From: Srinivas Kandagatla srinivas.kandagatla@linaro.org

This patch add dt bindings for Qualcomm APR bus driver

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- .../devicetree/bindings/soc/qcom/qcom,apr.txt | 28 ++++++++++++++++++++++ 1 file changed, 28 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..4e93213ae98d --- /dev/null +++ b/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt @@ -0,0 +1,28 @@ +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-<SOC-NAME>" example: "qcom,apr-msm8996" + + +- qcom,smd-channel: + Usage: required + Value type: <string> + Definition: standard SMD property specifying the SMD channel used for + communication with the APR on QDSP. + Should be "apr_audio_svc". + Described in soc/qcom/qcom,smd.txt + += 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-msm8996"; + qcom,smd-channels = "apr_audio_svc"; + };

From: Srinivas Kandagatla srinivas.kandagatla@linaro.org

This patch adds support to q6 AFE (Audio Front End) module on Q6DSP.

AFE module sits right at the other end of cpu where the codec/audio devices are connected.

AFE provides abstraced interfaces to both hardware and virtual devices. Each AFE tx/rx port can be configured to connect to one of the hardware devices like codec, hdmi, slimbus, i2s and so on. AFE services include starting, stopping, and if needed, any configurations of the ports.

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/Kconfig | 13 ++ sound/soc/qcom/Makefile | 5 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe.c | 503 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 30 +++ 5 files changed, 552 insertions(+) create mode 100644 sound/soc/qcom/qdsp6/Makefile create mode 100644 sound/soc/qcom/qdsp6/q6afe.c create mode 100644 sound/soc/qcom/qdsp6/q6afe.h

diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 8ec9a074b38b..1db92069a6a0 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -43,3 +43,16 @@ config SND_SOC_APQ8016_SBC Support for Qualcomm Technologies LPASS audio block in APQ8016 SOC-based systems. Say Y if you want to use audio devices on MI2S. + +config SND_SOC_QDSP6_AFE + tristate + default n + +config SND_SOC_QDSP6 + tristate "SoC ALSA audio driver for QDSP6" + select SND_SOC_QDSP6_AFE + help + To add support for MSM QDSP6 Soc Audio. + This will enable sound soc platform specific + audio drivers. This includes q6asm, q6adm, + q6afe interfaces to DSP using apr. diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index d5280355c24f..748f5e891dcf 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -13,6 +13,11 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o # Machine snd-soc-storm-objs := storm.o snd-soc-apq8016-sbc-objs := apq8016_sbc.o +snd-soc-msm8996-objs := apq8096.o

obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o +obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-msm8996.o + +#DSP lib +obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile new file mode 100644 index 000000000000..313e65f571db --- /dev/null +++ b/sound/soc/qcom/qdsp6/Makefile @@ -0,0 +1 @@ +obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c new file mode 100644 index 000000000000..a6a782e6f17d --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -0,0 +1,503 @@ +/* SPDX-License-Identifier: GPL-2.0 +* Copyright (c) 2011-2016, The Linux Foundation +* Copyright (c) 2017, Linaro Limited +*/ +#include <linux/slab.h> +#include <linux/kernel.h> +#include <linux/uaccess.h> +#include <linux/wait.h> +#include <linux/jiffies.h> +#include <linux/sched.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/soc/qcom/apr.h> +#include "common.h" +#include "q6afe.h" + +/* AFE CMDs */ +#define AFE_PORT_CMD_DEVICE_START 0x000100E5 +#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6 +#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF +#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106 +#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210 +#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C + +/* Port IDs */ +#define AFE_API_VERSION_HDMI_CONFIG 0x1 +#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E +#define TIMEOUT_MS 1000 +#define AFE_CMD_RESP_AVAIL 0 +#define AFE_CMD_RESP_NONE 1 + + +struct q6afev2 { + void *apr; + struct device *dev; + int state; + int status; + struct platform_device *daidev; + + struct mutex afe_cmd_lock; + struct list_head port_list; + spinlock_t port_list_lock; + struct list_head node; +}; + +struct afe_port_cmd_device_start { + struct apr_hdr hdr; + u16 port_id; + u16 reserved; +} __packed; + +struct afe_port_cmd_device_stop { + struct apr_hdr hdr; + u16 port_id; + u16 reserved; +/* Reserved for 32-bit alignment. This field must be set to 0.*/ +} __packed; + +struct afe_port_param_data_v2 { + u32 module_id; + u32 param_id; + u16 param_size; + u16 reserved; +} __packed; + +struct afe_port_cmd_set_param_v2 { + u16 port_id; + u16 payload_size; + u32 payload_address_lsw; + u32 payload_address_msw; + u32 mem_map_handle; +} __packed; + +struct afe_param_id_hdmi_multi_chan_audio_cfg { + u32 hdmi_cfg_minor_version; + u16 datatype; + u16 channel_allocation; + u32 sample_rate; + u16 bit_width; + u16 reserved; +} __packed; + +union afe_port_config { + struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch; +} __packed; + +struct q6afe_port { + wait_queue_head_t wait; + union afe_port_config port_cfg; + int token; + int id; + int cfg_type; + union { + struct q6afev2 *v2; + } afe; + struct list_head node; +}; + +struct afe_audioif_config_command { + struct apr_hdr hdr; + struct afe_port_cmd_set_param_v2 param; + struct afe_port_param_data_v2 pdata; + union afe_port_config port; +} __packed; + +struct afe_port_map { + int port_id; + int token; + int is_rx; +}; + +/* Port map of index vs real hw port ids */ +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}, +}; + +static struct q6afe_port *afe_find_port(struct q6afev2 *afe, int token) +{ + struct q6afe_port *p = NULL; + + spin_lock(&afe->port_list_lock); + list_for_each_entry(p, &afe->port_list, node) + if (p->token == token) + break; + + spin_unlock(&afe->port_list_lock); + return p; +} + +static int afe_callback(struct apr_device *adev, struct apr_client_data *data) +{ + struct q6afev2 *afe = dev_get_drvdata(&adev->dev);//priv; + struct q6afe_port *port; + + if (!data) { + dev_err(afe->dev, "%s: Invalid param data\n", __func__); + return -EINVAL; + } + + if (data->payload_size) { + uint32_t *payload = data->payload; + + if (data->opcode == APR_BASIC_RSP_RESULT) { + if (payload[1] != 0) { + afe->status = payload[1]; + dev_err(afe->dev, + "cmd = 0x%x returned error = 0x%x\n", + payload[0], payload[1]); + } + switch (payload[0]) { + case AFE_PORT_CMD_SET_PARAM_V2: + case AFE_PORT_CMD_DEVICE_STOP: + case AFE_PORT_CMD_DEVICE_START: + afe->state = AFE_CMD_RESP_AVAIL; + port = afe_find_port(afe, data->token); + if (port) + wake_up(&port->wait); + + break; + default: + dev_err(afe->dev, "Unknown cmd 0x%x\n", + payload[0]); + break; + } + } + } + return 0; +} +/** + * q6afe_get_port_id() - Get port id from a given port index + * + * @index: port index + * + * Return: Will be an negative on error or valid port_id on success + */ +int q6afe_get_port_id(int index) +{ + if (index < 0 || index > AFE_PORT_MAX) + return -EINVAL; + + return port_maps[index].port_id; +} +EXPORT_SYMBOL_GPL(q6afe_get_port_id); + +static int afe_apr_send_pkt(struct q6afev2 *afe, void *data, + wait_queue_head_t *wait) +{ + int ret; + + if (wait) + afe->state = AFE_CMD_RESP_NONE; + + afe->status = 0; + ret = apr_send_pkt(afe->apr, data); + if (ret > 0) { + if (wait) { + ret = wait_event_timeout(*wait, + (afe->state == + AFE_CMD_RESP_AVAIL), + msecs_to_jiffies(TIMEOUT_MS)); + if (!ret) { + ret = -ETIMEDOUT; + } else if (afe->status > 0) { + dev_err(afe->dev, "DSP returned error[%s]\n", + adsp_err_get_err_str(afe->status)); + ret = adsp_err_get_lnx_err_code(afe->status); + } else { + ret = 0; + } + } else { + ret = 0; + } + } else { + dev_err(afe->dev, "packet not transmitted\n"); + ret = -EINVAL; + } + + return ret; +} + +static int afe_send_cmd_port_start(struct q6afe_port *port) +{ + u16 port_id = port->id; + struct afe_port_cmd_device_start start; + struct q6afev2 *afe = port->afe.v2; + int ret, index; + + index = port->token; + start.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(APR_HDR_SIZE), + APR_PKT_VER); + start.hdr.pkt_size = sizeof(start); + start.hdr.src_port = 0; + start.hdr.dest_port = 0; + start.hdr.token = index; + start.hdr.opcode = AFE_PORT_CMD_DEVICE_START; + start.port_id = port_id; + + ret = afe_apr_send_pkt(afe, &start, &port->wait); + if (ret) + dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n", + port_id, ret); + + return ret; +} + +static int afe_port_start(struct q6afe_port *port, + union afe_port_config *afe_config) +{ + struct afe_audioif_config_command config; + struct q6afev2 *afe = port->afe.v2; + int ret = 0; + int port_id = port->id; + int cfg_type; + int index = 0; + + if (!afe_config) { + dev_err(afe->dev, "Error, no configuration data\n"); + ret = -EINVAL; + return ret; + } + + index = port->token; + + mutex_lock(&afe->afe_cmd_lock); + /* Also send the topology id here: */ + config.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(APR_HDR_SIZE), + APR_PKT_VER); + config.hdr.pkt_size = sizeof(config); + config.hdr.src_port = 0; + config.hdr.dest_port = 0; + config.hdr.token = index; + + cfg_type = port->cfg_type; + config.hdr.opcode = AFE_PORT_CMD_SET_PARAM_V2; + config.param.port_id = port_id; + config.param.payload_size = sizeof(config) - sizeof(struct apr_hdr) - + sizeof(config.param); + config.param.payload_address_lsw = 0x00; + config.param.payload_address_msw = 0x00; + config.param.mem_map_handle = 0x00; + config.pdata.module_id = AFE_MODULE_AUDIO_DEV_INTERFACE; + config.pdata.param_id = cfg_type; + config.pdata.param_size = sizeof(config.port); + + config.port = *afe_config; + + ret = afe_apr_send_pkt(afe, &config, &port->wait); + if (ret) { + dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n", + port_id, ret); + goto fail_cmd; + } + + ret = afe_send_cmd_port_start(port); + +fail_cmd: + mutex_unlock(&afe->afe_cmd_lock); + return ret; +} + +/** + * q6afe_port_stop() - Stop a afe port + * + * @port: Instance of port to stop + * + * Return: Will be an negative on packet size on success. + */ +int q6afe_port_stop(struct q6afe_port *port) +{ + int port_id = port->id; + struct afe_port_cmd_device_stop stop; + struct q6afev2 *afe = port->afe.v2; + int ret = 0; + int index = 0; + + port_id = port->id; + index = port->token; + if (index < 0 || index > AFE_PORT_MAX) { + dev_err(afe->dev, "AFE port index[%d] invalid!\n", index); + return -EINVAL; + } + + stop.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, + APR_HDR_LEN(APR_HDR_SIZE), + APR_PKT_VER); + stop.hdr.pkt_size = sizeof(stop); + stop.hdr.src_port = 0; + stop.hdr.dest_port = 0; + stop.hdr.token = index; + stop.hdr.opcode = AFE_PORT_CMD_DEVICE_STOP; + stop.port_id = port_id; + stop.reserved = 0; + + ret = afe_apr_send_pkt(afe, &stop, &port->wait); + if (ret) + dev_err(afe->dev, "AFE close failed %d\n", ret); + + return ret; +} +EXPORT_SYMBOL_GPL(q6afe_port_stop); + +/** + * q6afe_hdmi_port_prepare() - Prepare hdmi afe port. + * + * @port: Instance of afe port + * @cfg: HDMI configuration for the afe port + * + */ +void q6afe_hdmi_port_prepare(struct q6afe_port *port, + struct q6afe_hdmi_cfg *cfg) +{ + union afe_port_config *pcfg = &port->port_cfg; + + pcfg->hdmi_multi_ch.hdmi_cfg_minor_version = + AFE_API_VERSION_HDMI_CONFIG; + pcfg->hdmi_multi_ch.datatype = cfg->datatype; + pcfg->hdmi_multi_ch.channel_allocation = cfg->channel_allocation; + pcfg->hdmi_multi_ch.sample_rate = cfg->sample_rate; + pcfg->hdmi_multi_ch.bit_width = cfg->bit_width; +} +EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare); + +/** + * q6afe_port_start() - Start a afe port + * + * @port: Instance of port to start + * + * Return: Will be an negative on packet size on success. + */ +int q6afe_port_start(struct q6afe_port *port) +{ + return afe_port_start(port, &port->port_cfg); +} +EXPORT_SYMBOL_GPL(q6afe_port_start); + +/** + * q6afe_port_get_from_id() - Get port instance from a port id + * + * @dev: Pointer to afe child device. + * @id: port id + * + * Return: Will be an error pointer on error or a valid afe port + * on success. + */ +struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id) +{ + int port_id; + struct q6afev2 *afe = dev_get_drvdata(dev->parent); + struct q6afe_port *port; + int token; + int cfg_type; + + if (!afe) { + dev_err(dev, "Unable to find instance of afe service\n"); + return ERR_PTR(-ENOENT); + } + + token = id; + if (token < 0 || token > AFE_PORT_MAX) { + dev_err(dev, "AFE port token[%d] invalid!\n", token); + return ERR_PTR(-EINVAL); + } + + port_id = port_maps[id].port_id; + + port = devm_kzalloc(dev, sizeof(*port), GFP_KERNEL); + if (!port) + return ERR_PTR(-ENOMEM); + + init_waitqueue_head(&port->wait); + + port->token = token; + port->id = port_id; + + port->afe.v2 = afe; + switch (port_id) { + case AFE_PORT_ID_MULTICHAN_HDMI_RX: + cfg_type = AFE_PARAM_ID_HDMI_CONFIG; + break; + default: + dev_err(dev, "Invalid port id 0x%x\n", port_id); + return ERR_PTR(-EINVAL); + } + + port->cfg_type = cfg_type; + + spin_lock(&afe->port_list_lock); + list_add_tail(&port->node, &afe->port_list); + spin_unlock(&afe->port_list_lock); + + return port; + +} +EXPORT_SYMBOL_GPL(q6afe_port_get_from_id); + +/** + * q6afe_port_put() - Release port reference + * + * @port: Instance of port to put + */ +void q6afe_port_put(struct q6afe_port *port) +{ + struct q6afev2 *afe = port->afe.v2; + + spin_lock(&afe->port_list_lock); + list_del(&port->node); + spin_unlock(&afe->port_list_lock); +} +EXPORT_SYMBOL_GPL(q6afe_port_put); + +static int q6afev2_probe(struct apr_device *adev) +{ + struct q6afev2 *afe; + struct device *dev = &adev->dev; + + afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL); + if (!afe) + return -ENOMEM; + + afe->apr = adev; + mutex_init(&afe->afe_cmd_lock); + afe->dev = dev; + INIT_LIST_HEAD(&afe->port_list); + spin_lock_init(&afe->port_list_lock); + + dev_set_drvdata(dev, afe); + + afe->daidev = platform_device_register_data(&adev->dev, "q6afe_dai", + -1, NULL, 0); + return 0; +} + +static int q6afev2_remove(struct apr_device *adev) +{ + struct q6afev2 *afe = dev_get_drvdata(&adev->dev); + + platform_device_unregister(afe->daidev); + + return 0; +} + +static const struct apr_device_id q6asm_id[] = { + {"Q6AFE", APR_DOMAIN_ADSP, APR_SVC_AFE, APR_CLIENT_AUDIO}, + {} +}; + +static struct apr_driver qcom_q6afe_driver = { + .probe = q6afev2_probe, + .remove = q6afev2_remove, + .callback = afe_callback, + .id_table = q6asm_id, + .driver = { + .name = "qcom-q6afe", + }, +}; + +module_apr_driver(qcom_q6afe_driver); +MODULE_DESCRIPTION("Q6 Audio Front End"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h new file mode 100644 index 000000000000..c04541b94a47 --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6afe.h @@ -0,0 +1,30 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __Q6AFE_H__ +#define __Q6AFE_H__ + +/* Audio Front End (AFE) Ports */ +#define AFE_PORT_HDMI_RX 8 +#define AFE_PORT_MAX 9 + +#define MSM_AFE_PORT_TYPE_RX 0 +#define MSM_AFE_PORT_TYPE_TX 1 +#define AFE_MAX_PORTS AFE_PORT_MAX + +struct q6afe_hdmi_cfg { + u16 datatype; + u16 channel_allocation; + u32 sample_rate; + u16 bit_width; +}; + +struct q6afe_port; + +struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id); +int q6afe_port_start(struct q6afe_port *port); +int q6afe_port_stop(struct q6afe_port *port); +void q6afe_port_put(struct q6afe_port *port); +int q6afe_get_port_id(int index); +void q6afe_hdmi_port_prepare(struct q6afe_port *port, + struct q6afe_hdmi_cfg *cfg); + +#endif /* __Q6AFE_H__ */

