From: Srinivas Kandagatla srinivas.kandagatla@linaro.org

This patch add dt bindings for Qualcomm APR (Asynchronous Packet Router) bus driver. This bus is used for communicating with DSP which provides audio and various other services to cpu.

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- .../devicetree/bindings/soc/qcom/qcom,apr.txt | 83 ++++++++++++++++++++++ 1 file changed, 83 insertions(+) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt

diff --git a/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt b/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt new file mode 100644 index 000000000000..1b95fbfed348 --- /dev/null +++ b/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt @@ -0,0 +1,83 @@ +Qualcomm APR (Asynchronous Packet Router) binding + +This binding describes the Qualcomm APR. APR is a IPC protocol for +communication between Application processor and QDSP. APR is mainly +used for audio/voice services on the QDSP. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,apr-v<VERSION-NUMBER>", example "qcom,apr-v2" + +- qcom,apr-dest-domain-id + Usage: required + Value type: <prop-encoded-array> + Definition: Destination processor ID. + Possible values are : + 1 - APR simulator + 2 - PC + 3 - MODEM + 4 - ADSP + 5 - APPS + 6 - MODEM2 + 7 - APPS2 + += APR SERVICES +Each subnode of the APR node can represent service tied to this apr. The name +of the nodes are not important. The properties of these nodes are defined +by the individual bindings for the specific service +- but must contain the following property: + +- qcom,apr-svc-id + Usage: required + Value type: <prop-encoded-array> + Definition: APR Service ID, used for matching the service. + Possible values are : + 3 - DSP Core Service + 4 - Audio Front End Service. + 5 - Voice Stream Manager Service. + 6 - Voice processing manager. + 7 - Audio Stream Manager Service. + 8 - Audio Device Manager Service. + 9 - Multimode voice manager. + 10 - Core voice stream. + 11 - Core voice processor. + 12 - Ultrasound stream manager. + 13 - Listen stream manager. + +- qcom,apr-svc-name + Usage: required + Value type: <stringlist> + Definition: User readable name of a APR service. + += APR DEVICES: +Each subnode of the APR node can represent devices tied to this apr, like +sound-card. The properties of these nodes are defined by the individual +bindings for the specific device. + += EXAMPLE +The following example represents a QDSP based sound card on a MSM8996 device +which uses apr as communication between Apps and QDSP. + + apr { + compatible = "qcom,apr-v2"; + qcom,smd-channels = "apr_audio_svc"; + qcom,apr-dest-domain-id = <APR_DOMAIN_ADSP>; + + q6core { + compatible = "qcom,q6core"; + qcom,apr-svc-name = "CORE"; + qcom,apr-svc-id = <APR_SVC_ADSP_CORE>; + }; + + q6afe { + compatible = "qcom,q6afe"; + qcom,apr-svc-name = "AFE"; + qcom,apr-svc-id = <APR_SVC_AFE>; + }; + + audio { + compatible = "qcom,msm8996-snd-card"; + ... + }; + };

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 | 381 +++++++++++++++++++++++++++++++++++++ include/dt-bindings/soc/qcom,apr.h | 27 +++ include/linux/mod_devicetable.h | 11 ++ include/linux/soc/qcom/apr.h | 131 +++++++++++++ 6 files changed, 560 insertions(+) create mode 100644 drivers/soc/qcom/apr.c create mode 100644 include/dt-bindings/soc/qcom,apr.h create mode 100644 include/linux/soc/qcom/apr.h

diff --git a/drivers/soc/qcom/Kconfig b/drivers/soc/qcom/Kconfig index e050eb83341d..a1273e3f9eb5 100644 --- a/drivers/soc/qcom/Kconfig +++ b/drivers/soc/qcom/Kconfig @@ -107,4 +107,13 @@ 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 4e91e2a7c24c..92f9c8630291 100644 --- a/drivers/soc/qcom/Makefile +++ b/drivers/soc/qcom/Makefile @@ -13,3 +13,4 @@ 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..4db41c218851 --- /dev/null +++ b/drivers/soc/qcom/apr.c @@ -0,0 +1,381 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2011-2017, The Linux Foundation + * Copyright (c) 2018, Linaro Limited + */ + +#include <linux/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/dt-bindings/soc/qcom,apr.h b/include/dt-bindings/soc/qcom,apr.h new file mode 100644 index 000000000000..fb162ac4ca0e --- /dev/null +++ b/include/dt-bindings/soc/qcom,apr.h @@ -0,0 +1,27 @@ +#ifndef __DT_BINDINGS_QCOM_APR_H +#define __DT_BINDINGS_QCOM_APR_H + +/* Domain IDs */ +#define APR_DOMAIN_SIM 0x1 +#define APR_DOMAIN_PC 0x2 +#define APR_DOMAIN_MODEM 0x3 +#define APR_DOMAIN_ADSP 0x4 +#define APR_DOMAIN_APPS 0x5 +#define APR_DOMAIN_MAX 0x6 + +/* ADSP service IDs */ +#define APR_SVC_ADSP_CORE 0x3 +#define APR_SVC_AFE 0x4 +#define APR_SVC_VSM 0x5 +#define APR_SVC_VPM 0x6 +#define APR_SVC_ASM 0x7 +#define APR_SVC_ADM 0x8 +#define APR_SVC_ADSP_MVM 0x09 +#define APR_SVC_ADSP_CVS 0x0A +#define APR_SVC_ADSP_CVP 0x0B +#define APR_SVC_USM 0x0C +#define APR_SVC_LSM 0x0D +#define APR_SVC_VIDC 0x16 +#define APR_SVC_MAX 0x17 + +#endif /* __DT_BINDINGS_QCOM_APR_H */ 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..17bf0773e146 --- /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; + uint16_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_ */

From: Srinivas Kandagatla srinivas.kandagatla@linaro.org

This patch adds support to Q6ADM (Audio Device Manager) module in q6dsp. ADM performs routing between audio streams and AFE ports. It does Rate matching for streams going to devices driven by different clocks, it handles volume ramping, Mixing with channel and bit-width. ADM creates and destroys dynamic COPP services for device-related audio processing as needed.

This patch adds basic support to ADM.

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6adm.c | 634 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6adm.h | 26 ++ 4 files changed, 666 insertions(+) create mode 100644 sound/soc/qcom/qdsp6/q6adm.c create mode 100644 sound/soc/qcom/qdsp6/q6adm.h

diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index caeaf8b1b561..bf04e1301608 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -52,11 +52,16 @@ config SND_SOC_QDSP6_AFE tristate default n

