2021-02-07 10:38:04

by Shengjiu Wang

[permalink] [raw]
Subject: [PATCH v2 0/7] Add audio driver base on rpmsg on i.MX platform

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 v2:
- update codes and comments according to Mark's comments

Shengjiu Wang (7):
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
ASoC: dt-bindings: imx-rpmsg: Add binding doc for rpmsg machine driver

.../devicetree/bindings/sound/fsl,rpmsg.yaml | 80 ++
.../bindings/sound/imx-audio-rpmsg.yaml | 48 +
include/sound/soc-component.h | 3 +
sound/soc/fsl/Kconfig | 28 +
sound/soc/fsl/Makefile | 6 +
sound/soc/fsl/fsl_rpmsg.c | 252 +++++
sound/soc/fsl/fsl_rpmsg.h | 38 +
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 | 148 +++
sound/soc/soc-component.c | 14 +
sound/soc/soc-pcm.c | 2 +
13 files changed, 2201 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-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


2021-02-07 10:38:08

by Shengjiu Wang

[permalink] [raw]
Subject: [PATCH v2 1/7] ASoC: soc-component: Add snd_soc_pcm_component_ack

Add snd_soc_pcm_component_ack back, which can be used to get updated
buffer pointer in 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 5b47768222b7..2dc8c7e3d1a6 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 159bf88b9f8c..a9fbb2d26412 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 b79f064887d4..605acec48971 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2830,6 +2830,8 @@ 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

2021-02-07 10:38:45

by Shengjiu Wang

[permalink] [raw]
Subject: [PATCH v2 4/7] ASoC: imx-audio-rpmsg: Add rpmsg_driver for audio channel

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 message 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 a688c3c2efbc..84d9f0f1f75b 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

2021-02-07 10:38:57

by Shengjiu Wang

[permalink] [raw]
Subject: [PATCH v2 2/7] ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg

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 | 252 ++++++++++++++++++++++++++++++++++++++
sound/soc/fsl/fsl_rpmsg.h | 38 ++++++
4 files changed, 299 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..a688c3c2efbc 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 "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..6e218344df0d
--- /dev/null
+++ b/sound/soc/fsl/fsl_rpmsg.c
@@ -0,0 +1,252 @@
+// 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;
+ unsigned int rate = params_rate(params);
+ int ret;
+
+ /* 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);
+ }
+ }
+
+ ret = clk_prepare_enable(rpmsg->mclk);
+ if (ret)
+ dev_err(dai->dev, "failed to enable mclk: %d\n", ret);
+
+ 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);
+
+ clk_disable_unprepare(rpmsg->mclk);
+
+ 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;
+
+ 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,
+ .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..5648742def74
--- /dev/null
+++ b/sound/soc/fsl/fsl_rpmsg.h
@@ -0,0 +1,38 @@
+/* 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
+ * @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;
+ int force_lpa;
+ int enable_lpa;
+ int buffer_size;
+ int audioindex;
+ int version;
+};
+#endif /* __FSL_RPMSG_H */
--
2.27.0

2021-02-07 10:39:48

by Shengjiu Wang

[permalink] [raw]
Subject: [PATCH v2 3/7] ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg cpu dai driver

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 | 80 +++++++++++++++++++
1 file changed, 80 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..2d3ce10d42fc
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
@@ -0,0 +1,80 @@
+# 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]>
+
+properties:
+ compatible:
+ enum:
+ - fsl,imx7ulp-rpmsg
+ - fsl,imx8mn-rpmsg
+ - fsl,imx8mm-rpmsg
+ - fsl,imx8mp-rpmsg
+
+ 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
+ description: instance index for rpmsg image
+
+ fsl,version:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: rpmsg image version index
+
+ 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,codec-type:
+ $ref: /schemas/types.yaml#/definitions/uint32
+ description: Sometimes the codec is registered by
+ driver not the device tree, this items
+ can be used to distinguish codecs
+
+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;
+ status = "okay";
+ };
--
2.27.0

2021-02-07 10:39:52

by Shengjiu Wang

[permalink] [raw]
Subject: [PATCH v2 7/7] ASoC: dt-bindings: imx-rpmsg: Add binding doc for rpmsg machine driver

Imx-rpmsg is a new added machine driver for supporting audio on Cortex-M
core. The Cortex-M core will control the audio interface, DMA and audio
codec, setup the pipeline, the audio driver on Cortex-A core side is just
to communitcate with M core, it is a virtual sound card and don't touch
the hardware.