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 | 250 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 14 +++ 4 files changed, 270 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 a307880dc992..7ebdb879a8a3 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -52,10 +52,15 @@ 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" 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 052813ea7062..49dd3ccab27b 100644 --- a/sound/soc/qcom/qdsp6/Makefile +++ b/sound/soc/qcom/qdsp6/Makefile @@ -1,2 +1,3 @@ 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..9cc583afef4d --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -0,0 +1,250 @@ +/* SPDX-License-Identifier: GPL-2.0 +* Copyright (c) 2011-2016, The Linux Foundation +* Copyright (c) 2017, 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/platform_device.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/mm.h> +#include "q6asm.h" +#include "common.h" + +#define TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */ +#define SYNC_IO_MODE 0x0001 +#define ASYNC_IO_MODE 0x0002 + +struct audio_client { + int session; + app_cb cb; + int cmd_state; + void *priv; + uint32_t io_mode; + uint64_t time_stamp; + struct apr_device *adev; + struct mutex cmd_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]; +}; + +static int q6asm_session_alloc(struct audio_client *ac, struct q6asm *a) +{ + int n = -EINVAL; + + mutex_lock(&a->session_lock); + for (n = 1; n <= MAX_SESSIONS; n++) { + if (!a->session[n]) { + a->session[n] = ac; + break; + } + } + mutex_unlock(&a->session_lock); + + return n; +} + +static bool q6asm_is_valid_audio_client(struct audio_client *ac) +{ + struct q6asm *a = dev_get_drvdata(ac->dev->parent); + int n; + + for (n = 1; n <= MAX_SESSIONS; n++) { + if (a->session[n] == ac) + return 1; + } + + return 0; +} + +static void q6asm_session_free(struct audio_client *ac) +{ + struct q6asm *a = dev_get_drvdata(ac->dev->parent); + + if (!a) + return; + + mutex_lock(&a->session_lock); + a->session[ac->session] = 0; + ac->session = 0; + ac->perf_mode = LEGACY_PCM_MODE; + mutex_unlock(&a->session_lock); +} + +/** + * q6asm_audio_client_free() - Freee allocated audio client + * + * @ac: audio client to free + */ +void q6asm_audio_client_free(struct audio_client *ac) +{ + q6asm_session_free(ac); + 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); + goto err; + } + + if (!a->session[session_id]) { + dev_err(a->dev, "session not active: %d\n", session_id); + goto err; + } + return a->session[session_id]; +err: + return NULL; +} + +static int q6asm_srvc_callback(struct apr_device *adev, struct apr_client_data *data) +{ + struct q6asm *q6asm = dev_get_drvdata(&adev->dev); + struct audio_client *ac = NULL; + uint32_t sid = 0; + uint32_t *payload; + + if (!data) { + dev_err(&adev->dev, "%s: Invalid CB\n", __func__); + return 0; + } + + payload = 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, + app_cb cb, void *priv) +{ + struct q6asm *a = dev_get_drvdata(dev->parent); + struct audio_client *ac; + int n; + + ac = kzalloc(sizeof(struct audio_client), GFP_KERNEL); + if (!ac) + return NULL; + + n = q6asm_session_alloc(ac, a); + if (n <= 0) { + dev_err(dev, "ASM Session alloc fail n=%d\n", n); + kfree(ac); + return NULL; + } + + ac->session = n; + ac->cb = cb; + ac->dev = dev; + ac->priv = priv; + ac->io_mode = 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->cmd_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); + + q6asm->pcmdev = platform_device_register_data(&adev->dev, + "q6asm_dai", -1, NULL, 0); + + return 0; +} + +static int q6asm_remove(struct apr_device *adev) +{ + struct q6asm *q6asm = dev_get_drvdata(&adev->dev); + + platform_device_unregister(q6asm->pcmdev); + + return 0; +} + +static const struct apr_device_id q6asm_id[] = { + {"Q6ASM", APR_DOMAIN_ADSP, APR_SVC_ASM, APR_CLIENT_AUDIO}, + {} +}; + +static struct apr_driver qcom_q6asm_driver = { + .probe = q6asm_probe, + .remove = q6asm_remove, + .callback = q6asm_srvc_callback, + .id_table = q6asm_id, + .driver = { + .name = "qcom-q6asm", + }, +}; + +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..7a8a9039fd89 --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -0,0 +1,14 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __Q6_ASM_H__ +#define __Q6_ASM_H__ + +#define MAX_SESSIONS 16 + +typedef void (*app_cb) (uint32_t opcode, uint32_t token, + uint32_t *payload, void *priv); +struct audio_client; +struct audio_client *q6asm_audio_client_alloc(struct device *dev, + app_cb cb, void *priv); +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 --- sound/soc/qcom/qdsp6/q6asm.c | 530 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.h | 42 ++++ 2 files changed, 571 insertions(+), 1 deletion(-)

diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 4be92441f524..dabd6509ef99 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -8,16 +8,34 @@ #include <linux/soc/qcom/apr.h> #include <linux/device.h> #include <linux/platform_device.h> +#include <uapi/sound/asound.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/mm.h> #include "q6asm.h" #include "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 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 DEFAULT_APP_TYPE 0 +#define TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */ #define TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */ #define SYNC_IO_MODE 0x0001 #define ASYNC_IO_MODE 0x0002 @@ -42,6 +60,49 @@ 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[8]; +} __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 { dma_addr_t phys; uint32_t used; @@ -408,6 +469,111 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a, return NULL; }

