* [PATCH v4 01/24] soc: qcom dt-bindings: Add APR bus bindings
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 02/24] soc: qcom: Add APR bus driver srinivas.kandagatla
` (22 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch add dt bindings for Qualcomm APR (Asynchronous Packet Router)
bus driver. This bus is used for communicating with DSP which provides
audio and various other services to cpu.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
.../devicetree/bindings/soc/qcom/qcom,apr.txt | 85 ++++++++++++++++++++++
include/dt-bindings/soc/qcom,apr.h | 27 +++++++
2 files changed, 112 insertions(+)
create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt
create mode 100644 include/dt-bindings/soc/qcom,apr.h
diff --git a/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt b/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt
new file mode 100644
index 000000000000..6071a9038b4b
--- /dev/null
+++ b/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt
@@ -0,0 +1,85 @@
+Qualcomm APR (Asynchronous Packet Router) binding
+
+This binding describes the Qualcomm APR. APR is a IPC protocol for
+communication between Application processor and QDSP. APR is mainly
+used for audio/voice services on the QDSP.
+
+- compatible:
+ Usage: required
+ Value type: <stringlist>
+ Definition: must be "qcom,apr-v<VERSION-NUMBER>", example "qcom,apr-v2"
+
+- qcom,apr-dest-domain-id
+ Usage: required
+ Value type: <prop-encoded-array>
+ Definition: Destination processor ID.
+ Possible values are :
+ 1 - APR simulator
+ 2 - PC
+ 3 - MODEM
+ 4 - ADSP
+ 5 - APPS
+ 6 - MODEM2
+ 7 - APPS2
+
+= APR SERVICES
+Each subnode of the APR node represents service tied to this apr. The name
+of the nodes are not important. The properties of these nodes are defined
+by the individual bindings for the specific service
+- All APR services MUST contain the following property:
+
+- qcom,apr-svc-id
+ Usage: required
+ Value type: <prop-encoded-array>
+ Definition: APR Service ID, used for matching the service.
+ Possible values are :
+ 3 - DSP Core Service
+ 4 - Audio Front End Service.
+ 5 - Voice Stream Manager Service.
+ 6 - Voice processing manager.
+ 7 - Audio Stream Manager Service.
+ 8 - Audio Device Manager Service.
+ 9 - Multimode voice manager.
+ 10 - Core voice stream.
+ 11 - Core voice processor.
+ 12 - Ultrasound stream manager.
+ 13 - Listen stream manager.
+
+- qcom,apr-svc-name
+ Usage: required
+ Value type: <stringlist>
+ Definition: User readable name of a APR service.
+
+= EXAMPLE
+The following example represents a QDSP based sound card on a MSM8996 device
+which uses apr as communication between Apps and QDSP.
+
+ apr {
+ compatible = "qcom,apr-v2";
+ qcom,smd-channels = "apr_audio_svc";
+ qcom,apr-dest-domain-id = <APR_DOMAIN_ADSP>;
+
+ q6core {
+ compatible = "qcom,q6core";
+ qcom,apr-svc-name = "CORE";
+ qcom,apr-svc-id = <APR_SVC_ADSP_CORE>;
+ };
+
+ q6afe {
+ compatible = "qcom,q6afe";
+ qcom,apr-svc-name = "AFE";
+ qcom,apr-svc-id = <APR_SVC_AFE>;
+ };
+
+ q6asm {
+ compatible = "qcom,q6asm";
+ qcom,apr-svc-name = "ASM";
+ qcom,apr-svc-id = <APR_SVC_ASM>;
+ };
+
+ q6adm {
+ compatible = "qcom,q6adm";
+ qcom,apr-svc-name = "ADM";
+ qcom,apr-svc-id = <APR_SVC_ADM>;
+ };
+ };
diff --git a/include/dt-bindings/soc/qcom,apr.h b/include/dt-bindings/soc/qcom,apr.h
new file mode 100644
index 000000000000..905503f81885
--- /dev/null
+++ b/include/dt-bindings/soc/qcom,apr.h
@@ -0,0 +1,27 @@
+#ifndef __DT_BINDINGS_QCOM_APR_H
+#define __DT_BINDINGS_QCOM_APR_H
+
+/* Domain IDs */
+#define APR_DOMAIN_SIM 0x1
+#define APR_DOMAIN_PC 0x2
+#define APR_DOMAIN_MODEM 0x3
+#define APR_DOMAIN_ADSP 0x4
+#define APR_DOMAIN_APPS 0x5
+#define APR_DOMAIN_MAX 0x6
+
+/* ADSP service IDs */
+#define APR_SVC_ADSP_CORE 0x3
+#define APR_SVC_AFE 0x4
+#define APR_SVC_VSM 0x5
+#define APR_SVC_VPM 0x6
+#define APR_SVC_ASM 0x7
+#define APR_SVC_ADM 0x8
+#define APR_SVC_ADSP_MVM 0x09
+#define APR_SVC_ADSP_CVS 0x0A
+#define APR_SVC_ADSP_CVP 0x0B
+#define APR_SVC_USM 0x0C
+#define APR_SVC_LSM 0x0D
+#define APR_SVC_VIDC 0x16
+#define APR_SVC_MAX 0x17
+
+#endif /* __DT_BINDINGS_QCOM_APR_H */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 02/24] soc: qcom: Add APR bus driver
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 01/24] soc: qcom dt-bindings: Add APR bus bindings srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-14 14:12 ` [alsa-devel] " Rohit Kumar
2018-03-10 2:24 ` [PATCH v4 03/24] ASoC: qdsp6: q6common: Add qdsp6 helper functions srinivas.kandagatla
` (21 subsequent siblings)
23 siblings, 1 reply; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support toi APR bus (Asynchronous Packet Router) driver.
ARP driver is made as a bus driver so that the apr devices can added removed
more dynamically depending on the state of the services on the dsp.
APR is used for communication between application processor and QDSP to
use services on QDSP like Audio and others.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
drivers/soc/qcom/Kconfig | 9 +
drivers/soc/qcom/Makefile | 1 +
drivers/soc/qcom/apr.c | 379 ++++++++++++++++++++++++++++++++++++++++
include/linux/mod_devicetable.h | 11 ++
include/linux/soc/qcom/apr.h | 131 ++++++++++++++
5 files changed, 531 insertions(+)
create mode 100644 drivers/soc/qcom/apr.c
create mode 100644 include/linux/soc/qcom/apr.h
diff --git a/drivers/soc/qcom/Kconfig b/drivers/soc/qcom/Kconfig
index 7a729eee13a0..26a8332276e6 100644
--- a/drivers/soc/qcom/Kconfig
+++ b/drivers/soc/qcom/Kconfig
@@ -115,4 +115,13 @@ config QCOM_WCNSS_CTRL
Client driver for the WCNSS_CTRL SMD channel, used to download nv
firmware to a newly booted WCNSS chip.
+config QCOM_APR
+ tristate "Qualcomm APR Bus (Asynchronous Packet Router)"
+ depends on ARCH_QCOM
+ depends on RPMSG
+ help
+ Enable APR IPC protocol support between
+ application processor and QDSP6. APR is
+ used by audio driver to configure QDSP6
+ ASM, ADM and AFE modules.
endmenu
diff --git a/drivers/soc/qcom/Makefile b/drivers/soc/qcom/Makefile
index 7620b1416ee1..043da96e3b79 100644
--- a/drivers/soc/qcom/Makefile
+++ b/drivers/soc/qcom/Makefile
@@ -14,3 +14,4 @@ obj-$(CONFIG_QCOM_SMP2P) += smp2p.o
obj-$(CONFIG_QCOM_SMSM) += smsm.o
obj-$(CONFIG_QCOM_WCNSS_CTRL) += wcnss_ctrl.o
obj-$(CONFIG_ARCH_MSM8996) += kryo-l2-accessors.o
+obj-$(CONFIG_QCOM_APR) += apr.o
diff --git a/drivers/soc/qcom/apr.c b/drivers/soc/qcom/apr.c
new file mode 100644
index 000000000000..56ed3fb1f2f9
--- /dev/null
+++ b/drivers/soc/qcom/apr.c
@@ -0,0 +1,379 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/spinlock.h>
+#include <linux/list.h>
+#include <linux/slab.h>
+#include <linux/of_device.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/rpmsg.h>
+#include <linux/of.h>
+
+struct apr {
+ struct rpmsg_endpoint *ch;
+ struct device *dev;
+ spinlock_t svcs_lock;
+ struct list_head svcs;
+ int dest_domain_id;
+};
+
+/**
+ * apr_send_pkt() - Send a apr message from apr device
+ *
+ * @adev: Pointer to previously registered apr device.
+ * @buf: Pointer to buffer to send
+ *
+ * Return: Will be an negative on packet size on success.
+ */
+int apr_send_pkt(struct apr_device *adev, void *buf)
+{
+ struct apr *apr = dev_get_drvdata(adev->dev.parent);
+ struct apr_hdr *hdr;
+ unsigned long flags;
+ int ret;
+
+ spin_lock_irqsave(&adev->lock, flags);
+
+ hdr = (struct apr_hdr *)buf;
+ hdr->src_domain = APR_DOMAIN_APPS;
+ hdr->src_svc = adev->svc_id;
+ hdr->dest_domain = adev->domain_id;
+ hdr->dest_svc = adev->svc_id;
+
+ ret = rpmsg_send(apr->ch, buf, hdr->pkt_size);
+ if (ret) {
+ dev_err(&adev->dev, "Unable to send APR pkt %d\n",
+ hdr->pkt_size);
+ } else {
+ ret = hdr->pkt_size;
+ }
+
+ spin_unlock_irqrestore(&adev->lock, flags);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(apr_send_pkt);
+
+static void apr_dev_release(struct device *dev)
+{
+ struct apr_device *adev = to_apr_device(dev);
+
+ kfree(adev);
+}
+
+static int apr_callback(struct rpmsg_device *rpdev, void *buf,
+ int len, void *priv, u32 addr)
+{
+ struct apr *apr = dev_get_drvdata(&rpdev->dev);
+ struct apr_client_message data;
+ struct apr_device *p, *c_svc = NULL;
+ struct apr_driver *adrv = NULL;
+ struct apr_hdr *hdr;
+ uint16_t hdr_size;
+ uint16_t msg_type;
+ uint16_t ver;
+ uint16_t svc;
+
+ if (len <= APR_HDR_SIZE) {
+ dev_err(apr->dev, "APR: Improper apr pkt received:%p %d\n",
+ buf, len);
+ return -EINVAL;
+ }
+
+ hdr = buf;
+ ver = APR_HDR_FIELD_VER(hdr->hdr_field);
+ if (ver > APR_PKT_VER + 1)
+ return -EINVAL;
+
+ hdr_size = APR_HDR_FIELD_SIZE_BYTES(hdr->hdr_field);
+ if (hdr_size < APR_HDR_SIZE) {
+ dev_err(apr->dev, "APR: Wrong hdr size:%d\n", hdr_size);
+ return -EINVAL;
+ }
+
+ if (hdr->pkt_size < APR_HDR_SIZE) {
+ dev_err(apr->dev, "APR: Wrong paket size\n");
+ return -EINVAL;
+ }
+
+ msg_type = APR_HDR_FIELD_MT(hdr->hdr_field);
+ if (msg_type >= APR_MSG_TYPE_MAX && msg_type != APR_BASIC_RSP_RESULT) {
+ dev_err(apr->dev, "APR: Wrong message type: %d\n", msg_type);
+ return -EINVAL;
+ }
+
+ if (hdr->src_domain >= APR_DOMAIN_MAX ||
+ hdr->dest_domain >= APR_DOMAIN_MAX ||
+ hdr->src_svc >= APR_SVC_MAX ||
+ hdr->dest_svc >= APR_SVC_MAX) {
+ dev_err(apr->dev, "APR: Wrong APR header\n");
+ return -EINVAL;
+ }
+
+ svc = hdr->dest_svc;
+ spin_lock(&apr->svcs_lock);
+ list_for_each_entry(p, &apr->svcs, node) {
+ if (svc == p->svc_id) {
+ c_svc = p;
+ if (c_svc->dev.driver)
+ adrv = to_apr_driver(c_svc->dev.driver);
+ break;
+ }
+ }
+ spin_unlock(&apr->svcs_lock);
+
+ if (!adrv) {
+ dev_err(apr->dev, "APR: service is not registered\n");
+ return -EINVAL;
+ }
+
+ data.payload_size = hdr->pkt_size - hdr_size;
+ data.opcode = hdr->opcode;
+ data.src_port = hdr->src_port;
+ data.dest_port = hdr->dest_port;
+ data.token = hdr->token;
+ data.msg_type = msg_type;
+
+ if (data.payload_size > 0)
+ data.payload = buf + hdr_size;
+
+ adrv->callback(c_svc, &data);
+
+ return 0;
+}
+
+static int apr_device_match(struct device *dev, struct device_driver *drv)
+{
+ struct apr_device *adev = to_apr_device(dev);
+ struct apr_driver *adrv = to_apr_driver(drv);
+ const struct apr_device_id *id = adrv->id_table;
+
+ /* Attempt an OF style match first */
+ if (of_driver_match_device(dev, drv))
+ return 1;
+
+ if (!id)
+ return 0;
+
+ while (id->domain_id != 0 || id->svc_id != 0) {
+ if (id->domain_id == adev->domain_id &&
+ id->svc_id == adev->svc_id)
+ return 1;
+ id++;
+ }
+
+ return 0;
+}
+
+static int apr_device_probe(struct device *dev)
+{
+ struct apr_device *adev = to_apr_device(dev);
+ struct apr_driver *adrv = to_apr_driver(dev->driver);
+
+ return adrv->probe(adev);
+}
+
+static int apr_device_remove(struct device *dev)
+{
+ struct apr_device *adev = to_apr_device(dev);
+ struct apr_driver *adrv;
+ struct apr *apr = dev_get_drvdata(adev->dev.parent);
+
+ if (dev->driver) {
+ adrv = to_apr_driver(dev->driver);
+ if (adrv->remove)
+ adrv->remove(adev);
+ spin_lock(&apr->svcs_lock);
+ list_del(&adev->node);
+ spin_unlock(&apr->svcs_lock);
+ }
+
+ return 0;
+}
+
+struct bus_type aprbus_type = {
+ .name = "aprbus",
+ .match = apr_device_match,
+ .probe = apr_device_probe,
+ .remove = apr_device_remove,
+ .force_dma = true,
+};
+EXPORT_SYMBOL_GPL(aprbus_type);
+
+static int apr_add_device(struct device *dev, struct device_node *np,
+ const struct apr_device_id *id, bool is_svc)
+{
+ struct apr *apr = dev_get_drvdata(dev);
+ struct apr_device *adev = NULL;
+
+ adev = kzalloc(sizeof(*adev), GFP_KERNEL);
+ if (!adev)
+ return -ENOMEM;
+
+ spin_lock_init(&adev->lock);
+
+ if (is_svc) {
+ adev->svc_id = id->svc_id;
+ adev->domain_id = id->domain_id;
+ adev->version = id->svc_version;
+ dev_set_name(&adev->dev, "aprsvc:%s:%x:%x", id->name,
+ id->domain_id, id->svc_id);
+ } else {
+ dev_set_name(&adev->dev, "%s:%s", dev_name(dev), np->name);
+ }
+
+ adev->dev.bus = &aprbus_type;
+ adev->dev.parent = dev;
+ adev->dev.of_node = np;
+ adev->dev.release = apr_dev_release;
+ adev->dev.driver = NULL;
+
+ spin_lock(&apr->svcs_lock);
+ list_add_tail(&adev->node, &apr->svcs);
+ spin_unlock(&apr->svcs_lock);
+
+ dev_info(dev, "Adding APR dev: %s\n", dev_name(&adev->dev));
+
+ return device_register(&adev->dev);
+}
+
+static void of_register_apr_devices(struct device *dev)
+{
+ struct apr *apr = dev_get_drvdata(dev);
+ struct device_node *node;
+
+ for_each_child_of_node(dev->of_node, node) {
+ struct apr_device_id id = {0};
+ const char *svc_name;
+ bool is_svc = false;
+
+ if (of_find_property(node, "qcom,apr-svc-id", NULL) &&
+ of_find_property(node, "qcom,apr-svc-name", NULL)) {
+ /* svc node */
+ of_property_read_u32(node, "qcom,apr-svc-id",
+ &id.svc_id);
+ of_property_read_string(node, "qcom,apr-svc-name",
+ &svc_name);
+ id.domain_id = apr->dest_domain_id;
+
+ memcpy(id.name, svc_name, strlen(svc_name) + 1);
+ is_svc = true;
+ }
+
+ if (apr_add_device(dev, node, &id, is_svc))
+ dev_err(dev, "Failed to add arp %s svc\n", svc_name);
+ }
+}
+
+static int apr_probe(struct rpmsg_device *rpdev)
+{
+ struct device *dev = &rpdev->dev;
+ struct apr *apr;
+ int ret;
+
+ apr = devm_kzalloc(dev, sizeof(*apr), GFP_KERNEL);
+ if (!apr)
+ return -ENOMEM;
+
+ ret = of_property_read_u32(dev->of_node, "qcom,apr-dest-domain-id",
+ &apr->dest_domain_id);
+ if (ret) {
+ dev_err(dev, "APR Domain ID not specified in DT\n");
+ return ret;
+ }
+
+ dev_set_drvdata(dev, apr);
+ apr->ch = rpdev->ept;
+ apr->dev = dev;
+ INIT_LIST_HEAD(&apr->svcs);
+
+ of_register_apr_devices(dev);
+
+ return 0;
+}
+
+static int apr_remove_device(struct device *dev, void *null)
+{
+ struct apr_device *adev = to_apr_device(dev);
+
+ device_unregister(&adev->dev);
+
+ return 0;
+}
+
+static void apr_remove(struct rpmsg_device *rpdev)
+{
+ device_for_each_child(&rpdev->dev, NULL, apr_remove_device);
+}
+
+/*
+ * __apr_driver_register() - Client driver registration with aprbus
+ *
+ * @drv:Client driver to be associated with client-device.
+ * @owner: owning module/driver
+ *
+ * This API will register the client driver with the aprbus
+ * It is called from the driver's module-init function.
+ */
+int __apr_driver_register(struct apr_driver *drv, struct module *owner)
+{
+ drv->driver.bus = &aprbus_type;
+ drv->driver.owner = owner;
+
+ return driver_register(&drv->driver);
+}
+EXPORT_SYMBOL_GPL(__apr_driver_register);
+
+/*
+ * apr_driver_unregister() - Undo effect of apr_driver_register
+ *
+ * @drv: Client driver to be unregistered
+ */
+void apr_driver_unregister(struct apr_driver *drv)
+{
+ driver_unregister(&drv->driver);
+}
+EXPORT_SYMBOL_GPL(apr_driver_unregister);
+
+static const struct of_device_id apr_of_match[] = {
+ { .compatible = "qcom,apr"},
+ { .compatible = "qcom,apr-v2"},
+ {}
+};
+
+static struct rpmsg_driver apr_driver = {
+ .probe = apr_probe,
+ .remove = apr_remove,
+ .callback = apr_callback,
+ .drv = {
+ .name = "qcom,apr",
+ .of_match_table = apr_of_match,
+ },
+};
+
+static int __init apr_init(void)
+{
+ int ret;
+
+ ret = bus_register(&aprbus_type);
+ if (!ret)
+ ret = register_rpmsg_driver(&apr_driver);
+
+ return ret;
+}
+
+static void __exit apr_exit(void)
+{
+ bus_unregister(&aprbus_type);
+ unregister_rpmsg_driver(&apr_driver);
+}
+
+subsys_initcall(apr_init);
+module_exit(apr_exit);
+
+MODULE_LICENSE("GPL v2");
+MODULE_DESCRIPTION("Qualcomm APR Bus");
diff --git a/include/linux/mod_devicetable.h b/include/linux/mod_devicetable.h
index 48fb2b43c35a..c05c89b5a99e 100644
--- a/include/linux/mod_devicetable.h
+++ b/include/linux/mod_devicetable.h
@@ -471,6 +471,17 @@ struct slim_device_id {
kernel_ulong_t driver_data;
};
+#define APR_NAME_SIZE 32
+#define APR_MODULE_PREFIX "apr:"
+
+struct apr_device_id {
+ char name[APR_NAME_SIZE];
+ __u32 domain_id;
+ __u32 svc_id;
+ __u32 svc_version;
+ kernel_ulong_t driver_data; /* Data private to the driver */
+};
+
#define SPMI_NAME_SIZE 32
#define SPMI_MODULE_PREFIX "spmi:"
diff --git a/include/linux/soc/qcom/apr.h b/include/linux/soc/qcom/apr.h
new file mode 100644
index 000000000000..dbc9efe185d1
--- /dev/null
+++ b/include/linux/soc/qcom/apr.h
@@ -0,0 +1,131 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * Copyright (c) 2011-2017, The Linux Foundation
+ * Copyright (c) 2018, Linaro Limited
+ */
+
+#ifndef __QCOM_APR_H_
+#define __QCOM_APR_H_
+
+#include <linux/spinlock.h>
+#include <linux/device.h>
+#include <linux/mod_devicetable.h>
+#include <dt-bindings/soc/qcom,apr.h>
+
+#define APR_HDR_LEN(hdr_len) ((hdr_len)/4)
+
+/*
+ * HEADER field
+ * version:0:3
+ * header_size : 4:7
+ * message_type : 8:9
+ * reserved: 10:15
+ */
+#define APR_HDR_FIELD(msg_type, hdr_len, ver)\
+ (((msg_type & 0x3) << 8) | ((hdr_len & 0xF) << 4) | (ver & 0xF))
+
+#define APR_HDR_SIZE sizeof(struct apr_hdr)
+#define APR_SEQ_CMD_HDR_FIELD APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, \
+ APR_HDR_LEN(APR_HDR_SIZE), \
+ APR_PKT_VER)
+/* Version */
+#define APR_PKT_VER 0x0
+
+/* Command and Response Types */
+#define APR_MSG_TYPE_EVENT 0x0
+#define APR_MSG_TYPE_CMD_RSP 0x1
+#define APR_MSG_TYPE_SEQ_CMD 0x2
+#define APR_MSG_TYPE_NSEQ_CMD 0x3
+#define APR_MSG_TYPE_MAX 0x04
+
+/* APR Basic Response Message */
+#define APR_BASIC_RSP_RESULT 0x000110E8
+#define APR_RSP_ACCEPTED 0x000100BE
+
+struct aprv2_ibasic_rsp_result_t {
+ uint32_t opcode;
+ uint32_t status;
+};
+
+/* hdr field Ver [0:3], Size [4:7], Message type [8:10] */
+#define APR_HDR_FIELD_VER(h) (h & 0x000F)
+#define APR_HDR_FIELD_SIZE(h) ((h & 0x00F0) >> 4)
+#define APR_HDR_FIELD_SIZE_BYTES(h) (((h & 0x00F0) >> 4) * 4)
+#define APR_HDR_FIELD_MT(h) ((h & 0x0300) >> 8)
+
+struct apr_hdr {
+ uint16_t hdr_field;
+ uint16_t pkt_size;
+ uint8_t src_svc;
+ uint8_t src_domain;
+ uint16_t src_port;
+ uint8_t dest_svc;
+ uint8_t dest_domain;
+ uint16_t dest_port;
+ uint32_t token;
+ uint32_t opcode;
+};
+
+struct apr_client_message {
+ uint16_t payload_size;
+ uint16_t hdr_len;
+ uint16_t msg_type;
+ uint16_t src;
+ uint16_t dest_svc;
+ uint16_t src_port;
+ uint16_t dest_port;
+ uint32_t token;
+ uint32_t opcode;
+ void *payload;
+};
+
+/* Bits 0 to 15 -- Minor version, Bits 16 to 31 -- Major version */
+#define APR_SVC_MAJOR_VERSION(v) ((v >> 16) & 0xFF)
+#define APR_SVC_MINOR_VERSION(v) (v & 0xFF)
+
+struct apr_device {
+ struct device dev;
+ uint16_t svc_id;
+ uint16_t domain_id;
+ uint32_t version;
+ spinlock_t lock;
+ struct list_head node;
+};
+
+#define to_apr_device(d) container_of(d, struct apr_device, dev)
+
+struct apr_driver {
+ int (*probe)(struct apr_device *sl);
+ int (*remove)(struct apr_device *sl);
+ int (*callback)(struct apr_device *a,
+ struct apr_client_message *d);
+ struct device_driver driver;
+ const struct apr_device_id *id_table;
+};
+
+#define to_apr_driver(d) container_of(d, struct apr_driver, driver)
+
+/*
+ * use a macro to avoid include chaining to get THIS_MODULE
+ */
+#define apr_driver_register(drv) __apr_driver_register(drv, THIS_MODULE)
+
+int __apr_driver_register(struct apr_driver *drv, struct module *owner);
+void apr_driver_unregister(struct apr_driver *drv);
+
+/**
+ * module_apr_driver() - Helper macro for registering a aprbus driver
+ * @__aprbus_driver: aprbus_driver struct
+ *
+ * Helper macro for aprbus drivers which do not do anything special in
+ * module init/exit. This eliminates a lot of boilerplate. Each module
+ * may only use this macro once, and calling it replaces module_init()
+ * and module_exit()
+ */
+#define module_apr_driver(__apr_driver) \
+ module_driver(__apr_driver, apr_driver_register, \
+ apr_driver_unregister)
+
+int apr_send_pkt(struct apr_device *adev, void *buf);
+
+#endif /* __QCOM_APR_H_ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* Re: [alsa-devel] [PATCH v4 02/24] soc: qcom: Add APR bus driver
2018-03-10 2:24 ` [PATCH v4 02/24] soc: qcom: Add APR bus driver srinivas.kandagatla
@ 2018-03-14 14:12 ` Rohit Kumar
2018-03-14 14:15 ` Srinivas Kandagatla
0 siblings, 1 reply; 33+ messages in thread
From: Rohit Kumar @ 2018-03-14 14:12 UTC (permalink / raw)
To: srinivas.kandagatla, andy.gross, broonie, linux-arm-msm,
alsa-devel, robh+dt
Cc: mark.rutland, devicetree, bgoswami, rohkumar, gregkh, plai,
lgirdwood, tiwai, david.brown, linux-arm-kernel, spatakok,
linux-kernel
On 3/10/2018 7:54 AM, srinivas.kandagatla@linaro.org wrote:
> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
>
[..]
> +struct bus_type aprbus_type = {
> + .name = "aprbus",
> + .match = apr_device_match,
> + .probe = apr_device_probe,
> + .remove = apr_device_remove,
> + .force_dma = true,
There is no need of force_dma now as machine driver is not child of apr.
Please remove it.
> +};
> +EXPORT_SYMBOL_GPL(aprbus_type);
> +
>
^ permalink raw reply [flat|nested] 33+ messages in thread* Re: [alsa-devel] [PATCH v4 02/24] soc: qcom: Add APR bus driver
2018-03-14 14:12 ` [alsa-devel] " Rohit Kumar
@ 2018-03-14 14:15 ` Srinivas Kandagatla
0 siblings, 0 replies; 33+ messages in thread
From: Srinivas Kandagatla @ 2018-03-14 14:15 UTC (permalink / raw)
To: Rohit Kumar, andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: mark.rutland, devicetree, bgoswami, rohkumar, gregkh, plai,
lgirdwood, tiwai, david.brown, linux-arm-kernel, spatakok,
linux-kernel
On 14/03/18 14:12, Rohit Kumar wrote:
>
>
> On 3/10/2018 7:54 AM, srinivas.kandagatla@linaro.org wrote:
>> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
>>
> [..]
>> +struct bus_type aprbus_type = {
>> + .name = "aprbus",
>> + .match = apr_device_match,
>> + .probe = apr_device_probe,
>> + .remove = apr_device_remove,
>> + .force_dma = true,
>
> There is no need of force_dma now as machine driver is not child of apr.
> Please remove it.
Thanks Rohit, Will remove it.
>> +};
>> +EXPORT_SYMBOL_GPL(aprbus_type);
>> +
>>
>
^ permalink raw reply [flat|nested] 33+ messages in thread
* [PATCH v4 03/24] ASoC: qdsp6: q6common: Add qdsp6 helper functions
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 01/24] soc: qcom dt-bindings: Add APR bus bindings srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 02/24] soc: qcom: Add APR bus driver srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 04/24] ASoC: qdsp6: dt-bindings: Add q6afe dt bindings srinivas.kandagatla
` (20 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds some common helper functions like translating dsp error
to linux error codes and channel mappings etc.
These functions are used in all the following qdsp6 drivers.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/Kconfig | 13 ++++++++
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6dsp-common.c | 66 +++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6dsp-common.h | 24 ++++++++++++++
sound/soc/qcom/qdsp6/q6dsp-errno.h | 51 ++++++++++++++++++++++++++++
5 files changed, 155 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/Makefile
create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.c
create mode 100644 sound/soc/qcom/qdsp6/q6dsp-common.h
create mode 100644 sound/soc/qcom/qdsp6/q6dsp-errno.h
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 8ec9a074b38b..8247ade16dbe 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -43,3 +43,16 @@ config SND_SOC_APQ8016_SBC
Support for Qualcomm Technologies LPASS audio block in
APQ8016 SOC-based systems.
Say Y if you want to use audio devices on MI2S.
+
+config SND_SOC_QDSP6_COMMON
+ tristate
+
+config SND_SOC_QDSP6
+ tristate "SoC ALSA audio driver for QDSP6"
+ depends on QCOM_APR && HAS_DMA
+ select SND_SOC_QDSP6_COMMON
+ help
+ To add support for MSM QDSP6 Soc Audio.
+ This will enable sound soc platform specific
+ audio drivers. This includes q6asm, q6adm,
+ q6afe interfaces to DSP using apr.