+config SND_SOC_QDSP6_ADM + tristate + default n + config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" depends on QCOM_APR && HAS_DMA select SND_SOC_QDSP6_COMMON select SND_SOC_QDSP6_AFE + select SND_SOC_QDSP6_ADM help To add support for MSM QDSP6 Soc Audio. This will enable sound soc platform specific diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile index 9ec951e0833b..930944425051 100644 --- a/sound/soc/qcom/qdsp6/Makefile +++ b/sound/soc/qcom/qdsp6/Makefile @@ -1,2 +1,3 @@ obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o +obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o diff --git a/sound/soc/qcom/qdsp6/q6adm.c b/sound/soc/qcom/qdsp6/q6adm.c new file mode 100644 index 000000000000..e9c65e05882b --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6adm.c @@ -0,0 +1,634 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2011-2017, The Linux Foundation + * Copyright (c) 2018, Linaro Limited + */ + +#include <linux/slab.h> +#include <linux/wait.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <linux/module.h> +#include <linux/sched.h> +#include <linux/jiffies.h> +#include <linux/of.h> +#include <linux/wait.h> +#include <linux/soc/qcom/apr.h> +#include <linux/platform_device.h> +#include <sound/asound.h> +#include "q6adm.h" +#include "q6afe.h" +#include "q6dsp-errno.h" +#include "q6dsp-common.h" + +#define ADM_CMD_DEVICE_OPEN_V5 0x00010326 +#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329 +#define ADM_CMD_DEVICE_CLOSE_V5 0x00010327 +#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325 + +#define TIMEOUT_MS 1000 +#define RESET_COPP_ID 99 +#define INVALID_COPP_ID 0xFF +/* Definition for a legacy device session. */ +#define ADM_LEGACY_DEVICE_SESSION 0 +#define ADM_MATRIX_ID_AUDIO_RX 0 + +struct copp { + int afe_port; + int copp_idx; + int id; + int refcnt; + int topology; + int mode; + int stat; + int rate; + int bit_width; + int channels; + int app_type; + int acdb_id; + struct mutex lock; + wait_queue_head_t wait; + struct list_head node; + struct q6adm *adm; +}; + +struct q6adm { + struct apr_device *apr; + struct device *dev; + unsigned long copp_bitmap[AFE_MAX_PORTS]; + struct list_head copps_list; + spinlock_t copps_list_lock; + int matrix_map_stat; + struct mutex lock; + wait_queue_head_t matrix_map_wait; + void *routing_data; +}; + +struct adm_cmd_device_open_v5 { + struct apr_hdr hdr; + u16 flags; + u16 mode_of_operation; + u16 endpoint_id_1; + u16 endpoint_id_2; + u32 topology_id; + u16 dev_num_channel; + u16 bit_width; + u32 sample_rate; + u8 dev_channel_mapping[8]; +} __packed; + +struct adm_cmd_matrix_map_routings_v5 { + struct apr_hdr hdr; + u32 matrix_id; + u32 num_sessions; +} __packed; + +struct adm_session_map_node_v5 { + u16 session_id; + u16 num_copps; +} __packed; + +static struct copp *adm_find_copp(struct q6adm *adm, int port_idx, + int copp_idx) +{ + struct copp *c; + + spin_lock(&adm->copps_list_lock); + list_for_each_entry(c, &adm->copps_list, node) { + if ((port_idx == c->afe_port) && (copp_idx == c->copp_idx)) { + spin_unlock(&adm->copps_list_lock); + return c; + } + } + + spin_unlock(&adm->copps_list_lock); + return NULL; + +} + +static int adm_callback(struct apr_device *adev, + struct apr_client_message *data) +{ + struct aprv2_ibasic_rsp_result_t *result = data->payload; + int port_idx, copp_idx; + struct copp *copp; + struct q6adm *adm = dev_get_drvdata(&adev->dev); + + if (!data->payload_size) + return 0; + + copp_idx = (data->token) & 0XFF; + port_idx = ((data->token) >> 16) & 0xFF; + if (port_idx < 0 || port_idx >= AFE_MAX_PORTS) { + dev_err(&adev->dev, "Invalid port idx %d token %d\n", + port_idx, data->token); + return 0; + } + if (copp_idx < 0 || copp_idx >= MAX_COPPS_PER_PORT) { + dev_err(&adev->dev, "Invalid copp idx %d token %d\n", + copp_idx, data->token); + return 0; + } + + switch (data->opcode) { + case APR_BASIC_RSP_RESULT: { + if (result->status != 0) { + dev_err(&adev->dev, "cmd = 0x%x return error = 0x%x\n", + result->opcode, result->status); + } + switch (result->opcode) { + case ADM_CMD_DEVICE_OPEN_V5: + case ADM_CMD_DEVICE_CLOSE_V5: + copp = adm_find_copp(adm, port_idx, copp_idx); + if (IS_ERR_OR_NULL(copp)) + return 0; + + copp->stat = result->status; + wake_up(&copp->wait); + break; + case ADM_CMD_MATRIX_MAP_ROUTINGS_V5: + adm->matrix_map_stat = result->status; + wake_up(&adm->matrix_map_wait); + break; + + default: + dev_err(&adev->dev, "Unknown Cmd: 0x%x\n", + result->opcode); + break; + } + return 0; + } + case ADM_CMDRSP_DEVICE_OPEN_V5: { + struct adm_cmd_rsp_device_open_v5 { + u32 status; + u16 copp_id; + u16 reserved; + } __packed * open = data->payload; + + open = data->payload; + copp = adm_find_copp(adm, port_idx, copp_idx); + if (IS_ERR_OR_NULL(copp)) + return 0; + + if (open->copp_id == INVALID_COPP_ID) { + dev_err(&adev->dev, "Invalid coppid rxed %d\n", + open->copp_id); + copp->stat = ADSP_EBADPARAM; + wake_up(&copp->wait); + break; + } + copp->stat = result->opcode; + copp->id = open->copp_id; + wake_up(&copp->wait); + } + break; + default: + dev_err(&adev->dev, "Unknown cmd:0x%x\n", + data->opcode); + break; + } + + return 0; +} + +static struct copp *adm_alloc_copp(struct q6adm *adm, int port_idx) +{ + struct copp *c; + int idx; + + idx = find_first_zero_bit(&adm->copp_bitmap[port_idx], + MAX_COPPS_PER_PORT); + + if (idx > MAX_COPPS_PER_PORT) + return ERR_PTR(-EBUSY); + + c = kzalloc(sizeof(*c), GFP_KERNEL); + if (!c) + return ERR_PTR(-ENOMEM); + + set_bit(idx, &adm->copp_bitmap[port_idx]); + c->copp_idx = idx; + c->afe_port = port_idx; + c->adm = adm; + + mutex_init(&c->lock); + init_waitqueue_head(&c->wait); + + spin_lock(&adm->copps_list_lock); + list_add_tail(&c->node, &adm->copps_list); + spin_unlock(&adm->copps_list_lock); + + return c; +} + +static void adm_free_copp(struct q6adm *adm, struct copp *c, int port_idx) +{ + clear_bit(c->copp_idx, &adm->copp_bitmap[port_idx]); + spin_lock(&adm->copps_list_lock); + list_del(&c->node); + spin_unlock(&adm->copps_list_lock); + kfree(c); +} + +static struct copp *adm_find_matching_copp(struct q6adm *adm, + int port_id, int topology, + int mode, int rate, int channel_mode, + int bit_width, int app_type) +{ + struct copp *c; + + spin_lock(&adm->copps_list_lock); + + list_for_each_entry(c, &adm->copps_list, node) { + if ((port_id == c->afe_port) && (topology == c->topology) && + (mode == c->mode) && (rate == c->rate) && + (bit_width == c->bit_width) && (app_type == c->app_type)) { + spin_unlock(&adm->copps_list_lock); + return c; + } + } + spin_unlock(&adm->copps_list_lock); + + c = adm_alloc_copp(adm, port_id); + if (IS_ERR_OR_NULL(c)) + return ERR_CAST(c); + + mutex_lock(&c->lock); + c->refcnt = 0; + c->topology = topology; + c->mode = mode; + c->rate = rate; + c->channels = channel_mode; + c->bit_width = bit_width; + c->app_type = app_type; + mutex_unlock(&c->lock); + + return c; + +} + +static int q6adm_apr_send_copp_pkt(struct q6adm *adm, struct copp *copp, + void *data) +{ + struct device *dev = adm->dev; + int ret; + + mutex_lock(&copp->lock); + copp->stat = -1; + ret = apr_send_pkt(adm->apr, data); + if (ret < 0) { + dev_err(dev, "Failed to send APR packet\n"); + ret = -EINVAL; + goto err; + } + /* Wait for the callback with copp id */ + ret = wait_event_timeout(copp->wait, copp->stat >= 0, + msecs_to_jiffies(TIMEOUT_MS)); + if (!ret) { + dev_err(dev, "ADM copp cmd timedout\n"); + ret = -EINVAL; + } else if (copp->stat > 0) { + dev_err(dev, "DSP returned error[%s]\n", + q6dsp_strerror(copp->stat)); + ret = q6dsp_errno(copp->stat); + } + +err: + mutex_unlock(&copp->lock); + return ret; +} + +static int q6adm_device_open(struct q6adm *adm, struct copp *copp, int port_id, + int path, int topology, int channel_mode, + int bit_width, int rate) +{ + struct adm_cmd_device_open_v5 open = {0,}; + int afe_port = q6afe_get_port_id(port_id); + int ret; + + open.