+static int32_t q6asm_callback(struct apr_device *adev, + struct apr_client_data *data, int session_id) +{ + struct audio_client *ac;// = (struct audio_client *)priv; + uint32_t token; + uint32_t *payload; + uint32_t wakeup_flag = 1; + uint32_t client_event = 0; + struct q6asm *q6asm = dev_get_drvdata(&adev->dev); + + if (data == NULL) + return -EINVAL; + + ac = q6asm_get_audio_client(q6asm, session_id); + if (!q6asm_is_valid_audio_client(ac)) + return -EINVAL; + + payload = data->payload; + + if (data->opcode == APR_BASIC_RSP_RESULT) { + token = data->token; + switch (payload[0]) { + 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 (payload[1] != 0) { + dev_err(ac->dev, + "cmd = 0x%x returned error = 0x%x\n", + payload[0], payload[1]); + if (wakeup_flag) { + ac->cmd_state = payload[1]; + wake_up(&ac->cmd_wait); + } + return 0; + } + break; + default: + dev_err(ac->dev, "command[0x%x] not expecting rsp\n", + payload[0]); + break; + } + + if (ac->cmd_state && wakeup_flag) { + ac->cmd_state = 0; + wake_up(&ac->cmd_wait); + } + if (ac->cb) + ac->cb(client_event, data->token, + data->payload, ac->priv); + + return 0; + } + + switch (data->opcode) { + case ASM_DATA_EVENT_WRITE_DONE_V2:{ + struct audio_port_data *port = + &ac->port[SNDRV_PCM_STREAM_PLAYBACK]; + + client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE; + + if (ac->io_mode & SYNC_IO_MODE) { + dma_addr_t phys = port->buf[data->token].phys; + + if (lower_32_bits(phys) != payload[0] || + upper_32_bits(phys) != payload[1]) { + 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_data *data) { struct q6asm *a, *q6asm = dev_get_drvdata(&adev->dev); @@ -415,12 +581,16 @@ static int q6asm_srvc_callback(struct apr_device *adev, struct apr_client_data * struct audio_port_data *port; uint32_t dir = 0; uint32_t sid = 0; + int dest_port; uint32_t *payload;

if (!data) { dev_err(&adev->dev, "%s: Invalid CB\n", __func__); return 0; } + dest_port = (data->dest_port >> 8) & 0xFF; + if (dest_port) + return q6asm_callback(adev, data, dest_port);

payload = data->payload; sid = (data->token >> 8) & 0x0F; @@ -540,6 +710,364 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, } EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);

+static int __q6asm_open_write(struct audio_client *ac, uint32_t format, + uint16_t bits_per_sample, uint32_t stream_id, + bool is_gapless_mode) +{ + struct asm_stream_cmd_open_write_v3 open; + int rc; + + q6asm_add_hdr(ac, &open.hdr, sizeof(open), true, stream_id); + ac->cmd_state = -1; + + open.hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3; + open.mode_flags = 0x00; + open.mode_flags |= ASM_LEGACY_STREAM_SESSION; + if (is_gapless_mode) + open.mode_flags |= 1 << ASM_SHIFT_GAPLESS_MODE_FLAG; + + /* source endpoint : matrix */ + open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX; + open.bits_per_sample = bits_per_sample; + open.postprocopo_id = 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 = apr_send_pkt(ac->adev, (uint32_t *) &open); + if (rc < 0) + return rc; + + rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), 5 * HZ); + if (!rc) { + dev_err(ac->dev, "timeout on open write\n"); + return -ETIMEDOUT; + } + + if (ac->cmd_state > 0) + return adsp_err_get_lnx_err_code(ac->cmd_state); + + ac->io_mode |= TUN_WRITE_IO_MODE; + + return 0; +} + +/** + * 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) +{ + return __q6asm_open_write(ac, format, bits_per_sample, + ac->stream_id, false); +} +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; + int rc; + + q6asm_add_hdr(ac, &run.hdr, sizeof(run), true, ac->stream_id); + ac->cmd_state = -1; + + run.hdr.opcode = ASM_SESSION_CMD_RUN_V2; + run.flags = flags; + run.time_lsw = lsw_ts; + run.time_msw = msw_ts; + + rc = apr_send_pkt(ac->adev, (uint32_t *) &run); + if (rc < 0) + return rc; + + if (wait) { + rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), + 5 * HZ); + if (!rc) { + dev_err(ac->dev, "timeout on run cmd\n"); + return -ETIMEDOUT; + } + if (ac->cmd_state > 0) + return adsp_err_get_lnx_err_code(ac->cmd_state); + } + + return 0; +} + +/** + * 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, + bool use_default_chmap, + char *channel_map, + uint16_t bits_per_sample) +{ + struct asm_multi_channel_pcm_fmt_blk_v2 fmt; + u8 *channel_mapping; + int rc = 0; + + q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), true, ac->stream_id); + ac->cmd_state = -1; + + 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 (use_default_chmap) { + if (q6dsp_map_channels(channel_mapping, channels)) { + dev_err(ac->dev, " map channels failed %d\n", channels); + return -EINVAL; + } + } else { + memcpy(channel_mapping, channel_map, + PCM_FORMAT_MAX_NUM_CHANNEL); + } + + rc = apr_send_pkt(ac->adev, (uint32_t *) &fmt); + if (rc < 0) + goto fail_cmd; + + rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), 5 * HZ); + if (!rc) { + dev_err(ac->dev, "timeout on format update\n"); + return -ETIMEDOUT; + } + if (ac->cmd_state > 0) + return adsp_err_get_lnx_err_code(ac->cmd_state); + + return 0; +fail_cmd: + return rc; +} +EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm); + +/** + * q6asm_write_nolock() - 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_nolock(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 dsp_buf = 0; + int rc = 0; + + if (ac->io_mode & SYNC_IO_MODE) { + port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK]; + q6asm_add_hdr(ac, &write.hdr, sizeof(write), false, + ac->stream_id); + + dsp_buf = port->dsp_buf; + ab = &port->buf[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->max_buf_cnt) + port->dsp_buf = 0; + + rc = apr_send_pkt(ac->adev, (uint32_t *) &write); + if (rc < 0) + return rc; + } + + return 0; +} +EXPORT_SYMBOL_GPL(q6asm_write_nolock); + +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 & SYNC_IO_MODE) { + used = (ac->io_mode & TUN_WRITE_IO_MODE ? 1 : 0); + mutex_lock(&ac->cmd_lock); + for (loopcnt = 0; loopcnt <= SNDRV_PCM_STREAM_CAPTURE; + loopcnt++) { + port = &ac->port[loopcnt]; + cnt = port->max_buf_cnt - 1; + port->dsp_buf = 0; + while (cnt >= 0) { + if (!port->buf) + continue; + port->buf[cnt].used = used; + cnt--; + } + } + mutex_unlock(&ac->cmd_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); + ac->cmd_state = -1; + 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; + ac->cmd_state = 0; + break; + case CMD_CLOSE: + hdr.opcode = ASM_STREAM_CMD_CLOSE; + break; + default: + return -EINVAL; + } + + rc = apr_send_pkt(ac->adev, (uint32_t *) &hdr); + if (rc < 0) + return rc; + + if (!wait) + return 0; + + rc = wait_event_timeout(ac->cmd_wait, (ac->cmd_state >= 0), 5 * HZ); + if (!rc) { + dev_err(ac->dev, "timeout response for opcode[0x%x]\n", + hdr.opcode); + return -ETIMEDOUT; + } + if (ac->cmd_state > 0) + return adsp_err_get_lnx_err_code(ac->cmd_state); + + 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 e1409c368600..b4896059da79 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -2,7 +2,34 @@ #ifndef __Q6_ASM_H__ #define __Q6_ASM_H__