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
new file mode 100644
index 000000000000..accebdb49306
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -0,0 +1 @@
+obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
diff --git a/sound/soc/qcom/qdsp6/q6dsp-common.c b/sound/soc/qcom/qdsp6/q6dsp-common.c
new file mode 100644
index 000000000000..c342a073144c
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6dsp-common.c
@@ -0,0 +1,66 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation
+// Copyright (c) 2018, Linaro Limited
+
+#include "q6dsp-common.h"
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/string.h>
+#include <linux/errno.h>
+
+int q6dsp_map_channels(u8 ch_map[PCM_FORMAT_MAX_NUM_CHANNEL], int ch)
+{
+ memset(ch_map, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
+
+ switch (ch) {
+ case 1:
+ ch_map[0] = PCM_CHANNEL_FC;
+ break;
+ case 2:
+ ch_map[0] = PCM_CHANNEL_FL;
+ ch_map[1] = PCM_CHANNEL_FR;
+ break;
+ case 3:
+ ch_map[0] = PCM_CHANNEL_FL;
+ ch_map[1] = PCM_CHANNEL_FR;
+ ch_map[2] = PCM_CHANNEL_FC;
+ break;
+ case 4:
+ ch_map[0] = PCM_CHANNEL_FL;
+ ch_map[1] = PCM_CHANNEL_FR;
+ ch_map[2] = PCM_CHANNEL_LS;
+ ch_map[3] = PCM_CHANNEL_RS;
+ break;
+ case 5:
+ ch_map[0] = PCM_CHANNEL_FL;
+ ch_map[1] = PCM_CHANNEL_FR;
+ ch_map[2] = PCM_CHANNEL_FC;
+ ch_map[3] = PCM_CHANNEL_LS;
+ ch_map[4] = PCM_CHANNEL_RS;
+ break;
+ case 6:
+ ch_map[0] = PCM_CHANNEL_FL;
+ ch_map[1] = PCM_CHANNEL_FR;
+ ch_map[2] = PCM_CHANNEL_LFE;
+ ch_map[3] = PCM_CHANNEL_FC;
+ ch_map[4] = PCM_CHANNEL_LS;
+ ch_map[5] = PCM_CHANNEL_RS;
+ break;
+ case 8:
+ ch_map[0] = PCM_CHANNEL_FL;
+ ch_map[1] = PCM_CHANNEL_FR;
+ ch_map[2] = PCM_CHANNEL_LFE;
+ ch_map[3] = PCM_CHANNEL_FC;
+ ch_map[4] = PCM_CHANNEL_LS;
+ ch_map[5] = PCM_CHANNEL_RS;
+ ch_map[6] = PCM_CHANNEL_LB;
+ ch_map[7] = PCM_CHANNEL_RB;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6dsp_map_channels);
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6dsp-common.h b/sound/soc/qcom/qdsp6/q6dsp-common.h
new file mode 100644
index 000000000000..32386f4a6432
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6dsp-common.h
@@ -0,0 +1,24 @@
+// SPDX-License-Identifier: GPL-2.0
+
+#ifndef __Q6DSP_COMMON_H__
+#define __Q6DSP_COMMON_H__
+
+#include <linux/kernel.h>
+
+#define PCM_FORMAT_MAX_NUM_CHANNEL 8
+#define PCM_CHANNEL_NULL 0
+
+#define PCM_CHANNEL_FL 1 /* Front left channel. */
+#define PCM_CHANNEL_FR 2 /* Front right channel. */
+#define PCM_CHANNEL_FC 3 /* Front center channel. */
+#define PCM_CHANNEL_LS 4 /* Left surround channel. */
+#define PCM_CHANNEL_RS 5 /* Right surround channel. */
+#define PCM_CHANNEL_LFE 6 /* Low frequency effect channel. */
+#define PCM_CHANNEL_CS 7 /* Center surround channel; Rear center ch */
+#define PCM_CHANNEL_LB 8 /* Left back channel; Rear left channel. */
+#define PCM_CHANNEL_RB 9 /* Right back channel; Rear right channel. */
+#define PCM_CHANNELS 10 /* Top surround channel. */
+
+int q6dsp_map_channels(u8 ch_map[PCM_FORMAT_MAX_NUM_CHANNEL], int ch);
+
+#endif /* __Q6DSP_COMMON_H__ */
diff --git a/sound/soc/qcom/qdsp6/q6dsp-errno.h b/sound/soc/qcom/qdsp6/q6dsp-errno.h
new file mode 100644
index 000000000000..6285e1782824
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6dsp-errno.h
@@ -0,0 +1,51 @@
+// SPDX-License-Identifier: GPL-2.0
+
+#ifndef __Q6DSP_ERR_NO_H__
+#define __Q6DSP_ERR_NO_H__
+#include <linux/kernel.h>
+
+/* Success. The operation completed with no errors. */
+#define ADSP_EOK 0x00000000
+/* General failure. */
+#define ADSP_EFAILED 0x00000001
+/* Bad operation parameter. */
+#define ADSP_EBADPARAM 0x00000002
+/* Unsupported routine or operation. */
+#define ADSP_EUNSUPPORTED 0x00000003
+/* Unsupported version. */
+#define ADSP_EVERSION 0x00000004
+/* Unexpected problem encountered. */
+#define ADSP_EUNEXPECTED 0x00000005
+/* Unhandled problem occurred. */
+#define ADSP_EPANIC 0x00000006
+/* Unable to allocate resource. */
+#define ADSP_ENORESOURCE 0x00000007
+/* Invalid handle. */
+#define ADSP_EHANDLE 0x00000008
+/* Operation is already processed. */
+#define ADSP_EALREADY 0x00000009
+/* Operation is not ready to be processed. */
+#define ADSP_ENOTREADY 0x0000000A
+/* Operation is pending completion. */
+#define ADSP_EPENDING 0x0000000B
+/* Operation could not be accepted or processed. */
+#define ADSP_EBUSY 0x0000000C
+/* Operation aborted due to an error. */
+#define ADSP_EABORTED 0x0000000D
+/* Operation preempted by a higher priority. */
+#define ADSP_EPREEMPTED 0x0000000E
+/* Operation requests intervention to complete. */
+#define ADSP_ECONTINUE 0x0000000F
+/* Operation requests immediate intervention to complete. */
+#define ADSP_EIMMEDIATE 0x00000010
+/* Operation is not implemented. */
+#define ADSP_ENOTIMPL 0x00000011
+/* Operation needs more data or resources. */
+#define ADSP_ENEEDMORE 0x00000012
+/* Operation does not have memory. */
+#define ADSP_ENOMEMORY 0x00000014
+/* Item does not exist. */
+#define ADSP_ENOTEXIST 0x00000015
+/* Max count for adsp error code sent to HLOS*/
+
+#endif /*__Q6DSP_ERR_NO_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 04/24] ASoC: qdsp6: dt-bindings: Add q6afe dt bindings
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (2 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 03/24] ASoC: qdsp6: q6common: Add qdsp6 helper functions srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 05/24] ASoC: qdsp6: dt-bindings: Add q6adm " srinivas.kandagatla
` (19 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch add DT bindings for AFE (Audio Frontend) DSP module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
.../devicetree/bindings/sound/qcom,q6afe.txt | 107 +++++++++++++++++++++
include/dt-bindings/sound/qcom,q6afe.h | 31 ++++++
2 files changed, 138 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt
create mode 100644 include/dt-bindings/sound/qcom,q6afe.h
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
new file mode 100644
index 000000000000..1e4b82e7747a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt
@@ -0,0 +1,107 @@
+Qualcomm Audio Front End (Q6AFE) binding
+
+AFE is one of the APR audio service on Q6DSP
+Please refer to qcom,apr.txt for details of the common apr service bindings
+used by all apr services.
+
+- but must contain the following property:
+
+- compatible:
+ Usage: required
+ Value type: <stringlist>
+ Definition: must be "qcom,afe-v<MAJOR-NUMBER>.<MINOR-NUMBER>"
+ Or "qcom,afe" where the version number can be queried
+ from DSP.
+ example "qcom,afe-v2.0"
+
+= AFE DAIs (Digial Audio Interface)
+"dais" subnode of the AFE node represents dai specific configuration
+
+- #sound-dai-cells
+ Usage: required
+ Value type: <prop-encoded-array>
+ Definition: Must be 1
+
+- reg
+ Usage: required
+ Value type: <prop-encoded-array>
+ Definition: Must be dai id
+
+- interface-type
+ Usage: required
+ Value type: <stringlist>
+ Definition: Must be one of following
+ "mi2s"
+ "hdmi"
+ "pcm"
+ "slim"
+
+- qcom,sd-lines
+ Usage: required for mi2s interface
+ Value type: <prop-encoded-array>
+ Definition: Must be list of serial data lines used by this dai.
+ should be one or more of the 1-4 sd lines.
+
+= EXAMPLE
+
+q6afe {
+ compatible = "qcom,q6afe";
+ qcom,apr-svc-name = "AFE";
+ qcom,apr-svc-id = <APR_SVC_AFE>;
+
+ q6asmdai: dais {
+ #sound-dai-cells = <1>;
+ hdmi@1 {
+ reg = <1>;
+ interface-type = "hdmi";
+ };
+
+ prim-mi2s-rx@16 {
+ reg = <16>;
+ qcom,sd-lines = <1 3>;
+ interface-type = "mi2s";
+ };
+
+ prim-mi2s-tx@17 {
+ reg = <17>;
+ qcom,sd-lines = <2>;
+ interface-type = "mi2s";
+ };
+
+ sec-mi2s-rx@18 {
+ reg = <18>;
+ qcom,sd-lines = <1 4>;
+ interface-type = "mi2s";
+ };
+
+ sec-mi2s-tx@19 {
+ reg = <19>;
+ qcom,sd-lines = <2>;
+ interface-type = "mi2s";
+ };
+
+ tert-mi2s-rx@20 {
+ reg = <20>;
+ qcom,sd-lines = <2 4>;
+ interface-type = "mi2s";
+ };
+
+ tert-mi2s-tx@21 {
+ reg = <21>;
+ qcom,sd-lines = <1>;
+ interface-type = "mi2s";
+ };
+
+ quat-mi2s-rx@22 {
+ reg = <22>;
+ qcom,sd-lines = <1>;
+ interface-type = "mi2s";
+ };
+
+ quat-mi2s-tx@23 {
+ reg = <23>;
+ qcom,sd-lines = <2>;
+ interface-type = "mi2s";
+ };
+ };
+};
diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h
new file mode 100644
index 000000000000..3c7868394889
--- /dev/null
+++ b/include/dt-bindings/sound/qcom,q6afe.h
@@ -0,0 +1,31 @@
+// SPDX-License-Identifier: GPL-2.0
+#ifndef __DT_BINDINGS_Q6_AFE_H__
+#define __DT_BINDINGS_Q6_AFE_H__
+
+/* Audio Front End (AFE) virtual ports IDs */
+#define HDMI_RX 1
+#define SLIMBUS_0_RX 2
+#define SLIMBUS_0_TX 3
+#define SLIMBUS_1_RX 4
+#define SLIMBUS_1_TX 5
+#define SLIMBUS_2_RX 6
+#define SLIMBUS_2_TX 7
+#define SLIMBUS_3_RX 8
+#define SLIMBUS_3_TX 9
+#define SLIMBUS_4_RX 10
+#define SLIMBUS_4_TX 11
+#define SLIMBUS_5_RX 12
+#define SLIMBUS_5_TX 13
+#define SLIMBUS_6_RX 14
+#define SLIMBUS_6_TX 15
+#define PRIMARY_MI2S_RX 16
+#define PRIMARY_MI2S_TX 17
+#define SECONDARY_MI2S_RX 18
+#define SECONDARY_MI2S_TX 19
+#define TERTIARY_MI2S_RX 20
+#define TERTIARY_MI2S_TX 21
+#define QUATERNARY_MI2S_RX 22
+#define QUATERNARY_MI2S_TX 23
+
+#endif /* __DT_BINDINGS_Q6_AFE_H__ */
+
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 05/24] ASoC: qdsp6: dt-bindings: Add q6adm dt bindings
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (3 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 04/24] ASoC: qdsp6: dt-bindings: Add q6afe dt bindings srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 06/24] ASoC: qdsp6: dt-bindings: Add q6asm " srinivas.kandagatla
` (18 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch add DT bindings for ADM (Audio Device Manager) DSP module.
This module implements mixer controls to setup the connections between
AFE ports and ASM streams.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
.../devicetree/bindings/sound/qcom,q6adm.txt | 34 ++++++++++++++++++++++
1 file changed, 34 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.txt
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6adm.txt b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt
new file mode 100644
index 000000000000..ddd93c80bd8a
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,q6adm.txt
@@ -0,0 +1,34 @@
+Qualcomm Audio Device Manager (Q6ADM) binding
+
+Q6ADM is one of the APR audio service on Q6DSP.
+Please refer to qcom,apr.txt for details of the coommon apr service bindings
+used by the apr service device.
+
+- but must contain the following property:
+
+- compatible:
+ Usage: required
+ Value type: <stringlist>
+ Definition: must be "qcom,adm-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
+ Or "qcom,adm" where the version number can be queried
+ from DSP.
+ example "qcom,adm-v2.0"
+
+
+= ADM routing
+"routing" subnode of the ADM node represents adm routing specific configuration
+
+- #sound-dai-cells
+ Usage: required
+ Value type: <prop-encoded-array>
+ Definition: Must be 0
+
+= EXAMPLE
+q6adm: q6adm {
+ compatible = "qcom,q6adm";
+ qcom,apr-svc-name = "ADM";
+ qcom,apr-svc-id = <APR_SVC_ADM>;
+ q6routing: routing {
+ #sound-dai-cells = <0>;
+ };
+};
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 06/24] ASoC: qdsp6: dt-bindings: Add q6asm dt bindings
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (4 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 05/24] ASoC: qdsp6: dt-bindings: Add q6adm " srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 07/24] ASoC: qdsp6: q6afe: Add q6afe driver srinivas.kandagatla
` (17 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch add DT bindings for ASM (Audio Stream Manager) DSP module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
.../devicetree/bindings/sound/qcom,q6asm.txt | 34 ++++++++++++++++++++++
include/dt-bindings/sound/qcom,q6asm.h | 22 ++++++++++++++
2 files changed, 56 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt
create mode 100644 include/dt-bindings/sound/qcom,q6asm.h
diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt
new file mode 100644
index 000000000000..84703747db3c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt
@@ -0,0 +1,34 @@
+Qualcomm Audio Stream Manager (Q6ASM) binding
+
+Q6ASM is one of the APR audio service on Q6DSP.
+Please refer to qcom,apr.txt for details of the coommon apr service bindings
+used by the apr service device.
+
+- but must contain the following property:
+
+- compatible:
+ Usage: required
+ Value type: <stringlist>
+ Definition: must be "qcom,asm-v<MAJOR-NUMBER>.<MINOR-NUMBER>".
+ Or "qcom,asm" where the version number can be queried
+ from DSP.
+ example "qcom,asm-v2.0"
+
+= ASM DAIs (Digial Audio Interface)
+"dais" subnode of the ASM node represents dai specific configuration
+
+- #sound-dai-cells
+ Usage: required
+ Value type: <prop-encoded-array>
+ Definition: Must be 1
+
+= EXAMPLE
+
+q6asm: q6asm {
+ compatible = "qcom,q6asm";
+ qcom,apr-svc-name = "ASM";
+ qcom,apr-svc-id = <APR_SVC_ASM>;
+ q6asmdai: dais {
+ #sound-dai-cells = <1>;
+ };
+};
diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h
new file mode 100644
index 000000000000..2e11b15d930d
--- /dev/null
+++ b/include/dt-bindings/sound/qcom,q6asm.h
@@ -0,0 +1,22 @@
+// SPDX-License-Identifier: GPL-2.0
+#ifndef __DT_BINDINGS_Q6_ASM_H__
+#define __DT_BINDINGS_Q6_ASM_H__
+
+#define MSM_FRONTEND_DAI_MULTIMEDIA1 0
+#define MSM_FRONTEND_DAI_MULTIMEDIA2 1
+#define MSM_FRONTEND_DAI_MULTIMEDIA3 2
+#define MSM_FRONTEND_DAI_MULTIMEDIA4 3
+#define MSM_FRONTEND_DAI_MULTIMEDIA5 4
+#define MSM_FRONTEND_DAI_MULTIMEDIA6 5
+#define MSM_FRONTEND_DAI_MULTIMEDIA7 6
+#define MSM_FRONTEND_DAI_MULTIMEDIA8 7
+#define MSM_FRONTEND_DAI_MULTIMEDIA9 8
+#define MSM_FRONTEND_DAI_MULTIMEDIA10 9
+#define MSM_FRONTEND_DAI_MULTIMEDIA11 10
+#define MSM_FRONTEND_DAI_MULTIMEDIA12 11
+#define MSM_FRONTEND_DAI_MULTIMEDIA13 12
+#define MSM_FRONTEND_DAI_MULTIMEDIA14 13
+#define MSM_FRONTEND_DAI_MULTIMEDIA15 14
+#define MSM_FRONTEND_DAI_MULTIMEDIA16 15
+
+#endif /* __DT_BINDINGS_Q6_ASM_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 07/24] ASoC: qdsp6: q6afe: Add q6afe driver
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (5 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 06/24] ASoC: qdsp6: dt-bindings: Add q6asm " srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-13 11:29 ` [alsa-devel] " Rohit Kumar
2018-03-10 2:24 ` [PATCH v4 08/24] ASoC: qdsp6: qdafe: Add SLIMBus port Support srinivas.kandagatla
` (16 subsequent siblings)
23 siblings, 1 reply; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to Q6AFE (Audio Front End) module on Q6DSP.
AFE module sits right at the other end of cpu where the codec/audio
devices are connected.
AFE provides abstraced interfaces to both hardware and virtual devices.
Each AFE tx/rx port can be configured to connect to one of the hardware
devices like codec, hdmi, slimbus, i2s and so on. AFE services include
starting, stopping, and if needed, any configurations of the ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/Makefile | 3 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6afe.c | 503 ++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 35 +++
5 files changed, 546 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6afe.c
create mode 100644 sound/soc/qcom/qdsp6/q6afe.h
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 8247ade16dbe..f50c76d712c0 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -47,10 +47,14 @@ config SND_SOC_APQ8016_SBC
config SND_SOC_QDSP6_COMMON
tristate
+config SND_SOC_QDSP6_AFE
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
select SND_SOC_QDSP6_COMMON
+ select SND_SOC_QDSP6_AFE
help
To add support for MSM QDSP6 Soc Audio.
This will enable sound soc platform specific
diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
index d5280355c24f..0276717917c0 100644
--- a/sound/soc/qcom/Makefile
+++ b/sound/soc/qcom/Makefile
@@ -16,3 +16,6 @@ snd-soc-apq8016-sbc-objs := apq8016_sbc.o
obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
+
+#DSP lib
+obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index accebdb49306..9ec951e0833b 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -1 +1,2 @@
obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
+obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
new file mode 100644
index 000000000000..fd76f7296f36
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -0,0 +1,503 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/slab.h>
+#include <linux/kernel.h>
+#include <linux/uaccess.h>
+#include <linux/wait.h>
+#include <linux/jiffies.h>
+#include <linux/sched.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/spinlock.h>
+#include <linux/delay.h>
+#include <linux/soc/qcom/apr.h>
+#include "q6dsp-errno.h"
+#include "q6core.h"
+#include "q6afe.h"
+
+/* AFE CMDs */
+#define AFE_PORT_CMD_DEVICE_START 0x000100E5
+#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6
+#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF
+#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106
+#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210
+#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C
+
+/* Port IDs */
+#define AFE_API_VERSION_HDMI_CONFIG 0x1
+#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
+#define TIMEOUT_MS 1000
+#define AFE_CMD_RESP_AVAIL 0
+#define AFE_CMD_RESP_NONE 1
+
+struct q6afe {
+ struct apr_device *apr;
+ struct device *dev;
+ struct q6core_svc_api_info ainfo;
+ struct mutex lock;
+ struct list_head port_list;
+ spinlock_t port_list_lock;
+ struct list_head node;
+ struct platform_device *pdev_dais;
+};
+
+struct afe_port_cmd_device_start {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 reserved;
+} __packed;
+
+struct afe_port_cmd_device_stop {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.*/
+} __packed;
+
+struct afe_port_param_data_v2 {
+ u32 module_id;
+ u32 param_id;
+ u16 param_size;
+ u16 reserved;
+} __packed;
+
+struct afe_port_cmd_set_param_v2 {
+ u16 port_id;
+ u16 payload_size;
+ u32 payload_address_lsw;
+ u32 payload_address_msw;
+ u32 mem_map_handle;
+} __packed;
+
+struct afe_param_id_hdmi_multi_chan_audio_cfg {
+ u32 hdmi_cfg_minor_version;
+ u16 datatype;
+ u16 channel_allocation;
+ u32 sample_rate;
+ u16 bit_width;
+ u16 reserved;
+} __packed;
+
+union afe_port_config {
+ struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
+} __packed;
+
+struct q6afe_port {
+ wait_queue_head_t wait;
+ union afe_port_config port_cfg;
+ int token;
+ int id;
+ int cfg_type;
+ struct q6afe *afe;
+ struct list_head node;
+};
+
+struct afe_audioif_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ union afe_port_config pcfg;
+} __packed;
+
+struct afe_port_map {
+ int port_id;
+ int token;
+ int is_rx;
+ int is_dig_pcm;
+};
+
+/**
+ * Mapping between Virtual Port IDs to DSP AFE Port ID
+ * On B Family SoCs DSP Port IDs are consistent across multiple SoCs
+ * on A Family SoCs DSP port IDs are same as virtual Port IDs.
+ **/
+
+static struct afe_port_map port_maps[AFE_PORT_MAX] = {
+ [HDMI_RX] = { AFE_PORT_ID_MULTICHAN_HDMI_RX, HDMI_RX, 1, 1},
+};
+
+static struct q6afe_port *afe_find_port(struct q6afe *afe, int token)
+{
+ struct q6afe_port *p = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&afe->port_list_lock, flags);
+ list_for_each_entry(p, &afe->port_list, node)
+ if (p->token == token)
+ break;
+
+ spin_unlock_irqrestore(&afe->port_list_lock, flags);
+ return p;
+}
+
+static int q6afe_callback(struct apr_device *adev,
+ struct apr_client_message *data)
+{
+ struct q6afe *afe = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *res;
+ struct q6afe_port *port;
+
+ if (!data->payload_size)
+ return 0;
+
+ res = data->payload;
+ switch (data->opcode) {
+ case APR_BASIC_RSP_RESULT: {
+ if (res->status) {
+ dev_err(afe->dev, "cmd = 0x%x returned error = 0x%x\n",
+ res->opcode, res->status);
+ }
+ switch (res->opcode) {
+ case AFE_PORT_CMD_SET_PARAM_V2:
+ case AFE_PORT_CMD_DEVICE_STOP:
+ case AFE_PORT_CMD_DEVICE_START:
+ port = afe_find_port(afe, data->token);
+ if (port) {
+ port->result = *res;
+ wake_up(&port->wait);
+ }
+ break;
+ default:
+ dev_err(afe->dev, "Unknown cmd 0x%x\n", res->opcode);
+ break;
+ }
+ }
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+/**
+ * q6afe_get_port_id() - Get port id from a given port index
+ *
+ * @index: port index
+ *
+ * Return: Will be an negative on error or valid port_id on success
+ */
+int q6afe_get_port_id(int index)
+{
+ if (index < 0 || index > AFE_PORT_MAX)
+ return -EINVAL;
+
+ return port_maps[index].port_id;
+}
+EXPORT_SYMBOL_GPL(q6afe_get_port_id);
+
+static int afe_apr_send_pkt(struct q6afe *afe, void *data,
+ struct q6afe_port *port)
+{
+ wait_queue_head_t *wait = &port->wait;
+ struct apr_hdr *hdr = data;
+ int ret;
+
+ mutex_lock(&afe->lock);
+ port->result.opcode = 0;
+ port->result.status = 0;
+
+ ret = apr_send_pkt(afe->apr, data);
+ if (ret < 0) {
+ dev_err(afe->dev, "packet not transmitted (%d)\n", ret);
+ ret = -EINVAL;
+ goto err;
+ }
+
+ ret = wait_event_timeout(*wait, (port->result.opcode == hdr->opcode),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ ret = -ETIMEDOUT;
+ } else if (port->result.status > 0) {
+ dev_err(afe->dev, "DSP returned error[%x]\n",
+ port->result.status);
+ ret = -EINVAL;
+ } else {
+ ret = 0;
+ }
+
+err:
+ mutex_unlock(&afe->lock);
+
+ return ret;
+}
+
+static int q6afe_port_set_param_v2(struct q6afe_port *port, void *data,
+ int param_id, int psize)
+{
+ struct apr_hdr *hdr;
+ struct afe_port_cmd_set_param_v2 *param;
+ struct afe_port_param_data_v2 *pdata;
+ struct q6afe *afe = port->afe;
+ u16 port_id = port->id;
+ int ret;
+
+ hdr = data;
+ param = data + sizeof(*hdr);
+ pdata = data + sizeof(*hdr) + sizeof(*param);
+
+ hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ hdr->pkt_size = sizeof(*hdr) + sizeof(*param) +
+ sizeof(*pdata) + psize;
+ hdr->src_port = 0;
+ hdr->dest_port = 0;
+ hdr->token = port->token;
+ hdr->opcode = AFE_PORT_CMD_SET_PARAM_V2;
+ param->port_id = port_id;
+ param->payload_size = sizeof(*pdata) + psize;
+ param->payload_address_lsw = 0x00;
+ param->payload_address_msw = 0x00;
+ param->mem_map_handle = 0x00;
+ pdata->module_id = AFE_MODULE_AUDIO_DEV_INTERFACE;
+ pdata->param_id = param_id;
+ pdata->param_size = psize;
+
+ ret = afe_apr_send_pkt(afe, data, port);
+ if (ret)
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+
+
+ return ret;
+}
+
+/**
+ * q6afe_port_stop() - Stop a afe port
+ *
+ * @port: Instance of port to stop
+ *
+ * Return: Will be an negative on packet size on success.
+ */
+int q6afe_port_stop(struct q6afe_port *port)
+{
+ int port_id = port->id;
+ struct afe_port_cmd_device_stop stop;
+ struct q6afe *afe = port->afe;
+ int ret = 0;
+ int index;
+
+ port_id = port->id;
+ index = port->token;
+ if (index < 0 || index > AFE_PORT_MAX) {
+ dev_err(afe->dev, "AFE port index[%d] invalid!\n", index);
+ return -EINVAL;
+ }
+
+ stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ stop.hdr.pkt_size = sizeof(stop);
+ stop.hdr.src_port = 0;
+ stop.hdr.dest_port = 0;
+ stop.hdr.token = index;
+ stop.hdr.opcode = AFE_PORT_CMD_DEVICE_STOP;
+ stop.port_id = port_id;
+ stop.reserved = 0;
+
+ ret = afe_apr_send_pkt(afe, &stop, port);
+ if (ret)
+ dev_err(afe->dev, "AFE close failed %d\n", ret);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6afe_port_stop);
+
+/**
+ * q6afe_hdmi_port_prepare() - Prepare hdmi afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: HDMI configuration for the afe port
+ *
+ */
+void q6afe_hdmi_port_prepare(struct q6afe_port *port,
+ struct q6afe_hdmi_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+
+ pcfg->hdmi_multi_ch.hdmi_cfg_minor_version =
+ AFE_API_VERSION_HDMI_CONFIG;
+ pcfg->hdmi_multi_ch.datatype = cfg->datatype;
+ pcfg->hdmi_multi_ch.channel_allocation = cfg->channel_allocation;
+ pcfg->hdmi_multi_ch.sample_rate = cfg->sample_rate;
+ pcfg->hdmi_multi_ch.bit_width = cfg->bit_width;
+}
+EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
+
+/**
+ * q6afe_port_start() - Start a afe port
+ *
+ * @port: Instance of port to start
+ *
+ * Return: Will be an negative on packet size on success.
+ */
+int q6afe_port_start(struct q6afe_port *port)
+{
+ struct afe_audioif_config_command config = {0,};
+ struct afe_port_cmd_device_start start;
+ struct q6afe *afe = port->afe;
+ int port_id = port->id;
+ int ret, param_id = port->cfg_type;
+
+ config.pcfg = port->port_cfg;
+
+ ret = q6afe_port_set_param_v2(port, &config, param_id,
+ sizeof(config.pcfg));
+ if (ret) {
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+ return ret;
+ }
+
+ start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ start.hdr.pkt_size = sizeof(start);
+ start.hdr.src_port = 0;
+ start.hdr.dest_port = 0;
+ start.hdr.token = port->token;
+ start.hdr.opcode = AFE_PORT_CMD_DEVICE_START;
+ start.port_id = port_id;
+
+ ret = afe_apr_send_pkt(afe, &start, port);
+ if (ret)
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6afe_port_start);
+
+/**
+ * q6afe_port_get_from_id() - Get port instance from a port id
+ *
+ * @dev: Pointer to afe child device.
+ * @id: port id
+ *
+ * Return: Will be an error pointer on error or a valid afe port
+ * on success.
+ */
+struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
+{
+ int port_id;
+ struct q6afe *afe = dev_get_drvdata(dev->parent);
+ struct q6afe_port *port;
+ unsigned long flags;
+ int cfg_type;
+
+ if (id < 0 || id > AFE_PORT_MAX) {
+ dev_err(dev, "AFE port token[%d] invalid!\n", id);
+ return ERR_PTR(-EINVAL);
+ }
+
+ port_id = port_maps[id].port_id;
+
+ switch (port_id) {
+ case AFE_PORT_ID_MULTICHAN_HDMI_RX:
+ cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
+ break;
+ default:
+ dev_err(dev, "Invalid port id 0x%x\n", port_id);
+ return ERR_PTR(-EINVAL);
+ }
+
+ port = kzalloc(sizeof(*port), GFP_KERNEL);
+ if (!port)
+ return ERR_PTR(-ENOMEM);
+
+ init_waitqueue_head(&port->wait);
+
+ port->token = id;
+ port->id = port_id;
+ port->afe = afe;
+ port->cfg_type = cfg_type;
+
+ spin_lock_irqsave(&afe->port_list_lock, flags);
+ list_add_tail(&port->node, &afe->port_list);
+ spin_unlock_irqrestore(&afe->port_list_lock, flags);
+
+ return port;
+
+}
+EXPORT_SYMBOL_GPL(q6afe_port_get_from_id);
+
+/**
+ * q6afe_port_put() - Release port reference
+ *
+ * @port: Instance of port to put
+ */
+void q6afe_port_put(struct q6afe_port *port)
+{
+ struct q6afe *afe = port->afe;
+ unsigned long flags;
+
+ spin_lock_irqsave(&afe->port_list_lock, flags);
+ list_del(&port->node);
+ spin_unlock_irqrestore(&afe->port_list_lock, flags);
+ kfree(port);
+}
+EXPORT_SYMBOL_GPL(q6afe_port_put);
+
+static int q6afe_probe(struct apr_device *adev)
+{
+ struct q6afe *afe;
+ struct device *dev = &adev->dev;
+ struct device_node *dais_np;
+
+ afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL);
+ if (!afe)
+ return -ENOMEM;
+
+ q6core_get_svc_api_info(adev->svc_id, &afe->ainfo);
+ afe->apr = adev;
+ mutex_init(&afe->lock);
+ afe->dev = dev;
+ INIT_LIST_HEAD(&afe->port_list);
+ spin_lock_init(&afe->port_list_lock);
+
+ dev_set_drvdata(dev, afe);
+
+ dais_np = of_get_child_by_name(dev->of_node, "dais");
+ if (dais_np) {
+ afe->pdev_dais = of_platform_device_create(dais_np,
+ "q6afe-dai", dev);
+ of_node_put(dais_np);
+ }
+
+ return 0;
+}
+
+static int q6afe_remove(struct apr_device *adev)
+{
+ struct q6afe *afe = dev_get_drvdata(&adev->dev);
+
+ if (afe->pdev_dais)
+ of_platform_device_destroy(&afe->pdev_dais->dev, NULL);
+
+ return 0;
+}
+
+static const struct of_device_id q6afe_device_id[] = {
+ { .compatible = "qcom,q6afe" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6afe_device_id);
+
+static struct apr_driver qcom_q6afe_driver = {
+ .probe = q6afe_probe,
+ .remove = q6afe_remove,
+ .callback = q6afe_callback,
+ .driver = {
+ .name = "qcom-q6afe",
+ .of_match_table = of_match_ptr(q6afe_device_id),
+
+ },
+};
+
+module_apr_driver(qcom_q6afe_driver);
+MODULE_DESCRIPTION("Q6 Audio Front End");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
new file mode 100644
index 000000000000..63250f118df8
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -0,0 +1,35 @@
+// SPDX-License-Identifier: GPL-2.0
+
+#ifndef __Q6AFE_H__
+#define __Q6AFE_H__
+
+#include <dt-bindings/sound/qcom,q6afe.h>
+
+#define AFE_PORT_MAX 48
+
+#define MSM_AFE_PORT_TYPE_RX 0
+#define MSM_AFE_PORT_TYPE_TX 1
+#define AFE_MAX_PORTS AFE_PORT_MAX
+
+struct q6afe_hdmi_cfg {
+ u16 datatype;
+ u16 channel_allocation;
+ u32 sample_rate;
+ u16 bit_width;
+};
+
+struct q6afe_port_config {
+ struct q6afe_hdmi_cfg hdmi;
+};
+
+struct q6afe_port;
+
+struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id);
+int q6afe_port_start(struct q6afe_port *port);
+int q6afe_port_stop(struct q6afe_port *port);
+void q6afe_port_put(struct q6afe_port *port);
+int q6afe_get_port_id(int index);
+void q6afe_hdmi_port_prepare(struct q6afe_port *port,
+ struct q6afe_hdmi_cfg *cfg);
+
+#endif /* __Q6AFE_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* Re: [alsa-devel] [PATCH v4 07/24] ASoC: qdsp6: q6afe: Add q6afe driver
2018-03-10 2:24 ` [PATCH v4 07/24] ASoC: qdsp6: q6afe: Add q6afe driver srinivas.kandagatla
@ 2018-03-13 11:29 ` Rohit Kumar
2018-03-13 13:51 ` Srinivas Kandagatla
0 siblings, 1 reply; 33+ messages in thread
From: Rohit Kumar @ 2018-03-13 11:29 UTC (permalink / raw)
To: srinivas.kandagatla, andy.gross, broonie, linux-arm-msm,
alsa-devel, robh+dt
Cc: mark.rutland, devicetree, bgoswami, rohkumar, gregkh, plai,
lgirdwood, tiwai, david.brown, linux-arm-kernel, spatakok,
linux-kernel
On 3/10/2018 7:54 AM, srinivas.kandagatla@linaro.org wrote:
> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
>
> This patch adds support to Q6AFE (Audio Front End) module on Q6DSP.