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(APR_HDR_SIZE), + APR_PKT_VER); + open.hdr.pkt_size = sizeof(open); + open.hdr.src_svc = APR_SVC_ADM; + open.hdr.src_domain = APR_DOMAIN_APPS; + open.hdr.src_port = afe_port; + open.hdr.dest_svc = APR_SVC_ADM; + open.hdr.dest_domain = APR_DOMAIN_ADSP; + open.hdr.dest_port = afe_port; + open.hdr.token = port_id << 16 | copp->copp_idx; + open.hdr.opcode = ADM_CMD_DEVICE_OPEN_V5; + open.flags = ADM_LEGACY_DEVICE_SESSION; + open.mode_of_operation = path; + open.endpoint_id_1 = afe_port; + open.topology_id = topology; + open.dev_num_channel = channel_mode & 0x00FF; + open.bit_width = bit_width; + open.sample_rate = rate; + + ret = q6dsp_map_channels(&open.dev_channel_mapping[0], + channel_mode); + if (ret) + return ret; + + return q6adm_apr_send_copp_pkt(adm, copp, &open); +} + +/** + * q6adm_open() - open adm and grab a free copp + * + * @dev: Pointer to adm child device. + * @port_id: port id + * @path: playback or capture path. + * @rate: rate at which copp is required. + * @channel_mode: channel mode + * @topology: adm topology id + * @perf_mode: performace mode. + * @bit_width: audio sample bit width + * @app_type: Application type. + * @acdb_id: ACDB id + * + * Return: Will be an negative on error or a valid copp index on success. + */ +int q6adm_open(struct device *dev, int port_id, int path, int rate, + int channel_mode, int topology, int perf_mode, + uint16_t bit_width, int app_type, int acdb_id) +{ + struct q6adm *adm = dev_get_drvdata(dev); + struct copp *copp; + int ret = 0; + + if (port_id < 0) { + dev_err(dev, "Invalid port_id 0x%x\n", port_id); + return -EINVAL; + } + + copp = adm_find_matching_copp(adm, port_id, topology, perf_mode, + rate, channel_mode, bit_width, app_type); + + /* Create a COPP if port id are not enabled */ + if (copp->refcnt == 0) { + ret = q6adm_device_open(adm, copp, port_id, path, topology, + channel_mode, bit_width, rate); + if (ret < 0) + return ret; + } + mutex_lock(&copp->lock); + copp->refcnt++; + mutex_unlock(&copp->lock); + + return copp->copp_idx; +} +EXPORT_SYMBOL_GPL(q6adm_open); + +/** + * q6adm_set_routing_data() - set routing private data + * + * @dev: Pointer to adm device. + * @data: routing private data + * + */ +void q6adm_set_routing_data(struct device *dev, void *data) +{ + struct q6adm *adm = dev_get_drvdata(dev); + + adm->routing_data = data; +} +EXPORT_SYMBOL_GPL(q6adm_set_routing_data); + +/** + * q6adm_get_routing_data() - get routing private data + * + * @dev: Pointer to adm device. + * + * Return: pointer to routing private data + */ +void *q6adm_get_routing_data(struct device *dev) +{ + struct q6adm *adm = dev_get_drvdata(dev); + + return adm->routing_data; +} +EXPORT_SYMBOL_GPL(q6adm_get_routing_data); + +/** + * q6adm_matrix_map() - Map asm streams and afe ports using payload + * + * @dev: Pointer to adm child device. + * @path: playback or capture path. + * @payload_map: map between session id and afe ports. + * @perf_mode: Performace mode. + * + * Return: Will be an negative on error or a zero on success. + */ +int q6adm_matrix_map(struct device *dev, int path, + struct route_payload payload_map, int perf_mode) +{ + struct q6adm *adm = dev_get_drvdata(dev); + struct adm_cmd_matrix_map_routings_v5 *route; + struct adm_session_map_node_v5 *node; + uint16_t *copps_list; + int cmd_size, ret, i, copp_idx; + void *matrix_map = NULL; + struct copp *copp; + + /* Assumes port_ids have already been validated during adm_open */ + cmd_size = (sizeof(*route) + + sizeof(*node) + + (sizeof(uint32_t) * payload_map.num_copps)); + matrix_map = kzalloc(cmd_size, GFP_KERNEL); + if (!matrix_map) + return -ENOMEM; + + route = matrix_map; + route->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(APR_HDR_SIZE), + APR_PKT_VER); + route->hdr.pkt_size = cmd_size; + route->hdr.src_svc = 0; + route->hdr.src_domain = APR_DOMAIN_APPS; + route->hdr.dest_svc = APR_SVC_ADM; + route->hdr.dest_domain = APR_DOMAIN_ADSP; + route->hdr.token = 0; + route->hdr.opcode = ADM_CMD_MATRIX_MAP_ROUTINGS_V5; + route->num_sessions = 1; + + switch (path) { + case ADM_PATH_PLAYBACK: + route->matrix_id = ADM_MATRIX_ID_AUDIO_RX; + break; + default: + dev_err(dev, "Wrong path set[%d]\n", path); + + break; + } + + node = matrix_map + sizeof(*route); + node->session_id = payload_map.session_id; + node->num_copps = payload_map.num_copps; + copps_list = matrix_map + sizeof(*route) + sizeof(*node); + + for (i = 0; i < payload_map.num_copps; i++) { + int port_idx = payload_map.port_id[i]; + + if (port_idx < 0) { + dev_err(dev, "Invalid port_id 0x%x\n", + payload_map.port_id[i]); + ret = -EINVAL; + goto fail_cmd; + } + copp_idx = payload_map.copp_idx[i]; + + copp = adm_find_copp(adm, port_idx, copp_idx); + if (IS_ERR_OR_NULL(copp)) { + ret = -EINVAL; + goto fail_cmd; + } + + copps_list[i] = copp->id; + } + + mutex_lock(&adm->lock); + adm->matrix_map_stat = -1; + + ret = apr_send_pkt(adm->apr, matrix_map); + if (ret < 0) { + dev_err(dev, "routing for syream %d failed ret %d\n", + payload_map.session_id, ret); + ret = -EINVAL; + goto fail_cmd; + } + ret = wait_event_timeout(adm->matrix_map_wait, + adm->matrix_map_stat >= 0, + msecs_to_jiffies(TIMEOUT_MS)); + if (!ret) { + dev_err(dev, "routing for syream %d failed\n", + payload_map.session_id); + ret = -ETIMEDOUT; + goto fail_cmd; + } else if (adm->matrix_map_stat > 0) { + dev_err(dev, "DSP returned error[%s]\n", + q6dsp_strerror(adm->matrix_map_stat)); + ret = q6dsp_errno(adm->matrix_map_stat); + goto fail_cmd; + } + +fail_cmd: + mutex_unlock(&adm->lock); + kfree(matrix_map); + return ret; +} +EXPORT_SYMBOL_GPL(q6adm_matrix_map); + +static int q6adm_device_close(struct q6adm *adm, struct copp *copp, + int port_id, int copp_idx) +{ + struct apr_hdr close = {0}; + + close.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(APR_HDR_SIZE), + APR_PKT_VER); + close.pkt_size = sizeof(close); + close.src_svc = APR_SVC_ADM; + close.src_domain = APR_DOMAIN_APPS; + close.src_port = port_id; + close.dest_svc = APR_SVC_ADM; + close.dest_domain = APR_DOMAIN_ADSP; + close.dest_port = copp->id; + close.token = port_id << 16 | copp_idx; + close.opcode = ADM_CMD_DEVICE_CLOSE_V5; + + return q6adm_apr_send_copp_pkt(adm, copp, &close); +} + +/** + * q6adm_close() - Close adm copp + * + * @dev: Pointer to adm child device. + * @port_id: afe port id. + * @perf_mode: perf_mode mode + * @copp_idx: copp index to close + * + * Return: Will be an negative on error or a zero on success. + */ +int q6adm_close(struct device *dev, int port_id, int perf_mode, int copp_idx) +{ + struct q6adm *adm = dev_get_drvdata(dev); + struct copp *copp; + + if (port_id < 0) { + dev_err(dev, "Invalid port_id 0x%x\n", port_id); + return -EINVAL; + } + + if ((copp_idx < 0) || (copp_idx >= MAX_COPPS_PER_PORT)) { + dev_err(dev, "Invalid copp idx: %d\n", copp_idx); + return -EINVAL; + } + + copp = adm_find_copp(adm, port_id, copp_idx); + if (IS_ERR_OR_NULL(copp)) + return -EINVAL; + + mutex_lock(&copp->lock); + copp->refcnt--; + mutex_unlock(&copp->lock); + if (!copp->refcnt) { + int ret = q6adm_device_close(adm, copp, port_id, copp_idx); + + if (ret < 0) + return ret; + + adm_free_copp(adm, copp, port_id); + } + + return 0; +} +EXPORT_SYMBOL_GPL(q6adm_close); + +static int q6adm_probe(struct apr_device *adev) +{ + struct q6adm *adm; + + adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL); + if (!adm) + return -ENOMEM; + + adm->apr = adev; + dev_set_drvdata(&adev->dev, adm); + adm->dev = &adev->dev; + adm->matrix_map_stat = 0; + mutex_init(&adm->lock); + init_waitqueue_head(&adm->matrix_map_wait); + + INIT_LIST_HEAD(&adm->copps_list); + spin_lock_init(&adm->copps_list_lock); + + return q6pcm_routing_probe(adm->dev); +} + +static int q6adm_exit(struct apr_device *adev) +{ + q6pcm_routing_remove(&adev->dev); + return 0; +} + +static const struct of_device_id q6adm_device_id[] = { + { .compatible = "qcom,q6adm" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6adm_device_id); + +static struct apr_driver qcom_q6adm_driver = { + .probe = q6adm_probe, + .remove = q6adm_exit, + .callback = adm_callback, + .driver = { + .name = "qcom-q6adm", + .of_match_table = of_match_ptr(q6adm_device_id), + }, +}; + +module_apr_driver(qcom_q6adm_driver); +MODULE_DESCRIPTION("Q6 Audio Device Manager"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/qdsp6/q6adm.h b/sound/soc/qcom/qdsp6/q6adm.h new file mode 100644 index 000000000000..8f89210e2fcf --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6adm.h @@ -0,0 +1,26 @@ +// SPDX-License-Identifier: GPL-2.0 +#ifndef __Q6_ADM_V2_H__ +#define __Q6_ADM_V2_H__ + +#define ADM_PATH_PLAYBACK 0x1 +#define MAX_COPPS_PER_PORT 8 +#define NULL_COPP_TOPOLOGY 0x00010312 + +/* multiple copp per stream. */ +struct route_payload { + int num_copps; + int session_id; + int copp_idx[MAX_COPPS_PER_PORT]; + int port_id[MAX_COPPS_PER_PORT]; +}; + +void *q6adm_get_routing_data(struct device *dev); +void q6adm_set_routing_data(struct device *dev, void *data); +int q6adm_open(struct device *dev, int port_id, int path, int rate, + int channel_mode, int topology, int perf_mode, + uint16_t bit_width, int app_type, int acdb_id); +int q6adm_close(struct device *dev, int port, int topology, int perf_mode); +int q6adm_matrix_map(struct device *dev, int path, + struct route_payload payload_map, int perf_mode); + +#endif /* __Q6_ADM_V2_H__ */