Signed-off-by: Shengjiu Wang <[email protected]>
---
.../bindings/sound/imx-audio-rpmsg.yaml | 48 +++++++++++++++++++
1 file changed, 48 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml

diff --git a/Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml b/Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml
new file mode 100644
index 000000000000..b941aeb80678
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml
@@ -0,0 +1,48 @@
+# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/imx-audio-rpmsg.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: NXP i.MX audio complex with rpmsg
+
+maintainers:
+ - Shengjiu Wang <[email protected]>
+
+properties:
+ compatible:
+ enum:
+ - fsl,imx-audio-rpmsg
+
+ model:
+ $ref: /schemas/types.yaml#/definitions/string
+ description: User specified audio sound card name
+
+ audio-cpu:
+ description: The phandle of an CPU DAI controller
+
+ rpmsg-out:
+ description: |
+ This is a boolean property. If present, the transmitting function
+ will be enabled,
+
+ rpmsg-in:
+ description: |
+ This is a boolean property. If present, the receiving function
+ will be enabled.
+
+required:
+ - compatible
+ - model
+ - audio-cpu
+
+additionalProperties: false
+
+examples:
+ - |
+ sound-rpmsg {
+ compatible = "fsl,imx-audio-rpmsg";
+ model = "ak4497-audio";
+ audio-cpu = <&rpmsg_audio>;
+ rpmsg-out;
+ };
--
2.27.0

2021-02-07 10:40:06

by Shengjiu Wang

[permalink] [raw]
Subject: [PATCH v2 5/7] ASoC: imx-pcm-rpmsg: Add platform driver for audio base on rpmsg

platform driver base 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
wakeup.

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 84d9f0f1f75b..749c44fc0759 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

2021-02-07 10:41:09

by Shengjiu Wang

[permalink] [raw]
Subject: [PATCH v2 6/7] ASoC: imx-rpmsg: Add machine driver for audio base on rpmsg

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 | 148 ++++++++++++++++++++++++++++++++++++++
3 files changed, 162 insertions(+)
create mode 100644 sound/soc/fsl/imx-rpmsg.c

diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 749c44fc0759..3557866d3fa2 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..a87dcbce4f36
--- /dev/null
+++ b/sound/soc/fsl/imx-rpmsg.c
@@ -0,0 +1,148 @@
+// 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 platform_device *cpu_pdev;
+ struct of_phandle_args args;
+ struct device_node *cpu_np;
+ struct imx_rpmsg *data;
+ int ret;
+
+ dlc = devm_kzalloc(&pdev->dev, 3 * sizeof(*dlc), GFP_KERNEL);
+ if (!dlc)
+ return -ENOMEM;
+
+ cpu_np = of_parse_phandle(pdev->dev.of_node, "audio-cpu", 0);
+ if (!cpu_np) {
+ dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+ if (!data) {
+ ret = -ENOMEM;
+ goto fail;
+ }
+
+ cpu_pdev = of_find_device_by_node(cpu_np);
+ if (!cpu_pdev) {
+ dev_err(&pdev->dev, "failed to find rpmsg platform device\n");
+ ret = -EINVAL;
+ goto fail;
+ }
+
+ ret = of_reserved_mem_device_init_by_idx(&pdev->dev, pdev->dev.of_node, 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(pdev->dev.of_node,
+ "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(&cpu_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(pdev->dev.of_node, "rpmsg-out"))
+ data->dai.capture_only = false;
+
+ if (of_property_read_bool(pdev->dev.of_node, "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 = snd_soc_of_parse_card_name(&data->card, "model");
+ 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:
+ if (cpu_np)
+ of_node_put(cpu_np);
+ return ret;
+}
+
+static const struct of_device_id imx_rpmsg_dt_ids[] = {
+ { .compatible = "fsl,imx-audio-rpmsg", },
+ { /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_rpmsg_dt_ids);
+
+static struct platform_driver imx_rpmsg_driver = {
+ .driver = {
+ .name = "imx-audio-rpmsg",
+ .owner = THIS_MODULE,
+ .pm = &snd_soc_pm_ops,
+ .of_match_table = imx_rpmsg_dt_ids,
+ },
+ .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-rpmsg");
+MODULE_LICENSE("GPL v2");
--
2.27.0