>
[..]
> + if (res->status) {
> + dev_err(afe->dev, "cmd = 0x%x returned error = 0x%x\n",
> + res->opcode, res->status);
> + }
> + switch (res->opcode) {
> + case AFE_PORT_CMD_SET_PARAM_V2:
> + case AFE_PORT_CMD_DEVICE_STOP:
> + case AFE_PORT_CMD_DEVICE_START:
case AFE_SVC_CMD_SET_PARAM: needs to be added
> + port = afe_find_port(afe, data->token);
> + if (port) {
> + port->result = *res;
> + wake_up(&port->wait);
> + }
[..]
> +
> +static int q6afe_port_set_param_v2(struct q6afe_port *port, void *data,
> + int param_id, int psize)
> +{
> + struct apr_hdr *hdr;
> + struct afe_port_cmd_set_param_v2 *param;
> + struct afe_port_param_data_v2 *pdata;
> + struct q6afe *afe = port->afe;
> + u16 port_id = port->id;
> + int ret;
> +
> + hdr = data;
> + param = data + sizeof(*hdr);
> + pdata = data + sizeof(*hdr) + sizeof(*param);
> +
> + hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
> + APR_HDR_LEN(APR_HDR_SIZE),
> + APR_PKT_VER);
> + hdr->pkt_size = sizeof(*hdr) + sizeof(*param) +
> + sizeof(*pdata) + psize;
> + hdr->src_port = 0;
> + hdr->dest_port = 0;
> + hdr->token = port->token;
> + hdr->opcode = AFE_PORT_CMD_SET_PARAM_V2;
> + param->port_id = port_id;
> + param->payload_size = sizeof(*pdata) + psize;
> + param->payload_address_lsw = 0x00;
> + param->payload_address_msw = 0x00;
> + param->mem_map_handle = 0x00;
> + pdata->module_id = AFE_MODULE_AUDIO_DEV_INTERFACE;
> + pdata->param_id = param_id;
> + pdata->param_size = psize;
> +
> + ret = afe_apr_send_pkt(afe, data, port);
> + if (ret)
> + dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
we should add %s to distinguish different functions with similar error
logs. Also, q6afe_port_set_param_v2() can be called for different
purpose. AFE enable for port 0x%x failed is suitable only for port
start failure. Error message needs to be updated here.
> + port_id, ret);
> +
> +
> + return ret;
> +}
> +
^ permalink raw reply [flat|nested] 33+ messages in thread* Re: [alsa-devel] [PATCH v4 07/24] ASoC: qdsp6: q6afe: Add q6afe driver
2018-03-13 11:29 ` [alsa-devel] " Rohit Kumar
@ 2018-03-13 13:51 ` Srinivas Kandagatla
0 siblings, 0 replies; 33+ messages in thread
From: Srinivas Kandagatla @ 2018-03-13 13:51 UTC (permalink / raw)
To: Rohit Kumar, andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: mark.rutland, devicetree, bgoswami, rohkumar, gregkh, plai,
lgirdwood, tiwai, david.brown, linux-arm-kernel, spatakok,
linux-kernel
Thanks Rohit for the review,
On 13/03/18 11:29, Rohit Kumar wrote:
>
>
> On 3/10/2018 7:54 AM, srinivas.kandagatla@linaro.org wrote:
>> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
>>
>> This patch adds support to Q6AFE (Audio Front End) module on Q6DSP.
>>
> [..]
>> + if (res->status) {
>> + dev_err(afe->dev, "cmd = 0x%x returned error = 0x%x\n",
>> + res->opcode, res->status);
>> + }
>> + switch (res->opcode) {
>> + case AFE_PORT_CMD_SET_PARAM_V2:
>> + case AFE_PORT_CMD_DEVICE_STOP:
>> + case AFE_PORT_CMD_DEVICE_START:
>
> case AFE_SVC_CMD_SET_PARAM: needs to be added
thanks for spotting this, Will add this back
>> + port = afe_find_port(afe, data->token);
>> + if (port) {
>> + port->result = *res;
>> + wake_up(&port->wait);
>> + }
> [..]
>> +
>> +static int q6afe_port_set_param_v2(struct q6afe_port *port, void *data,
>> + int param_id, int psize)
>> +{
...
>> + ret = afe_apr_send_pkt(afe, data, port);
>> + if (ret)
>> + dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
>
> we should add %s to distinguish different functions with similar error
> logs. Also, q6afe_port_set_param_v2() can be called for different
> purpose. AFE enable for port 0x%x failed is suitable only for port
> start failure. Error message needs to be updated here.
>
q6afe_callback already has a extra level of error message which shows
the opcode and error code in failure cases.
thanks,
srini
>> + port_id, ret);
>> +
>> +
>> + return ret;
>> +}
>> +
^ permalink raw reply [flat|nested] 33+ messages in thread
* [PATCH v4 08/24] ASoC: qdsp6: qdafe: Add SLIMBus port Support
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (6 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 07/24] ASoC: qdsp6: q6afe: Add q6afe driver srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 09/24] ASoC: qdsp6: q6afe: Add support to MI2S ports srinivas.kandagatla
` (15 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to 6 SLIMBus AFE ports, which are used as
backend dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/qdsp6/q6afe.c | 129 +++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 14 +++++
2 files changed, 143 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index fd76f7296f36..de50ddd9faa4 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -26,9 +26,45 @@
#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210
#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C
+#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235
+
+#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212
+
/* Port IDs */
#define AFE_API_VERSION_HDMI_CONFIG 0x1
#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
+
+#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
+
+/* SLIMbus Rx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000
+/* SLIMbus Tx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX 0x4001
+/* SLIMbus Rx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX 0x4002
+/* SLIMbus Tx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX 0x4003
+/* SLIMbus Rx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX 0x4004
+/* SLIMbus Tx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX 0x4005
+/* SLIMbus Rx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX 0x4006
+/* SLIMbus Tx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX 0x4007
+/* SLIMbus Rx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX 0x4008
+/* SLIMbus Tx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX 0x4009
+/* SLIMbus Rx port on channel 5. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX 0x400a
+/* SLIMbus Tx port on channel 5. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_TX 0x400b
+/* SLIMbus Rx port on channel 6. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX 0x400c
+/* SLIMbus Tx port on channel 6. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX 0x400d
+
#define TIMEOUT_MS 1000
#define AFE_CMD_RESP_AVAIL 0
#define AFE_CMD_RESP_NONE 1
@@ -81,8 +117,53 @@ struct afe_param_id_hdmi_multi_chan_audio_cfg {
u16 reserved;
} __packed;
+struct afe_param_id_slimbus_cfg {
+ u32 sb_cfg_minor_version;
+/* Minor version used for tracking the version of the SLIMBUS
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_SLIMBUS_CONFIG
+ */
+
+ u16 slimbus_dev_id;
+/* SLIMbus hardware device ID, which is required to handle
+ * multiple SLIMbus hardware blocks.
+ * Supported values: - #AFE_SLIMBUS_DEVICE_1 - #AFE_SLIMBUS_DEVICE_2
+ */
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+ u16 data_format;
+/* Data format supported by the SLIMbus hardware. The default is
+ * 0 (#AFE_SB_DATA_FORMAT_NOT_INDICATED), which indicates the
+ * hardware does not perform any format conversions before the data
+ * transfer.
+ */
+ u16 num_channels;
+/* Number of channels.
+ * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+ */
+ u8 shared_ch_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+/* Mapping of shared channel IDs (128 to 255) to which the
+ * master port is to be connected.
+ * Shared_channel_mapping[i] represents the shared channel assigned
+ * for audio channel i in multichannel audio data.
+ */
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_192K
+ */
+} __packed;
+
+
union afe_port_config {
struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
+ struct afe_param_id_slimbus_cfg slim_cfg;
} __packed;
struct q6afe_port {
@@ -117,6 +198,20 @@ struct afe_port_map {
static struct afe_port_map port_maps[AFE_PORT_MAX] = {
[HDMI_RX] = { AFE_PORT_ID_MULTICHAN_HDMI_RX, HDMI_RX, 1, 1},
+ [SLIMBUS_0_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX,
+ SLIMBUS_0_RX, 1, 1},
+ [SLIMBUS_1_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX,
+ SLIMBUS_1_RX, 1, 1},
+ [SLIMBUS_2_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX,
+ SLIMBUS_2_RX, 1, 1},
+ [SLIMBUS_3_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX,
+ SLIMBUS_3_RX, 1, 1},
+ [SLIMBUS_4_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX,
+ SLIMBUS_4_RX, 1, 1},
+ [SLIMBUS_5_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX,
+ SLIMBUS_5_RX, 1, 1},
+ [SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
+ SLIMBUS_6_RX, 1, 1},
};
static struct q6afe_port *afe_find_port(struct q6afe *afe, int token)
@@ -307,6 +402,31 @@ int q6afe_port_stop(struct q6afe_port *port)
}
EXPORT_SYMBOL_GPL(q6afe_port_stop);
+/**
+ * q6afe_slim_port_prepare() - Prepare slim afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: SLIM configuration for the afe port
+ *
+ */
+void q6afe_slim_port_prepare(struct q6afe_port *port,
+ struct q6afe_slim_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+
+ pcfg->slim_cfg.sb_cfg_minor_version = AFE_API_VERSION_SLIMBUS_CONFIG;
+ pcfg->slim_cfg.sample_rate = cfg->sample_rate;
+ pcfg->slim_cfg.bit_width = cfg->bit_width;
+ pcfg->slim_cfg.num_channels = cfg->num_channels;
+ pcfg->slim_cfg.data_format = cfg->data_format;
+ pcfg->slim_cfg.shared_ch_mapping[0] = cfg->ch_mapping[0];
+ pcfg->slim_cfg.shared_ch_mapping[1] = cfg->ch_mapping[1];
+ pcfg->slim_cfg.shared_ch_mapping[2] = cfg->ch_mapping[2];
+ pcfg->slim_cfg.shared_ch_mapping[3] = cfg->ch_mapping[3];
+
+}
+EXPORT_SYMBOL_GPL(q6afe_slim_port_prepare);
+
/**
* q6afe_hdmi_port_prepare() - Prepare hdmi afe port.
*
@@ -400,6 +520,15 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
case AFE_PORT_ID_MULTICHAN_HDMI_RX:
cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
break;
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX:
+ case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX:
+ cfg_type = AFE_PARAM_ID_SLIMBUS_CONFIG;
+ break;
default:
dev_err(dev, "Invalid port id 0x%x\n", port_id);
return ERR_PTR(-EINVAL);
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index 63250f118df8..155d6e47d7ce 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -11,6 +11,9 @@
#define MSM_AFE_PORT_TYPE_TX 1
#define AFE_MAX_PORTS AFE_PORT_MAX
+#define AFE_MAX_CHAN_COUNT 8
+#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
+
struct q6afe_hdmi_cfg {
u16 datatype;
u16 channel_allocation;
@@ -18,8 +21,17 @@ struct q6afe_hdmi_cfg {
u16 bit_width;
};
+struct q6afe_slim_cfg {
+ u32 sample_rate;
+ u16 bit_width;
+ u16 data_format;
+ u16 num_channels;
+ u8 ch_mapping[AFE_MAX_CHAN_COUNT];
+};
+
struct q6afe_port_config {
struct q6afe_hdmi_cfg hdmi;
+ struct q6afe_slim_cfg slim;
};
struct q6afe_port;
@@ -31,5 +43,7 @@ void q6afe_port_put(struct q6afe_port *port);
int q6afe_get_port_id(int index);
void q6afe_hdmi_port_prepare(struct q6afe_port *port,
struct q6afe_hdmi_cfg *cfg);
+void q6afe_slim_port_prepare(struct q6afe_port *port,
+ struct q6afe_slim_cfg *cfg);
#endif /* __Q6AFE_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 09/24] ASoC: qdsp6: q6afe: Add support to MI2S ports
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (7 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 08/24] ASoC: qdsp6: qdafe: Add SLIMBus port Support srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 10/24] ASoC: qdsp6: q6afe: Add support to MI2S sysclks srinivas.kandagatla
` (14 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to 4 MI2S ports on LPASS.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/qdsp6/q6afe.c | 228 +++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 13 +++
2 files changed, 241 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index de50ddd9faa4..2da5e3b14e38 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -14,6 +14,10 @@
#include <linux/spinlock.h>
#include <linux/delay.h>
#include <linux/soc/qcom/apr.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
#include "q6dsp-errno.h"
#include "q6core.h"
#include "q6afe.h"
@@ -28,7 +32,36 @@
#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235
+#define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238
+#define AFE_PARAM_ID_INTERNAL_DIGITAL_CDC_CLK_CONFIG 0x00010239
+
#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212
+#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D
+
+/* I2S config specific */
+#define AFE_API_VERSION_I2S_CONFIG 0x1
+#define AFE_PORT_I2S_SD0 0x1
+#define AFE_PORT_I2S_SD1 0x2
+#define AFE_PORT_I2S_SD2 0x3
+#define AFE_PORT_I2S_SD3 0x4
+#define AFE_PORT_I2S_SD0_MASK BIT(0x1)
+#define AFE_PORT_I2S_SD1_MASK BIT(0x2)
+#define AFE_PORT_I2S_SD2_MASK BIT(0x3)
+#define AFE_PORT_I2S_SD3_MASK BIT(0x4)
+#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1)
+#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3)
+#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1)
+#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1)
+#define AFE_PORT_I2S_QUAD01 0x5
+#define AFE_PORT_I2S_QUAD23 0x6
+#define AFE_PORT_I2S_6CHS 0x7
+#define AFE_PORT_I2S_8CHS 0x8
+#define AFE_PORT_I2S_MONO 0x0
+#define AFE_PORT_I2S_STEREO 0x1
+#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0
+#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1
+#define AFE_LINEAR_PCM_DATA 0x0
+
/* Port IDs */
#define AFE_API_VERSION_HDMI_CONFIG 0x1
@@ -64,6 +97,19 @@
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX 0x400c
/* SLIMbus Tx port on channel 6. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX 0x400d
+#define AFE_PORT_ID_PRIMARY_MI2S_RX 0x1000
+#define AFE_PORT_ID_PRIMARY_MI2S_TX 0x1001
+#define AFE_PORT_ID_SECONDARY_MI2S_RX 0x1002
+#define AFE_PORT_ID_SECONDARY_MI2S_TX 0x1003
+#define AFE_PORT_ID_TERTIARY_MI2S_RX 0x1004
+#define AFE_PORT_ID_TERTIARY_MI2S_TX 0x1005
+#define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006
+#define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007
+
+#define Q6AFE_LPASS_MODE_CLK1_VALID 1
+#define Q6AFE_LPASS_MODE_CLK2_VALID 2
+#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
+#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
#define TIMEOUT_MS 1000
#define AFE_CMD_RESP_AVAIL 0
@@ -160,10 +206,21 @@ struct afe_param_id_slimbus_cfg {
*/
} __packed;
+struct afe_param_id_i2s_cfg {
+ u32 i2s_cfg_minor_version;
+ u16 bit_width;
+ u16 channel_mode;
+ u16 mono_stereo;
+ u16 ws_src;
+ u32 sample_rate;
+ u16 data_format;
+ u16 reserved;
+} __packed;
union afe_port_config {
struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
struct afe_param_id_slimbus_cfg slim_cfg;
+ struct afe_param_id_i2s_cfg i2s_cfg;
} __packed;
struct q6afe_port {
@@ -210,6 +267,22 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
SLIMBUS_4_RX, 1, 1},
[SLIMBUS_5_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX,
SLIMBUS_5_RX, 1, 1},
+ [QUATERNARY_MI2S_RX] = { AFE_PORT_ID_QUATERNARY_MI2S_RX,
+ QUATERNARY_MI2S_RX, 1, 1},
+ [QUATERNARY_MI2S_TX] = { AFE_PORT_ID_QUATERNARY_MI2S_TX,
+ QUATERNARY_MI2S_TX, 0, 1},
+ [SECONDARY_MI2S_RX] = { AFE_PORT_ID_SECONDARY_MI2S_RX,
+ SECONDARY_MI2S_RX, 1, 1},
+ [SECONDARY_MI2S_TX] = { AFE_PORT_ID_SECONDARY_MI2S_TX,
+ SECONDARY_MI2S_TX, 0, 1},
+ [TERTIARY_MI2S_RX] = { AFE_PORT_ID_TERTIARY_MI2S_RX,
+ TERTIARY_MI2S_RX, 1, 1},
+ [TERTIARY_MI2S_TX] = { AFE_PORT_ID_TERTIARY_MI2S_TX,
+ TERTIARY_MI2S_TX, 0, 1},
+ [PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX,
+ PRIMARY_MI2S_RX, 1, 1},
+ [PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX,
+ PRIMARY_MI2S_RX, 0, 1},
[SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
SLIMBUS_6_RX, 1, 1},
};
@@ -448,6 +521,150 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
}
EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
+/**
+ * q6afe_i2s_port_prepare() - Prepare i2s afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: I2S configuration for the afe port
+ *
+ Return: Will be an negative on error and zero on success.
+ */
+int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+ struct device *dev = port->afe->dev;
+ int num_sd_lines;
+
+ pcfg->i2s_cfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ pcfg->i2s_cfg.sample_rate = cfg->sample_rate;
+ pcfg->i2s_cfg.bit_width = cfg->bit_width;
+ pcfg->i2s_cfg.data_format = AFE_LINEAR_PCM_DATA;
+
+ switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* CPU is slave */
+ pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL;
+ break;
+ default:
+ break;
+ }
+
+ num_sd_lines = hweight_long(cfg->sd_line_mask);
+
+ switch (num_sd_lines) {
+ case 0:
+ dev_err(dev, "no line is assigned\n");
+ return -EINVAL;
+ case 1:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD0;
+ break;
+ case AFE_PORT_I2S_SD1_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD1;
+ break;
+ case AFE_PORT_I2S_SD2_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD2;
+ break;
+ case AFE_PORT_I2S_SD3_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD3;
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ case 2:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_1_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_QUAD01;
+ break;
+ case AFE_PORT_I2S_SD2_3_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_QUAD23;
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ case 3:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_1_2_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_6CHS;
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ case 4:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_1_2_3_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_8CHS;
+
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+
+ switch (cfg->num_channels) {
+ case 1:
+ case 2:
+ switch (pcfg->i2s_cfg.channel_mode) {
+ case AFE_PORT_I2S_QUAD01:
+ case AFE_PORT_I2S_6CHS:
+ case AFE_PORT_I2S_8CHS:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD0;
+ break;
+ case AFE_PORT_I2S_QUAD23:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD2;
+ break;
+ }
+
+ if (cfg->num_channels == 2)
+ pcfg->i2s_cfg.mono_stereo = AFE_PORT_I2S_STEREO;
+ else
+ pcfg->i2s_cfg.mono_stereo = AFE_PORT_I2S_MONO;
+
+ break;
+ case 3:
+ case 4:
+ if (pcfg->i2s_cfg.channel_mode < AFE_PORT_I2S_QUAD01) {
+ dev_err(dev, "Invalid Channel mode\n");
+ return -EINVAL;
+ }
+ break;
+ case 5:
+ case 6:
+ if (pcfg->i2s_cfg.channel_mode < AFE_PORT_I2S_6CHS) {
+ dev_err(dev, "Invalid Channel mode\n");
+ return -EINVAL;
+ }
+ break;
+ case 7:
+ case 8:
+ if (pcfg->i2s_cfg.channel_mode < AFE_PORT_I2S_8CHS) {
+ dev_err(dev, "Invalid Channel mode\n");
+ return -EINVAL;
+ }
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6afe_i2s_port_prepare);
+
/**
* q6afe_port_start() - Start a afe port
*
@@ -529,6 +746,17 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX:
cfg_type = AFE_PARAM_ID_SLIMBUS_CONFIG;
break;
+
+ case AFE_PORT_ID_PRIMARY_MI2S_RX:
+ case AFE_PORT_ID_PRIMARY_MI2S_TX:
+ case AFE_PORT_ID_SECONDARY_MI2S_RX:
+ case AFE_PORT_ID_SECONDARY_MI2S_TX:
+ case AFE_PORT_ID_TERTIARY_MI2S_RX:
+ case AFE_PORT_ID_TERTIARY_MI2S_TX:
+ case AFE_PORT_ID_QUATERNARY_MI2S_RX:
+ case AFE_PORT_ID_QUATERNARY_MI2S_TX:
+ cfg_type = AFE_PARAM_ID_I2S_CONFIG;
+ break;
default:
dev_err(dev, "Invalid port id 0x%x\n", port_id);
return ERR_PTR(-EINVAL);
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index 155d6e47d7ce..05c61014abea 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -11,6 +11,8 @@
#define MSM_AFE_PORT_TYPE_TX 1
#define AFE_MAX_PORTS AFE_PORT_MAX
+#define Q6AFE_MAX_MI2S_LINES 4
+
#define AFE_MAX_CHAN_COUNT 8
#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
@@ -29,9 +31,19 @@ struct q6afe_slim_cfg {
u8 ch_mapping[AFE_MAX_CHAN_COUNT];
};
+struct q6afe_i2s_cfg {
+ u32 sample_rate;
+ u16 bit_width;
+ u16 data_format;
+ u16 num_channels;
+ u32 sd_line_mask;
+ int fmt;
+};
+
struct q6afe_port_config {
struct q6afe_hdmi_cfg hdmi;
struct q6afe_slim_cfg slim;
+ struct q6afe_i2s_cfg i2s_cfg;
};
struct q6afe_port;
@@ -45,5 +57,6 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
struct q6afe_hdmi_cfg *cfg);
void q6afe_slim_port_prepare(struct q6afe_port *port,
struct q6afe_slim_cfg *cfg);
+int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg);
#endif /* __Q6AFE_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 10/24] ASoC: qdsp6: q6afe: Add support to MI2S sysclks
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (8 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 09/24] ASoC: qdsp6: q6afe: Add support to MI2S ports srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 11/24] ASoC: qdsp6: q6adm: Add q6adm driver srinivas.kandagatla
` (13 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to LPASS Bit clock, LPASS Digital
core clock and OSR clock. These clocks are required for both
MI2S and PCM setup.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/qdsp6/q6afe.c | 211 ++++++++++++++++++++++++++++++++++++++++++-
sound/soc/qcom/qdsp6/q6afe.h | 131 +++++++++++++++++++++++++++
2 files changed, 340 insertions(+), 2 deletions(-)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 2da5e3b14e38..40ba53d13e98 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -26,6 +26,7 @@
#define AFE_PORT_CMD_DEVICE_START 0x000100E5
#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6
#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF
+#define AFE_SVC_CMD_SET_PARAM 0x000100f3
#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106
#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210
#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C
@@ -68,6 +69,11 @@
#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
+/* Clock set API version */
+#define AFE_API_VERSION_CLOCK_SET 1
+#define Q6AFE_LPASS_CLK_CONFIG_API_VERSION 0x1
+#define AFE_MODULE_CLOCK_SET 0x0001028F
+#define AFE_PARAM_ID_CLOCK_SET 0x00010290
/* SLIMbus Rx port on channel 0. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000
@@ -146,6 +152,13 @@ struct afe_port_param_data_v2 {
u16 reserved;
} __packed;
+struct afe_svc_cmd_set_param {
+ uint32_t payload_size;
+ uint32_t payload_address_lsw;
+ uint32_t payload_address_msw;
+ uint32_t mem_map_handle;
+} __packed;
+
struct afe_port_cmd_set_param_v2 {
u16 port_id;
u16 payload_size;
@@ -206,6 +219,23 @@ struct afe_param_id_slimbus_cfg {
*/
} __packed;
+struct afe_clk_cfg {
+ u32 i2s_cfg_minor_version;
+ u32 clk_val1;
+ u32 clk_val2;
+ u16 clk_src;
+ u16 clk_root;
+ u16 clk_set_mode;
+ u16 reserved;
+} __packed;
+
+struct afe_digital_clk_cfg {
+ u32 i2s_cfg_minor_version;
+ u32 clk_val;
+ u16 clk_root;
+ u16 reserved;
+} __packed;
+
struct afe_param_id_i2s_cfg {
u32 i2s_cfg_minor_version;
u16 bit_width;
@@ -223,9 +253,43 @@ union afe_port_config {
struct afe_param_id_i2s_cfg i2s_cfg;
} __packed;
+
+struct afe_clk_set {
+ uint32_t clk_set_minor_version;
+ uint32_t clk_id;
+ uint32_t clk_freq_in_hz;
+ uint16_t clk_attri;
+ uint16_t clk_root;
+ uint32_t enable;
+};
+
+struct afe_lpass_clk_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_clk_cfg clk_cfg;
+} __packed;
+
+struct afe_lpass_clk_config_command_v2 {
+ struct apr_hdr hdr;
+ struct afe_svc_cmd_set_param param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_clk_set clk_cfg;
+} __packed;
+
+
+
+struct afe_lpass_digital_clk_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ struct afe_digital_clk_cfg clk_cfg;
+} __packed;
+
struct q6afe_port {
wait_queue_head_t wait;
union afe_port_config port_cfg;
+ struct aprv2_ibasic_rsp_result_t result;
int token;
int id;
int cfg_type;
@@ -393,6 +457,46 @@ static int afe_apr_send_pkt(struct q6afe *afe, void *data,
return ret;
}
+static int q6afe_port_set_param(struct q6afe_port *port, void *data,
+ int param_id, int module_id, int psize)
+{
+ struct apr_hdr *hdr;
+ struct afe_svc_cmd_set_param *param;
+ struct afe_port_param_data_v2 *pdata;
+ struct q6afe *afe = port->afe;
+ u16 port_id = port->id;
+ int ret;
+
+ hdr = data;
+ param = data + sizeof(*hdr);
+ pdata = data + sizeof(*hdr) + sizeof(*param);
+
+ hdr->hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ hdr->pkt_size = sizeof(*hdr) + sizeof(*param) +
+ sizeof(*pdata) + psize;
+ hdr->src_port = 0;
+ hdr->dest_port = 0;
+ hdr->token = port->token;
+ hdr->opcode = AFE_SVC_CMD_SET_PARAM;
+
+ param->payload_size = sizeof(*pdata) + psize;
+ param->payload_address_lsw = 0x00;
+ param->payload_address_msw = 0x00;
+ param->mem_map_handle = 0x00;
+ pdata->module_id = module_id;
+ pdata->param_id = param_id;
+ pdata->param_size = psize;
+
+ ret = afe_apr_send_pkt(afe, data, port);
+ if (ret)
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+
+ return ret;
+}
+
static int q6afe_port_set_param_v2(struct q6afe_port *port, void *data,
int param_id, int psize)
{
@@ -434,6 +538,110 @@ static int q6afe_port_set_param_v2(struct q6afe_port *port, void *data,
return ret;
}
+static int q6afe_set_lpass_clock(struct q6afe_port *port,
+ struct afe_clk_cfg *cfg)
+{
+ struct afe_lpass_clk_config_command clk_cfg = {0};
+ int param_id = AFE_PARAM_ID_LPAIF_CLK_CONFIG;
+ struct q6afe *afe = port->afe;
+
+ if (!cfg) {
+ dev_err(afe->dev, "clock cfg is NULL\n");
+ return -EINVAL;
+ }
+
+ clk_cfg.clk_cfg = *cfg;
+
+ return q6afe_port_set_param_v2(port, &clk_cfg, param_id, sizeof(*cfg));
+}
+
+static int q6afe_set_lpass_clock_v2(struct q6afe_port *port,
+ struct afe_clk_set *cfg)
+{
+ struct afe_lpass_clk_config_command_v2 clk_cfg = {0};
+ int param_id = AFE_PARAM_ID_CLOCK_SET;
+ int module_id = AFE_MODULE_CLOCK_SET;
+ struct q6afe *afe = port->afe;
+
+ if (!cfg) {
+ dev_err(afe->dev, "clock cfg is NULL\n");
+ return -EINVAL;
+ }
+
+ clk_cfg.clk_cfg = *cfg;
+
+ return q6afe_port_set_param(port, &clk_cfg, param_id,
+ module_id, sizeof(*cfg));
+}
+
+static int q6afe_set_digital_codec_core_clock(struct q6afe_port *port,
+ struct afe_digital_clk_cfg *cfg)
+{
+ struct afe_lpass_digital_clk_config_command clk_cfg = {0};
+ int param_id = AFE_PARAM_ID_INTERNAL_DIGITAL_CDC_CLK_CONFIG;
+ struct q6afe *afe = port->afe;
+
+ if (!cfg) {
+ dev_err(afe->dev, "clock cfg is NULL\n");
+ return -EINVAL;
+ }
+
+ clk_cfg.clk_cfg = *cfg;
+
+ return q6afe_port_set_param_v2(port, &clk_cfg, param_id, sizeof(*cfg));
+}
+
+int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
+ int clk_src, int clk_root,
+ unsigned int freq, int dir)
+{
+ struct afe_clk_cfg ccfg = {0,};
+ struct afe_clk_set cset = {0,};
+ struct afe_digital_clk_cfg dcfg = {0,};
+ int ret;
+
+ switch (clk_id) {
+ case LPAIF_DIG_CLK:
+ dcfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ dcfg.clk_val = freq;
+ dcfg.clk_root = clk_root;
+ ret = q6afe_set_digital_codec_core_clock(port, &dcfg);
+ break;
+ case LPAIF_BIT_CLK:
+ ccfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ ccfg.clk_val1 = freq;
+ ccfg.clk_src = clk_src;
+ ccfg.clk_root = clk_root;
+ ccfg.clk_set_mode = Q6AFE_LPASS_MODE_CLK1_VALID;
+ ret = q6afe_set_lpass_clock(port, &ccfg);
+ break;
+
+ case LPAIF_OSR_CLK:
+ ccfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ ccfg.clk_val2 = freq;
+ ccfg.clk_src = clk_src;
+ ccfg.clk_root = clk_root;
+ ccfg.clk_set_mode = Q6AFE_LPASS_MODE_CLK2_VALID;
+ ret = q6afe_set_lpass_clock(port, &ccfg);
+ break;
+ case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1:
+ cset.clk_set_minor_version = AFE_API_VERSION_CLOCK_SET;
+ cset.clk_id = clk_id;
+ cset.clk_freq_in_hz = freq;
+ cset.clk_attri = clk_src;
+ cset.clk_root = clk_root;
+ cset.enable = !!freq;
+ ret = q6afe_set_lpass_clock_v2(port, &cset);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6afe_port_set_sysclk);
+
/**
* q6afe_port_stop() - Stop a afe port
*
@@ -526,8 +734,7 @@ EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
*
* @port: Instance of afe port
* @cfg: I2S configuration for the afe port
- *
- Return: Will be an negative on error and zero on success.
+ * Return: Will be an negative on error and zero on success.