+/* ASM client callback events */ +#define CMD_PAUSE 0x0001 +#define ASM_CLIENT_EVENT_CMD_PAUSE_DONE 0x1001 +#define CMD_FLUSH 0x0002 +#define ASM_CLIENT_EVENT_CMD_FLUSH_DONE 0x1002 +#define CMD_EOS 0x0003 +#define ASM_CLIENT_EVENT_CMD_EOS_DONE 0x1003 +#define CMD_CLOSE 0x0004 +#define ASM_CLIENT_EVENT_CMD_CLOSE_DONE 0x1004 +#define CMD_OUT_FLUSH 0x0005 +#define ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE 0x1005 +#define CMD_SUSPEND 0x0006 +#define ASM_CLIENT_EVENT_CMD_SUSPEND_DONE 0x1006 +#define ASM_CLIENT_EVENT_CMD_RUN_DONE 0x1008 +#define ASM_CLIENT_EVENT_DATA_WRITE_DONE 0x1009 + +#define 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 MAX_SESSIONS 16 +#define NO_TIMESTAMP 0xFF00 +#define FORMAT_LINEAR_PCM 0x0000

typedef void (*app_cb) (uint32_t opcode, uint32_t token, uint32_t *payload, void *priv); @@ -10,6 +37,21 @@ struct audio_client; struct audio_client *q6asm_audio_client_alloc(struct device *dev, app_cb cb, void *priv); void q6asm_audio_client_free(struct audio_client *ac); +int q6asm_write_nolock(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, + bool use_default_chmap, + char *channel_map, + 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 q6 routing driver which configures route between ASM and AFE module using ADM apis.

This driver uses dapm widgets to setup the matrix between AFE ports and ASM streams.

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6routing.c | 386 +++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6routing.h | 9 + 4 files changed, 401 insertions(+) create mode 100644 sound/soc/qcom/qdsp6/q6routing.c create mode 100644 sound/soc/qcom/qdsp6/q6routing.h

diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 121b9c957024..dd8fb0cde614 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -60,12 +60,17 @@ config SND_SOC_QDSP6_CORE tristate default n

