* [PATCH v9 01/15] ASoC: qdsp6: q6afe: Add q6afe driver
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
@ 2018-05-18 12:55 ` Srinivas Kandagatla
2018-05-21 15:50 ` Applied "ASoC: qdsp6: q6afe: Add q6afe driver" to the asoc tree Mark Brown
2018-05-18 12:55 ` [PATCH v9 02/15] ASoC: qdsp6: qdafe: Add SLIMBus port Support Srinivas Kandagatla
` (13 subsequent siblings)
14 siblings, 1 reply; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:55 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
This patch adds support to Q6AFE (Audio Front End) module on Q6DSP.
AFE module sits right at the other end of cpu where the codec/audio
devices are connected.
AFE provides abstraced interfaces to both hardware and virtual devices.
Each AFE tx/rx port can be configured to connect to one of the hardware
devices like codec, hdmi, slimbus, i2s and so on. AFE services include
starting, stopping, and if needed, any configurations of the ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6afe.c | 549 ++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 35 +++
4 files changed, 589 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6afe.c
create mode 100644 sound/soc/qcom/qdsp6/q6afe.h
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 37ee0d958145..bb0a2afb0563 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -47,11 +47,15 @@ config SND_SOC_QDSP6_COMMON
config SND_SOC_QDSP6_CORE
tristate
+config SND_SOC_QDSP6_AFE
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_CORE
+ select SND_SOC_QDSP6_AFE
help
To add support for MSM QDSP6 Soc Audio.
This will enable sound soc platform specific
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 03b8e89c9731..7ff666bd10ca 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_CORE) += q6core.o
+obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
new file mode 100644
index 000000000000..5e0032c13aab
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -0,0 +1,549 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/slab.h>
+#include <linux/kernel.h>
+#include <linux/uaccess.h>
+#include <linux/wait.h>
+#include <linux/jiffies.h>
+#include <linux/sched.h>
+#include <linux/module.h>
+#include <linux/kref.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/spinlock.h>
+#include <linux/delay.h>
+#include <linux/soc/qcom/apr.h>
+#include "q6dsp-errno.h"
+#include "q6core.h"
+#include "q6afe.h"
+
+/* AFE CMDs */
+#define AFE_PORT_CMD_DEVICE_START 0x000100E5
+#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6
+#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF
+#define AFE_SVC_CMD_SET_PARAM 0x000100f3
+#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106
+#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210
+#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C
+
+/* Port IDs */
+#define AFE_API_VERSION_HDMI_CONFIG 0x1
+#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
+#define TIMEOUT_MS 1000
+#define AFE_CMD_RESP_AVAIL 0
+#define AFE_CMD_RESP_NONE 1
+
+struct q6afe {
+ struct apr_device *apr;
+ struct device *dev;
+ struct q6core_svc_api_info ainfo;
+ struct mutex lock;
+ struct list_head port_list;
+ spinlock_t port_list_lock;
+ struct platform_device *pdev_dais;
+};
+
+struct afe_port_cmd_device_start {
+ u16 port_id;
+ u16 reserved;
+} __packed;
+
+struct afe_port_cmd_device_stop {
+ 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;
+ struct aprv2_ibasic_rsp_result_t result;
+ int token;
+ int id;
+ int cfg_type;
+ struct q6afe *afe;
+ struct kref refcount;
+ struct list_head node;
+};
+
+struct afe_port_map {
+ int port_id;
+ int token;
+ int is_rx;
+ int is_dig_pcm;
+};
+
+/*
+ * Mapping between Virtual Port IDs to DSP AFE Port ID
+ * On B Family SoCs DSP Port IDs are consistent across multiple SoCs
+ * on A Family SoCs DSP port IDs are same as virtual Port IDs.
+ */
+
+static struct afe_port_map port_maps[AFE_PORT_MAX] = {
+ [HDMI_RX] = { AFE_PORT_ID_MULTICHAN_HDMI_RX, HDMI_RX, 1, 1},
+};
+
+static void q6afe_port_free(struct kref *ref)
+{
+ struct q6afe_port *port;
+ struct q6afe *afe;
+ unsigned long flags;
+
+ port = container_of(ref, struct q6afe_port, refcount);
+ afe = port->afe;
+ spin_lock_irqsave(&afe->port_list_lock, flags);
+ list_del(&port->node);
+ spin_unlock_irqrestore(&afe->port_list_lock, flags);
+ kfree(port);
+}
+
+static struct q6afe_port *q6afe_find_port(struct q6afe *afe, int token)
+{
+ struct q6afe_port *p = NULL;
+ struct q6afe_port *ret = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&afe->port_list_lock, flags);
+ list_for_each_entry(p, &afe->port_list, node)
+ if (p->token == token) {
+ ret = p;
+ kref_get(&p->refcount);
+ break;
+ }
+
+ spin_unlock_irqrestore(&afe->port_list_lock, flags);
+ return ret;
+}
+
+static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data)
+{
+ struct q6afe *afe = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *res;
+ struct apr_hdr *hdr = &data->hdr;
+ struct q6afe_port *port;
+
+ if (!data->payload_size)
+ return 0;
+
+ res = data->payload;
+ switch (hdr->opcode) {
+ case APR_BASIC_RSP_RESULT: {
+ if (res->status) {
+ dev_err(afe->dev, "cmd = 0x%x returned error = 0x%x\n",
+ res->opcode, res->status);
+ }
+ switch (res->opcode) {
+ case AFE_PORT_CMD_SET_PARAM_V2:
+ case AFE_PORT_CMD_DEVICE_STOP:
+ case AFE_PORT_CMD_DEVICE_START:
+ case AFE_SVC_CMD_SET_PARAM:
+ port = q6afe_find_port(afe, hdr->token);
+ if (port) {
+ port->result = *res;
+ wake_up(&port->wait);
+ kref_put(&port->refcount, q6afe_port_free);
+ }
+ break;
+ default:
+ dev_err(afe->dev, "Unknown cmd 0x%x\n", res->opcode);
+ break;
+ }
+ }
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+/**
+ * q6afe_get_port_id() - Get port id from a given port index
+ *
+ * @index: port index
+ *
+ * Return: Will be an negative on error or valid port_id on success
+ */
+int q6afe_get_port_id(int index)
+{
+ if (index < 0 || index > AFE_PORT_MAX)
+ return -EINVAL;
+
+ return port_maps[index].port_id;
+}
+EXPORT_SYMBOL_GPL(q6afe_get_port_id);
+
+static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt,
+ struct q6afe_port *port)
+{
+ wait_queue_head_t *wait = &port->wait;
+ struct apr_hdr *hdr = &pkt->hdr;
+ int ret;
+
+ mutex_lock(&afe->lock);
+ port->result.opcode = 0;
+ port->result.status = 0;
+
+ ret = apr_send_pkt(afe->apr, pkt);
+ if (ret < 0) {
+ dev_err(afe->dev, "packet not transmitted (%d)\n", ret);
+ ret = -EINVAL;
+ goto err;
+ }
+
+ ret = wait_event_timeout(*wait, (port->result.opcode == hdr->opcode),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ ret = -ETIMEDOUT;
+ } else if (port->result.status > 0) {
+ dev_err(afe->dev, "DSP returned error[%x]\n",
+ port->result.status);
+ ret = -EINVAL;
+ } else {
+ ret = 0;
+ }
+
+err:
+ mutex_unlock(&afe->lock);
+
+ return ret;
+}
+
+static int q6afe_port_set_param_v2(struct q6afe_port *port, void *data,
+ int param_id, int module_id, int psize)
+{
+ struct afe_port_cmd_set_param_v2 *param;
+ struct afe_port_param_data_v2 *pdata;
+ struct q6afe *afe = port->afe;
+ struct apr_pkt *pkt;
+ u16 port_id = port->id;
+ int ret, pkt_size;
+ void *p, *pl;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*param) + sizeof(*pdata) + psize;
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ param = p + APR_HDR_SIZE;
+ pdata = p + APR_HDR_SIZE + sizeof(*param);
+ pl = p + APR_HDR_SIZE + sizeof(*param) + sizeof(*pdata);
+ memcpy(pl, data, psize);
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.token = port->token;
+ pkt->hdr.opcode = AFE_PORT_CMD_SET_PARAM_V2;
+
+ param->port_id = port_id;
+ param->payload_size = sizeof(*pdata) + psize;
+ param->payload_address_lsw = 0x00;
+ param->payload_address_msw = 0x00;
+ param->mem_map_handle = 0x00;
+ pdata->module_id = module_id;
+ pdata->param_id = param_id;
+ pdata->param_size = psize;
+
+ ret = afe_apr_send_pkt(afe, pkt, port);
+ if (ret)
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+
+ kfree(pkt);
+ 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)
+{
+ struct afe_port_cmd_device_stop *stop;
+ struct q6afe *afe = port->afe;
+ struct apr_pkt *pkt;
+ int port_id = port->id;
+ int ret = 0;
+ int index, pkt_size;
+ void *p;
+
+ 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;
+ }
+
+ pkt_size = APR_HDR_SIZE + sizeof(*stop);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ stop = p + APR_HDR_SIZE;
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.token = index;
+ pkt->hdr.opcode = AFE_PORT_CMD_DEVICE_STOP;
+ stop->port_id = port_id;
+ stop->reserved = 0;
+
+ ret = afe_apr_send_pkt(afe, pkt, port);
+ if (ret)
+ dev_err(afe->dev, "AFE close failed %d\n", ret);
+
+ kfree(pkt);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6afe_port_stop);
+
+/**
+ * q6afe_hdmi_port_prepare() - Prepare hdmi afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: HDMI configuration for the afe port
+ *
+ */
+void q6afe_hdmi_port_prepare(struct q6afe_port *port,
+ struct q6afe_hdmi_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+
+ pcfg->hdmi_multi_ch.hdmi_cfg_minor_version =
+ AFE_API_VERSION_HDMI_CONFIG;
+ pcfg->hdmi_multi_ch.datatype = cfg->datatype;
+ pcfg->hdmi_multi_ch.channel_allocation = cfg->channel_allocation;
+ pcfg->hdmi_multi_ch.sample_rate = cfg->sample_rate;
+ pcfg->hdmi_multi_ch.bit_width = cfg->bit_width;
+}
+EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
+
+/**
+ * q6afe_port_start() - Start a afe port
+ *
+ * @port: Instance of port to start
+ *
+ * Return: Will be an negative on packet size on success.
+ */
+int q6afe_port_start(struct q6afe_port *port)
+{
+ struct afe_port_cmd_device_start *start;
+ struct q6afe *afe = port->afe;
+ int port_id = port->id;
+ int ret, param_id = port->cfg_type;
+ struct apr_pkt *pkt;
+ int pkt_size;
+ void *p;
+
+ ret = q6afe_port_set_param_v2(port, &port->port_cfg, param_id,
+ AFE_MODULE_AUDIO_DEV_INTERFACE,
+ sizeof(port->port_cfg));
+ if (ret) {
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+ return ret;
+ }
+
+ pkt_size = APR_HDR_SIZE + sizeof(*start);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ start = p + APR_HDR_SIZE;
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.token = port->token;
+ pkt->hdr.opcode = AFE_PORT_CMD_DEVICE_START;
+
+ start->port_id = port_id;
+
+ ret = afe_apr_send_pkt(afe, pkt, port);
+ if (ret)
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+
+ kfree(pkt);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6afe_port_start);
+
+/**
+ * q6afe_port_get_from_id() - Get port instance from a port id
+ *
+ * @dev: Pointer to afe child device.
+ * @id: port id
+ *
+ * Return: Will be an error pointer on error or a valid afe port
+ * on success.
+ */
+struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
+{
+ int port_id;
+ struct q6afe *afe = dev_get_drvdata(dev->parent);
+ struct q6afe_port *port;
+ unsigned long flags;
+ int cfg_type;
+
+ if (id < 0 || id > AFE_PORT_MAX) {
+ dev_err(dev, "AFE port token[%d] invalid!\n", id);
+ return ERR_PTR(-EINVAL);
+ }
+
+ /* if port is multiple times bind/unbind before callback finishes */
+ port = q6afe_find_port(afe, id);
+ if (port) {
+ dev_err(dev, "AFE Port already open\n");
+ return port;
+ }
+
+ port_id = port_maps[id].port_id;
+
+ switch (port_id) {
+ case AFE_PORT_ID_MULTICHAN_HDMI_RX:
+ cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
+ break;
+ default:
+ dev_err(dev, "Invalid port id 0x%x\n", port_id);
+ return ERR_PTR(-EINVAL);
+ }
+
+ port = kzalloc(sizeof(*port), GFP_KERNEL);
+ if (!port)
+ return ERR_PTR(-ENOMEM);
+
+ init_waitqueue_head(&port->wait);
+
+ port->token = id;
+ port->id = port_id;
+ port->afe = afe;
+ port->cfg_type = cfg_type;
+ kref_init(&port->refcount);
+
+ spin_lock_irqsave(&afe->port_list_lock, flags);
+ list_add_tail(&port->node, &afe->port_list);
+ spin_unlock_irqrestore(&afe->port_list_lock, flags);
+
+ return port;
+
+}
+EXPORT_SYMBOL_GPL(q6afe_port_get_from_id);
+
+/**
+ * q6afe_port_put() - Release port reference
+ *
+ * @port: Instance of port to put
+ */
+void q6afe_port_put(struct q6afe_port *port)
+{
+ kref_put(&port->refcount, q6afe_port_free);
+}
+EXPORT_SYMBOL_GPL(q6afe_port_put);
+
+static int q6afe_probe(struct apr_device *adev)
+{
+ struct q6afe *afe;
+ struct device *dev = &adev->dev;
+ struct device_node *dais_np;
+
+ afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL);
+ if (!afe)
+ return -ENOMEM;
+
+ q6core_get_svc_api_info(adev->svc_id, &afe->ainfo);
+ afe->apr = adev;
+ mutex_init(&afe->lock);
+ afe->dev = dev;
+ INIT_LIST_HEAD(&afe->port_list);
+ spin_lock_init(&afe->port_list_lock);
+
+ dev_set_drvdata(dev, afe);
+
+ dais_np = of_get_child_by_name(dev->of_node, "dais");
+ if (dais_np) {
+ afe->pdev_dais = of_platform_device_create(dais_np,
+ "q6afe-dai", dev);
+ of_node_put(dais_np);
+ }
+
+ return 0;
+}
+
+static int q6afe_remove(struct apr_device *adev)
+{
+ struct q6afe *afe = dev_get_drvdata(&adev->dev);
+
+ if (afe->pdev_dais)
+ of_platform_device_destroy(&afe->pdev_dais->dev, NULL);
+
+ return 0;
+}
+
+static const struct of_device_id q6afe_device_id[] = {
+ { .compatible = "qcom,q6afe" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6afe_device_id);
+
+static struct apr_driver qcom_q6afe_driver = {
+ .probe = q6afe_probe,
+ .remove = q6afe_remove,
+ .callback = q6afe_callback,
+ .driver = {
+ .name = "qcom-q6afe",
+ .of_match_table = of_match_ptr(q6afe_device_id),
+
+ },
+};
+
+module_apr_driver(qcom_q6afe_driver);
+MODULE_DESCRIPTION("Q6 Audio Front End");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
new file mode 100644
index 000000000000..3bd991a7c42d
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -0,0 +1,35 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+
+#ifndef __Q6AFE_H__
+#define __Q6AFE_H__
+
+#include <dt-bindings/sound/qcom,q6afe.h>
+
+#define AFE_PORT_MAX 48
+
+#define MSM_AFE_PORT_TYPE_RX 0
+#define MSM_AFE_PORT_TYPE_TX 1
+#define AFE_MAX_PORTS AFE_PORT_MAX
+
+struct q6afe_hdmi_cfg {
+ u16 datatype;
+ u16 channel_allocation;
+ u32 sample_rate;
+ u16 bit_width;
+};
+
+struct q6afe_port_config {
+ struct q6afe_hdmi_cfg hdmi;
+};
+
+struct q6afe_port;
+
+struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id);
+int q6afe_port_start(struct q6afe_port *port);
+int q6afe_port_stop(struct q6afe_port *port);
+void q6afe_port_put(struct q6afe_port *port);
+int q6afe_get_port_id(int index);
+void q6afe_hdmi_port_prepare(struct q6afe_port *port,
+ struct q6afe_hdmi_cfg *cfg);
+
+#endif /* __Q6AFE_H__ */
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* Applied "ASoC: qdsp6: q6afe: Add q6afe driver" to the asoc tree
2018-05-18 12:55 ` [PATCH v9 01/15] ASoC: qdsp6: q6afe: Add q6afe driver Srinivas Kandagatla
@ 2018-05-21 15:50 ` Mark Brown
0 siblings, 0 replies; 25+ messages in thread
From: Mark Brown @ 2018-05-21 15:50 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, broonie, linux-arm-msm, alsa-devel, bgoswami,
mark.rutland, devicetree, rohkumar, gregkh, plai, lgirdwood,
tiwai, david.brown, robh+dt, linux-arm-kernel, spatakok,
linux-kernel, alsa-devel
The patch
ASoC: qdsp6: q6afe: Add q6afe driver
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From 7fa2d70f976657111a5ea4f3d16a738ddaa10c4f Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:55:56 +0100
Subject: [PATCH] ASoC: qdsp6: q6afe: Add q6afe driver
This patch adds support to Q6AFE (Audio Front End) module on Q6DSP.
AFE module sits right at the other end of cpu where the codec/audio
devices are connected.
AFE provides abstraced interfaces to both hardware and virtual devices.
Each AFE tx/rx port can be configured to connect to one of the hardware
devices like codec, hdmi, slimbus, i2s and so on. AFE services include
starting, stopping, and if needed, any configurations of the ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6afe.c | 549 ++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 35 +++
4 files changed, 589 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6afe.c
create mode 100644 sound/soc/qcom/qdsp6/q6afe.h
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 37ee0d958145..bb0a2afb0563 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -47,11 +47,15 @@ config SND_SOC_QDSP6_COMMON
config SND_SOC_QDSP6_CORE
tristate
+config SND_SOC_QDSP6_AFE
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_CORE
+ select SND_SOC_QDSP6_AFE
help
To add support for MSM QDSP6 Soc Audio.
This will enable sound soc platform specific
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 03b8e89c9731..7ff666bd10ca 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-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
+obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
new file mode 100644
index 000000000000..5e0032c13aab
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -0,0 +1,549 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/slab.h>
+#include <linux/kernel.h>
+#include <linux/uaccess.h>
+#include <linux/wait.h>
+#include <linux/jiffies.h>
+#include <linux/sched.h>
+#include <linux/module.h>
+#include <linux/kref.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/spinlock.h>
+#include <linux/delay.h>
+#include <linux/soc/qcom/apr.h>
+#include "q6dsp-errno.h"
+#include "q6core.h"
+#include "q6afe.h"
+
+/* AFE CMDs */
+#define AFE_PORT_CMD_DEVICE_START 0x000100E5
+#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6
+#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF
+#define AFE_SVC_CMD_SET_PARAM 0x000100f3
+#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106
+#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210
+#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C
+
+/* Port IDs */
+#define AFE_API_VERSION_HDMI_CONFIG 0x1
+#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
+#define TIMEOUT_MS 1000
+#define AFE_CMD_RESP_AVAIL 0
+#define AFE_CMD_RESP_NONE 1
+
+struct q6afe {
+ struct apr_device *apr;
+ struct device *dev;
+ struct q6core_svc_api_info ainfo;
+ struct mutex lock;
+ struct list_head port_list;
+ spinlock_t port_list_lock;
+ struct platform_device *pdev_dais;
+};
+
+struct afe_port_cmd_device_start {
+ u16 port_id;
+ u16 reserved;
+} __packed;
+
+struct afe_port_cmd_device_stop {
+ 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;
+ struct aprv2_ibasic_rsp_result_t result;
+ int token;
+ int id;
+ int cfg_type;
+ struct q6afe *afe;
+ struct kref refcount;
+ struct list_head node;
+};
+
+struct afe_port_map {
+ int port_id;
+ int token;
+ int is_rx;
+ int is_dig_pcm;
+};
+
+/*
+ * Mapping between Virtual Port IDs to DSP AFE Port ID
+ * On B Family SoCs DSP Port IDs are consistent across multiple SoCs
+ * on A Family SoCs DSP port IDs are same as virtual Port IDs.
+ */
+
+static struct afe_port_map port_maps[AFE_PORT_MAX] = {
+ [HDMI_RX] = { AFE_PORT_ID_MULTICHAN_HDMI_RX, HDMI_RX, 1, 1},
+};
+
+static void q6afe_port_free(struct kref *ref)
+{
+ struct q6afe_port *port;
+ struct q6afe *afe;
+ unsigned long flags;
+
+ port = container_of(ref, struct q6afe_port, refcount);
+ afe = port->afe;
+ spin_lock_irqsave(&afe->port_list_lock, flags);
+ list_del(&port->node);
+ spin_unlock_irqrestore(&afe->port_list_lock, flags);
+ kfree(port);
+}
+
+static struct q6afe_port *q6afe_find_port(struct q6afe *afe, int token)
+{
+ struct q6afe_port *p = NULL;
+ struct q6afe_port *ret = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&afe->port_list_lock, flags);
+ list_for_each_entry(p, &afe->port_list, node)
+ if (p->token == token) {
+ ret = p;
+ kref_get(&p->refcount);
+ break;
+ }
+
+ spin_unlock_irqrestore(&afe->port_list_lock, flags);
+ return ret;
+}
+
+static int q6afe_callback(struct apr_device *adev, struct apr_resp_pkt *data)
+{
+ struct q6afe *afe = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *res;
+ struct apr_hdr *hdr = &data->hdr;
+ struct q6afe_port *port;
+
+ if (!data->payload_size)
+ return 0;
+
+ res = data->payload;
+ switch (hdr->opcode) {
+ case APR_BASIC_RSP_RESULT: {
+ if (res->status) {
+ dev_err(afe->dev, "cmd = 0x%x returned error = 0x%x\n",
+ res->opcode, res->status);
+ }
+ switch (res->opcode) {
+ case AFE_PORT_CMD_SET_PARAM_V2:
+ case AFE_PORT_CMD_DEVICE_STOP:
+ case AFE_PORT_CMD_DEVICE_START:
+ case AFE_SVC_CMD_SET_PARAM:
+ port = q6afe_find_port(afe, hdr->token);
+ if (port) {
+ port->result = *res;
+ wake_up(&port->wait);
+ kref_put(&port->refcount, q6afe_port_free);
+ }
+ break;
+ default:
+ dev_err(afe->dev, "Unknown cmd 0x%x\n", res->opcode);
+ break;
+ }
+ }
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+
+/**
+ * q6afe_get_port_id() - Get port id from a given port index
+ *
+ * @index: port index
+ *
+ * Return: Will be an negative on error or valid port_id on success
+ */
+int q6afe_get_port_id(int index)
+{
+ if (index < 0 || index > AFE_PORT_MAX)
+ return -EINVAL;
+
+ return port_maps[index].port_id;
+}
+EXPORT_SYMBOL_GPL(q6afe_get_port_id);
+
+static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt,
+ struct q6afe_port *port)
+{
+ wait_queue_head_t *wait = &port->wait;
+ struct apr_hdr *hdr = &pkt->hdr;
+ int ret;
+
+ mutex_lock(&afe->lock);
+ port->result.opcode = 0;
+ port->result.status = 0;
+
+ ret = apr_send_pkt(afe->apr, pkt);
+ if (ret < 0) {
+ dev_err(afe->dev, "packet not transmitted (%d)\n", ret);
+ ret = -EINVAL;
+ goto err;
+ }
+
+ ret = wait_event_timeout(*wait, (port->result.opcode == hdr->opcode),
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ ret = -ETIMEDOUT;
+ } else if (port->result.status > 0) {
+ dev_err(afe->dev, "DSP returned error[%x]\n",
+ port->result.status);
+ ret = -EINVAL;
+ } else {
+ ret = 0;
+ }
+
+err:
+ mutex_unlock(&afe->lock);
+
+ return ret;
+}
+
+static int q6afe_port_set_param_v2(struct q6afe_port *port, void *data,
+ int param_id, int module_id, int psize)
+{
+ struct afe_port_cmd_set_param_v2 *param;
+ struct afe_port_param_data_v2 *pdata;
+ struct q6afe *afe = port->afe;
+ struct apr_pkt *pkt;
+ u16 port_id = port->id;
+ int ret, pkt_size;
+ void *p, *pl;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*param) + sizeof(*pdata) + psize;
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ param = p + APR_HDR_SIZE;
+ pdata = p + APR_HDR_SIZE + sizeof(*param);
+ pl = p + APR_HDR_SIZE + sizeof(*param) + sizeof(*pdata);
+ memcpy(pl, data, psize);
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.token = port->token;
+ pkt->hdr.opcode = AFE_PORT_CMD_SET_PARAM_V2;
+
+ param->port_id = port_id;
+ param->payload_size = sizeof(*pdata) + psize;
+ param->payload_address_lsw = 0x00;
+ param->payload_address_msw = 0x00;
+ param->mem_map_handle = 0x00;
+ pdata->module_id = module_id;
+ pdata->param_id = param_id;
+ pdata->param_size = psize;
+
+ ret = afe_apr_send_pkt(afe, pkt, port);
+ if (ret)
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+
+ kfree(pkt);
+ 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)
+{
+ struct afe_port_cmd_device_stop *stop;
+ struct q6afe *afe = port->afe;
+ struct apr_pkt *pkt;
+ int port_id = port->id;
+ int ret = 0;
+ int index, pkt_size;
+ void *p;
+
+ 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;
+ }
+
+ pkt_size = APR_HDR_SIZE + sizeof(*stop);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ stop = p + APR_HDR_SIZE;
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.token = index;
+ pkt->hdr.opcode = AFE_PORT_CMD_DEVICE_STOP;
+ stop->port_id = port_id;
+ stop->reserved = 0;
+
+ ret = afe_apr_send_pkt(afe, pkt, port);
+ if (ret)
+ dev_err(afe->dev, "AFE close failed %d\n", ret);
+
+ kfree(pkt);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6afe_port_stop);
+
+/**
+ * q6afe_hdmi_port_prepare() - Prepare hdmi afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: HDMI configuration for the afe port
+ *
+ */
+void q6afe_hdmi_port_prepare(struct q6afe_port *port,
+ struct q6afe_hdmi_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+
+ pcfg->hdmi_multi_ch.hdmi_cfg_minor_version =
+ AFE_API_VERSION_HDMI_CONFIG;
+ pcfg->hdmi_multi_ch.datatype = cfg->datatype;
+ pcfg->hdmi_multi_ch.channel_allocation = cfg->channel_allocation;
+ pcfg->hdmi_multi_ch.sample_rate = cfg->sample_rate;
+ pcfg->hdmi_multi_ch.bit_width = cfg->bit_width;
+}
+EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
+
+/**
+ * q6afe_port_start() - Start a afe port
+ *
+ * @port: Instance of port to start
+ *
+ * Return: Will be an negative on packet size on success.