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 | 5 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6asm.c | 252 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 18 +++ 4 files changed, 276 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 bf04e1301608..a14d960b8fe4 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -56,12 +56,17 @@ config SND_SOC_QDSP6_ADM tristate default n

+config SND_SOC_QDSP6_ASM + tristate + default n + 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..768d9b446da9 --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -0,0 +1,252 @@ +// 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/of.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/mm.h> +#include "q6asm.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; + int cmd_state; + void *priv; + uint32_t io_mode; + struct apr_device *adev; + struct mutex lock; + wait_queue_head_t cmd_wait; + int perf_mode; + int stream_id; + struct device *dev; +}; + +struct q6asm { + struct apr_device *adev; + int mem_state; + struct device *dev; + wait_queue_head_t mem_wait; + struct mutex session_lock; + struct platform_device *pcmdev; + struct audio_client *session[MAX_SESSIONS + 1]; + void *dai_data; +}; + +static bool q6asm_is_valid_audio_client(struct audio_client *ac) +{ + struct q6asm *a = dev_get_drvdata(ac->dev); + 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); + + mutex_lock(&a->session_lock); + a->session[ac->session] = NULL; + mutex_unlock(&a->session_lock); + 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]; +} + +/** + * q6asm_set_dai_data() - set dai private data + * + * @dev: Pointer to asm device. + * @data: dai private data + * + */ +void q6asm_set_dai_data(struct device *dev, void *data) +{ + struct q6asm *a = dev_get_drvdata(dev); + + a->dai_data = data; +} +EXPORT_SYMBOL_GPL(q6asm_set_dai_data); + +/** + * q6asm_get_dai_data() - get dai private data + * + * @dev: Pointer to asm device. + * + * Return: pointer to dai private data + */ +void *q6asm_get_dai_data(struct device *dev) +{ + struct q6asm *a = dev_get_drvdata(dev); + + return a->dai_data; +} +EXPORT_SYMBOL_GPL(q6asm_get_dai_data); + +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) +{ + struct q6asm *a = dev_get_drvdata(dev); + 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); + + mutex_lock(&a->session_lock); + a->session[stream_id + 1] = ac; + mutex_unlock(&a->session_lock); + + ac->session = stream_id + 1; + ac->cb = cb; + ac->dev = dev; + ac->priv = priv; + ac->io_mode = ASM_SYNC_IO_MODE; + ac->perf_mode = LEGACY_PCM_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); + ac->cmd_state = 0; + + return ac; +} +EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc); + + +static int q6asm_probe(struct apr_device *adev) +{ + struct q6asm *q6asm; + + q6asm = devm_kzalloc(&adev->dev, sizeof(*q6asm), GFP_KERNEL); + if (!q6asm) + return -ENOMEM; + + q6asm->dev = &adev->dev; + q6asm->adev = adev; + q6asm->mem_state = 0; + init_waitqueue_head(&q6asm->mem_wait); + mutex_init(&q6asm->session_lock); + dev_set_drvdata(&adev->dev, q6asm); + + return q6asm_dai_probe(&adev->dev); +} + +static int q6asm_remove(struct apr_device *adev) +{ + return q6asm_dai_remove(&adev->dev); +} + +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..b13fe0d2ea61 --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -0,0 +1,18 @@ +// SPDX-License-Identifier: GPL-2.0 +#ifndef __Q6_ASM_H__ +#define __Q6_ASM_H__ + +#define MAX_SESSIONS 16 + +void q6asm_set_dai_data(struct device *dev, void *data); +void *q6asm_get_dai_data(struct device *dev); + +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); +void q6asm_audio_client_free(struct audio_client *ac); +int q6asm_get_session_id(struct audio_client *ac); +#endif /* __Q6_ASM_H__ */

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 --- include/dt-bindings/sound/qcom,q6asm.h | 22 ++ sound/soc/qcom/qdsp6/q6asm.c | 503 ++++++++++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.h | 41 +++ 3 files changed, 564 insertions(+), 2 deletions(-) create mode 100644 include/dt-bindings/sound/qcom,q6asm.h

diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h new file mode 100644 index 000000000000..4e85bf804cec --- /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__ */ diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 412275edb15c..0ee1e30a8d8e 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -10,6 +10,7 @@ #include <linux/soc/qcom/apr.h> #include <linux/device.h> #include <linux/of.h> +#include <uapi/sound/asound.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/mm.h> @@ -17,10 +18,26 @@ #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_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_SESSION_CMD_RUN_V2 0x00010DAA +#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5 +#define ASM_DATA_CMD_WRITE_V2 0x00010DAB +#define ASM_SESSION_CMD_SUSPEND 0x00010DEC +#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3 + +#define ASM_LEGACY_STREAM_SESSION 0 +#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 +63,49 @@ 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_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; @@ -131,7 +191,7 @@ static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,

rc = wait_event_timeout(a->mem_wait, (a->mem_state <= 0), 5 * HZ); if (!rc) { - dev_err(a->dev, "CMD timeout \n"); + dev_err(a->dev, "CMD timeout\n"); rc = -ETIMEDOUT; } else if (a->mem_state < 0) { rc = q6dsp_errno(a->mem_state); @@ -395,6 +455,108 @@ void *q6asm_get_dai_data(struct device *dev) } EXPORT_SYMBOL_GPL(q6asm_get_dai_data);