*/
int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg)
{
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index 05c61014abea..f21c89a25c29 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -16,6 +16,134 @@
#define AFE_MAX_CHAN_COUNT 8
#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
+#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
+#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
+
+#define LPAIF_DIG_CLK 1
+#define LPAIF_BIT_CLK 2
+#define LPAIF_OSR_CLK 3
+
+/* Clock ID for Primary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT 0x100
+/* Clock ID for Primary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_EBIT 0x101
+/* Clock ID for Secondary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT 0x102
+/* Clock ID for Secondary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_EBIT 0x103
+/* Clock ID for Tertiary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_IBIT 0x104
+/* Clock ID for Tertiary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_EBIT 0x105
+/* Clock ID for Quartnery I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT 0x106
+/* Clock ID for Quartnery I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_EBIT 0x107
+/* Clock ID for Speaker I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_IBIT 0x108
+/* Clock ID for Speaker I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_EBIT 0x109
+/* Clock ID for Speaker I2S OSR */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_OSR 0x10A
+
+/* Clock ID for QUINARY I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_IBIT 0x10B
+/* Clock ID for QUINARY I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_EBIT 0x10C
+/* Clock ID for SENARY I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_IBIT 0x10D
+/* Clock ID for SENARY I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_EBIT 0x10E
+/* Clock ID for INT0 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT0_MI2S_IBIT 0x10F
+/* Clock ID for INT1 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT1_MI2S_IBIT 0x110
+/* Clock ID for INT2 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT2_MI2S_IBIT 0x111
+/* Clock ID for INT3 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT3_MI2S_IBIT 0x112
+/* Clock ID for INT4 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT4_MI2S_IBIT 0x113
+/* Clock ID for INT5 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT5_MI2S_IBIT 0x114
+/* Clock ID for INT6 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT6_MI2S_IBIT 0x115
+
+/* Clock ID for QUINARY MI2S OSR CLK */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR 0x116
+
+/* Clock ID for Primary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT 0x200
+/* Clock ID for Primary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_EBIT 0x201
+/* Clock ID for Secondary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_IBIT 0x202
+/* Clock ID for Secondary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_EBIT 0x203
+/* Clock ID for Tertiary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_IBIT 0x204
+/* Clock ID for Tertiary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_EBIT 0x205
+/* Clock ID for Quartery PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT 0x206
+/* Clock ID for Quartery PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_EBIT 0x207
+/* Clock ID for Quinary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_PCM_IBIT 0x208
+/* Clock ID for Quinary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_PCM_EBIT 0x209
+/* Clock ID for QUINARY PCM OSR */
+#define Q6AFE_LPASS_CLK_ID_QUI_PCM_OSR 0x20A
+
+/** Clock ID for Primary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT 0x200
+/** Clock ID for Primary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_EBIT 0x201
+/** Clock ID for Secondary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_IBIT 0x202
+/** Clock ID for Secondary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_EBIT 0x203
+/** Clock ID for Tertiary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_IBIT 0x204
+/** Clock ID for Tertiary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_EBIT 0x205
+/** Clock ID for Quartery TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT 0x206
+/** Clock ID for Quartery TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_EBIT 0x207
+/** Clock ID for Quinary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_IBIT 0x208
+/** Clock ID for Quinary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT 0x209
+/** Clock ID for Quinary TDM OSR */
+#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_OSR 0x20A
+
+/* Clock ID for MCLK1 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_1 0x300
+/* Clock ID for MCLK2 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_2 0x301
+/* Clock ID for MCLK3 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_3 0x302
+/* Clock ID for MCLK4 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_4 0x304
+/* Clock ID for Internal Digital Codec Core */
+#define Q6AFE_LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE 0x303
+/* Clock ID for INT MCLK0 */
+#define Q6AFE_LPASS_CLK_ID_INT_MCLK_0 0x305
+/* Clock ID for INT MCLK1 */
+#define Q6AFE_LPASS_CLK_ID_INT_MCLK_1 0x306
+
+/* Clock attribute for invalid use (reserved for internal usage) */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVALID 0x0
+/* Clock attribute for no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO 0x1
+/* Clock attribute for dividend couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND 0x2
+/* Clock attribute for divisor couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR 0x3
+/* Clock attribute for invert and no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO 0x4
+
struct q6afe_hdmi_cfg {
u16 datatype;
u16 channel_allocation;
@@ -59,4 +187,7 @@ void q6afe_slim_port_prepare(struct q6afe_port *port,
struct q6afe_slim_cfg *cfg);
int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg);
+int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
+ int clk_src, int clk_root,
+ unsigned int freq, int dir);
#endif /* __Q6AFE_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 11/24] ASoC: qdsp6: q6adm: Add q6adm driver
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (9 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 10/24] ASoC: qdsp6: q6afe: Add support to MI2S sysclks srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 12/24] ASoC: qdsp6: q6asm: Add q6asm driver srinivas.kandagatla
` (12 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to Q6ADM (Audio Device Manager) module in
q6dsp. ADM performs routing between audio streams and AFE ports.
It does Rate matching for streams going to devices driven by
different clocks, it handles volume ramping, Mixing with channel
and bit-width. ADM creates and destroys dynamic COPP services
for device-related audio processing as needed.
This patch adds basic support to ADM.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6adm.c | 664 ++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6adm.h | 25 ++
4 files changed, 694 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6adm.c
create mode 100644 sound/soc/qcom/qdsp6/q6adm.h
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index f50c76d712c0..5090ffa75648 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -50,11 +50,15 @@ config SND_SOC_QDSP6_COMMON
config SND_SOC_QDSP6_AFE
tristate
+config SND_SOC_QDSP6_ADM
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_AFE
+ select SND_SOC_QDSP6_ADM
help
To add support for MSM QDSP6 Soc Audio.
This will enable sound soc platform specific
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 9ec951e0833b..930944425051 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -1,2 +1,3 @@
obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
+obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c
new file mode 100644
index 000000000000..4b4c604cb998
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -0,0 +1,664 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/slab.h>
+#include <linux/wait.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/module.h>
+#include <linux/sched.h>
+#include <linux/jiffies.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/kref.h>
+#include <linux/wait.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/platform_device.h>
+#include <sound/asound.h>
+#include "q6adm.h"
+#include "q6afe.h"
+#include "q6core.h"
+#include "q6dsp-errno.h"
+#include "q6dsp-common.h"
+
+#define ADM_CMD_DEVICE_OPEN_V5 0x00010326
+#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329
+#define ADM_CMD_DEVICE_CLOSE_V5 0x00010327
+#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325
+
+#define TIMEOUT_MS 1000
+#define RESET_COPP_ID 99
+#define INVALID_COPP_ID 0xFF
+/* Definition for a legacy device session. */
+#define ADM_LEGACY_DEVICE_SESSION 0
+#define ADM_MATRIX_ID_AUDIO_RX 0
+#define ADM_MATRIX_ID_AUDIO_TX 1
+
+struct copp {
+ int afe_port;
+ int copp_idx;
+ int id;
+ int topology;
+ int mode;
+ int rate;
+ int bit_width;
+ int channels;
+ int app_type;
+ int acdb_id;
+
+ struct aprv2_ibasic_rsp_result_t result;
+ struct kref refcount;
+ wait_queue_head_t wait;
+ struct list_head node;
+ struct q6adm *adm;
+};
+
+struct q6adm {
+ struct apr_device *apr;
+ struct device *dev;
+ struct q6core_svc_api_info ainfo;
+ unsigned long copp_bitmap[AFE_MAX_PORTS];
+ struct list_head copps_list;
+ spinlock_t copps_list_lock;
+ struct aprv2_ibasic_rsp_result_t result;
+ struct mutex lock;
+ wait_queue_head_t matrix_map_wait;
+ void *routing_data;
+ struct platform_device *pdev_routing;
+};
+
+struct adm_cmd_device_open_v5 {
+ struct apr_hdr hdr;
+ u16 flags;
+ u16 mode_of_operation;
+ u16 endpoint_id_1;
+ u16 endpoint_id_2;
+ u32 topology_id;
+ u16 dev_num_channel;
+ u16 bit_width;
+ u32 sample_rate;
+ u8 dev_channel_mapping[8];
+} __packed;
+
+struct adm_cmd_matrix_map_routings_v5 {
+ struct apr_hdr hdr;
+ u32 matrix_id;
+ u32 num_sessions;
+} __packed;
+
+struct adm_session_map_node_v5 {
+ u16 session_id;
+ u16 num_copps;
+} __packed;
+
+static struct copp *adm_find_copp(struct q6adm *adm, int port_idx,
+ int copp_idx)
+{
+ struct copp *c = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+ list_for_each_entry(c, &adm->copps_list, node) {
+ if ((port_idx == c->afe_port) && (copp_idx == c->copp_idx))
+ break;
+ }
+
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+
+ return c;
+
+}
+
+static int q6adm_callback(struct apr_device *adev,
+ struct apr_client_message *data)
+{
+ struct aprv2_ibasic_rsp_result_t *result = data->payload;
+ int port_idx, copp_idx;
+ struct copp *copp;
+ struct q6adm *adm = dev_get_drvdata(&adev->dev);
+
+ if (!data->payload_size)
+ return 0;
+
+ copp_idx = (data->token) & 0XFF;
+ port_idx = ((data->token) >> 16) & 0xFF;
+ if (port_idx < 0 || port_idx >= AFE_MAX_PORTS) {
+ dev_err(&adev->dev, "Invalid port idx %d token %d\n",
+ port_idx, data->token);
+ return 0;
+ }
+ if (copp_idx < 0 || copp_idx >= MAX_COPPS_PER_PORT) {
+ dev_err(&adev->dev, "Invalid copp idx %d token %d\n",
+ copp_idx, data->token);
+ return 0;
+ }
+
+ switch (data->opcode) {
+ case APR_BASIC_RSP_RESULT: {
+ if (result->status != 0) {
+ dev_err(&adev->dev, "cmd = 0x%x return error = 0x%x\n",
+ result->opcode, result->status);
+ }
+ switch (result->opcode) {
+ case ADM_CMD_DEVICE_OPEN_V5:
+ case ADM_CMD_DEVICE_CLOSE_V5:
+ copp = adm_find_copp(adm, port_idx, copp_idx);
+ if (!copp)
+ return 0;
+
+ copp->result = *result;
+ wake_up(&copp->wait);
+ break;
+ case ADM_CMD_MATRIX_MAP_ROUTINGS_V5:
+ adm->result = *result;
+ wake_up(&adm->matrix_map_wait);
+ break;
+
+ default:
+ dev_err(&adev->dev, "Unknown Cmd: 0x%x\n",
+ result->opcode);
+ break;
+ }
+ return 0;
+ }
+ case ADM_CMDRSP_DEVICE_OPEN_V5: {
+ struct adm_cmd_rsp_device_open_v5 {
+ u32 status;
+ u16 copp_id;
+ u16 reserved;
+ } __packed * open = data->payload;
+
+ open = data->payload;
+ copp = adm_find_copp(adm, port_idx, copp_idx);
+ if (!copp)
+ return 0;
+
+ if (open->copp_id == INVALID_COPP_ID) {
+ dev_err(&adev->dev, "Invalid coppid rxed %d\n",
+ open->copp_id);
+ copp->result.status = ADSP_EBADPARAM;
+ wake_up(&copp->wait);
+ break;
+ }
+ copp->result.opcode = data->opcode;
+ copp->id = open->copp_id;
+ wake_up(&copp->wait);
+ }
+ break;
+ default:
+ dev_err(&adev->dev, "Unknown cmd:0x%x\n",
+ data->opcode);
+ break;
+ }
+
+ return 0;
+}
+
+static struct copp *adm_alloc_copp(struct q6adm *adm, int port_idx)
+{
+ struct copp *c;
+ int idx;
+
+ idx = find_first_zero_bit(&adm->copp_bitmap[port_idx],
+ MAX_COPPS_PER_PORT);
+
+ if (idx > MAX_COPPS_PER_PORT)
+ return ERR_PTR(-EBUSY);
+
+ c = kzalloc(sizeof(*c), GFP_KERNEL);
+ if (!c)
+ return ERR_PTR(-ENOMEM);
+
+ set_bit(idx, &adm->copp_bitmap[port_idx]);
+ c->copp_idx = idx;
+ c->afe_port = port_idx;
+ c->adm = adm;
+
+ init_waitqueue_head(&c->wait);
+
+ return c;
+}
+
+static int q6adm_apr_send_copp_pkt(struct q6adm *adm, struct copp *copp,
+ void *data, uint32_t rsp_opcode)
+{
+ struct device *dev = adm->dev;
+ struct apr_hdr *hdr = data;
+ uint32_t opcode = hdr->opcode;
+ int ret;
+
+ mutex_lock(&adm->lock);
+ copp->result.opcode = 0;
+ copp->result.status = 0;
+ ret = apr_send_pkt(adm->apr, data);
+ if (ret < 0) {
+ dev_err(dev, "Failed to send APR packet\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ /* Wait for the callback with copp id */
+ if (rsp_opcode)
+ ret = wait_event_timeout(copp->wait,
+ (copp->result.opcode == opcode) ||
+ (copp->result.opcode == rsp_opcode),
+ msecs_to_jiffies(TIMEOUT_MS));
+ else
+ ret = wait_event_timeout(copp->wait,
+ (copp->result.opcode == opcode),
+ msecs_to_jiffies(TIMEOUT_MS));
+
+ if (!ret) {
+ dev_err(dev, "ADM copp cmd timedout\n");
+ ret = -EINVAL;
+ } else if (copp->result.status > 0) {
+ dev_err(dev, "DSP returned error[%d]\n",
+ copp->result.status);
+ ret = -EINVAL;
+ }
+
+err:
+ mutex_unlock(&adm->lock);
+ return ret;
+}
+
+static int q6adm_device_close(struct q6adm *adm, struct copp *copp,
+ int port_id, int copp_idx)
+{
+ struct apr_hdr close = {0};
+
+ close.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ close.pkt_size = sizeof(close);
+ close.src_svc = APR_SVC_ADM;
+ close.src_domain = APR_DOMAIN_APPS;
+ close.src_port = port_id;
+ close.dest_svc = APR_SVC_ADM;
+ close.dest_domain = APR_DOMAIN_ADSP;
+ close.dest_port = copp->id;
+ close.token = port_id << 16 | copp_idx;
+ close.opcode = ADM_CMD_DEVICE_CLOSE_V5;
+
+ return q6adm_apr_send_copp_pkt(adm, copp, &close, 0);
+}
+
+static void adm_free_copp(struct kref *ref)
+{
+ struct copp *c = container_of(ref, struct copp, refcount);
+ struct q6adm *adm = c->adm;
+ int port_idx = c->afe_port;
+ int copp_idx = c->copp_idx;
+ unsigned long flags;
+ int ret;
+
+ ret = q6adm_device_close(adm, c, port_idx, copp_idx);
+ if (ret < 0)
+ dev_err(adm->dev, "Failed to close copp %d\n", ret);
+
+ clear_bit(c->copp_idx, &adm->copp_bitmap[port_idx]);
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+ list_del(&c->node);
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+ kfree(c);
+}
+
+static struct copp *adm_find_matching_copp(struct q6adm *adm,
+ int port_id, int topology,
+ int mode, int rate, int channel_mode,
+ int bit_width, int app_type)
+{
+ unsigned long flags;
+ struct copp *c;
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+
+ list_for_each_entry(c, &adm->copps_list, node) {
+ if ((port_id == c->afe_port) && (topology == c->topology) &&
+ (mode == c->mode) && (rate == c->rate) &&
+ (bit_width == c->bit_width) && (app_type == c->app_type)) {
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+ return c;
+ }
+ }
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+
+ c = adm_alloc_copp(adm, port_id);
+ if (IS_ERR_OR_NULL(c))
+ return ERR_CAST(c);
+
+ kref_init(&c->refcount);
+ c->topology = topology;
+ c->mode = mode;
+ c->rate = rate;
+ c->channels = channel_mode;
+ c->bit_width = bit_width;
+ c->app_type = app_type;
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+ list_add_tail(&c->node, &adm->copps_list);
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+
+ return c;
+
+}
+
+static int q6adm_device_open(struct q6adm *adm, struct copp *copp, int port_id,
+ int path, int topology, int channel_mode,
+ int bit_width, int rate)
+{
+ struct adm_cmd_device_open_v5 open = {0,};
+ int afe_port = q6afe_get_port_id(port_id);
+ int ret;
+
+ open.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ open.hdr.pkt_size = sizeof(open);
+ open.hdr.src_svc = APR_SVC_ADM;
+ open.hdr.src_domain = APR_DOMAIN_APPS;
+ open.hdr.src_port = afe_port;
+ open.hdr.dest_svc = APR_SVC_ADM;
+ open.hdr.dest_domain = APR_DOMAIN_ADSP;
+ open.hdr.dest_port = afe_port;
+ open.hdr.token = port_id << 16 | copp->copp_idx;
+ open.hdr.opcode = ADM_CMD_DEVICE_OPEN_V5;
+ open.flags = ADM_LEGACY_DEVICE_SESSION;
+ open.mode_of_operation = path;
+ open.endpoint_id_1 = afe_port;
+ open.topology_id = topology;
+ open.dev_num_channel = channel_mode & 0x00FF;
+ open.bit_width = bit_width;
+ open.sample_rate = rate;
+
+ ret = q6dsp_map_channels(&open.dev_channel_mapping[0],
+ channel_mode);
+ if (ret)
+ return ret;
+
+ return q6adm_apr_send_copp_pkt(adm, copp, &open,
+ ADM_CMDRSP_DEVICE_OPEN_V5);
+}
+
+/**
+ * q6adm_open() - open adm and grab a free copp
+ *
+ * @dev: Pointer to adm child device.
+ * @port_id: port id
+ * @path: playback or capture path.
+ * @rate: rate at which copp is required.
+ * @channel_mode: channel mode
+ * @topology: adm topology id
+ * @perf_mode: performace mode.
+ * @bit_width: audio sample bit width
+ * @app_type: Application type.
+ * @acdb_id: ACDB id
+ *
+ * Return: Will be an negative on error or a valid copp index on success.
+ */
+int q6adm_open(struct device *dev, int port_id, int path, int rate,
+ int channel_mode, int topology, int perf_mode,
+ uint16_t bit_width, int app_type, int acdb_id)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+ struct copp *copp;
+ int ret = 0;
+
+ if (port_id < 0) {
+ dev_err(dev, "Invalid port_id 0x%x\n", port_id);
+ return -EINVAL;
+ }
+
+ copp = adm_find_matching_copp(adm, port_id, topology, perf_mode,
+ rate, channel_mode, bit_width, app_type);
+ if (kref_read(&copp->refcount) == 1) {
+ /* not initialized yet */
+ ret = q6adm_device_open(adm, copp, port_id, path, topology,
+ channel_mode, bit_width, rate);
+ if (ret < 0)
+ return ret;
+ }
+
+ kref_get(&copp->refcount);
+
+ return copp->copp_idx;
+}
+EXPORT_SYMBOL_GPL(q6adm_open);
+
+/**
+ * q6adm_set_routing_data() - set routing private data
+ *
+ * @dev: Pointer to adm device.
+ * @data: routing private data
+ *
+ */
+void q6adm_set_routing_data(struct device *dev, void *data)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+
+ adm->routing_data = data;
+}
+EXPORT_SYMBOL_GPL(q6adm_set_routing_data);
+
+/**
+ * q6adm_get_routing_data() - get routing private data
+ *
+ * @dev: Pointer to adm device.
+ *
+ * Return: pointer to routing private data
+ */
+void *q6adm_get_routing_data(struct device *dev)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+
+ return adm->routing_data;
+}
+EXPORT_SYMBOL_GPL(q6adm_get_routing_data);
+
+/**
+ * q6adm_matrix_map() - Map asm streams and afe ports using payload
+ *
+ * @dev: Pointer to adm child device.
+ * @path: playback or capture path.
+ * @payload_map: map between session id and afe ports.
+ * @perf_mode: Performace mode.
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6adm_matrix_map(struct device *dev, int path,
+ struct route_payload payload_map, int perf_mode)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+ struct adm_cmd_matrix_map_routings_v5 *route;
+ struct adm_session_map_node_v5 *node;
+ uint16_t *copps_list;
+ int cmd_size, ret, i, copp_idx;
+ void *matrix_map = NULL;
+ struct copp *copp;
+
+ /* Assumes port_ids have already been validated during adm_open */
+ cmd_size = (sizeof(*route) +
+ sizeof(*node) +
+ (sizeof(uint32_t) * payload_map.num_copps));
+ matrix_map = kzalloc(cmd_size, GFP_KERNEL);
+ if (!matrix_map)
+ return -ENOMEM;
+
+ route = matrix_map;
+ route->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ route->hdr.pkt_size = cmd_size;
+ route->hdr.src_svc = 0;
+ route->hdr.src_domain = APR_DOMAIN_APPS;
+ route->hdr.dest_svc = APR_SVC_ADM;
+ route->hdr.dest_domain = APR_DOMAIN_ADSP;
+ route->hdr.token = 0;
+ route->hdr.opcode = ADM_CMD_MATRIX_MAP_ROUTINGS_V5;
+ route->num_sessions = 1;
+
+ switch (path) {
+ case ADM_PATH_PLAYBACK:
+ route->matrix_id = ADM_MATRIX_ID_AUDIO_RX;
+ break;
+ case ADM_PATH_LIVE_REC:
+ route->matrix_id = ADM_MATRIX_ID_AUDIO_TX;
+ break;
+ default:
+ dev_err(dev, "Wrong path set[%d]\n", path);
+
+ break;
+ }
+
+ node = matrix_map + sizeof(*route);
+ node->session_id = payload_map.session_id;
+ node->num_copps = payload_map.num_copps;
+ copps_list = matrix_map + sizeof(*route) + sizeof(*node);
+
+ for (i = 0; i < payload_map.num_copps; i++) {
+ int port_idx = payload_map.port_id[i];
+
+ if (port_idx < 0) {
+ dev_err(dev, "Invalid port_id 0x%x\n",
+ payload_map.port_id[i]);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+ copp_idx = payload_map.copp_idx[i];
+
+ copp = adm_find_copp(adm, port_idx, copp_idx);
+ if (!copp) {
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+ copps_list[i] = copp->id;
+ }
+
+ mutex_lock(&adm->lock);
+ adm->result.status = 0;
+ adm->result.opcode = 0;
+
+ ret = apr_send_pkt(adm->apr, matrix_map);
+ if (ret < 0) {
+ dev_err(dev, "routing for syream %d failed ret %d\n",
+ payload_map.session_id, ret);
+ goto fail_cmd;
+ }
+ ret = wait_event_timeout(adm->matrix_map_wait,
+ adm->result.opcode == route->hdr.opcode,
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ dev_err(dev, "routing for syream %d failed\n",
+ payload_map.session_id);
+ ret = -ETIMEDOUT;
+ goto fail_cmd;
+ } else if (adm->result.status > 0) {
+ dev_err(dev, "DSP returned error[%d]\n",
+ adm->result.status);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+fail_cmd:
+ mutex_unlock(&adm->lock);
+ kfree(matrix_map);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6adm_matrix_map);
+
+/**
+ * q6adm_close() - Close adm copp
+ *
+ * @dev: Pointer to adm child device.
+ * @port_id: afe port id.
+ * @perf_mode: perf_mode mode
+ * @copp_idx: copp index to close
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6adm_close(struct device *dev, int port_id, int perf_mode, int copp_idx)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+ struct copp *copp;
+
+ if (port_id < 0) {
+ dev_err(dev, "Invalid port_id 0x%x\n", port_id);
+ return -EINVAL;
+ }
+
+ if ((copp_idx < 0) || (copp_idx >= MAX_COPPS_PER_PORT)) {
+ dev_err(dev, "Invalid copp idx: %d\n", copp_idx);
+ return -EINVAL;
+ }
+
+ copp = adm_find_copp(adm, port_id, copp_idx);
+ if (!copp)
+ return -EINVAL;
+
+ kref_put(&copp->refcount, adm_free_copp);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6adm_close);
+
+static int q6adm_probe(struct apr_device *adev)
+{
+ struct device *dev = &adev->dev;
+ struct device_node *dais_np;
+ struct q6adm *adm;
+
+ adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL);
+ if (!adm)
+ return -ENOMEM;
+
+ adm->apr = adev;
+ dev_set_drvdata(&adev->dev, adm);
+ adm->dev = dev;
+ q6core_get_svc_api_info(adev->svc_id, &adm->ainfo);
+ mutex_init(&adm->lock);
+ init_waitqueue_head(&adm->matrix_map_wait);
+
+ INIT_LIST_HEAD(&adm->copps_list);
+ spin_lock_init(&adm->copps_list_lock);
+
+ dais_np = of_get_child_by_name(dev->of_node, "routing");
+ if (dais_np) {
+ adm->pdev_routing = of_platform_device_create(dais_np,
+ "q6routing", dev);
+ of_node_put(dais_np);
+ }
+
+ return 0;
+}
+
+static int q6adm_remove(struct apr_device *adev)
+{
+ struct q6adm *adm = dev_get_drvdata(&adev->dev);
+
+ if (adm->pdev_routing)
+ of_platform_device_destroy(&adm->pdev_routing->dev, NULL);
+
+ return 0;
+}
+
+static const struct of_device_id q6adm_device_id[] = {
+ { .compatible = "qcom,q6adm" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6adm_device_id);
+
+static struct apr_driver qcom_q6adm_driver = {
+ .probe = q6adm_probe,
+ .remove = q6adm_remove,
+ .callback = q6adm_callback,
+ .driver = {
+ .name = "qcom-q6adm",
+ .of_match_table = of_match_ptr(q6adm_device_id),
+ },
+};
+
+module_apr_driver(qcom_q6adm_driver);
+MODULE_DESCRIPTION("Q6 Audio Device Manager");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6adm.h b/sound/soc/qcom/qdsp6/q6adm.h
new file mode 100644
index 000000000000..7ad278399ec0
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6adm.h
@@ -0,0 +1,25 @@
+// SPDX-License-Identifier: GPL-2.0
+#ifndef __Q6_ADM_V2_H__
+#define __Q6_ADM_V2_H__
+
+#define ADM_PATH_PLAYBACK 0x1
+#define ADM_PATH_LIVE_REC 0x2
+#define MAX_COPPS_PER_PORT 8
+#define NULL_COPP_TOPOLOGY 0x00010312
+
+/* multiple copp per stream. */
+struct route_payload {
+ int num_copps;
+ int session_id;
+ int copp_idx[MAX_COPPS_PER_PORT];
+ int port_id[MAX_COPPS_PER_PORT];
+};
+
+int q6adm_open(struct device *dev, int port_id, int path, int rate,
+ int channel_mode, int topology, int perf_mode,
+ uint16_t bit_width, int app_type, int acdb_id);
+int q6adm_close(struct device *dev, int port, int topology, int perf_mode);
+int q6adm_matrix_map(struct device *dev, int path,
+ struct route_payload payload_map, int perf_mode);
+
+#endif /* __Q6_ADM_V2_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 12/24] ASoC: qdsp6: q6asm: Add q6asm driver
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (10 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 11/24] ASoC: qdsp6: q6adm: Add q6adm driver srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 13/24] ASoC: qdsp6: q6asm: Add support to memory map and unmap srinivas.kandagatla
` (11 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds basic support to Q6 ASM (Audio Stream Manager) module on
Q6DSP. ASM supports up to 8 concurrent streams. each stream can be setup
as playback/capture. ASM provides top control functions like
Pause/flush/resume for playback and record. ASM can Create/destroy encoder,
decoder and also provides POPP dynamic services.
This patch adds support to basic features to allow hdmi playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6asm.c | 230 ++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6asm.h | 15 +++
4 files changed, 250 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6asm.c
create mode 100644 sound/soc/qcom/qdsp6/q6asm.h
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 5090ffa75648..a07181290624 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -53,12 +53,16 @@ config SND_SOC_QDSP6_AFE
config SND_SOC_QDSP6_ADM
tristate
+config SND_SOC_QDSP6_ASM
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_AFE
select SND_SOC_QDSP6_ADM
+ select SND_SOC_QDSP6_ASM
help
To add support for MSM QDSP6 Soc Audio.
This will enable sound soc platform specific
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 930944425051..eea962315ab3 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -1,3 +1,4 @@
obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
+obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
new file mode 100644
index 000000000000..4ed64d32c6b0
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -0,0 +1,230 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/mutex.h>
+#include <linux/wait.h>
+#include <linux/module.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/device.h>
+#include <linux/spinlock.h>
+#include <linux/of.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/mm.h>
+#include "q6asm.h"
+#include "q6core.h"
+#include "q6dsp-errno.h"
+#include "q6dsp-common.h"
+
+#define ASM_SYNC_IO_MODE 0x0001
+#define ASM_ASYNC_IO_MODE 0x0002
+#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
+#define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
+
+struct audio_client {
+ int session;
+ q6asm_cb cb;
+ void *priv;
+ uint32_t io_mode;
+ struct apr_device *adev;
+ struct mutex lock;
+ wait_queue_head_t cmd_wait;
+ struct aprv2_ibasic_rsp_result_t result;
+ int perf_mode;
+ int stream_id;
+ struct device *dev;
+};
+
+struct q6asm {
+ struct apr_device *adev;
+ struct device *dev;
+ struct q6core_svc_api_info ainfo;
+ wait_queue_head_t mem_wait;
+ struct platform_device *pcmdev;
+ struct audio_client *session[MAX_SESSIONS + 1];
+ struct platform_device *pdev_dais;
+};
+
+static bool q6asm_is_valid_audio_client(struct audio_client *ac)
+{
+ struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+ int n;
+
+ if (!ac)
+ return false;
+
+ for (n = 1; n <= MAX_SESSIONS; n++) {
+ if (a->session[n] == ac)
+ return true;
+ }
+
+ return false;
+}
+
+/**
+ * q6asm_audio_client_free() - Freee allocated audio client
+ *
+ * @ac: audio client to free
+ */
+void q6asm_audio_client_free(struct audio_client *ac)
+{
+ struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+
+ a->session[ac->session] = NULL;
+ kfree(ac);
+}
+EXPORT_SYMBOL_GPL(q6asm_audio_client_free);
+
+static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
+ int session_id)
+{
+ if ((session_id <= 0) || (session_id > MAX_SESSIONS)) {
+ dev_err(a->dev, "invalid session: %d\n", session_id);
+ return NULL;
+ }
+
+ if (!a->session[session_id]) {
+ dev_err(a->dev, "session not active: %d\n", session_id);
+ return NULL;
+ }
+
+ return a->session[session_id];
+}
+
+static int q6asm_srvc_callback(struct apr_device *adev,
+ struct apr_client_message *data)
+{
+ struct aprv2_ibasic_rsp_result_t *result;
+ struct q6asm *a, *q6asm = dev_get_drvdata(&adev->dev);
+ struct audio_client *ac = NULL;
+ struct audio_port_data *port;
+ uint32_t dir = 0;
+ uint32_t sid = 0;
+
+ result = data->payload;
+ sid = (data->token >> 8) & 0x0F;
+ ac = q6asm_get_audio_client(q6asm, sid);
+ if (!ac) {
+ dev_err(&adev->dev, "Audio Client not active\n");
+ return 0;
+ }
+
+ if (ac->cb)
+ ac->cb(data->opcode, data->token, data->payload, ac->priv);
+
+ return 0;
+}
+
+/**
+ * q6asm_get_session_id() - get session id for audio client
+ *
+ * @ac: audio client pointer
+ *
+ * Return: Will be an session id of the audio client.
+ */
+int q6asm_get_session_id(struct audio_client *c)
+{
+ return c->session;
+}
+EXPORT_SYMBOL_GPL(q6asm_get_session_id);
+
+/**
+ * q6asm_audio_client_alloc() - Allocate a new audio client
+ *
+ * @dev: Pointer to asm child device.
+ * @cb: event callback.
+ * @priv: private data associated with this client.
+ *
+ * Return: Will be an error pointer on error or a valid audio client
+ * on success.