+ */
+int q6afe_port_start(struct q6afe_port *port)
+{
+ struct afe_port_cmd_device_start *start;
+ struct q6afe *afe = port->afe;
+ int port_id = port->id;
+ int ret, param_id = port->cfg_type;
+ struct apr_pkt *pkt;
+ int pkt_size;
+ void *p;
+
+ ret = q6afe_port_set_param_v2(port, &port->port_cfg, param_id,
+ AFE_MODULE_AUDIO_DEV_INTERFACE,
+ sizeof(port->port_cfg));
+ if (ret) {
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+ return ret;
+ }
+
+ pkt_size = APR_HDR_SIZE + sizeof(*start);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ start = p + APR_HDR_SIZE;
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.token = port->token;
+ pkt->hdr.opcode = AFE_PORT_CMD_DEVICE_START;
+
+ start->port_id = port_id;
+
+ ret = afe_apr_send_pkt(afe, pkt, port);
+ if (ret)
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+
+ kfree(pkt);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6afe_port_start);
+
+/**
+ * q6afe_port_get_from_id() - Get port instance from a port id
+ *
+ * @dev: Pointer to afe child device.
+ * @id: port id
+ *
+ * Return: Will be an error pointer on error or a valid afe port
+ * on success.
+ */
+struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
+{
+ int port_id;
+ struct q6afe *afe = dev_get_drvdata(dev->parent);
+ struct q6afe_port *port;
+ unsigned long flags;
+ int cfg_type;
+
+ if (id < 0 || id > AFE_PORT_MAX) {
+ dev_err(dev, "AFE port token[%d] invalid!\n", id);
+ return ERR_PTR(-EINVAL);
+ }
+
+ /* if port is multiple times bind/unbind before callback finishes */
+ port = q6afe_find_port(afe, id);
+ if (port) {
+ dev_err(dev, "AFE Port already open\n");
+ return port;
+ }
+
+ port_id = port_maps[id].port_id;
+
+ switch (port_id) {
+ case AFE_PORT_ID_MULTICHAN_HDMI_RX:
+ cfg_type = AFE_PARAM_ID_HDMI_CONFIG;
+ break;
+ default:
+ dev_err(dev, "Invalid port id 0x%x\n", port_id);
+ return ERR_PTR(-EINVAL);
+ }
+
+ port = kzalloc(sizeof(*port), GFP_KERNEL);
+ if (!port)
+ return ERR_PTR(-ENOMEM);
+
+ init_waitqueue_head(&port->wait);
+
+ port->token = id;
+ port->id = port_id;
+ port->afe = afe;
+ port->cfg_type = cfg_type;
+ kref_init(&port->refcount);
+
+ spin_lock_irqsave(&afe->port_list_lock, flags);
+ list_add_tail(&port->node, &afe->port_list);
+ spin_unlock_irqrestore(&afe->port_list_lock, flags);
+
+ return port;
+
+}
+EXPORT_SYMBOL_GPL(q6afe_port_get_from_id);
+
+/**
+ * q6afe_port_put() - Release port reference
+ *
+ * @port: Instance of port to put
+ */
+void q6afe_port_put(struct q6afe_port *port)
+{
+ kref_put(&port->refcount, q6afe_port_free);
+}
+EXPORT_SYMBOL_GPL(q6afe_port_put);
+
+static int q6afe_probe(struct apr_device *adev)
+{
+ struct q6afe *afe;
+ struct device *dev = &adev->dev;
+ struct device_node *dais_np;
+
+ afe = devm_kzalloc(dev, sizeof(*afe), GFP_KERNEL);
+ if (!afe)
+ return -ENOMEM;
+
+ q6core_get_svc_api_info(adev->svc_id, &afe->ainfo);
+ afe->apr = adev;
+ mutex_init(&afe->lock);
+ afe->dev = dev;
+ INIT_LIST_HEAD(&afe->port_list);
+ spin_lock_init(&afe->port_list_lock);
+
+ dev_set_drvdata(dev, afe);
+
+ dais_np = of_get_child_by_name(dev->of_node, "dais");
+ if (dais_np) {
+ afe->pdev_dais = of_platform_device_create(dais_np,
+ "q6afe-dai", dev);
+ of_node_put(dais_np);
+ }
+
+ return 0;
+}
+
+static int q6afe_remove(struct apr_device *adev)
+{
+ struct q6afe *afe = dev_get_drvdata(&adev->dev);
+
+ if (afe->pdev_dais)
+ of_platform_device_destroy(&afe->pdev_dais->dev, NULL);
+
+ return 0;
+}
+
+static const struct of_device_id q6afe_device_id[] = {
+ { .compatible = "qcom,q6afe" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6afe_device_id);
+
+static struct apr_driver qcom_q6afe_driver = {
+ .probe = q6afe_probe,
+ .remove = q6afe_remove,
+ .callback = q6afe_callback,
+ .driver = {
+ .name = "qcom-q6afe",
+ .of_match_table = of_match_ptr(q6afe_device_id),
+
+ },
+};
+
+module_apr_driver(qcom_q6afe_driver);
+MODULE_DESCRIPTION("Q6 Audio Front End");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
new file mode 100644
index 000000000000..3bd991a7c42d
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -0,0 +1,35 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+
+#ifndef __Q6AFE_H__
+#define __Q6AFE_H__
+
+#include <dt-bindings/sound/qcom,q6afe.h>
+
+#define AFE_PORT_MAX 48
+
+#define MSM_AFE_PORT_TYPE_RX 0
+#define MSM_AFE_PORT_TYPE_TX 1
+#define AFE_MAX_PORTS AFE_PORT_MAX
+
+struct q6afe_hdmi_cfg {
+ u16 datatype;
+ u16 channel_allocation;
+ u32 sample_rate;
+ u16 bit_width;
+};
+
+struct q6afe_port_config {
+ struct q6afe_hdmi_cfg hdmi;
+};
+
+struct q6afe_port;
+
+struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id);
+int q6afe_port_start(struct q6afe_port *port);
+int q6afe_port_stop(struct q6afe_port *port);
+void q6afe_port_put(struct q6afe_port *port);
+int q6afe_get_port_id(int index);
+void q6afe_hdmi_port_prepare(struct q6afe_port *port,
+ struct q6afe_hdmi_cfg *cfg);
+
+#endif /* __Q6AFE_H__ */
--
2.17.0
^ permalink raw reply related [flat|nested] 25+ messages in thread
* [PATCH v9 02/15] ASoC: qdsp6: qdafe: Add SLIMBus port Support
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
2018-05-18 12:55 ` [PATCH v9 01/15] ASoC: qdsp6: q6afe: Add q6afe driver Srinivas Kandagatla
@ 2018-05-18 12:55 ` Srinivas Kandagatla
2018-05-21 15:49 ` Applied "ASoC: qdsp6: qdafe: Add SLIMBus port Support" to the asoc tree Mark Brown
2018-05-18 12:55 ` [PATCH v9 03/15] ASoC: qdsp6: q6afe: Add support to MI2S ports Srinivas Kandagatla
` (12 subsequent siblings)
14 siblings, 1 reply; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:55 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
This patch adds support to 6 SLIMBus AFE ports, which are used as
backend dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/qdsp6/q6afe.c | 129 +++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 14 +++++
2 files changed, 143 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 5e0032c13aab..735c0b5eb55a 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -28,9 +28,45 @@
#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
@@ -80,8 +116,53 @@ 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 {
@@ -111,6 +192,20 @@ struct afe_port_map {
static struct afe_port_map port_maps[AFE_PORT_MAX] = {
[HDMI_RX] = { AFE_PORT_ID_MULTICHAN_HDMI_RX, HDMI_RX, 1, 1},
+ [SLIMBUS_0_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX,
+ SLIMBUS_0_RX, 1, 1},
+ [SLIMBUS_1_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX,
+ SLIMBUS_1_RX, 1, 1},
+ [SLIMBUS_2_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX,
+ SLIMBUS_2_RX, 1, 1},
+ [SLIMBUS_3_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX,
+ SLIMBUS_3_RX, 1, 1},
+ [SLIMBUS_4_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX,
+ SLIMBUS_4_RX, 1, 1},
+ [SLIMBUS_5_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX,
+ SLIMBUS_5_RX, 1, 1},
+ [SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
+ SLIMBUS_6_RX, 1, 1},
};
static void q6afe_port_free(struct kref *ref)
@@ -340,6 +435,31 @@ int q6afe_port_stop(struct q6afe_port *port)
}
EXPORT_SYMBOL_GPL(q6afe_port_stop);
+/**
+ * q6afe_slim_port_prepare() - Prepare slim afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: SLIM configuration for the afe port
+ *
+ */
+void q6afe_slim_port_prepare(struct q6afe_port *port,
+ struct q6afe_slim_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+
+ pcfg->slim_cfg.sb_cfg_minor_version = AFE_API_VERSION_SLIMBUS_CONFIG;
+ pcfg->slim_cfg.sample_rate = cfg->sample_rate;
+ pcfg->slim_cfg.bit_width = cfg->bit_width;
+ pcfg->slim_cfg.num_channels = cfg->num_channels;
+ pcfg->slim_cfg.data_format = cfg->data_format;
+ pcfg->slim_cfg.shared_ch_mapping[0] = cfg->ch_mapping[0];
+ pcfg->slim_cfg.shared_ch_mapping[1] = cfg->ch_mapping[1];
+ pcfg->slim_cfg.shared_ch_mapping[2] = cfg->ch_mapping[2];
+ pcfg->slim_cfg.shared_ch_mapping[3] = cfg->ch_mapping[3];
+
+}
+EXPORT_SYMBOL_GPL(q6afe_slim_port_prepare);
+
/**
* q6afe_hdmi_port_prepare() - Prepare hdmi afe port.
*
@@ -451,6 +571,15 @@ 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 3bd991a7c42d..5659966c6b1e 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -11,6 +11,9 @@
#define MSM_AFE_PORT_TYPE_TX 1
#define AFE_MAX_PORTS AFE_PORT_MAX
+#define AFE_MAX_CHAN_COUNT 8
+#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
+
struct q6afe_hdmi_cfg {
u16 datatype;
u16 channel_allocation;
@@ -18,8 +21,17 @@ struct q6afe_hdmi_cfg {
u16 bit_width;
};
+struct q6afe_slim_cfg {
+ u32 sample_rate;
+ u16 bit_width;
+ u16 data_format;
+ u16 num_channels;
+ u8 ch_mapping[AFE_MAX_CHAN_COUNT];
+};
+
struct q6afe_port_config {
struct q6afe_hdmi_cfg hdmi;
+ struct q6afe_slim_cfg slim;
};
struct q6afe_port;
@@ -31,5 +43,7 @@ void q6afe_port_put(struct q6afe_port *port);
int q6afe_get_port_id(int index);
void q6afe_hdmi_port_prepare(struct q6afe_port *port,
struct q6afe_hdmi_cfg *cfg);
+void q6afe_slim_port_prepare(struct q6afe_port *port,
+ struct q6afe_slim_cfg *cfg);
#endif /* __Q6AFE_H__ */
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* Applied "ASoC: qdsp6: qdafe: Add SLIMBus port Support" to the asoc tree
2018-05-18 12:55 ` [PATCH v9 02/15] ASoC: qdsp6: qdafe: Add SLIMBus port Support Srinivas Kandagatla
@ 2018-05-21 15:49 ` Mark Brown
0 siblings, 0 replies; 25+ messages in thread
From: Mark Brown @ 2018-05-21 15:49 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, broonie, linux-arm-msm, alsa-devel, bgoswami,
mark.rutland, devicetree, rohkumar, gregkh, plai, lgirdwood,
tiwai, david.brown, robh+dt, linux-arm-kernel, spatakok,
linux-kernel, alsa-devel
The patch
ASoC: qdsp6: qdafe: Add SLIMBus port Support
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From 4d430d5a0e9d4d065e0447366caf48d5f580d7fb Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:55:57 +0100
Subject: [PATCH] ASoC: qdsp6: qdafe: Add SLIMBus port Support
This patch adds support to 6 SLIMBus AFE ports, which are used as
backend dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6afe.c | 129 +++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 14 ++++
2 files changed, 143 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 5e0032c13aab..735c0b5eb55a 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -28,9 +28,45 @@
#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
@@ -80,8 +116,53 @@ 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 {
@@ -111,6 +192,20 @@ struct afe_port_map {
static struct afe_port_map port_maps[AFE_PORT_MAX] = {
[HDMI_RX] = { AFE_PORT_ID_MULTICHAN_HDMI_RX, HDMI_RX, 1, 1},
+ [SLIMBUS_0_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX,
+ SLIMBUS_0_RX, 1, 1},
+ [SLIMBUS_1_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX,
+ SLIMBUS_1_RX, 1, 1},
+ [SLIMBUS_2_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX,
+ SLIMBUS_2_RX, 1, 1},
+ [SLIMBUS_3_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX,
+ SLIMBUS_3_RX, 1, 1},
+ [SLIMBUS_4_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX,
+ SLIMBUS_4_RX, 1, 1},
+ [SLIMBUS_5_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX,
+ SLIMBUS_5_RX, 1, 1},
+ [SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
+ SLIMBUS_6_RX, 1, 1},
};
static void q6afe_port_free(struct kref *ref)
@@ -340,6 +435,31 @@ int q6afe_port_stop(struct q6afe_port *port)
}
EXPORT_SYMBOL_GPL(q6afe_port_stop);
+/**
+ * q6afe_slim_port_prepare() - Prepare slim afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: SLIM configuration for the afe port
+ *
+ */
+void q6afe_slim_port_prepare(struct q6afe_port *port,
+ struct q6afe_slim_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+
+ pcfg->slim_cfg.sb_cfg_minor_version = AFE_API_VERSION_SLIMBUS_CONFIG;
+ pcfg->slim_cfg.sample_rate = cfg->sample_rate;
+ pcfg->slim_cfg.bit_width = cfg->bit_width;
+ pcfg->slim_cfg.num_channels = cfg->num_channels;
+ pcfg->slim_cfg.data_format = cfg->data_format;
+ pcfg->slim_cfg.shared_ch_mapping[0] = cfg->ch_mapping[0];
+ pcfg->slim_cfg.shared_ch_mapping[1] = cfg->ch_mapping[1];
+ pcfg->slim_cfg.shared_ch_mapping[2] = cfg->ch_mapping[2];
+ pcfg->slim_cfg.shared_ch_mapping[3] = cfg->ch_mapping[3];
+
+}
+EXPORT_SYMBOL_GPL(q6afe_slim_port_prepare);
+
/**
* q6afe_hdmi_port_prepare() - Prepare hdmi afe port.
*
@@ -451,6 +571,15 @@ 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 3bd991a7c42d..5659966c6b1e 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -11,6 +11,9 @@
#define MSM_AFE_PORT_TYPE_TX 1
#define AFE_MAX_PORTS AFE_PORT_MAX
+#define AFE_MAX_CHAN_COUNT 8
+#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
+
struct q6afe_hdmi_cfg {
u16 datatype;
u16 channel_allocation;
@@ -18,8 +21,17 @@ struct q6afe_hdmi_cfg {
u16 bit_width;
};
+struct q6afe_slim_cfg {
+ u32 sample_rate;
+ u16 bit_width;
+ u16 data_format;
+ u16 num_channels;
+ u8 ch_mapping[AFE_MAX_CHAN_COUNT];
+};
+
struct q6afe_port_config {
struct q6afe_hdmi_cfg hdmi;
+ struct q6afe_slim_cfg slim;
};
struct q6afe_port;
@@ -31,5 +43,7 @@ void q6afe_port_put(struct q6afe_port *port);
int q6afe_get_port_id(int index);
void q6afe_hdmi_port_prepare(struct q6afe_port *port,
struct q6afe_hdmi_cfg *cfg);
+void q6afe_slim_port_prepare(struct q6afe_port *port,
+ struct q6afe_slim_cfg *cfg);
#endif /* __Q6AFE_H__ */
--
2.17.0
^ permalink raw reply related [flat|nested] 25+ messages in thread
* [PATCH v9 03/15] ASoC: qdsp6: q6afe: Add support to MI2S ports
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
2018-05-18 12:55 ` [PATCH v9 01/15] ASoC: qdsp6: q6afe: Add q6afe driver Srinivas Kandagatla
2018-05-18 12:55 ` [PATCH v9 02/15] ASoC: qdsp6: qdafe: Add SLIMBus port Support Srinivas Kandagatla
@ 2018-05-18 12:55 ` Srinivas Kandagatla
2018-05-21 15:49 ` Applied "ASoC: qdsp6: q6afe: Add support to MI2S ports" to the asoc tree Mark Brown
2018-05-18 12:55 ` [PATCH v9 04/15] ASoC: qdsp6: q6afe: Add support to MI2S sysclks Srinivas Kandagatla
` (11 subsequent siblings)
14 siblings, 1 reply; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:55 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
This patch adds support to 4 MI2S ports on LPASS.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/qdsp6/q6afe.c | 224 +++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 13 +++
2 files changed, 237 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 735c0b5eb55a..0004369b3661 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -15,6 +15,10 @@
#include <linux/spinlock.h>
#include <linux/delay.h>
#include <linux/soc/qcom/apr.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
#include "q6dsp-errno.h"
#include "q6core.h"
#include "q6afe.h"
@@ -31,6 +35,32 @@
#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_SD0_MASK BIT(0x1)
+#define AFE_PORT_I2S_SD1_MASK BIT(0x2)
+#define AFE_PORT_I2S_SD2_MASK BIT(0x3)
+#define AFE_PORT_I2S_SD3_MASK BIT(0x4)
+#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1)
+#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3)
+#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1)
+#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1)
+#define AFE_PORT_I2S_QUAD01 0x5
+#define AFE_PORT_I2S_QUAD23 0x6
+#define AFE_PORT_I2S_6CHS 0x7
+#define AFE_PORT_I2S_8CHS 0x8
+#define AFE_PORT_I2S_MONO 0x0
+#define AFE_PORT_I2S_STEREO 0x1
+#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0
+#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1
+#define AFE_LINEAR_PCM_DATA 0x0
+
/* Port IDs */
#define AFE_API_VERSION_HDMI_CONFIG 0x1
@@ -66,6 +96,19 @@
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX 0x400c
/* SLIMbus Tx port on channel 6. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX 0x400d
+#define AFE_PORT_ID_PRIMARY_MI2S_RX 0x1000
+#define AFE_PORT_ID_PRIMARY_MI2S_TX 0x1001
+#define AFE_PORT_ID_SECONDARY_MI2S_RX 0x1002
+#define AFE_PORT_ID_SECONDARY_MI2S_TX 0x1003
+#define AFE_PORT_ID_TERTIARY_MI2S_RX 0x1004
+#define AFE_PORT_ID_TERTIARY_MI2S_TX 0x1005
+#define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006
+#define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007
+
+#define Q6AFE_LPASS_MODE_CLK1_VALID 1
+#define Q6AFE_LPASS_MODE_CLK2_VALID 2
+#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
+#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
#define TIMEOUT_MS 1000
#define AFE_CMD_RESP_AVAIL 0
@@ -159,10 +202,21 @@ 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 {
@@ -206,6 +260,22 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
SLIMBUS_5_RX, 1, 1},
[SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
SLIMBUS_6_RX, 1, 1},
+ [PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX,
+ PRIMARY_MI2S_RX, 1, 1},
+ [PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX,
+ PRIMARY_MI2S_RX, 0, 1},
+ [SECONDARY_MI2S_RX] = { AFE_PORT_ID_SECONDARY_MI2S_RX,
+ SECONDARY_MI2S_RX, 1, 1},
+ [SECONDARY_MI2S_TX] = { AFE_PORT_ID_SECONDARY_MI2S_TX,
+ SECONDARY_MI2S_TX, 0, 1},
+ [TERTIARY_MI2S_RX] = { AFE_PORT_ID_TERTIARY_MI2S_RX,
+ TERTIARY_MI2S_RX, 1, 1},
+ [TERTIARY_MI2S_TX] = { AFE_PORT_ID_TERTIARY_MI2S_TX,
+ TERTIARY_MI2S_TX, 0, 1},
+ [QUATERNARY_MI2S_RX] = { AFE_PORT_ID_QUATERNARY_MI2S_RX,
+ QUATERNARY_MI2S_RX, 1, 1},
+ [QUATERNARY_MI2S_TX] = { AFE_PORT_ID_QUATERNARY_MI2S_TX,
+ QUATERNARY_MI2S_TX, 0, 1},
};
static void q6afe_port_free(struct kref *ref)
@@ -481,6 +551,149 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
}
EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
+/**
+ * q6afe_i2s_port_prepare() - Prepare i2s afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: I2S configuration for the afe port
+ * Return: Will be an negative on error and zero on success.