+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_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->cmd_state = -result->status; + wake_up(&ac->cmd_wait); + return 0; + } + break; + default: + dev_err(ac->dev, "command[0x%x] not expecting rsp\n", + result->opcode); + break; + } + + if (ac->cmd_state) { + ac->cmd_state = 0; + 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 = 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); + return -EINVAL; + } + token = data->token; + port->buf[token].used = 1; + } + 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) { @@ -404,6 +566,11 @@ 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; @@ -519,6 +686,338 @@ 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) +{ + int rc; + + mutex_lock(&ac->lock); + ac->cmd_state = 1; + + rc = apr_send_pkt(ac->adev, cmd); + if (rc < 0) + goto err; + + rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state <= 0), 5 * HZ); + if (!rc) { + dev_err(ac->dev, "CMD timeout\n"); + rc = -ETIMEDOUT; + goto err; + } + + if (ac->cmd_state > 0) + rc = q6dsp_errno(ac->cmd_state); + +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_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) +{ + int cnt = 0; + int loopcnt = 0; + int used; + struct audio_port_data *port = NULL; + + if (!(ac->io_mode & ASM_SYNC_IO_MODE)) + return; + + used = (ac->io_mode & ASM_TUN_WRITE_IO_MODE ? 1 : 0); + mutex_lock(&ac->lock); + 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--; + } + } + mutex_unlock(&ac->lock); +} + +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 a4f9fe636b7e..b5ef90bb724b 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -1,8 +1,35 @@ // 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 + +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

void q6asm_set_dai_data(struct device *dev, void *data); void *q6asm_get_dai_data(struct device *dev); @@ -14,6 +41,20 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb, void *priv, int session_id); 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_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,

From: Srinivas Kandagatla srinivas.kandagatla@linaro.org

This patch adds support to q6afe backend dais driver.

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/Makefile | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 280 +++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 3 + 3 files changed, 284 insertions(+), 1 deletion(-) create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c

diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile index 660afcab98fd..c7833842b878 100644 --- a/sound/soc/qcom/qdsp6/Makefile +++ b/sound/soc/qcom/qdsp6/Makefile @@ -1,5 +1,5 @@ obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o -obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o +obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o q6afe-dai.o obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o 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/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c new file mode 100644 index 000000000000..f6a618e9db9e --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -0,0 +1,280 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2011-2016, The Linux Foundation + * Copyright (c) 2018, Linaro Limited + */ + +#include <linux/err.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include "q6afe.h" + +struct q6afe_dai_data { + struct q6afe_port *port[AFE_PORT_MAX]; + struct q6afe_port_config port_config[AFE_PORT_MAX]; + bool is_port_started[AFE_PORT_MAX]; +}; + +static int q6hdmi_format_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct q6afe_dai_data *dai_data = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + + dai_data->port_config[AFE_PORT_HDMI_RX].hdmi.datatype = value; + + return 0; +} + +static int q6hdmi_format_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + + struct q6afe_dai_data *dai_data = kcontrol->private_data; + + ucontrol->value.integer.value[0] = + dai_data->port_config[AFE_PORT_HDMI_RX].hdmi.datatype; + + return 0; +} + +static const char * const hdmi_format[] = { + "LPCM", + "Compr" +}; + +static const struct soc_enum hdmi_config_enum[] = { + SOC_ENUM_SINGLE_EXT(2, hdmi_format), +}; + +static const struct snd_kcontrol_new q6afe_config_controls[] = { + SOC_ENUM_EXT("HDMI RX Format", hdmi_config_enum[0], + q6hdmi_format_get, + q6hdmi_format_put), +}; + +static int q6hdmi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = q6afe_get_dai_data(dai->dev); + int channels = params_channels(params); + struct q6afe_hdmi_cfg *hdmi = &dai_data->port_config[dai->id].hdmi; + + hdmi->sample_rate = params_rate(params); + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hdmi->bit_width = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + hdmi->bit_width = 24; + break; + } + + /*refer to HDMI spec CEA-861-E: Table 28 Audio InfoFrame Data Byte 4*/ + switch (channels) { + case 2: + hdmi->channel_allocation = 0; + break; + case 3: + hdmi->channel_allocation = 0x02; + break; + case 4: + hdmi->channel_allocation = 0x06; + break; + case 5: + hdmi->channel_allocation = 0x0A; + break; + case 6: + hdmi->channel_allocation = 0x0B; + break; + case 7: + hdmi->channel_allocation = 0x12; + break; + case 8: + hdmi->channel_allocation = 0x13; + break; + default: + dev_err(dai->dev, "invalid Channels = %u\n", channels); + return -EINVAL; + } + + return 0; +} + +static int q6afe_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = q6afe_get_dai_data(dai->dev); + + dai_data->is_port_started[dai->id] = false; + + return 0; +} + +static void q6afe_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = q6afe_get_dai_data(dai->dev); + int rc; + + rc = q6afe_port_stop(dai_data->port[dai->id]); + if (rc < 0) + dev_err(dai->dev, "fail to close AFE port\n"); + + dai_data->is_port_started[dai->id] = false; + +} + +static int q6afe_dai_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = q6afe_get_dai_data(dai->dev); + int rc; + + if (dai_data->is_port_started[dai->id]) { + /* stop the port and restart with new port config */ + rc = q6afe_port_stop(dai_data->port[dai->id]); + if (rc < 0) { + dev_err(dai->dev, "fail to close AFE port\n"); + return rc; + } + } + + if (dai->id == AFE_PORT_HDMI_RX) + q6afe_hdmi_port_prepare(dai_data->port[dai->id], + &dai_data->port_config[dai->id].hdmi); + + rc = q6afe_port_start(dai_data->port[dai->id]); + if (rc < 0) { + dev_err(dai->dev, "fail to start AFE port %x\n", dai->id); + return rc; + } + dai_data->is_port_started[dai->id] = true; + + return 0; +} + +static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { + {"HDMI Playback", NULL, "HDMI_RX"}, +}; + +static struct snd_soc_dai_ops q6hdmi_ops = { + .prepare = q6afe_dai_prepare, + .hw_params = q6hdmi_hw_params, + .shutdown = q6afe_dai_shutdown, + .startup = q6afe_dai_startup, +}; + +static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = q6afe_get_dai_data(dai->dev); + struct snd_soc_dapm_context *dapm; + struct q6afe_port *port; + + dapm = snd_soc_component_get_dapm(dai->component); + + port = q6afe_port_get_from_id(dai->dev, dai->id); + if (IS_ERR(port)) { + dev_err(dai->dev, "Unable to get afe port\n"); + return -EINVAL; + } + dai_data->port[dai->id] = port; + + return 0; +} + +static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai) +{ + struct q6afe_dai_data *dai_data = q6afe_get_dai_data(dai->dev); + + q6afe_port_put(dai_data->port[dai->id]); + + return 0; +} + +static struct snd_soc_dai_driver q6afe_dais[] = { + { + .playback = { + .stream_name = "HDMI Playback", + .rates = SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + .channels_min = 2, + .channels_max = 8, + .rate_max = 192000, + .rate_min = 48000, + }, + .ops = &q6hdmi_ops, + .id = AFE_PORT_HDMI_RX, + .name = "HDMI", + .probe = msm_dai_q6_dai_probe, + .remove = msm_dai_q6_dai_remove, + }, +}; + +static int q6afe_of_xlate_dai_name(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name) +{ + int id = args->args[0]; + int i, ret = -EINVAL; + + for (i = 0; i < ARRAY_SIZE(q6afe_dais); i++) { + if (q6afe_dais[i].id == id) { + *dai_name = q6afe_dais[i].name; + ret = 0; + break; + } + } + + return ret; +} + +static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { + SND_SOC_DAPM_AIF_OUT("HDMI_RX", "HDMI Playback", 0, 0, 0, 0), +}; + +static const struct snd_soc_component_driver q6afe_dai_component = { + .name = "q6afe-dai-component", + .dapm_widgets = q6afe_dai_widgets, + .num_dapm_widgets = ARRAY_SIZE(q6afe_dai_widgets), + .controls = q6afe_config_controls, + .num_controls = ARRAY_SIZE(q6afe_config_controls), + .dapm_routes = q6afe_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(q6afe_dapm_routes), + .of_xlate_dai_name = q6afe_of_xlate_dai_name, + +}; + +int q6afe_dai_dev_probe(struct device *dev) +{ + int rc = 0; + struct q6afe_dai_data *dai_data; + + dai_data = devm_kzalloc(dev, sizeof(*dai_data), GFP_KERNEL); + if (!dai_data) + rc = -ENOMEM; + + q6afe_set_dai_data(dev, dai_data); + + return devm_snd_soc_register_component(dev, &q6afe_dai_component, + q6afe_dais, ARRAY_SIZE(q6afe_dais)); +} +EXPORT_SYMBOL_GPL(q6afe_dai_dev_probe); + +int q6afe_dai_dev_remove(struct device *dev) +{ + return 0; +} +EXPORT_SYMBOL_GPL(q6afe_dai_dev_remove); +MODULE_DESCRIPTION("Q6 Audio Fronend dai driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index 43df524f01bb..647ed2d15545 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -26,6 +26,9 @@ struct q6afe_port; void q6afe_set_dai_data(struct device *dev, void *data); void *q6afe_get_dai_data(struct device *dev);