+ */
+struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
+ void *priv, int stream_id,
+ int perf_mode)
+{
+ struct q6asm *a = dev_get_drvdata(dev->parent);
+ struct audio_client *ac;
+
+ if (stream_id + 1 > MAX_SESSIONS)
+ return ERR_PTR(-EINVAL);
+
+ ac = kzalloc(sizeof(*ac), GFP_KERNEL);
+ if (!ac)
+ return ERR_PTR(-ENOMEM);
+
+ a->session[stream_id + 1] = ac;
+ ac->session = stream_id + 1;
+ ac->cb = cb;
+ ac->dev = dev;
+ ac->priv = priv;
+ ac->io_mode = ASM_SYNC_IO_MODE;
+ ac->perf_mode = perf_mode;
+ /* DSP expects stream id from 1 */
+ ac->stream_id = 1;
+ ac->adev = a->adev;
+
+ init_waitqueue_head(&ac->cmd_wait);
+ mutex_init(&ac->lock);
+ spin_lock_init(&ac->buf_lock);
+
+ return ac;
+}
+EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
+
+
+static int q6asm_probe(struct apr_device *adev)
+{
+ struct device *dev = &adev->dev;
+ struct device_node *dais_np;
+ struct q6asm *q6asm;
+
+ q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL);
+ if (!q6asm)
+ return -ENOMEM;
+
+ q6core_get_svc_api_info(adev->svc_id, &q6asm->ainfo);
+
+ q6asm->dev = dev;
+ q6asm->adev = adev;
+ init_waitqueue_head(&q6asm->mem_wait);
+ dev_set_drvdata(dev, q6asm);
+
+ dais_np = of_get_child_by_name(dev->of_node, "dais");
+ if (dais_np) {
+ q6asm->pdev_dais = of_platform_device_create(dais_np,
+ "q6asm-dai", dev);
+ of_node_put(dais_np);
+ }
+
+ return 0;
+}
+
+static int q6asm_remove(struct apr_device *adev)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+
+ if (q6asm->pdev_dais)
+ of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL);
+
+ return 0;
+}
+static const struct of_device_id q6asm_device_id[] = {
+ { .compatible = "qcom,q6asm" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6asm_device_id);
+
+static struct apr_driver qcom_q6asm_driver = {
+ .probe = q6asm_probe,
+ .remove = q6asm_remove,
+ .callback = q6asm_srvc_callback,
+ .driver = {
+ .name = "qcom-q6asm",
+ .of_match_table = of_match_ptr(q6asm_device_id),
+ },
+};
+
+module_apr_driver(qcom_q6asm_driver);
+MODULE_DESCRIPTION("Q6 Audio Stream Manager driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
new file mode 100644
index 000000000000..6a68afc5d8b6
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -0,0 +1,15 @@
+// SPDX-License-Identifier: GPL-2.0
+#ifndef __Q6_ASM_H__
+#define __Q6_ASM_H__
+
+#define MAX_SESSIONS 16
+
+typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
+ void *payload, void *priv);
+struct audio_client;
+struct audio_client *q6asm_audio_client_alloc(struct device *dev,
+ q6asm_cb cb, void *priv,
+ int session_id, int perf_mode);
+void q6asm_audio_client_free(struct audio_client *ac);
+int q6asm_get_session_id(struct audio_client *ac);
+#endif /* __Q6_ASM_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 13/24] ASoC: qdsp6: q6asm: Add support to memory map and unmap
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (11 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 12/24] ASoC: qdsp6: q6asm: Add q6asm driver srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 14/24] ASoC: qdsp6: q6asm: Add support to audio stream apis srinivas.kandagatla
` (10 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to memory map and unmap regions commands in
q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/qdsp6/q6asm.c | 338 +++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6asm.h | 5 +
2 files changed, 343 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 4ed64d32c6b0..8a2ca2ff9ce8 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -17,10 +17,47 @@
#include "q6dsp-errno.h"
#include "q6dsp-common.h"
+#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
+#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
+#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
+
#define ASM_SYNC_IO_MODE 0x0001
#define ASM_ASYNC_IO_MODE 0x0002
#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
#define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
+#define ASM_SHIFT_GAPLESS_MODE_FLAG 31
+#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3
+
+struct avs_cmd_shared_mem_map_regions {
+ struct apr_hdr hdr;
+ u16 mem_pool_id;
+ u16 num_regions;
+ u32 property_flag;
+} __packed;
+
+struct avs_shared_map_region_payload {
+ u32 shm_addr_lsw;
+ u32 shm_addr_msw;
+ u32 mem_size_bytes;
+} __packed;
+
+struct avs_cmd_shared_mem_unmap_regions {
+ struct apr_hdr hdr;
+ u32 mem_map_handle;
+} __packed;
+
+struct audio_buffer {
+ phys_addr_t phys;
+ uint32_t used;
+ uint32_t size; /* size of buffer */
+};
+
+struct audio_port_data {
+ struct audio_buffer *buf;
+ uint32_t num_periods;
+ uint32_t dsp_buf;
+ uint32_t mem_map_handle;
+};
struct audio_client {
int session;
@@ -29,6 +66,9 @@ struct audio_client {
uint32_t io_mode;
struct apr_device *adev;
struct mutex lock;
+ spinlock_t buf_lock;
+ /* idx:1 out port, 0: in port */
+ struct audio_port_data port[2];
wait_queue_head_t cmd_wait;
struct aprv2_ibasic_rsp_result_t result;
int perf_mode;
@@ -62,6 +102,268 @@ static bool q6asm_is_valid_audio_client(struct audio_client *ac)
return false;
}
+static inline void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, bool cmd_flg,
+ uint32_t stream_id)
+{
+ hdr->hdr_field = APR_SEQ_CMD_HDR_FIELD;
+ hdr->src_svc = ac->adev->svc_id;
+ hdr->src_domain = APR_DOMAIN_APPS;
+ hdr->dest_svc = APR_SVC_ASM;
+ hdr->dest_domain = APR_DOMAIN_ADSP;
+ hdr->src_port = ((ac->session << 8) & 0xFF00) | (stream_id);
+ hdr->dest_port = ((ac->session << 8) & 0xFF00) | (stream_id);
+ hdr->pkt_size = pkt_size;
+ if (cmd_flg)
+ hdr->token = ac->session;
+}
+
+static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,
+ void *data, uint32_t rsp_opcode)
+{
+ struct apr_hdr *hdr = data;
+ int rc;
+
+ mutex_lock(&ac->lock);
+ ac->result.opcode = 0;
+ ac->result.status = 0;
+ rc = apr_send_pkt(a->adev, data);
+ if (rc < 0)
+ goto err;
+
+ if (rsp_opcode)
+ rc = wait_event_timeout(a->mem_wait,
+ (ac->result.opcode == hdr->opcode) ||
+ (ac->result.opcode == rsp_opcode),
+ 5 * HZ);
+ else
+ rc = wait_event_timeout(a->mem_wait,
+ (ac->result.opcode == hdr->opcode),
+ 5 * HZ);
+
+ if (!rc) {
+ dev_err(a->dev, "CMD timeout\n");
+ rc = -ETIMEDOUT;
+ } else if (ac->result.status > 0) {
+ dev_err(a->dev, "DSP returned error[%x]\n",
+ ac->result.status);
+ rc = -EINVAL;
+ }
+
+err:
+ mutex_unlock(&ac->lock);
+ return rc;
+}
+
+static int __q6asm_memory_unmap(struct audio_client *ac,
+ phys_addr_t buf_add, int dir)
+{
+ struct avs_cmd_shared_mem_unmap_regions mem_unmap;
+ struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+ int rc;
+
+ if (ac->port[dir].mem_map_handle == 0) {
+ dev_err(ac->dev, "invalid mem handle\n");
+ return -EINVAL;
+ }
+
+ mem_unmap.hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
+ mem_unmap.hdr.src_port = 0;
+ mem_unmap.hdr.dest_port = 0;
+ mem_unmap.hdr.pkt_size = sizeof(mem_unmap);
+ mem_unmap.hdr.token = ((ac->session << 8) | dir);
+
+ mem_unmap.hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
+ mem_unmap.mem_map_handle = ac->port[dir].mem_map_handle;
+
+ rc = q6asm_apr_send_session_pkt(a, ac, &mem_unmap, 0);
+ if (rc < 0)
+ return rc;
+
+ ac->port[dir].mem_map_handle = 0;
+
+ return 0;
+}
+
+
+static void q6asm_audio_client_free_buf(struct audio_client *ac,
+ struct audio_port_data *port)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&ac->buf_lock, flags);
+
+ port->num_periods = 0;
+ kfree(port->buf);
+ port->buf = NULL;
+
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+}
+
+/**
+ * q6asm_unmap_memory_regions() - unmap memory regions in the dsp.
+ *
+ * @dir: direction of audio stream
+ * @ac: audio client instanace
+ *
+ * Return: Will be an negative value on failure or zero on success
+ */
+int q6asm_unmap_memory_regions(unsigned int dir, struct audio_client *ac)
+{
+ struct audio_port_data *port;
+ int cnt = 0;
+ int rc = 0;
+
+ port = &ac->port[dir];
+ if (!port->buf) {
+ rc = -EINVAL;
+ goto err;
+ }
+
+ cnt = port->num_periods - 1;
+ if (cnt >= 0) {
+ rc = __q6asm_memory_unmap(ac, port->buf[dir].phys, dir);
+ if (rc < 0) {
+ dev_err(ac->dev, "%s: Memory_unmap_regions failed %d\n",
+ __func__, rc);
+ goto err;
+ }
+ }
+
+ q6asm_audio_client_free_buf(ac, port);
+
+err:
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_unmap_memory_regions);
+
+static int __q6asm_memory_map_regions(struct audio_client *ac, int dir,
+ size_t period_sz, unsigned int periods,
+ bool is_contiguous)
+{
+ struct avs_cmd_shared_mem_map_regions *cmd = NULL;
+ struct avs_shared_map_region_payload *mregions = NULL;
+ struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+ struct audio_port_data *port = NULL;
+ struct audio_buffer *ab = NULL;
+ void *mmap_region_cmd = NULL;
+ unsigned long flags;
+ uint32_t num_regions, buf_sz;
+ int rc, i, cmd_size;
+
+ if (is_contiguous) {
+ num_regions = 1;
+ buf_sz = period_sz * periods;
+ } else {
+ buf_sz = period_sz;
+ num_regions = periods;
+ }
+
+ /* DSP expects size should be aligned to 4K */
+ buf_sz = ALIGN(buf_sz, 4096);
+
+ cmd_size = sizeof(*cmd) + (sizeof(*mregions) * num_regions);
+ mmap_region_cmd = kzalloc(cmd_size, GFP_KERNEL);
+ if (!mmap_region_cmd)
+ return -ENOMEM;
+
+ cmd = mmap_region_cmd;
+
+ cmd->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
+ cmd->hdr.src_port = 0;
+ cmd->hdr.dest_port = 0;
+ cmd->hdr.pkt_size = cmd_size;
+ cmd->hdr.token = ((ac->session << 8) | dir);
+
+
+ cmd->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
+ cmd->mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
+ cmd->num_regions = num_regions;
+ cmd->property_flag = 0x00;
+
+ mregions = mmap_region_cmd + sizeof(*cmd);
+
+ port = &ac->port[dir];
+
+ spin_lock_irqsave(&ac->buf_lock, flags);
+ for (i = 0; i < num_regions; i++) {
+ ab = &port->buf[i];
+ mregions->shm_addr_lsw = lower_32_bits(ab->phys);
+ mregions->shm_addr_msw = upper_32_bits(ab->phys);
+ mregions->mem_size_bytes = buf_sz;
+ ++mregions;
+ }
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+
+ rc = q6asm_apr_send_session_pkt(a, ac, mmap_region_cmd,
+ ASM_CMDRSP_SHARED_MEM_MAP_REGIONS);
+
+ kfree(mmap_region_cmd);
+
+ return rc;
+}
+
+/**
+ * q6asm_map_memory_regions() - map memory regions in the dsp.
+ *
+ * @dir: direction of audio stream
+ * @ac: audio client instanace
+ * @phys: physcial address that needs mapping.
+ * @period_sz: audio period size
+ * @periods: number of periods
+ *
+ * Return: Will be an negative value on failure or zero on success
+ */
+int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac,
+ phys_addr_t phys,
+ size_t period_sz, unsigned int periods)
+{
+ struct audio_buffer *buf;
+ unsigned long flags;
+ int cnt;
+ int rc;
+
+ spin_lock_irqsave(&ac->buf_lock, flags);
+ if (ac->port[dir].buf) {
+ dev_err(ac->dev, "Buffer already allocated\n");
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return 0;
+ }
+
+ buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC);
+ if (!buf) {
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return -ENOMEM;
+ }
+
+
+ ac->port[dir].buf = buf;
+
+ buf[0].phys = phys;
+ buf[0].used = !!dir;
+ buf[0].size = period_sz;
+
+ for (cnt = 1; cnt < periods; cnt++) {
+ if (period_sz > 0) {
+ buf[cnt].phys = buf[0].phys + (cnt * period_sz);
+ buf[cnt].used = dir ^ 1;
+ buf[cnt].size = period_sz;
+ }
+ }
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+
+ ac->port[dir].num_periods = periods;
+
+ rc = __q6asm_memory_map_regions(ac, dir, period_sz, periods, 1);
+ if (rc < 0) {
+ dev_err(ac->dev, "Memory_map_regions failed\n");
+ q6asm_audio_client_free_buf(ac, &ac->port[dir]);
+ }
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_map_memory_regions);
+
/**
* q6asm_audio_client_free() - Freee allocated audio client
*
@@ -110,6 +412,42 @@ static int q6asm_srvc_callback(struct apr_device *adev,
return 0;
}
+ a = dev_get_drvdata(ac->dev->parent);
+ dir = (data->token & 0x0F);
+ port = &ac->port[dir];
+
+ switch (data->opcode)
+ case APR_BASIC_RSP_RESULT: {
+ switch (result->opcode) {
+ case ASM_CMD_SHARED_MEM_MAP_REGIONS:
+ case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
+ ac->result = *result;
+ wake_up(&a->mem_wait);
+ break;
+ default:
+ dev_err(&adev->dev, "command[0x%x] not expecting rsp\n",
+ result->opcode);
+ break;
+ }
+ return 0;
+ case ASM_CMDRSP_SHARED_MEM_MAP_REGIONS:
+ ac->result.status = 0;
+ ac->result.opcode = data->opcode;
+ ac->port[dir].mem_map_handle = result->opcode;
+ wake_up(&a->mem_wait);
+ break;
+ case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
+ ac->result.opcode = data->opcode;
+ ac->result.status = 0;
+ ac->port[dir].mem_map_handle = 0;
+ wake_up(&a->mem_wait);
+ break;
+ default:
+ dev_dbg(&adev->dev, "command[0x%x]success [0x%x]\n",
+ result->opcode, result->status);
+ break;
+ }
+
if (ac->cb)
ac->cb(data->opcode, data->token, data->payload, ac->priv);
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index 6a68afc5d8b6..faa49ca1cde6 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -12,4 +12,9 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
int session_id, int perf_mode);
void q6asm_audio_client_free(struct audio_client *ac);
int q6asm_get_session_id(struct audio_client *ac);
+int q6asm_map_memory_regions(unsigned int dir,
+ struct audio_client *ac,
+ phys_addr_t phys,
+ size_t bufsz, unsigned int bufcnt);
+int q6asm_unmap_memory_regions(unsigned int dir, struct audio_client *ac);
#endif /* __Q6_ASM_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 14/24] ASoC: qdsp6: q6asm: Add support to audio stream apis
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (12 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 13/24] ASoC: qdsp6: q6asm: Add support to memory map and unmap srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-04-09 11:53 ` [alsa-devel] " Rohit Kumar
2018-03-10 2:24 ` [PATCH v4 15/24] ASoC: qdsp6: q6core: Add q6core driver srinivas.kandagatla
` (9 subsequent siblings)
23 siblings, 1 reply; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to open, write and media format commands
in the q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/qdsp6/q6asm.c | 744 ++++++++++++++++++++++++++++++++++++++++++-
sound/soc/qcom/qdsp6/q6asm.h | 49 +++
2 files changed, 792 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 8a2ca2ff9ce8..ddbf526358bd 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -9,6 +9,8 @@
#include <linux/device.h>
#include <linux/spinlock.h>
#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <uapi/sound/asound.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/mm.h>
@@ -17,10 +19,36 @@
#include "q6dsp-errno.h"
#include "q6dsp-common.h"
+#define ASM_STREAM_CMD_CLOSE 0x00010BCD
+#define ASM_STREAM_CMD_FLUSH 0x00010BCE
+#define ASM_SESSION_CMD_PAUSE 0x00010BD3
+#define ASM_DATA_CMD_EOS 0x00010BDB
+#define ASM_DEFAULT_POPP_TOPOLOGY 0x00010BE4
+#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
+#define ASM_STREAM_POSTPROC_TOPO_ID_NONE 0x00010C68
#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
-
+#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
+#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99
+#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
+#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
+#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
+#define ASM_DATA_CMD_READ_V2 0x00010DAC
+#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
+#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
+#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
+#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+
+
+#define ASM_LEGACY_STREAM_SESSION 0
+/* Bit shift for the stream_perf_mode subfield. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29
+#define ASM_END_POINT_DEVICE_MATRIX 0
+#define ASM_DEFAULT_APP_TYPE 0
#define ASM_SYNC_IO_MODE 0x0001
#define ASM_ASYNC_IO_MODE 0x0002
#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
@@ -46,6 +74,96 @@ struct avs_cmd_shared_mem_unmap_regions {
u32 mem_map_handle;
} __packed;
+struct asm_data_cmd_media_fmt_update_v2 {
+ u32 fmt_blk_size;
+} __packed;
+
+struct asm_multi_channel_pcm_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 num_channels;
+ u16 bits_per_sample;
+ u32 sample_rate;
+ u16 is_signed;
+ u16 reserved;
+ u8 channel_mapping[PCM_FORMAT_MAX_NUM_CHANNEL];
+} __packed;
+
+struct asm_stream_cmd_set_encdec_param {
+ u32 param_id;
+ u32 param_size;
+} __packed;
+
+struct asm_enc_cfg_blk_param_v2 {
+ u32 frames_per_buf;
+ u32 enc_cfg_blk_size;
+} __packed;
+
+struct asm_multi_channel_pcm_enc_cfg_v2 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ uint16_t num_channels;
+ uint16_t bits_per_sample;
+ uint32_t sample_rate;
+ uint16_t is_signed;
+ uint16_t reserved;
+ uint8_t channel_mapping[8];
+} __packed;
+
+struct asm_data_cmd_read_v2 {
+ struct apr_hdr hdr;
+ u32 buf_addr_lsw;
+ u32 buf_addr_msw;
+ u32 mem_map_handle;
+ u32 buf_size;
+ u32 seq_id;
+} __packed;
+
+struct asm_data_cmd_read_v2_done {
+ u32 status;
+ u32 buf_addr_lsw;
+ u32 buf_addr_msw;
+};
+
+struct asm_stream_cmd_open_read_v3 {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+ u32 src_endpointype;
+ u32 preprocopo_id;
+ u32 enc_cfg_id;
+ u16 bits_per_sample;
+ u16 reserved;
+} __packed;
+
+struct asm_data_cmd_write_v2 {
+ struct apr_hdr hdr;
+ u32 buf_addr_lsw;
+ u32 buf_addr_msw;
+ u32 mem_map_handle;
+ u32 buf_size;
+ u32 seq_id;
+ u32 timestamp_lsw;
+ u32 timestamp_msw;
+ u32 flags;
+} __packed;
+
+struct asm_stream_cmd_open_write_v3 {
+ struct apr_hdr hdr;
+ uint32_t mode_flags;
+ uint16_t sink_endpointype;
+ uint16_t bits_per_sample;
+ uint32_t postprocopo_id;
+ uint32_t dec_fmt_id;
+} __packed;
+
+struct asm_session_cmd_run_v2 {
+ struct apr_hdr hdr;
+ u32 flags;
+ u32 time_lsw;
+ u32 time_msw;
+} __packed;
+
struct audio_buffer {
phys_addr_t phys;
uint32_t used;
@@ -394,6 +512,153 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
return a->session[session_id];
}
+static int32_t q6asm_stream_callback(struct apr_device *adev,
+ struct apr_client_message *data,
+ int session_id)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *result;
+ struct audio_port_data *port;
+ struct audio_client *ac;
+ uint32_t token;
+ uint32_t client_event = 0;
+
+ ac = q6asm_get_audio_client(q6asm, session_id);
+ if (!ac)/* Audio client might already be freed by now */
+ return 0;
+
+ if (!q6asm_is_valid_audio_client(ac))
+ return -EINVAL;
+
+ result = data->payload;
+
+ switch (data->opcode) {
+ case APR_BASIC_RSP_RESULT:
+ token = data->token;
+ switch (result->opcode) {
+ case ASM_SESSION_CMD_PAUSE:
+ client_event = ASM_CLIENT_EVENT_CMD_PAUSE_DONE;
+ break;
+ case ASM_SESSION_CMD_SUSPEND:
+ client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
+ break;
+ case ASM_DATA_CMD_EOS:
+ client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
+ break;
+ break;
+ case ASM_STREAM_CMD_FLUSH:
+ client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
+ break;
+ case ASM_SESSION_CMD_RUN_V2:
+ client_event = ASM_CLIENT_EVENT_CMD_RUN_DONE;
+ break;
+
+ case ASM_STREAM_CMD_FLUSH_READBUFS:
+ if (token != ac->session) {
+ dev_err(ac->dev, "session invalid\n");
+ return -EINVAL;
+ }
+ case ASM_STREAM_CMD_CLOSE:
+ client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
+ break;
+ case ASM_STREAM_CMD_OPEN_WRITE_V3:
+ case ASM_STREAM_CMD_OPEN_READ_V3:
+ case ASM_STREAM_CMD_OPEN_READWRITE_V2:
+ case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
+ case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+ if (result->status != 0) {
+ dev_err(ac->dev,
+ "cmd = 0x%x returned error = 0x%x\n",
+ result->opcode, result->status);
+ ac->result = *result;
+ wake_up(&ac->cmd_wait);
+ return 0;
+ }
+ break;
+ default:
+ dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
+ result->opcode);
+ break;
+ }
+
+ ac->result = *result;
+ wake_up(&ac->cmd_wait);
+
+ if (ac->cb)
+ ac->cb(client_event, data->token,
+ data->payload, ac->priv);
+
+ return 0;
+
+ case ASM_DATA_EVENT_WRITE_DONE_V2:
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+
+ client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
+
+ if (ac->io_mode & ASM_SYNC_IO_MODE) {
+ phys_addr_t phys;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ac->buf_lock, flags);
+ if (!port->buf) {
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return 0;
+ }
+
+ phys = port->buf[data->token].phys;
+
+ if (lower_32_bits(phys) != result->opcode ||
+ upper_32_bits(phys) != result->status) {
+ dev_err(ac->dev, "Expected addr %pa\n",
+ &port->buf[data->token].phys);
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return -EINVAL;
+ }
+ token = data->token;
+ port->buf[token].used = 1;
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ }
+ break;
+ case ASM_DATA_EVENT_READ_DONE_V2:
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ client_event = ASM_CLIENT_EVENT_DATA_READ_DONE;
+
+ if (ac->io_mode & ASM_SYNC_IO_MODE) {
+ struct asm_data_cmd_read_v2_done *done = data->payload;
+ unsigned long flags;
+ phys_addr_t phys;
+
+ spin_lock_irqsave(&ac->buf_lock, flags);
+ if (!port->buf) {
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return 0;
+ }
+
+ phys = port->buf[data->token].phys;
+ token = data->token;
+ port->buf[token].used = 0;
+
+ if (upper_32_bits(phys) != done->buf_addr_msw ||
+ lower_32_bits(phys) != done->buf_addr_lsw) {
+ dev_err(ac->dev, "Expected addr %pa %08x-%08x\n",
+ &port->buf[data->token].phys,
+ done->buf_addr_lsw,
+ done->buf_addr_msw);
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ return -EINVAL;
+ }
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+ }
+
+ break;
+ }
+
+ if (ac->cb)
+ ac->cb(client_event, data->token, data->payload, ac->priv);
+
+ return 0;
+}
+
static int q6asm_srvc_callback(struct apr_device *adev,
struct apr_client_message *data)
{
@@ -403,6 +668,11 @@ static int q6asm_srvc_callback(struct apr_device *adev,
struct audio_port_data *port;
uint32_t dir = 0;
uint32_t sid = 0;
+ int session_id;
+
+ session_id = (data->dest_port >> 8) & 0xFF;
+ if (session_id)
+ return q6asm_stream_callback(adev, data, session_id);
result = data->payload;
sid = (data->token >> 8) & 0x0F;
@@ -510,6 +780,478 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
}
EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
+static int q6asm_ac_send_cmd_sync(struct audio_client *ac, void *cmd)
+{
+ struct apr_hdr *hdr = cmd;
+ int rc;
+
+ mutex_lock(&ac->lock);
+ ac->result.opcode = 0;
+ ac->result.status = 0;
+
+ rc = apr_send_pkt(ac->adev, cmd);
+ if (rc < 0)
+ goto err;
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (ac->result.opcode == hdr->opcode), 5 * HZ);
+ if (!rc) {
+ dev_err(ac->dev, "CMD timeout\n");
+ rc = -ETIMEDOUT;
+ goto err;
+ }
+
+ if (ac->result.status > 0) {
+ dev_err(ac->dev, "DSP returned error[%x]\n",
+ ac->result.status);
+ rc = -EINVAL;
+ }
+
+
+err:
+ mutex_unlock(&ac->lock);
+ return rc;
+}
+
+/**
+ * q6asm_open_write() - Open audio client for writing
+ *
+ * @ac: audio client pointer
+ * @format: audio sample format
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_open_write(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample)
+{
+ struct asm_stream_cmd_open_write_v3 open;
+ int rc;
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, ac->stream_id);
+
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
+ open.mode_flags = 0x00;
+ open.mode_flags |= ASM_LEGACY_STREAM_SESSION;
+
+ /* source endpoint : matrix */
+ open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+ open.bits_per_sample = bits_per_sample;
+ open.postprocopo_id = ASM_DEFAULT_POPP_TOPOLOGY;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open.dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ default:
+ dev_err(ac->dev, "Invalid format 0x%x\n", format);
+ return -EINVAL;
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, &open);
+ if (rc < 0)
+ return rc;
+
+ ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6asm_open_write);
+
+static int __q6asm_run(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts, bool wait)
+{
+ struct asm_session_cmd_run_v2 run;
+
+ q6asm_add_hdr(ac, &run.hdr, sizeof(run), true, ac->stream_id);
+
+ run.hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+ run.flags = flags;
+ run.time_lsw = lsw_ts;
+ run.time_msw = msw_ts;
+ if (wait)
+ return q6asm_ac_send_cmd_sync(ac, &run);
+ else
+ return apr_send_pkt(ac->adev, &run);
+
+}
+
+/**
+ * q6asm_run() - start the audio client
+ *
+ * @ac: audio client pointer
+ * @flags: flags associated with write
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_run(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts)
+{
+ return __q6asm_run(ac, flags, msw_ts, lsw_ts, true);
+}
+EXPORT_SYMBOL_GPL(q6asm_run);
+
+/**
+ * q6asm_run_nowait() - start the audio client withou blocking
+ *
+ * @ac: audio client pointer
+ * @flags: flags associated with write
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts)
+{
+ return __q6asm_run(ac, flags, msw_ts, lsw_ts, false);
+}
+EXPORT_SYMBOL_GPL(q6asm_run_nowait);
+
+/**
+ * q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration
+ *
+ * @ac: audio client pointer
+ * @rate: audio sample rate
+ * @channels: number of audio channels.
+ * @use_default_chmap: flag to use default ch map.
+ * @channel_map: channel map pointer
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ u8 channel_map[PCM_FORMAT_MAX_NUM_CHANNEL],
+ uint16_t bits_per_sample)
+{
+ struct asm_multi_channel_pcm_fmt_blk_v2 fmt;
+ u8 *channel_mapping;
+ int rc;
+
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), true, ac->stream_id);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt.fmt_blk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
+ sizeof(fmt.fmt_blk);
+ fmt.num_channels = channels;
+ fmt.bits_per_sample = bits_per_sample;
+ fmt.sample_rate = rate;
+ fmt.is_signed = 1;
+
+ channel_mapping = fmt.channel_mapping;
+
+ if (channel_map) {
+ memcpy(channel_mapping, channel_map,
+ PCM_FORMAT_MAX_NUM_CHANNEL);
+ } else {
+ if (q6dsp_map_channels(channel_mapping, channels)) {
+ dev_err(ac->dev, " map channels failed %d\n", channels);
+ return -EINVAL;
+ }
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, &fmt);
+ if (rc < 0)
+ goto fail_cmd;
+
+ return 0;
+fail_cmd:
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
+
+/**
+ * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
+ *
+ * @ac: audio client pointer
+ * @rate: audio sample rate
+ * @channels: number of audio channels.
+ * @use_default_chmap: flag to use default ch map.
+ * @channel_map: channel map pointer
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+ uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
+{
+ struct asm_multi_channel_pcm_enc_cfg_v2 enc_cfg;
+ u8 *channel_mapping;
+ u32 frames_per_buf = 0;
+
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), true, ac->stream_id);
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg.encdec.param_size = sizeof(enc_cfg) - sizeof(enc_cfg.hdr) -
+ sizeof(enc_cfg.encdec);
+ enc_cfg.encblk.frames_per_buf = frames_per_buf;
+ enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ enc_cfg.num_channels = channels;
+ enc_cfg.bits_per_sample = bits_per_sample;
+ enc_cfg.sample_rate = rate;
+ enc_cfg.is_signed = 1;
+ channel_mapping = enc_cfg.channel_mapping;
+
+ memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL);
+
+ if (q6dsp_map_channels(channel_mapping, channels))
+ return -EINVAL;
+
+
+ return q6asm_ac_send_cmd_sync(ac, &enc_cfg);
+}
+EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
+
+/**
+ * q6asm_read() - read data of period size from audio client
+ *
+ * @ac: audio client pointer
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_read(struct audio_client *ac)
+{
+ struct asm_data_cmd_read_v2 read;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ int rc;
+
+ if (!(ac->io_mode & ASM_SYNC_IO_MODE))
+ return 0;
+
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ q6asm_add_hdr(ac, &read.hdr, sizeof(read), false, ac->stream_id);
+ ab = &port->buf[port->dsp_buf];
+ read.hdr.opcode = ASM_DATA_CMD_READ_V2;
+ read.buf_addr_lsw = lower_32_bits(ab->phys);
+ read.buf_addr_msw = upper_32_bits(ab->phys);
+ read.mem_map_handle = ac->port[SNDRV_PCM_STREAM_CAPTURE].mem_map_handle;
+
+ read.buf_size = ab->size;
+ read.seq_id = port->dsp_buf;
+ read.hdr.token = port->dsp_buf;
+
+ port->dsp_buf++;
+
+ if (port->dsp_buf >= port->num_periods)
+ port->dsp_buf = 0;
+
+ rc = apr_send_pkt(ac->adev, &read);
+ if (rc < 0) {
+ pr_err("read op[0x%x]rc[%d]\n", read.hdr.opcode, rc);
+ return rc;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6asm_read);
+
+static int __q6asm_open_read(struct audio_client *ac,
+ uint32_t format, uint16_t bits_per_sample)
+{
+ struct asm_stream_cmd_open_read_v3 open;
+
+ q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, ac->stream_id);
+ open.hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
+ /* Stream prio : High, provide meta info with encoded frames */
+ open.src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+
+ open.preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_NONE;
+ open.bits_per_sample = bits_per_sample;
+ open.mode_flags = 0x0;
+
+ open.mode_flags |= ASM_LEGACY_STREAM_SESSION <<
+ ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open.mode_flags |= 0x00;
+ open.enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", format);
+ }
+
+ return q6asm_ac_send_cmd_sync(ac, &open);
+}
+
+/**
+ * q6asm_open_read() - Open audio client for reading
+ *
+ * @ac: audio client pointer
+ * @format: audio sample format
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_open_read(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample)
+{
+ return __q6asm_open_read(ac, format, bits_per_sample);
+}
+EXPORT_SYMBOL_GPL(q6asm_open_read);
+
+/**
+ * q6asm_write_async() - non blocking write
+ *
+ * @ac: audio client pointer
+ * @len: lenght in bytes
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ * @flags: flags associated with write
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags)
+{
+ struct asm_data_cmd_write_v2 write;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ int rc = 0;
+
+ if (!(ac->io_mode & ASM_SYNC_IO_MODE))
+ return 0;
+
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+ q6asm_add_hdr(ac, &write.hdr, sizeof(write), false,
+ ac->stream_id);
+
+ ab = &port->buf[port->dsp_buf];
+
+ write.hdr.token = port->dsp_buf;
+ write.hdr.opcode = ASM_DATA_CMD_WRITE_V2;
+ write.buf_addr_lsw = lower_32_bits(ab->phys);
+ write.buf_addr_msw = upper_32_bits(ab->phys);
+ write.buf_size = len;
+ write.seq_id = port->dsp_buf;
+ write.timestamp_lsw = lsw_ts;
+ write.timestamp_msw = msw_ts;
+ write.mem_map_handle =
+ ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle;
+
+ if (flags == NO_TIMESTAMP)
+ write.flags = (flags & 0x800000FF);
+ else
+ write.flags = (0x80000000 | flags);
+
+ port->dsp_buf++;
+
+ if (port->dsp_buf >= port->num_periods)
+ port->dsp_buf = 0;
+
+ rc = apr_send_pkt(ac->adev, &write);
+ if (rc < 0)
+ return rc;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6asm_write_async);
+
+static void q6asm_reset_buf_state(struct audio_client *ac)
+{
+ struct audio_port_data *port = NULL;
+ unsigned long flags;
+ int loopcnt = 0;
+ int cnt = 0;
+ int used;
+
+ if (!(ac->io_mode & ASM_SYNC_IO_MODE))
+ return;
+
+ used = (ac->io_mode & ASM_TUN_WRITE_IO_MODE ? 1 : 0);
+ spin_lock_irqsave(&ac->buf_lock, flags);
+ for (loopcnt = 0; loopcnt <= SNDRV_PCM_STREAM_CAPTURE; loopcnt++) {
+ port = &ac->port[loopcnt];
+ cnt = port->num_periods - 1;
+ port->dsp_buf = 0;
+ while (cnt >= 0) {
+ if (!port->buf)
+ continue;
+ port->buf[cnt].used = used;
+ cnt--;
+ }
+ }
+ spin_unlock_irqrestore(&ac->buf_lock, flags);
+}
+
+static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
+{
+ int stream_id = ac->stream_id;
+ struct apr_hdr hdr;
+ int rc;
+
+ q6asm_add_hdr(ac, &hdr, sizeof(hdr), true, stream_id);
+
+ switch (cmd) {
+ case CMD_PAUSE:
+ hdr.opcode = ASM_SESSION_CMD_PAUSE;
+ break;
+ case CMD_SUSPEND:
+ hdr.opcode = ASM_SESSION_CMD_SUSPEND;
+ break;
+ case CMD_FLUSH:
+ hdr.opcode = ASM_STREAM_CMD_FLUSH;
+ break;
+ case CMD_OUT_FLUSH:
+ hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
+ break;
+ case CMD_EOS:
+ hdr.opcode = ASM_DATA_CMD_EOS;
+ break;
+ case CMD_CLOSE:
+ hdr.opcode = ASM_STREAM_CMD_CLOSE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (wait)
+ rc = q6asm_ac_send_cmd_sync(ac, &hdr);
+ else
+ return apr_send_pkt(ac->adev, &hdr);
+
+ if (rc < 0)
+ return rc;
+
+ if (cmd == CMD_FLUSH)
+ q6asm_reset_buf_state(ac);
+
+ return 0;
+}
+
+/**
+ * q6asm_cmd() - run cmd on audio client
+ *
+ * @ac: audio client pointer
+ * @cmd: command to run on audio client.