+ */
+int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+ struct device *dev = port->afe->dev;
+ int num_sd_lines;
+
+ pcfg->i2s_cfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ pcfg->i2s_cfg.sample_rate = cfg->sample_rate;
+ pcfg->i2s_cfg.bit_width = cfg->bit_width;
+ pcfg->i2s_cfg.data_format = AFE_LINEAR_PCM_DATA;
+
+ switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* CPU is slave */
+ pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL;
+ break;
+ default:
+ break;
+ }
+
+ num_sd_lines = hweight_long(cfg->sd_line_mask);
+
+ switch (num_sd_lines) {
+ case 0:
+ dev_err(dev, "no line is assigned\n");
+ return -EINVAL;
+ case 1:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD0;
+ break;
+ case AFE_PORT_I2S_SD1_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD1;
+ break;
+ case AFE_PORT_I2S_SD2_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD2;
+ break;
+ case AFE_PORT_I2S_SD3_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD3;
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ case 2:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_1_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_QUAD01;
+ break;
+ case AFE_PORT_I2S_SD2_3_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_QUAD23;
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ case 3:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_1_2_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_6CHS;
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ case 4:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_1_2_3_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_8CHS;
+
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+
+ switch (cfg->num_channels) {
+ case 1:
+ case 2:
+ switch (pcfg->i2s_cfg.channel_mode) {
+ case AFE_PORT_I2S_QUAD01:
+ case AFE_PORT_I2S_6CHS:
+ case AFE_PORT_I2S_8CHS:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD0;
+ break;
+ case AFE_PORT_I2S_QUAD23:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD2;
+ break;
+ }
+
+ if (cfg->num_channels == 2)
+ pcfg->i2s_cfg.mono_stereo = AFE_PORT_I2S_STEREO;
+ else
+ pcfg->i2s_cfg.mono_stereo = AFE_PORT_I2S_MONO;
+
+ break;
+ case 3:
+ case 4:
+ if (pcfg->i2s_cfg.channel_mode < AFE_PORT_I2S_QUAD01) {
+ dev_err(dev, "Invalid Channel mode\n");
+ return -EINVAL;
+ }
+ break;
+ case 5:
+ case 6:
+ if (pcfg->i2s_cfg.channel_mode < AFE_PORT_I2S_6CHS) {
+ dev_err(dev, "Invalid Channel mode\n");
+ return -EINVAL;
+ }
+ break;
+ case 7:
+ case 8:
+ if (pcfg->i2s_cfg.channel_mode < AFE_PORT_I2S_8CHS) {
+ dev_err(dev, "Invalid Channel mode\n");
+ return -EINVAL;
+ }
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6afe_i2s_port_prepare);
+
/**
* q6afe_port_start() - Start a afe port
*
@@ -580,6 +793,17 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX:
cfg_type = AFE_PARAM_ID_SLIMBUS_CONFIG;
break;
+
+ case AFE_PORT_ID_PRIMARY_MI2S_RX:
+ case AFE_PORT_ID_PRIMARY_MI2S_TX:
+ case AFE_PORT_ID_SECONDARY_MI2S_RX:
+ case AFE_PORT_ID_SECONDARY_MI2S_TX:
+ case AFE_PORT_ID_TERTIARY_MI2S_RX:
+ case AFE_PORT_ID_TERTIARY_MI2S_TX:
+ case AFE_PORT_ID_QUATERNARY_MI2S_RX:
+ case AFE_PORT_ID_QUATERNARY_MI2S_TX:
+ cfg_type = AFE_PARAM_ID_I2S_CONFIG;
+ break;
default:
dev_err(dev, "Invalid port id 0x%x\n", port_id);
return ERR_PTR(-EINVAL);
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index 5659966c6b1e..3cb3bb4985a9 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -11,6 +11,8 @@
#define MSM_AFE_PORT_TYPE_TX 1
#define AFE_MAX_PORTS AFE_PORT_MAX
+#define Q6AFE_MAX_MI2S_LINES 4
+
#define AFE_MAX_CHAN_COUNT 8
#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
@@ -29,9 +31,19 @@ struct q6afe_slim_cfg {
u8 ch_mapping[AFE_MAX_CHAN_COUNT];
};
+struct q6afe_i2s_cfg {
+ u32 sample_rate;
+ u16 bit_width;
+ u16 data_format;
+ u16 num_channels;
+ u32 sd_line_mask;
+ int fmt;
+};
+
struct q6afe_port_config {
struct q6afe_hdmi_cfg hdmi;
struct q6afe_slim_cfg slim;
+ struct q6afe_i2s_cfg i2s_cfg;
};
struct q6afe_port;
@@ -45,5 +57,6 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
struct q6afe_hdmi_cfg *cfg);
void q6afe_slim_port_prepare(struct q6afe_port *port,
struct q6afe_slim_cfg *cfg);
+int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg);
#endif /* __Q6AFE_H__ */
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* Applied "ASoC: qdsp6: q6afe: Add support to MI2S ports" to the asoc tree
2018-05-18 12:55 ` [PATCH v9 03/15] ASoC: qdsp6: q6afe: Add support to MI2S ports Srinivas Kandagatla
@ 2018-05-21 15:49 ` Mark Brown
0 siblings, 0 replies; 25+ messages in thread
From: Mark Brown @ 2018-05-21 15:49 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, broonie, linux-arm-msm, alsa-devel, bgoswami,
mark.rutland, devicetree, rohkumar, gregkh, plai, lgirdwood,
tiwai, david.brown, robh+dt, linux-arm-kernel, spatakok,
linux-kernel, alsa-devel
The patch
ASoC: qdsp6: q6afe: Add support to MI2S ports
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From d3839145722bae49e3a02d65f64ce49c75d7c8e1 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:55:58 +0100
Subject: [PATCH] ASoC: qdsp6: q6afe: Add support to MI2S ports
This patch adds support to 4 MI2S ports on LPASS.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6afe.c | 224 +++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 13 ++
2 files changed, 237 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 735c0b5eb55a..0004369b3661 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -15,6 +15,10 @@
#include <linux/spinlock.h>
#include <linux/delay.h>
#include <linux/soc/qcom/apr.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
#include "q6dsp-errno.h"
#include "q6core.h"
#include "q6afe.h"
@@ -31,6 +35,32 @@
#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_SD0_MASK BIT(0x1)
+#define AFE_PORT_I2S_SD1_MASK BIT(0x2)
+#define AFE_PORT_I2S_SD2_MASK BIT(0x3)
+#define AFE_PORT_I2S_SD3_MASK BIT(0x4)
+#define AFE_PORT_I2S_SD0_1_MASK GENMASK(2, 1)
+#define AFE_PORT_I2S_SD2_3_MASK GENMASK(4, 3)
+#define AFE_PORT_I2S_SD0_1_2_MASK GENMASK(3, 1)
+#define AFE_PORT_I2S_SD0_1_2_3_MASK GENMASK(4, 1)
+#define AFE_PORT_I2S_QUAD01 0x5
+#define AFE_PORT_I2S_QUAD23 0x6
+#define AFE_PORT_I2S_6CHS 0x7
+#define AFE_PORT_I2S_8CHS 0x8
+#define AFE_PORT_I2S_MONO 0x0
+#define AFE_PORT_I2S_STEREO 0x1
+#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0
+#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1
+#define AFE_LINEAR_PCM_DATA 0x0
+
/* Port IDs */
#define AFE_API_VERSION_HDMI_CONFIG 0x1
@@ -66,6 +96,19 @@
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX 0x400c
/* SLIMbus Tx port on channel 6. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_TX 0x400d
+#define AFE_PORT_ID_PRIMARY_MI2S_RX 0x1000
+#define AFE_PORT_ID_PRIMARY_MI2S_TX 0x1001
+#define AFE_PORT_ID_SECONDARY_MI2S_RX 0x1002
+#define AFE_PORT_ID_SECONDARY_MI2S_TX 0x1003
+#define AFE_PORT_ID_TERTIARY_MI2S_RX 0x1004
+#define AFE_PORT_ID_TERTIARY_MI2S_TX 0x1005
+#define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006
+#define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007
+
+#define Q6AFE_LPASS_MODE_CLK1_VALID 1
+#define Q6AFE_LPASS_MODE_CLK2_VALID 2
+#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
+#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
#define TIMEOUT_MS 1000
#define AFE_CMD_RESP_AVAIL 0
@@ -159,10 +202,21 @@ 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 {
@@ -206,6 +260,22 @@ static struct afe_port_map port_maps[AFE_PORT_MAX] = {
SLIMBUS_5_RX, 1, 1},
[SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX,
SLIMBUS_6_RX, 1, 1},
+ [PRIMARY_MI2S_RX] = { AFE_PORT_ID_PRIMARY_MI2S_RX,
+ PRIMARY_MI2S_RX, 1, 1},
+ [PRIMARY_MI2S_TX] = { AFE_PORT_ID_PRIMARY_MI2S_TX,
+ PRIMARY_MI2S_RX, 0, 1},
+ [SECONDARY_MI2S_RX] = { AFE_PORT_ID_SECONDARY_MI2S_RX,
+ SECONDARY_MI2S_RX, 1, 1},
+ [SECONDARY_MI2S_TX] = { AFE_PORT_ID_SECONDARY_MI2S_TX,
+ SECONDARY_MI2S_TX, 0, 1},
+ [TERTIARY_MI2S_RX] = { AFE_PORT_ID_TERTIARY_MI2S_RX,
+ TERTIARY_MI2S_RX, 1, 1},
+ [TERTIARY_MI2S_TX] = { AFE_PORT_ID_TERTIARY_MI2S_TX,
+ TERTIARY_MI2S_TX, 0, 1},
+ [QUATERNARY_MI2S_RX] = { AFE_PORT_ID_QUATERNARY_MI2S_RX,
+ QUATERNARY_MI2S_RX, 1, 1},
+ [QUATERNARY_MI2S_TX] = { AFE_PORT_ID_QUATERNARY_MI2S_TX,
+ QUATERNARY_MI2S_TX, 0, 1},
};
static void q6afe_port_free(struct kref *ref)
@@ -481,6 +551,149 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
}
EXPORT_SYMBOL_GPL(q6afe_hdmi_port_prepare);
+/**
+ * q6afe_i2s_port_prepare() - Prepare i2s afe port.
+ *
+ * @port: Instance of afe port
+ * @cfg: I2S configuration for the afe port
+ * Return: Will be an negative on error and zero on success.
+ */
+int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg)
+{
+ union afe_port_config *pcfg = &port->port_cfg;
+ struct device *dev = port->afe->dev;
+ int num_sd_lines;
+
+ pcfg->i2s_cfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ pcfg->i2s_cfg.sample_rate = cfg->sample_rate;
+ pcfg->i2s_cfg.bit_width = cfg->bit_width;
+ pcfg->i2s_cfg.data_format = AFE_LINEAR_PCM_DATA;
+
+ switch (cfg->fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFS:
+ pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL;
+ break;
+ case SND_SOC_DAIFMT_CBM_CFM:
+ /* CPU is slave */
+ pcfg->i2s_cfg.ws_src = AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL;
+ break;
+ default:
+ break;
+ }
+
+ num_sd_lines = hweight_long(cfg->sd_line_mask);
+
+ switch (num_sd_lines) {
+ case 0:
+ dev_err(dev, "no line is assigned\n");
+ return -EINVAL;
+ case 1:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD0;
+ break;
+ case AFE_PORT_I2S_SD1_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD1;
+ break;
+ case AFE_PORT_I2S_SD2_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD2;
+ break;
+ case AFE_PORT_I2S_SD3_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD3;
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ case 2:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_1_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_QUAD01;
+ break;
+ case AFE_PORT_I2S_SD2_3_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_QUAD23;
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ case 3:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_1_2_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_6CHS;
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ case 4:
+ switch (cfg->sd_line_mask) {
+ case AFE_PORT_I2S_SD0_1_2_3_MASK:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_8CHS;
+
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+ break;
+ default:
+ dev_err(dev, "Invalid SD lines\n");
+ return -EINVAL;
+ }
+
+ switch (cfg->num_channels) {
+ case 1:
+ case 2:
+ switch (pcfg->i2s_cfg.channel_mode) {
+ case AFE_PORT_I2S_QUAD01:
+ case AFE_PORT_I2S_6CHS:
+ case AFE_PORT_I2S_8CHS:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD0;
+ break;
+ case AFE_PORT_I2S_QUAD23:
+ pcfg->i2s_cfg.channel_mode = AFE_PORT_I2S_SD2;
+ break;
+ }
+
+ if (cfg->num_channels == 2)
+ pcfg->i2s_cfg.mono_stereo = AFE_PORT_I2S_STEREO;
+ else
+ pcfg->i2s_cfg.mono_stereo = AFE_PORT_I2S_MONO;
+
+ break;
+ case 3:
+ case 4:
+ if (pcfg->i2s_cfg.channel_mode < AFE_PORT_I2S_QUAD01) {
+ dev_err(dev, "Invalid Channel mode\n");
+ return -EINVAL;
+ }
+ break;
+ case 5:
+ case 6:
+ if (pcfg->i2s_cfg.channel_mode < AFE_PORT_I2S_6CHS) {
+ dev_err(dev, "Invalid Channel mode\n");
+ return -EINVAL;
+ }
+ break;
+ case 7:
+ case 8:
+ if (pcfg->i2s_cfg.channel_mode < AFE_PORT_I2S_8CHS) {
+ dev_err(dev, "Invalid Channel mode\n");
+ return -EINVAL;
+ }
+ break;
+ default:
+ break;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6afe_i2s_port_prepare);
+
/**
* q6afe_port_start() - Start a afe port
*
@@ -580,6 +793,17 @@ struct q6afe_port *q6afe_port_get_from_id(struct device *dev, int id)
case AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX:
cfg_type = AFE_PARAM_ID_SLIMBUS_CONFIG;
break;
+
+ case AFE_PORT_ID_PRIMARY_MI2S_RX:
+ case AFE_PORT_ID_PRIMARY_MI2S_TX:
+ case AFE_PORT_ID_SECONDARY_MI2S_RX:
+ case AFE_PORT_ID_SECONDARY_MI2S_TX:
+ case AFE_PORT_ID_TERTIARY_MI2S_RX:
+ case AFE_PORT_ID_TERTIARY_MI2S_TX:
+ case AFE_PORT_ID_QUATERNARY_MI2S_RX:
+ case AFE_PORT_ID_QUATERNARY_MI2S_TX:
+ cfg_type = AFE_PARAM_ID_I2S_CONFIG;
+ break;
default:
dev_err(dev, "Invalid port id 0x%x\n", port_id);
return ERR_PTR(-EINVAL);
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index 5659966c6b1e..3cb3bb4985a9 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -11,6 +11,8 @@
#define MSM_AFE_PORT_TYPE_TX 1
#define AFE_MAX_PORTS AFE_PORT_MAX
+#define Q6AFE_MAX_MI2S_LINES 4
+
#define AFE_MAX_CHAN_COUNT 8
#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
@@ -29,9 +31,19 @@ struct q6afe_slim_cfg {
u8 ch_mapping[AFE_MAX_CHAN_COUNT];
};
+struct q6afe_i2s_cfg {
+ u32 sample_rate;
+ u16 bit_width;
+ u16 data_format;
+ u16 num_channels;
+ u32 sd_line_mask;
+ int fmt;
+};
+
struct q6afe_port_config {
struct q6afe_hdmi_cfg hdmi;
struct q6afe_slim_cfg slim;
+ struct q6afe_i2s_cfg i2s_cfg;
};
struct q6afe_port;
@@ -45,5 +57,6 @@ void q6afe_hdmi_port_prepare(struct q6afe_port *port,
struct q6afe_hdmi_cfg *cfg);
void q6afe_slim_port_prepare(struct q6afe_port *port,
struct q6afe_slim_cfg *cfg);
+int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg);
#endif /* __Q6AFE_H__ */
--
2.17.0
^ permalink raw reply related [flat|nested] 25+ messages in thread
* [PATCH v9 04/15] ASoC: qdsp6: q6afe: Add support to MI2S sysclks
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (2 preceding siblings ...)
2018-05-18 12:55 ` [PATCH v9 03/15] ASoC: qdsp6: q6afe: Add support to MI2S ports Srinivas Kandagatla
@ 2018-05-18 12:55 ` Srinivas Kandagatla
2018-05-21 15:49 ` Applied "ASoC: qdsp6: q6afe: Add support to MI2S sysclks" to the asoc tree Mark Brown
2018-05-18 12:56 ` [PATCH v9 05/15] ASoC: qdsp6: q6adm: Add q6adm driver Srinivas Kandagatla
` (10 subsequent siblings)
14 siblings, 1 reply; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:55 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
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>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/qdsp6/q6afe.c | 166 +++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 131 ++++++++++++++++++++++++++++++++++
2 files changed, 297 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 0004369b3661..de0030068ecb 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -34,6 +34,9 @@
#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235
+#define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238
+#define AFE_PARAM_ID_INT_DIGITAL_CDC_CLK_CONFIG 0x00010239
+
#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212
#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D
@@ -67,6 +70,11 @@
#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
+/* Clock set API version */
+#define AFE_API_VERSION_CLOCK_SET 1
+#define Q6AFE_LPASS_CLK_CONFIG_API_VERSION 0x1
+#define AFE_MODULE_CLOCK_SET 0x0001028F
+#define AFE_PARAM_ID_CLOCK_SET 0x00010290
/* SLIMbus Rx port on channel 0. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000
@@ -142,6 +150,13 @@ struct afe_port_param_data_v2 {
u16 reserved;
} __packed;
+struct afe_svc_cmd_set_param {
+ uint32_t payload_size;
+ uint32_t payload_address_lsw;
+ uint32_t payload_address_msw;
+ uint32_t mem_map_handle;
+} __packed;
+
struct afe_port_cmd_set_param_v2 {
u16 port_id;
u16 payload_size;
@@ -202,6 +217,23 @@ 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;
@@ -219,6 +251,16 @@ union afe_port_config {
struct afe_param_id_i2s_cfg i2s_cfg;
} __packed;
+
+struct afe_clk_set {
+ uint32_t clk_set_minor_version;
+ uint32_t clk_id;
+ uint32_t clk_freq_in_hz;
+ uint16_t clk_attri;
+ uint16_t clk_root;
+ uint32_t enable;
+};
+
struct q6afe_port {
wait_queue_head_t wait;
union afe_port_config port_cfg;
@@ -404,6 +446,54 @@ static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt,
return ret;
}
+static int q6afe_port_set_param(struct q6afe_port *port, void *data,
+ int param_id, int module_id, int psize)
+{
+ struct afe_svc_cmd_set_param *param;
+ struct afe_port_param_data_v2 *pdata;
+ struct q6afe *afe = port->afe;
+ struct apr_pkt *pkt;
+ u16 port_id = port->id;
+ int ret, pkt_size;
+ void *p, *pl;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*param) + sizeof(*pdata) + psize;
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ param = p + APR_HDR_SIZE;
+ pdata = p + APR_HDR_SIZE + sizeof(*param);
+ pl = p + APR_HDR_SIZE + sizeof(*param) + sizeof(*pdata);
+ memcpy(pl, data, psize);
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.token = port->token;
+ pkt->hdr.opcode = AFE_SVC_CMD_SET_PARAM;
+
+ param->payload_size = sizeof(*pdata) + psize;
+ param->payload_address_lsw = 0x00;
+ param->payload_address_msw = 0x00;
+ param->mem_map_handle = 0x00;
+ pdata->module_id = module_id;
+ pdata->param_id = param_id;
+ pdata->param_size = psize;
+
+ ret = afe_apr_send_pkt(afe, pkt, port);
+ if (ret)
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+
+ kfree(pkt);
+ return ret;
+}
+
static int q6afe_port_set_param_v2(struct q6afe_port *port, void *data,
int param_id, int module_id, int psize)
{
@@ -453,6 +543,82 @@ 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)
+{
+ return q6afe_port_set_param_v2(port, cfg,
+ AFE_PARAM_ID_LPAIF_CLK_CONFIG,
+ AFE_MODULE_AUDIO_DEV_INTERFACE,
+ sizeof(*cfg));
+}
+
+static int q6afe_set_lpass_clock_v2(struct q6afe_port *port,
+ struct afe_clk_set *cfg)
+{
+ return q6afe_port_set_param(port, cfg, AFE_PARAM_ID_CLOCK_SET,
+ AFE_MODULE_CLOCK_SET, sizeof(*cfg));
+}
+
+static int q6afe_set_digital_codec_core_clock(struct q6afe_port *port,
+ struct afe_digital_clk_cfg *cfg)
+{
+ return q6afe_port_set_param_v2(port, cfg,
+ AFE_PARAM_ID_INT_DIGITAL_CDC_CLK_CONFIG,
+ AFE_MODULE_AUDIO_DEV_INTERFACE,
+ sizeof(*cfg));
+}
+
+int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
+ int clk_src, int clk_root,
+ unsigned int freq, int dir)
+{
+ struct afe_clk_cfg ccfg = {0,};
+ struct afe_clk_set cset = {0,};
+ struct afe_digital_clk_cfg dcfg = {0,};
+ int ret;
+
+ switch (clk_id) {
+ case LPAIF_DIG_CLK:
+ dcfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ dcfg.clk_val = freq;
+ dcfg.clk_root = clk_root;
+ ret = q6afe_set_digital_codec_core_clock(port, &dcfg);
+ break;
+ case LPAIF_BIT_CLK:
+ ccfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ ccfg.clk_val1 = freq;
+ ccfg.clk_src = clk_src;
+ ccfg.clk_root = clk_root;
+ ccfg.clk_set_mode = Q6AFE_LPASS_MODE_CLK1_VALID;
+ ret = q6afe_set_lpass_clock(port, &ccfg);
+ break;
+
+ case LPAIF_OSR_CLK:
+ ccfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ ccfg.clk_val2 = freq;
+ ccfg.clk_src = clk_src;
+ ccfg.clk_root = clk_root;
+ ccfg.clk_set_mode = Q6AFE_LPASS_MODE_CLK2_VALID;
+ ret = q6afe_set_lpass_clock(port, &ccfg);
+ break;
+ case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1:
+ cset.clk_set_minor_version = AFE_API_VERSION_CLOCK_SET;
+ cset.clk_id = clk_id;
+ cset.clk_freq_in_hz = freq;
+ cset.clk_attri = clk_src;
+ cset.clk_root = clk_root;
+ cset.enable = !!freq;
+ ret = q6afe_set_lpass_clock_v2(port, &cset);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6afe_port_set_sysclk);
+
/**
* q6afe_port_stop() - Stop a afe port
*
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index 3cb3bb4985a9..5ca54a9bdfd5 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -16,6 +16,134 @@
#define AFE_MAX_CHAN_COUNT 8
#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
+#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
+#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
+
+#define LPAIF_DIG_CLK 1
+#define LPAIF_BIT_CLK 2
+#define LPAIF_OSR_CLK 3
+
+/* Clock ID for Primary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT 0x100
+/* Clock ID for Primary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_EBIT 0x101
+/* Clock ID for Secondary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT 0x102
+/* Clock ID for Secondary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_EBIT 0x103
+/* Clock ID for Tertiary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_IBIT 0x104
+/* Clock ID for Tertiary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_EBIT 0x105
+/* Clock ID for Quartnery I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT 0x106
+/* Clock ID for Quartnery I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_EBIT 0x107
+/* Clock ID for Speaker I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_IBIT 0x108
+/* Clock ID for Speaker I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_EBIT 0x109
+/* Clock ID for Speaker I2S OSR */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_OSR 0x10A
+
+/* Clock ID for QUINARY I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_IBIT 0x10B
+/* Clock ID for QUINARY I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_EBIT 0x10C
+/* Clock ID for SENARY I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_IBIT 0x10D
+/* Clock ID for SENARY I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_EBIT 0x10E
+/* Clock ID for INT0 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT0_MI2S_IBIT 0x10F
+/* Clock ID for INT1 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT1_MI2S_IBIT 0x110
+/* Clock ID for INT2 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT2_MI2S_IBIT 0x111
+/* Clock ID for INT3 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT3_MI2S_IBIT 0x112
+/* Clock ID for INT4 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT4_MI2S_IBIT 0x113
+/* Clock ID for INT5 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT5_MI2S_IBIT 0x114
+/* Clock ID for INT6 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT6_MI2S_IBIT 0x115
+
+/* Clock ID for QUINARY MI2S OSR CLK */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR 0x116
+
+/* Clock ID for Primary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT 0x200
+/* Clock ID for Primary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_EBIT 0x201
+/* Clock ID for Secondary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_IBIT 0x202
+/* Clock ID for Secondary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_EBIT 0x203
+/* Clock ID for Tertiary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_IBIT 0x204
+/* Clock ID for Tertiary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_EBIT 0x205
+/* Clock ID for Quartery PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT 0x206
+/* Clock ID for Quartery PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_EBIT 0x207
+/* Clock ID for Quinary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_PCM_IBIT 0x208
+/* Clock ID for Quinary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_PCM_EBIT 0x209
+/* Clock ID for QUINARY PCM OSR */
+#define Q6AFE_LPASS_CLK_ID_QUI_PCM_OSR 0x20A
+
+/** Clock ID for Primary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT 0x200
+/** Clock ID for Primary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_EBIT 0x201
+/** Clock ID for Secondary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_IBIT 0x202
+/** Clock ID for Secondary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_EBIT 0x203
+/** Clock ID for Tertiary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_IBIT 0x204
+/** Clock ID for Tertiary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_EBIT 0x205
+/** Clock ID for Quartery TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT 0x206
+/** Clock ID for Quartery TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_EBIT 0x207
+/** Clock ID for Quinary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_IBIT 0x208
+/** Clock ID for Quinary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT 0x209
+/** Clock ID for Quinary TDM OSR */
+#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_OSR 0x20A
+
+/* Clock ID for MCLK1 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_1 0x300
+/* Clock ID for MCLK2 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_2 0x301
+/* Clock ID for MCLK3 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_3 0x302
+/* Clock ID for MCLK4 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_4 0x304
+/* Clock ID for Internal Digital Codec Core */
+#define Q6AFE_LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE 0x303
+/* Clock ID for INT MCLK0 */
+#define Q6AFE_LPASS_CLK_ID_INT_MCLK_0 0x305
+/* Clock ID for INT MCLK1 */
+#define Q6AFE_LPASS_CLK_ID_INT_MCLK_1 0x306
+
+/* Clock attribute for invalid use (reserved for internal usage) */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVALID 0x0
+/* Clock attribute for no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO 0x1
+/* Clock attribute for dividend couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND 0x2
+/* Clock attribute for divisor couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR 0x3
+/* Clock attribute for invert and no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO 0x4
+
struct q6afe_hdmi_cfg {
u16 datatype;
u16 channel_allocation;
@@ -59,4 +187,7 @@ void q6afe_slim_port_prepare(struct q6afe_port *port,
struct q6afe_slim_cfg *cfg);
int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg);
+int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
+ int clk_src, int clk_root,
+ unsigned int freq, int dir);
#endif /* __Q6AFE_H__ */
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* Applied "ASoC: qdsp6: q6afe: Add support to MI2S sysclks" to the asoc tree
2018-05-18 12:55 ` [PATCH v9 04/15] ASoC: qdsp6: q6afe: Add support to MI2S sysclks Srinivas Kandagatla
@ 2018-05-21 15:49 ` Mark Brown