+config SND_SOC_QDSP6_ROUTING + tristate + default n + config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" select SND_SOC_QDSP6_AFE select SND_SOC_QDSP6_ADM select SND_SOC_QDSP6_ASM select SND_SOC_QDSP6_CORE + select SND_SOC_QDSP6_ROUTING 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 ad7f10691e54..c1ad060a2341 100644 --- a/sound/soc/qcom/qdsp6/Makefile +++ b/sound/soc/qcom/qdsp6/Makefile @@ -2,3 +2,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o +obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c new file mode 100644 index 000000000000..f5f12d61a1ee --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -0,0 +1,386 @@ +/* SPDX-License-Identifier: GPL-2.0 +* Copyright (c) 2011-2016, The Linux Foundation +* Copyright (c) 2017, Linaro Limited +*/ +#include <linux/init.h> +#include <linux/err.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/bitops.h> +#include <linux/mutex.h> +#include <linux/of_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <sound/control.h> +#include <sound/asound.h> +#include <sound/pcm_params.h> +#include "q6afe.h" +#include "q6asm.h" +#include "q6adm.h" +#include "q6routing.h" + +struct session_data { + int state; + int port_id; + int path_type; + int app_type; + int acdb_id; + int sample_rate; + int bits_per_sample; + int channels; + int format; + int perf_mode; + int numcopps; + int fedai_id; + unsigned long copp_map; +}; + +struct msm_routing_data { + struct session_data sessions[MAX_SESSIONS]; + struct device *dev; + struct mutex lock; +}; + +static struct msm_routing_data *routing_data; + +/** + * q6routing_reg_phy_stream() - Register a new stream for route setup + * + * @fedai_id: Frontend dai id. + * @perf_mode: Performace mode. + * @stream_id: ASM stream id to map. + * @stream_type: Direction of stream + * + * Return: Will be an negative on error or a zero on success. + */ +int q6routing_reg_phy_stream(int fedai_id, int perf_mode, + int stream_id, int stream_type) +{ + int j, topology, num_copps = 0; + struct route_payload payload; + int copp_idx; + struct session_data *session; + + if (!routing_data) { + pr_err("Routing driver not yet ready\n"); + return -EINVAL; + } + + session = &routing_data->sessions[stream_id - 1]; + mutex_lock(&routing_data->lock); + session->fedai_id = fedai_id; + payload.num_copps = 0; /* only RX needs to use payload */ + topology = NULL_COPP_TOPOLOGY; + copp_idx = q6adm_open(routing_data->dev, session->port_id, + session->path_type, session->sample_rate, + session->channels, topology, perf_mode, + session->bits_per_sample, 0, 0); + if ((copp_idx < 0) || (copp_idx >= MAX_COPPS_PER_PORT)) { + mutex_unlock(&routing_data->lock); + return -EINVAL; + } + + set_bit(copp_idx, &session->copp_map); + for (j = 0; j < MAX_COPPS_PER_PORT; j++) { + unsigned long copp = session->copp_map; + + if (test_bit(j, &copp)) { + payload.port_id[num_copps] = session->port_id; + payload.copp_idx[num_copps] = j; + num_copps++; + } + } + + if (num_copps) { + payload.num_copps = num_copps; + payload.session_id = stream_id; + q6adm_matrix_map(routing_data->dev, session->path_type, + payload, perf_mode); + } + mutex_unlock(&routing_data->lock); + + return 0; +} +EXPORT_SYMBOL_GPL(q6routing_reg_phy_stream); + +static struct session_data *routing_get_session(struct msm_routing_data *data, + int port_id, int port_type) +{ + int i; + + for (i = 0; i < MAX_SESSIONS; i++) + if (port_id == data->sessions[i].port_id) + return &data->sessions[i]; + + return NULL; +} + +static struct session_data *get_session_from_id(struct msm_routing_data *data, + int fedai_id) +{ + int i; + + for (i = 0; i < MAX_SESSIONS; i++) { + if (fedai_id == data->sessions[i].fedai_id) + return &data->sessions[i]; + } + + return NULL; +} +/** + * q6routing_dereg_phy_stream() - Deregister a stream + * + * @fedai_id: Frontend dai id. + * @stream_type: Direction of stream + * + * Return: Will be an negative on error or a zero on success. + */ +void q6routing_dereg_phy_stream(int fedai_id, int stream_type) +{ + struct session_data *session; + int idx; + + session = get_session_from_id(routing_data, fedai_id); + if (!session) + return; + + for_each_set_bit(idx, &session->copp_map, MAX_COPPS_PER_PORT) + q6adm_close(routing_data->dev, session->port_id, + session->perf_mode, idx); + + session->fedai_id = -1; + session->copp_map = 0; +} +EXPORT_SYMBOL_GPL(q6routing_dereg_phy_stream); + +static int msm_routing_get_audio_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int session_id = mc->shift; + struct snd_soc_platform *platform = snd_soc_dapm_to_platform(dapm); + struct msm_routing_data *priv = snd_soc_platform_get_drvdata(platform); + struct session_data *session = &priv->sessions[session_id]; + + if (session->port_id != -1) + ucontrol->value.integer.value[0] = 1; + else + ucontrol->value.integer.value[0] = 0; + + return 0; +} + +static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_platform *platform = snd_soc_dapm_to_platform(dapm); + struct msm_routing_data *data = snd_soc_platform_get_drvdata(platform); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_dapm_update *update = NULL; + int be_id = mc->reg; + int session_id = mc->shift; + struct session_data *session = &data->sessions[session_id]; + + if (ucontrol->value.integer.value[0]) { + session->port_id = be_id; + snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update); + } else { + session->port_id = -1; + snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update); + } + + return 1; +} + +static const struct snd_kcontrol_new hdmi_mixer_controls[] = { + SOC_SINGLE_EXT("MultiMedia1", AFE_PORT_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, + msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia2", AFE_PORT_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, + msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia3", AFE_PORT_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, + msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia4", AFE_PORT_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, + msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia5", AFE_PORT_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, + msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia6", AFE_PORT_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, + msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia7", AFE_PORT_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, + msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("MultiMedia8", AFE_PORT_HDMI_RX, + MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, + msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + +static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { + /* Frontend AIF */ + /* Widget name equals to Front-End DAI name<Need confirmation>, + * Stream name must contains substring of front-end dai name + */ + SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, 0, 0, 0), + + /* Mixer definitions */ + SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0, + hdmi_mixer_controls, + ARRAY_SIZE(hdmi_mixer_controls)), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"HDMI Mixer", "MultiMedia1", "MM_DL1"}, + {"HDMI Mixer", "MultiMedia2", "MM_DL2"}, + {"HDMI Mixer", "MultiMedia3", "MM_DL3"}, + {"HDMI Mixer", "MultiMedia4", "MM_DL4"}, + {"HDMI Mixer", "MultiMedia5", "MM_DL5"}, + {"HDMI Mixer", "MultiMedia6", "MM_DL6"}, + {"HDMI Mixer", "MultiMedia7", "MM_DL7"}, + {"HDMI Mixer", "MultiMedia8", "MM_DL8"}, + {"HDMI", NULL, "HDMI Mixer"}, + {"HDMI-RX", NULL, "HDMI"}, +}; + +static int routing_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int be_id = rtd->cpu_dai->id; + struct snd_soc_platform *platform = rtd->platform; + struct msm_routing_data *data = snd_soc_platform_get_drvdata(platform); + struct session_data *session; + int port_id, port_type, path_type, bits_per_sample; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + path_type = ADM_PATH_PLAYBACK; + port_type = MSM_AFE_PORT_TYPE_RX; + } + + port_id = be_id; + + session = routing_get_session(data, port_id, port_type); + + if (!session) { + pr_err("No session matrix setup yet..\n"); + return -EINVAL; + } + + mutex_lock(&data->lock); + + session->path_type = path_type; + session->sample_rate = params_rate(params); + session->channels = params_channels(params); + session->format = params_format(params); + + if (session->format == SNDRV_PCM_FORMAT_S16_LE) + session->bits_per_sample = 16; + else if (session->format == SNDRV_PCM_FORMAT_S24_LE) + bits_per_sample = 24; + + mutex_unlock(&data->lock); + return 0; +} + +static int routing_close(struct snd_pcm_substream *substream) +{ + return 0; +} + +static int routing_prepare(struct snd_pcm_substream *substream) +{ + return 0; +} + +static struct snd_pcm_ops q6pcm_routing_ops = { + .hw_params = routing_hw_params, + .close = routing_close, + .prepare = routing_prepare, +}; + +/* Not used but frame seems to require it */ +static int msm_routing_probe(struct snd_soc_platform *platform) +{ + int i; + + for (i = 0; i < MAX_SESSIONS; i++) + routing_data->sessions[i].port_id = -1; + + snd_soc_platform_set_drvdata(platform, routing_data); + + return 0; +} + +static struct snd_soc_platform_driver msm_soc_routing_platform = { + .ops = &q6pcm_routing_ops, + .probe = msm_routing_probe, + .component_driver = { + .dapm_widgets = msm_qdsp6_widgets, + .num_dapm_widgets = ARRAY_SIZE(msm_qdsp6_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), + }, +}; + +static int q6pcm_routing_probe(struct platform_device *pdev) +{ + + routing_data = devm_kzalloc(&pdev->dev, + sizeof(*routing_data), GFP_KERNEL); + if (!routing_data) + return -ENOMEM; + + routing_data->dev = &pdev->dev; + + mutex_init(&routing_data->lock); + dev_set_drvdata(&pdev->dev, routing_data); + + return devm_snd_soc_register_platform(&pdev->dev, + &msm_soc_routing_platform); +} + +static int q6pcm_routing_remove(struct platform_device *pdev) +{ + return 0; +} + +static struct platform_driver q6pcm_routing_driver = { + .driver = { + .name = "q6routing", + .owner = THIS_MODULE, + }, + .probe = q6pcm_routing_probe, + .remove = q6pcm_routing_remove, +}; + +module_platform_driver(q6pcm_routing_driver); + +MODULE_DESCRIPTION("Q6 Routing platform"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/qdsp6/q6routing.h b/sound/soc/qcom/qdsp6/q6routing.h new file mode 100644 index 000000000000..7f0feb196acc --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6routing.h @@ -0,0 +1,9 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _Q6_PCM_ROUTING_H +#define _Q6_PCM_ROUTING_H + +int q6routing_reg_phy_stream(int fedai_id, int perf_mode, + int stream_id, int stream_type); +void q6routing_dereg_phy_stream(int fedai_id, int stream_type); + +#endif /*_Q6_PCM_ROUTING_H */

From: Srinivas Kandagatla srinivas.kandagatla@linaro.org

This patch adds support to q6asm dai driver which configures Q6ASM streams to pass pcm data. Currently the driver only exposes 2 playback streams for hdmi playback support, it can be easily extended to add all 8 streams.