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_cmd(struct audio_client *ac, int cmd)
+{
+ return __q6asm_cmd(ac, cmd, true);
+}
+EXPORT_SYMBOL_GPL(q6asm_cmd);
+
+/**
+ * q6asm_cmd_nowait() - non blocking, run cmd on audio client
+ *
+ * @ac: audio client pointer
+ * @cmd: command to run on audio client.
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd)
+{
+ return __q6asm_cmd(ac, cmd, false);
+}
+EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
static int q6asm_probe(struct apr_device *adev)
{
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index faa49ca1cde6..9ed15e0edaf0 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -1,8 +1,36 @@
// SPDX-License-Identifier: GPL-2.0
#ifndef __Q6_ASM_H__
#define __Q6_ASM_H__
+#include "q6dsp-common.h"
+#include <dt-bindings/sound/qcom,q6asm.h>
+
+/* ASM client callback events */
+#define CMD_PAUSE 0x0001
+#define ASM_CLIENT_EVENT_CMD_PAUSE_DONE 0x1001
+#define CMD_FLUSH 0x0002
+#define ASM_CLIENT_EVENT_CMD_FLUSH_DONE 0x1002
+#define CMD_EOS 0x0003
+#define ASM_CLIENT_EVENT_CMD_EOS_DONE 0x1003
+#define CMD_CLOSE 0x0004
+#define ASM_CLIENT_EVENT_CMD_CLOSE_DONE 0x1004
+#define CMD_OUT_FLUSH 0x0005
+#define ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE 0x1005
+#define CMD_SUSPEND 0x0006
+#define ASM_CLIENT_EVENT_CMD_SUSPEND_DONE 0x1006
+#define ASM_CLIENT_EVENT_CMD_RUN_DONE 0x1008
+#define ASM_CLIENT_EVENT_DATA_WRITE_DONE 0x1009
+#define ASM_CLIENT_EVENT_DATA_READ_DONE 0x100a
+
+enum {
+ LEGACY_PCM_MODE = 0,
+ LOW_LATENCY_PCM_MODE,
+ ULTRA_LOW_LATENCY_PCM_MODE,
+ ULL_POST_PROCESSING_PCM_MODE,
+};
#define MAX_SESSIONS 16
+#define NO_TIMESTAMP 0xFF00
+#define FORMAT_LINEAR_PCM 0x0000
typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
void *payload, void *priv);
@@ -11,6 +39,27 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
q6asm_cb cb, void *priv,
int session_id, int perf_mode);
void q6asm_audio_client_free(struct audio_client *ac);
+int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags);
+int q6asm_open_write(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample);
+
+int q6asm_open_read(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample);
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+ uint32_t rate, uint32_t channels, uint16_t bits_per_sample);
+int q6asm_read(struct audio_client *ac);
+
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ u8 channel_map[PCM_FORMAT_MAX_NUM_CHANNEL],
+ uint16_t bits_per_sample);
+int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
+ uint32_t lsw_ts);
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
+ uint32_t lsw_ts);
+int q6asm_cmd(struct audio_client *ac, int cmd);
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
int q6asm_get_session_id(struct audio_client *ac);
int q6asm_map_memory_regions(unsigned int dir,
struct audio_client *ac,
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* Re: [alsa-devel] [PATCH v4 14/24] ASoC: qdsp6: q6asm: Add support to audio stream apis
2018-03-10 2:24 ` [PATCH v4 14/24] ASoC: qdsp6: q6asm: Add support to audio stream apis srinivas.kandagatla
@ 2018-04-09 11:53 ` Rohit Kumar
0 siblings, 0 replies; 33+ messages in thread
From: Rohit Kumar @ 2018-04-09 11:53 UTC (permalink / raw)
To: srinivas.kandagatla, andy.gross, broonie, linux-arm-msm,
alsa-devel, robh+dt
Cc: mark.rutland, devicetree, bgoswami, rohkumar, gregkh, plai,
lgirdwood, tiwai, david.brown, linux-arm-kernel, spatakok,
linux-kernel
On 3/10/2018 7:54 AM, srinivas.kandagatla@linaro.org wrote:
> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
>
> This patch adds support to open, write and media format commands
> in the q6asm module.
>
> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-Tested-by: Rohit kumar <rohitkr@codeaurora.org>
> ---
> sound/soc/qcom/qdsp6/q6asm.c | 744 ++++++++++++++++++++++++++++++++++++++++++-
> sound/soc/qcom/qdsp6/q6asm.h | 49 +++
> 2 files changed, 792 insertions(+), 1 deletion(-)
>
> diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
> index 8a2ca2ff9ce8..ddbf526358bd 100644
> --- a/sound/soc/qcom/qdsp6/q6asm.c
> +++ b/sound/soc/qcom/qdsp6/q6asm.c
> @@ -9,6 +9,8 @@
> #include <linux/device.h>
> #include <linux/spinlock.h>
> #include <linux/of.h>
> +#include <linux/of_platform.h>
> +#include <uapi/sound/asound.h>
> #include <linux/delay.h>
> #include <linux/slab.h>
> #include <linux/mm.h>
> @@ -17,10 +19,36 @@
> #include "q6dsp-errno.h"
> #include "q6dsp-common.h"
>
> +#define ASM_STREAM_CMD_CLOSE 0x00010BCD
> +#define ASM_STREAM_CMD_FLUSH 0x00010BCE
> +#define ASM_SESSION_CMD_PAUSE 0x00010BD3
> +#define ASM_DATA_CMD_EOS 0x00010BDB
> +#define ASM_DEFAULT_POPP_TOPOLOGY 0x00010BE4
Use NULL topology instead of default topology.
#define ASM_NULL_POPP_TOPOLOGY 0x00010C68
> +#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
> +#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
[..]
> + q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, ac->stream_id);
> +
> + open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
> + open.mode_flags = 0x00;
> + open.mode_flags |= ASM_LEGACY_STREAM_SESSION;
> +
> + /* source endpoint : matrix */
> + open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
> + open.bits_per_sample = bits_per_sample;
> + open.postprocopo_id = ASM_DEFAULT_POPP_TOPOLOGY;
open.postprocopo_id = ASM_NULL_POPP_TOPOLOGY;
> +
> + switch (format) {
> + case FORMAT_LINEAR_PCM:
> + open.dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
> + break;
> + default:
> + dev_err(ac->dev, "Invalid format 0x%x\n", format);
> + return -EINVAL;
> + }
> +
> + rc = q6asm_ac_send_cmd_sync(ac, &open);
> + if (rc < 0)
> + return rc;
> +
> + ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
> +
> + return 0;
> +}
> +EXPORT_SYMBOL_GPL(q6asm_open_write);
^ permalink raw reply [flat|nested] 33+ messages in thread
* [PATCH v4 15/24] ASoC: qdsp6: q6core: Add q6core driver
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (13 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 14/24] ASoC: qdsp6: q6asm: Add support to audio stream apis srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-13 11:46 ` [alsa-devel] " Rohit Kumar
2018-03-10 2:24 ` [PATCH v4 16/24] ASoC: qdsp6: q6routing: Add q6routing driver srinivas.kandagatla
` (8 subsequent siblings)
23 siblings, 1 reply; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to core apr service, which is used to query
status of other static and dynamic services on the dsp.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6core.c | 381 ++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6core.h | 15 ++
4 files changed, 401 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6core.c
create mode 100644 sound/soc/qcom/qdsp6/q6core.h
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index a07181290624..488ee93ee084 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -56,6 +56,9 @@ config SND_SOC_QDSP6_ADM
config SND_SOC_QDSP6_ASM
tristate
+config SND_SOC_QDSP6_CORE
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
@@ -63,6 +66,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_AFE
select SND_SOC_QDSP6_ADM
select SND_SOC_QDSP6_ASM
+ select SND_SOC_QDSP6_CORE
help
To add support for MSM QDSP6 Soc Audio.
This will enable sound soc platform specific
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index eea962315ab3..61f089bc0d25 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -2,3 +2,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
+obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c
new file mode 100644
index 000000000000..0d9cef4b7544
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6core.c
@@ -0,0 +1,381 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/slab.h>
+#include <linux/wait.h>
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/sched.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/jiffies.h>
+#include <linux/wait.h>
+#include <linux/soc/qcom/apr.h>
+#include "q6core.h"
+#include "q6dsp-errno.h"
+
+#define ADSP_STATE_READY_TIMEOUT_MS 3000
+#define Q6_READY_TIMEOUT_MS 100
+#define AVCS_CMD_ADSP_EVENT_GET_STATE 0x0001290C
+#define AVCS_CMDRSP_ADSP_EVENT_GET_STATE 0x0001290D
+#define AVCS_GET_VERSIONS 0x00012905
+#define AVCS_GET_VERSIONS_RSP 0x00012906
+#define AVCS_CMD_GET_FWK_VERSION 0x001292c
+#define AVCS_CMDRSP_GET_FWK_VERSION 0x001292d
+
+struct avcs_svc_info {
+ uint32_t service_id;
+ uint32_t version;
+} __packed;
+
+struct avcs_cmdrsp_get_version {
+ uint32_t build_id;
+ uint32_t num_services;
+ struct avcs_svc_info svc_api_info[];
+} __packed;
+
+/* for ADSP2.8 and above */
+struct avcs_svc_api_info {
+ uint32_t service_id;
+ uint32_t api_version;
+ uint32_t api_branch_version;
+} __packed;
+
+struct avcs_cmdrsp_get_fwk_version {
+ uint32_t build_major_version;
+ uint32_t build_minor_version;
+ uint32_t build_branch_version;
+ uint32_t build_subbranch_version;
+ uint32_t num_services;
+ struct avcs_svc_api_info svc_api_info[];
+} __packed;
+
+struct q6core {
+ struct apr_device *adev;
+ wait_queue_head_t wait;
+ uint32_t avcs_state;
+ struct mutex lock;
+ bool resp_received;
+ uint32_t num_services;
+ struct avcs_cmdrsp_get_fwk_version *fwk_version;
+ struct avcs_cmdrsp_get_version *svc_version;
+ bool fwk_version_supported;
+ bool get_state_supported;
+ bool get_version_supported;
+ bool is_version_requested;
+};
+
+struct q6core *g_core;
+
+static int q6core_callback(struct apr_device *adev,
+ struct apr_client_message *data)
+{
+ struct q6core *core = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *result;
+
+ result = data->payload;
+ switch (data->opcode) {
+ case APR_BASIC_RSP_RESULT:{
+ result = data->payload;
+ switch (result->opcode) {
+ case AVCS_GET_VERSIONS:
+ if (result->status == ADSP_EUNSUPPORTED)
+ core->get_version_supported = false;
+ core->resp_received = true;
+ break;
+ case AVCS_CMD_GET_FWK_VERSION:
+ if (result->status == ADSP_EUNSUPPORTED)
+ core->fwk_version_supported = false;
+ core->resp_received = true;
+ break;
+ case AVCS_CMD_ADSP_EVENT_GET_STATE:
+ if (result->status == ADSP_EUNSUPPORTED)
+ core->get_state_supported = false;
+ core->resp_received = true;
+ break;
+ }
+ break;
+ }
+ case AVCS_CMDRSP_GET_FWK_VERSION: {
+ struct avcs_cmdrsp_get_fwk_version *fwk;
+ int bytes;
+
+ fwk = data->payload;
+ core->fwk_version_supported = true;
+ bytes = sizeof(*fwk) + fwk->num_services *
+ sizeof(fwk->svc_api_info[0]);
+
+ core->fwk_version = kzalloc(bytes, GFP_ATOMIC);
+ if (!core->fwk_version)
+ return -ENOMEM;
+
+ memcpy(core->fwk_version, data->payload, bytes);
+
+ core->resp_received = true;
+
+ break;
+ }
+ case AVCS_GET_VERSIONS_RSP: {
+ struct avcs_cmdrsp_get_version *v;
+ int len;
+
+ v = data->payload;
+ core->get_version_supported = true;
+
+ len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]);
+
+ core->svc_version = kzalloc(len, GFP_ATOMIC);
+ if (!core->svc_version)
+ return -ENOMEM;
+
+ memcpy(core->svc_version, data->payload, len);
+
+ core->resp_received = true;
+
+ break;
+ }
+ case AVCS_CMDRSP_ADSP_EVENT_GET_STATE:
+ core->get_state_supported = true;
+ core->avcs_state = result->opcode;
+
+ core->resp_received = true;
+ break;
+ default:
+ dev_err(&adev->dev, "Message id from adsp core svc: 0x%x\n",
+ data->opcode);
+ break;
+ }
+
+ if (core->resp_received)
+ wake_up(&core->wait);
+
+ return 0;
+}
+
+static int q6core_get_fwk_versions(struct q6core *core)
+{
+ struct apr_device *adev = core->adev;
+ struct apr_hdr hdr = {0};
+ int rc;
+
+ hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ hdr.pkt_size = APR_HDR_SIZE;
+ hdr.opcode = AVCS_CMD_GET_FWK_VERSION;
+
+ rc = apr_send_pkt(adev, &hdr);
+ if (rc < 0)
+ return rc;
+
+ rc = wait_event_timeout(core->wait, (core->resp_received),
+ msecs_to_jiffies(Q6_READY_TIMEOUT_MS));
+ if (rc > 0 && core->resp_received) {
+ core->resp_received = false;
+
+ if (!core->fwk_version_supported)
+ return -ENOTSUPP;
+ else
+ return 0;
+ }
+
+
+ return rc;
+}
+
+static int q6core_get_svc_versions(struct q6core *core)
+{
+ struct apr_device *adev = core->adev;
+ struct apr_hdr hdr = {0};
+ int rc;
+
+ core->get_version_supported = true;
+ hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ hdr.pkt_size = APR_HDR_SIZE;
+ hdr.opcode = AVCS_GET_VERSIONS;
+
+ rc = apr_send_pkt(adev, &hdr);
+ if (rc < 0)
+ return rc;
+
+ rc = wait_event_timeout(core->wait, (core->resp_received),
+ msecs_to_jiffies(Q6_READY_TIMEOUT_MS));
+ if (rc > 0 && core->resp_received) {
+ core->resp_received = false;
+ return 0;
+ }
+
+ return rc;
+}
+
+static bool __q6core_is_adsp_ready(struct q6core *core)
+{
+ struct apr_device *adev = core->adev;
+ struct apr_hdr hdr = {0};
+ int rc;
+
+ core->get_state_supported = false;
+
+ hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
+ hdr.pkt_size = APR_HDR_SIZE;
+ hdr.opcode = AVCS_CMD_ADSP_EVENT_GET_STATE;
+
+ rc = apr_send_pkt(adev, &hdr);
+ if (rc < 0)
+ return false;
+
+ rc = wait_event_timeout(core->wait, (core->resp_received),
+ msecs_to_jiffies(Q6_READY_TIMEOUT_MS));
+ if (rc > 0 && core->resp_received) {
+ core->resp_received = false;
+
+ if (core->avcs_state == 0x1)
+ return true;
+ }
+
+ /* assume that the adsp is up if we not support this command */
+ if (!core->get_state_supported)
+ return true;
+
+ return false;
+}
+
+/**
+ * q6core_get_svc_api_info() - Get version number of a service.
+ *
+ * @svc_id: service id of the service.
+ * @info: Valid struct pointer to fill svc api information.
+ *
+ * Return: zero on success and error code on failure or unsupported
+ */
+int q6core_get_svc_api_info(int svc_id, struct q6core_svc_api_info *ainfo)
+{
+ int i;
+ int ret = -ENOTSUPP;
+
+ if (!g_core || !ainfo)
+ return 0;
+
+ mutex_lock(&g_core->lock);
+ if (!g_core->is_version_requested) {
+ if (q6core_get_fwk_versions(g_core) == -ENOTSUPP)
+ q6core_get_svc_versions(g_core);
+ g_core->is_version_requested = true;
+ }
+
+ if (g_core->fwk_version_supported) {
+ for (i = 0; i < g_core->fwk_version->num_services; i++) {
+ struct avcs_svc_api_info *info;
+
+ info = &g_core->fwk_version->svc_api_info[i];
+ if (svc_id != info->service_id)
+ continue;
+
+ ainfo->api_version = info->api_version;
+ ainfo->api_branch_version = info->api_branch_version;
+ ret = 0;
+ break;
+ }
+ } else if (g_core->get_version_supported) {
+ for (i = 0; i < g_core->svc_version->num_services; i++) {
+ struct avcs_svc_info *info;
+
+ info = &g_core->svc_version->svc_api_info[i];
+ if (svc_id != info->service_id)
+ continue;
+
+ ainfo->api_version = info->version;
+ ainfo->api_branch_version = 0;
+ ret = 0;
+ break;
+ }
+ }
+
+ mutex_unlock(&g_core->lock);
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6core_get_svc_api_info);
+
+/**
+ * q6core_is_adsp_ready() - Get status of adsp
+ *
+ * Return: Will be an true if adsp is ready and false if not.
+ */
+bool q6core_is_adsp_ready(void)
+{
+ unsigned long timeout;
+ bool ret = false;
+
+ if (!g_core)
+ return false;
+
+ mutex_lock(&g_core->lock);
+ timeout = jiffies + msecs_to_jiffies(ADSP_STATE_READY_TIMEOUT_MS);
+ for (;;) {
+ if (__q6core_is_adsp_ready(g_core)) {
+ ret = true;
+ break;
+ }
+
+ if (!time_after(timeout, jiffies)) {
+ ret = false;
+ break;
+ }
+ }
+
+ mutex_unlock(&g_core->lock);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6core_is_adsp_ready);
+
+static int q6core_probe(struct apr_device *adev)
+{
+ g_core = kzalloc(sizeof(*g_core), GFP_KERNEL);
+ if (!g_core)
+ return -ENOMEM;
+
+ dev_set_drvdata(&adev->dev, g_core);
+
+ mutex_init(&g_core->lock);
+ g_core->adev = adev;
+ init_waitqueue_head(&g_core->wait);
+ return 0;
+}
+
+static int q6core_exit(struct apr_device *adev)
+{
+ struct q6core *core = dev_get_drvdata(&adev->dev);
+
+ if (core->fwk_version_supported)
+ kfree(core->fwk_version);
+ if (core->get_version_supported)
+ kfree(core->svc_version);
+
+ kfree(core);
+ g_core = NULL;
+
+ return 0;
+}
+
+static const struct of_device_id q6core_device_id[] = {
+ { .compatible = "qcom,q6core" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6core_device_id);
+
+static struct apr_driver qcom_q6core_driver = {
+ .probe = q6core_probe,
+ .remove = q6core_exit,
+ .callback = q6core_callback,
+ .driver = {
+ .name = "qcom-q6core",
+ .of_match_table = of_match_ptr(q6core_device_id),
+ },
+};
+
+module_apr_driver(qcom_q6core_driver);
+MODULE_DESCRIPTION("q6 core");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6core.h b/sound/soc/qcom/qdsp6/q6core.h
new file mode 100644
index 000000000000..7678abd8fd46
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6core.h
@@ -0,0 +1,15 @@
+// SPDX-License-Identifier: GPL-2.0
+
+#ifndef __Q6CORE_H__
+#define __Q6CORE_H__
+
+struct q6core_svc_api_info {
+ uint32_t service_id;
+ uint32_t api_version;
+ uint32_t api_branch_version;
+};
+
+bool q6core_is_adsp_ready(void);
+int q6core_get_svc_api_info(int svc_id, struct q6core_svc_api_info *ainfo);
+
+#endif /* __Q6CORE_H__ */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* Re: [alsa-devel] [PATCH v4 15/24] ASoC: qdsp6: q6core: Add q6core driver
2018-03-10 2:24 ` [PATCH v4 15/24] ASoC: qdsp6: q6core: Add q6core driver srinivas.kandagatla
@ 2018-03-13 11:46 ` Rohit Kumar
2018-03-13 13:55 ` Srinivas Kandagatla
0 siblings, 1 reply; 33+ messages in thread
From: Rohit Kumar @ 2018-03-13 11:46 UTC (permalink / raw)
To: srinivas.kandagatla, andy.gross, broonie, linux-arm-msm,
alsa-devel, robh+dt
Cc: mark.rutland, devicetree, bgoswami, rohkumar, gregkh, plai,
lgirdwood, tiwai, david.brown, linux-arm-kernel, spatakok,
linux-kernel
On 3/10/2018 7:54 AM, srinivas.kandagatla@linaro.org wrote:
> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
[..]
> +static int q6core_get_svc_versions(struct q6core *core)
> +{
> + struct apr_device *adev = core->adev;
> + struct apr_hdr hdr = {0};
> + int rc;
> +
> + core->get_version_supported = true;
core->get_version_supported should be set to true only after we get proper response from adsp in callback(). In case,we get wrong response from adsp, memory for g_core->svc_version
will not get allocated and there will be NULL pointer dereference in q6core_get_svc_api_info() in below statement
+ } else if (g_core->get_version_supported) {
+ for (i = 0; i < g_core->svc_version->num_services; i++) {
> + hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
> + APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
> + hdr.pkt_size = APR_HDR_SIZE;
> + hdr.opcode = AVCS_GET_VERSIONS;
> +
> + rc = apr_send_pkt(adev, &hdr);
> + if (rc < 0)
> + return rc;
> +
^ permalink raw reply [flat|nested] 33+ messages in thread* Re: [alsa-devel] [PATCH v4 15/24] ASoC: qdsp6: q6core: Add q6core driver
2018-03-13 11:46 ` [alsa-devel] " Rohit Kumar
@ 2018-03-13 13:55 ` Srinivas Kandagatla
0 siblings, 0 replies; 33+ messages in thread
From: Srinivas Kandagatla @ 2018-03-13 13:55 UTC (permalink / raw)
To: Rohit Kumar, andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: mark.rutland, devicetree, bgoswami, rohkumar, gregkh, plai,
lgirdwood, tiwai, david.brown, linux-arm-kernel, spatakok,
linux-kernel
Thanks Rohit for review comments,
On 13/03/18 11:46, Rohit Kumar wrote:
>
>
> On 3/10/2018 7:54 AM, srinivas.kandagatla@linaro.org wrote:
>> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
> [..]
>> +static int q6core_get_svc_versions(struct q6core *core)
>> +{
>> + struct apr_device *adev = core->adev;
>> + struct apr_hdr hdr = {0};
>> + int rc;
>> +
>> + core->get_version_supported = true;
>
> core->get_version_supported should be set to true only after we get
> proper response from adsp in callback(). In case,we get wrong response
> from adsp, memory for g_core->svc_version
> will not get allocated and there will be NULL pointer dereference in
> q6core_get_svc_api_info() in below statement
I totally agree with you , this line seems to be a leftover from
previous patches, It should be removed.
thanks,
srini
> + } else if (g_core->get_version_supported) {
> + for (i = 0; i < g_core->svc_version->num_services; i++) {
>
>
>> + hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
>> + APR_HDR_LEN(APR_HDR_SIZE), APR_PKT_VER);
>> + hdr.pkt_size = APR_HDR_SIZE;
>> + hdr.opcode = AVCS_GET_VERSIONS;
>> +
>> + rc = apr_send_pkt(adev, &hdr);
>> + if (rc < 0)
>> + return rc;
>> +
^ permalink raw reply [flat|nested] 33+ messages in thread
* [PATCH v4 16/24] ASoC: qdsp6: q6routing: Add q6routing driver
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (14 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 15/24] ASoC: qdsp6: q6core: Add q6core driver srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 17/24] ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers srinivas.kandagatla
` (7 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to q6 routing driver which configures route
between ASM and AFE module using ADM apis.
This driver uses dapm widgets to setup the matrix between AFE ports and
ASM streams.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6routing.c | 392 +++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6routing.h | 9 +
4 files changed, 406 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6routing.c
create mode 100644 sound/soc/qcom/qdsp6/q6routing.h
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 488ee93ee084..7a868a6fac1b 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -53,6 +53,9 @@ config SND_SOC_QDSP6_AFE
config SND_SOC_QDSP6_ADM
tristate
+config SND_SOC_QDSP6_ROUTING
+ tristate
+
config SND_SOC_QDSP6_ASM
tristate
@@ -65,6 +68,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_AFE
select SND_SOC_QDSP6_ADM
+ select SND_SOC_QDSP6_ROUTING
select SND_SOC_QDSP6_ASM
select SND_SOC_QDSP6_CORE
help
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 61f089bc0d25..6640bceccc18 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -1,5 +1,6 @@
obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
+obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
new file mode 100644
index 000000000000..b72bd9045fea
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -0,0 +1,392 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_platform.h>
+#include <linux/bitops.h>
+#include <linux/component.h>
+#include <linux/mutex.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/control.h>
+#include <sound/asound.h>
+#include <sound/pcm_params.h>
+#include "q6afe.h"
+#include "q6asm.h"
+#include "q6adm.h"
+#include "q6routing.h"
+
+struct session_data {
+ int state;
+ int port_id;
+ int path_type;
+ int app_type;
+ int acdb_id;
+ int sample_rate;
+ int bits_per_sample;
+ int channels;
+ int perf_mode;
+ int numcopps;
+ int fedai_id;
+ unsigned long copp_map;
+};
+
+struct msm_routing_data {
+ struct session_data sessions[MAX_SESSIONS];
+ struct session_data port_data[AFE_MAX_PORTS];
+ struct device *dev;
+ struct mutex lock;
+};
+
+static struct msm_routing_data *routing_data;
+
+/**
+ * q6routing_stream_open() - Register a new stream for route setup
+ *
+ * @fedai_id: Frontend dai id.
+ * @perf_mode: Performance mode.
+ * @stream_id: ASM stream id to map.
+ * @stream_type: Direction of stream
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6routing_stream_open(int fedai_id, int perf_mode,
+ int stream_id, int stream_type)
+{
+ int j, topology, num_copps = 0;
+ struct route_payload payload;
+ int copp_idx;
+ struct session_data *session, *pdata;
+
+ if (!routing_data) {
+ pr_err("Routing driver not yet ready\n");
+ return -EINVAL;
+ }
+
+ session = &routing_data->sessions[stream_id - 1];
+ pdata = &routing_data->port_data[session->port_id];
+
+ mutex_lock(&routing_data->lock);
+ session->fedai_id = fedai_id;
+
+ session->path_type = pdata->path_type;
+ session->sample_rate = pdata->sample_rate;
+ session->channels = pdata->channels;
+ session->bits_per_sample = pdata->bits_per_sample;
+
+ payload.num_copps = 0; /* only RX needs to use payload */
+ topology = NULL_COPP_TOPOLOGY;
+ copp_idx = q6adm_open(routing_data->dev, session->port_id,
+ session->path_type, session->sample_rate,
+ session->channels, topology, perf_mode,
+ session->bits_per_sample, 0, 0);
+
+ if (copp_idx < 0) {
+ mutex_unlock(&routing_data->lock);
+ return -EINVAL;
+ }
+
+ set_bit(copp_idx, &session->copp_map);
+
+ for_each_set_bit(j, &session->copp_map, MAX_COPPS_PER_PORT) {
+ payload.port_id[num_copps] = session->port_id;
+ payload.copp_idx[num_copps] = j;
+ num_copps++;
+ }
+
+ if (num_copps) {
+ payload.num_copps = num_copps;
+ payload.session_id = stream_id;
+ q6adm_matrix_map(routing_data->dev, session->path_type,
+ payload, perf_mode);
+ }
+ mutex_unlock(&routing_data->lock);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6routing_stream_open);
+
+static struct session_data *get_session_from_id(struct msm_routing_data *data,
+ int fedai_id)
+{
+ int i;
+
+ for (i = 0; i < MAX_SESSIONS; i++) {
+ if (fedai_id == data->sessions[i].fedai_id)
+ return &data->sessions[i];
+ }
+
+ return NULL;
+}
+/**
+ * q6routing_stream_close() - Deregister a stream
+ *
+ * @fedai_id: Frontend dai id.