0 siblings, 0 replies; 25+ messages in thread
From: Mark Brown @ 2018-05-21 15:49 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, broonie, linux-arm-msm, alsa-devel, bgoswami,
mark.rutland, devicetree, rohkumar, gregkh, plai, lgirdwood,
tiwai, david.brown, robh+dt, linux-arm-kernel, spatakok,
linux-kernel, alsa-devel
The patch
ASoC: qdsp6: q6afe: Add support to MI2S sysclks
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From a4ae3af59bd58514b5417d7d802a37e82bfcfd8f Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:55:59 +0100
Subject: [PATCH] ASoC: qdsp6: q6afe: Add support to MI2S sysclks
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>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6afe.c | 166 +++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6afe.h | 131 +++++++++++++++++++++++++++
2 files changed, 297 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c
index 0004369b3661..de0030068ecb 100644
--- a/sound/soc/qcom/qdsp6/q6afe.c
+++ b/sound/soc/qcom/qdsp6/q6afe.c
@@ -34,6 +34,9 @@
#define AFE_PARAM_ID_CDC_SLIMBUS_SLAVE_CFG 0x00010235
+#define AFE_PARAM_ID_LPAIF_CLK_CONFIG 0x00010238
+#define AFE_PARAM_ID_INT_DIGITAL_CDC_CLK_CONFIG 0x00010239
+
#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212
#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D
@@ -67,6 +70,11 @@
#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
+/* Clock set API version */
+#define AFE_API_VERSION_CLOCK_SET 1
+#define Q6AFE_LPASS_CLK_CONFIG_API_VERSION 0x1
+#define AFE_MODULE_CLOCK_SET 0x0001028F
+#define AFE_PARAM_ID_CLOCK_SET 0x00010290
/* SLIMbus Rx port on channel 0. */
#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000
@@ -142,6 +150,13 @@ struct afe_port_param_data_v2 {
u16 reserved;
} __packed;
+struct afe_svc_cmd_set_param {
+ uint32_t payload_size;
+ uint32_t payload_address_lsw;
+ uint32_t payload_address_msw;
+ uint32_t mem_map_handle;
+} __packed;
+
struct afe_port_cmd_set_param_v2 {
u16 port_id;
u16 payload_size;
@@ -202,6 +217,23 @@ 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;
@@ -219,6 +251,16 @@ union afe_port_config {
struct afe_param_id_i2s_cfg i2s_cfg;
} __packed;
+
+struct afe_clk_set {
+ uint32_t clk_set_minor_version;
+ uint32_t clk_id;
+ uint32_t clk_freq_in_hz;
+ uint16_t clk_attri;
+ uint16_t clk_root;
+ uint32_t enable;
+};
+
struct q6afe_port {
wait_queue_head_t wait;
union afe_port_config port_cfg;
@@ -404,6 +446,54 @@ static int afe_apr_send_pkt(struct q6afe *afe, struct apr_pkt *pkt,
return ret;
}
+static int q6afe_port_set_param(struct q6afe_port *port, void *data,
+ int param_id, int module_id, int psize)
+{
+ struct afe_svc_cmd_set_param *param;
+ struct afe_port_param_data_v2 *pdata;
+ struct q6afe *afe = port->afe;
+ struct apr_pkt *pkt;
+ u16 port_id = port->id;
+ int ret, pkt_size;
+ void *p, *pl;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*param) + sizeof(*pdata) + psize;
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ param = p + APR_HDR_SIZE;
+ pdata = p + APR_HDR_SIZE + sizeof(*param);
+ pl = p + APR_HDR_SIZE + sizeof(*param) + sizeof(*pdata);
+ memcpy(pl, data, psize);
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.token = port->token;
+ pkt->hdr.opcode = AFE_SVC_CMD_SET_PARAM;
+
+ param->payload_size = sizeof(*pdata) + psize;
+ param->payload_address_lsw = 0x00;
+ param->payload_address_msw = 0x00;
+ param->mem_map_handle = 0x00;
+ pdata->module_id = module_id;
+ pdata->param_id = param_id;
+ pdata->param_size = psize;
+
+ ret = afe_apr_send_pkt(afe, pkt, port);
+ if (ret)
+ dev_err(afe->dev, "AFE enable for port 0x%x failed %d\n",
+ port_id, ret);
+
+ kfree(pkt);
+ return ret;
+}
+
static int q6afe_port_set_param_v2(struct q6afe_port *port, void *data,
int param_id, int module_id, int psize)
{
@@ -453,6 +543,82 @@ 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)
+{
+ return q6afe_port_set_param_v2(port, cfg,
+ AFE_PARAM_ID_LPAIF_CLK_CONFIG,
+ AFE_MODULE_AUDIO_DEV_INTERFACE,
+ sizeof(*cfg));
+}
+
+static int q6afe_set_lpass_clock_v2(struct q6afe_port *port,
+ struct afe_clk_set *cfg)
+{
+ return q6afe_port_set_param(port, cfg, AFE_PARAM_ID_CLOCK_SET,
+ AFE_MODULE_CLOCK_SET, sizeof(*cfg));
+}
+
+static int q6afe_set_digital_codec_core_clock(struct q6afe_port *port,
+ struct afe_digital_clk_cfg *cfg)
+{
+ return q6afe_port_set_param_v2(port, cfg,
+ AFE_PARAM_ID_INT_DIGITAL_CDC_CLK_CONFIG,
+ AFE_MODULE_AUDIO_DEV_INTERFACE,
+ sizeof(*cfg));
+}
+
+int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
+ int clk_src, int clk_root,
+ unsigned int freq, int dir)
+{
+ struct afe_clk_cfg ccfg = {0,};
+ struct afe_clk_set cset = {0,};
+ struct afe_digital_clk_cfg dcfg = {0,};
+ int ret;
+
+ switch (clk_id) {
+ case LPAIF_DIG_CLK:
+ dcfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ dcfg.clk_val = freq;
+ dcfg.clk_root = clk_root;
+ ret = q6afe_set_digital_codec_core_clock(port, &dcfg);
+ break;
+ case LPAIF_BIT_CLK:
+ ccfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ ccfg.clk_val1 = freq;
+ ccfg.clk_src = clk_src;
+ ccfg.clk_root = clk_root;
+ ccfg.clk_set_mode = Q6AFE_LPASS_MODE_CLK1_VALID;
+ ret = q6afe_set_lpass_clock(port, &ccfg);
+ break;
+
+ case LPAIF_OSR_CLK:
+ ccfg.i2s_cfg_minor_version = AFE_API_VERSION_I2S_CONFIG;
+ ccfg.clk_val2 = freq;
+ ccfg.clk_src = clk_src;
+ ccfg.clk_root = clk_root;
+ ccfg.clk_set_mode = Q6AFE_LPASS_MODE_CLK2_VALID;
+ ret = q6afe_set_lpass_clock(port, &ccfg);
+ break;
+ case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1:
+ cset.clk_set_minor_version = AFE_API_VERSION_CLOCK_SET;
+ cset.clk_id = clk_id;
+ cset.clk_freq_in_hz = freq;
+ cset.clk_attri = clk_src;
+ cset.clk_root = clk_root;
+ cset.enable = !!freq;
+ ret = q6afe_set_lpass_clock_v2(port, &cset);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6afe_port_set_sysclk);
+
/**
* q6afe_port_stop() - Stop a afe port
*
diff --git a/sound/soc/qcom/qdsp6/q6afe.h b/sound/soc/qcom/qdsp6/q6afe.h
index 3cb3bb4985a9..5ca54a9bdfd5 100644
--- a/sound/soc/qcom/qdsp6/q6afe.h
+++ b/sound/soc/qcom/qdsp6/q6afe.h
@@ -16,6 +16,134 @@
#define AFE_MAX_CHAN_COUNT 8
#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
+#define Q6AFE_LPASS_CLK_SRC_INTERNAL 1
+#define Q6AFE_LPASS_CLK_ROOT_DEFAULT 0
+
+#define LPAIF_DIG_CLK 1
+#define LPAIF_BIT_CLK 2
+#define LPAIF_OSR_CLK 3
+
+/* Clock ID for Primary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT 0x100
+/* Clock ID for Primary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_MI2S_EBIT 0x101
+/* Clock ID for Secondary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_IBIT 0x102
+/* Clock ID for Secondary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_MI2S_EBIT 0x103
+/* Clock ID for Tertiary I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_IBIT 0x104
+/* Clock ID for Tertiary I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_MI2S_EBIT 0x105
+/* Clock ID for Quartnery I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_IBIT 0x106
+/* Clock ID for Quartnery I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_MI2S_EBIT 0x107
+/* Clock ID for Speaker I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_IBIT 0x108
+/* Clock ID for Speaker I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_EBIT 0x109
+/* Clock ID for Speaker I2S OSR */
+#define Q6AFE_LPASS_CLK_ID_SPEAKER_I2S_OSR 0x10A
+
+/* Clock ID for QUINARY I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_IBIT 0x10B
+/* Clock ID for QUINARY I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_EBIT 0x10C
+/* Clock ID for SENARY I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_IBIT 0x10D
+/* Clock ID for SENARY I2S EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEN_MI2S_EBIT 0x10E
+/* Clock ID for INT0 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT0_MI2S_IBIT 0x10F
+/* Clock ID for INT1 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT1_MI2S_IBIT 0x110
+/* Clock ID for INT2 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT2_MI2S_IBIT 0x111
+/* Clock ID for INT3 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT3_MI2S_IBIT 0x112
+/* Clock ID for INT4 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT4_MI2S_IBIT 0x113
+/* Clock ID for INT5 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT5_MI2S_IBIT 0x114
+/* Clock ID for INT6 I2S IBIT */
+#define Q6AFE_LPASS_CLK_ID_INT6_MI2S_IBIT 0x115
+
+/* Clock ID for QUINARY MI2S OSR CLK */
+#define Q6AFE_LPASS_CLK_ID_QUI_MI2S_OSR 0x116
+
+/* Clock ID for Primary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_IBIT 0x200
+/* Clock ID for Primary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_PCM_EBIT 0x201
+/* Clock ID for Secondary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_IBIT 0x202
+/* Clock ID for Secondary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_PCM_EBIT 0x203
+/* Clock ID for Tertiary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_IBIT 0x204
+/* Clock ID for Tertiary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_PCM_EBIT 0x205
+/* Clock ID for Quartery PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_IBIT 0x206
+/* Clock ID for Quartery PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_PCM_EBIT 0x207
+/* Clock ID for Quinary PCM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_PCM_IBIT 0x208
+/* Clock ID for Quinary PCM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_PCM_EBIT 0x209
+/* Clock ID for QUINARY PCM OSR */
+#define Q6AFE_LPASS_CLK_ID_QUI_PCM_OSR 0x20A
+
+/** Clock ID for Primary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_IBIT 0x200
+/** Clock ID for Primary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_PRI_TDM_EBIT 0x201
+/** Clock ID for Secondary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_IBIT 0x202
+/** Clock ID for Secondary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_SEC_TDM_EBIT 0x203
+/** Clock ID for Tertiary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_IBIT 0x204
+/** Clock ID for Tertiary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_TER_TDM_EBIT 0x205
+/** Clock ID for Quartery TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_IBIT 0x206
+/** Clock ID for Quartery TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUAD_TDM_EBIT 0x207
+/** Clock ID for Quinary TDM IBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_IBIT 0x208
+/** Clock ID for Quinary TDM EBIT */
+#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_EBIT 0x209
+/** Clock ID for Quinary TDM OSR */
+#define Q6AFE_LPASS_CLK_ID_QUIN_TDM_OSR 0x20A
+
+/* Clock ID for MCLK1 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_1 0x300
+/* Clock ID for MCLK2 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_2 0x301
+/* Clock ID for MCLK3 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_3 0x302
+/* Clock ID for MCLK4 */
+#define Q6AFE_LPASS_CLK_ID_MCLK_4 0x304
+/* Clock ID for Internal Digital Codec Core */
+#define Q6AFE_LPASS_CLK_ID_INTERNAL_DIGITAL_CODEC_CORE 0x303
+/* Clock ID for INT MCLK0 */
+#define Q6AFE_LPASS_CLK_ID_INT_MCLK_0 0x305
+/* Clock ID for INT MCLK1 */
+#define Q6AFE_LPASS_CLK_ID_INT_MCLK_1 0x306
+
+/* Clock attribute for invalid use (reserved for internal usage) */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVALID 0x0
+/* Clock attribute for no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO 0x1
+/* Clock attribute for dividend couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVIDEND 0x2
+/* Clock attribute for divisor couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_DIVISOR 0x3
+/* Clock attribute for invert and no couple case */
+#define Q6AFE_LPASS_CLK_ATTRIBUTE_INVERT_COUPLE_NO 0x4
+
struct q6afe_hdmi_cfg {
u16 datatype;
u16 channel_allocation;
@@ -59,4 +187,7 @@ void q6afe_slim_port_prepare(struct q6afe_port *port,
struct q6afe_slim_cfg *cfg);
int q6afe_i2s_port_prepare(struct q6afe_port *port, struct q6afe_i2s_cfg *cfg);
+int q6afe_port_set_sysclk(struct q6afe_port *port, int clk_id,
+ int clk_src, int clk_root,
+ unsigned int freq, int dir);
#endif /* __Q6AFE_H__ */
--
2.17.0
^ permalink raw reply related [flat|nested] 25+ messages in thread
* [PATCH v9 05/15] ASoC: qdsp6: q6adm: Add q6adm driver
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (3 preceding siblings ...)
2018-05-18 12:55 ` [PATCH v9 04/15] ASoC: qdsp6: q6afe: Add support to MI2S sysclks Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-21 15:48 ` Applied "ASoC: qdsp6: q6adm: Add q6adm driver" to the asoc tree Mark Brown
2018-05-18 12:56 ` [PATCH v9 06/15] ASoC: qdsp6: q6asm: Add q6asm driver Srinivas Kandagatla
` (9 subsequent siblings)
14 siblings, 1 reply; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
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>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6adm.c | 646 ++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6adm.h | 27 ++
4 files changed, 678 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 bb0a2afb0563..971127edbc23 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -50,12 +50,16 @@ config SND_SOC_QDSP6_CORE
config SND_SOC_QDSP6_AFE
tristate
+config SND_SOC_QDSP6_ADM
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_CORE
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 7ff666bd10ca..95cdb3a12694 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_CORE) += q6core.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..9983c665a941
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -0,0 +1,646 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/slab.h>
+#include <linux/wait.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/module.h>
+#include <linux/sched.h>
+#include <linux/jiffies.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/kref.h>
+#include <linux/wait.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/platform_device.h>
+#include <sound/asound.h>
+#include "q6adm.h"
+#include "q6afe.h"
+#include "q6core.h"
+#include "q6dsp-errno.h"
+#include "q6dsp-common.h"
+
+#define ADM_CMD_DEVICE_OPEN_V5 0x00010326
+#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329
+#define ADM_CMD_DEVICE_CLOSE_V5 0x00010327
+#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325
+
+#define TIMEOUT_MS 1000
+#define RESET_COPP_ID 99
+#define INVALID_COPP_ID 0xFF
+/* Definition for a legacy device session. */
+#define ADM_LEGACY_DEVICE_SESSION 0
+#define ADM_MATRIX_ID_AUDIO_RX 0
+#define ADM_MATRIX_ID_AUDIO_TX 1
+
+struct q6copp {
+ int afe_port;
+ int copp_idx;
+ int id;
+ int topology;
+ int mode;
+ int rate;
+ int bit_width;
+ int channels;
+ int app_type;
+ int acdb_id;
+
+ struct aprv2_ibasic_rsp_result_t result;
+ struct kref refcount;
+ wait_queue_head_t wait;
+ struct list_head node;
+ struct q6adm *adm;
+};
+
+struct q6adm {
+ struct apr_device *apr;
+ struct device *dev;
+ struct q6core_svc_api_info ainfo;
+ unsigned long copp_bitmap[AFE_MAX_PORTS];
+ struct list_head copps_list;
+ spinlock_t copps_list_lock;
+ struct aprv2_ibasic_rsp_result_t result;
+ struct mutex lock;
+ wait_queue_head_t matrix_map_wait;
+ struct platform_device *pdev_routing;
+};
+
+struct q6adm_cmd_device_open_v5 {
+ 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 q6adm_cmd_matrix_map_routings_v5 {
+ u32 matrix_id;
+ u32 num_sessions;
+} __packed;
+
+struct q6adm_session_map_node_v5 {
+ u16 session_id;
+ u16 num_copps;
+} __packed;
+
+static struct q6copp *q6adm_find_copp(struct q6adm *adm, int port_idx,
+ int copp_idx)
+{
+ struct q6copp *c = NULL;
+ struct q6copp *ret = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+ list_for_each_entry(c, &adm->copps_list, node) {
+ if ((port_idx == c->afe_port) && (copp_idx == c->copp_idx)) {
+ ret = c;
+ kref_get(&c->refcount);
+ break;
+ }
+ }
+
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+
+ return ret;
+
+}
+
+static void q6adm_free_copp(struct kref *ref)
+{
+ struct q6copp *c = container_of(ref, struct q6copp, refcount);
+ struct q6adm *adm = c->adm;
+ unsigned long flags;
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+ clear_bit(c->copp_idx, &adm->copp_bitmap[c->afe_port]);
+ list_del(&c->node);
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+ kfree(c);
+}
+
+static int q6adm_callback(struct apr_device *adev, struct apr_resp_pkt *data)
+{
+ struct aprv2_ibasic_rsp_result_t *result = data->payload;
+ int port_idx, copp_idx;
+ struct apr_hdr *hdr = &data->hdr;
+ struct q6copp *copp;
+ struct q6adm *adm = dev_get_drvdata(&adev->dev);
+
+ if (!data->payload_size)
+ return 0;
+
+ copp_idx = (hdr->token) & 0XFF;
+ port_idx = ((hdr->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, hdr->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, hdr->token);
+ return 0;
+ }
+
+ switch (hdr->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 = q6adm_find_copp(adm, port_idx, copp_idx);
+ if (!copp)
+ return 0;
+
+ copp->result = *result;
+ wake_up(&copp->wait);
+ kref_put(&copp->refcount, q6adm_free_copp);
+ break;
+ case ADM_CMD_MATRIX_MAP_ROUTINGS_V5:
+ adm->result = *result;
+ wake_up(&adm->matrix_map_wait);
+ break;
+
+ default:
+ dev_err(&adev->dev, "Unknown Cmd: 0x%x\n",
+ result->opcode);
+ break;
+ }
+ return 0;
+ }
+ case ADM_CMDRSP_DEVICE_OPEN_V5: {
+ struct adm_cmd_rsp_device_open_v5 {
+ u32 status;
+ u16 copp_id;
+ u16 reserved;
+ } __packed * open = data->payload;
+
+ copp = q6adm_find_copp(adm, port_idx, copp_idx);
+ if (!copp)
+ return 0;
+
+ if (open->copp_id == INVALID_COPP_ID) {
+ dev_err(&adev->dev, "Invalid coppid rxed %d\n",
+ open->copp_id);
+ copp->result.status = ADSP_EBADPARAM;
+ wake_up(&copp->wait);
+ kref_put(&copp->refcount, q6adm_free_copp);
+ break;
+ }
+ copp->result.opcode = hdr->opcode;
+ copp->id = open->copp_id;
+ wake_up(&copp->wait);
+ kref_put(&copp->refcount, q6adm_free_copp);
+ }
+ break;
+ default:
+ dev_err(&adev->dev, "Unknown cmd:0x%x\n",
+ hdr->opcode);
+ break;
+ }
+
+ return 0;
+}
+
+static struct q6copp *q6adm_alloc_copp(struct q6adm *adm, int port_idx)
+{
+ struct q6copp *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_ATOMIC);
+ if (!c)
+ return ERR_PTR(-ENOMEM);
+
+ set_bit(idx, &adm->copp_bitmap[port_idx]);
+ c->copp_idx = idx;
+ c->afe_port = port_idx;
+ c->adm = adm;
+
+ init_waitqueue_head(&c->wait);
+
+ return c;
+}
+
+static int q6adm_apr_send_copp_pkt(struct q6adm *adm, struct q6copp *copp,
+ struct apr_pkt *pkt, uint32_t rsp_opcode)
+{
+ struct device *dev = adm->dev;
+ uint32_t opcode = pkt->hdr.opcode;
+ int ret;
+
+ mutex_lock(&adm->lock);
+ copp->result.opcode = 0;
+ copp->result.status = 0;
+ ret = apr_send_pkt(adm->apr, pkt);
+ if (ret < 0) {
+ dev_err(dev, "Failed to send APR packet\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ /* Wait for the callback with copp id */
+ if (rsp_opcode)
+ ret = wait_event_timeout(copp->wait,
+ (copp->result.opcode == opcode) ||
+ (copp->result.opcode == rsp_opcode),
+ msecs_to_jiffies(TIMEOUT_MS));
+ else
+ ret = wait_event_timeout(copp->wait,
+ (copp->result.opcode == opcode),
+ msecs_to_jiffies(TIMEOUT_MS));
+
+ if (!ret) {
+ dev_err(dev, "ADM copp cmd timedout\n");
+ ret = -ETIMEDOUT;
+ } else if (copp->result.status > 0) {
+ dev_err(dev, "DSP returned error[%d]\n",
+ copp->result.status);
+ ret = -EINVAL;
+ }
+
+err:
+ mutex_unlock(&adm->lock);
+ return ret;
+}
+
+static int q6adm_device_close(struct q6adm *adm, struct q6copp *copp,
+ int port_id, int copp_idx)
+{
+ struct apr_pkt close;
+
+ close.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ close.hdr.pkt_size = sizeof(close);
+ close.hdr.src_port = port_id;
+ close.hdr.dest_port = copp->id;
+ close.hdr.token = port_id << 16 | copp_idx;
+ close.hdr.opcode = ADM_CMD_DEVICE_CLOSE_V5;
+
+ return q6adm_apr_send_copp_pkt(adm, copp, &close, 0);
+}
+
+static struct q6copp *q6adm_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 q6copp *c = NULL;
+ struct q6copp *ret = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+
+ list_for_each_entry(c, &adm->copps_list, node) {
+ if ((port_id == c->afe_port) && (topology == c->topology) &&
+ (mode == c->mode) && (rate == c->rate) &&
+ (bit_width == c->bit_width) && (app_type == c->app_type)) {
+ ret = c;
+ kref_get(&c->refcount);
+ }
+ }
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+
+ return ret;
+}
+
+static int q6adm_device_open(struct q6adm *adm, struct q6copp *copp,
+ int port_id, int path, int topology,
+ int channel_mode, int bit_width, int rate)
+{
+ struct q6adm_cmd_device_open_v5 *open;
+ int afe_port = q6afe_get_port_id(port_id);
+ struct apr_pkt *pkt;
+ void *p;
+ int ret, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*open);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ open = p + APR_HDR_SIZE;
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = afe_port;
+ pkt->hdr.dest_port = afe_port;
+ pkt->hdr.token = port_id << 16 | copp->copp_idx;
+ pkt->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)
+ goto err;
+
+ ret = q6adm_apr_send_copp_pkt(adm, copp, pkt,
+ ADM_CMDRSP_DEVICE_OPEN_V5);
+
+err:
+ kfree(pkt);
+ return ret;
+}
+
+/**
+ * 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 pointer on success.
+ */
+struct q6copp *q6adm_open(struct device *dev, int port_id, int path, int rate,
+ int channel_mode, int topology, int perf_mode,
+ uint16_t bit_width, int app_type, int acdb_id)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+ struct q6copp *copp;
+ unsigned long flags;
+ int ret = 0;
+
+ if (port_id < 0) {
+ dev_err(dev, "Invalid port_id 0x%x\n", port_id);
+ return ERR_PTR(-EINVAL);
+ }
+
+ copp = q6adm_find_matching_copp(adm, port_id, topology, perf_mode,
+ rate, channel_mode, bit_width, app_type);
+ if (copp) {
+ dev_err(dev, "Found Matching Copp 0x%x\n", copp->copp_idx);
+ return copp;
+ }
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+ copp = q6adm_alloc_copp(adm, port_id);
+ if (IS_ERR_OR_NULL(copp)) {
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+ return ERR_CAST(copp);
+ }
+
+ list_add_tail(&copp->node, &adm->copps_list);
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+
+ kref_init(&copp->refcount);
+ copp->topology = topology;
+ copp->mode = perf_mode;
+ copp->rate = rate;
+ copp->channels = channel_mode;
+ copp->bit_width = bit_width;
+ copp->app_type = app_type;
+
+
+ ret = q6adm_device_open(adm, copp, port_id, path, topology,
+ channel_mode, bit_width, rate);
+ if (ret < 0) {
+ kref_put(&copp->refcount, q6adm_free_copp);
+ return ERR_PTR(ret);
+ }
+
+ return copp;
+}
+EXPORT_SYMBOL_GPL(q6adm_open);
+
+/**
+ * q6adm_get_copp_id() - get copp index
+ *
+ * @copp: Pointer to valid copp
+ *
+ * Return: Will be an negative on error or a valid copp index on success.
+ **/
+int q6adm_get_copp_id(struct q6copp *copp)
+{
+ if (!copp)
+ return -EINVAL;
+
+ return copp->copp_idx;
+}
+EXPORT_SYMBOL_GPL(q6adm_get_copp_id);
+
+/**
+ * q6adm_matrix_map() - Map asm streams and afe ports using payload
+ *
+ * @dev: Pointer to adm child device.
+ * @path: playback or capture path.
+ * @payload_map: map between session id and afe ports.
+ * @perf_mode: Performace mode.
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6adm_matrix_map(struct device *dev, int path,
+ struct route_payload payload_map, int perf_mode)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+ struct q6adm_cmd_matrix_map_routings_v5 *route;
+ struct q6adm_session_map_node_v5 *node;
+ struct apr_pkt *pkt;
+ uint16_t *copps_list;
+ int pkt_size, ret, i, copp_idx;
+ void *matrix_map = NULL;
+ struct q6copp *copp;
+
+ /* Assumes port_ids have already been validated during adm_open */
+ pkt_size = (APR_HDR_SIZE + sizeof(*route) + sizeof(*node) +
+ (sizeof(uint32_t) * payload_map.num_copps));
+
+ matrix_map = kzalloc(pkt_size, GFP_KERNEL);
+ if (!matrix_map)
+ return -ENOMEM;
+
+ pkt = matrix_map;
+ route = matrix_map + APR_HDR_SIZE;
+ node = matrix_map + APR_HDR_SIZE + sizeof(*route);
+ copps_list = matrix_map + APR_HDR_SIZE + sizeof(*route) + sizeof(*node);
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.token = 0;
+ pkt->hdr.opcode = ADM_CMD_MATRIX_MAP_ROUTINGS_V5;
+ route->num_sessions = 1;
+
+ switch (path) {
+ case ADM_PATH_PLAYBACK:
+ route->matrix_id = ADM_MATRIX_ID_AUDIO_RX;
+ break;
+ case ADM_PATH_LIVE_REC:
+ route->matrix_id = ADM_MATRIX_ID_AUDIO_TX;
+ break;
+ default:
+ dev_err(dev, "Wrong path set[%d]\n", path);
+ break;
+ }
+
+ node->session_id = payload_map.session_id;
+ node->num_copps = payload_map.num_copps;
+
+ 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]);
+ kfree(pkt);
+ return -EINVAL;
+ }
+ copp_idx = payload_map.copp_idx[i];
+
+ copp = q6adm_find_copp(adm, port_idx, copp_idx);
+ if (!copp) {
+ kfree(pkt);
+ return -EINVAL;
+ }
+
+ copps_list[i] = copp->id;
+ kref_put(&copp->refcount, q6adm_free_copp);
+ }
+
+ mutex_lock(&adm->lock);
+ adm->result.status = 0;
+ adm->result.opcode = 0;
+
+ ret = apr_send_pkt(adm->apr, pkt);
+ if (ret < 0) {
+ dev_err(dev, "routing for stream %d failed ret %d\n",
+ payload_map.session_id, ret);
+ goto fail_cmd;
+ }
+ ret = wait_event_timeout(adm->matrix_map_wait,
+ adm->result.opcode == pkt->hdr.opcode,
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ dev_err(dev, "routing for stream %d failed\n",
+ payload_map.session_id);
+ ret = -ETIMEDOUT;
+ goto fail_cmd;
+ } else if (adm->result.status > 0) {
+ dev_err(dev, "DSP returned error[%d]\n",
+ adm->result.status);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+fail_cmd:
+ mutex_unlock(&adm->lock);
+ kfree(pkt);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6adm_matrix_map);
+
+/**
+ * q6adm_close() - Close adm copp
+ *
+ * @dev: Pointer to adm child device.
