info->dev, "rpmsg channel not ready\n");
+ mutex_unlock(&info->msg_lock);
+ return -EINVAL;
+ }
+
+ dev_dbg(&rpdev->dev, "send cmd %d\n", msg->s_msg.header.cmd);
+
+ if (!(msg->s_msg.header.type == MSG_TYPE_C))
+ reinit_completion(&info->cmd_complete);
+
+ ret = rpmsg_send(rpdev->ept, (void *)&msg->s_msg,
+ sizeof(struct rpmsg_s_msg));
+ if (ret) {
+ dev_err(&rpdev->dev, "rpmsg_send failed: %d\n", ret);
+ mutex_unlock(&info->msg_lock);
+ return ret;
+ }
+
+ /* No receive msg for TYPE_C command */
+ if (msg->s_msg.header.type == MSG_TYPE_C) {
+ mutex_unlock(&info->msg_lock);
+ return 0;
+ }
+
+ /* wait response from rpmsg */
+ ret = wait_for_completion_timeout(&info->cmd_complete,
+ msecs_to_jiffies(RPMSG_TIMEOUT));
+ if (!ret) {
+ dev_err(&rpdev->dev, "rpmsg_send cmd %d timeout!\n",
+ msg->s_msg.header.cmd);
+ mutex_unlock(&info->msg_lock);
+ return -ETIMEDOUT;
+ }
+
+ memcpy(&msg->r_msg, &info->r_msg, sizeof(struct rpmsg_r_msg));
+ memcpy(&info->msg[msg->r_msg.header.cmd].r_msg,
+ &msg->r_msg, sizeof(struct rpmsg_r_msg));
+
+ /*
+ * Reset the buffer pointer to be zero, actully we have
+ * set the buffer pointer to be zero in imx_rpmsg_terminate_all
+ * But if there is timer task queued in queue, after it is
+ * executed the buffer pointer will be changed, so need to
+ * reset it again with TERMINATE command.
+ */
+ switch (msg->s_msg.header.cmd) {
+ case TX_TERMINATE:
+ info->msg[TX_POINTER].r_msg.param.buffer_offset = 0;
+ break;
+ case RX_TERMINATE:
+ info->msg[RX_POINTER].r_msg.param.buffer_offset = 0;
+ break;
+ default:
+ break;
+ }
+
+ dev_dbg(&rpdev->dev, "cmd:%d, resp %d\n", msg->s_msg.header.cmd,
+ info->r_msg.param.resp);
+
+ mutex_unlock(&info->msg_lock);
+
+ return 0;
+}
+
+static int imx_rpmsg_insert_workqueue(struct snd_pcm_substream *substream,
+ struct rpmsg_msg *msg,
+ struct rpmsg_info *info)
+{
+ unsigned long flags;
+ int ret = 0;
+
+ /*
+ * Queue the work to workqueue.
+ * If the queue is full, drop the message.
+ */
+ spin_lock_irqsave(&info->wq_lock, flags);
+ if (info->work_write_index != info->work_read_index) {
+ int index = info->work_write_index;
+
+ memcpy(&info->work_list[index].msg, msg,
+ sizeof(struct rpmsg_s_msg));
+
+ queue_work(info->rpmsg_wq, &info->work_list[index].work);
+ info->work_write_index++;
+ info->work_write_index %= WORK_MAX_NUM;
+ } else {
+ info->msg_drop_count[substream->stream]++;
+ ret = -EPIPE;
+ }
+ spin_unlock_irqrestore(&info->wq_lock, flags);
+
+ return ret;
+}
+
+static int imx_rpmsg_pcm_hw_params(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct rpmsg_msg *msg;
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_HW_PARAM];
+ msg->s_msg.header.cmd = TX_HW_PARAM;
+ } else {
+ msg = &info->msg[RX_HW_PARAM];
+ msg->s_msg.header.cmd = RX_HW_PARAM;
+ }
+
+ msg->s_msg.param.rate = params_rate(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ msg->s_msg.param.format = RPMSG_S16_LE;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ msg->s_msg.param.format = RPMSG_S24_LE;
+ break;
+ case SNDRV_PCM_FORMAT_DSD_U16_LE:
+ msg->s_msg.param.format = SNDRV_PCM_FORMAT_DSD_U16_LE;
+ break;
+ case SNDRV_PCM_FORMAT_DSD_U32_LE:
+ msg->s_msg.param.format = SNDRV_PCM_FORMAT_DSD_U32_LE;
+ break;
+ default:
+ msg->s_msg.param.format = RPMSG_S32_LE;
+ break;
+ }
+
+ switch (params_channels(params)) {
+ case 1:
+ msg->s_msg.param.channels = RPMSG_CH_LEFT;
+ break;
+ case 2:
+ msg->s_msg.param.channels = RPMSG_CH_STEREO;
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ info->send_message(msg, info);
+
+ return ret;
+}
+
+static int imx_rpmsg_pcm_hw_free(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+
+static snd_pcm_uframes_t imx_rpmsg_pcm_pointer(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ struct rpmsg_msg *msg;
+ unsigned int pos = 0;
+ int buffer_tail = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ msg = &info->msg[TX_PERIOD_DONE + MSG_TYPE_A_NUM];
+ else
+ msg = &info->msg[RX_PERIOD_DONE + MSG_TYPE_A_NUM];
+
+ buffer_tail = msg->r_msg.param.buffer_tail;
+ pos = buffer_tail * snd_pcm_lib_period_bytes(substream);
+
+ return bytes_to_frames(substream->runtime, pos);
+}
+
+static void imx_rpmsg_timer_callback(struct timer_list *t)
+{
+ struct stream_timer *stream_timer =
+ from_timer(stream_timer, t, timer);
+ struct snd_pcm_substream *substream = stream_timer->substream;
+ struct rpmsg_info *info = stream_timer->info;
+ struct rpmsg_msg *msg;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_PERIOD_DONE + MSG_TYPE_A_NUM];
+ msg->s_msg.header.cmd = TX_PERIOD_DONE;
+ } else {
+ msg = &info->msg[RX_PERIOD_DONE + MSG_TYPE_A_NUM];
+ msg->s_msg.header.cmd = RX_PERIOD_DONE;
+ }
+
+ imx_rpmsg_insert_workqueue(substream, msg, info);
+}
+
+static int imx_rpmsg_pcm_open(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ struct rpmsg_msg *msg;
+ int ret = 0;
+ int cmd;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_OPEN];
+ msg->s_msg.header.cmd = TX_OPEN;
+
+ /* reinitialize buffer counter*/
+ cmd = TX_PERIOD_DONE + MSG_TYPE_A_NUM;
+ info->msg[cmd].s_msg.param.buffer_tail = 0;
+ info->msg[cmd].r_msg.param.buffer_tail = 0;
+ info->msg[TX_POINTER].r_msg.param.buffer_offset = 0;
+
+ } else {
+ msg = &info->msg[RX_OPEN];
+ msg->s_msg.header.cmd = RX_OPEN;
+
+ /* reinitialize buffer counter*/
+ cmd = RX_PERIOD_DONE + MSG_TYPE_A_NUM;
+ info->msg[cmd].s_msg.param.buffer_tail = 0;
+ info->msg[cmd].r_msg.param.buffer_tail = 0;
+ info->msg[RX_POINTER].r_msg.param.buffer_offset = 0;
+ }
+
+ info->send_message(msg, info);
+
+ imx_rpmsg_pcm_hardware.period_bytes_max =
+ imx_rpmsg_pcm_hardware.buffer_bytes_max / 2;
+
+ snd_soc_set_runtime_hwparams(substream, &imx_rpmsg_pcm_hardware);
+
+ ret = snd_pcm_hw_constraint_integer(substream->runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ return ret;
+
+ info->msg_drop_count[substream->stream] = 0;
+
+ /* Create timer*/
+ info->stream_timer[substream->stream].info = info;
+ info->stream_timer[substream->stream].substream = substream;
+ timer_setup(&info->stream_timer[substream->stream].timer,
+ imx_rpmsg_timer_callback, 0);
+ return ret;
+}
+
+static int imx_rpmsg_pcm_close(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ struct rpmsg_msg *msg;
+ int ret = 0;
+
+ /* Flush work in workqueue to make TX_CLOSE is the last message */
+ flush_workqueue(info->rpmsg_wq);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_CLOSE];
+ msg->s_msg.header.cmd = TX_CLOSE;
+ } else {
+ msg = &info->msg[RX_CLOSE];
+ msg->s_msg.header.cmd = RX_CLOSE;
+ }
+
+ info->send_message(msg, info);
+
+ del_timer(&info->stream_timer[substream->stream].timer);
+
+ rtd->dai_link->ignore_suspend = 0;
+
+ if (info->msg_drop_count[substream->stream])
+ dev_warn(rtd->dev, "Msg is dropped!, number is %d\n",
+ info->msg_drop_count[substream->stream]);
+
+ return ret;
+}
+
+static int imx_rpmsg_pcm_prepare(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev);
+
+ /*
+ * NON-MMAP mode, NONBLOCK, Version 2, enable lpa in dts
+ * four conditions to determine the lpa is enabled.
+ */
+ if ((runtime->access == SNDRV_PCM_ACCESS_RW_INTERLEAVED ||
+ runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED) &&
+ rpmsg->version == API_VERSION_V2 &&
+ rpmsg->enable_lpa) {
+ /*
+ * Ignore suspend operation in low power mode
+ * M core will continue playback music on A core suspend.