+ * @stream_type: Direction of stream
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+void q6routing_stream_close(int fedai_id, int stream_type)
+{
+ struct session_data *session;
+ int idx;
+
+ session = get_session_from_id(routing_data, fedai_id);
+ if (!session)
+ return;
+
+ for_each_set_bit(idx, &session->copp_map, MAX_COPPS_PER_PORT)
+ q6adm_close(routing_data->dev, session->port_id,
+ session->perf_mode, idx);
+
+ session->fedai_id = -1;
+ session->copp_map = 0;
+}
+EXPORT_SYMBOL_GPL(q6routing_stream_close);
+
+static int msm_routing_get_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int session_id = mc->shift;
+ struct snd_soc_platform *platform = snd_soc_dapm_to_platform(dapm);
+ struct msm_routing_data *priv = dev_get_drvdata(platform->dev);
+ struct session_data *session = &priv->sessions[session_id];
+
+ if (session->port_id == mc->reg)
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_platform *platform = snd_soc_dapm_to_platform(dapm);
+ struct msm_routing_data *data = dev_get_drvdata(platform->dev);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_dapm_update *update = NULL;
+ int be_id = mc->reg;
+ int session_id = mc->shift;
+ struct session_data *session = &data->sessions[session_id];
+
+ if (ucontrol->value.integer.value[0]) {
+ session->port_id = be_id;
+ snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update);
+ } else {
+ session->port_id = -1;
+ snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update);
+ }
+
+ return 1;
+}
+
+static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
+ /* Frontend AIF */
+ SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0),
+
+ /* Mixer definitions */
+ SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0,
+ hdmi_mixer_controls,
+ ARRAY_SIZE(hdmi_mixer_controls)),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"HDMI Mixer", "MultiMedia1", "MM_DL1"},
+ {"HDMI Mixer", "MultiMedia2", "MM_DL2"},
+ {"HDMI Mixer", "MultiMedia3", "MM_DL3"},
+ {"HDMI Mixer", "MultiMedia4", "MM_DL4"},
+ {"HDMI Mixer", "MultiMedia5", "MM_DL5"},
+ {"HDMI Mixer", "MultiMedia6", "MM_DL6"},
+ {"HDMI Mixer", "MultiMedia7", "MM_DL7"},
+ {"HDMI Mixer", "MultiMedia8", "MM_DL8"},
+ {"HDMI_RX", NULL, "HDMI Mixer"},
+};
+
+static int routing_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ unsigned int be_id = rtd->cpu_dai->id;
+ struct snd_soc_platform *platform = rtd->platform;
+ struct msm_routing_data *data = dev_get_drvdata(platform->dev);
+ struct session_data *session;
+ int path_type;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ path_type = ADM_PATH_PLAYBACK;
+ else
+ path_type = ADM_PATH_LIVE_REC;
+
+ if (be_id > AFE_MAX_PORTS)
+ return -EINVAL;
+
+ session = &data->port_data[be_id];
+
+ mutex_lock(&data->lock);
+
+ session->path_type = path_type;
+ session->sample_rate = params_rate(params);
+ session->channels = params_channels(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ session->bits_per_sample = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ session->bits_per_sample = 24;
+ break;
+ default:
+ break;
+ }
+
+ mutex_unlock(&data->lock);
+ return 0;
+}
+
+static struct snd_pcm_ops q6pcm_routing_ops = {
+ .hw_params = routing_hw_params,
+};
+
+static int msm_routing_probe(struct snd_soc_platform *platform)
+{
+ int i;
+
+ for (i = 0; i < MAX_SESSIONS; i++)
+ routing_data->sessions[i].port_id = -1;
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver msm_soc_routing_platform = {
+ .ops = &q6pcm_routing_ops,
+ .probe = msm_routing_probe,
+ .component_driver = {
+ .name = "q6routing-component",
+ .dapm_widgets = msm_qdsp6_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(msm_qdsp6_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
+ },
+};
+
+static int q6routing_dai_bind(struct device *dev, struct device *master,
+ void *data)
+{
+ routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL);
+ if (!routing_data)
+ return -ENOMEM;
+
+ routing_data->dev = dev;
+
+ mutex_init(&routing_data->lock);
+ dev_set_drvdata(dev, routing_data);
+
+ return snd_soc_register_platform(dev,
+ &msm_soc_routing_platform);
+}
+static void q6routing_dai_unbind(struct device *dev, struct device *master,
+ void *d)
+{
+ struct msm_routing_data *data = dev_get_drvdata(dev);
+
+ snd_soc_unregister_platform(dev);
+
+ kfree(data);
+
+ routing_data = NULL;
+}
+
+static const struct component_ops q6routing_dai_comp_ops = {
+ .bind = q6routing_dai_bind,
+ .unbind = q6routing_dai_unbind,
+};
+
+static int q6pcm_routing_probe(struct platform_device *pdev)
+{
+ return component_add(&pdev->dev, &q6routing_dai_comp_ops);
+}
+
+static int q6pcm_routing_remove(struct platform_device *pdev)
+{
+ component_del(&pdev->dev, &q6routing_dai_comp_ops);
+ return 0;
+}
+
+static struct platform_driver q6pcm_routing_platform_driver = {
+ .driver = {
+ .name = "q6routing",
+ },
+ .probe = q6pcm_routing_probe,
+ .remove = q6pcm_routing_remove,
+};
+module_platform_driver(q6pcm_routing_platform_driver);
+
+MODULE_DESCRIPTION("Q6 Routing platform");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6routing.h b/sound/soc/qcom/qdsp6/q6routing.h
new file mode 100644
index 000000000000..aa41e4f45742
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6routing.h
@@ -0,0 +1,9 @@
+// SPDX-License-Identifier: GPL-2.0
+#ifndef _Q6_PCM_ROUTING_H
+#define _Q6_PCM_ROUTING_H
+
+int q6routing_stream_open(int fedai_id, int perf_mode,
+ int stream_id, int stream_type);
+void q6routing_stream_close(int fedai_id, int stream_type);
+
+#endif /*_Q6_PCM_ROUTING_H */
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 17/24] ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (15 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 16/24] ASoC: qdsp6: q6routing: Add q6routing driver srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 18/24] ASoC: qdsp6: q6routing: Add support to MI2S Mixers srinivas.kandagatla
` (6 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to SLIMBus related mixers to control mux between
ASM stream and AFE port.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/qdsp6/q6routing.c | 261 +++++++++++++++++++++++++++++++++++++++
1 file changed, 261 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index b72bd9045fea..c6be775167b8 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -232,6 +232,180 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
msm_routing_put_audio_mixer),
};
+static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_0_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_1_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_1_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_2_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_2_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_2_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_2_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_2_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_2_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_2_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_2_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_2_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_3_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_3_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_3_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_3_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_3_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_3_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_3_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_3_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_3_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_4_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_4_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_4_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_4_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_5_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_5_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_5_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_5_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_5_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_5_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_5_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_5_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_5_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new slimbus_6_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_6_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_6_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_6_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_6_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_6_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_6_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_6_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_6_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
/* Frontend AIF */
SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
@@ -255,6 +429,28 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0,
hdmi_mixer_controls,
ARRAY_SIZE(hdmi_mixer_controls)),
+
+ SND_SOC_DAPM_MIXER("SLIMBUS_0_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_rx_mixer_controls,
+ ARRAY_SIZE(slimbus_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_1_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_1_rx_mixer_controls,
+ ARRAY_SIZE(slimbus_1_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_2_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_2_rx_mixer_controls,
+ ARRAY_SIZE(slimbus_2_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_3_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_3_rx_mixer_controls,
+ ARRAY_SIZE(slimbus_3_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_4_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_4_rx_mixer_controls,
+ ARRAY_SIZE(slimbus_4_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_5_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_5_rx_mixer_controls,
+ ARRAY_SIZE(slimbus_5_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SLIMBUS_6_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ slimbus_6_rx_mixer_controls,
+ ARRAY_SIZE(slimbus_6_rx_mixer_controls)),
};
static const struct snd_soc_dapm_route intercon[] = {
@@ -267,6 +463,71 @@ static const struct snd_soc_dapm_route intercon[] = {
{"HDMI Mixer", "MultiMedia7", "MM_DL7"},
{"HDMI Mixer", "MultiMedia8", "MM_DL8"},
{"HDMI_RX", NULL, "HDMI Mixer"},
+
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Audio Mixer"},
+
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_1_RX", NULL, "SLIMBUS_1_RX Audio Mixer"},
+
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_2_RX", NULL, "SLIMBUS_2_RX Audio Mixer"},
+
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_3_RX", NULL, "SLIMBUS_3_RX Audio Mixer"},
+
+ {"SLIMBUS_4_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_4_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_4_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_4_RX", NULL, "SLIMBUS_4_RX Audio Mixer"},
+
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_5_RX", NULL, "SLIMBUS_5_RX Audio Mixer"},
+
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_6_RX", NULL, "SLIMBUS_6_RX Audio Mixer"},
};
static int routing_hw_params(struct snd_pcm_substream *substream,
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 18/24] ASoC: qdsp6: q6routing: Add support to MI2S Mixers
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (16 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 17/24] ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-05-21 15:47 ` Applied "ASoC: qdsp6: q6routing: Add support to MI2S Mixers" to the asoc tree Mark Brown
2018-03-10 2:24 ` [PATCH v4 19/24] ASoC: qdsp6: q6afe: Add q6afe dai driver srinivas.kandagatla
` (5 subsequent siblings)
23 siblings, 1 reply; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch add support to MI2S mixers required to select path between
ASM stream and AFE ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/qdsp6/q6routing.c | 329 +++++++++++++++++++++++++++++++++++++++
1 file changed, 329 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index c6be775167b8..710c2ae652c7 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -232,6 +232,103 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
msm_routing_put_audio_mixer),
};
+static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new secondary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+
static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_0_RX,
MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
@@ -406,6 +503,126 @@ static const struct snd_kcontrol_new slimbus_6_rx_mixer_controls[] = {
msm_routing_put_audio_mixer),
};
+static const struct snd_kcontrol_new mmul1_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul2_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul3_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul4_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul5_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul6_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul7_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul8_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
/* Frontend AIF */
SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
@@ -451,6 +668,35 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
SND_SOC_DAPM_MIXER("SLIMBUS_6_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
slimbus_6_rx_mixer_controls,
ARRAY_SIZE(slimbus_6_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("PRI_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ primary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(primary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SEC_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ secondary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(secondary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("QUAT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ quaternary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(quaternary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("TERT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ tertiary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(tertiary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia1 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul2_mixer_controls, ARRAY_SIZE(mmul2_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia3 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul3_mixer_controls, ARRAY_SIZE(mmul3_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia4 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul4_mixer_controls, ARRAY_SIZE(mmul4_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia5 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul5_mixer_controls, ARRAY_SIZE(mmul5_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia6 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul6_mixer_controls, ARRAY_SIZE(mmul6_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia7 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul7_mixer_controls, ARRAY_SIZE(mmul7_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)),
+
};
static const struct snd_soc_dapm_route intercon[] = {
@@ -528,6 +774,89 @@ static const struct snd_soc_dapm_route intercon[] = {
{"SLIMBUS_6_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
{"SLIMBUS_6_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
{"SLIMBUS_6_RX", NULL, "SLIMBUS_6_RX Audio Mixer"},
+
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"QUAT_MI2S_RX", NULL, "QUAT_MI2S_RX Audio Mixer"},
+
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"TERT_MI2S_RX", NULL, "TERT_MI2S_RX Audio Mixer"},
+
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia6", "MM_DL5"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia8", "MM_DL7"},
+ {"SEC_MI2S_RX", NULL, "SEC_MI2S_RX Audio Mixer"},
+
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"PRI_MI2S_RX", NULL, "PRI_MI2S_RX Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia1 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia1 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia1 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia2 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia2 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia2 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia2 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia3 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia3 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia3 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia3 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia4 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia4 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia4 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia4 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia5 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia5 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia5 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia5 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia6 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia6 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia6 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia6 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia7 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia7 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia7 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia7 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia8 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia8 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia8 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia8 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MM_UL1", NULL, "MultiMedia1 Mixer"},
+ {"MM_UL2", NULL, "MultiMedia2 Mixer"},
+ {"MM_UL3", NULL, "MultiMedia3 Mixer"},
+ {"MM_UL4", NULL, "MultiMedia4 Mixer"},
+ {"MM_UL5", NULL, "MultiMedia5 Mixer"},
+ {"MM_UL6", NULL, "MultiMedia6 Mixer"},
+ {"MM_UL7", NULL, "MultiMedia7 Mixer"},
+ {"MM_UL8", NULL, "MultiMedia8 Mixer"},
};
static int routing_hw_params(struct snd_pcm_substream *substream,
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* Applied "ASoC: qdsp6: q6routing: Add support to MI2S Mixers" to the asoc tree
2018-03-10 2:24 ` [PATCH v4 18/24] ASoC: qdsp6: q6routing: Add support to MI2S Mixers srinivas.kandagatla
@ 2018-05-21 15:47 ` Mark Brown
0 siblings, 0 replies; 33+ messages in thread
From: Mark Brown @ 2018-05-21 15:47 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, andy.gross, broonie, linux-arm-msm, alsa-devel,
robh+dt, mark.rutland, devicetree, bgoswami, rohkumar, gregkh,
plai, lgirdwood, tiwai, david.brown, srinivas.kandagatla,
linux-arm-kernel, spatakok, linux-kernel, alsa-devel
The patch
ASoC: qdsp6: q6routing: Add support to MI2S Mixers
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From c8add3fd49fe1547bf2d3209a42543657caa3dad Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:56:06 +0100
Subject: [PATCH] ASoC: qdsp6: q6routing: Add support to MI2S Mixers
This patch add support to MI2S mixers required to select path between
ASM stream and AFE ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6routing.c | 329 +++++++++++++++++++++++++++++++
1 file changed, 329 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 43086913060a..08c25c26adf4 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -241,6 +241,103 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
msm_routing_put_audio_mixer),
};
+static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new secondary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+
static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_0_RX,
MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
@@ -415,6 +512,126 @@ static const struct snd_kcontrol_new slimbus_6_rx_mixer_controls[] = {
msm_routing_put_audio_mixer),
};
+static const struct snd_kcontrol_new mmul1_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul2_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul3_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul4_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul5_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul6_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul7_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul8_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
/* Frontend AIF */
SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
@@ -460,6 +677,35 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
SND_SOC_DAPM_MIXER("SLIMBUS_6_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
slimbus_6_rx_mixer_controls,
ARRAY_SIZE(slimbus_6_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("PRI_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ primary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(primary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SEC_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ secondary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(secondary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("QUAT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ quaternary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(quaternary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("TERT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ tertiary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(tertiary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia1 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul2_mixer_controls, ARRAY_SIZE(mmul2_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia3 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul3_mixer_controls, ARRAY_SIZE(mmul3_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia4 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul4_mixer_controls, ARRAY_SIZE(mmul4_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia5 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul5_mixer_controls, ARRAY_SIZE(mmul5_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia6 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul6_mixer_controls, ARRAY_SIZE(mmul6_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia7 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul7_mixer_controls, ARRAY_SIZE(mmul7_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)),
+
};
static const struct snd_soc_dapm_route intercon[] = {
@@ -537,6 +783,89 @@ static const struct snd_soc_dapm_route intercon[] = {
{"SLIMBUS_6_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
{"SLIMBUS_6_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
{"SLIMBUS_6_RX", NULL, "SLIMBUS_6_RX Audio Mixer"},
+
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"QUAT_MI2S_RX", NULL, "QUAT_MI2S_RX Audio Mixer"},
+
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"TERT_MI2S_RX", NULL, "TERT_MI2S_RX Audio Mixer"},
+
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia6", "MM_DL5"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia8", "MM_DL7"},
+ {"SEC_MI2S_RX", NULL, "SEC_MI2S_RX Audio Mixer"},
+
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"PRI_MI2S_RX", NULL, "PRI_MI2S_RX Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia1 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia1 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia1 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia2 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia2 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia2 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia2 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia3 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia3 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia3 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia3 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia4 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia4 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia4 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia4 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia5 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia5 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia5 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia5 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia6 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia6 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia6 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia6 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia7 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia7 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia7 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia7 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia8 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia8 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia8 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia8 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MM_UL1", NULL, "MultiMedia1 Mixer"},
+ {"MM_UL2", NULL, "MultiMedia2 Mixer"},
+ {"MM_UL3", NULL, "MultiMedia3 Mixer"},
+ {"MM_UL4", NULL, "MultiMedia4 Mixer"},
+ {"MM_UL5", NULL, "MultiMedia5 Mixer"},
+ {"MM_UL6", NULL, "MultiMedia6 Mixer"},
+ {"MM_UL7", NULL, "MultiMedia7 Mixer"},
+ {"MM_UL8", NULL, "MultiMedia8 Mixer"},
};
static int routing_hw_params(struct snd_pcm_substream *substream,
--
2.17.0
^ permalink raw reply related [flat|nested] 33+ messages in thread
* [PATCH v4 19/24] ASoC: qdsp6: q6afe: Add q6afe dai driver
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (17 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 18/24] ASoC: qdsp6: q6routing: Add support to MI2S Mixers srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 20/24] ASoC: qdsp6: q6asm: Add q6asm " srinivas.kandagatla
` (4 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to q6afe backend dais driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6afe-dai.c | 746 +++++++++++++++++++++++++++++++++++++++
3 files changed, 751 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 7a868a6fac1b..9b630686eb19 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -50,6 +50,9 @@ config SND_SOC_QDSP6_COMMON
config SND_SOC_QDSP6_AFE
tristate
+config SND_SOC_QDSP6_AFE_DAI
+ tristate
+
config SND_SOC_QDSP6_ADM
tristate
@@ -67,6 +70,7 @@ config SND_SOC_QDSP6
depends on QCOM_APR && HAS_DMA
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_AFE
+ select SND_SOC_QDSP6_AFE_DAI
select SND_SOC_QDSP6_ADM
select SND_SOC_QDSP6_ROUTING
select SND_SOC_QDSP6_ASM
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 6640bceccc18..6a8ae698f22f 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -1,5 +1,6 @@
obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
+obj-$(CONFIG_SND_SOC_QDSP6_AFE_DAI) += q6afe-dai.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
new file mode 100644
index 000000000000..5cb8b59a8cd9
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -0,0 +1,746 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2016, The Linux Foundation
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/err.h>
+#include <linux/init.h>
+#include <linux/component.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include "q6afe.h"
+
+struct q6afe_dai_priv_data {
+ uint32_t sd_line_mask;
+};
+
+struct q6afe_dai_data {
+ struct q6afe_port *port[AFE_PORT_MAX];
+ struct q6afe_port_config port_config[AFE_PORT_MAX];
+ bool is_port_started[AFE_PORT_MAX];
+ struct q6afe_dai_priv_data priv[AFE_PORT_MAX];
+};
+
+static int q6slim_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim;
+
+ slim->num_channels = params_channels(params);
+ slim->sample_rate = params_rate(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_SPECIAL:
+ slim->bit_width = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ slim->bit_width = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ slim->bit_width = 32;
+ break;
+ default:
+ pr_err("%s: format %d\n",
+ __func__, params_format(params));
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int q6hdmi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int channels = params_channels(params);
+ struct q6afe_hdmi_cfg *hdmi = &dai_data->port_config[dai->id].hdmi;
+
+ hdmi->sample_rate = params_rate(params);
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hdmi->bit_width = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ hdmi->bit_width = 24;
+ break;
+ }
+
+ /* HDMI spec CEA-861-E: Table 28 Audio InfoFrame Data Byte 4 */
+ switch (channels) {
+ case 2:
+ hdmi->channel_allocation = 0;
+ break;
+ case 3:
+ hdmi->channel_allocation = 0x02;
+ break;
+ case 4:
+ hdmi->channel_allocation = 0x06;
+ break;
+ case 5:
+ hdmi->channel_allocation = 0x0A;
+ break;
+ case 6:
+ hdmi->channel_allocation = 0x0B;
+ break;
+ case 7:
+ hdmi->channel_allocation = 0x12;
+ break;
+ case 8:
+ hdmi->channel_allocation = 0x13;
+ break;
+ default:
+ dev_err(dai->dev, "invalid Channels = %u\n", channels);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int q6i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_i2s_cfg *i2s = &dai_data->port_config[dai->id].i2s_cfg;
+
+ i2s->sample_rate = params_rate(params);
+ i2s->bit_width = params_width(params);
+ i2s->num_channels = params_channels(params);
+ i2s->sd_line_mask = dai_data->priv[dai->id].sd_line_mask;
+
+ return 0;
+}
+
+static int q6i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_i2s_cfg *i2s = &dai_data->port_config[dai->id].i2s_cfg;
+
+ i2s->fmt = fmt;
+
+ return 0;
+}
+
+static void q6afe_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc;
+
+ rc = q6afe_port_stop(dai_data->port[dai->id]);
+ if (rc < 0)
+ dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
+
+ dai_data->is_port_started[dai->id] = false;
+
+}
+
+static int q6afe_mi2s_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc;
+
+ if (dai_data->is_port_started[dai->id]) {
+ /* stop the port and restart with new port config */
+ rc = q6afe_port_stop(dai_data->port[dai->id]);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
+ return rc;
+ }
+ }
+
+ rc = q6afe_i2s_port_prepare(dai_data->port[dai->id],
+ &dai_data->port_config[dai->id].i2s_cfg);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to prepare AFE port %x\n", dai->id);
+ return rc;
+ }
+
+ rc = q6afe_port_start(dai_data->port[dai->id]);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to start AFE port %x\n", dai->id);
+ return rc;
+ }
+ dai_data->is_port_started[dai->id] = true;
+
+ return 0;
+}
+
+static int q6afe_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc;
+
+ if (dai_data->is_port_started[dai->id]) {
+ /* stop the port and restart with new port config */
+ rc = q6afe_port_stop(dai_data->port[dai->id]);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
+ return rc;
+ }
+ }
+
+ if (dai->id == HDMI_RX)
+ q6afe_hdmi_port_prepare(dai_data->port[dai->id],
+ &dai_data->port_config[dai->id].hdmi);
+ else if (dai->id >= SLIMBUS_0_RX && dai->id <= SLIMBUS_6_TX)
+ q6afe_slim_port_prepare(dai_data->port[dai->id],
+ &dai_data->port_config[dai->id].slim);
+
+ rc = q6afe_port_start(dai_data->port[dai->id]);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to start AFE port %x\n", dai->id);
+ return rc;
+ }
+ dai_data->is_port_started[dai->id] = true;
+
+ return 0;
+}
+
+static int q6slim_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id];
+ int i;
+
+ if (!rx_slot) {
+ pr_err("%s: rx slot not found\n", __func__);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < rx_num; i++) {
+ pcfg->slim.ch_mapping[i] = rx_slot[i];
+ pr_debug("%s: find number of channels[%d] ch[%d]\n",
+ __func__, i, rx_slot[i]);
+ }
+
+ pcfg->slim.num_channels = rx_num;
+
+ pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__,
+ (dai->id - SLIMBUS_0_RX) / 2, rx_num,
+ pcfg->slim.ch_mapping[0],
+ pcfg->slim.ch_mapping[1]);
+
+ return 0;
+}
+
+static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_port *port = dai_data->port[dai->id];
+
+ switch (clk_id) {
+ case LPAIF_DIG_CLK:
+ return q6afe_port_set_sysclk(port, clk_id, 0, 5, freq, dir);
+ case LPAIF_BIT_CLK:
+ case LPAIF_OSR_CLK:
+ return q6afe_port_set_sysclk(port, clk_id,
+ Q6AFE_LPASS_CLK_SRC_INTERNAL,
+ Q6AFE_LPASS_CLK_ROOT_DEFAULT,
+ freq, dir);
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
+ {"HDMI Playback", NULL, "HDMI_RX"},
+ {"Slimbus1 Playback", NULL, "SLIMBUS_1_RX"},
+ {"Slimbus2 Playback", NULL, "SLIMBUS_2_RX"},
+ {"Slimbus3 Playback", NULL, "SLIMBUS_3_RX"},
+ {"Slimbus4 Playback", NULL, "SLIMBUS_4_RX"},
+ {"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"},
+ {"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"},
+
+ {"Primary MI2S Playback", NULL, "PRI_MI2S_RX"},
+ {"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"},
+ {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"},
+ {"Quaternary MI2S Playback", NULL, "QUAT_MI2S_RX"},
+
+ {"TERT_MI2S_TX", NULL, "Tertiary MI2S Capture"},
+ {"PRI_MI2S_TX", NULL, "Primary MI2S Capture"},
+ {"SEC_MI2S_TX", NULL, "Secondary MI2S Capture"},
+ {"QUAT_MI2S_TX", NULL, "Quaternary MI2S Capture"},
+};
+
+static struct snd_soc_dai_ops q6hdmi_ops = {
+ .prepare = q6afe_dai_prepare,
+ .hw_params = q6hdmi_hw_params,
+ .shutdown = q6afe_dai_shutdown,
+};
+
+static struct snd_soc_dai_ops q6i2s_ops = {
+ .prepare = q6afe_mi2s_prepare,
+ .hw_params = q6i2s_hw_params,
+ .set_fmt = q6i2s_set_fmt,
+ .shutdown = q6afe_dai_shutdown,
+ .set_sysclk = q6afe_mi2s_set_sysclk,
+};
+
+static struct snd_soc_dai_ops q6slim_ops = {
+ .prepare = q6afe_dai_prepare,
+ .hw_params = q6slim_hw_params,
+ .shutdown = q6afe_dai_shutdown,
+ .set_channel_map = q6slim_set_channel_map,
+};
+
+static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct snd_soc_dapm_context *dapm;
+ struct q6afe_port *port;
+
+ dapm = snd_soc_component_get_dapm(dai->component);
+
+ port = q6afe_port_get_from_id(dai->dev, dai->id);
+ if (IS_ERR(port)) {
+ dev_err(dai->dev, "Unable to get afe port\n");
+ return -EINVAL;
+ }
+ dai_data->port[dai->id] = port;
+
+ return 0;
+}
+
+static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ q6afe_port_put(dai_data->port[dai->id]);
+ dai_data->port[dai->id] = NULL;
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver q6afe_dais[] = {
+ {
+ .playback = {
+ .stream_name = "HDMI Playback",
+ .rates = SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 2,
+ .channels_max = 8,
+ .rate_max = 192000,
+ .rate_min = 48000,
+ },
+ .ops = &q6hdmi_ops,
+ .id = HDMI_RX,
+ .name = "HDMI",
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .name = "SLIMBUS_0_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_0_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ .playback = {
+ .stream_name = "Slimbus Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ }, {
+ .playback = {
+ .stream_name = "Slimbus1 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_1_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_1_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus2 Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_2_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_2_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus3 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_3_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_3_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus4 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_4_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_4_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus5 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_5_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_5_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus6 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .ops = &q6slim_ops,
+ .name = "SLIMBUS_6_RX",
+ .id = SLIMBUS_6_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Primary MI2S Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = PRIMARY_MI2S_RX,
+ .name = "PRI_MI2S_RX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .capture = {
+ .stream_name = "Primary MI2S Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = PRIMARY_MI2S_TX,
+ .name = "PRI_MI2S_TX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Secondary MI2S Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .name = "SEC_MI2S_RX",
+ .id = SECONDARY_MI2S_RX,
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .capture = {
+ .stream_name = "Secondary MI2S Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = SECONDARY_MI2S_TX,
+ .name = "SEC_MI2S_TX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Tertiary MI2S Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .name = "TERT_MI2S_RX",
+ .id = TERTIARY_MI2S_RX,
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .capture = {
+ .stream_name = "Tertiary MI2S Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = TERTIARY_MI2S_TX,
+ .name = "TERT_MI2S_TX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Quaternary MI2S Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .name = "QUAT_MI2S_RX",
+ .id = QUATERNARY_MI2S_RX,
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .capture = {
+ .stream_name = "Quaternary MI2S Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = QUATERNARY_MI2S_TX,
+ .name = "QUAT_MI2S_TX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ },
+};
+
+static int q6afe_of_xlate_dai_name(struct snd_soc_component *component,
+ struct of_phandle_args *args,
+ const char **dai_name)
+{
+ int id = args->args[0];
+ int ret = -EINVAL;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(q6afe_dais); i++) {
+ if (q6afe_dais[i].id == id) {
+ *dai_name = q6afe_dais[i].name;
+ ret = 0;
+ break;
+ }
+ }
+
+ return ret;
+}
+
+static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
+ SND_SOC_DAPM_AIF_OUT("HDMI_RX", "HDMI Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_RX", "Slimbus Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_RX", "Slimbus1 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_RX", "Slimbus2 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_RX", "Slimbus3 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("TERT_MI2S_RX", "Tertiary MI2S Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("TERT_MI2S_TX", "Tertiary MI2S Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX", "Secondary MI2S Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SEC_MI2S_TX", "Secondary MI2S Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX_SD1",
+ "Secondary MI2S Playback SD1",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("PRI_MI2S_RX", "Primary MI2S Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("PRI_MI2S_TX", "Primary MI2S Capture",
+ 0, 0, 0, 0),
+};
+
+static const struct snd_soc_component_driver q6afe_dai_component = {
+ .name = "q6afe-dai-component",
+ .dapm_widgets = q6afe_dai_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(q6afe_dai_widgets),
+ .dapm_routes = q6afe_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(q6afe_dapm_routes),
+ .of_xlate_dai_name = q6afe_of_xlate_dai_name,
+
+};
+
+static void of_q6afe_parse_dai_data(struct device *dev,
+ struct q6afe_dai_data *data)
+{
+ struct device_node *node;
+ int ret;
+
+ for_each_child_of_node(dev->of_node, node) {
+ unsigned int lines[Q6AFE_MAX_MI2S_LINES];
+ struct q6afe_dai_priv_data *priv;
+ const char *if_type;
+ int id, i, num_lines;
+
+ ret = of_property_read_u32(node, "reg", &id);
+ if (ret || id > AFE_PORT_MAX) {
+ dev_err(dev, "valid dai id not found:%d\n", ret);
+ continue;
+ }
+
+ if (of_property_read_string(node, "interface-type", &if_type))
+ continue;
+
+ priv = &data->priv[id];
+
+ /* MI2S specific properties */
+ if (!strcmp(if_type, "mi2s")) {
+ ret = of_property_read_variable_u32_array(node,
+ "qcom,sd-lines", lines, 0,
+ Q6AFE_MAX_MI2S_LINES);
+ if (ret < 0)
+ num_lines = 0;
+ else
+ num_lines = ret;
+
+ priv->sd_line_mask = 0;
+
+ for (i = 0; i < num_lines; i++)
+ priv->sd_line_mask |= BIT(lines[i]);
+ }
+ }
+}
+
+static int q6afe_dai_bind(struct device *dev, struct device *master, void *data)
+{
+ struct q6afe_dai_data *dai_data;
+ int rc = 0;
+
+ dai_data = kzalloc(sizeof(*dai_data), GFP_KERNEL);
+ if (!dai_data)
+ rc = -ENOMEM;
+
+ dev_set_drvdata(dev, dai_data);
+
+ of_q6afe_parse_dai_data(dev, dai_data);
+
+ return snd_soc_register_component(dev, &q6afe_dai_component,
+ q6afe_dais, ARRAY_SIZE(q6afe_dais));
+}
+
+static void q6afe_dai_unbind(struct device *dev, struct device *master,
+ void *data)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dev);
+
+ snd_soc_unregister_component(dev);
+ kfree(dai_data);
+}
+
+static const struct component_ops q6afe_dai_comp_ops = {
+ .bind = q6afe_dai_bind,
+ .unbind = q6afe_dai_unbind,
+};
+
+static int q6afe_dai_dev_probe(struct platform_device *pdev)
+{
+ return component_add(&pdev->dev, &q6afe_dai_comp_ops);
+}
+
+static int q6afe_dai_dev_remove(struct platform_device *pdev)
+{
+ component_del(&pdev->dev, &q6afe_dai_comp_ops);
+ return 0;
+}
+
+static struct platform_driver q6afe_dai_platform_driver = {
+ .driver = {
+ .name = "q6afe-dai",
+ },
+ .probe = q6afe_dai_dev_probe,
+ .remove = q6afe_dai_dev_remove,
+};
+module_platform_driver(q6afe_dai_platform_driver);
+
+MODULE_DESCRIPTION("Q6 Audio Fronend dai driver");
+MODULE_LICENSE("GPL v2");
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 20/24] ASoC: qdsp6: q6asm: Add q6asm dai driver
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (18 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 19/24] ASoC: qdsp6: q6afe: Add q6afe dai driver srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 21/24] ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings srinivas.kandagatla
` (3 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to q6asm dai driver which configures Q6ASM streams
to pass pcm data.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6asm-dai.c | 632 +++++++++++++++++++++++++++++++++++++++
3 files changed, 637 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 9b630686eb19..5bdf39f6003f 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -62,6 +62,9 @@ config SND_SOC_QDSP6_ROUTING
config SND_SOC_QDSP6_ASM
tristate
+config SND_SOC_QDSP6_ASM_DAI
+ tristate
+
config SND_SOC_QDSP6_CORE
tristate
@@ -74,6 +77,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_ADM
select SND_SOC_QDSP6_ROUTING
select SND_SOC_QDSP6_ASM
+ select SND_SOC_QDSP6_ASM_DAI
select SND_SOC_QDSP6_CORE
help
To add support for MSM QDSP6 Soc Audio.