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/Kconfig | 6 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6asm-dai.c | 534 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 541 insertions(+) create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c

diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 003ce182691c..ecd1e4ba834d 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -68,6 +68,11 @@ config SND_SOC_QDSP6_AFE_DAI tristate default n

+config SND_SOC_QDSP6_ASM_DAI + tristate + default n + + config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" select SND_SOC_QDSP6_AFE @@ -76,6 +81,7 @@ config SND_SOC_QDSP6 select SND_SOC_QDSP6_CORE select SND_SOC_QDSP6_ROUTING select SND_SOC_QDSP6_AFE_DAI + select SND_SOC_QDSP6_ASM_DAI 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 bd8bd02bf09e..03576a442fb5 100644 --- a/sound/soc/qcom/qdsp6/Makefile +++ b/sound/soc/qcom/qdsp6/Makefile @@ -4,3 +4,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o obj-$(CONFIG_SND_SOC_QDSP6_AFE_DAI) += q6afe-dai.o +obj-$(CONFIG_SND_SOC_QDSP6_ASM_DAI) += q6asm-dai.o diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c new file mode 100644 index 000000000000..709c5de230fa --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -0,0 +1,534 @@ +/* SPDX-License-Identifier: GPL-2.0 +* Copyright (c) 2011-2016, The Linux Foundation +* Copyright (c) 2017, Linaro Limited +*/ +#include <linux/init.h> +#include <linux/err.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm.h> +#include <asm/dma.h> +#include <linux/dma-mapping.h> +#include <linux/of_device.h> +#include <sound/pcm_params.h> +#include "q6asm.h" +#include "q6routing.h" +#include "common.h" + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 8 +#define PLAYBACK_MAX_PERIOD_SIZE 65536 +#define PLAYBACK_MIN_PERIOD_SIZE 128 + +enum stream_state { + IDLE = 0, + STOPPED, + RUNNING, +}; + +struct q6asm_dai_rtd { + struct snd_pcm_substream *substream; + dma_addr_t phys; + unsigned int pcm_size; + unsigned int pcm_count; + unsigned int pcm_irq_pos; /* IRQ position */ + unsigned int periods; + uint16_t bits_per_sample; + uint16_t source; /* Encoding source bit mask */ + + struct audio_client *audio_client; + uint16_t session_id; + + enum stream_state state; + bool set_channel_map; + char channel_map[8]; +}; + +struct q6asm_dai_data { + u64 sid; +}; + +static struct snd_pcm_hardware q6asm_dai_hardware_playback = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE), + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 1, + .channels_max = 8, + .buffer_bytes_max = (PLAYBACK_MAX_NUM_PERIODS * + PLAYBACK_MAX_PERIOD_SIZE), + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, + .fifo_size = 0, +}; + +/* Conventional and unconventional sample rate supported */ +static unsigned int supported_sample_rates[] = { + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, + 88200, 96000, 176400, 192000 +}; + +static struct snd_pcm_hw_constraint_list constraints_sample_rates = { + .count = ARRAY_SIZE(supported_sample_rates), + .list = supported_sample_rates, + .mask = 0, +}; + +static void event_handler(uint32_t opcode, uint32_t token, + uint32_t *payload, void *priv) +{ + struct q6asm_dai_rtd *prtd = priv; + struct snd_pcm_substream *substream = prtd->substream; + + switch (opcode) { + case ASM_CLIENT_EVENT_CMD_RUN_DONE: + q6asm_write_nolock(prtd->audio_client, + prtd->pcm_count, 0, 0, NO_TIMESTAMP); + break; + case ASM_CLIENT_EVENT_CMD_EOS_DONE: + prtd->state = STOPPED; + break; + case ASM_CLIENT_EVENT_DATA_WRITE_DONE: { + prtd->pcm_irq_pos += prtd->pcm_count; + snd_pcm_period_elapsed(substream); + if (prtd->state == RUNNING) + q6asm_write_nolock(prtd->audio_client, + prtd->pcm_count, 0, 0, NO_TIMESTAMP); + + break; + } + default: + break; + } +} + +static int q6asm_dai_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct q6asm_dai_data *pdata; + int ret; + + pdata = dev_get_drvdata(soc_prtd->platform->dev); + if (!pdata) + return -EINVAL; + + if (!prtd || !prtd->audio_client) { + pr_err("%s: private data null or audio client freed\n", + __func__); + return -EINVAL; + } + + prtd->pcm_count = snd_pcm_lib_period_bytes(substream); + prtd->pcm_irq_pos = 0; + /* rate and channels are sent to audio driver */ + if (prtd->state) { + /* clear the previous setup if any */ + q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_unmap_memory_regions(substream->stream, + prtd->audio_client); + q6routing_dereg_phy_stream(soc_prtd->dai_link->id, + SNDRV_PCM_STREAM_PLAYBACK); + } + + ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client, + prtd->phys, + (prtd->pcm_size / prtd->periods), + prtd->periods); + + if (ret < 0) { + pr_err("Audio Start: Buffer Allocation failed rc = %d\n", + ret); + return -ENOMEM; + } + + ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM, + prtd->bits_per_sample); + if (ret < 0) { + pr_err("%s: q6asm_open_write failed\n", __func__); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + return -ENOMEM; + } + + prtd->session_id = q6asm_get_session_id(prtd->audio_client); + ret = q6routing_reg_phy_stream(soc_prtd->dai_link->id, LEGACY_PCM_MODE, + prtd->session_id, substream->stream); + if (ret) { + pr_err("%s: stream reg failed ret:%d\n", __func__, ret); + return ret; + } + + ret = q6asm_media_format_block_multi_ch_pcm( + prtd->audio_client, runtime->rate, + runtime->channels, !prtd->set_channel_map, + prtd->channel_map, prtd->bits_per_sample); + if (ret < 0) + pr_info("%s: CMD Format block failed\n", __func__); + + prtd->state = RUNNING; + + return 0; +} + +static int q6asm_dai_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + struct snd_pcm_runtime *runtime = substream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0); + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + prtd->state = STOPPED; + ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE); + break; + default: + ret = -EINVAL; + break; + } + + return ret; +} + +static int q6asm_dai_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; + struct q6asm_dai_rtd *prtd; + struct q6asm_dai_data *pdata; + struct device *dev = soc_prtd->platform->dev; + int ret = 0; + + pdata = dev_get_drvdata(dev); + if (!pdata) { + pr_err("Platform data not found ..\n"); + return -EINVAL; + } + + prtd = kzalloc(sizeof(struct q6asm_dai_rtd), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + prtd->substream = substream; + prtd->audio_client = q6asm_audio_client_alloc(dev, + (app_cb)event_handler, prtd); + if (!prtd->audio_client) { + pr_info("%s: Could not allocate memory\n", __func__); + kfree(prtd); + return -ENOMEM; + } + +// prtd->audio_client->dev = dev; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = q6asm_dai_hardware_playback; + + ret = snd_pcm_hw_constraint_list(runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_sample_rates); + if (ret < 0) + pr_info("snd_pcm_hw_constraint_list failed\n"); + /* Ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + pr_info("snd_pcm_hw_constraint_integer failed\n"); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE, + PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE); + if (ret < 0) { + pr_err("constraint for buffer bytes min max ret = %d\n", + ret); + } + } + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); + if (ret < 0) { + pr_err("constraint for period bytes step ret = %d\n", + ret); + } + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); + if (ret < 0) { + pr_err("constraint for buffer bytes step ret = %d\n", + ret); + } + + prtd->set_channel_map = false; + runtime->private_data = prtd; + + snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback); + + runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max; + + + if (pdata->sid < 0) + prtd->phys = substream->dma_buffer.addr; + else + prtd->phys = substream->dma_buffer.addr | (pdata->sid << 32); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int q6asm_dai_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; + struct q6asm_dai_rtd *prtd = runtime->private_data; + + if (prtd->audio_client) { + q6asm_cmd(prtd->audio_client, CMD_CLOSE); + q6asm_unmap_memory_regions(substream->stream, + prtd->audio_client); + q6asm_audio_client_free(prtd->audio_client); + } + q6routing_dereg_phy_stream(soc_prtd->dai_link->id, + SNDRV_PCM_STREAM_PLAYBACK); + kfree(prtd); + return 0; +} + +static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_pcm_substream *substream) +{ + + struct snd_pcm_runtime *runtime = substream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + + if (prtd->pcm_irq_pos >= prtd->pcm_size) + prtd->pcm_irq_pos = 0; + + return bytes_to_frames(runtime, (prtd->pcm_irq_pos)); +} + +static int q6asm_dai_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_prtd = substream->private_data; + struct snd_card *card = soc_prtd->card->snd_card; + + return dma_mmap_coherent(card->dev, vma, + runtime->dma_area, runtime->dma_addr, + runtime->dma_bytes); +} + +static int q6asm_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + + prtd->pcm_size = params_buffer_bytes(params); + prtd->periods = params_periods(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + prtd->bits_per_sample = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + prtd->bits_per_sample = 24; + break; + } + + return 0; +} + +static struct snd_pcm_ops q6asm_dai_ops = { + .open = q6asm_dai_open, + .hw_params = q6asm_dai_hw_params, + .close = q6asm_dai_close, + .ioctl = snd_pcm_lib_ioctl, + .prepare = q6asm_dai_prepare, + .trigger = q6asm_dai_trigger, + .pointer = q6asm_dai_pointer, + .mmap = q6asm_dai_mmap, +}; + +static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + struct snd_pcm_substream *substream; + struct snd_card *card = rtd->card->snd_card; + struct device *dev = card->dev; + struct device_node *node = dev->of_node; + struct q6asm_dai_data *pdata = dev_get_drvdata(rtd->platform->dev); + struct of_phandle_args args; + + int size, ret = 0; + + ret = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args); + if (ret < 0) + pdata->sid = -1; + else + pdata->sid = args.args[0]; + + + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + size = q6asm_dai_hardware_playback.buffer_bytes_max; + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size, + &substream->dma_buffer); + if (ret) { + dev_err(dev, "Cannot allocate buffer(s)\n"); + return ret; + } + + return ret; +} + +static void q6asm_dai_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + int i; + + for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) { + substream = pcm->streams[i].substream; + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + } +} + +static struct snd_soc_platform_driver q6asm_soc_platform = { + .ops = &q6asm_dai_ops, + .pcm_new = q6asm_dai_pcm_new, + .pcm_free = q6asm_dai_pcm_free, + +}; + +static const struct snd_soc_dapm_route afe_pcm_routes[] = { + {"MM_DL1", NULL, "MultiMedia1 Playback" }, + {"MM_DL2", NULL, "MultiMedia2 Playback" }, + +}; + +static int fe_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm; + + dapm = snd_soc_component_get_dapm(dai->component); + snd_soc_dapm_add_routes(dapm, afe_pcm_routes, + ARRAY_SIZE(afe_pcm_routes)); + + return 0; +} + +static const struct snd_soc_component_driver q6asm_fe_dai_component = { + .name = "q6asm-fe-dai", +}; + +static struct snd_soc_dai_driver q6asm_fe_dais[] = { + { + .playback = { + .stream_name = "MultiMedia1 Playback", + .rates = (SNDRV_PCM_RATE_8000_192000| + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, + .name = "MM_DL1", + .probe = fe_dai_probe, + .id = MSM_FRONTEND_DAI_MULTIMEDIA1, + }, + { + .playback = { + .stream_name = "MultiMedia2 Playback", + .rates = (SNDRV_PCM_RATE_8000_192000| + SNDRV_PCM_RATE_KNOT), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE), + .channels_min = 1, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 192000, + }, + .name = "MM_DL2", + .probe = fe_dai_probe, + .id = MSM_FRONTEND_DAI_MULTIMEDIA2, + }, +}; + +static int q6asm_dai_probe(struct platform_device *pdev) +{ + struct q6asm_dai_data *pdata; + struct device *dev = &pdev->dev; + int rc; + + pdata = devm_kzalloc(dev, sizeof(struct q6asm_dai_data), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + + dev_set_drvdata(dev, pdata); + + rc = snd_soc_register_platform(dev, &q6asm_soc_platform); + if (rc) { + dev_err(&pdev->dev, "err_dai_platform\n"); + return rc; + } + + rc = snd_soc_register_component(dev, &q6asm_fe_dai_component, + q6asm_fe_dais, + ARRAY_SIZE(q6asm_fe_dais)); + if (rc) + dev_err(dev, "err_dai_component\n"); + + return rc; + +} + +static int q6asm_dai_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver q6asm_dai_driver = { + .driver = { + .name = "q6asm_dai", + .owner = THIS_MODULE, + }, + .probe = q6asm_dai_probe, + .remove = q6asm_dai_remove, +}; + +module_platform_driver(q6asm_dai_driver); + +MODULE_DESCRIPTION("PCM module platform driver"); +MODULE_LICENSE("GPL v2");