+ * @copp: pointer to previously opened copp
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6adm_close(struct device *dev, struct q6copp *copp)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+ int ret = 0;
+
+ ret = q6adm_device_close(adm, copp, copp->afe_port, copp->copp_idx);
+ if (ret < 0) {
+ dev_err(adm->dev, "Failed to close copp %d\n", ret);
+ return ret;
+ }
+
+ kref_put(&copp->refcount, q6adm_free_copp);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6adm_close);
+
+static int q6adm_probe(struct apr_device *adev)
+{
+ struct device *dev = &adev->dev;
+ struct device_node *dais_np;
+ struct q6adm *adm;
+
+ adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL);
+ if (!adm)
+ return -ENOMEM;
+
+ adm->apr = adev;
+ dev_set_drvdata(&adev->dev, adm);
+ adm->dev = dev;
+ q6core_get_svc_api_info(adev->svc_id, &adm->ainfo);
+ mutex_init(&adm->lock);
+ init_waitqueue_head(&adm->matrix_map_wait);
+
+ INIT_LIST_HEAD(&adm->copps_list);
+ spin_lock_init(&adm->copps_list_lock);
+
+ dais_np = of_get_child_by_name(dev->of_node, "routing");
+ if (dais_np) {
+ adm->pdev_routing = of_platform_device_create(dais_np,
+ "q6routing", dev);
+ of_node_put(dais_np);
+ }
+
+ return 0;
+}
+
+static int q6adm_remove(struct apr_device *adev)
+{
+ struct q6adm *adm = dev_get_drvdata(&adev->dev);
+
+ if (adm->pdev_routing)
+ of_platform_device_destroy(&adm->pdev_routing->dev, NULL);
+
+ return 0;
+}
+
+static const struct of_device_id q6adm_device_id[] = {
+ { .compatible = "qcom,q6adm" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6adm_device_id);
+
+static struct apr_driver qcom_q6adm_driver = {
+ .probe = q6adm_probe,
+ .remove = q6adm_remove,
+ .callback = q6adm_callback,
+ .driver = {
+ .name = "qcom-q6adm",
+ .of_match_table = of_match_ptr(q6adm_device_id),
+ },
+};
+
+module_apr_driver(qcom_q6adm_driver);
+MODULE_DESCRIPTION("Q6 Audio Device Manager");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6adm.h b/sound/soc/qcom/qdsp6/q6adm.h
new file mode 100644
index 000000000000..4f56999b7fab
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6adm.h
@@ -0,0 +1,27 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __Q6_ADM_V2_H__
+#define __Q6_ADM_V2_H__
+
+#define ADM_PATH_PLAYBACK 0x1
+#define ADM_PATH_LIVE_REC 0x2
+#define MAX_COPPS_PER_PORT 8
+#define NULL_COPP_TOPOLOGY 0x00010312
+
+/* multiple copp per stream. */
+struct route_payload {
+ int num_copps;
+ int session_id;
+ int copp_idx[MAX_COPPS_PER_PORT];
+ int port_id[MAX_COPPS_PER_PORT];
+};
+
+struct q6copp;
+struct q6copp *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, struct q6copp *copp);
+int q6adm_get_copp_id(struct q6copp *copp);
+int q6adm_matrix_map(struct device *dev, int path,
+ struct route_payload payload_map, int perf_mode);
+
+#endif /* __Q6_ADM_V2_H__ */
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* Applied "ASoC: qdsp6: q6adm: Add q6adm driver" to the asoc tree
2018-05-18 12:56 ` [PATCH v9 05/15] ASoC: qdsp6: q6adm: Add q6adm driver Srinivas Kandagatla
@ 2018-05-21 15:48 ` Mark Brown
0 siblings, 0 replies; 25+ messages in thread
From: Mark Brown @ 2018-05-21 15:48 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, broonie, linux-arm-msm, alsa-devel, bgoswami,
mark.rutland, devicetree, rohkumar, gregkh, plai, lgirdwood,
tiwai, david.brown, robh+dt, linux-arm-kernel, spatakok,
linux-kernel, alsa-devel
The patch
ASoC: qdsp6: q6adm: Add q6adm driver
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From 7b20b2be51e1a9c250535e1a4df42e1cd20c7ecf Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:56:00 +0100
Subject: [PATCH] ASoC: qdsp6: q6adm: Add q6adm driver
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>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6adm.c | 646 ++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6adm.h | 27 ++
4 files changed, 678 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 bb0a2afb0563..971127edbc23 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -50,12 +50,16 @@ config SND_SOC_QDSP6_CORE
config SND_SOC_QDSP6_AFE
tristate
+config SND_SOC_QDSP6_ADM
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_CORE
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 7ff666bd10ca..95cdb3a12694 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_CORE) += q6core.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..9983c665a941
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6adm.c
@@ -0,0 +1,646 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/slab.h>
+#include <linux/wait.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <linux/module.h>
+#include <linux/sched.h>
+#include <linux/jiffies.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/kref.h>
+#include <linux/wait.h>
+#include <linux/soc/qcom/apr.h>
+#include <linux/platform_device.h>
+#include <sound/asound.h>
+#include "q6adm.h"
+#include "q6afe.h"
+#include "q6core.h"
+#include "q6dsp-errno.h"
+#include "q6dsp-common.h"
+
+#define ADM_CMD_DEVICE_OPEN_V5 0x00010326
+#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329
+#define ADM_CMD_DEVICE_CLOSE_V5 0x00010327
+#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325
+
+#define TIMEOUT_MS 1000
+#define RESET_COPP_ID 99
+#define INVALID_COPP_ID 0xFF
+/* Definition for a legacy device session. */
+#define ADM_LEGACY_DEVICE_SESSION 0
+#define ADM_MATRIX_ID_AUDIO_RX 0
+#define ADM_MATRIX_ID_AUDIO_TX 1
+
+struct q6copp {
+ int afe_port;
+ int copp_idx;
+ int id;
+ int topology;
+ int mode;
+ int rate;
+ int bit_width;
+ int channels;
+ int app_type;
+ int acdb_id;
+
+ struct aprv2_ibasic_rsp_result_t result;
+ struct kref refcount;
+ wait_queue_head_t wait;
+ struct list_head node;
+ struct q6adm *adm;
+};
+
+struct q6adm {
+ struct apr_device *apr;
+ struct device *dev;
+ struct q6core_svc_api_info ainfo;
+ unsigned long copp_bitmap[AFE_MAX_PORTS];
+ struct list_head copps_list;
+ spinlock_t copps_list_lock;
+ struct aprv2_ibasic_rsp_result_t result;
+ struct mutex lock;
+ wait_queue_head_t matrix_map_wait;
+ struct platform_device *pdev_routing;
+};
+
+struct q6adm_cmd_device_open_v5 {
+ 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 q6adm_cmd_matrix_map_routings_v5 {
+ u32 matrix_id;
+ u32 num_sessions;
+} __packed;
+
+struct q6adm_session_map_node_v5 {
+ u16 session_id;
+ u16 num_copps;
+} __packed;
+
+static struct q6copp *q6adm_find_copp(struct q6adm *adm, int port_idx,
+ int copp_idx)
+{
+ struct q6copp *c = NULL;
+ struct q6copp *ret = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+ list_for_each_entry(c, &adm->copps_list, node) {
+ if ((port_idx == c->afe_port) && (copp_idx == c->copp_idx)) {
+ ret = c;
+ kref_get(&c->refcount);
+ break;
+ }
+ }
+
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+
+ return ret;
+
+}
+
+static void q6adm_free_copp(struct kref *ref)
+{
+ struct q6copp *c = container_of(ref, struct q6copp, refcount);
+ struct q6adm *adm = c->adm;
+ unsigned long flags;
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+ clear_bit(c->copp_idx, &adm->copp_bitmap[c->afe_port]);
+ list_del(&c->node);
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+ kfree(c);
+}
+
+static int q6adm_callback(struct apr_device *adev, struct apr_resp_pkt *data)
+{
+ struct aprv2_ibasic_rsp_result_t *result = data->payload;
+ int port_idx, copp_idx;
+ struct apr_hdr *hdr = &data->hdr;
+ struct q6copp *copp;
+ struct q6adm *adm = dev_get_drvdata(&adev->dev);
+
+ if (!data->payload_size)
+ return 0;
+
+ copp_idx = (hdr->token) & 0XFF;
+ port_idx = ((hdr->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, hdr->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, hdr->token);
+ return 0;
+ }
+
+ switch (hdr->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 = q6adm_find_copp(adm, port_idx, copp_idx);
+ if (!copp)
+ return 0;
+
+ copp->result = *result;
+ wake_up(&copp->wait);
+ kref_put(&copp->refcount, q6adm_free_copp);
+ break;
+ case ADM_CMD_MATRIX_MAP_ROUTINGS_V5:
+ adm->result = *result;
+ wake_up(&adm->matrix_map_wait);
+ break;
+
+ default:
+ dev_err(&adev->dev, "Unknown Cmd: 0x%x\n",
+ result->opcode);
+ break;
+ }
+ return 0;
+ }
+ case ADM_CMDRSP_DEVICE_OPEN_V5: {
+ struct adm_cmd_rsp_device_open_v5 {
+ u32 status;
+ u16 copp_id;
+ u16 reserved;
+ } __packed * open = data->payload;
+
+ copp = q6adm_find_copp(adm, port_idx, copp_idx);
+ if (!copp)
+ return 0;
+
+ if (open->copp_id == INVALID_COPP_ID) {
+ dev_err(&adev->dev, "Invalid coppid rxed %d\n",
+ open->copp_id);
+ copp->result.status = ADSP_EBADPARAM;
+ wake_up(&copp->wait);
+ kref_put(&copp->refcount, q6adm_free_copp);
+ break;
+ }
+ copp->result.opcode = hdr->opcode;
+ copp->id = open->copp_id;
+ wake_up(&copp->wait);
+ kref_put(&copp->refcount, q6adm_free_copp);
+ }
+ break;
+ default:
+ dev_err(&adev->dev, "Unknown cmd:0x%x\n",
+ hdr->opcode);
+ break;
+ }
+
+ return 0;
+}
+
+static struct q6copp *q6adm_alloc_copp(struct q6adm *adm, int port_idx)
+{
+ struct q6copp *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_ATOMIC);
+ if (!c)
+ return ERR_PTR(-ENOMEM);
+
+ set_bit(idx, &adm->copp_bitmap[port_idx]);
+ c->copp_idx = idx;
+ c->afe_port = port_idx;
+ c->adm = adm;
+
+ init_waitqueue_head(&c->wait);
+
+ return c;
+}
+
+static int q6adm_apr_send_copp_pkt(struct q6adm *adm, struct q6copp *copp,
+ struct apr_pkt *pkt, uint32_t rsp_opcode)
+{
+ struct device *dev = adm->dev;
+ uint32_t opcode = pkt->hdr.opcode;
+ int ret;
+
+ mutex_lock(&adm->lock);
+ copp->result.opcode = 0;
+ copp->result.status = 0;
+ ret = apr_send_pkt(adm->apr, pkt);
+ if (ret < 0) {
+ dev_err(dev, "Failed to send APR packet\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ /* Wait for the callback with copp id */
+ if (rsp_opcode)
+ ret = wait_event_timeout(copp->wait,
+ (copp->result.opcode == opcode) ||
+ (copp->result.opcode == rsp_opcode),
+ msecs_to_jiffies(TIMEOUT_MS));
+ else
+ ret = wait_event_timeout(copp->wait,
+ (copp->result.opcode == opcode),
+ msecs_to_jiffies(TIMEOUT_MS));
+
+ if (!ret) {
+ dev_err(dev, "ADM copp cmd timedout\n");
+ ret = -ETIMEDOUT;
+ } else if (copp->result.status > 0) {
+ dev_err(dev, "DSP returned error[%d]\n",
+ copp->result.status);
+ ret = -EINVAL;
+ }
+
+err:
+ mutex_unlock(&adm->lock);
+ return ret;
+}
+
+static int q6adm_device_close(struct q6adm *adm, struct q6copp *copp,
+ int port_id, int copp_idx)
+{
+ struct apr_pkt close;
+
+ close.hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ close.hdr.pkt_size = sizeof(close);
+ close.hdr.src_port = port_id;
+ close.hdr.dest_port = copp->id;
+ close.hdr.token = port_id << 16 | copp_idx;
+ close.hdr.opcode = ADM_CMD_DEVICE_CLOSE_V5;
+
+ return q6adm_apr_send_copp_pkt(adm, copp, &close, 0);
+}
+
+static struct q6copp *q6adm_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 q6copp *c = NULL;
+ struct q6copp *ret = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+
+ list_for_each_entry(c, &adm->copps_list, node) {
+ if ((port_id == c->afe_port) && (topology == c->topology) &&
+ (mode == c->mode) && (rate == c->rate) &&
+ (bit_width == c->bit_width) && (app_type == c->app_type)) {
+ ret = c;
+ kref_get(&c->refcount);
+ }
+ }
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+
+ return ret;
+}
+
+static int q6adm_device_open(struct q6adm *adm, struct q6copp *copp,
+ int port_id, int path, int topology,
+ int channel_mode, int bit_width, int rate)
+{
+ struct q6adm_cmd_device_open_v5 *open;
+ int afe_port = q6afe_get_port_id(port_id);
+ struct apr_pkt *pkt;
+ void *p;
+ int ret, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*open);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ open = p + APR_HDR_SIZE;
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.src_port = afe_port;
+ pkt->hdr.dest_port = afe_port;
+ pkt->hdr.token = port_id << 16 | copp->copp_idx;
+ pkt->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)
+ goto err;
+
+ ret = q6adm_apr_send_copp_pkt(adm, copp, pkt,
+ ADM_CMDRSP_DEVICE_OPEN_V5);
+
+err:
+ kfree(pkt);
+ return ret;
+}
+
+/**
+ * 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 pointer on success.
+ */
+struct q6copp *q6adm_open(struct device *dev, int port_id, int path, int rate,
+ int channel_mode, int topology, int perf_mode,
+ uint16_t bit_width, int app_type, int acdb_id)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+ struct q6copp *copp;
+ unsigned long flags;
+ int ret = 0;
+
+ if (port_id < 0) {
+ dev_err(dev, "Invalid port_id 0x%x\n", port_id);
+ return ERR_PTR(-EINVAL);
+ }
+
+ copp = q6adm_find_matching_copp(adm, port_id, topology, perf_mode,
+ rate, channel_mode, bit_width, app_type);
+ if (copp) {
+ dev_err(dev, "Found Matching Copp 0x%x\n", copp->copp_idx);
+ return copp;
+ }
+
+ spin_lock_irqsave(&adm->copps_list_lock, flags);
+ copp = q6adm_alloc_copp(adm, port_id);
+ if (IS_ERR_OR_NULL(copp)) {
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+ return ERR_CAST(copp);
+ }
+
+ list_add_tail(&copp->node, &adm->copps_list);
+ spin_unlock_irqrestore(&adm->copps_list_lock, flags);
+
+ kref_init(&copp->refcount);
+ copp->topology = topology;
+ copp->mode = perf_mode;
+ copp->rate = rate;
+ copp->channels = channel_mode;
+ copp->bit_width = bit_width;
+ copp->app_type = app_type;
+
+
+ ret = q6adm_device_open(adm, copp, port_id, path, topology,
+ channel_mode, bit_width, rate);
+ if (ret < 0) {
+ kref_put(&copp->refcount, q6adm_free_copp);
+ return ERR_PTR(ret);
+ }
+
+ return copp;
+}
+EXPORT_SYMBOL_GPL(q6adm_open);
+
+/**
+ * q6adm_get_copp_id() - get copp index
+ *
+ * @copp: Pointer to valid copp
+ *
+ * Return: Will be an negative on error or a valid copp index on success.
+ **/
+int q6adm_get_copp_id(struct q6copp *copp)
+{
+ if (!copp)
+ return -EINVAL;
+
+ return copp->copp_idx;
+}
+EXPORT_SYMBOL_GPL(q6adm_get_copp_id);
+
+/**
+ * q6adm_matrix_map() - Map asm streams and afe ports using payload
+ *
+ * @dev: Pointer to adm child device.
+ * @path: playback or capture path.
+ * @payload_map: map between session id and afe ports.
+ * @perf_mode: Performace mode.
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6adm_matrix_map(struct device *dev, int path,
+ struct route_payload payload_map, int perf_mode)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+ struct q6adm_cmd_matrix_map_routings_v5 *route;
+ struct q6adm_session_map_node_v5 *node;
+ struct apr_pkt *pkt;
+ uint16_t *copps_list;
+ int pkt_size, ret, i, copp_idx;
+ void *matrix_map = NULL;
+ struct q6copp *copp;
+
+ /* Assumes port_ids have already been validated during adm_open */
+ pkt_size = (APR_HDR_SIZE + sizeof(*route) + sizeof(*node) +
+ (sizeof(uint32_t) * payload_map.num_copps));
+
+ matrix_map = kzalloc(pkt_size, GFP_KERNEL);
+ if (!matrix_map)
+ return -ENOMEM;
+
+ pkt = matrix_map;
+ route = matrix_map + APR_HDR_SIZE;
+ node = matrix_map + APR_HDR_SIZE + sizeof(*route);
+ copps_list = matrix_map + APR_HDR_SIZE + sizeof(*route) + sizeof(*node);
+
+ pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD,
+ APR_HDR_LEN(APR_HDR_SIZE),
+ APR_PKT_VER);
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.token = 0;
+ pkt->hdr.opcode = ADM_CMD_MATRIX_MAP_ROUTINGS_V5;
+ route->num_sessions = 1;
+
+ switch (path) {
+ case ADM_PATH_PLAYBACK:
+ route->matrix_id = ADM_MATRIX_ID_AUDIO_RX;
+ break;
+ case ADM_PATH_LIVE_REC:
+ route->matrix_id = ADM_MATRIX_ID_AUDIO_TX;
+ break;
+ default:
+ dev_err(dev, "Wrong path set[%d]\n", path);
+ break;
+ }
+
+ node->session_id = payload_map.session_id;
+ node->num_copps = payload_map.num_copps;
+
+ 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]);
+ kfree(pkt);
+ return -EINVAL;
+ }
+ copp_idx = payload_map.copp_idx[i];
+
+ copp = q6adm_find_copp(adm, port_idx, copp_idx);
+ if (!copp) {
+ kfree(pkt);
+ return -EINVAL;
+ }
+
+ copps_list[i] = copp->id;
+ kref_put(&copp->refcount, q6adm_free_copp);
+ }
+
+ mutex_lock(&adm->lock);
+ adm->result.status = 0;
+ adm->result.opcode = 0;
+
+ ret = apr_send_pkt(adm->apr, pkt);
+ if (ret < 0) {
+ dev_err(dev, "routing for stream %d failed ret %d\n",
+ payload_map.session_id, ret);
+ goto fail_cmd;
+ }
+ ret = wait_event_timeout(adm->matrix_map_wait,
+ adm->result.opcode == pkt->hdr.opcode,
+ msecs_to_jiffies(TIMEOUT_MS));
+ if (!ret) {
+ dev_err(dev, "routing for stream %d failed\n",
+ payload_map.session_id);
+ ret = -ETIMEDOUT;
+ goto fail_cmd;
+ } else if (adm->result.status > 0) {
+ dev_err(dev, "DSP returned error[%d]\n",
+ adm->result.status);
+ ret = -EINVAL;
+ goto fail_cmd;
+ }
+
+fail_cmd:
+ mutex_unlock(&adm->lock);
+ kfree(pkt);
+ return ret;
+}
+EXPORT_SYMBOL_GPL(q6adm_matrix_map);
+
+/**
+ * q6adm_close() - Close adm copp
+ *
+ * @dev: Pointer to adm child device.
+ * @copp: pointer to previously opened copp
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6adm_close(struct device *dev, struct q6copp *copp)
+{
+ struct q6adm *adm = dev_get_drvdata(dev->parent);
+ int ret = 0;
+
+ ret = q6adm_device_close(adm, copp, copp->afe_port, copp->copp_idx);
+ if (ret < 0) {
+ dev_err(adm->dev, "Failed to close copp %d\n", ret);
+ return ret;
+ }
+
+ kref_put(&copp->refcount, q6adm_free_copp);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6adm_close);
+
+static int q6adm_probe(struct apr_device *adev)
+{
+ struct device *dev = &adev->dev;
+ struct device_node *dais_np;
+ struct q6adm *adm;
+
+ adm = devm_kzalloc(&adev->dev, sizeof(*adm), GFP_KERNEL);
+ if (!adm)
+ return -ENOMEM;
+
+ adm->apr = adev;
+ dev_set_drvdata(&adev->dev, adm);
+ adm->dev = dev;
+ q6core_get_svc_api_info(adev->svc_id, &adm->ainfo);
+ mutex_init(&adm->lock);
+ init_waitqueue_head(&adm->matrix_map_wait);
+
+ INIT_LIST_HEAD(&adm->copps_list);
+ spin_lock_init(&adm->copps_list_lock);
+
+ dais_np = of_get_child_by_name(dev->of_node, "routing");
+ if (dais_np) {
+ adm->pdev_routing = of_platform_device_create(dais_np,
+ "q6routing", dev);
+ of_node_put(dais_np);
+ }
+
+ return 0;
+}
+
+static int q6adm_remove(struct apr_device *adev)
+{
+ struct q6adm *adm = dev_get_drvdata(&adev->dev);
+
+ if (adm->pdev_routing)
+ of_platform_device_destroy(&adm->pdev_routing->dev, NULL);
+
+ return 0;
+}
+
+static const struct of_device_id q6adm_device_id[] = {
+ { .compatible = "qcom,q6adm" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6adm_device_id);
+
+static struct apr_driver qcom_q6adm_driver = {
+ .probe = q6adm_probe,
+ .remove = q6adm_remove,
+ .callback = q6adm_callback,
+ .driver = {
+ .name = "qcom-q6adm",
+ .of_match_table = of_match_ptr(q6adm_device_id),
+ },
+};
+
+module_apr_driver(qcom_q6adm_driver);
+MODULE_DESCRIPTION("Q6 Audio Device Manager");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6adm.h b/sound/soc/qcom/qdsp6/q6adm.h
new file mode 100644
index 000000000000..4f56999b7fab
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6adm.h
@@ -0,0 +1,27 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __Q6_ADM_V2_H__
+#define __Q6_ADM_V2_H__
+
+#define ADM_PATH_PLAYBACK 0x1
+#define ADM_PATH_LIVE_REC 0x2
+#define MAX_COPPS_PER_PORT 8
+#define NULL_COPP_TOPOLOGY 0x00010312
+
+/* multiple copp per stream. */
+struct route_payload {
+ int num_copps;
+ int session_id;
+ int copp_idx[MAX_COPPS_PER_PORT];
+ int port_id[MAX_COPPS_PER_PORT];
+};
+
+struct q6copp;
+struct q6copp *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, struct q6copp *copp);
+int q6adm_get_copp_id(struct q6copp *copp);
+int q6adm_matrix_map(struct device *dev, int path,
+ struct route_payload payload_map, int perf_mode);
+
+#endif /* __Q6_ADM_V2_H__ */
--
2.17.0
^ permalink raw reply related [flat|nested] 25+ messages in thread
* [PATCH v9 06/15] ASoC: qdsp6: q6asm: Add q6asm driver
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (4 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 05/15] ASoC: qdsp6: q6adm: Add q6adm driver Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-21 15:47 ` Applied "ASoC: qdsp6: q6asm: Add q6asm driver" to the asoc tree Mark Brown
2018-05-18 12:56 ` [PATCH v9 07/15] ASoC: qdsp6: q6asm: Add support to memory map and unmap Srinivas Kandagatla
` (8 subsequent siblings)
14 siblings, 1 reply; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
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>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6asm.c | 252 ++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6asm.h | 15 +++
4 files changed, 272 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 971127edbc23..941774abd94f 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -53,6 +53,9 @@ config SND_SOC_QDSP6_AFE
config SND_SOC_QDSP6_ADM
tristate
+config SND_SOC_QDSP6_ASM
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
@@ -60,6 +63,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_CORE
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 95cdb3a12694..01d9dcf3375c 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -2,3 +2,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.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..585fcfbada6a
--- /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. All rights reserved.
+// 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_platform.h>
+#include <linux/spinlock.h>
+#include <linux/kref.h>
+#include <linux/of.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/mm.h>
+#include "q6asm.h"
+#include "q6core.h"
+#include "q6dsp-errno.h"
+#include "q6dsp-common.h"
+
+#define ASM_SYNC_IO_MODE 0x0001
+#define ASM_ASYNC_IO_MODE 0x0002
+#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
+#define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
+
+struct q6asm {
+ struct apr_device *adev;
+ struct device *dev;
+ struct q6core_svc_api_info ainfo;
+ wait_queue_head_t mem_wait;
+ struct platform_device *pcmdev;
+ spinlock_t slock;
+ struct audio_client *session[MAX_SESSIONS + 1];
+ struct platform_device *pdev_dais;
+};
+
+struct audio_client {
+ int session;
+ q6asm_cb cb;
+ void *priv;
+ uint32_t io_mode;
+ struct apr_device *adev;
+ struct mutex cmd_lock;
+ spinlock_t lock;
+ struct kref refcount;
+ wait_queue_head_t cmd_wait;
+ struct aprv2_ibasic_rsp_result_t result;
+ int perf_mode;
+ int stream_id;
+ struct q6asm *q6asm;
+ struct device *dev;
+};
+
+static void q6asm_audio_client_release(struct kref *ref)
+{
+ struct audio_client *ac;
+ struct q6asm *a;
+ unsigned long flags;
+
+ ac = container_of(ref, struct audio_client, refcount);
+ a = ac->q6asm;
+
+ spin_lock_irqsave(&a->slock, flags);
+ a->session[ac->session] = NULL;
+ spin_unlock_irqrestore(&a->slock, flags);
+
+ kfree(ac);
+}
+
+/**
+ * q6asm_audio_client_free() - Freee allocated audio client
+ *
+ * @ac: audio client to free
+ */
+void q6asm_audio_client_free(struct audio_client *ac)
+{
+ kref_put(&ac->refcount, q6asm_audio_client_release);
+}
+EXPORT_SYMBOL_GPL(q6asm_audio_client_free);
+
+static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
+ int session_id)
+{
+ struct audio_client *ac = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&a->slock, flags);
+ if ((session_id <= 0) || (session_id > MAX_SESSIONS)) {
+ dev_err(a->dev, "invalid session: %d\n", session_id);
+ goto err;
+ }
+
+ /* check for valid session */
+ if (!a->session[session_id])
+ goto err;
+ else if (a->session[session_id]->session != session_id)
+ goto err;
+
+ ac = a->session[session_id];
+ kref_get(&ac->refcount);
+err:
+ spin_unlock_irqrestore(&a->slock, flags);
+ return ac;
+}
+
+static int q6asm_srvc_callback(struct apr_device *adev,
+ struct apr_resp_pkt *data)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+ struct audio_client *ac = NULL;
+ struct apr_hdr *hdr = &data->hdr;
+ uint32_t sid = 0;
+
+ sid = (hdr->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(hdr->opcode, hdr->token, data->payload, ac->priv);
+
+ kref_put(&ac->refcount, q6asm_audio_client_release);
+
+ return 0;
+}
+
+/**
+ * q6asm_get_session_id() - get session id for audio client
+ *
+ * @c: 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.