2021-02-08 12:15:48

by Mark Brown

[permalink] [raw]
Subject: Re: [PATCH v2 2/7] ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg

On Sun, Feb 07, 2021 at 06:23:50PM +0800, Shengjiu Wang wrote:

> +static int fsl_rpmsg_hw_params(struct snd_pcm_substream *substream,
> + struct snd_pcm_hw_params *params,
> + struct snd_soc_dai *dai)
> +{

...

> + ret = clk_prepare_enable(rpmsg->mclk);
> + if (ret)
> + dev_err(dai->dev, "failed to enable mclk: %d\n", ret);
> +
> + 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);
> +
> + clk_disable_unprepare(rpmsg->mclk);

hw_params() can be called multiple times and there's no need for it to
be balanced with hw_free(), I'd move this to a different callback (DAPM
should work well).


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2021-02-09 09:23:43

by Shengjiu Wang

[permalink] [raw]
Subject: Re: [PATCH v2 2/7] ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg

On Mon, Feb 8, 2021 at 7:53 PM Mark Brown <[email protected]> wrote:
>
> On Sun, Feb 07, 2021 at 06:23:50PM +0800, Shengjiu Wang wrote:
>
> > +static int fsl_rpmsg_hw_params(struct snd_pcm_substream *substream,
> > + struct snd_pcm_hw_params *params,
> > + struct snd_soc_dai *dai)
> > +{
>
> ...
>
> > + ret = clk_prepare_enable(rpmsg->mclk);
> > + if (ret)
> > + dev_err(dai->dev, "failed to enable mclk: %d\n", ret);
> > +
> > + 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);
> > +
> > + clk_disable_unprepare(rpmsg->mclk);
>
> hw_params() can be called multiple times and there's no need for it to
> be balanced with hw_free(), I'd move this to a different callback (DAPM
> should work well).

Which callback should I use? Is there an example?

best regards
wang shengjiu

2021-02-10 01:11:20

by Mark Brown

[permalink] [raw]
Subject: Re: [PATCH v2 2/7] ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg

On Tue, Feb 09, 2021 at 05:16:16PM +0800, Shengjiu Wang wrote:
> On Mon, Feb 8, 2021 at 7:53 PM Mark Brown <[email protected]> wrote:

> > hw_params() can be called multiple times and there's no need for it to
> > be balanced with hw_free(), I'd move this to a different callback (DAPM
> > should work well).

> Which callback should I use? Is there an example?

Like I say I'd actually recommend moving this control to DAPM.


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2021-02-10 08:36:51

by Shengjiu Wang

[permalink] [raw]
Subject: Re: [PATCH v2 2/7] ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg

On Wed, Feb 10, 2021 at 6:30 AM Mark Brown <[email protected]> wrote:
>
> On Tue, Feb 09, 2021 at 05:16:16PM +0800, Shengjiu Wang wrote:
> > On Mon, Feb 8, 2021 at 7:53 PM Mark Brown <[email protected]> wrote:
>
> > > hw_params() can be called multiple times and there's no need for it to
> > > be balanced with hw_free(), I'd move this to a different callback (DAPM
> > > should work well).
>
> > Which callback should I use? Is there an example?
>
> Like I say I'd actually recommend moving this control to DAPM.