From: Srinivas Kandagatla srinivas.kandagatla@linaro.org

uThis patch adds support to DB820c machine driver.

Signed-off-by: Srinivas Kandagatla srinivas.kandagatla@linaro.org --- sound/soc/qcom/Kconfig | 9 +++- sound/soc/qcom/apq8096.c | 124 +++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 132 insertions(+), 1 deletion(-) create mode 100644 sound/soc/qcom/apq8096.c

diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index ecd1e4ba834d..748ccc3edefa 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -72,7 +72,6 @@ config SND_SOC_QDSP6_ASM_DAI tristate default n

- config SND_SOC_QDSP6 tristate "SoC ALSA audio driver for QDSP6" select SND_SOC_QDSP6_AFE @@ -87,3 +86,11 @@ 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" + 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..5b2036317f71 --- /dev/null +++ b/sound/soc/qcom/apq8096.c @@ -0,0 +1,124 @@ +/* + * Copyright (c) 2017 The Linux Foundation. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 and + * only version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <linux/dma-mapping.h> +#include <sound/pcm.h> + +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 struct snd_soc_dai_link msm8996_dai_links[] = { + /* FrontEnd DAI Links */ + { + .name = "MultiMedia1 Playback", + .stream_name = "MultiMedia1", + .cpu_dai_name = "MM_DL1", + .platform_name = "q6asm_dai", + .dynamic = 1, + .dpcm_playback = 1, + + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + }, + /* Backend DAI Links */ + { + .name = "HDMI Playback", + .stream_name = "q6afe_dai", + .cpu_dai_name = "HDMI", + .platform_name = "q6routing", + .no_pcm = 1, + .dpcm_playback = 1, + .be_hw_params_fixup = msm8996_be_hw_params_fixup, + .codec_dai_name = "i2s-hifi", + .codec_name = "hdmi-audio-codec.0.auto", + }, +}; + +static int apq8096_sbc_parse_of(struct snd_soc_card *card) +{ + struct device *dev = card->dev; + int ret; + + ret = snd_soc_of_parse_card_name(card, "qcom,model"); + if (ret) + dev_err(dev, "Error parsing card name: %d\n", ret); + + return ret; +} + +static int msm_snd_apq8096_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_card *card; + + card = devm_kzalloc(&pdev->dev, sizeof(*card), GFP_KERNEL); + if (!card) + return -ENOMEM; + + card->dev = &pdev->dev; + + ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + card->dai_link = msm8996_dai_links; + card->num_links = ARRAY_SIZE(msm8996_dai_links); + + ret = apq8096_sbc_parse_of(card); + if (ret) { + dev_err(&pdev->dev, "Error parsing OF data\n"); + return ret; + } + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "sound card register failed (%d)!\n", ret); + else + dev_err(&pdev->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 platform_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_platform_driver(msm_snd_apq8096_driver); +MODULE_AUTHOR("Srinivas Kandagatla <srinivas.kandagatla@linaro.org"); +MODULE_DESCRIPTION("APQ8096 ASoC Machine Driver"); +MODULE_LICENSE("GPL v2");