+ * @stream_id: stream id
+ * @perf_mode: performace mode for this client
+ *
+ * Return: Will be an error pointer on error or a valid audio client
+ * on success.
+ */
+struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
+ void *priv, int stream_id,
+ int perf_mode)
+{
+ struct q6asm *a = dev_get_drvdata(dev->parent);
+ struct audio_client *ac;
+ unsigned long flags;
+
+ ac = q6asm_get_audio_client(a, stream_id + 1);
+ if (ac) {
+ dev_err(dev, "Audio Client already active\n");
+ return ac;
+ }
+
+ ac = kzalloc(sizeof(*ac), GFP_KERNEL);
+ if (!ac)
+ return ERR_PTR(-ENOMEM);
+
+ spin_lock_irqsave(&a->slock, flags);
+ a->session[stream_id + 1] = ac;
+ spin_unlock_irqrestore(&a->slock, flags);
+ ac->session = stream_id + 1;
+ ac->cb = cb;
+ ac->dev = dev;
+ ac->q6asm = a;
+ ac->priv = priv;
+ ac->io_mode = ASM_SYNC_IO_MODE;
+ ac->perf_mode = perf_mode;
+ /* DSP expects stream id from 1 */
+ ac->stream_id = 1;
+ ac->adev = a->adev;
+ kref_init(&ac->refcount);
+
+ init_waitqueue_head(&ac->cmd_wait);
+ mutex_init(&ac->cmd_lock);
+ spin_lock_init(&ac->lock);
+
+ return ac;
+}
+EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
+
+
+static int q6asm_probe(struct apr_device *adev)
+{
+ struct device *dev = &adev->dev;
+ struct device_node *dais_np;
+ struct q6asm *q6asm;
+
+ q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL);
+ if (!q6asm)
+ return -ENOMEM;
+
+ q6core_get_svc_api_info(adev->svc_id, &q6asm->ainfo);
+
+ q6asm->dev = dev;
+ q6asm->adev = adev;
+ init_waitqueue_head(&q6asm->mem_wait);
+ spin_lock_init(&q6asm->slock);
+ dev_set_drvdata(dev, q6asm);
+
+ dais_np = of_get_child_by_name(dev->of_node, "dais");
+ if (dais_np) {
+ q6asm->pdev_dais = of_platform_device_create(dais_np,
+ "q6asm-dai", dev);
+ of_node_put(dais_np);
+ }
+
+ return 0;
+}
+
+static int q6asm_remove(struct apr_device *adev)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+
+ if (q6asm->pdev_dais)
+ of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL);
+
+ return 0;
+}
+static const struct of_device_id q6asm_device_id[] = {
+ { .compatible = "qcom,q6asm" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6asm_device_id);
+
+static struct apr_driver qcom_q6asm_driver = {
+ .probe = q6asm_probe,
+ .remove = q6asm_remove,
+ .callback = q6asm_srvc_callback,
+ .driver = {
+ .name = "qcom-q6asm",
+ .of_match_table = of_match_ptr(q6asm_device_id),
+ },
+};
+
+module_apr_driver(qcom_q6asm_driver);
+MODULE_DESCRIPTION("Q6 Audio Stream Manager driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
new file mode 100644
index 000000000000..b7816e6384e7
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -0,0 +1,15 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __Q6_ASM_H__
+#define __Q6_ASM_H__
+
+#define MAX_SESSIONS 8
+
+typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
+ void *payload, void *priv);
+struct audio_client;
+struct audio_client *q6asm_audio_client_alloc(struct device *dev,
+ q6asm_cb cb, void *priv,
+ int session_id, int perf_mode);
+void q6asm_audio_client_free(struct audio_client *ac);
+int q6asm_get_session_id(struct audio_client *ac);
+#endif /* __Q6_ASM_H__ */
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* Applied "ASoC: qdsp6: q6asm: Add q6asm driver" to the asoc tree
2018-05-18 12:56 ` [PATCH v9 06/15] ASoC: qdsp6: q6asm: Add q6asm driver Srinivas Kandagatla
@ 2018-05-21 15:47 ` Mark Brown
0 siblings, 0 replies; 25+ messages in thread
From: Mark Brown @ 2018-05-21 15:47 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, broonie, linux-arm-msm, alsa-devel, bgoswami,
mark.rutland, devicetree, rohkumar, gregkh, plai, lgirdwood,
tiwai, david.brown, robh+dt, linux-arm-kernel, spatakok,
linux-kernel, alsa-devel
The patch
ASoC: qdsp6: q6asm: Add q6asm driver
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From a13e872314ae97d3ec8c70de6a01d5904182b049 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:56:01 +0100
Subject: [PATCH] ASoC: qdsp6: q6asm: Add q6asm driver
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>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6asm.c | 252 ++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6asm.h | 15 ++
4 files changed, 272 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 971127edbc23..941774abd94f 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -53,6 +53,9 @@ config SND_SOC_QDSP6_AFE
config SND_SOC_QDSP6_ADM
tristate
+config SND_SOC_QDSP6_ASM
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
@@ -60,6 +63,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_CORE
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 95cdb3a12694..01d9dcf3375c 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -2,3 +2,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
obj-$(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..585fcfbada6a
--- /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. All rights reserved.
+// 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_platform.h>
+#include <linux/spinlock.h>
+#include <linux/kref.h>
+#include <linux/of.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/mm.h>
+#include "q6asm.h"
+#include "q6core.h"
+#include "q6dsp-errno.h"
+#include "q6dsp-common.h"
+
+#define ASM_SYNC_IO_MODE 0x0001
+#define ASM_ASYNC_IO_MODE 0x0002
+#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
+#define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
+
+struct q6asm {
+ struct apr_device *adev;
+ struct device *dev;
+ struct q6core_svc_api_info ainfo;
+ wait_queue_head_t mem_wait;
+ struct platform_device *pcmdev;
+ spinlock_t slock;
+ struct audio_client *session[MAX_SESSIONS + 1];
+ struct platform_device *pdev_dais;
+};
+
+struct audio_client {
+ int session;
+ q6asm_cb cb;
+ void *priv;
+ uint32_t io_mode;
+ struct apr_device *adev;
+ struct mutex cmd_lock;
+ spinlock_t lock;
+ struct kref refcount;
+ wait_queue_head_t cmd_wait;
+ struct aprv2_ibasic_rsp_result_t result;
+ int perf_mode;
+ int stream_id;
+ struct q6asm *q6asm;
+ struct device *dev;
+};
+
+static void q6asm_audio_client_release(struct kref *ref)
+{
+ struct audio_client *ac;
+ struct q6asm *a;
+ unsigned long flags;
+
+ ac = container_of(ref, struct audio_client, refcount);
+ a = ac->q6asm;
+
+ spin_lock_irqsave(&a->slock, flags);
+ a->session[ac->session] = NULL;
+ spin_unlock_irqrestore(&a->slock, flags);
+
+ kfree(ac);
+}
+
+/**
+ * q6asm_audio_client_free() - Freee allocated audio client
+ *
+ * @ac: audio client to free
+ */
+void q6asm_audio_client_free(struct audio_client *ac)
+{
+ kref_put(&ac->refcount, q6asm_audio_client_release);
+}
+EXPORT_SYMBOL_GPL(q6asm_audio_client_free);
+
+static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
+ int session_id)
+{
+ struct audio_client *ac = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&a->slock, flags);
+ if ((session_id <= 0) || (session_id > MAX_SESSIONS)) {
+ dev_err(a->dev, "invalid session: %d\n", session_id);
+ goto err;
+ }
+
+ /* check for valid session */
+ if (!a->session[session_id])
+ goto err;
+ else if (a->session[session_id]->session != session_id)
+ goto err;
+
+ ac = a->session[session_id];
+ kref_get(&ac->refcount);
+err:
+ spin_unlock_irqrestore(&a->slock, flags);
+ return ac;
+}
+
+static int q6asm_srvc_callback(struct apr_device *adev,
+ struct apr_resp_pkt *data)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+ struct audio_client *ac = NULL;
+ struct apr_hdr *hdr = &data->hdr;
+ uint32_t sid = 0;
+
+ sid = (hdr->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(hdr->opcode, hdr->token, data->payload, ac->priv);
+
+ kref_put(&ac->refcount, q6asm_audio_client_release);
+
+ return 0;
+}
+
+/**
+ * q6asm_get_session_id() - get session id for audio client
+ *
+ * @c: 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.
+ * @stream_id: stream id
+ * @perf_mode: performace mode for this client
+ *
+ * Return: Will be an error pointer on error or a valid audio client
+ * on success.
+ */
+struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
+ void *priv, int stream_id,
+ int perf_mode)
+{
+ struct q6asm *a = dev_get_drvdata(dev->parent);
+ struct audio_client *ac;
+ unsigned long flags;
+
+ ac = q6asm_get_audio_client(a, stream_id + 1);
+ if (ac) {
+ dev_err(dev, "Audio Client already active\n");
+ return ac;
+ }
+
+ ac = kzalloc(sizeof(*ac), GFP_KERNEL);
+ if (!ac)
+ return ERR_PTR(-ENOMEM);
+
+ spin_lock_irqsave(&a->slock, flags);
+ a->session[stream_id + 1] = ac;
+ spin_unlock_irqrestore(&a->slock, flags);
+ ac->session = stream_id + 1;
+ ac->cb = cb;
+ ac->dev = dev;
+ ac->q6asm = a;
+ ac->priv = priv;
+ ac->io_mode = ASM_SYNC_IO_MODE;
+ ac->perf_mode = perf_mode;
+ /* DSP expects stream id from 1 */
+ ac->stream_id = 1;
+ ac->adev = a->adev;
+ kref_init(&ac->refcount);
+
+ init_waitqueue_head(&ac->cmd_wait);
+ mutex_init(&ac->cmd_lock);
+ spin_lock_init(&ac->lock);
+
+ return ac;
+}
+EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
+
+
+static int q6asm_probe(struct apr_device *adev)
+{
+ struct device *dev = &adev->dev;
+ struct device_node *dais_np;
+ struct q6asm *q6asm;
+
+ q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL);
+ if (!q6asm)
+ return -ENOMEM;
+
+ q6core_get_svc_api_info(adev->svc_id, &q6asm->ainfo);
+
+ q6asm->dev = dev;
+ q6asm->adev = adev;
+ init_waitqueue_head(&q6asm->mem_wait);
+ spin_lock_init(&q6asm->slock);
+ dev_set_drvdata(dev, q6asm);
+
+ dais_np = of_get_child_by_name(dev->of_node, "dais");
+ if (dais_np) {
+ q6asm->pdev_dais = of_platform_device_create(dais_np,
+ "q6asm-dai", dev);
+ of_node_put(dais_np);
+ }
+
+ return 0;
+}
+
+static int q6asm_remove(struct apr_device *adev)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+
+ if (q6asm->pdev_dais)
+ of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL);
+
+ return 0;
+}
+static const struct of_device_id q6asm_device_id[] = {
+ { .compatible = "qcom,q6asm" },
+ {},
+};
+MODULE_DEVICE_TABLE(of, q6asm_device_id);
+
+static struct apr_driver qcom_q6asm_driver = {
+ .probe = q6asm_probe,
+ .remove = q6asm_remove,
+ .callback = q6asm_srvc_callback,
+ .driver = {
+ .name = "qcom-q6asm",
+ .of_match_table = of_match_ptr(q6asm_device_id),
+ },
+};
+
+module_apr_driver(qcom_q6asm_driver);
+MODULE_DESCRIPTION("Q6 Audio Stream Manager driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
new file mode 100644
index 000000000000..b7816e6384e7
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -0,0 +1,15 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef __Q6_ASM_H__
+#define __Q6_ASM_H__
+
+#define MAX_SESSIONS 8
+
+typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
+ void *payload, void *priv);
+struct audio_client;
+struct audio_client *q6asm_audio_client_alloc(struct device *dev,
+ q6asm_cb cb, void *priv,
+ int session_id, int perf_mode);
+void q6asm_audio_client_free(struct audio_client *ac);
+int q6asm_get_session_id(struct audio_client *ac);
+#endif /* __Q6_ASM_H__ */
--
2.17.0
^ permalink raw reply related [flat|nested] 25+ messages in thread
* [PATCH v9 07/15] ASoC: qdsp6: q6asm: Add support to memory map and unmap
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (5 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 06/15] ASoC: qdsp6: q6asm: Add q6asm driver Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-21 15:47 ` Applied "ASoC: qdsp6: q6asm: Add support to memory map and unmap" to the asoc tree Mark Brown
2018-05-18 12:56 ` [PATCH v9 08/15] ASoC: qdsp6: q6asm: Add support to audio stream apis Srinivas Kandagatla
` (7 subsequent siblings)
14 siblings, 1 reply; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
This patch adds support to memory map and unmap regions commands in
q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/qdsp6/q6asm.c | 347 +++++++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6asm.h | 5 +
2 files changed, 352 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 585fcfbada6a..a20d243ed10a 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -19,10 +19,44 @@
#include "q6dsp-errno.h"
#include "q6dsp-common.h"
+#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
+#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
+#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
+
#define ASM_SYNC_IO_MODE 0x0001
#define ASM_ASYNC_IO_MODE 0x0002
#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
#define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
+#define ASM_SHIFT_GAPLESS_MODE_FLAG 31
+#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3
+
+struct avs_cmd_shared_mem_map_regions {
+ u16 mem_pool_id;
+ u16 num_regions;
+ u32 property_flag;
+} __packed;
+
+struct avs_shared_map_region_payload {
+ u32 shm_addr_lsw;
+ u32 shm_addr_msw;
+ u32 mem_size_bytes;
+} __packed;
+
+struct avs_cmd_shared_mem_unmap_regions {
+ u32 mem_map_handle;
+} __packed;
+
+struct audio_buffer {
+ phys_addr_t phys;
+ uint32_t size; /* size of buffer */
+};
+
+struct audio_port_data {
+ struct audio_buffer *buf;
+ uint32_t num_periods;
+ uint32_t dsp_buf;
+ uint32_t mem_map_handle;
+};
struct q6asm {
struct apr_device *adev;
@@ -44,6 +78,8 @@ struct audio_client {
struct mutex cmd_lock;
spinlock_t lock;
struct kref refcount;
+ /* idx:1 out port, 0: in port */
+ struct audio_port_data port[2];
wait_queue_head_t cmd_wait;
struct aprv2_ibasic_rsp_result_t result;
int perf_mode;
@@ -52,6 +88,275 @@ struct audio_client {
struct device *dev;
};
+static inline void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, bool cmd_flg,
+ uint32_t stream_id)
+{
+ hdr->hdr_field = APR_SEQ_CMD_HDR_FIELD;
+ hdr->src_port = ((ac->session << 8) & 0xFF00) | (stream_id);
+ hdr->dest_port = ((ac->session << 8) & 0xFF00) | (stream_id);
+ hdr->pkt_size = pkt_size;
+ if (cmd_flg)
+ hdr->token = ac->session;
+}
+
+static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,
+ struct apr_pkt *pkt, uint32_t rsp_opcode)
+{
+ struct apr_hdr *hdr = &pkt->hdr;
+ int rc;
+
+ mutex_lock(&ac->cmd_lock);
+ ac->result.opcode = 0;
+ ac->result.status = 0;
+ rc = apr_send_pkt(a->adev, pkt);
+ if (rc < 0)
+ goto err;
+
+ if (rsp_opcode)
+ rc = wait_event_timeout(a->mem_wait,
+ (ac->result.opcode == hdr->opcode) ||
+ (ac->result.opcode == rsp_opcode),
+ 5 * HZ);
+ else
+ rc = wait_event_timeout(a->mem_wait,
+ (ac->result.opcode == hdr->opcode),
+ 5 * HZ);
+
+ if (!rc) {
+ dev_err(a->dev, "CMD timeout\n");
+ rc = -ETIMEDOUT;
+ } else if (ac->result.status > 0) {
+ dev_err(a->dev, "DSP returned error[%x]\n",
+ ac->result.status);
+ rc = -EINVAL;
+ }
+
+err:
+ mutex_unlock(&ac->cmd_lock);
+ return rc;
+}
+
+static int __q6asm_memory_unmap(struct audio_client *ac,
+ phys_addr_t buf_add, int dir)
+{
+ struct avs_cmd_shared_mem_unmap_regions *mem_unmap;
+ struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+ struct apr_pkt *pkt;
+ int rc, pkt_size;
+ void *p;
+
+ if (ac->port[dir].mem_map_handle == 0) {
+ dev_err(ac->dev, "invalid mem handle\n");
+ return -EINVAL;
+ }
+
+ pkt_size = APR_HDR_SIZE + sizeof(*mem_unmap);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ mem_unmap = p + APR_HDR_SIZE;
+
+ pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.token = ((ac->session << 8) | dir);
+
+ pkt->hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
+ mem_unmap->mem_map_handle = ac->port[dir].mem_map_handle;
+
+ rc = q6asm_apr_send_session_pkt(a, ac, pkt, 0);
+ if (rc < 0) {
+ kfree(pkt);
+ return rc;
+ }
+
+ ac->port[dir].mem_map_handle = 0;
+
+ kfree(pkt);
+ return 0;
+}
+
+
+static void q6asm_audio_client_free_buf(struct audio_client *ac,
+ struct audio_port_data *port)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port->num_periods = 0;
+ kfree(port->buf);
+ port->buf = NULL;
+ spin_unlock_irqrestore(&ac->lock, flags);
+}
+
+/**
+ * q6asm_unmap_memory_regions() - unmap memory regions in the dsp.
+ *
+ * @dir: direction of audio stream
+ * @ac: audio client instanace
+ *
+ * Return: Will be an negative value on failure or zero on success
+ */
+int q6asm_unmap_memory_regions(unsigned int dir, struct audio_client *ac)
+{
+ struct audio_port_data *port;
+ int cnt = 0;
+ int rc = 0;
+
+ port = &ac->port[dir];
+ if (!port->buf) {
+ rc = -EINVAL;
+ goto err;
+ }
+
+ cnt = port->num_periods - 1;
+ if (cnt >= 0) {
+ rc = __q6asm_memory_unmap(ac, port->buf[dir].phys, dir);
+ if (rc < 0) {
+ dev_err(ac->dev, "%s: Memory_unmap_regions failed %d\n",
+ __func__, rc);
+ goto err;
+ }
+ }
+
+ q6asm_audio_client_free_buf(ac, port);
+
+err:
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_unmap_memory_regions);
+
+static int __q6asm_memory_map_regions(struct audio_client *ac, int dir,
+ size_t period_sz, unsigned int periods,
+ bool is_contiguous)
+{
+ struct avs_cmd_shared_mem_map_regions *cmd = NULL;
+ struct avs_shared_map_region_payload *mregions = NULL;
+ struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+ struct audio_port_data *port = NULL;
+ struct audio_buffer *ab = NULL;
+ struct apr_pkt *pkt;
+ void *p;
+ unsigned long flags;
+ uint32_t num_regions, buf_sz;
+ int rc, i, pkt_size;
+
+ if (is_contiguous) {
+ num_regions = 1;
+ buf_sz = period_sz * periods;
+ } else {
+ buf_sz = period_sz;
+ num_regions = periods;
+ }
+
+ /* DSP expects size should be aligned to 4K */
+ buf_sz = ALIGN(buf_sz, 4096);
+
+ pkt_size = APR_HDR_SIZE + sizeof(*cmd) +
+ (sizeof(*mregions) * num_regions);
+
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ cmd = p + APR_HDR_SIZE;
+ mregions = p + APR_HDR_SIZE + sizeof(*cmd);
+
+ pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.token = ((ac->session << 8) | dir);
+ pkt->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
+
+ cmd->mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
+ cmd->num_regions = num_regions;
+ cmd->property_flag = 0x00;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[dir];
+
+ for (i = 0; i < num_regions; i++) {
+ ab = &port->buf[i];
+ mregions->shm_addr_lsw = lower_32_bits(ab->phys);
+ mregions->shm_addr_msw = upper_32_bits(ab->phys);
+ mregions->mem_size_bytes = buf_sz;
+ ++mregions;
+ }
+ spin_unlock_irqrestore(&ac->lock, flags);
+
+ rc = q6asm_apr_send_session_pkt(a, ac, pkt,
+ ASM_CMDRSP_SHARED_MEM_MAP_REGIONS);
+
+ kfree(pkt);
+
+ return rc;
+}
+
+/**
+ * q6asm_map_memory_regions() - map memory regions in the dsp.
+ *
+ * @dir: direction of audio stream
+ * @ac: audio client instanace
+ * @phys: physcial address that needs mapping.
+ * @period_sz: audio period size
+ * @periods: number of periods
+ *
+ * Return: Will be an negative value on failure or zero on success
+ */
+int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac,
+ phys_addr_t phys,
+ size_t period_sz, unsigned int periods)
+{
+ struct audio_buffer *buf;
+ unsigned long flags;
+ int cnt;
+ int rc;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ if (ac->port[dir].buf) {
+ dev_err(ac->dev, "Buffer already allocated\n");
+ spin_unlock_irqrestore(&ac->lock, flags);
+ return 0;
+ }
+
+ buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC);
+ if (!buf) {
+ spin_unlock_irqrestore(&ac->lock, flags);
+ return -ENOMEM;
+ }
+
+
+ ac->port[dir].buf = buf;
+
+ buf[0].phys = phys;
+ buf[0].size = period_sz;
+
+ for (cnt = 1; cnt < periods; cnt++) {
+ if (period_sz > 0) {
+ buf[cnt].phys = buf[0].phys + (cnt * period_sz);
+ buf[cnt].size = period_sz;
+ }
+ }
+ ac->port[dir].num_periods = periods;
+
+ spin_unlock_irqrestore(&ac->lock, flags);
+
+ rc = __q6asm_memory_map_regions(ac, dir, period_sz, periods, 1);
+ if (rc < 0) {
+ dev_err(ac->dev, "Memory_map_regions failed\n");
+ q6asm_audio_client_free_buf(ac, &ac->port[dir]);
+ }
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_map_memory_regions);
+
static void q6asm_audio_client_release(struct kref *ref)
{
struct audio_client *ac;
@@ -108,9 +413,13 @@ static int q6asm_srvc_callback(struct apr_device *adev,
struct apr_resp_pkt *data)
{
struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *result;
+ struct audio_port_data *port;
struct audio_client *ac = NULL;
struct apr_hdr *hdr = &data->hdr;
+ struct q6asm *a;
uint32_t sid = 0;
+ uint32_t dir = 0;
sid = (hdr->token >> 8) & 0x0F;
ac = q6asm_get_audio_client(q6asm, sid);
@@ -119,9 +428,47 @@ static int q6asm_srvc_callback(struct apr_device *adev,
return 0;
}
+ a = dev_get_drvdata(ac->dev->parent);
+ dir = (hdr->token & 0x0F);
+ port = &ac->port[dir];
+ result = data->payload;
+
+ switch (hdr->opcode) {
+ case APR_BASIC_RSP_RESULT:
+ switch (result->opcode) {
+ case ASM_CMD_SHARED_MEM_MAP_REGIONS:
+ case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
+ ac->result = *result;
+ wake_up(&a->mem_wait);
+ break;
+ default:
+ dev_err(&adev->dev, "command[0x%x] not expecting rsp\n",
+ result->opcode);
+ break;
+ }
+ goto done;
+ case ASM_CMDRSP_SHARED_MEM_MAP_REGIONS:
+ ac->result.status = 0;
+ ac->result.opcode = hdr->opcode;
+ port->mem_map_handle = result->opcode;
+ wake_up(&a->mem_wait);
+ break;
+ case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
+ ac->result.opcode = hdr->opcode;
+ ac->result.status = 0;
+ port->mem_map_handle = 0;
+ wake_up(&a->mem_wait);
+ break;
+ default:
+ dev_dbg(&adev->dev, "command[0x%x]success [0x%x]\n",
+ result->opcode, result->status);
+ break;
+ }
+
if (ac->cb)
ac->cb(hdr->opcode, hdr->token, data->payload, ac->priv);
+done:
kref_put(&ac->refcount, q6asm_audio_client_release);
return 0;
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index b7816e6384e7..8c317b7b63c3 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -12,4 +12,9 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
int session_id, int perf_mode);
void q6asm_audio_client_free(struct audio_client *ac);
int q6asm_get_session_id(struct audio_client *ac);
+int q6asm_map_memory_regions(unsigned int dir,
+ struct audio_client *ac,
+ phys_addr_t phys,
+ size_t bufsz, unsigned int bufcnt);
+int q6asm_unmap_memory_regions(unsigned int dir, struct audio_client *ac);
#endif /* __Q6_ASM_H__ */
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* Applied "ASoC: qdsp6: q6asm: Add support to memory map and unmap" to the asoc tree
2018-05-18 12:56 ` [PATCH v9 07/15] ASoC: qdsp6: q6asm: Add support to memory map and unmap Srinivas Kandagatla
@ 2018-05-21 15:47 ` Mark Brown
0 siblings, 0 replies; 25+ messages in thread
From: Mark Brown @ 2018-05-21 15:47 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, broonie, linux-arm-msm, alsa-devel, bgoswami,
mark.rutland, devicetree, rohkumar, gregkh, plai, lgirdwood,
tiwai, david.brown, robh+dt, linux-arm-kernel, spatakok,
linux-kernel, alsa-devel
The patch
ASoC: qdsp6: q6asm: Add support to memory map and unmap
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From a2a5d30218fdf6ad869248ef46bf35b912dc1686 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:56:02 +0100
Subject: [PATCH] ASoC: qdsp6: q6asm: Add support to memory map and unmap
This patch adds support to memory map and unmap regions commands in
q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6asm.c | 347 +++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6asm.h | 5 +
2 files changed, 352 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 585fcfbada6a..a20d243ed10a 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -19,10 +19,44 @@
#include "q6dsp-errno.h"
#include "q6dsp-common.h"
+#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
+#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
+#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
+
#define ASM_SYNC_IO_MODE 0x0001
#define ASM_ASYNC_IO_MODE 0x0002
#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
#define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
+#define ASM_SHIFT_GAPLESS_MODE_FLAG 31
+#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3
+
+struct avs_cmd_shared_mem_map_regions {
+ u16 mem_pool_id;
+ u16 num_regions;
+ u32 property_flag;
+} __packed;
+
+struct avs_shared_map_region_payload {
+ u32 shm_addr_lsw;
+ u32 shm_addr_msw;
+ u32 mem_size_bytes;
+} __packed;
+
+struct avs_cmd_shared_mem_unmap_regions {
+ u32 mem_map_handle;
+} __packed;
+
+struct audio_buffer {
+ phys_addr_t phys;
+ uint32_t size; /* size of buffer */
+};
+
+struct audio_port_data {
+ struct audio_buffer *buf;
+ uint32_t num_periods;
+ uint32_t dsp_buf;
+ uint32_t mem_map_handle;
+};
struct q6asm {
struct apr_device *adev;
@@ -44,6 +78,8 @@ struct audio_client {
struct mutex cmd_lock;
spinlock_t lock;
struct kref refcount;
+ /* idx:1 out port, 0: in port */
+ struct audio_port_data port[2];
wait_queue_head_t cmd_wait;
struct aprv2_ibasic_rsp_result_t result;
int perf_mode;
@@ -52,6 +88,275 @@ struct audio_client {
struct device *dev;
};
+static inline void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
+ uint32_t pkt_size, bool cmd_flg,
+ uint32_t stream_id)
+{
+ hdr->hdr_field = APR_SEQ_CMD_HDR_FIELD;
+ hdr->src_port = ((ac->session << 8) & 0xFF00) | (stream_id);
+ hdr->dest_port = ((ac->session << 8) & 0xFF00) | (stream_id);
+ hdr->pkt_size = pkt_size;
+ if (cmd_flg)
+ hdr->token = ac->session;
+}
+
+static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,
+ struct apr_pkt *pkt, uint32_t rsp_opcode)
+{
+ struct apr_hdr *hdr = &pkt->hdr;
+ int rc;
+
+ mutex_lock(&ac->cmd_lock);
+ ac->result.opcode = 0;
+ ac->result.status = 0;
+ rc = apr_send_pkt(a->adev, pkt);
+ if (rc < 0)
+ goto err;
+
+ if (rsp_opcode)
+ rc = wait_event_timeout(a->mem_wait,
+ (ac->result.opcode == hdr->opcode) ||
+ (ac->result.opcode == rsp_opcode),
+ 5 * HZ);
+ else
+ rc = wait_event_timeout(a->mem_wait,
+ (ac->result.opcode == hdr->opcode),
+ 5 * HZ);
+
+ if (!rc) {
+ dev_err(a->dev, "CMD timeout\n");
+ rc = -ETIMEDOUT;
+ } else if (ac->result.status > 0) {
+ dev_err(a->dev, "DSP returned error[%x]\n",
+ ac->result.status);
+ rc = -EINVAL;
+ }
+
+err:
+ mutex_unlock(&ac->cmd_lock);
+ return rc;
+}
+
+static int __q6asm_memory_unmap(struct audio_client *ac,
+ phys_addr_t buf_add, int dir)
+{
+ struct avs_cmd_shared_mem_unmap_regions *mem_unmap;
+ struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+ struct apr_pkt *pkt;
+ int rc, pkt_size;
+ void *p;
+
+ if (ac->port[dir].mem_map_handle == 0) {
+ dev_err(ac->dev, "invalid mem handle\n");
+ return -EINVAL;
+ }
+
+ pkt_size = APR_HDR_SIZE + sizeof(*mem_unmap);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ mem_unmap = p + APR_HDR_SIZE;
+
+ pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.token = ((ac->session << 8) | dir);
+
+ pkt->hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
+ mem_unmap->mem_map_handle = ac->port[dir].mem_map_handle;
+
+ rc = q6asm_apr_send_session_pkt(a, ac, pkt, 0);
+ if (rc < 0) {
+ kfree(pkt);
+ return rc;
+ }
+
+ ac->port[dir].mem_map_handle = 0;
+
+ kfree(pkt);
+ return 0;
+}
+
+
+static void q6asm_audio_client_free_buf(struct audio_client *ac,
+ struct audio_port_data *port)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port->num_periods = 0;
+ kfree(port->buf);
+ port->buf = NULL;
+ spin_unlock_irqrestore(&ac->lock, flags);
+}
+
+/**
+ * q6asm_unmap_memory_regions() - unmap memory regions in the dsp.