I may understand your point, you suggest to use the .set_bias_level
interface. But in my case I need to enable the clock in earlier stage
and keep the clock on when system go to suspend.

I am not sure .set_bias_level can met my requirement. we start
the Chinese new year holiday now, so currently I can't do test for this
recommendation.

Maybe we can keep current implementation, can we?
Later after I do the test, I can submit another patch for it.

Best regards
Wang Shengjiu

2021-02-10 15:43:20

by Mark Brown

[permalink] [raw]
Subject: Re: [PATCH v2 2/7] ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg

On Wed, Feb 10, 2021 at 02:35:29PM +0800, Shengjiu Wang wrote:
> On Wed, Feb 10, 2021 at 6:30 AM Mark Brown <[email protected]> wrote:

> > Like I say I'd actually recommend moving this control to DAPM.

> I may understand your point, you suggest to use the .set_bias_level
> interface. But in my case I need to enable the clock in earlier stage
> and keep the clock on when system go to suspend.

The device can be kept alive over system suspend if that's needed, or
possibly it sounds like runtime PM is a better fit? There's callbacks
in the core to keep the device runtime PM enabled while it's open which
is probably about the time range you're looking for.

> I am not sure .set_bias_level can met my requirement. we start
> the Chinese new year holiday now, so currently I can't do test for this
> recommendation.


> Maybe we can keep current implementation, can we?
> Later after I do the test, I can submit another patch for it.

Well, the current version is clearly going to leak clock enables with
valid userspace so


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2021-02-10 22:16:54

by Rob Herring

[permalink] [raw]
Subject: Re: [PATCH v2 3/7] ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg cpu dai driver

On Sun, Feb 07, 2021 at 06:23:51PM +0800, Shengjiu Wang wrote:
> 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 | 80 +++++++++++++++++++
> 1 file changed, 80 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..2d3ce10d42fc
> --- /dev/null
> +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> @@ -0,0 +1,80 @@
> +# 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]>
> +
> +properties:
> + compatible:
> + enum:
> + - fsl,imx7ulp-rpmsg
> + - fsl,imx8mn-rpmsg
> + - fsl,imx8mm-rpmsg
> + - fsl,imx8mp-rpmsg

rpmsg is a protocol. What's the h/w block?

> +
> + 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
> + description: instance index for rpmsg image
> +
> + fsl,version:
> + $ref: /schemas/types.yaml#/definitions/uint32
> + description: rpmsg image version index

What are these 2 used for?

> +
> + 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,codec-type:
> + $ref: /schemas/types.yaml#/definitions/uint32
> + description: Sometimes the codec is registered by
> + driver not the device tree, this items
> + can be used to distinguish codecs

0-2^32 are valid values?

> +
> +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;
> + status = "okay";

Don't show status in examples.

> + };
> --
> 2.27.0
>

2021-02-10 22:19:33

by Rob Herring

[permalink] [raw]
Subject: Re: [PATCH v2 7/7] ASoC: dt-bindings: imx-rpmsg: Add binding doc for rpmsg machine driver

On Sun, Feb 07, 2021 at 06:23:55PM +0800, Shengjiu Wang wrote:
> Imx-rpmsg is a new added machine driver for supporting audio on Cortex-M
> core. The Cortex-M core will control the audio interface, DMA and audio
> codec, setup the pipeline, the audio driver on Cortex-A core side is just
> to communitcate with M core, it is a virtual sound card and don't touch
> the hardware.

I don't understand why there are 2 nodes for this other than you happen
to want to split this into 2 Linux drivers. It's 1 h/w thing.

