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

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

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

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

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

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

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

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

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

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

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

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

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