+ *
+ * @dir: direction of audio stream
+ * @ac: audio client instanace
+ *
+ * Return: Will be an negative value on failure or zero on success
+ */
+int q6asm_unmap_memory_regions(unsigned int dir, struct audio_client *ac)
+{
+ struct audio_port_data *port;
+ int cnt = 0;
+ int rc = 0;
+
+ port = &ac->port[dir];
+ if (!port->buf) {
+ rc = -EINVAL;
+ goto err;
+ }
+
+ cnt = port->num_periods - 1;
+ if (cnt >= 0) {
+ rc = __q6asm_memory_unmap(ac, port->buf[dir].phys, dir);
+ if (rc < 0) {
+ dev_err(ac->dev, "%s: Memory_unmap_regions failed %d\n",
+ __func__, rc);
+ goto err;
+ }
+ }
+
+ q6asm_audio_client_free_buf(ac, port);
+
+err:
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_unmap_memory_regions);
+
+static int __q6asm_memory_map_regions(struct audio_client *ac, int dir,
+ size_t period_sz, unsigned int periods,
+ bool is_contiguous)
+{
+ struct avs_cmd_shared_mem_map_regions *cmd = NULL;
+ struct avs_shared_map_region_payload *mregions = NULL;
+ struct q6asm *a = dev_get_drvdata(ac->dev->parent);
+ struct audio_port_data *port = NULL;
+ struct audio_buffer *ab = NULL;
+ struct apr_pkt *pkt;
+ void *p;
+ unsigned long flags;
+ uint32_t num_regions, buf_sz;
+ int rc, i, pkt_size;
+
+ if (is_contiguous) {
+ num_regions = 1;
+ buf_sz = period_sz * periods;
+ } else {
+ buf_sz = period_sz;
+ num_regions = periods;
+ }
+
+ /* DSP expects size should be aligned to 4K */
+ buf_sz = ALIGN(buf_sz, 4096);
+
+ pkt_size = APR_HDR_SIZE + sizeof(*cmd) +
+ (sizeof(*mregions) * num_regions);
+
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ cmd = p + APR_HDR_SIZE;
+ mregions = p + APR_HDR_SIZE + sizeof(*cmd);
+
+ pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
+ pkt->hdr.src_port = 0;
+ pkt->hdr.dest_port = 0;
+ pkt->hdr.pkt_size = pkt_size;
+ pkt->hdr.token = ((ac->session << 8) | dir);
+ pkt->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
+
+ cmd->mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
+ cmd->num_regions = num_regions;
+ cmd->property_flag = 0x00;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[dir];
+
+ for (i = 0; i < num_regions; i++) {
+ ab = &port->buf[i];
+ mregions->shm_addr_lsw = lower_32_bits(ab->phys);
+ mregions->shm_addr_msw = upper_32_bits(ab->phys);
+ mregions->mem_size_bytes = buf_sz;
+ ++mregions;
+ }
+ spin_unlock_irqrestore(&ac->lock, flags);
+
+ rc = q6asm_apr_send_session_pkt(a, ac, pkt,
+ ASM_CMDRSP_SHARED_MEM_MAP_REGIONS);
+
+ kfree(pkt);
+
+ return rc;
+}
+
+/**
+ * q6asm_map_memory_regions() - map memory regions in the dsp.
+ *
+ * @dir: direction of audio stream
+ * @ac: audio client instanace
+ * @phys: physcial address that needs mapping.
+ * @period_sz: audio period size
+ * @periods: number of periods
+ *
+ * Return: Will be an negative value on failure or zero on success
+ */
+int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac,
+ phys_addr_t phys,
+ size_t period_sz, unsigned int periods)
+{
+ struct audio_buffer *buf;
+ unsigned long flags;
+ int cnt;
+ int rc;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ if (ac->port[dir].buf) {
+ dev_err(ac->dev, "Buffer already allocated\n");
+ spin_unlock_irqrestore(&ac->lock, flags);
+ return 0;
+ }
+
+ buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC);
+ if (!buf) {
+ spin_unlock_irqrestore(&ac->lock, flags);
+ return -ENOMEM;
+ }
+
+
+ ac->port[dir].buf = buf;
+
+ buf[0].phys = phys;
+ buf[0].size = period_sz;
+
+ for (cnt = 1; cnt < periods; cnt++) {
+ if (period_sz > 0) {
+ buf[cnt].phys = buf[0].phys + (cnt * period_sz);
+ buf[cnt].size = period_sz;
+ }
+ }
+ ac->port[dir].num_periods = periods;
+
+ spin_unlock_irqrestore(&ac->lock, flags);
+
+ rc = __q6asm_memory_map_regions(ac, dir, period_sz, periods, 1);
+ if (rc < 0) {
+ dev_err(ac->dev, "Memory_map_regions failed\n");
+ q6asm_audio_client_free_buf(ac, &ac->port[dir]);
+ }
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_map_memory_regions);
+
static void q6asm_audio_client_release(struct kref *ref)
{
struct audio_client *ac;
@@ -108,9 +413,13 @@ static int q6asm_srvc_callback(struct apr_device *adev,
struct apr_resp_pkt *data)
{
struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *result;
+ struct audio_port_data *port;
struct audio_client *ac = NULL;
struct apr_hdr *hdr = &data->hdr;
+ struct q6asm *a;
uint32_t sid = 0;
+ uint32_t dir = 0;
sid = (hdr->token >> 8) & 0x0F;
ac = q6asm_get_audio_client(q6asm, sid);
@@ -119,9 +428,47 @@ static int q6asm_srvc_callback(struct apr_device *adev,
return 0;
}
+ a = dev_get_drvdata(ac->dev->parent);
+ dir = (hdr->token & 0x0F);
+ port = &ac->port[dir];
+ result = data->payload;
+
+ switch (hdr->opcode) {
+ case APR_BASIC_RSP_RESULT:
+ switch (result->opcode) {
+ case ASM_CMD_SHARED_MEM_MAP_REGIONS:
+ case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
+ ac->result = *result;
+ wake_up(&a->mem_wait);
+ break;
+ default:
+ dev_err(&adev->dev, "command[0x%x] not expecting rsp\n",
+ result->opcode);
+ break;
+ }
+ goto done;
+ case ASM_CMDRSP_SHARED_MEM_MAP_REGIONS:
+ ac->result.status = 0;
+ ac->result.opcode = hdr->opcode;
+ port->mem_map_handle = result->opcode;
+ wake_up(&a->mem_wait);
+ break;
+ case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
+ ac->result.opcode = hdr->opcode;
+ ac->result.status = 0;
+ port->mem_map_handle = 0;
+ wake_up(&a->mem_wait);
+ break;
+ default:
+ dev_dbg(&adev->dev, "command[0x%x]success [0x%x]\n",
+ result->opcode, result->status);
+ break;
+ }
+
if (ac->cb)
ac->cb(hdr->opcode, hdr->token, data->payload, ac->priv);
+done:
kref_put(&ac->refcount, q6asm_audio_client_release);
return 0;
diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h
index b7816e6384e7..8c317b7b63c3 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -12,4 +12,9 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
int session_id, int perf_mode);
void q6asm_audio_client_free(struct audio_client *ac);
int q6asm_get_session_id(struct audio_client *ac);
+int q6asm_map_memory_regions(unsigned int dir,
+ struct audio_client *ac,
+ phys_addr_t phys,
+ size_t bufsz, unsigned int bufcnt);
+int q6asm_unmap_memory_regions(unsigned int dir, struct audio_client *ac);
#endif /* __Q6_ASM_H__ */
--
2.17.0
^ permalink raw reply related [flat|nested] 25+ messages in thread
* [PATCH v9 08/15] ASoC: qdsp6: q6asm: Add support to audio stream apis
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (6 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 07/15] ASoC: qdsp6: q6asm: Add support to memory map and unmap Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-21 15:47 ` Applied "ASoC: qdsp6: q6asm: Add support to audio stream apis" to the asoc tree Mark Brown
2018-05-18 12:56 ` [PATCH v9 09/15] ASoC: qdsp6: q6routing: Add q6routing driver Srinivas Kandagatla
` (6 subsequent siblings)
14 siblings, 1 reply; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
This patch adds support to open, write and media format commands
in the q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/qdsp6/q6asm.c | 802 ++++++++++++++++++++++++++++++++++++++++++-
sound/soc/qcom/qdsp6/q6asm.h | 49 +++
2 files changed, 850 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index a20d243ed10a..530852385cad 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -11,6 +11,8 @@
#include <linux/spinlock.h>
#include <linux/kref.h>
#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <uapi/sound/asound.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/mm.h>
@@ -19,10 +21,36 @@
#include "q6dsp-errno.h"
#include "q6dsp-common.h"
+#define ASM_STREAM_CMD_CLOSE 0x00010BCD
+#define ASM_STREAM_CMD_FLUSH 0x00010BCE
+#define ASM_SESSION_CMD_PAUSE 0x00010BD3
+#define ASM_DATA_CMD_EOS 0x00010BDB
+#define ASM_NULL_POPP_TOPOLOGY 0x00010C68
+#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
+#define ASM_STREAM_POSTPROC_TOPO_ID_NONE 0x00010C68
#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
-
+#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
+#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99
+#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
+#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
+#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
+#define ASM_DATA_CMD_READ_V2 0x00010DAC
+#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
+#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
+#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
+#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+
+
+#define ASM_LEGACY_STREAM_SESSION 0
+/* Bit shift for the stream_perf_mode subfield. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29
+#define ASM_END_POINT_DEVICE_MATRIX 0
+#define ASM_DEFAULT_APP_TYPE 0
#define ASM_SYNC_IO_MODE 0x0001
#define ASM_ASYNC_IO_MODE 0x0002
#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
@@ -46,6 +74,89 @@ 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 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_MAX_NUM_CHANNEL];
+} __packed;
+
+struct asm_stream_cmd_set_encdec_param {
+ u32 param_id;
+ u32 param_size;
+} __packed;
+
+struct asm_enc_cfg_blk_param_v2 {
+ u32 frames_per_buf;
+ u32 enc_cfg_blk_size;
+} __packed;
+
+struct asm_multi_channel_pcm_enc_cfg_v2 {
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ uint16_t num_channels;
+ uint16_t bits_per_sample;
+ uint32_t sample_rate;
+ uint16_t is_signed;
+ uint16_t reserved;
+ uint8_t channel_mapping[8];
+} __packed;
+
+struct asm_data_cmd_read_v2 {
+ u32 buf_addr_lsw;
+ u32 buf_addr_msw;
+ u32 mem_map_handle;
+ u32 buf_size;
+ u32 seq_id;
+} __packed;
+
+struct asm_data_cmd_read_v2_done {
+ u32 status;
+ u32 buf_addr_lsw;
+ u32 buf_addr_msw;
+};
+
+struct asm_stream_cmd_open_read_v3 {
+ u32 mode_flags;
+ u32 src_endpointype;
+ u32 preprocopo_id;
+ u32 enc_cfg_id;
+ u16 bits_per_sample;
+ u16 reserved;
+} __packed;
+
+struct asm_data_cmd_write_v2 {
+ 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 {
+ 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 {
+ u32 flags;
+ u32 time_lsw;
+ u32 time_msw;
+} __packed;
+
struct audio_buffer {
phys_addr_t phys;
uint32_t size; /* size of buffer */
@@ -409,6 +520,149 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
return ac;
}
+static int32_t q6asm_stream_callback(struct apr_device *adev,
+ struct apr_resp_pkt *data,
+ int session_id)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *result;
+ struct apr_hdr *hdr = &data->hdr;
+ struct audio_port_data *port;
+ struct audio_client *ac;
+ uint32_t client_event = 0;
+ int ret = 0;
+
+ ac = q6asm_get_audio_client(q6asm, session_id);
+ if (!ac)/* Audio client might already be freed by now */
+ return 0;
+
+ result = data->payload;
+
+ switch (hdr->opcode) {
+ case APR_BASIC_RSP_RESULT:
+ 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;
+ 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_CLOSE:
+ client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
+ break;
+ case ASM_STREAM_CMD_FLUSH_READBUFS:
+ client_event = ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE;
+ break;
+ case ASM_STREAM_CMD_OPEN_WRITE_V3:
+ case ASM_STREAM_CMD_OPEN_READ_V3:
+ case ASM_STREAM_CMD_OPEN_READWRITE_V2:
+ case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
+ case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+ if (result->status != 0) {
+ dev_err(ac->dev,
+ "cmd = 0x%x returned error = 0x%x\n",
+ result->opcode, result->status);
+ ac->result = *result;
+ wake_up(&ac->cmd_wait);
+ ret = 0;
+ goto done;
+ }
+ break;
+ default:
+ dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
+ result->opcode);
+ break;
+ }
+
+ ac->result = *result;
+ wake_up(&ac->cmd_wait);
+
+ if (ac->cb)
+ ac->cb(client_event, hdr->token,
+ data->payload, ac->priv);
+
+ ret = 0;
+ goto done;
+
+ case ASM_DATA_EVENT_WRITE_DONE_V2:
+ client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
+ if (ac->io_mode & ASM_SYNC_IO_MODE) {
+ phys_addr_t phys;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ac->lock, flags);
+
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+
+ if (!port->buf) {
+ spin_unlock_irqrestore(&ac->lock, flags);
+ ret = 0;
+ goto done;
+ }
+
+ phys = port->buf[hdr->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[hdr->token].phys);
+ spin_unlock_irqrestore(&ac->lock, flags);
+ ret = -EINVAL;
+ goto done;
+ }
+ spin_unlock_irqrestore(&ac->lock, flags);
+ }
+ break;
+ case ASM_DATA_EVENT_READ_DONE_V2:
+ client_event = ASM_CLIENT_EVENT_DATA_READ_DONE;
+ if (ac->io_mode & ASM_SYNC_IO_MODE) {
+ struct asm_data_cmd_read_v2_done *done = data->payload;
+ unsigned long flags;
+ phys_addr_t phys;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ if (!port->buf) {
+ spin_unlock_irqrestore(&ac->lock, flags);
+ ret = 0;
+ goto done;
+ }
+
+ phys = port->buf[hdr->token].phys;
+
+ if (upper_32_bits(phys) != done->buf_addr_msw ||
+ lower_32_bits(phys) != done->buf_addr_lsw) {
+ dev_err(ac->dev, "Expected addr %pa %08x-%08x\n",
+ &port->buf[hdr->token].phys,
+ done->buf_addr_lsw,
+ done->buf_addr_msw);
+ spin_unlock_irqrestore(&ac->lock, flags);
+ ret = -EINVAL;
+ goto done;
+ }
+ spin_unlock_irqrestore(&ac->lock, flags);
+ }
+
+ break;
+ }
+
+ if (ac->cb)
+ ac->cb(client_event, hdr->token, data->payload, ac->priv);
+
+done:
+ kref_put(&ac->refcount, q6asm_audio_client_release);
+ return ret;
+}
+
static int q6asm_srvc_callback(struct apr_device *adev,
struct apr_resp_pkt *data)
{
@@ -420,6 +674,11 @@ static int q6asm_srvc_callback(struct apr_device *adev,
struct q6asm *a;
uint32_t sid = 0;
uint32_t dir = 0;
+ int session_id;
+
+ session_id = (hdr->dest_port >> 8) & 0xFF;
+ if (session_id)
+ return q6asm_stream_callback(adev, data, session_id);
sid = (hdr->token >> 8) & 0x0F;
ac = q6asm_get_audio_client(q6asm, sid);
@@ -540,6 +799,547 @@ 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, struct apr_pkt *pkt)
+{
+ struct apr_hdr *hdr = &pkt->hdr;
+ int rc;
+
+ mutex_lock(&ac->cmd_lock);
+ ac->result.opcode = 0;
+ ac->result.status = 0;
+
+ rc = apr_send_pkt(ac->adev, pkt);
+ if (rc < 0)
+ goto err;
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (ac->result.opcode == hdr->opcode), 5 * HZ);
+ if (!rc) {
+ dev_err(ac->dev, "CMD timeout\n");
+ rc = -ETIMEDOUT;
+ goto err;
+ }
+
+ if (ac->result.status > 0) {
+ dev_err(ac->dev, "DSP returned error[%x]\n",
+ ac->result.status);
+ rc = -EINVAL;
+ } else {
+ rc = 0;
+ }
+
+
+err:
+ mutex_unlock(&ac->cmd_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;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*open);
+
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ open = p + APR_HDR_SIZE;
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->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_NULL_POPP_TOPOLOGY;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ default:
+ dev_err(ac->dev, "Invalid format 0x%x\n", format);
+ rc = -EINVAL;
+ goto err;
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ if (rc < 0)
+ goto err;
+
+ ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
+
+err:
+ kfree(pkt);
+ return rc;
+}
+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;
+ struct apr_pkt *pkt;
+ int pkt_size, rc;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*run);
+ p = kzalloc(pkt_size, GFP_ATOMIC);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ run = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+ run->flags = flags;
+ run->time_lsw = lsw_ts;
+ run->time_msw = msw_ts;
+ if (wait) {
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ } else {
+ rc = apr_send_pkt(ac->adev, pkt);
+ if (rc == pkt_size)
+ rc = 0;
+ }
+
+ kfree(pkt);
+ return rc;
+}
+
+/**
+ * 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.
+ * @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_MAX_NUM_CHANNEL],
+ uint16_t bits_per_sample)
+{
+ struct asm_multi_channel_pcm_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ u8 *channel_mapping;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - 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_MAX_NUM_CHANNEL);
+ } else {
+ if (q6dsp_map_channels(channel_mapping, channels)) {
+ dev_err(ac->dev, " map channels failed %d\n", channels);
+ rc = -EINVAL;
+ goto err;
+ }
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+
+err:
+ kfree(pkt);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
+
+/**
+ * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
+ *
+ * @ac: audio client pointer
+ * @rate: audio sample rate
+ * @channels: number of audio channels.