+ */
+ rtd->dai_link->ignore_suspend = 1;
+ rpmsg->force_lpa = 1;
+ } else {
+ rpmsg->force_lpa = 0;
+ }
+
+ return 0;
+}
+
+static int imx_rpmsg_pcm_mmap(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ return dma_mmap_wc(substream->pcm->card->dev, vma,
+ runtime->dma_area,
+ runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static void imx_rpmsg_pcm_dma_complete(void *arg)
+{
+ struct snd_pcm_substream *substream = arg;
+
+ snd_pcm_period_elapsed(substream);
+}
+
+static int imx_rpmsg_prepare_and_submit(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ struct rpmsg_msg *msg;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_BUFFER];
+ msg->s_msg.header.cmd = TX_BUFFER;
+ } else {
+ msg = &info->msg[RX_BUFFER];
+ msg->s_msg.header.cmd = RX_BUFFER;
+ }
+
+ /* Send buffer address and buffer size */
+ msg->s_msg.param.buffer_addr = substream->runtime->dma_addr;
+ msg->s_msg.param.buffer_size = snd_pcm_lib_buffer_bytes(substream);
+ msg->s_msg.param.period_size = snd_pcm_lib_period_bytes(substream);
+ msg->s_msg.param.buffer_tail = 0;
+
+ info->num_period[substream->stream] = msg->s_msg.param.buffer_size /
+ msg->s_msg.param.period_size;
+
+ info->callback[substream->stream] = imx_rpmsg_pcm_dma_complete;
+ info->callback_param[substream->stream] = substream;
+
+ return imx_rpmsg_insert_workqueue(substream, msg, info);
+}
+
+static int imx_rpmsg_async_issue_pending(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ struct rpmsg_msg *msg;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_START];
+ msg->s_msg.header.cmd = TX_START;
+ } else {
+ msg = &info->msg[RX_START];
+ msg->s_msg.header.cmd = RX_START;
+ }
+
+ return imx_rpmsg_insert_workqueue(substream, msg, info);
+}
+
+static int imx_rpmsg_restart(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ struct rpmsg_msg *msg;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_RESTART];
+ msg->s_msg.header.cmd = TX_RESTART;
+ } else {
+ msg = &info->msg[RX_RESTART];
+ msg->s_msg.header.cmd = RX_RESTART;
+ }
+
+ return imx_rpmsg_insert_workqueue(substream, msg, info);
+}
+
+static int imx_rpmsg_pause(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ struct rpmsg_msg *msg;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_PAUSE];
+ msg->s_msg.header.cmd = TX_PAUSE;
+ } else {
+ msg = &info->msg[RX_PAUSE];
+ msg->s_msg.header.cmd = RX_PAUSE;
+ }
+
+ return imx_rpmsg_insert_workqueue(substream, msg, info);
+}
+
+static int imx_rpmsg_terminate_all(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ struct rpmsg_msg *msg;
+ int cmd;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_TERMINATE];
+ msg->s_msg.header.cmd = TX_TERMINATE;
+ /* Clear buffer count*/
+ cmd = TX_PERIOD_DONE + MSG_TYPE_A_NUM;
+ info->msg[cmd].s_msg.param.buffer_tail = 0;
+ info->msg[cmd].r_msg.param.buffer_tail = 0;
+ info->msg[TX_POINTER].r_msg.param.buffer_offset = 0;
+ } else {
+ msg = &info->msg[RX_TERMINATE];
+ msg->s_msg.header.cmd = RX_TERMINATE;
+ /* Clear buffer count*/
+ cmd = RX_PERIOD_DONE + MSG_TYPE_A_NUM;
+ info->msg[cmd].s_msg.param.buffer_tail = 0;
+ info->msg[cmd].r_msg.param.buffer_tail = 0;
+ info->msg[RX_POINTER].r_msg.param.buffer_offset = 0;
+ }
+
+ del_timer(&info->stream_timer[substream->stream].timer);
+
+ return imx_rpmsg_insert_workqueue(substream, msg, info);
+}
+
+static int imx_rpmsg_pcm_trigger(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev);
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ ret = imx_rpmsg_prepare_and_submit(component, substream);
+ if (ret)
+ return ret;
+ ret = imx_rpmsg_async_issue_pending(component, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_RESUME:
+ if (rpmsg->force_lpa)
+ break;
+ fallthrough;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = imx_rpmsg_restart(component, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ if (!rpmsg->force_lpa) {
+ if (runtime->info & SNDRV_PCM_INFO_PAUSE)
+ ret = imx_rpmsg_pause(component, substream);
+ else
+ ret = imx_rpmsg_terminate_all(component, substream);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = imx_rpmsg_pause(component, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ ret = imx_rpmsg_terminate_all(component, substream);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (ret)
+ return ret;
+
+ return 0;
+}
+
+/*
+ * imx_rpmsg_pcm_ack
+ *
+ * Send the period index to M core through rpmsg, but not send
+ * all the period index to M core, reduce some unnessesary msg
+ * to reduce the pressure of rpmsg bandwidth.
+ */
+static int imx_rpmsg_pcm_ack(struct snd_soc_component *component,
+ struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev);
+ struct rpmsg_info *info = dev_get_drvdata(component->dev);
+ snd_pcm_uframes_t period_size = runtime->period_size;
+ snd_pcm_sframes_t avail;
+ struct timer_list *timer;
+ struct rpmsg_msg *msg;
+ unsigned long flags;
+ int buffer_tail = 0;
+ int written_num = 0;
+
+ if (!rpmsg->force_lpa)
+ return 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ msg = &info->msg[TX_PERIOD_DONE + MSG_TYPE_A_NUM];
+ msg->s_msg.header.cmd = TX_PERIOD_DONE;
+ } else {
+ msg = &info->msg[RX_PERIOD_DONE + MSG_TYPE_A_NUM];
+ msg->s_msg.header.cmd = RX_PERIOD_DONE;
+ }
+
+ msg->s_msg.header.type = MSG_TYPE_C;
+
+ buffer_tail = (frames_to_bytes(runtime, runtime->control->appl_ptr) %
+ snd_pcm_lib_buffer_bytes(substream));
+ buffer_tail = buffer_tail / snd_pcm_lib_period_bytes(substream);
+
+ /* There is update for period index */
+ if (buffer_tail != msg->s_msg.param.buffer_tail) {
+ written_num = buffer_tail - msg->s_msg.param.buffer_tail;
+ if (written_num < 0)
+ written_num += runtime->periods;
+
+ msg->s_msg.param.buffer_tail = buffer_tail;
+
+ /* The notification message is updated to latest */
+ spin_lock_irqsave(&info->lock[substream->stream], flags);
+ memcpy(&info->notify[substream->stream], msg,
+ sizeof(struct rpmsg_s_msg));
+ info->notify_updated[substream->stream] = true;
+ spin_unlock_irqrestore(&info->lock[substream->stream], flags);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ avail = snd_pcm_playback_hw_avail(runtime);
+ else
+ avail = snd_pcm_capture_hw_avail(runtime);
+
+ timer = &info->stream_timer[substream->stream].timer;
+ /*
+ * If the data in the buffer is less than one period before
+ * this fill, which means the data may not enough on M
+ * core side, we need to send message immediately to let
+ * M core know the pointer is updated.
+ * if there is more than one period data in the buffer before
+ * this fill, which means the data is enough on M core side,
+ * we can delay one period (using timer) to send the message
+ * for reduce the message number in workqueue, because the
+ * pointer may be updated by ack function later, we can
+ * send latest pointer to M core side.
+ */
+ if ((avail - written_num * period_size) <= period_size) {
+ imx_rpmsg_insert_workqueue(substream, msg, info);
+ } else if (rpmsg->force_lpa && !timer_pending(timer)) {
+ int time_msec;
+
+ time_msec = (int)(runtime->period_size * 1000 / runtime->rate);
+ mod_timer(timer, jiffies + msecs_to_jiffies(time_msec));
+ }
+ }
+
+ return 0;
+}
+
+static int imx_rpmsg_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+ int stream, int size)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_wc(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+ if (!buf->area)
+ return -ENOMEM;
+
+ buf->bytes = size;
+ return 0;
+}
+
+static void imx_rpmsg_pcm_free_dma_buffers(struct snd_soc_component *component,
+ struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+ int stream;
+
+ for (stream = SNDRV_PCM_STREAM_PLAYBACK;
+ stream < SNDRV_PCM_STREAM_LAST; stream++) {
+ substream = pcm->streams[stream].substream;
+ if (!substream)
+ continue;
+
+ buf = &substream->dma_buffer;
+ if (!buf->area)
+ continue;
+
+ dma_free_wc(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+}
+
+static int imx_rpmsg_pcm_new(struct snd_soc_component *component,
+ struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+ struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+ struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev);
+ int ret;
+
+ ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+ if (ret)
+ return ret;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = imx_rpmsg_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ rpmsg->buffer_size);
+ if (ret)
+ goto out;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = imx_rpmsg_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ rpmsg->buffer_size);
+ if (ret)
+ goto out;
+ }
+
+ imx_rpmsg_pcm_hardware.buffer_bytes_max = rpmsg->buffer_size;
+out:
+ /* free preallocated buffers in case of error */
+ if (ret)
+ imx_rpmsg_pcm_free_dma_buffers(component, pcm);
+
+ return ret;
+}
+
+static const struct snd_soc_component_driver imx_rpmsg_soc_component = {
+ .name = IMX_PCM_DRV_NAME,
+ .pcm_construct = imx_rpmsg_pcm_new,
+ .pcm_destruct = imx_rpmsg_pcm_free_dma_buffers,
+ .open = imx_rpmsg_pcm_open,
+ .close = imx_rpmsg_pcm_close,
+ .hw_params = imx_rpmsg_pcm_hw_params,
+ .hw_free = imx_rpmsg_pcm_hw_free,
+ .trigger = imx_rpmsg_pcm_trigger,
+ .pointer = imx_rpmsg_pcm_pointer,
+ .mmap = imx_rpmsg_pcm_mmap,
+ .ack = imx_rpmsg_pcm_ack,
+ .prepare = imx_rpmsg_pcm_prepare,
+};
+
+static void imx_rpmsg_pcm_work(struct work_struct *work)
+{
+ struct work_of_rpmsg *work_of_rpmsg;
+ bool is_notification = false;
+ struct rpmsg_info *info;
+ struct rpmsg_msg msg;
+ unsigned long flags;
+
+ work_of_rpmsg = container_of(work, struct work_of_rpmsg, work);
+ info = work_of_rpmsg->info;
+
+ /*
+ * Every work in the work queue, first we check if there
+ * is update for period is filled, because there may be not
+ * enough data in M core side, need to let M core know
+ * data is updated immediately.