+int q6afe_dai_dev_probe(struct device *dev); +int q6afe_dai_dev_remove(struct device *dev); + struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id); int q6afe_port_start(struct q6afe_port *port); int q6afe_port_stop(struct q6afe_port *port);

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 --- include/dt-bindings/sound/qcom,q6afe.h | 14 ++++ sound/soc/qcom/qdsp6/q6afe.c | 129 +++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 16 +++- 3 files changed, 158 insertions(+), 1 deletion(-)

diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h index b4d82cccdc86..e9004ee39f72 100644 --- a/include/dt-bindings/sound/qcom,q6afe.h +++ b/include/dt-bindings/sound/qcom,q6afe.h @@ -4,6 +4,20 @@

/* Audio Front End (AFE) Ports */ #define AFE_PORT_HDMI_RX 8 +#define SLIMBUS_0_RX 15 +#define SLIMBUS_0_TX 16 +#define SLIMBUS_1_RX 17 +#define SLIMBUS_1_TX 18 +#define SLIMBUS_2_RX 19 +#define SLIMBUS_2_TX 20 +#define SLIMBUS_3_RX 21 +#define SLIMBUS_3_TX 22 +#define SLIMBUS_4_RX 23 +#define SLIMBUS_4_TX 24 +#define SLIMBUS_5_RX 25 +#define SLIMBUS_5_TX 26 +#define SLIMBUS_6_RX 45 +#define SLIMBUS_6_TX 46

#endif /* __DT_BINDINGS_Q6_AFE_H__ */

diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 0a5af06bb50e..637390f5421e 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -25,9 +25,45 @@ #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 @@ -82,8 +118,53 @@ 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 { @@ -114,6 +195,20 @@ struct afe_port_map { static struct afe_port_map port_maps[AFE_PORT_MAX] = { [AFE_PORT_HDMI_RX] = { AFE_PORT_ID_MULTICHAN_HDMI_RX, AFE_PORT_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) @@ -372,6 +467,31 @@ void *q6afe_get_dai_data(struct device *dev) } EXPORT_SYMBOL_GPL(q6afe_get_dai_data);

+/** + * 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. * @@ -433,6 +553,15 @@ 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 647ed2d15545..aeacf1f2c9a9 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -5,12 +5,15 @@

#include <dt-bindings/sound/qcom,q6afe.h>

-#define AFE_PORT_MAX 9 +#define AFE_PORT_MAX 48

#define MSM_AFE_PORT_TYPE_RX 0 #define MSM_AFE_PORT_TYPE_TX 1 #define AFE_MAX_PORTS AFE_PORT_MAX

+#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; @@ -36,5 +48,7 @@ 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__ */

From: Srinivas Kandagatla srinivas.kandagatla@linaro.org

This patch adds support to SLIMBus related mixers to control mux between ASM stream and AFE port.

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/qdsp6/q6routing.c | 261 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 261 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 828243c58569..8c9b5d899ba7 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -232,6 +232,180 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = { msm_routing_put_audio_mixer), };

+static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = { + SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_0_RX, + MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_kcontrol_new slimbus_1_rx_mixer_controls[] = { + SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_1_RX, + MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_1_RX, + MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_1_RX, + MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_1_RX, + MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_1_RX, + MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_1_RX, + MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_1_RX, + MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_1_RX, + MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_kcontrol_new slimbus_2_rx_mixer_controls[] = { + SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_2_RX, + MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_2_RX, + MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_2_RX, + MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_2_RX, + MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_2_RX, + MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_2_RX, + MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_2_RX, + MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_2_RX, + MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_kcontrol_new slimbus_3_rx_mixer_controls[] = { + SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_3_RX, + MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_3_RX, + MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_3_RX, + MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_3_RX, + MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_3_RX, + MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_3_RX, + MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_3_RX, + MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_3_RX, + MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_kcontrol_new slimbus_4_rx_mixer_controls[] = { + SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_4_RX, + MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_4_RX, + MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_4_RX, + MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_kcontrol_new slimbus_5_rx_mixer_controls[] = { + SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_5_RX, + MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_kcontrol_new slimbus_6_rx_mixer_controls[] = { + SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia2", SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia3", SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia4", SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia5", SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia6", SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia7", SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia8", SLIMBUS_6_RX, + MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { /* Frontend AIF */ SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0), @@ -247,6 +421,28 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0, hdmi_mixer_controls, ARRAY_SIZE(hdmi_mixer_controls)), + + SND_SOC_DAPM_MIXER("SLIMBUS_0_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + slimbus_rx_mixer_controls, + ARRAY_SIZE(slimbus_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("SLIMBUS_1_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + slimbus_1_rx_mixer_controls, + ARRAY_SIZE(slimbus_1_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("SLIMBUS_2_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + slimbus_2_rx_mixer_controls, + ARRAY_SIZE(slimbus_2_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("SLIMBUS_3_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + slimbus_3_rx_mixer_controls, + ARRAY_SIZE(slimbus_3_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("SLIMBUS_4_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + slimbus_4_rx_mixer_controls, + ARRAY_SIZE(slimbus_4_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("SLIMBUS_5_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + slimbus_5_rx_mixer_controls, + ARRAY_SIZE(slimbus_5_rx_mixer_controls)), + SND_SOC_DAPM_MIXER("SLIMBUS_6_RX Audio Mixer", SND_SOC_NOPM, 0, 0, + slimbus_6_rx_mixer_controls, + ARRAY_SIZE(slimbus_6_rx_mixer_controls)), };