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 6a8ae698f22f..af44711eba24 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -4,4 +4,5 @@ obj-$(CONFIG_SND_SOC_QDSP6_AFE_DAI) += q6afe-dai.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
+obj-$(CONFIG_SND_SOC_QDSP6_ASM_DAI) += q6asm-dai.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
new file mode 100644
index 000000000000..c411070b67eb
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -0,0 +1,632 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2016, The Linux Foundation
+// Copyright (c) 2017, Linaro Limited
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/component.h>
+#include <sound/soc.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/of_device.h>
+#include <sound/pcm_params.h>
+#include "q6asm.h"
+#include "q6routing.h"
+#include "q6dsp-errno.h"
+
+#define PLAYBACK_MIN_NUM_PERIODS 2
+#define PLAYBACK_MAX_NUM_PERIODS 8
+#define PLAYBACK_MAX_PERIOD_SIZE 65536
+#define PLAYBACK_MIN_PERIOD_SIZE 128
+#define CAPTURE_MIN_NUM_PERIODS 2
+#define CAPTURE_MAX_NUM_PERIODS 8
+#define CAPTURE_MAX_PERIOD_SIZE 4096
+#define CAPTURE_MIN_PERIOD_SIZE 320
+#define SID_MASK_DEFAULT 0xFF
+
+enum stream_state {
+ Q6ASM_STREAM_IDLE = 0,
+ Q6ASM_STREAM_STOPPED,
+ Q6ASM_STREAM_RUNNING,
+};
+
+struct q6asm_dai_rtd {
+ struct snd_pcm_substream *substream;
+ phys_addr_t phys;
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ unsigned int periods;
+ uint16_t bits_per_sample;
+ uint16_t source; /* Encoding source bit mask */
+ struct audio_client *audio_client;
+ uint16_t session_id;
+ enum stream_state state;
+};
+
+struct q6asm_dai_data {
+ long long int sid;
+};
+
+static struct snd_pcm_hardware q6asm_dai_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 4,
+ .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS *
+ CAPTURE_MAX_PERIOD_SIZE,
+ .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE,
+ .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE,
+ .periods_min = CAPTURE_MIN_NUM_PERIODS,
+ .periods_max = CAPTURE_MAX_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = (PLAYBACK_MAX_NUM_PERIODS *
+ PLAYBACK_MAX_PERIOD_SIZE),
+ .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE,
+ .periods_min = PLAYBACK_MIN_NUM_PERIODS,
+ .periods_max = PLAYBACK_MAX_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+#define Q6ASM_FEDAI_DRIVER(num) { \
+ .playback = { \
+ .stream_name = "MultiMedia"#num" Playback", \
+ .rates = (SNDRV_PCM_RATE_8000_192000| \
+ SNDRV_PCM_RATE_KNOT), \
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE), \
+ .channels_min = 1, \
+ .channels_max = 8, \
+ .rate_min = 8000, \
+ .rate_max = 192000, \
+ }, \
+ .capture = { \
+ .stream_name = "MultiMedia"#num" Capture", \
+ .rates = (SNDRV_PCM_RATE_8000_48000| \
+ SNDRV_PCM_RATE_KNOT), \
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE), \
+ .channels_min = 1, \
+ .channels_max = 4, \
+ .rate_min = 8000, \
+ .rate_max = 48000, \
+ }, \
+ .name = "MultiMedia"#num, \
+ .probe = fe_dai_probe, \
+ .id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \
+ }
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
+ 88200, 96000, 176400, 192000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode, uint32_t token,
+ uint32_t *payload, void *priv)
+{
+ struct q6asm_dai_rtd *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+
+ switch (opcode) {
+ case ASM_CLIENT_EVENT_CMD_RUN_DONE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ q6asm_write_async(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ break;
+ case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ break;
+ case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ snd_pcm_period_elapsed(substream);
+ if (prtd->state == Q6ASM_STREAM_RUNNING)
+ q6asm_write_async(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+
+ break;
+ }
+ case ASM_CLIENT_EVENT_DATA_READ_DONE:
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ snd_pcm_period_elapsed(substream);
+ if (prtd->state == Q6ASM_STREAM_RUNNING)
+ q6asm_read(prtd->audio_client);
+
+ break;
+ default:
+ break;
+ }
+}
+
+static int q6asm_dai_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct q6asm_dai_data *pdata;
+ int ret, i;
+
+ pdata = snd_soc_platform_get_drvdata(soc_prtd->platform);
+ if (!pdata)
+ return -EINVAL;
+
+ if (!prtd || !prtd->audio_client) {
+ pr_err("%s: private data null or audio client freed\n",
+ __func__);
+ return -EINVAL;
+ }
+
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ if (prtd->state) {
+ /* clear the previous setup if any */
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_unmap_memory_regions(substream->stream,
+ prtd->audio_client);
+ q6routing_stream_close(soc_prtd->dai_link->id,
+ substream->stream);
+ }
+
+ ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client,
+ prtd->phys,
+ (prtd->pcm_size / prtd->periods),
+ prtd->periods);
+
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
+ ret);
+ return -ENOMEM;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
+ prtd->bits_per_sample);
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
+ prtd->bits_per_sample);
+ }
+
+ if (ret < 0) {
+ pr_err("%s: q6asm_open_write failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ return -ENOMEM;
+ }
+
+ prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+ ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
+ prtd->session_id, substream->stream);
+ if (ret) {
+ pr_err("%s: stream reg failed ret:%d\n", __func__, ret);
+ return ret;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = q6asm_media_format_block_multi_ch_pcm(
+ prtd->audio_client, runtime->rate,
+ runtime->channels, NULL,
+ prtd->bits_per_sample);
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
+ runtime->rate, runtime->channels,
+ prtd->bits_per_sample);
+
+ /* Queue the buffers */
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read(prtd->audio_client);
+
+ }
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+
+ prtd->state = Q6ASM_STREAM_RUNNING;
+
+ return 0;
+}
+
+static int q6asm_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int q6asm_dai_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = soc_prtd->cpu_dai;
+
+ struct q6asm_dai_rtd *prtd;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = soc_prtd->platform->dev;
+ int ret = 0;
+ int stream_id;
+
+ stream_id = cpu_dai->driver->id;
+
+ pdata = snd_soc_platform_get_drvdata(soc_prtd->platform);
+ if (!pdata) {
+ pr_err("Platform data not found ..\n");
+ return -EINVAL;
+ }
+
+ prtd = kzalloc(sizeof(struct q6asm_dai_rtd), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(dev,
+ (q6asm_cb)event_handler, prtd, stream_id,
+ LEGACY_PCM_MODE);
+ if (!prtd->audio_client) {
+ pr_info("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ runtime->hw = q6asm_dai_hardware_playback;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ runtime->hw = q6asm_dai_hardware_capture;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_integer failed\n");
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
+ PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes min max ret = %d\n",
+ ret);
+ }
+ }
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (ret < 0) {
+ pr_err("constraint for period bytes step ret = %d\n",
+ ret);
+ }
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes step ret = %d\n",
+ ret);
+ }
+
+ runtime->private_data = prtd;
+
+ snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback);
+
+ runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max;
+
+
+ if (pdata->sid < 0)
+ prtd->phys = substream->dma_buffer.addr;
+ else
+ prtd->phys = substream->dma_buffer.addr | (pdata->sid << 32);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int q6asm_dai_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ if (prtd->audio_client) {
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_unmap_memory_regions(substream->stream,
+ prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ }
+ q6routing_stream_close(soc_prtd->dai_link->id,
+ substream->stream);
+ kfree(prtd);
+ return 0;
+}
+
+static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int q6asm_dai_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct device *dev = soc_prtd->platform->dev;
+
+ return dma_mmap_coherent(dev, vma,
+ runtime->dma_area, runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static int q6asm_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ prtd->pcm_size = params_buffer_bytes(params);
+ prtd->periods = params_periods(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ prtd->bits_per_sample = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ prtd->bits_per_sample = 24;
+ break;
+ }
+
+ return 0;
+}
+
+static struct snd_pcm_ops q6asm_dai_ops = {
+ .open = q6asm_dai_open,
+ .hw_params = q6asm_dai_hw_params,
+ .close = q6asm_dai_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = q6asm_dai_prepare,
+ .trigger = q6asm_dai_trigger,
+ .pointer = q6asm_dai_pointer,
+ .mmap = q6asm_dai_mmap,
+};
+
+static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm_substream *psubstream, *csubstream;
+ struct q6asm_dai_data *pdata;
+ struct snd_pcm *pcm = rtd->pcm;
+ struct device *dev;
+ int size, ret;
+
+ dev = rtd->platform->dev;
+ pdata = snd_soc_platform_get_drvdata(rtd->platform);
+
+ size = q6asm_dai_hardware_playback.buffer_bytes_max;
+
+ psubstream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (psubstream) {
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+ &psubstream->dma_buffer);
+ if (ret) {
+ dev_err(dev, "Cannot allocate buffer(s)\n");
+ return ret;
+ }
+ }
+
+ csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ if (csubstream) {
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+ &csubstream->dma_buffer);
+ if (ret) {
+ dev_err(dev, "Cannot allocate buffer(s)\n");
+ if (psubstream)
+ snd_dma_free_pages(&psubstream->dma_buffer);
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+ substream = pcm->streams[i].substream;
+ if (substream) {
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+ }
+}
+
+static struct snd_soc_platform_driver q6asm_soc_platform = {
+ .ops = &q6asm_dai_ops,
+ .pcm_new = q6asm_dai_pcm_new,
+ .pcm_free = q6asm_dai_pcm_free,
+
+};
+
+static const struct snd_soc_dapm_route afe_pcm_routes[] = {
+ {"MM_DL1", NULL, "MultiMedia1 Playback" },
+ {"MM_DL2", NULL, "MultiMedia2 Playback" },
+ {"MM_DL3", NULL, "MultiMedia3 Playback" },
+ {"MM_DL4", NULL, "MultiMedia4 Playback" },
+ {"MM_DL5", NULL, "MultiMedia5 Playback" },
+ {"MM_DL6", NULL, "MultiMedia6 Playback" },
+ {"MM_DL7", NULL, "MultiMedia7 Playback" },
+ {"MM_DL7", NULL, "MultiMedia8 Playback" },
+ {"MultiMedia1 Capture", NULL, "MM_UL1"},
+ {"MultiMedia2 Capture", NULL, "MM_UL2"},
+ {"MultiMedia3 Capture", NULL, "MM_UL3"},
+ {"MultiMedia4 Capture", NULL, "MM_UL4"},
+ {"MultiMedia5 Capture", NULL, "MM_UL5"},
+ {"MultiMedia6 Capture", NULL, "MM_UL6"},
+ {"MultiMedia7 Capture", NULL, "MM_UL7"},
+ {"MultiMedia8 Capture", NULL, "MM_UL8"},
+
+};
+
+static int fe_dai_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_dapm_context *dapm;
+
+ dapm = snd_soc_component_get_dapm(dai->component);
+ snd_soc_dapm_add_routes(dapm, afe_pcm_routes,
+ ARRAY_SIZE(afe_pcm_routes));
+
+ return 0;
+}
+
+static const struct snd_soc_component_driver q6asm_fe_dai_component = {
+ .name = "q6asm-fe-dai",
+};
+
+static struct snd_soc_dai_driver q6asm_fe_dais[] = {
+ Q6ASM_FEDAI_DRIVER(1),
+ Q6ASM_FEDAI_DRIVER(2),
+ Q6ASM_FEDAI_DRIVER(3),
+ Q6ASM_FEDAI_DRIVER(4),
+ Q6ASM_FEDAI_DRIVER(5),
+ Q6ASM_FEDAI_DRIVER(6),
+ Q6ASM_FEDAI_DRIVER(7),
+ Q6ASM_FEDAI_DRIVER(8),
+};
+
+static int q6asm_dai_bind(struct device *dev, struct device *master, void *data)
+{
+ struct device_node *node = dev->of_node;
+ struct of_phandle_args args;
+ struct q6asm_dai_data *pdata;
+ int rc;
+
+ pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL);
+ if (!pdata)
+ return -ENOMEM;
+
+ rc = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
+ if (rc < 0)
+ pdata->sid = -1;
+ else
+ pdata->sid = args.args[0] & SID_MASK_DEFAULT;
+
+ dev_set_drvdata(dev, pdata);
+
+ rc = snd_soc_register_platform(dev, &q6asm_soc_platform);
+ if (rc) {
+ dev_err(dev, "err_dai_platform\n");
+ return rc;
+ }
+
+ return snd_soc_register_component(dev, &q6asm_fe_dai_component,
+ q6asm_fe_dais,
+ ARRAY_SIZE(q6asm_fe_dais));
+}
+static void q6asm_dai_unbind(struct device *dev, struct device *master,
+ void *data)
+{
+ struct q6asm_dai_data *pdata = dev_get_drvdata(dev);
+
+ snd_soc_unregister_platform(dev);
+ snd_soc_unregister_component(dev);
+
+ kfree(pdata);
+
+}
+
+static const struct component_ops q6asm_dai_comp_ops = {
+ .bind = q6asm_dai_bind,
+ .unbind = q6asm_dai_unbind,
+};
+
+static int q6asm_dai_probe(struct platform_device *pdev)
+{
+ return component_add(&pdev->dev, &q6asm_dai_comp_ops);
+}
+
+static int q6asm_dai_dev_remove(struct platform_device *pdev)
+{
+ component_del(&pdev->dev, &q6asm_dai_comp_ops);
+ return 0;
+}
+
+static struct platform_driver q6asm_dai_platform_driver = {
+ .driver = {
+ .name = "q6asm-dai",
+ },
+ .probe = q6asm_dai_probe,
+ .remove = q6asm_dai_dev_remove,
+};
+module_platform_driver(q6asm_dai_platform_driver);
+
+MODULE_DESCRIPTION("Q6ASM dai driver");
+MODULE_LICENSE("GPL v2");
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 21/24] ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (19 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 20/24] ASoC: qdsp6: q6asm: Add q6asm " srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 22/24] ASoC: qcom: apq8096: Add db820c machine driver srinivas.kandagatla
` (2 subsequent siblings)
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Add devicetree bindings documentation file for Qualcomm apq8096 sound card.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
.../devicetree/bindings/sound/qcom,apq8096.txt | 80 ++++++++++++++++++++++
1 file changed, 80 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt
diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt
new file mode 100644
index 000000000000..fddc9f0ce3a5
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt
@@ -0,0 +1,80 @@
+* Qualcomm Technologies APQ8096 ASoC sound card driver
+
+This binding describes the APQ8096 sound card, which uses qdsp for audio.
+
+- compatible:
+ Usage: required
+ Value type: <stringlist>
+ Definition: must be "qcom,apq8096-sndcard"
+
+- qcom,audio-routing:
+ Usage: Optional
+ Value type: <stringlist>
+ Definition: A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the
+ connection's sink, the second being the connection's
+ source. Valid names could be power supplies, MicBias
+ of codec and the jacks on the board:
+
+= dailinks
+Each subnode of sndcard represents either a dailink, and subnodes of each
+dailinks would be cpu/codec/platform dais.
+
+- link-name:
+ Usage: required
+ Value type: <string>
+ Definition: User friendly name for dai link
+
+= CPU, PLATFORM, CODEC dais subnodes
+- cpu:
+ Usage: required
+ Value type: <subnode>
+ Definition: cpu dai sub-node
+
+- codec:
+ Usage: required
+ Value type: <subnode>
+ Definition: codec dai sub-node
+
+- platform:
+ Usage: opional
+ Value type: <subnode>
+ Definition: platform dai sub-node
+
+- sound-dai:
+ Usage: required
+ Value type: <phandle>
+ Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
+
+Example:
+
+audio {
+ compatible = "qcom,apq8096-sndcard";
+ qcom,model = "DB820c";
+ com,audio-routing = "RX_BIAS", "MCLK";
+
+ mm1-dai-link {
+ link-name = "MultiMedia1";
+ cpu {
+ sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+ };
+ platform {
+ sound-dai = <&q6asmdai>;
+ };
+ };
+
+ hdmi-dai-link {
+ link-name = "HDMI Playback";
+ cpu {
+ sound-dai = <&q6afe HDMI_RX>;
+ };
+
+ platform {
+ sound-dai = <&q6adm>;
+ };
+
+ codec {
+ sound-dai = <&hdmi 0>;
+ };
+ };
+};
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 22/24] ASoC: qcom: apq8096: Add db820c machine driver
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (20 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 21/24] ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 23/24] arm64: dts: db820c: add sound card dt entry srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 24/24] MAINTAINERS: Add myself as co-maintainer of qcom audio srinivas.kandagatla
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds support to DB820c machine driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
sound/soc/qcom/Kconfig | 9 ++
sound/soc/qcom/Makefile | 2 +
sound/soc/qcom/apq8096.c | 235 +++++++++++++++++++++++++++++++++++++++++++++++
3 files changed, 246 insertions(+)
create mode 100644 sound/soc/qcom/apq8096.c
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 5bdf39f6003f..9d8007e2f3cf 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -84,3 +84,12 @@ config SND_SOC_QDSP6
This will enable sound soc platform specific
audio drivers. This includes q6asm, q6adm,
q6afe interfaces to DSP using apr.
+
+config SND_SOC_MSM8996
+ tristate "SoC Machine driver for MSM8996 and APQ8096 boards"
+ depends on QCOM_APR
+ select SND_SOC_QDSP6
+ help
+ Support for Qualcomm Technologies LPASS audio block in
+ APQ8096 SoC-based systems.
+ Say Y if you want to use audio device on this SoCs
diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
index 0276717917c0..206945bb9ba1 100644
--- a/sound/soc/qcom/Makefile
+++ b/sound/soc/qcom/Makefile
@@ -13,9 +13,11 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o
# Machine
snd-soc-storm-objs := storm.o
snd-soc-apq8016-sbc-objs := apq8016_sbc.o
+snd-soc-apq8096-objs := apq8096.o
obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
+obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o
#DSP lib
obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
new file mode 100644
index 000000000000..e9b61dc0aa63
--- /dev/null
+++ b/sound/soc/qcom/apq8096.c
@@ -0,0 +1,235 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/soc/qcom/apr.h>
+#include <linux/module.h>
+#include <linux/component.h>
+#include <linux/platform_device.h>
+#include <linux/of_device.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+
+static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ return 0;
+}
+
+static int apq8096_sbc_parse_of(struct snd_soc_card *card)
+{
+ struct device_node *np, *codec, *platform, *cpu, *node;
+ struct device *dev = card->dev;
+ struct snd_soc_dai_link *link;
+ int ret, num_links;
+
+ ret = snd_soc_of_parse_card_name(card, "qcom,model");
+ if (ret) {
+ dev_err(dev, "Error parsing card name: %d\n", ret);
+ return ret;
+ }
+
+ node = dev->of_node;
+
+ /* DAPM routes */
+ if (of_property_read_bool(node, "qcom,audio-routing")) {
+ ret = snd_soc_of_parse_audio_routing(card,
+ "qcom,audio-routing");
+ if (ret)
+ return ret;
+ }
+
+ /* Populate links */
+ num_links = of_get_child_count(node);
+
+ /* Allocate the DAI link array */
+ card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL);
+ if (!card->dai_link)
+ return -ENOMEM;
+
+ card->num_links = num_links;
+ link = card->dai_link;
+
+ for_each_child_of_node(node, np) {
+ cpu = of_get_child_by_name(np, "cpu");
+ platform = of_get_child_by_name(np, "platform");
+ codec = of_get_child_by_name(np, "codec");
+
+ if (!cpu) {
+ dev_err(dev, "Can't find cpu DT node\n");
+ return -EINVAL;
+ }
+
+ link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0);
+ if (!link->cpu_of_node) {
+ dev_err(card->dev, "error getting cpu phandle\n");
+ return -EINVAL;
+ }
+
+ link->platform_of_node = of_parse_phandle(platform,
+ "sound-dai", 0);
+ if (!link->platform_of_node) {
+ dev_err(card->dev, "error getting platform phandle\n");
+ return -EINVAL;
+ }
+
+ ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
+ if (ret) {
+ dev_err(card->dev, "error getting cpu dai name\n");
+ return ret;
+ }
+
+ if (codec) {
+ ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
+
+ if (ret < 0) {
+ dev_err(card->dev, "error getting codec dai name\n");
+ return ret;
+ }
+ link->no_pcm = 1;
+ link->ignore_suspend = 1;
+ link->ignore_pmdown_time = 1;
+ link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
+ } else {
+ link->codec_dai_name = "snd-soc-dummy-dai";
+ link->codec_name = "snd-soc-dummy";
+ link->dynamic = 1;
+ }
+
+ ret = of_property_read_string(np, "link-name", &link->name);
+ if (ret) {
+ dev_err(card->dev, "error getting codec dai_link name\n");
+ return ret;
+ }
+
+ link->dpcm_playback = 1;
+ link->dpcm_capture = 1;
+ link->stream_name = link->name;
+ link++;
+ }
+
+ return ret;
+}
+
+static int apq8096_bind(struct device *dev)
+{
+ struct snd_soc_card *card;
+ int ret;
+
+ card = kzalloc(sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
+ component_bind_all(dev, card);
+ card->dev = dev;
+ ret = apq8096_sbc_parse_of(card);
+ if (ret) {
+ dev_err(dev, "Error parsing OF data\n");
+ component_unbind_all(dev, card);
+ return ret;
+ }
+
+ return snd_soc_register_card(card);
+}
+
+static void apq8096_unbind(struct device *dev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(dev);
+
+ component_unbind_all(dev, card);
+ snd_soc_unregister_card(card);
+ kfree(card->dai_link);
+ kfree(card);
+}
+
+static const struct component_master_ops apq8096_ops = {
+ .bind = apq8096_bind,
+ .unbind = apq8096_unbind,
+};
+
+static int apq8016_compare_of(struct device *dev, void *data)
+{
+ return dev->of_node == data;
+}
+
+static void apq8016_release_of(struct device *dev, void *data)
+{
+ of_node_put(data);
+}
+
+static int add_audio_components(struct device *dev,
+ struct component_match **matchptr)
+{
+ struct device_node *np, *platform, *cpu, *node, *dai_node;
+
+ node = dev->of_node;
+
+ for_each_child_of_node(node, np) {
+ cpu = of_get_child_by_name(np, "cpu");
+ if (cpu) {
+ dai_node = of_parse_phandle(cpu, "sound-dai", 0);
+ of_node_get(dai_node);
+ component_match_add_release(dev, matchptr,
+ apq8016_release_of,
+ apq8016_compare_of,
+ dai_node);
+ }
+
+ platform = of_get_child_by_name(np, "platform");
+ if (platform) {
+ dai_node = of_parse_phandle(platform, "sound-dai", 0);
+ component_match_add_release(dev, matchptr,
+ apq8016_release_of,
+ apq8016_compare_of,
+ dai_node);
+ }
+ }
+
+ return 0;
+}
+
+static int apq8096_platform_probe(struct platform_device *pdev)
+{
+ struct component_match *match = NULL;
+ int ret;
+
+ ret = add_audio_components(&pdev->dev, &match);
+ if (ret)
+ return ret;
+
+ return component_master_add_with_match(&pdev->dev, &apq8096_ops, match);
+}
+
+static int apq8096_platform_remove(struct platform_device *pdev)
+{
+ component_master_del(&pdev->dev, &apq8096_ops);
+
+ return 0;
+}
+
+static const struct of_device_id msm_snd_apq8096_dt_match[] = {
+ {.compatible = "qcom,apq8096-sndcard"},
+ {}
+};
+
+static struct platform_driver msm_snd_apq8096_driver = {
+ .probe = apq8096_platform_probe,
+ .remove = apq8096_platform_remove,
+ .driver = {
+ .name = "msm-snd-apq8096",
+ .owner = THIS_MODULE,
+ .of_match_table = msm_snd_apq8096_dt_match,
+ },
+};
+module_platform_driver(msm_snd_apq8096_driver);
+MODULE_AUTHOR("Srinivas Kandagatla <srinivas.kandagatla@linaro.org");
+MODULE_DESCRIPTION("APQ8096 ASoC Machine Driver");
+MODULE_LICENSE("GPL v2");
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 23/24] arm64: dts: db820c: add sound card dt entry
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (21 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 22/24] ASoC: qcom: apq8096: Add db820c machine driver srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
2018-03-10 2:24 ` [PATCH v4 24/24] MAINTAINERS: Add myself as co-maintainer of qcom audio srinivas.kandagatla
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
This patch adds a sound card entry for basic hdmi playback using adsp.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi | 34 +++++++++++++
arch/arm64/boot/dts/qcom/msm8996.dtsi | 75 +++++++++++++++++++++++++++-
2 files changed, 108 insertions(+), 1 deletion(-)
diff --git a/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi b/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi
index 0be734be95f9..d69ffd042835 100644
--- a/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi
+++ b/arch/arm64/boot/dts/qcom/apq8096-db820c.dtsi
@@ -18,6 +18,8 @@
#include "apq8096-db820c-pmic-pins.dtsi"
#include <dt-bindings/input/input.h>
#include <dt-bindings/gpio/gpio.h>
+#include <dt-bindings/sound/qcom,q6afe.h>
+#include <dt-bindings/sound/qcom,q6asm.h>
/ {
aliases {
@@ -222,6 +224,7 @@
core-vdda-supply = <&pm8994_l12>;
core-vcc-supply = <&pm8994_s4>;
+ #sound-dai-cells = <1>;
};
};
};
@@ -397,3 +400,34 @@
};
};
};
+
+&sound {
+ compatible = "qcom,apq8096-sndcard";
+ qcom,model = "DB820c";
+ qcom,audio-routing =
+ "RX_BIAS", "MCLK";
+ mm1-dai-link {
+ link-name = "MultiMedia1";
+ cpu {
+ sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+ };
+ platform {
+ sound-dai = <&q6asmdai>;
+ };
+ };
+
+ hdmi-dai-link {
+ link-name = "HDMI Playback";
+ cpu {
+ sound-dai = <&q6afedai HDMI_RX>;
+ };
+
+ platform {
+ sound-dai = <&q6routing>;
+ };
+
+ codec {
+ sound-dai = <&hdmi 0>;
+ };
+ };
+};
diff --git a/arch/arm64/boot/dts/qcom/msm8996.dtsi b/arch/arm64/boot/dts/qcom/msm8996.dtsi
index 6e04827879c0..8fc9354e84c5 100644
--- a/arch/arm64/boot/dts/qcom/msm8996.dtsi
+++ b/arch/arm64/boot/dts/qcom/msm8996.dtsi
@@ -15,6 +15,7 @@
#include <dt-bindings/clock/qcom,mmcc-msm8996.h>
#include <dt-bindings/clock/qcom,rpmcc.h>
#include <dt-bindings/thermal/thermal.h>
+#include <dt-bindings/soc/qcom,apr.h>
/ {
model = "Qualcomm Technologies, Inc. MSM8996";
@@ -1560,6 +1561,37 @@
"ref_clk";
};
};
+
+ lpass_q6_smmu: arm,smmu-lpass_q6@1600000 {
+ compatible = "qcom,msm8996-smmu-v2";
+ reg = <0x1600000 0x20000>;
+ #iommu-cells = <1>;
+ power-domains = <&gcc HLOS1_VOTE_LPASS_CORE_GDSC>;
+
+ #global-interrupts = <1>;
+ interrupts = <GIC_SPI 404 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 226 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 393 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 394 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 395 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 396 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 397 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 398 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 399 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 400 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 401 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 402 IRQ_TYPE_LEVEL_HIGH>,
+ <GIC_SPI 403 IRQ_TYPE_LEVEL_HIGH>;
+
+ clocks = <&gcc GCC_HLOS1_VOTE_LPASS_CORE_SMMU_CLK>,
+ <&gcc GCC_HLOS1_VOTE_LPASS_ADSP_SMMU_CLK>;
+ clock-names = "iface", "bus";
+ status = "okay";
+ };
+ };
+
+ sound: sound {
+
};
adsp-pil {
@@ -1583,11 +1615,52 @@
smd-edge {
interrupts = <GIC_SPI 156 IRQ_TYPE_EDGE_RISING>;
-
label = "lpass";
qcom,ipc = <&apcs 16 8>;
qcom,smd-edge = <1>;
qcom,remote-pid = <2>;
+
+ apr {
+ compatible = "qcom,apr-v2";
+ qcom,smd-channels = "apr_audio_svc";
+ qcom,apr-dest-domain-id = <APR_DOMAIN_ADSP>;
+
+ q6core {
+ qcom,apr-svc-name = "CORE";
+ qcom,apr-svc-id = <APR_SVC_ADSP_CORE>;
+ compatible = "qcom,q6core";
+ };
+
+ q6afe: q6afe {
+ compatible = "qcom,q6afe";
+ qcom,apr-svc-name = "AFE";
+ qcom,apr-svc-id = <APR_SVC_AFE>;
+ q6afedai: dais {
+ #sound-dai-cells = <1>;
+ };
+ };
+
+ q6asm: q6asm {
+ compatible = "qcom,q6asm";
+ qcom,apr-svc-name = "ASM";
+ qcom,apr-svc-id = <APR_SVC_ASM>;
+ q6asmdai: dais {
+ #sound-dai-cells = <1>;
+ power-domains = <&gcc HLOS1_VOTE_LPASS_ADSP_GDSC>;
+ iommus = <&lpass_q6_smmu 1>;
+ };
+ };
+
+ q6adm: q6adm {
+ compatible = "qcom,q6adm";
+ qcom,apr-svc-name = "ADM";
+ qcom,apr-svc-id = <APR_SVC_ADM>;
+ q6routing: routing {
+ #sound-dai-cells = <0>;
+ };
+ };
+
+ };
};
};
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread* [PATCH v4 24/24] MAINTAINERS: Add myself as co-maintainer of qcom audio
2018-03-10 2:24 [PATCH v4 00/24] ASoC: qcom: Add support to QDSP based Audio srinivas.kandagatla
` (22 preceding siblings ...)
2018-03-10 2:24 ` [PATCH v4 23/24] arm64: dts: db820c: add sound card dt entry srinivas.kandagatla
@ 2018-03-10 2:24 ` srinivas.kandagatla
23 siblings, 0 replies; 33+ messages in thread
From: srinivas.kandagatla @ 2018-03-10 2:24 UTC (permalink / raw)
To: andy.gross, broonie, linux-arm-msm, alsa-devel, robh+dt
Cc: gregkh, david.brown, mark.rutland, lgirdwood, plai, bgoswami,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, srinivas.kandagatla
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Add myself as co-maintainer of qcom audio drivers
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
---
MAINTAINERS | 1 +
1 file changed, 1 insertion(+)
diff --git a/MAINTAINERS b/MAINTAINERS
index 93a12af4f180..7463122a1184 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -11343,6 +11343,7 @@ F: drivers/crypto/qat/
QCOM AUDIO (ASoC) DRIVERS
M: Patrick Lai <plai@codeaurora.org>
M: Banajit Goswami <bgoswami@codeaurora.org>
+M: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
L: alsa-devel@alsa-project.org (moderated for non-subscribers)
S: Supported
F: sound/soc/qcom/
--
2.15.1
^ permalink raw reply related [flat|nested] 33+ messages in thread