+ */
+ spin_lock_irqsave(&info->lock[TX], flags);
+ if (info->notify_updated[TX]) {
+ memcpy(&msg, &info->notify[TX], sizeof(struct rpmsg_s_msg));
+ info->notify_updated[TX] = false;
+ spin_unlock_irqrestore(&info->lock[TX], flags);
+ info->send_message(&msg, info);
+ } else {
+ spin_unlock_irqrestore(&info->lock[TX], flags);
+ }
+
+ spin_lock_irqsave(&info->lock[RX], flags);
+ if (info->notify_updated[RX]) {
+ memcpy(&msg, &info->notify[RX], sizeof(struct rpmsg_s_msg));
+ info->notify_updated[RX] = false;
+ spin_unlock_irqrestore(&info->lock[RX], flags);
+ info->send_message(&msg, info);
+ } else {
+ spin_unlock_irqrestore(&info->lock[RX], flags);
+ }
+
+ /* Skip the notification message for it has been processed above */
+ if (work_of_rpmsg->msg.s_msg.header.type == MSG_TYPE_C &&
+ (work_of_rpmsg->msg.s_msg.header.cmd == TX_PERIOD_DONE ||
+ work_of_rpmsg->msg.s_msg.header.cmd == RX_PERIOD_DONE))
+ is_notification = true;
+
+ if (!is_notification)
+ info->send_message(&work_of_rpmsg->msg, info);
+
+ /* update read index */
+ spin_lock_irqsave(&info->wq_lock, flags);
+ info->work_read_index++;
+ info->work_read_index %= WORK_MAX_NUM;
+ spin_unlock_irqrestore(&info->wq_lock, flags);
+}
+
+static int imx_rpmsg_pcm_probe(struct platform_device *pdev)
+{
+ struct snd_soc_component *component;
+ struct rpmsg_info *info;
+ int ret, i;
+
+ info = devm_kzalloc(&pdev->dev, sizeof(*info), GFP_KERNEL);
+ if (!info)
+ return -ENOMEM;
+
+ platform_set_drvdata(pdev, info);
+
+ info->rpdev = container_of(pdev->dev.parent, struct rpmsg_device, dev);
+ info->dev = &pdev->dev;
+ /* Setup work queue */
+ info->rpmsg_wq = alloc_ordered_workqueue("rpmsg_audio",
+ WQ_HIGHPRI |
+ WQ_UNBOUND |
+ WQ_FREEZABLE);
+ if (!info->rpmsg_wq) {
+ dev_err(&pdev->dev, "workqueue create failed\n");
+ return -ENOMEM;
+ }
+
+ /* Write index initialize 1, make it differ with the read index */
+ info->work_write_index = 1;
+ info->send_message = imx_rpmsg_pcm_send_message;
+
+ for (i = 0; i < WORK_MAX_NUM; i++) {
+ INIT_WORK(&info->work_list[i].work, imx_rpmsg_pcm_work);
+ info->work_list[i].info = info;
+ }
+
+ /* Initialize msg */
+ for (i = 0; i < MSG_MAX_NUM; i++) {
+ info->msg[i].s_msg.header.cate = IMX_RPMSG_AUDIO;
+ info->msg[i].s_msg.header.major = IMX_RMPSG_MAJOR;
+ info->msg[i].s_msg.header.minor = IMX_RMPSG_MINOR;
+ info->msg[i].s_msg.header.type = MSG_TYPE_A;
+ info->msg[i].s_msg.param.audioindex = 0;
+ }
+
+ init_completion(&info->cmd_complete);
+ mutex_init(&info->msg_lock);
+ spin_lock_init(&info->lock[TX]);
+ spin_lock_init(&info->lock[RX]);
+ spin_lock_init(&info->wq_lock);
+
+ ret = devm_snd_soc_register_component(&pdev->dev,
+ &imx_rpmsg_soc_component,
+ NULL, 0);
+ if (ret)
+ goto fail;
+
+ component = snd_soc_lookup_component(&pdev->dev, IMX_PCM_DRV_NAME);
+ if (!component) {
+ ret = -EINVAL;
+ goto fail;
+ }
+#ifdef CONFIG_DEBUG_FS
+ component->debugfs_prefix = "rpmsg";
+#endif
+
+ return 0;
+
+fail:
+ if (info->rpmsg_wq)
+ destroy_workqueue(info->rpmsg_wq);
+
+ return ret;
+}
+
+static int imx_rpmsg_pcm_remove(struct platform_device *pdev)
+{
+ struct rpmsg_info *info = platform_get_drvdata(pdev);
+
+ if (info->rpmsg_wq)
+ destroy_workqueue(info->rpmsg_wq);
+
+ return 0;
+}
+
+#ifdef CONFIG_PM
+static int imx_rpmsg_pcm_runtime_resume(struct device *dev)
+{
+ struct rpmsg_info *info = dev_get_drvdata(dev);
+
+ cpu_latency_qos_add_request(&info->pm_qos_req, 0);
+
+ return 0;
+}
+
+static int imx_rpmsg_pcm_runtime_suspend(struct device *dev)
+{
+ struct rpmsg_info *info = dev_get_drvdata(dev);
+
+ cpu_latency_qos_remove_request(&info->pm_qos_req);
+
+ return 0;
+}
+#endif
+
+#ifdef CONFIG_PM_SLEEP
+static int imx_rpmsg_pcm_suspend(struct device *dev)
+{
+ struct rpmsg_info *info = dev_get_drvdata(dev);
+ struct rpmsg_msg *rpmsg_tx;
+ struct rpmsg_msg *rpmsg_rx;
+
+ rpmsg_tx = &info->msg[TX_SUSPEND];
+ rpmsg_rx = &info->msg[RX_SUSPEND];
+
+ rpmsg_tx->s_msg.header.cmd = TX_SUSPEND;
+ info->send_message(rpmsg_tx, info);
+
+ rpmsg_rx->s_msg.header.cmd = RX_SUSPEND;
+ info->send_message(rpmsg_rx, info);
+
+ return 0;
+}
+
+static int imx_rpmsg_pcm_resume(struct device *dev)
+{
+ struct rpmsg_info *info = dev_get_drvdata(dev);
+ struct rpmsg_msg *rpmsg_tx;
+ struct rpmsg_msg *rpmsg_rx;
+
+ rpmsg_tx = &info->msg[TX_RESUME];
+ rpmsg_rx = &info->msg[RX_RESUME];
+
+ rpmsg_tx->s_msg.header.cmd = TX_RESUME;
+ info->send_message(rpmsg_tx, info);
+
+ rpmsg_rx->s_msg.header.cmd = RX_RESUME;
+ info->send_message(rpmsg_rx, info);
+
+ return 0;
+}
+#endif /* CONFIG_PM_SLEEP */
+
+static const struct dev_pm_ops imx_rpmsg_pcm_pm_ops = {
+ SET_RUNTIME_PM_OPS(imx_rpmsg_pcm_runtime_suspend,
+ imx_rpmsg_pcm_runtime_resume,
+ NULL)
+ SET_SYSTEM_SLEEP_PM_OPS(imx_rpmsg_pcm_suspend,
+ imx_rpmsg_pcm_resume)
+};
+
+static struct platform_driver imx_pcm_rpmsg_driver = {
+ .probe = imx_rpmsg_pcm_probe,
+ .remove = imx_rpmsg_pcm_remove,
+ .driver = {
+ .name = IMX_PCM_DRV_NAME,
+ .pm = &imx_rpmsg_pcm_pm_ops,
+ },
+};
+module_platform_driver(imx_pcm_rpmsg_driver);
+
+MODULE_DESCRIPTION("Freescale SoC Audio RPMSG PCM interface");
+MODULE_AUTHOR("Shengjiu Wang <[email protected]>");
+MODULE_ALIAS("platform:" IMX_PCM_DRV_NAME);
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/fsl/imx-pcm-rpmsg.h b/sound/soc/fsl/imx-pcm-rpmsg.h
new file mode 100644
index 000000000000..308d153920a3
--- /dev/null
+++ b/sound/soc/fsl/imx-pcm-rpmsg.h
@@ -0,0 +1,512 @@
+/* SPDX-License-Identifier: GPL-2.0+ */
+/*
+ * Copyright 2017-2021 NXP
+ *
+ ******************************************************************************
+ * Communication stack of audio with rpmsg
+ ******************************************************************************
+ * Packet structure:
+ * A SRTM message consists of a 10 bytes header followed by 0~N bytes of data
+ *
+ * +---------------+-------------------------------+
+ * | | Content |
+ * +---------------+-------------------------------+
+ * | Byte Offset | 7 6 5 4 3 2 1 0 |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 0 | Category |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 1 ~ 2 | Version |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 3 | Type |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 4 | Command |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 5 | Reserved0 |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 6 | Reserved1 |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 7 | Reserved2 |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 8 | Reserved3 |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 9 | Reserved4 |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | 10 | DATA 0 |
+ * +---------------+---+---+---+---+---+---+---+---+
+ * : : : : : : : : : : : : :
+ * +---------------+---+---+---+---+---+---+---+---+
+ * | N + 10 - 1 | DATA N-1 |
+ * +---------------+---+---+---+---+---+---+---+---+
+ *
+ * +----------+------------+------------------------------------------------+
+ * | Field | Byte | |
+ * +----------+------------+------------------------------------------------+
+ * | Category | 0 | The destination category. |
+ * +----------+------------+------------------------------------------------+
+ * | Version | 1 ~ 2 | The category version of the sender of the |
+ * | | | packet. |
+ * | | | The first byte represent the major version of |
+ * | | | the packet.The second byte represent the minor |
+ * | | | version of the packet. |
+ * +----------+------------+------------------------------------------------+
+ * | Type | 3 | The message type of current message packet. |
+ * +----------+------------+------------------------------------------------+
+ * | Command | 4 | The command byte sent to remote processor/SoC. |
+ * +----------+------------+------------------------------------------------+
+ * | Reserved | 5 ~ 9 | Reserved field for future extension. |
+ * +----------+------------+------------------------------------------------+
+ * | Data | N | The data payload of the message packet. |
+ * +----------+------------+------------------------------------------------+
+ *
+ * Audio control:
+ * SRTM Audio Control Category Request Command Table:
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | Category | Version | Type | Command | Data | Function |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x00 | Data[0]: Audio Device Index | Open a TX Instance. |
+ * | | | | | Data[1]: format | |
+ * | | | | | Data[2]: channels | |
+ * | | | | | Data[3-6]: samplerate | |
+ * | | | | | Data[7-10]: buffer_addr | |
+ * | | | | | Data[11-14]: buffer_size | |
+ * | | | | | Data[15-18]: period_size | |
+ * | | | | | Data[19-22]: buffer_tail | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x01 | Data[0]: Audio Device Index | Start a TX Instance. |
+ * | | | | | Same as above command | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x02 | Data[0]: Audio Device Index | Pause a TX Instance. |
+ * | | | | | Same as above command | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x03 | Data[0]: Audio Device Index | Resume a TX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x04 | Data[0]: Audio Device Index | Stop a TX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x05 | Data[0]: Audio Device Index | Close a TX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x06 | Data[0]: Audio Device Index | Set Parameters for |
+ * | | | | | Data[1]: format | a TX Instance. |
+ * | | | | | Data[2]: channels | |
+ * | | | | | Data[3-6]: samplerate | |
+ * | | | | | Data[7-22]: reserved | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x07 | Data[0]: Audio Device Index | Set TX Buffer. |
+ * | | | | | Data[1-6]: reserved | |
+ * | | | | | Data[7-10]: buffer_addr | |
+ * | | | | | Data[11-14]: buffer_size | |
+ * | | | | | Data[15-18]: period_size | |
+ * | | | | | Data[19-22]: buffer_tail | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x08 | Data[0]: Audio Device Index | Suspend a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x09 | Data[0]: Audio Device Index | Resume a TX Instance. |
+ * | | | | | Data[1]: format | |
+ * | | | | | Data[2]: channels | |
+ * | | | | | Data[3-6]: samplerate | |
+ * | | | | | Data[7-10]: buffer_addr | |
+ * | | | | | Data[11-14]: buffer_size | |
+ * | | | | | Data[15-18]: period_size | |
+ * | | | | | Data[19-22]: buffer_tail | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x0A | Data[0]: Audio Device Index | Open a RX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x0B | Data[0]: Audio Device Index | Start a RX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x0C | Data[0]: Audio Device Index | Pause a RX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x0D | Data[0]: Audio Device Index | Resume a RX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x0E | Data[0]: Audio Device Index | Stop a RX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x0F | Data[0]: Audio Device Index | Close a RX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x10 | Data[0]: Audio Device Index | Set Parameters for |
+ * | | | | | Data[1]: format | a RX Instance. |
+ * | | | | | Data[2]: channels | |
+ * | | | | | Data[3-6]: samplerate | |
+ * | | | | | Data[7-22]: reserved | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x11 | Data[0]: Audio Device Index | Set RX Buffer. |
+ * | | | | | Data[1-6]: reserved | |
+ * | | | | | Data[7-10]: buffer_addr | |
+ * | | | | | Data[11-14]: buffer_size | |
+ * | | | | | Data[15-18]: period_size | |
+ * | | | | | Data[19-22]: buffer_tail | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x12 | Data[0]: Audio Device Index | Suspend a RX Instance.|
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x13 | Data[0]: Audio Device Index | Resume a RX Instance. |
+ * | | | | | Data[1]: format | |
+ * | | | | | Data[2]: channels | |
+ * | | | | | Data[3-6]: samplerate | |
+ * | | | | | Data[7-10]: buffer_addr | |
+ * | | | | | Data[11-14]: buffer_size | |
+ * | | | | | Data[15-18]: period_size | |
+ * | | | | | Data[19-22]: buffer_tail | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x14 | Data[0]: Audio Device Index | Set register value |
+ * | | | | | Data[1-6]: reserved | to codec |
+ * | | | | | Data[7-10]: register | |
+ * | | | | | Data[11-14]: value | |
+ * | | | | | Data[15-22]: reserved | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x00 | 0x15 | Data[0]: Audio Device Index | Get register value |
+ * | | | | | Data[1-6]: reserved | from codec |
+ * | | | | | Data[7-10]: register | |
+ * | | | | | Data[11-22]: reserved | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * Note 1: See <List of Sample Format> for available value of
+ * Sample Format;
+ * Note 2: See <List of Audio Channels> for available value of Channels;
+ * Note 3: Sample Rate of Set Parameters for an Audio TX Instance
+ * Command and Set Parameters for an Audio RX Instance Command is
+ * in little-endian format.