>
> Signed-off-by: Shengjiu Wang <[email protected]>
> ---
> .../bindings/sound/imx-audio-rpmsg.yaml | 48 +++++++++++++++++++
> 1 file changed, 48 insertions(+)
> create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml
>
> diff --git a/Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml b/Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml
> new file mode 100644
> index 000000000000..b941aeb80678
> --- /dev/null
> +++ b/Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml
> @@ -0,0 +1,48 @@
> +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
> +%YAML 1.2
> +---
> +$id: http://devicetree.org/schemas/sound/imx-audio-rpmsg.yaml#
> +$schema: http://devicetree.org/meta-schemas/core.yaml#
> +
> +title: NXP i.MX audio complex with rpmsg
> +
> +maintainers:
> + - Shengjiu Wang <[email protected]>
> +
> +properties:
> + compatible:
> + enum:
> + - fsl,imx-audio-rpmsg
> +
> + model:
> + $ref: /schemas/types.yaml#/definitions/string
> + description: User specified audio sound card name
> +
> + audio-cpu:
> + description: The phandle of an CPU DAI controller
> +
> + rpmsg-out:
> + description: |
> + This is a boolean property. If present, the transmitting function
> + will be enabled,
> +
> + rpmsg-in:
> + description: |
> + This is a boolean property. If present, the receiving function
> + will be enabled.
> +
> +required:
> + - compatible
> + - model
> + - audio-cpu
> +
> +additionalProperties: false
> +
> +examples:
> + - |
> + sound-rpmsg {
> + compatible = "fsl,imx-audio-rpmsg";
> + model = "ak4497-audio";
> + audio-cpu = <&rpmsg_audio>;
> + rpmsg-out;
> + };
> --
> 2.27.0
>

2021-02-18 08:28:28

by Shengjiu Wang

[permalink] [raw]
Subject: Re: [PATCH v2 7/7] ASoC: dt-bindings: imx-rpmsg: Add binding doc for rpmsg machine driver

On Thu, Feb 11, 2021 at 6:18 AM Rob Herring <[email protected]> wrote:
>
> On Sun, Feb 07, 2021 at 06:23:55PM +0800, Shengjiu Wang wrote:
> > Imx-rpmsg is a new added machine driver for supporting audio on Cortex-M
> > core. The Cortex-M core will control the audio interface, DMA and audio
> > codec, setup the pipeline, the audio driver on Cortex-A core side is just
> > to communitcate with M core, it is a virtual sound card and don't touch
> > the hardware.
>
> I don't understand why there are 2 nodes for this other than you happen
> to want to split this into 2 Linux drivers. It's 1 h/w thing.

This one is for the sound card machine driver. Another one is
for the sound card cpu dai driver. so there are 2 nodes.

>
> >
> > Signed-off-by: Shengjiu Wang <[email protected]>
> > ---
> > .../bindings/sound/imx-audio-rpmsg.yaml | 48 +++++++++++++++++++
> > 1 file changed, 48 insertions(+)
> > create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml
> >
> > diff --git a/Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml b/Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml
> > new file mode 100644
> > index 000000000000..b941aeb80678
> > --- /dev/null
> > +++ b/Documentation/devicetree/bindings/sound/imx-audio-rpmsg.yaml
> > @@ -0,0 +1,48 @@
> > +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause)
> > +%YAML 1.2
> > +---
> > +$id: http://devicetree.org/schemas/sound/imx-audio-rpmsg.yaml#
> > +$schema: http://devicetree.org/meta-schemas/core.yaml#
> > +
> > +title: NXP i.MX audio complex with rpmsg
> > +
> > +maintainers:
> > + - Shengjiu Wang <[email protected]>
> > +
> > +properties:
> > + compatible:
> > + enum:
> > + - fsl,imx-audio-rpmsg
> > +
> > + model:
> > + $ref: /schemas/types.yaml#/definitions/string
> > + description: User specified audio sound card name
> > +
> > + audio-cpu:
> > + description: The phandle of an CPU DAI controller
> > +
> > + rpmsg-out:
> > + description: |
> > + This is a boolean property. If present, the transmitting function
> > + will be enabled,
> > +
> > + rpmsg-in:
> > + description: |
> > + This is a boolean property. If present, the receiving function
> > + will be enabled.
> > +
> > +required:
> > + - compatible
> > + - model
> > + - audio-cpu
> > +
> > +additionalProperties: false
> > +
> > +examples:
> > + - |
> > + sound-rpmsg {
> > + compatible = "fsl,imx-audio-rpmsg";
> > + model = "ak4497-audio";
> > + audio-cpu = <&rpmsg_audio>;
> > + rpmsg-out;
> > + };
> > --
> > 2.27.0
> >