static const struct snd_soc_dapm_route intercon[] = { @@ -259,6 +455,71 @@ static const struct snd_soc_dapm_route intercon[] = { {"HDMI Mixer", "MultiMedia7", "MM_DL7"}, {"HDMI Mixer", "MultiMedia8", "MM_DL8"}, {"HDMI_RX", NULL, "HDMI Mixer"}, + + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia3", "MM_DL3"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia4", "MM_DL4"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia5", "MM_DL5"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia6", "MM_DL6"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia7", "MM_DL7"}, + {"SLIMBUS_0_RX Audio Mixer", "MultiMedia8", "MM_DL8"}, + {"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Audio Mixer"}, + + {"SLIMBUS_1_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, + {"SLIMBUS_1_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, + {"SLIMBUS_1_RX Audio Mixer", "MultiMedia3", "MM_DL3"}, + {"SLIMBUS_1_RX Audio Mixer", "MultiMedia4", "MM_DL4"}, + {"SLIMBUS_1_RX Audio Mixer", "MultiMedia5", "MM_DL5"}, + {"SLIMBUS_1_RX Audio Mixer", "MultiMedia6", "MM_DL6"}, + {"SLIMBUS_1_RX Audio Mixer", "MultiMedia7", "MM_DL7"}, + {"SLIMBUS_1_RX Audio Mixer", "MultiMedia8", "MM_DL8"}, + {"SLIMBUS_1_RX", NULL, "SLIMBUS_1_RX Audio Mixer"}, + + {"SLIMBUS_2_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, + {"SLIMBUS_2_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, + {"SLIMBUS_2_RX Audio Mixer", "MultiMedia3", "MM_DL3"}, + {"SLIMBUS_2_RX Audio Mixer", "MultiMedia4", "MM_DL4"}, + {"SLIMBUS_2_RX Audio Mixer", "MultiMedia5", "MM_DL5"}, + {"SLIMBUS_2_RX Audio Mixer", "MultiMedia6", "MM_DL6"}, + {"SLIMBUS_2_RX Audio Mixer", "MultiMedia7", "MM_DL7"}, + {"SLIMBUS_2_RX Audio Mixer", "MultiMedia8", "MM_DL8"}, + {"SLIMBUS_2_RX", NULL, "SLIMBUS_2_RX Audio Mixer"}, + + {"SLIMBUS_3_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, + {"SLIMBUS_3_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, + {"SLIMBUS_3_RX Audio Mixer", "MultiMedia3", "MM_DL3"}, + {"SLIMBUS_3_RX Audio Mixer", "MultiMedia4", "MM_DL4"}, + {"SLIMBUS_3_RX Audio Mixer", "MultiMedia5", "MM_DL5"}, + {"SLIMBUS_3_RX Audio Mixer", "MultiMedia6", "MM_DL6"}, + {"SLIMBUS_3_RX Audio Mixer", "MultiMedia7", "MM_DL7"}, + {"SLIMBUS_3_RX Audio Mixer", "MultiMedia8", "MM_DL8"}, + {"SLIMBUS_3_RX", NULL, "SLIMBUS_3_RX Audio Mixer"}, + + {"SLIMBUS_4_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, + {"SLIMBUS_4_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, + {"SLIMBUS_4_RX Audio Mixer", "MultiMedia5", "MM_DL5"}, + {"SLIMBUS_4_RX", NULL, "SLIMBUS_4_RX Audio Mixer"}, + + {"SLIMBUS_5_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, + {"SLIMBUS_5_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, + {"SLIMBUS_5_RX Audio Mixer", "MultiMedia3", "MM_DL3"}, + {"SLIMBUS_5_RX Audio Mixer", "MultiMedia4", "MM_DL4"}, + {"SLIMBUS_5_RX Audio Mixer", "MultiMedia5", "MM_DL5"}, + {"SLIMBUS_5_RX Audio Mixer", "MultiMedia6", "MM_DL6"}, + {"SLIMBUS_5_RX Audio Mixer", "MultiMedia7", "MM_DL7"}, + {"SLIMBUS_5_RX Audio Mixer", "MultiMedia8", "MM_DL8"}, + {"SLIMBUS_5_RX", NULL, "SLIMBUS_5_RX Audio Mixer"}, + + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia1", "MM_DL1"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia2", "MM_DL2"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia3", "MM_DL3"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia4", "MM_DL4"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia5", "MM_DL5"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia6", "MM_DL6"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia7", "MM_DL7"}, + {"SLIMBUS_6_RX Audio Mixer", "MultiMedia8", "MM_DL8"}, + {"SLIMBUS_6_RX", NULL, "SLIMBUS_6_RX Audio Mixer"}, };

static int routing_hw_params(struct snd_pcm_substream *substream,

From: Srinivas Kandagatla srinivas.kandagatla@linaro.org

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- include/dt-bindings/sound/qcom,q6afe.h | 10 +++ sound/soc/qcom/qdsp6/q6afe.c | 111 +++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 10 +++ 3 files changed, 131 insertions(+)

diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h index e9004ee39f72..3cd862262369 100644 --- a/include/dt-bindings/sound/qcom,q6afe.h +++ b/include/dt-bindings/sound/qcom,q6afe.h @@ -16,6 +16,16 @@ #define SLIMBUS_4_TX 24 #define SLIMBUS_5_RX 25 #define SLIMBUS_5_TX 26 +#define QUATERNARY_MI2S_RX 34 +#define QUATERNARY_MI2S_TX 35 +#define SECONDARY_MI2S_RX 36 +#define SECONDARY_MI2S_TX 37 +#define TERTIARY_MI2S_RX 38 +#define TERTIARY_MI2S_TX 39 +#define PRIMARY_MI2S_RX 40 +#define PRIMARY_MI2S_TX 41 +#define SECONDARY_PCM_RX 42 +#define SECONDARY_PCM_TX 43 #define SLIMBUS_6_RX 45 #define SLIMBUS_6_TX 46

diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 637390f5421e..c04caea5481c 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -14,6 +14,10 @@ #include <linux/of.h> #include <linux/delay.h> #include <linux/soc/qcom/apr.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> #include "q6dsp-errno.h" #include "q6afe.h"

@@ -28,6 +32,24 @@ #define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235

#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212 +#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D + +/* I2S config specific */ +#define AFE_API_VERSION_I2S_CONFIG 0x1 +#define AFE_PORT_I2S_SD0 0x1 +#define AFE_PORT_I2S_SD1 0x2 +#define AFE_PORT_I2S_SD2 0x3 +#define AFE_PORT_I2S_SD3 0x4 +#define AFE_PORT_I2S_QUAD01 0x5 +#define AFE_PORT_I2S_QUAD23 0x6 +#define AFE_PORT_I2S_6CHS 0x7 +#define AFE_PORT_I2S_8CHS 0x8 +#define AFE_PORT_I2S_MONO 0x0 +#define AFE_PORT_I2S_STEREO 0x1 +#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0 +#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1 +#define AFE_LINEAR_PCM_DATA 0x0 +

/* Port IDs */ #define AFE_API_VERSION_HDMI_CONFIG 0x1 @@ -63,6 +85,14 @@ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX 0x400c /* SLIMbus Tx port on channel 6. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX 0x400d +#define AFE_PORT_ID_PRIMARY_MI2S_RX 0x1000 +#define AFE_PORT_ID_PRIMARY_MI2S_TX 0x1001 +#define AFE_PORT_ID_SECONDARY_MI2S_RX 0x1002 +#define AFE_PORT_ID_SECONDARY_MI2S_TX 0x1003 +#define AFE_PORT_ID_TERTIARY_MI2S_RX 0x1004 +#define AFE_PORT_ID_TERTIARY_MI2S_TX 0x1005 +#define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006 +#define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007

#define TIMEOUT_MS 1000 #define AFE_CMD_RESP_AVAIL 0 @@ -161,10 +191,21 @@ struct afe_param_id_slimbus_cfg { */ } __packed;

+struct afe_param_id_i2s_cfg { + u32 i2s_cfg_minor_version; + u16 bit_width; + u16 channel_mode; + u16 mono_stereo; + u16 ws_src; + u32 sample_rate; + u16 data_format; + u16 reserved; +} __packed;

union afe_port_config { struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch; struct afe_param_id_slimbus_cfg slim_cfg; + struct afe_param_id_i2s_cfg i2s_cfg; } __packed;

struct q6afe_port { @@ -207,6 +248,14 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = { SLIMBUS_4_RX, 1, 1}, [SLIMBUS_5_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX, SLIMBUS_5_RX, 1, 1}, + [QUATERNARY_MI2S_RX] = { AFE_PORT_ID_QUATERNARY_MI2S_RX, + QUATERNARY_MI2S_RX, 1, 1}, + [SECONDARY_MI2S_RX] = { AFE_PORT_ID_SECONDARY_MI2S_RX, + SECONDARY_MI2S_RX, 1, 1}, + [TERTIARY_MI2S_RX] = { AFE_PORT_ID_TERTIARY_MI2S_RX, + TERTIARY_MI2S_RX, 1, 1}, + [PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX, + PRIMARY_MI2S_RX, 1, 1}, [SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX, SLIMBUS_6_RX, 1, 1}, }; @@ -513,6 +562,61 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port, } EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);