+ *
+ * SRTM Audio Control Category Response Command Table:
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | Category | Version | Type | Command | Data | Function |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x00 | Data[0]: Audio Device Index | Reply for Open |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x01 | Data[0]: Audio Device Index | Reply for Start |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x02 | Data[0]: Audio Device Index | Reply for Pause |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x03 | Data[0]: Audio Device Index | Reply for Resume |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x04 | Data[0]: Audio Device Index | Reply for Stop |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x05 | Data[0]: Audio Device Index | Reply for Close |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x06 | Data[0]: Audio Device Index | Reply for Set Param |
+ * | | | | | Data[1]: Return code | for a TX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x07 | Data[0]: Audio Device Index | Reply for Set |
+ * | | | | | Data[1]: Return code | TX Buffer |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x08 | Data[0]: Audio Device Index | Reply for Suspend |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x09 | Data[0]: Audio Device Index | Reply for Resume |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x0A | Data[0]: Audio Device Index | Reply for Open |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x0B | Data[0]: Audio Device Index | Reply for Start |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x0C | Data[0]: Audio Device Index | Reply for Pause |
+ * | | | | | Data[1]: Return code | a TX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x0D | Data[0]: Audio Device Index | Reply for Resume |
+ * | | | | | Data[1]: Return code | a RX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x0E | Data[0]: Audio Device Index | Reply for Stop |
+ * | | | | | Data[1]: Return code | a RX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x0F | Data[0]: Audio Device Index | Reply for Close |
+ * | | | | | Data[1]: Return code | a RX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x10 | Data[0]: Audio Device Index | Reply for Set Param |
+ * | | | | | Data[1]: Return code | for a RX Instance. |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x11 | Data[0]: Audio Device Index | Reply for Set |
+ * | | | | | Data[1]: Return code | RX Buffer |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x12 | Data[0]: Audio Device Index | Reply for Suspend |
+ * | | | | | Data[1]: Return code | a RX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x13 | Data[0]: Audio Device Index | Reply for Resume |
+ * | | | | | Data[1]: Return code | a RX Instance |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x14 | Data[0]: Audio Device Index | Reply for Set codec |
+ * | | | | | Data[1]: Return code | register value |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x01 | 0x15 | Data[0]: Audio Device Index | Reply for Get codec |
+ * | | | | | Data[1]: Return code | register value |
+ * | | | | | Data[2-6]: reserved | |
+ * | | | | | Data[7-10]: register | |
+ * | | | | | Data[11-14]: value | |
+ * | | | | | Data[15-22]: reserved | |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ *
+ * SRTM Audio Control Category Notification Command Table:
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | Category | Version | Type | Command | Data | Function |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x02 | 0x00 | Data[0]: Audio Device Index | Notify one TX period |
+ * | | | | | | is finished |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ * | 0x03 | 0x0100 | 0x02 | 0x01 | Data[0]: Audio Device Index | Notify one RX period |
+ * | | | | | | is finished |
+ * +----------+---------+------+---------+-------------------------------+-----------------------+
+ *
+ * List of Sample Format:
+ * +------------------+-----------------------+
+ * | Sample Format | Description |
+ * +------------------+-----------------------+
+ * | 0x0 | S16_LE |
+ * +------------------+-----------------------+
+ * | 0x1 | S24_LE |
+ * +------------------+-----------------------+
+ *
+ * List of Audio Channels
+ * +------------------+-----------------------+
+ * | Audio Channel | Description |
+ * +------------------+-----------------------+
+ * | 0x0 | Left Channel |
+ * +------------------+-----------------------+
+ * | 0x1 | Right Channel |
+ * +------------------+---------------- ------+
+ * | 0x2 | Left & Right Channel |
+ * +------------------+-----------------------+
+ *
+ */
+
+#ifndef _IMX_PCM_RPMSG_H
+#define _IMX_PCM_RPMSG_H
+
+#include <linux/pm_qos.h>
+#include <linux/interrupt.h>
+#include <sound/dmaengine_pcm.h>
+
+#define RPMSG_TIMEOUT 1000
+
+/* RPMSG Command (TYPE A)*/
+#define TX_OPEN 0x0
+#define TX_START 0x1
+#define TX_PAUSE 0x2
+#define TX_RESTART 0x3
+#define TX_TERMINATE 0x4
+#define TX_CLOSE 0x5
+#define TX_HW_PARAM 0x6
+#define TX_BUFFER 0x7
+#define TX_SUSPEND 0x8
+#define TX_RESUME 0x9
+
+#define RX_OPEN 0xA
+#define RX_START 0xB
+#define RX_PAUSE 0xC
+#define RX_RESTART 0xD
+#define RX_TERMINATE 0xE
+#define RX_CLOSE 0xF
+#define RX_HW_PARAM 0x10
+#define RX_BUFFER 0x11
+#define RX_SUSPEND 0x12
+#define RX_RESUME 0x13
+#define SET_CODEC_VALUE 0x14
+#define GET_CODEC_VALUE 0x15
+#define TX_POINTER 0x16
+#define RX_POINTER 0x17
+/* Total msg numver for type A */
+#define MSG_TYPE_A_NUM 0x18
+
+/* RPMSG Command (TYPE C)*/
+#define TX_PERIOD_DONE 0x0
+#define RX_PERIOD_DONE 0x1
+/* Total msg numver for type C */
+#define MSG_TYPE_C_NUM 0x2
+
+#define MSG_MAX_NUM (MSG_TYPE_A_NUM + MSG_TYPE_C_NUM)
+
+#define MSG_TYPE_A 0x0
+#define MSG_TYPE_B 0x1
+#define MSG_TYPE_C 0x2
+
+#define RESP_NONE 0x0
+#define RESP_NOT_ALLOWED 0x1
+#define RESP_SUCCESS 0x2
+#define RESP_FAILED 0x3
+
+#define RPMSG_S16_LE 0x0
+#define RPMSG_S24_LE 0x1
+#define RPMSG_S32_LE 0x2
+#define RPMSG_DSD_U16_LE 0x3
+#define RPMSG_DSD_U24_LE 0x4
+#define RPMSG_DSD_U32_LE 0x5
+
+#define RPMSG_CH_LEFT 0x0
+#define RPMSG_CH_RIGHT 0x1
+#define RPMSG_CH_STEREO 0x2
+
+#define WORK_MAX_NUM 0x30
+
+/* Category define */
+#define IMX_RMPSG_LIFECYCLE 1
+#define IMX_RPMSG_PMIC 2
+#define IMX_RPMSG_AUDIO 3
+#define IMX_RPMSG_KEY 4
+#define IMX_RPMSG_GPIO 5
+#define IMX_RPMSG_RTC 6
+#define IMX_RPMSG_SENSOR 7
+
+/* rpmsg version */
+#define IMX_RMPSG_MAJOR 1
+#define IMX_RMPSG_MINOR 0
+
+#define TX SNDRV_PCM_STREAM_PLAYBACK
+#define RX SNDRV_PCM_STREAM_CAPTURE
+
+/**
+ * struct rpmsg_head: rpmsg header structure
+ *
+ * @cate: category
+ * @major: major version
+ * @minor: minor version
+ * @type: message type (A/B/C)
+ * @cmd: message command
+ * @reserved: reserved space
+ */
+struct rpmsg_head {
+ u8 cate;
+ u8 major;
+ u8 minor;
+ u8 type;
+ u8 cmd;
+ u8 reserved[5];
+} __packed;
+
+/**
+ * struct param_s: sent rpmsg parameter
+ *
+ * @audioindex: audio instance index
+ * @format: audio format
+ * @channels: audio channel number
+ * @rate: sample rate
+ * @buffer_addr: dma buffer physical address or register for SET_CODEC_VALUE
+ * @buffer_size: dma buffer size or register value for SET_CODEC_VALUE
+ * @period_size: period size
+ * @buffer_tail: current period index
+ */
+struct param_s {
+ unsigned char audioindex;
+ unsigned char format;
+ unsigned char channels;
+ unsigned int rate;
+ unsigned int buffer_addr;
+ unsigned int buffer_size;
+ unsigned int period_size;
+ unsigned int buffer_tail;
+} __packed;
+
+/**
+ * struct param_s: send rpmsg parameter
+ *
+ * @audioindex: audio instance index
+ * @resp: response value