2021-02-18 08:28:36

by Shengjiu Wang

[permalink] [raw]
Subject: Re: [PATCH v2 3/7] ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg cpu dai driver

On Thu, Feb 11, 2021 at 6:13 AM Rob Herring <[email protected]> wrote:
>
> On Sun, Feb 07, 2021 at 06:23:51PM +0800, Shengjiu Wang wrote:
> > 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 | 80 +++++++++++++++++++
> > 1 file changed, 80 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..2d3ce10d42fc
> > --- /dev/null
> > +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml
> > @@ -0,0 +1,80 @@
> > +# 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]>
> > +
> > +properties:
> > + compatible:
> > + enum:
> > + - fsl,imx7ulp-rpmsg
> > + - fsl,imx8mn-rpmsg
> > + - fsl,imx8mm-rpmsg
> > + - fsl,imx8mp-rpmsg
>
> rpmsg is a protocol. What's the h/w block?

On Linux side this driver is a virtual driver, it is running
on Arm Cortex-A core. The h/w block is controlled by
another core (cortex-M core). so this driver actually
doesn't touch any hardware, it just does configuration
for rpmsg channel.

>
> > +
> > + 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
> > + description: instance index for rpmsg image
> > +
> > + fsl,version:
> > + $ref: /schemas/types.yaml#/definitions/uint32
> > + description: rpmsg image version index
>
> What are these 2 used for?

fsl,audioindex: As we may support multiple instance, for example
two sound card with one rpmsg channel, this is the instance index.

fsl,version: There are maybe different image running on M core, this
is the image version, different image has different function.


>
> > +
> > + 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,codec-type:
> > + $ref: /schemas/types.yaml#/definitions/uint32
> > + description: Sometimes the codec is registered by
> > + driver not the device tree, this items
> > + can be used to distinguish codecs
>
> 0-2^32 are valid values?

I should add range for it.

>
> > +
> > +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;
> > + status = "okay";
>
> Don't show status in examples.

ok, will remove it.

>
> > + };
> > --
> > 2.27.0
> >

2021-02-18 09:23:35

by Shengjiu Wang

[permalink] [raw]
Subject: Re: [PATCH v2 2/7] ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg

On Wed, Feb 10, 2021 at 11:39 PM Mark Brown <[email protected]> wrote:
>
> On Wed, Feb 10, 2021 at 02:35:29PM +0800, Shengjiu Wang wrote:
> > On Wed, Feb 10, 2021 at 6:30 AM Mark Brown <[email protected]> wrote:
>
> > > Like I say I'd actually recommend moving this control to DAPM.
>
> > I may understand your point, you suggest to use the .set_bias_level
> > interface. But in my case I need to enable the clock in earlier stage
> > and keep the clock on when system go to suspend.
>
> The device can be kept alive over system suspend if that's needed, or
> possibly it sounds like runtime PM is a better fit? There's callbacks
> in the core to keep the device runtime PM enabled while it's open which
> is probably about the time range you're looking for.

Before enabling the clock, I need to reparent the clock according to
the sample rate, Maybe the hw_params is the right place to do
these things.

Can I add a flag:
"rpmsg->mclk_streams & BIT(substream->stream)"
for avoiding multiple calls of hw_params function before enabling
clock?

2021-02-22 14:20:34

by Mark Brown

[permalink] [raw]
Subject: Re: [PATCH v2 2/7] ASoC: fsl_rpmsg: Add CPU DAI driver for audio base on rpmsg

On Thu, Feb 18, 2021 at 03:57:11PM +0800, Shengjiu Wang wrote:

> Can I add a flag:
> "rpmsg->mclk_streams & BIT(substream->stream)"
> for avoiding multiple calls of hw_params function before enabling
> clock?