+/** + * q6afe_i2s_port_prepare() - Prepare i2s afe port. + * + * @port: Instance of afe port + * @cfg: I2S configuration for the afe port + * + */ +void q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg) +{ + union afe_port_config *pcfg = &port->port_cfg; + + pcfg->i2s_cfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG; + pcfg->i2s_cfg.sample_rate = cfg->sample_rate; + pcfg->i2s_cfg.bit_width = cfg->bit_width; + pcfg->i2s_cfg.data_format = AFE_LINEAR_PCM_DATA; + + switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL; + break; + case SND_SOC_DAIFMT_CBM_CFM: + /* CPU is slave */ + pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL; + break; + default: + break; + } + + switch (cfg->num_channels) { + case 1: + pcfg->i2s_cfg.mono_stereo = AFE_PORT_I2S_MONO; + pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD0; + break; + case 2: + pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD0; + pcfg->i2s_cfg.mono_stereo = AFE_PORT_I2S_STEREO; + break; + case 3: + case 4: + pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_QUAD01; + break; + case 5: + case 6: + pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_6CHS; + break; + case 7: + case 8: + pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_8CHS; + break; + default: + break; + } +} +EXPORT_SYMBOL_GPL(q6afe_i2s_port_prepare); + /** * q6afe_port_start() - Start a afe port * @@ -562,6 +666,13 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX: cfg_type = AFE_PARAM_ID_SLIMBUS_CONFIG; break; + + case AFE_PORT_ID_PRIMARY_MI2S_RX: + case AFE_PORT_ID_SECONDARY_MI2S_RX: + case AFE_PORT_ID_TERTIARY_MI2S_RX: + case AFE_PORT_ID_QUATERNARY_MI2S_RX: + cfg_type = AFE_PARAM_ID_I2S_CONFIG; + break; default: dev_err(dev, "Invalid port id 0x%x\n", port_id); return ERR_PTR(-EINVAL); diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index aeacf1f2c9a9..9114d68a79cb 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -29,9 +29,18 @@ struct q6afe_slim_cfg { u8 ch_mapping[AFE_MAX_CHAN_COUNT]; };

+struct q6afe_i2s_cfg { + u32 sample_rate; + u16 bit_width; + u16 data_format; + u16 num_channels; + int fmt; +}; + struct q6afe_port_config { struct q6afe_hdmi_cfg hdmi; struct q6afe_slim_cfg slim; + struct q6afe_i2s_cfg i2s_cfg; };

struct q6afe_port; @@ -50,5 +59,6 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port, struct q6afe_hdmi_cfg *cfg); void q6afe_slim_port_prepare(struct q6afe_port *port, struct q6afe_slim_cfg *cfg); +void q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg);

#endif /* __Q6AFE_H__ */

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 | 115 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 10 ++++ 2 files changed, 125 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index c04caea5481c..ea6fa3848007 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -31,6 +31,9 @@

#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235

+#define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238 +#define AFE_PARAM_ID_INTERNAL_DIGITAL_CDC_CLK_CONFIG 0x00010239 + #define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212 #define AFE_PARAM_ID_I2S_CONFIG 0x0001020D

@@ -94,6 +97,11 @@ #define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006 #define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007

+#define Q6AFE_LPASS_MODE_CLK1_VALID 1 +#define Q6AFE_LPASS_MODE_CLK2_VALID 2 +#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1 +#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0 + #define TIMEOUT_MS 1000 #define AFE_CMD_RESP_AVAIL 0 #define AFE_CMD_RESP_NONE 1 @@ -191,6 +199,23 @@ 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; @@ -208,6 +233,21 @@ union afe_port_config { struct afe_param_id_i2s_cfg i2s_cfg; } __packed;

+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_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; + +/* This param id is used to configure internal clk */ struct q6afe_port { wait_queue_head_t wait; union afe_port_config port_cfg; @@ -425,6 +465,81 @@ 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_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_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; + default: + ret = -EINVAL; + break; + } + + return ret; +} +EXPORT_SYMBOL_GPL(q6afe_port_set_sysclk); + static int afe_port_start(struct q6afe_port *port, union afe_port_config *afe_config) { diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h index 9114d68a79cb..c469f6de38d2 100644 --- a/sound/soc/qcom/qdsp6/q6afe.h +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -14,6 +14,13 @@ #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 + struct q6afe_hdmi_cfg { u16 datatype; u16 channel_allocation; @@ -61,4 +68,7 @@ void q6afe_slim_port_prepare(struct q6afe_port *port, struct q6afe_slim_cfg *cfg); void 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__ */

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 | 142 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 142 insertions(+)

diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 8c9b5d899ba7..8ffcc49a79a1 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -232,6 +232,102 @@ 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, @@ -443,6 +539,18 @@ 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)), };

static const struct snd_soc_dapm_route intercon[] = { @@ -520,6 +628,40 @@ 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"}, };

static int routing_hw_params(struct snd_pcm_substream *substream,

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 | 8 +++ sound/soc/qcom/apq8096.c | 173 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 181 insertions(+) create mode 100644 sound/soc/qcom/apq8096.c

diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 8c2d65e0a28e..fa4b575c086c 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -77,3 +77,11 @@ 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_QDSP6V2 + default n + help + To add support for SoC audio on MSM8996 and APQ8096 boards diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c new file mode 100644 index 000000000000..b12d2398557a --- /dev/null +++ b/sound/soc/qcom/apq8096.c @@ -0,0 +1,173 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2018, Linaro Limited + */ + +#include <linux/soc/qcom/apr.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> + +static int msm8996_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 *dev = card->dev; + struct snd_soc_dai_link *link; + struct device_node *np, *codec, *platform, *cpu, *node = dev->of_node; + int ret, num_links; + bool is_fe; + + + ret = snd_soc_of_parse_card_name(card, "qcom,model"); + if (ret) + dev_err(dev, "Error parsing card name: %d\n", ret); + + if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) + ret = snd_soc_of_parse_audio_routing(card, + "qcom,audio-routing"); + + /* Populate links */ + num_links = of_get_child_count(node); + + dev_info(dev, "Found %d child audio dai links..\n", num_links); + /* Allocate the private data and the DAI link array */ + card->dai_link = devm_kzalloc(dev, sizeof(*link) * num_links, + GFP_KERNEL); + if (!card->dai_link) + return -ENOMEM; + + card->num_links = num_links; + + link = card->dai_link; + + for_each_child_of_node(node, np) { + is_fe = false; + if (of_property_read_bool(np, "is-fe")) + is_fe = true; + + if (is_fe) { + /* BE is dummy */ + link->codec_of_node = NULL; + link->codec_dai_name = "snd-soc-dummy-dai"; + link->codec_name = "snd-soc-dummy"; + + /* FE settings */ + link->dynamic = 1; + link->dpcm_playback = 1; + + } else { + link->no_pcm = 1; + link->dpcm_playback = 1; + link->ignore_suspend = 1; + link->ignore_pmdown_time = 1; + link->be_hw_params_fixup = msm8996_be_hw_params_fixup; + } + + 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; + } + } + + 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->stream_name = link->name; + link++; + } + + return ret; +} + +static int msm_snd_apq8096_probe(struct apr_device *adev) +{ + int ret; + struct snd_soc_card *card; + + card = devm_kzalloc(&adev->dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->dev = &adev->dev; + + ret = apq8096_sbc_parse_of(card); + if (ret) { + dev_err(&adev->dev, "Error parsing OF data\n"); + return ret; + } + + ret = devm_snd_soc_register_card(&adev->dev, card); + if (ret) + dev_err(&adev->dev, "sound card register failed (%d)!\n", ret); + else + dev_err(&adev->dev, "sound card register Sucessfull\n"); + + return ret; +} + +static const struct of_device_id msm_snd_apq8096_dt_match[] = { + {.compatible = "qcom,apq8096-sndcard"}, + {} +}; + +static struct apr_driver msm_snd_apq8096_driver = { + .probe = msm_snd_apq8096_probe, + .driver = { + .name = "msm-snd-apq8096", + .owner = THIS_MODULE, + .of_match_table = msm_snd_apq8096_dt_match, + }, +}; +module_apr_driver(msm_snd_apq8096_driver); +MODULE_AUTHOR("Srinivas Kandagatla <srinivas.kandagatla@linaro.org"); +MODULE_DESCRIPTION("APQ8096 ASoC Machine Driver"); +MODULE_LICENSE("GPL v2");