+ * @reserved1: reserved space
+ * @buffer_offset: the consumed offset of buffer
+ * @reg_addr: register addr of codec
+ * @reg_data: register value of codec
+ * @reserved2: reserved space
+ * @buffer_tail: current period index
+ */
+struct param_r {
+ unsigned char audioindex;
+ unsigned char resp;
+ unsigned char reserved1[1];
+ unsigned int buffer_offset;
+ unsigned int reg_addr;
+ unsigned int reg_data;
+ unsigned char reserved2[4];
+ unsigned int buffer_tail;
+} __packed;
+
+/* Struct of sent message */
+struct rpmsg_s_msg {
+ struct rpmsg_head header;
+ struct param_s param;
+};
+
+/* Struct of received message */
+struct rpmsg_r_msg {
+ struct rpmsg_head header;
+ struct param_r param;
+};
+
+/* Struct of rpmsg */
+struct rpmsg_msg {
+ struct rpmsg_s_msg s_msg;
+ struct rpmsg_r_msg r_msg;
+};
+
+/* Struct of rpmsg for workqueue */
+struct work_of_rpmsg {
+ struct rpmsg_info *info;
+ /* Sent msg for each work */
+ struct rpmsg_msg msg;
+ struct work_struct work;
+};
+
+/* Struct of timer */
+struct stream_timer {
+ struct timer_list timer;
+ struct rpmsg_info *info;
+ struct snd_pcm_substream *substream;
+};
+
+typedef void (*dma_callback)(void *arg);
+
+/**
+ * struct rpmsg_info: rpmsg audio information
+ *
+ * @rpdev: pointer of rpmsg_device
+ * @dev: pointer for imx_pcm_rpmsg device
+ * @cmd_complete: command is finished
+ * @pm_qos_req: request of pm qos
+ * @r_msg: received rpmsg
+ * @msg: array of rpmsg
+ * @notify: notification msg (type C) for TX & RX
+ * @notify_updated: notification flag for TX & RX
+ * @rpmsg_wq: rpmsg workqueue
+ * @work_list: array of work list for workqueue
+ * @work_write_index: write index of work list
+ * @work_read_index: read index of work list
+ * @msg_drop_count: counter of dropped msg for TX & RX
+ * @num_period: period number for TX & RX
+ * @callback_param: parameter for period elapse callback for TX & RX
+ * @callback: period elapse callback for TX & RX
+ * @send_message: function pointer for send message
+ * @lock: spin lock for TX & RX
+ * @wq_lock: lock for work queue
+ * @msg_lock: lock for send message
+ * @stream_timer: timer for tigger workqueue
+ */
+struct rpmsg_info {
+ struct rpmsg_device *rpdev;
+ struct device *dev;
+ struct completion cmd_complete;
+ struct pm_qos_request pm_qos_req;
+
+ /* Received msg (global) */
+ struct rpmsg_r_msg r_msg;
+ struct rpmsg_msg msg[MSG_MAX_NUM];
+ /* period done */
+ struct rpmsg_msg notify[2];
+ bool notify_updated[2];
+
+ struct workqueue_struct *rpmsg_wq;
+ struct work_of_rpmsg work_list[WORK_MAX_NUM];
+ int work_write_index;
+ int work_read_index;
+ int msg_drop_count[2];
+ int num_period[2];
+ void *callback_param[2];
+ dma_callback callback[2];
+ int (*send_message)(struct rpmsg_msg *msg, struct rpmsg_info *info);
+ spinlock_t lock[2]; /* spin lock for resource protection */
+ spinlock_t wq_lock; /* spin lock for resource protection */
+ struct mutex msg_lock; /* mutex for resource protection */
+ struct stream_timer stream_timer[2];
+};
+
+#define IMX_PCM_DRV_NAME "imx_pcm_rpmsg"
+
+#endif /* IMX_PCM_RPMSG_H */
--
2.27.0
fsl_rpmsg cpu dai driver is driver for rpmsg audio, which is mainly used
for getting the user's configuration from device tree and configure the
clocks which is used by Cortex-M core. So in this document define the
needed property.
Signed-off-by: Shengjiu Wang <[email protected]>
---
.../devicetree/bindings/sound/fsl,rpmsg.yaml | 118 ++++++++++++++++++
1 file changed, 118 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
new file mode 100644
index 000000000000..5731c1fbc0a6
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
@@ -0,0 +1,118 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/fsl,rpmsg.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP Audio RPMSG CPU DAI Controller
+
+maintainers:
+ - Shengjiu Wang <[email protected]>
+
+description: |
+ fsl_rpmsg cpu dai driver is virtual driver for rpmsg audio, which doesn't
+ touch hardware. It is mainly used for getting the user's configuration
+ from device tree and configure the clocks which is used by Cortex-M core.
+ So in this document define the needed property.
+
+properties:
+ compatible:
+ enum:
+ - fsl,imx7ulp-rpmsg
+ - fsl,imx8mn-rpmsg
+ - fsl,imx8mm-rpmsg
+ - fsl,imx8mp-rpmsg
+
+ model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: User specified audio sound card name
+
+ clocks:
+ items:
+ - description: Peripheral clock for register access
+ - description: Master clock
+ - description: DMA clock for DMA register access
+ - description: Parent clock for multiple of 8kHz sample rates
+ - description: Parent clock for multiple of 11kHz sample rates
+ minItems: 5
+
+ clock-names:
+ items:
+ - const: ipg
+ - const: mclk
+ - const: dma
+ - const: pll8k
+ - const: pll11k
+ minItems: 5
+
+ power-domains:
+ maxItems: 1
+
+ fsl,audioindex:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1]
+ default: 0
+ description: Instance index for sound card in
+ M core side, which share one rpmsg
+ channel.
+
+ fsl,version:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [1, 2]
+ default: 2
+ description: The version of M core image, which is
+ to make driver compatible with different image.
+
+ fsl,buffer-size:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: pre allocate dma buffer size
+
+ fsl,enable-lpa:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description: enable low power audio path.
+
+ fsl,rpmsg-out:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description: |
+ This is a boolean property. If present, the transmitting function
+ will be enabled.
+
+ fsl,rpmsg-in:
+ $ref: /schemas/types.yaml#/definitions/flag
+ description: |
+ This is a boolean property. If present, the receiving function
+ will be enabled.
+
+ fsl,codec-type:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ enum: [0, 1, 2]
+ default: 0
+ description: Sometimes the codec is registered by
+ driver not by the device tree, this items
+ can be used to distinguish codecs.
+
+ audio-codec:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: The phandle of the audio codec
+
+ memory-region:
+ $ref: /schemas/types.yaml#/definitions/phandle
+ description: phandle to the reserved memory nodes
+
+required:
+ - compatible
+ - fsl,audioindex
+ - fsl,version
+ - fsl,buffer-size
+
+additionalProperties: false
+
+examples:
+ - |
+ rpmsg_audio: rpmsg_audio {
+ compatible = "fsl,imx8mn-rpmsg";
+ fsl,audioindex = <0> ;
+ fsl,version = <2>;
+ fsl,buffer-size = <0x6000000>;
+ fsl,enable-lpa;
+ };
--
2.27.0
The platform device is not registered by device tree or
cpu dai driver, it is registered by the rpmsg channel,
So add a dedicated machine driver to handle this case.
Signed-off-by: Shengjiu Wang <[email protected]>
---
sound/soc/fsl/Kconfig | 12 ++++
sound/soc/fsl/Makefile | 2 +
sound/soc/fsl/imx-rpmsg.c | 127 ++++++++++++++++++++++++++++++++++++++
3 files changed, 141 insertions(+)
create mode 100644 sound/soc/fsl/imx-rpmsg.c
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 3b94882aee99..ad69026a7fd1 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -334,6 +334,18 @@ config SND_SOC_IMX_HDMI
Say Y if you want to add support for SoC audio on an i.MX board with
IMX HDMI.
+config SND_SOC_IMX_RPMSG
+ tristate "SoC Audio support for i.MX boards with rpmsg"
+ depends on RPMSG
+ select SND_SOC_IMX_PCM_RPMSG
+ select SND_SOC_IMX_AUDIO_RPMSG
+ select SND_SOC_FSL_RPMSG
+ help
+ SoC Audio support for i.MX boards with rpmsg.
+ There should be rpmsg devices defined in other core (M core)
+ Say Y if you want to add support for SoC audio on an i.MX board with
+ a rpmsg devices.