Yes, if you do local refcounting that'd avoid the issue.


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2021-02-25 06:11:51

by Fabio Estevam

[permalink] [raw]
Subject: Re: [PATCH v2 3/7] ASoC: dt-bindings: fsl_rpmsg: Add binding doc for rpmsg cpu dai driver

On Thu, Feb 18, 2021 at 4:21 AM Shengjiu Wang <[email protected]> wrote:

> > rpmsg is a protocol. What's the h/w block?
>
> On Linux side this driver is a virtual driver, it is running
> on Arm Cortex-A core. The h/w block is controlled by
> another core (cortex-M core). so this driver actually
> doesn't touch any hardware, it just does configuration
> for rpmsg channel.
> fsl,version: There are maybe different image running on M core, this
> is the image version, different image has different function.

To answer Rob's question: the hardware block that handles these
messages is the Message Unit block.

2021-03-04 20:07:31

by Rob Herring

[permalink] [raw]
Subject: Re: [PATCH v2 7/7] ASoC: dt-bindings: imx-rpmsg: Add binding doc for rpmsg machine driver

On Thu, Feb 18, 2021 at 1:23 AM Shengjiu Wang <[email protected]> wrote:
>
> On Thu, Feb 11, 2021 at 6:18 AM Rob Herring <[email protected]> wrote:
> >
> > On Sun, Feb 07, 2021 at 06:23:55PM +0800, Shengjiu Wang wrote:
> > > Imx-rpmsg is a new added machine driver for supporting audio on Cortex-M
> > > core. The Cortex-M core will control the audio interface, DMA and audio
> > > codec, setup the pipeline, the audio driver on Cortex-A core side is just
> > > to communitcate with M core, it is a virtual sound card and don't touch
> > > the hardware.
> >
> > I don't understand why there are 2 nodes for this other than you happen
> > to want to split this into 2 Linux drivers. It's 1 h/w thing.
>
> This one is for the sound card machine driver. Another one is
> for the sound card cpu dai driver. so there are 2 nodes.

You are explaining this to me in terms of drivers. Explain it in terms
of h/w blocks.

Rob

2021-03-05 02:56:55

by Shengjiu Wang

[permalink] [raw]
Subject: Re: [PATCH v2 7/7] ASoC: dt-bindings: imx-rpmsg: Add binding doc for rpmsg machine driver

Hi

On Fri, Mar 5, 2021 at 4:05 AM Rob Herring <[email protected]> wrote:
>
> On Thu, Feb 18, 2021 at 1:23 AM Shengjiu Wang <[email protected]> wrote:
> >
> > On Thu, Feb 11, 2021 at 6:18 AM Rob Herring <[email protected]> wrote:
> > >
> > > On Sun, Feb 07, 2021 at 06:23:55PM +0800, Shengjiu Wang wrote:
> > > > Imx-rpmsg is a new added machine driver for supporting audio on Cortex-M
> > > > core. The Cortex-M core will control the audio interface, DMA and audio
> > > > codec, setup the pipeline, the audio driver on Cortex-A core side is just
> > > > to communitcate with M core, it is a virtual sound card and don't touch
> > > > the hardware.
> > >
> > > I don't understand why there are 2 nodes for this other than you happen
> > > to want to split this into 2 Linux drivers. It's 1 h/w thing.
> >
> > This one is for the sound card machine driver. Another one is
> > for the sound card cpu dai driver. so there are 2 nodes.
>
> You are explaining this to me in terms of drivers. Explain it in terms
> of h/w blocks.
>

Yes, there is only 1 h/w block, which is (MU) message unit

As the sound card needs a cpu dai node and sound card node,
so from the driver's perspective, I use two nodes.

Seems It is hard to only use one node for my case.
or do you have any suggestions?

Best regards
Wang shengjiu