+
endif # SND_IMX_SOC
endmenu
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index ce4f4324c3a2..f146ce464acd 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -70,6 +70,7 @@ snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
snd-soc-imx-spdif-objs := imx-spdif.o
snd-soc-imx-audmix-objs := imx-audmix.o
snd-soc-imx-hdmi-objs := imx-hdmi.o
+snd-soc-imx-rpmsg-objs := imx-rpmsg.o
obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o
@@ -77,3 +78,4 @@ obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o
obj-$(CONFIG_SND_SOC_IMX_HDMI) += snd-soc-imx-hdmi.o
+obj-$(CONFIG_SND_SOC_IMX_RPMSG) += snd-soc-imx-rpmsg.o
diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c
new file mode 100644
index 000000000000..c26531525a7b
--- /dev/null
+++ b/sound/soc/fsl/imx-rpmsg.c
@@ -0,0 +1,127 @@
+// SPDX-License-Identifier: GPL-2.0+
+// Copyright 2017-2020 NXP
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <linux/of_reserved_mem.h>
+#include <linux/i2c.h>
+#include <linux/of_gpio.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+#include <linux/clk.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/control.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dapm.h>
+#include "imx-pcm-rpmsg.h"
+
+struct imx_rpmsg {
+ struct snd_soc_dai_link dai;
+ struct snd_soc_card card;
+};
+
+static int imx_rpmsg_probe(struct platform_device *pdev)
+{
+ struct snd_soc_dai_link_component *dlc;
+ struct device *dev = pdev->dev.parent;
+ /* rpmsg_pdev is the platform device for the rpmsg node that probed us */
+ struct platform_device *rpmsg_pdev = to_platform_device(dev);
+ struct device_node *np = rpmsg_pdev->dev.of_node;
+ struct of_phandle_args args;
+ struct imx_rpmsg *data;
+ int ret = 0;
+
+ dlc = devm_kzalloc(&pdev->dev, 3 * sizeof(*dlc), GFP_KERNEL);
+ if (!dlc)
+ return -ENOMEM;
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
+ ret = of_reserved_mem_device_init_by_idx(&pdev->dev, np, 0);
+ if (ret)
+ dev_warn(&pdev->dev, "no reserved DMA memory\n");
+
+ data->dai.cpus = &dlc[0];
+ data->dai.num_cpus = 1;
+ data->dai.platforms = &dlc[1];
+ data->dai.num_platforms = 1;
+ data->dai.codecs = &dlc[2];
+ data->dai.num_codecs = 1;
+
+ data->dai.name = "rpmsg hifi";
+ data->dai.stream_name = "rpmsg hifi";
+ data->dai.dai_fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS;
+
+ /* Optional codec node */
+ ret = of_parse_phandle_with_fixed_args(np, "audio-codec", 0, 0, &args);
+ if (ret) {
+ data->dai.codecs->dai_name = "snd-soc-dummy-dai";
+ data->dai.codecs->name = "snd-soc-dummy";
+ } else {
+ data->dai.codecs->of_node = args.np;
+ ret = snd_soc_get_dai_name(&args, &data->dai.codecs->dai_name);
+ if (ret) {
+ dev_err(&pdev->dev, "Unable to get codec_dai_name\n");
+ goto fail;
+ }
+ }
+
+ data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev);
+ data->dai.platforms->name = IMX_PCM_DRV_NAME;
+ data->dai.playback_only = true;
+ data->dai.capture_only = true;
+ data->card.num_links = 1;
+ data->card.dai_link = &data->dai;
+
+ if (of_property_read_bool(np, "fsl,rpmsg-out"))
+ data->dai.capture_only = false;
+
+ if (of_property_read_bool(np, "fsl,rpmsg-in"))
+ data->dai.playback_only = false;
+
+ if (data->dai.playback_only && data->dai.capture_only) {
+ dev_err(&pdev->dev, "no enabled rpmsg DAI link\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ data->card.dev = &pdev->dev;
+ data->card.owner = THIS_MODULE;
+
+ ret = of_property_read_string_index(np, "model", 0, &data->card.name);
+ if (ret)
+ goto fail;
+
+ platform_set_drvdata(pdev, &data->card);
+ snd_soc_card_set_drvdata(&data->card, data);
+ ret = devm_snd_soc_register_card(&pdev->dev, &data->card);
+ if (ret) {
+ dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+ goto fail;
+ }
+
+fail:
+ return ret;
+}
+
+static struct platform_driver imx_rpmsg_driver = {
+ .driver = {
+ .name = "imx-audio-rpmsg",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = imx_rpmsg_probe,
+};
+module_platform_driver(imx_rpmsg_driver);
+
+MODULE_DESCRIPTION("Freescale SoC Audio RPMSG Machine Driver");
+MODULE_AUTHOR("Shengjiu Wang <[email protected]>");
+MODULE_ALIAS("platform:imx-audio-rpmsg");
+MODULE_LICENSE("GPL v2");
--
2.27.0
Hi Shengjiu,
I love your patch! Yet something to improve:
[auto build test ERROR on asoc/for-next]
[also build test ERROR on shawnguo/for-next v5.12-rc2]
[If your patch is applied to the wrong git tree, kindly drop us a note.
And when submitting patch, we suggest to use '--base' as documented in
https://git-scm.com/docs/git-format-patch]
url: https://github.com/0day-ci/linux/commits/Shengjiu-Wang/Add-audio-driver-base-on-rpmsg-on-i-MX-platform/20210308-213802
base: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next
config: sh-allmodconfig (attached as .config)
compiler: sh4-linux-gcc (GCC) 9.3.0
reproduce (this is a W=1 build):
wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross
chmod +x ~/bin/make.cross
# https://github.com/0day-ci/linux/commit/0cabc4a6bce0db6d63de8cfffbdc5fe53c4ea6f5
git remote add linux-review https://github.com/0day-ci/linux
git fetch --no-tags linux-review Shengjiu-Wang/Add-audio-driver-base-on-rpmsg-on-i-MX-platform/20210308-213802
git checkout 0cabc4a6bce0db6d63de8cfffbdc5fe53c4ea6f5
# save the attached .config to linux build tree
COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=sh
If you fix the issue, kindly add following tag as appropriate
Reported-by: kernel test robot <[email protected]>
All errors (new ones prefixed by >>, old ones prefixed by <<):
>> ERROR: modpost: "__clk_get_name" [sound/soc/fsl/snd-soc-fsl-rpmsg.ko] undefined!
ERROR: modpost: "__delay" [drivers/net/mdio/mdio-cavium.ko] undefined!
ERROR: modpost: "__udivdi3" [fs/btrfs/btrfs.ko] undefined!
ERROR: modpost: "__umoddi3" [fs/btrfs/btrfs.ko] undefined!
Kconfig warnings: (for reference only)
WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC
Depends on SOUND && !UML && SND && SND_SOC && SND_ATMEL_SOC && HAS_DMA
Selected by
- SND_ATMEL_SOC_SSC && SOUND && !UML && SND && SND_SOC && SND_ATMEL_SOC
- SND_ATMEL_SOC_SSC_PDC && SOUND && !UML && SND && SND_SOC && SND_ATMEL_SOC && ATMEL_SSC
---
0-DAY CI Kernel Test Service, Intel Corporation
https://lists.01.org/hyperkitty/list/[email protected]
On Mon, Mar 08, 2021 at 09:22:27PM +0800, Shengjiu Wang wrote:
> fsl_rpmsg cpu dai driver is driver for rpmsg audio, which is mainly used
Bindings describe h/w blocks, not drivers.
> for getting the user's configuration from device tree and configure the
> clocks which is used by Cortex-M core. So in this document define the
> needed property.
>
> Signed-off-by: Shengjiu Wang <[email protected]>
> ---
> .../devicetree/bindings/sound/fsl,rpmsg.yaml | 118 ++++++++++++++++++
> 1 file changed, 118 insertions(+)
> create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
>
> diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> new file mode 100644
> index 000000000000..5731c1fbc0a6
> --- /dev/null
> +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> @@ -0,0 +1,118 @@
> +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
> +%YAML 1.2
> +---
> +$id: http://devicetree.org/schemas/sound/fsl,rpmsg.yaml#
> +$schema: http://devicetree.org/meta-schemas/core.yaml#
> +
> +title: NXP Audio RPMSG CPU DAI Controller
> +
> +maintainers:
> + - Shengjiu Wang <[email protected]>
> +
> +description: |
> + fsl_rpmsg cpu dai driver is virtual driver for rpmsg audio, which doesn't
> + touch hardware. It is mainly used for getting the user's configuration
> + from device tree and configure the clocks which is used by Cortex-M core.
> + So in this document define the needed property.
> +
> +properties:
> + compatible:
> + enum:
> + - fsl,imx7ulp-rpmsg
> + - fsl,imx8mn-rpmsg
> + - fsl,imx8mm-rpmsg
> + - fsl,imx8mp-rpmsg
> +
> + model:
> + $ref: /schemas/types.yaml#/definitions/string
> + description: User specified audio sound card name
> +
> + clocks:
> + items:
> + - description: Peripheral clock for register access
> + - description: Master clock
> + - description: DMA clock for DMA register access
> + - description: Parent clock for multiple of 8kHz sample rates
> + - description: Parent clock for multiple of 11kHz sample rates
> + minItems: 5
If this doesn't touch hardware, what are these clocks for?
You don't need 'minItems' unless it's less than the number of 'items'.
> +
> + clock-names:
> + items:
> + - const: ipg
> + - const: mclk
> + - const: dma
> + - const: pll8k
> + - const: pll11k
> + minItems: 5
> +
> + power-domains:
> + maxItems: 1
> +
> + fsl,audioindex:
> + $ref: /schemas/types.yaml#/definitions/uint32
> + enum: [0, 1]
> + default: 0
> + description: Instance index for sound card in
> + M core side, which share one rpmsg
> + channel.
We don't do indexes in DT. What's this numbering tied to?
> +
> + fsl,version:
version of what?
This seems odd at best.
> + $ref: /schemas/types.yaml#/definitions/uint32
> + enum: [1, 2]
You're going to update this with every new firmware version?
> + default: 2
> + description: The version of M core image, which is
> + to make driver compatible with different image.
> +
> + fsl,buffer-size:
> + $ref: /schemas/types.yaml#/definitions/uint32
> + description: pre allocate dma buffer size
How can you have DMA, this doesn't touch h/w?
> +
> + fsl,enable-lpa:
> + $ref: /schemas/types.yaml#/definitions/flag
> + description: enable low power audio path.
> +
> + fsl,rpmsg-out:
> + $ref: /schemas/types.yaml#/definitions/flag
> + description: |
> + This is a boolean property. If present, the transmitting function
> + will be enabled.
> +
> + fsl,rpmsg-in:
> + $ref: /schemas/types.yaml#/definitions/flag
> + description: |
> + This is a boolean property. If present, the receiving function
> + will be enabled.
> +
> + fsl,codec-type:
> + $ref: /schemas/types.yaml#/definitions/uint32
> + enum: [0, 1, 2]
> + default: 0
> + description: Sometimes the codec is registered by
> + driver not by the device tree, this items
> + can be used to distinguish codecs.
How does one decide what value to use?
> +
> + audio-codec:
> + $ref: /schemas/types.yaml#/definitions/phandle
> + description: The phandle of the audio codec
The codec is controlled from the Linux side?
> +
> + memory-region:
> + $ref: /schemas/types.yaml#/definitions/phandle
> + description: phandle to the reserved memory nodes
> +
> +required:
> + - compatible
> + - fsl,audioindex
> + - fsl,version
> + - fsl,buffer-size
> +
> +additionalProperties: false
> +
> +examples:
> + - |
> + rpmsg_audio: rpmsg_audio {
> + compatible = "fsl,imx8mn-rpmsg";
> + fsl,audioindex = <0> ;
> + fsl,version = <2>;
> + fsl,buffer-size = <0x6000000>;
> + fsl,enable-lpa;
How does this work? Don't you need somewhere to put the 'rpmsg' data?
> + };
> --
> 2.27.0
>
Hi Rob
On Wed, Mar 10, 2021 at 10:49 AM Rob Herring <[email protected]> wrote:
>
> On Mon, Mar 08, 2021 at 09:22:27PM +0800, Shengjiu Wang wrote:
> > fsl_rpmsg cpu dai driver is driver for rpmsg audio, which is mainly used
>
> Bindings describe h/w blocks, not drivers.
I will modify the descriptions. but here it is a virtual device. the
mapping in real h/w is cortex M core, Cortex M core controls the SAI,
DMA interface. What we see from Linux side is a audio service
through rpmsg channel.
>
> > for getting the user's configuration from device tree and configure the
> > clocks which is used by Cortex-M core. So in this document define the
> > needed property.
> >
> > Signed-off-by: Shengjiu Wang <[email protected]>
> > ---
> > .../devicetree/bindings/sound/fsl,rpmsg.yaml | 118 ++++++++++++++++++
> > 1 file changed, 118 insertions(+)
> > create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> >
> > diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> > new file mode 100644
> > index 000000000000..5731c1fbc0a6
> > --- /dev/null
> > +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> > @@ -0,0 +1,118 @@
> > +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
> > +%YAML 1.2
> > +---
> > +$id: http://devicetree.org/schemas/sound/fsl,rpmsg.yaml#
> > +$schema: http://devicetree.org/meta-schemas/core.yaml#
> > +
> > +title: NXP Audio RPMSG CPU DAI Controller
> > +
> > +maintainers:
> > + - Shengjiu Wang <[email protected]>
> > +
> > +description: |
> > + fsl_rpmsg cpu dai driver is virtual driver for rpmsg audio, which doesn't
> > + touch hardware. It is mainly used for getting the user's configuration
> > + from device tree and configure the clocks which is used by Cortex-M core.
> > + So in this document define the needed property.
> > +
> > +properties:
> > + compatible:
> > + enum:
> > + - fsl,imx7ulp-rpmsg
> > + - fsl,imx8mn-rpmsg
> > + - fsl,imx8mm-rpmsg
> > + - fsl,imx8mp-rpmsg
> > +
> > + model:
> > + $ref: /schemas/types.yaml#/definitions/string
> > + description: User specified audio sound card name
> > +
> > + clocks:
> > + items:
> > + - description: Peripheral clock for register access
> > + - description: Master clock
> > + - description: DMA clock for DMA register access
> > + - description: Parent clock for multiple of 8kHz sample rates
> > + - description: Parent clock for multiple of 11kHz sample rates
> > + minItems: 5
>
> If this doesn't touch hardware, what are these clocks for?
When the cortex-M core support audio service, these clock
needed prepared & enabled by ALSA driver.
>
> You don't need 'minItems' unless it's less than the number of 'items'.
Ok, I will remove this minItems.
>
> > +
> > + clock-names:
> > + items:
> > + - const: ipg
> > + - const: mclk
> > + - const: dma
> > + - const: pll8k
> > + - const: pll11k
> > + minItems: 5
> > +
> > + power-domains:
> > + maxItems: 1
> > +
> > + fsl,audioindex:
> > + $ref: /schemas/types.yaml#/definitions/uint32
> > + enum: [0, 1]
> > + default: 0
> > + description: Instance index for sound card in
> > + M core side, which share one rpmsg
> > + channel.
>
> We don't do indexes in DT. What's this numbering tied to?
I will remove it. it is not necessary
>
> > +
> > + fsl,version:
>
> version of what?
>
> This seems odd at best.
>
I will remove it. it is not necessary
> > + $ref: /schemas/types.yaml#/definitions/uint32
> > + enum: [1, 2]
>
> You're going to update this with every new firmware version?
>
> > + default: 2
> > + description: The version of M core image, which is
> > + to make driver compatible with different image.
> > +
> > + fsl,buffer-size:
> > + $ref: /schemas/types.yaml#/definitions/uint32
> > + description: pre allocate dma buffer size
>
> How can you have DMA, this doesn't touch h/w?
The DMA is handled by M core image for sound playback
and capture. we need to allocated buffer in Linux side.
here just make the buffer size to be configurable.
>
> > +
> > + fsl,enable-lpa:
> > + $ref: /schemas/types.yaml#/definitions/flag
> > + description: enable low power audio path.
> > +
> > + fsl,rpmsg-out:
> > + $ref: /schemas/types.yaml#/definitions/flag
> > + description: |
> > + This is a boolean property. If present, the transmitting function
> > + will be enabled.
> > +
> > + fsl,rpmsg-in:
> > + $ref: /schemas/types.yaml#/definitions/flag
> > + description: |
> > + This is a boolean property. If present, the receiving function
> > + will be enabled.
> > +
> > + fsl,codec-type:
> > + $ref: /schemas/types.yaml#/definitions/uint32
> > + enum: [0, 1, 2]
> > + default: 0
> > + description: Sometimes the codec is registered by
> > + driver not by the device tree, this items
> > + can be used to distinguish codecs.
>
> How does one decide what value to use?
I will add more description:
0: dummy codec
1: WM8960 codec
2: AK4497 codec
>
> > +
> > + audio-codec:
> > + $ref: /schemas/types.yaml#/definitions/phandle
> > + description: The phandle of the audio codec
>
> The codec is controlled from the Linux side?
yes.
>
> > +
> > + memory-region:
> > + $ref: /schemas/types.yaml#/definitions/phandle
> > + description: phandle to the reserved memory nodes
> > +
> > +required:
> > + - compatible
> > + - fsl,audioindex
> > + - fsl,version
> > + - fsl,buffer-size
> > +
> > +additionalProperties: false
> > +
> > +examples:
> > + - |
> > + rpmsg_audio: rpmsg_audio {
> > + compatible = "fsl,imx8mn-rpmsg";
> > + fsl,audioindex = <0> ;
> > + fsl,version = <2>;
> > + fsl,buffer-size = <0x6000000>;
> > + fsl,enable-lpa;
>
> How does this work? Don't you need somewhere to put the 'rpmsg' data?
The rpmsg data is not handled in this "rpmsg_audio" device, it is just to
prepare the resource for rpmsg audio function, the clock, the memory
the power...
The rpmsg data is handled in imx-pcm-rpmsg.c and imx-audio-rpmsg.c
These devices is registered by imx remoteproc driver.
I will update this document in v5
Best regards
Wang Shengjiu
On Wed, Mar 10, 2021 at 6:33 AM Shengjiu Wang <[email protected]> wrote:
>
> Hi Rob
>
> On Wed, Mar 10, 2021 at 10:49 AM Rob Herring <[email protected]> wrote:
> >
> > On Mon, Mar 08, 2021 at 09:22:27PM +0800, Shengjiu Wang wrote:
> > > fsl_rpmsg cpu dai driver is driver for rpmsg audio, which is mainly used
> >
> > Bindings describe h/w blocks, not drivers.
>
> I will modify the descriptions. but here it is a virtual device. the
> mapping in real h/w is cortex M core, Cortex M core controls the SAI,
> DMA interface. What we see from Linux side is a audio service
> through rpmsg channel.
It's describing the h/w from the view of the OS. It's not important
that it's a Cortex-M, but how you interface to it whether that's
shared registers, mailbox, etc. And it's what resources the block uses
that the OS controls.
> > > for getting the user's configuration from device tree and configure the
> > > clocks which is used by Cortex-M core. So in this document define the
> > > needed property.
> > >
> > > Signed-off-by: Shengjiu Wang <[email protected]>
> > > ---
> > > .../devicetree/bindings/sound/fsl,rpmsg.yaml | 118 ++++++++++++++++++
> > > 1 file changed, 118 insertions(+)
> > > create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> > >
> > > diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> > > new file mode 100644
> > > index 000000000000..5731c1fbc0a6
> > > --- /dev/null
> > > +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> > > @@ -0,0 +1,118 @@
> > > +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
> > > +%YAML 1.2
> > > +---
> > > +$id: http://devicetree.org/schemas/sound/fsl,rpmsg.yaml#
> > > +$schema: http://devicetree.org/meta-schemas/core.yaml#
> > > +
> > > +title: NXP Audio RPMSG CPU DAI Controller
> > > +
> > > +maintainers:
> > > + - Shengjiu Wang <[email protected]>
> > > +
> > > +description: |
> > > + fsl_rpmsg cpu dai driver is virtual driver for rpmsg audio, which doesn't
> > > + touch hardware. It is mainly used for getting the user's configuration
> > > + from device tree and configure the clocks which is used by Cortex-M core.
> > > + So in this document define the needed property.
> > > +
> > > +properties:
> > > + compatible:
> > > + enum:
> > > + - fsl,imx7ulp-rpmsg
> > > + - fsl,imx8mn-rpmsg
> > > + - fsl,imx8mm-rpmsg
> > > + - fsl,imx8mp-rpmsg
> > > +
> > > + model:
> > > + $ref: /schemas/types.yaml#/definitions/string
> > > + description: User specified audio sound card name
> > > +
> > > + clocks:
> > > + items:
> > > + - description: Peripheral clock for register access
> > > + - description: Master clock
> > > + - description: DMA clock for DMA register access
> > > + - description: Parent clock for multiple of 8kHz sample rates
> > > + - description: Parent clock for multiple of 11kHz sample rates
> > > + minItems: 5
> >
> > If this doesn't touch hardware, what are these clocks for?
>
> When the cortex-M core support audio service, these clock
> needed prepared & enabled by ALSA driver.
>
> >
> > You don't need 'minItems' unless it's less than the number of 'items'.
>
> Ok, I will remove this minItems.
>
> >
> > > +
> > > + clock-names:
> > > + items:
> > > + - const: ipg
> > > + - const: mclk
> > > + - const: dma
> > > + - const: pll8k
> > > + - const: pll11k
> > > + minItems: 5
> > > +
> > > + power-domains:
> > > + maxItems: 1
> > > +
> > > + fsl,audioindex:
> > > + $ref: /schemas/types.yaml#/definitions/uint32
> > > + enum: [0, 1]
> > > + default: 0
> > > + description: Instance index for sound card in
> > > + M core side, which share one rpmsg
> > > + channel.
> >
> > We don't do indexes in DT. What's this numbering tied to?
>
> I will remove it. it is not necessary
>
> >
> > > +
> > > + fsl,version:
> >
> > version of what?
> >
> > This seems odd at best.
> >
>
> I will remove it. it is not necessary
>
> > > + $ref: /schemas/types.yaml#/definitions/uint32
> > > + enum: [1, 2]
> >
> > You're going to update this with every new firmware version?
> >
> > > + default: 2
> > > + description: The version of M core image, which is
> > > + to make driver compatible with different image.
> > > +
> > > + fsl,buffer-size:
> > > + $ref: /schemas/types.yaml#/definitions/uint32
> > > + description: pre allocate dma buffer size
> >
> > How can you have DMA, this doesn't touch h/w?
>
> The DMA is handled by M core image for sound playback
> and capture. we need to allocated buffer in Linux side.
> here just make the buffer size to be configurable.
Do we set audio buffer sizes for other audio devices in DT? If not,
why is this special? If so, why is it not common.
> > > + fsl,enable-lpa:
> > > + $ref: /schemas/types.yaml#/definitions/flag
> > > + description: enable low power audio path.
> > > +
> > > + fsl,rpmsg-out:
> > > + $ref: /schemas/types.yaml#/definitions/flag
> > > + description: |
> > > + This is a boolean property. If present, the transmitting function
> > > + will be enabled.
> > > +
> > > + fsl,rpmsg-in:
> > > + $ref: /schemas/types.yaml#/definitions/flag
> > > + description: |
> > > + This is a boolean property. If present, the receiving function
> > > + will be enabled.
> > > +
> > > + fsl,codec-type:
> > > + $ref: /schemas/types.yaml#/definitions/uint32
> > > + enum: [0, 1, 2]
> > > + default: 0
> > > + description: Sometimes the codec is registered by
> > > + driver not by the device tree, this items
> > > + can be used to distinguish codecs.
> >
> > How does one decide what value to use?
>
> I will add more description:
> 0: dummy codec
> 1: WM8960 codec
> 2: AK4497 codec
I assume the last 2 cases have nodes in DT (pointed to by
'audio-codec'), so this is redundant.
> > > +
> > > + audio-codec:
> > > + $ref: /schemas/types.yaml#/definitions/phandle
> > > + description: The phandle of the audio codec
> >
> > The codec is controlled from the Linux side?
>
> yes.
>
> >
> > > +
> > > + memory-region:
> > > + $ref: /schemas/types.yaml#/definitions/phandle
> > > + description: phandle to the reserved memory nodes
> > > +
> > > +required:
> > > + - compatible
> > > + - fsl,audioindex
> > > + - fsl,version
> > > + - fsl,buffer-size
> > > +
> > > +additionalProperties: false
> > > +
> > > +examples:
> > > + - |
> > > + rpmsg_audio: rpmsg_audio {
> > > + compatible = "fsl,imx8mn-rpmsg";
> > > + fsl,audioindex = <0> ;
> > > + fsl,version = <2>;
> > > + fsl,buffer-size = <0x6000000>;
> > > + fsl,enable-lpa;
> >
> > How does this work? Don't you need somewhere to put the 'rpmsg' data?
>
> The rpmsg data is not handled in this "rpmsg_audio" device, it is just to
> prepare the resource for rpmsg audio function, the clock, the memory
> the power...
>
> The rpmsg data is handled in imx-pcm-rpmsg.c and imx-audio-rpmsg.c
> These devices is registered by imx remoteproc driver.
Then what is 'memory-region' for? You need that, a mailbox, or ???
somewhere in DT.
Rob
Hi Rob
On Thu, Mar 11, 2021 at 5:12 AM Rob Herring <[email protected]> wrote:
>
> On Wed, Mar 10, 2021 at 6:33 AM Shengjiu Wang <[email protected]> wrote:
> >
> > Hi Rob
> >
> > On Wed, Mar 10, 2021 at 10:49 AM Rob Herring <[email protected]> wrote:
> > >
> > > On Mon, Mar 08, 2021 at 09:22:27PM +0800, Shengjiu Wang wrote:
> > > > fsl_rpmsg cpu dai driver is driver for rpmsg audio, which is mainly used
> > >
> > > Bindings describe h/w blocks, not drivers.
> >
> > I will modify the descriptions. but here it is a virtual device. the
> > mapping in real h/w is cortex M core, Cortex M core controls the SAI,
> > DMA interface. What we see from Linux side is a audio service
> > through rpmsg channel.
>
> It's describing the h/w from the view of the OS. It's not important
> that it's a Cortex-M, but how you interface to it whether that's
> shared registers, mailbox, etc. And it's what resources the block uses
> that the OS controls.
ok.
>
> > > > for getting the user's configuration from device tree and configure the
> > > > clocks which is used by Cortex-M core. So in this document define the
> > > > needed property.
> > > >
> > > > Signed-off-by: Shengjiu Wang <[email protected]>
> > > > ---
> > > > .../devicetree/bindings/sound/fsl,rpmsg.yaml | 118 ++++++++++++++++++
> > > > 1 file changed, 118 insertions(+)
> > > > create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> > > >
> > > > diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> > > > new file mode 100644
> > > > index 000000000000..5731c1fbc0a6
> > > > --- /dev/null
> > > > +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> > > > @@ -0,0 +1,118 @@
> > > > +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
> > > > +%YAML 1.2
> > > > +---
> > > > +$id: http://devicetree.org/schemas/sound/fsl,rpmsg.yaml#
> > > > +$schema: http://devicetree.org/meta-schemas/core.yaml#
> > > > +
> > > > +title: NXP Audio RPMSG CPU DAI Controller
> > > > +
> > > > +maintainers:
> > > > + - Shengjiu Wang <[email protected]>
> > > > +
> > > > +description: |
> > > > + fsl_rpmsg cpu dai driver is virtual driver for rpmsg audio, which doesn't
> > > > + touch hardware. It is mainly used for getting the user's configuration
> > > > + from device tree and configure the clocks which is used by Cortex-M core.
> > > > + So in this document define the needed property.
> > > > +
> > > > +properties:
> > > > + compatible:
> > > > + enum:
> > > > + - fsl,imx7ulp-rpmsg
> > > > + - fsl,imx8mn-rpmsg
> > > > + - fsl,imx8mm-rpmsg
> > > > + - fsl,imx8mp-rpmsg
> > > > +
> > > > + model:
> > > > + $ref: /schemas/types.yaml#/definitions/string
> > > > + description: User specified audio sound card name
> > > > +
> > > > + clocks:
> > > > + items:
> > > > + - description: Peripheral clock for register access
> > > > + - description: Master clock
> > > > + - description: DMA clock for DMA register access
> > > > + - description: Parent clock for multiple of 8kHz sample rates
> > > > + - description: Parent clock for multiple of 11kHz sample rates
> > > > + minItems: 5
> > >
> > > If this doesn't touch hardware, what are these clocks for?
> >
> > When the cortex-M core support audio service, these clock
> > needed prepared & enabled by ALSA driver.
> >
> > >
> > > You don't need 'minItems' unless it's less than the number of 'items'.
> >
> > Ok, I will remove this minItems.
> >
> > >
> > > > +
> > > > + clock-names:
> > > > + items:
> > > > + - const: ipg
> > > > + - const: mclk
> > > > + - const: dma
> > > > + - const: pll8k
> > > > + - const: pll11k
> > > > + minItems: 5
> > > > +
> > > > + power-domains:
> > > > + maxItems: 1
> > > > +
> > > > + fsl,audioindex:
> > > > + $ref: /schemas/types.yaml#/definitions/uint32
> > > > + enum: [0, 1]
> > > > + default: 0
> > > > + description: Instance index for sound card in
> > > > + M core side, which share one rpmsg
> > > > + channel.
> > >
> > > We don't do indexes in DT. What's this numbering tied to?
> >
> > I will remove it. it is not necessary
> >
> > >
> > > > +
> > > > + fsl,version:
> > >
> > > version of what?
> > >
> > > This seems odd at best.
> > >
> >
> > I will remove it. it is not necessary
> >
> > > > + $ref: /schemas/types.yaml#/definitions/uint32
> > > > + enum: [1, 2]
> > >
> > > You're going to update this with every new firmware version?
> > >
> > > > + default: 2
> > > > + description: The version of M core image, which is
> > > > + to make driver compatible with different image.
> > > > +
> > > > + fsl,buffer-size:
> > > > + $ref: /schemas/types.yaml#/definitions/uint32
> > > > + description: pre allocate dma buffer size
> > >
> > > How can you have DMA, this doesn't touch h/w?
> >
> > The DMA is handled by M core image for sound playback
> > and capture. we need to allocated buffer in Linux side.
> > here just make the buffer size to be configurable.
>
> Do we set audio buffer sizes for other audio devices in DT? If not,
> why is this special? If so, why is it not common.
No. I will move it to driver.
>
> > > > + fsl,enable-lpa:
> > > > + $ref: /schemas/types.yaml#/definitions/flag
> > > > + description: enable low power audio path.
> > > > +
> > > > + fsl,rpmsg-out:
> > > > + $ref: /schemas/types.yaml#/definitions/flag
> > > > + description: |
> > > > + This is a boolean property. If present, the transmitting function
> > > > + will be enabled.
> > > > +
> > > > + fsl,rpmsg-in:
> > > > + $ref: /schemas/types.yaml#/definitions/flag
> > > > + description: |
> > > > + This is a boolean property. If present, the receiving function
> > > > + will be enabled.
> > > > +
> > > > + fsl,codec-type:
> > > > + $ref: /schemas/types.yaml#/definitions/uint32
> > > > + enum: [0, 1, 2]
> > > > + default: 0
> > > > + description: Sometimes the codec is registered by
> > > > + driver not by the device tree, this items
> > > > + can be used to distinguish codecs.
> > >
> > > How does one decide what value to use?
> >
> > I will add more description:
> > 0: dummy codec
> > 1: WM8960 codec
> > 2: AK4497 codec
>
> I assume the last 2 cases have nodes in DT (pointed to by
> 'audio-codec'), so this is redundant.
Ok, will remove it.
>
> > > > +
> > > > + audio-codec:
> > > > + $ref: /schemas/types.yaml#/definitions/phandle
> > > > + description: The phandle of the audio codec
> > >
> > > The codec is controlled from the Linux side?
> >
> > yes.
> >
> > >
> > > > +
> > > > + memory-region:
> > > > + $ref: /schemas/types.yaml#/definitions/phandle
> > > > + description: phandle to the reserved memory nodes
> > > > +
> > > > +required:
> > > > + - compatible
> > > > + - fsl,audioindex
> > > > + - fsl,version
> > > > + - fsl,buffer-size
> > > > +
> > > > +additionalProperties: false
> > > > +
> > > > +examples:
> > > > + - |
> > > > + rpmsg_audio: rpmsg_audio {
> > > > + compatible = "fsl,imx8mn-rpmsg";
> > > > + fsl,audioindex = <0> ;
> > > > + fsl,version = <2>;
> > > > + fsl,buffer-size = <0x6000000>;
> > > > + fsl,enable-lpa;
> > >
> > > How does this work? Don't you need somewhere to put the 'rpmsg' data?
> >
> > The rpmsg data is not handled in this "rpmsg_audio" device, it is just to
> > prepare the resource for rpmsg audio function, the clock, the memory
> > the power...
> >
> > The rpmsg data is handled in imx-pcm-rpmsg.c and imx-audio-rpmsg.c
> > These devices is registered by imx remoteproc driver.
>
> Then what is 'memory-region' for? You need that, a mailbox, or ???
> somewhere in DT.
>
The M core can't access all the DDR memory space on some platform,
so use 'memory-region' reserve a specific memory for dma buffer
which M core can access.
best regards
wang shengjiu