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+ uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
+{
+ struct asm_multi_channel_pcm_enc_cfg_v2 *enc_cfg;
+ struct apr_pkt *pkt;
+ u8 *channel_mapping;
+ u32 frames_per_buf = 0;
+ int pkt_size, rc;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*enc_cfg);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ enc_cfg = p + APR_HDR_SIZE;
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg->encdec.param_size = sizeof(*enc_cfg) - sizeof(enc_cfg->encdec);
+ enc_cfg->encblk.frames_per_buf = frames_per_buf;
+ enc_cfg->encblk.enc_cfg_blk_size = enc_cfg->encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ enc_cfg->num_channels = channels;
+ enc_cfg->bits_per_sample = bits_per_sample;
+ enc_cfg->sample_rate = rate;
+ enc_cfg->is_signed = 1;
+ channel_mapping = enc_cfg->channel_mapping;
+
+ if (q6dsp_map_channels(channel_mapping, channels)) {
+ rc = -EINVAL;
+ goto err;
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+err:
+ kfree(pkt);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
+
+/**
+ * q6asm_read() - read data of period size from audio client
+ *
+ * @ac: audio client pointer
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_read(struct audio_client *ac)
+{
+ struct asm_data_cmd_read_v2 *read;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ struct apr_pkt *pkt;
+ unsigned long flags;
+ int pkt_size;
+ int rc = 0;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*read);
+ p = kzalloc(pkt_size, GFP_ATOMIC);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ read = p + APR_HDR_SIZE;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+ ab = &port->buf[port->dsp_buf];
+ pkt->hdr.opcode = ASM_DATA_CMD_READ_V2;
+ read->buf_addr_lsw = lower_32_bits(ab->phys);
+ read->buf_addr_msw = upper_32_bits(ab->phys);
+ read->mem_map_handle = port->mem_map_handle;
+
+ read->buf_size = ab->size;
+ read->seq_id = port->dsp_buf;
+ pkt->hdr.token = port->dsp_buf;
+
+ port->dsp_buf++;
+
+ if (port->dsp_buf >= port->num_periods)
+ port->dsp_buf = 0;
+
+ spin_unlock_irqrestore(&ac->lock, flags);
+ rc = apr_send_pkt(ac->adev, pkt);
+ if (rc == pkt_size)
+ rc = 0;
+ else
+ pr_err("read op[0x%x]rc[%d]\n", pkt->hdr.opcode, rc);
+
+ kfree(pkt);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_read);
+
+static int __q6asm_open_read(struct audio_client *ac,
+ uint32_t format, uint16_t bits_per_sample)
+{
+ struct asm_stream_cmd_open_read_v3 *open;
+ struct apr_pkt *pkt;
+ int pkt_size, rc;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*open);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ open = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
+ /* Stream prio : High, provide meta info with encoded frames */
+ open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+
+ open->preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_NONE;
+ open->bits_per_sample = bits_per_sample;
+ open->mode_flags = 0x0;
+
+ open->mode_flags |= ASM_LEGACY_STREAM_SESSION <<
+ ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open->mode_flags |= 0x00;
+ open->enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", format);
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+
+ kfree(pkt);
+ return rc;
+}
+
+/**
+ * q6asm_open_read() - Open audio client for reading
+ *
+ * @ac: audio client pointer
+ * @format: audio sample format
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_open_read(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample)
+{
+ return __q6asm_open_read(ac, format, bits_per_sample);
+}
+EXPORT_SYMBOL_GPL(q6asm_open_read);
+
+/**
+ * q6asm_write_async() - non blocking write
+ *
+ * @ac: audio client pointer
+ * @len: lenght in bytes
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ * @wflags: 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 wflags)
+{
+ struct asm_data_cmd_write_v2 *write;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ unsigned long flags;
+ struct apr_pkt *pkt;
+ int pkt_size;
+ int rc = 0;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*write);
+ p = kzalloc(pkt_size, GFP_ATOMIC);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ write = p + APR_HDR_SIZE;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+
+ ab = &port->buf[port->dsp_buf];
+ pkt->hdr.token = port->dsp_buf;
+ pkt->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 (wflags == NO_TIMESTAMP)
+ write->flags = (wflags & 0x800000FF);
+ else
+ write->flags = (0x80000000 | wflags);
+
+ port->dsp_buf++;
+
+ if (port->dsp_buf >= port->num_periods)
+ port->dsp_buf = 0;
+
+ spin_unlock_irqrestore(&ac->lock, flags);
+ rc = apr_send_pkt(ac->adev, pkt);
+ if (rc == pkt_size)
+ rc = 0;
+
+ kfree(pkt);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_write_async);
+
+static void q6asm_reset_buf_state(struct audio_client *ac)
+{
+ struct audio_port_data *port = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+ port->dsp_buf = 0;
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ port->dsp_buf = 0;
+ spin_unlock_irqrestore(&ac->lock, flags);
+}
+
+static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
+{
+ int stream_id = ac->stream_id;
+ struct apr_pkt pkt;
+ int rc;
+
+ q6asm_add_hdr(ac, &pkt.hdr, APR_HDR_SIZE, true, stream_id);
+
+ switch (cmd) {
+ case CMD_PAUSE:
+ pkt.hdr.opcode = ASM_SESSION_CMD_PAUSE;
+ break;
+ case CMD_SUSPEND:
+ pkt.hdr.opcode = ASM_SESSION_CMD_SUSPEND;
+ break;
+ case CMD_FLUSH:
+ pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH;
+ break;
+ case CMD_OUT_FLUSH:
+ pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
+ break;
+ case CMD_EOS:
+ pkt.hdr.opcode = ASM_DATA_CMD_EOS;
+ break;
+ case CMD_CLOSE:
+ pkt.hdr.opcode = ASM_STREAM_CMD_CLOSE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (wait)
+ rc = q6asm_ac_send_cmd_sync(ac, &pkt);
+ else
+ return apr_send_pkt(ac->adev, &pkt);
+
+ 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 8c317b7b63c3..9f5fb573e4a0 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -1,8 +1,36 @@
/* SPDX-License-Identifier: GPL-2.0 */
#ifndef __Q6_ASM_H__
#define __Q6_ASM_H__
+#include "q6dsp-common.h"
+#include <dt-bindings/sound/qcom,q6asm.h>
+
+/* ASM client callback events */
+#define CMD_PAUSE 0x0001
+#define ASM_CLIENT_EVENT_CMD_PAUSE_DONE 0x1001
+#define CMD_FLUSH 0x0002
+#define ASM_CLIENT_EVENT_CMD_FLUSH_DONE 0x1002
+#define CMD_EOS 0x0003
+#define ASM_CLIENT_EVENT_CMD_EOS_DONE 0x1003
+#define CMD_CLOSE 0x0004
+#define ASM_CLIENT_EVENT_CMD_CLOSE_DONE 0x1004
+#define CMD_OUT_FLUSH 0x0005
+#define ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE 0x1005
+#define CMD_SUSPEND 0x0006
+#define ASM_CLIENT_EVENT_CMD_SUSPEND_DONE 0x1006
+#define ASM_CLIENT_EVENT_CMD_RUN_DONE 0x1008
+#define ASM_CLIENT_EVENT_DATA_WRITE_DONE 0x1009
+#define ASM_CLIENT_EVENT_DATA_READ_DONE 0x100a
+
+enum {
+ LEGACY_PCM_MODE = 0,
+ LOW_LATENCY_PCM_MODE,
+ ULTRA_LOW_LATENCY_PCM_MODE,
+ ULL_POST_PROCESSING_PCM_MODE,
+};
#define MAX_SESSIONS 8
+#define NO_TIMESTAMP 0xFF00
+#define FORMAT_LINEAR_PCM 0x0000
typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
void *payload, void *priv);
@@ -11,6 +39,27 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
q6asm_cb cb, void *priv,
int session_id, int perf_mode);
void q6asm_audio_client_free(struct audio_client *ac);
+int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags);
+int q6asm_open_write(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample);
+
+int q6asm_open_read(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample);
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+ uint32_t rate, uint32_t channels, uint16_t bits_per_sample);
+int q6asm_read(struct audio_client *ac);
+
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ u8 channel_map[PCM_MAX_NUM_CHANNEL],
+ uint16_t bits_per_sample);
+int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
+ uint32_t lsw_ts);
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
+ uint32_t lsw_ts);
+int q6asm_cmd(struct audio_client *ac, int cmd);
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
int q6asm_get_session_id(struct audio_client *ac);
int q6asm_map_memory_regions(unsigned int dir,
struct audio_client *ac,
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* Applied "ASoC: qdsp6: q6asm: Add support to audio stream apis" to the asoc tree
2018-05-18 12:56 ` [PATCH v9 08/15] ASoC: qdsp6: q6asm: Add support to audio stream apis Srinivas Kandagatla
@ 2018-05-21 15:47 ` Mark Brown
0 siblings, 0 replies; 25+ messages in thread
From: Mark Brown @ 2018-05-21 15:47 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, broonie, linux-arm-msm, alsa-devel, bgoswami,
mark.rutland, devicetree, rohkumar, gregkh, plai, lgirdwood,
tiwai, david.brown, robh+dt, linux-arm-kernel, spatakok,
linux-kernel, alsa-devel
The patch
ASoC: qdsp6: q6asm: Add support to audio stream apis
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From 68fd8480bb7baaf361e983f75d8571f25e017c67 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:56:03 +0100
Subject: [PATCH] ASoC: qdsp6: q6asm: Add support to audio stream apis
This patch adds support to open, write and media format commands
in the q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/qdsp6/q6asm.c | 802 ++++++++++++++++++++++++++++++++++-
sound/soc/qcom/qdsp6/q6asm.h | 49 +++
2 files changed, 850 insertions(+), 1 deletion(-)
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index a20d243ed10a..530852385cad 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -11,6 +11,8 @@
#include <linux/spinlock.h>
#include <linux/kref.h>
#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <uapi/sound/asound.h>
#include <linux/delay.h>
#include <linux/slab.h>
#include <linux/mm.h>
@@ -19,10 +21,36 @@
#include "q6dsp-errno.h"
#include "q6dsp-common.h"
+#define ASM_STREAM_CMD_CLOSE 0x00010BCD
+#define ASM_STREAM_CMD_FLUSH 0x00010BCE
+#define ASM_SESSION_CMD_PAUSE 0x00010BD3
+#define ASM_DATA_CMD_EOS 0x00010BDB
+#define ASM_NULL_POPP_TOPOLOGY 0x00010C68
+#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
+#define ASM_STREAM_POSTPROC_TOPO_ID_NONE 0x00010C68
#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
-
+#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
+#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99
+#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
+#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
+#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
+#define ASM_DATA_CMD_READ_V2 0x00010DAC
+#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
+#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
+#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
+#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+
+
+#define ASM_LEGACY_STREAM_SESSION 0
+/* Bit shift for the stream_perf_mode subfield. */
+#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29
+#define ASM_END_POINT_DEVICE_MATRIX 0
+#define ASM_DEFAULT_APP_TYPE 0
#define ASM_SYNC_IO_MODE 0x0001
#define ASM_ASYNC_IO_MODE 0x0002
#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
@@ -46,6 +74,89 @@ 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 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_MAX_NUM_CHANNEL];
+} __packed;
+
+struct asm_stream_cmd_set_encdec_param {
+ u32 param_id;
+ u32 param_size;
+} __packed;
+
+struct asm_enc_cfg_blk_param_v2 {
+ u32 frames_per_buf;
+ u32 enc_cfg_blk_size;
+} __packed;
+
+struct asm_multi_channel_pcm_enc_cfg_v2 {
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ uint16_t num_channels;
+ uint16_t bits_per_sample;
+ uint32_t sample_rate;
+ uint16_t is_signed;
+ uint16_t reserved;
+ uint8_t channel_mapping[8];
+} __packed;
+
+struct asm_data_cmd_read_v2 {
+ u32 buf_addr_lsw;
+ u32 buf_addr_msw;
+ u32 mem_map_handle;
+ u32 buf_size;
+ u32 seq_id;
+} __packed;
+
+struct asm_data_cmd_read_v2_done {
+ u32 status;
+ u32 buf_addr_lsw;
+ u32 buf_addr_msw;
+};
+
+struct asm_stream_cmd_open_read_v3 {
+ u32 mode_flags;
+ u32 src_endpointype;
+ u32 preprocopo_id;
+ u32 enc_cfg_id;
+ u16 bits_per_sample;
+ u16 reserved;
+} __packed;
+
+struct asm_data_cmd_write_v2 {
+ 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 {
+ 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 {
+ u32 flags;
+ u32 time_lsw;
+ u32 time_msw;
+} __packed;
+
struct audio_buffer {
phys_addr_t phys;
uint32_t size; /* size of buffer */
@@ -409,6 +520,149 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
return ac;
}
+static int32_t q6asm_stream_callback(struct apr_device *adev,
+ struct apr_resp_pkt *data,
+ int session_id)
+{
+ struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
+ struct aprv2_ibasic_rsp_result_t *result;
+ struct apr_hdr *hdr = &data->hdr;
+ struct audio_port_data *port;
+ struct audio_client *ac;
+ uint32_t client_event = 0;
+ int ret = 0;
+
+ ac = q6asm_get_audio_client(q6asm, session_id);
+ if (!ac)/* Audio client might already be freed by now */
+ return 0;
+
+ result = data->payload;
+
+ switch (hdr->opcode) {
+ case APR_BASIC_RSP_RESULT:
+ 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;
+ 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_CLOSE:
+ client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
+ break;
+ case ASM_STREAM_CMD_FLUSH_READBUFS:
+ client_event = ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE;
+ break;
+ case ASM_STREAM_CMD_OPEN_WRITE_V3:
+ case ASM_STREAM_CMD_OPEN_READ_V3:
+ case ASM_STREAM_CMD_OPEN_READWRITE_V2:
+ case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
+ case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+ if (result->status != 0) {
+ dev_err(ac->dev,
+ "cmd = 0x%x returned error = 0x%x\n",
+ result->opcode, result->status);
+ ac->result = *result;
+ wake_up(&ac->cmd_wait);
+ ret = 0;
+ goto done;
+ }
+ break;
+ default:
+ dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
+ result->opcode);
+ break;
+ }
+
+ ac->result = *result;
+ wake_up(&ac->cmd_wait);
+
+ if (ac->cb)
+ ac->cb(client_event, hdr->token,
+ data->payload, ac->priv);
+
+ ret = 0;
+ goto done;
+
+ case ASM_DATA_EVENT_WRITE_DONE_V2:
+ client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
+ if (ac->io_mode & ASM_SYNC_IO_MODE) {
+ phys_addr_t phys;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ac->lock, flags);
+
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+
+ if (!port->buf) {
+ spin_unlock_irqrestore(&ac->lock, flags);
+ ret = 0;
+ goto done;
+ }
+
+ phys = port->buf[hdr->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[hdr->token].phys);
+ spin_unlock_irqrestore(&ac->lock, flags);
+ ret = -EINVAL;
+ goto done;
+ }
+ spin_unlock_irqrestore(&ac->lock, flags);
+ }
+ break;
+ case ASM_DATA_EVENT_READ_DONE_V2:
+ client_event = ASM_CLIENT_EVENT_DATA_READ_DONE;
+ if (ac->io_mode & ASM_SYNC_IO_MODE) {
+ struct asm_data_cmd_read_v2_done *done = data->payload;
+ unsigned long flags;
+ phys_addr_t phys;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ if (!port->buf) {
+ spin_unlock_irqrestore(&ac->lock, flags);
+ ret = 0;
+ goto done;
+ }
+
+ phys = port->buf[hdr->token].phys;
+
+ if (upper_32_bits(phys) != done->buf_addr_msw ||
+ lower_32_bits(phys) != done->buf_addr_lsw) {
+ dev_err(ac->dev, "Expected addr %pa %08x-%08x\n",
+ &port->buf[hdr->token].phys,
+ done->buf_addr_lsw,
+ done->buf_addr_msw);
+ spin_unlock_irqrestore(&ac->lock, flags);
+ ret = -EINVAL;
+ goto done;
+ }
+ spin_unlock_irqrestore(&ac->lock, flags);
+ }
+
+ break;
+ }
+
+ if (ac->cb)
+ ac->cb(client_event, hdr->token, data->payload, ac->priv);
+
+done:
+ kref_put(&ac->refcount, q6asm_audio_client_release);
+ return ret;
+}
+
static int q6asm_srvc_callback(struct apr_device *adev,
struct apr_resp_pkt *data)
{
@@ -420,6 +674,11 @@ static int q6asm_srvc_callback(struct apr_device *adev,
struct q6asm *a;
uint32_t sid = 0;
uint32_t dir = 0;
+ int session_id;
+
+ session_id = (hdr->dest_port >> 8) & 0xFF;
+ if (session_id)
+ return q6asm_stream_callback(adev, data, session_id);
sid = (hdr->token >> 8) & 0x0F;
ac = q6asm_get_audio_client(q6asm, sid);
@@ -540,6 +799,547 @@ 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, struct apr_pkt *pkt)
+{
+ struct apr_hdr *hdr = &pkt->hdr;
+ int rc;
+
+ mutex_lock(&ac->cmd_lock);
+ ac->result.opcode = 0;
+ ac->result.status = 0;
+
+ rc = apr_send_pkt(ac->adev, pkt);
+ if (rc < 0)
+ goto err;
+
+ rc = wait_event_timeout(ac->cmd_wait,
+ (ac->result.opcode == hdr->opcode), 5 * HZ);
+ if (!rc) {
+ dev_err(ac->dev, "CMD timeout\n");
+ rc = -ETIMEDOUT;
+ goto err;
+ }
+
+ if (ac->result.status > 0) {
+ dev_err(ac->dev, "DSP returned error[%x]\n",
+ ac->result.status);
+ rc = -EINVAL;
+ } else {
+ rc = 0;
+ }
+
+
+err:
+ mutex_unlock(&ac->cmd_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;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*open);
+
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ open = p + APR_HDR_SIZE;
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->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_NULL_POPP_TOPOLOGY;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ default:
+ dev_err(ac->dev, "Invalid format 0x%x\n", format);
+ rc = -EINVAL;
+ goto err;
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ if (rc < 0)
+ goto err;
+
+ ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
+
+err:
+ kfree(pkt);
+ return rc;
+}
+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;
+ struct apr_pkt *pkt;
+ int pkt_size, rc;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*run);
+ p = kzalloc(pkt_size, GFP_ATOMIC);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ run = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2;
+ run->flags = flags;
+ run->time_lsw = lsw_ts;
+ run->time_msw = msw_ts;
+ if (wait) {
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ } else {
+ rc = apr_send_pkt(ac->adev, pkt);
+ if (rc == pkt_size)
+ rc = 0;
+ }
+
+ kfree(pkt);
+ return rc;
+}
+
+/**
+ * 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.
+ * @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_MAX_NUM_CHANNEL],
+ uint16_t bits_per_sample)
+{
+ struct asm_multi_channel_pcm_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ u8 *channel_mapping;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - 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_MAX_NUM_CHANNEL);
+ } else {
+ if (q6dsp_map_channels(channel_mapping, channels)) {
+ dev_err(ac->dev, " map channels failed %d\n", channels);
+ rc = -EINVAL;
+ goto err;
+ }
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+
+err:
+ kfree(pkt);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
+
+/**
+ * q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
+ *
+ * @ac: audio client pointer
+ * @rate: audio sample rate
+ * @channels: number of audio channels.
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+ uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
+{
+ struct asm_multi_channel_pcm_enc_cfg_v2 *enc_cfg;
+ struct apr_pkt *pkt;
+ u8 *channel_mapping;
+ u32 frames_per_buf = 0;
+ int pkt_size, rc;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*enc_cfg);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ enc_cfg = p + APR_HDR_SIZE;
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+
+ pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg->encdec.param_size = sizeof(*enc_cfg) - sizeof(enc_cfg->encdec);
+ enc_cfg->encblk.frames_per_buf = frames_per_buf;
+ enc_cfg->encblk.enc_cfg_blk_size = enc_cfg->encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ enc_cfg->num_channels = channels;
+ enc_cfg->bits_per_sample = bits_per_sample;
+ enc_cfg->sample_rate = rate;
+ enc_cfg->is_signed = 1;
+ channel_mapping = enc_cfg->channel_mapping;
+
+ if (q6dsp_map_channels(channel_mapping, channels)) {
+ rc = -EINVAL;
+ goto err;
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+err:
+ kfree(pkt);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
+
+/**
+ * q6asm_read() - read data of period size from audio client
+ *
+ * @ac: audio client pointer
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_read(struct audio_client *ac)
+{
+ struct asm_data_cmd_read_v2 *read;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ struct apr_pkt *pkt;
+ unsigned long flags;
+ int pkt_size;
+ int rc = 0;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*read);
+ p = kzalloc(pkt_size, GFP_ATOMIC);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ read = p + APR_HDR_SIZE;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+ ab = &port->buf[port->dsp_buf];
+ pkt->hdr.opcode = ASM_DATA_CMD_READ_V2;
+ read->buf_addr_lsw = lower_32_bits(ab->phys);
+ read->buf_addr_msw = upper_32_bits(ab->phys);
+ read->mem_map_handle = port->mem_map_handle;
+
+ read->buf_size = ab->size;
+ read->seq_id = port->dsp_buf;
+ pkt->hdr.token = port->dsp_buf;
+
+ port->dsp_buf++;
+
+ if (port->dsp_buf >= port->num_periods)
+ port->dsp_buf = 0;
+
+ spin_unlock_irqrestore(&ac->lock, flags);
+ rc = apr_send_pkt(ac->adev, pkt);
+ if (rc == pkt_size)
+ rc = 0;
+ else
+ pr_err("read op[0x%x]rc[%d]\n", pkt->hdr.opcode, rc);
+
+ kfree(pkt);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_read);
+
+static int __q6asm_open_read(struct audio_client *ac,
+ uint32_t format, uint16_t bits_per_sample)
+{
+ struct asm_stream_cmd_open_read_v3 *open;
+ struct apr_pkt *pkt;
+ int pkt_size, rc;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*open);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ open = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
+ /* Stream prio : High, provide meta info with encoded frames */
+ open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
+
+ open->preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_NONE;
+ open->bits_per_sample = bits_per_sample;
+ open->mode_flags = 0x0;
+
+ open->mode_flags |= ASM_LEGACY_STREAM_SESSION <<
+ ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
+
+ switch (format) {
+ case FORMAT_LINEAR_PCM:
+ open->mode_flags |= 0x00;
+ open->enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
+ break;
+ default:
+ pr_err("Invalid format[%d]\n", format);
+ }
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+
+ kfree(pkt);
+ return rc;
+}
+
+/**
+ * q6asm_open_read() - Open audio client for reading
+ *
+ * @ac: audio client pointer
+ * @format: audio sample format
+ * @bits_per_sample: bits per sample
+ *
+ * Return: Will be an negative value on error or zero on success
+ */
+int q6asm_open_read(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample)
+{
+ return __q6asm_open_read(ac, format, bits_per_sample);
+}
+EXPORT_SYMBOL_GPL(q6asm_open_read);
+
+/**
+ * q6asm_write_async() - non blocking write
+ *
+ * @ac: audio client pointer
+ * @len: lenght in bytes
+ * @msw_ts: timestamp msw
+ * @lsw_ts: timestamp lsw
+ * @wflags: 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 wflags)
+{
+ struct asm_data_cmd_write_v2 *write;
+ struct audio_port_data *port;
+ struct audio_buffer *ab;
+ unsigned long flags;
+ struct apr_pkt *pkt;
+ int pkt_size;
+ int rc = 0;
+ void *p;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*write);
+ p = kzalloc(pkt_size, GFP_ATOMIC);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ write = p + APR_HDR_SIZE;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+
+ ab = &port->buf[port->dsp_buf];
+ pkt->hdr.token = port->dsp_buf;
+ pkt->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 (wflags == NO_TIMESTAMP)
+ write->flags = (wflags & 0x800000FF);
+ else
+ write->flags = (0x80000000 | wflags);
+
+ port->dsp_buf++;
+
+ if (port->dsp_buf >= port->num_periods)
+ port->dsp_buf = 0;
+
+ spin_unlock_irqrestore(&ac->lock, flags);
+ rc = apr_send_pkt(ac->adev, pkt);
+ if (rc == pkt_size)
+ rc = 0;
+
+ kfree(pkt);
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_write_async);
+
+static void q6asm_reset_buf_state(struct audio_client *ac)
+{
+ struct audio_port_data *port = NULL;
+ unsigned long flags;
+
+ spin_lock_irqsave(&ac->lock, flags);
+ port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
+ port->dsp_buf = 0;
+ port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
+ port->dsp_buf = 0;
+ spin_unlock_irqrestore(&ac->lock, flags);
+}
+
+static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
+{
+ int stream_id = ac->stream_id;
+ struct apr_pkt pkt;
+ int rc;
+
+ q6asm_add_hdr(ac, &pkt.hdr, APR_HDR_SIZE, true, stream_id);
+
+ switch (cmd) {
+ case CMD_PAUSE:
+ pkt.hdr.opcode = ASM_SESSION_CMD_PAUSE;
+ break;
+ case CMD_SUSPEND:
+ pkt.hdr.opcode = ASM_SESSION_CMD_SUSPEND;
+ break;
+ case CMD_FLUSH:
+ pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH;
+ break;
+ case CMD_OUT_FLUSH:
+ pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
+ break;
+ case CMD_EOS:
+ pkt.hdr.opcode = ASM_DATA_CMD_EOS;
+ break;
+ case CMD_CLOSE:
+ pkt.hdr.opcode = ASM_STREAM_CMD_CLOSE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (wait)
+ rc = q6asm_ac_send_cmd_sync(ac, &pkt);
+ else
+ return apr_send_pkt(ac->adev, &pkt);
+
+ 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 8c317b7b63c3..9f5fb573e4a0 100644
--- a/sound/soc/qcom/qdsp6/q6asm.h
+++ b/sound/soc/qcom/qdsp6/q6asm.h
@@ -1,8 +1,36 @@
/* SPDX-License-Identifier: GPL-2.0 */
#ifndef __Q6_ASM_H__
#define __Q6_ASM_H__
+#include "q6dsp-common.h"
+#include <dt-bindings/sound/qcom,q6asm.h>
+
+/* ASM client callback events */
+#define CMD_PAUSE 0x0001
+#define ASM_CLIENT_EVENT_CMD_PAUSE_DONE 0x1001
+#define CMD_FLUSH 0x0002
+#define ASM_CLIENT_EVENT_CMD_FLUSH_DONE 0x1002
+#define CMD_EOS 0x0003
+#define ASM_CLIENT_EVENT_CMD_EOS_DONE 0x1003
+#define CMD_CLOSE 0x0004
+#define ASM_CLIENT_EVENT_CMD_CLOSE_DONE 0x1004
+#define CMD_OUT_FLUSH 0x0005
+#define ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE 0x1005
+#define CMD_SUSPEND 0x0006
+#define ASM_CLIENT_EVENT_CMD_SUSPEND_DONE 0x1006
+#define ASM_CLIENT_EVENT_CMD_RUN_DONE 0x1008
+#define ASM_CLIENT_EVENT_DATA_WRITE_DONE 0x1009
+#define ASM_CLIENT_EVENT_DATA_READ_DONE 0x100a
+
+enum {
+ LEGACY_PCM_MODE = 0,
+ LOW_LATENCY_PCM_MODE,
+ ULTRA_LOW_LATENCY_PCM_MODE,
+ ULL_POST_PROCESSING_PCM_MODE,
+};
#define MAX_SESSIONS 8
+#define NO_TIMESTAMP 0xFF00
+#define FORMAT_LINEAR_PCM 0x0000
typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
void *payload, void *priv);
@@ -11,6 +39,27 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev,
q6asm_cb cb, void *priv,
int session_id, int perf_mode);
void q6asm_audio_client_free(struct audio_client *ac);
+int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags);
+int q6asm_open_write(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample);
+
+int q6asm_open_read(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample);
+int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
+ uint32_t rate, uint32_t channels, uint16_t bits_per_sample);
+int q6asm_read(struct audio_client *ac);
+
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ u8 channel_map[PCM_MAX_NUM_CHANNEL],
+ uint16_t bits_per_sample);
+int q6asm_run(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
+ uint32_t lsw_ts);
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags, uint32_t msw_ts,
+ uint32_t lsw_ts);
+int q6asm_cmd(struct audio_client *ac, int cmd);
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
int q6asm_get_session_id(struct audio_client *ac);
int q6asm_map_memory_regions(unsigned int dir,
struct audio_client *ac,
--
2.17.0
^ permalink raw reply related [flat|nested] 25+ messages in thread
* [PATCH v9 09/15] ASoC: qdsp6: q6routing: Add q6routing driver
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (7 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 08/15] ASoC: qdsp6: q6asm: Add support to audio stream apis Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-21 15:47 ` Applied "ASoC: qdsp6: q6routing: Add q6routing driver" to the asoc tree Mark Brown
2018-05-18 12:56 ` [PATCH v9 10/15] ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers Srinivas Kandagatla
` (5 subsequent siblings)
14 siblings, 1 reply; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
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>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6routing.c | 400 +++++++++++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6routing.h | 9 +
4 files changed, 414 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 941774abd94f..43f9ed85efa8 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -53,6 +53,9 @@ config SND_SOC_QDSP6_AFE
config SND_SOC_QDSP6_ADM
tristate
+config SND_SOC_QDSP6_ROUTING
+ tristate
+
config SND_SOC_QDSP6_ASM
tristate
@@ -63,6 +66,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_CORE
select SND_SOC_QDSP6_AFE
select SND_SOC_QDSP6_ADM
+ select SND_SOC_QDSP6_ROUTING
select SND_SOC_QDSP6_ASM
help
To add support for MSM QDSP6 Soc Audio.
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 01d9dcf3375c..0e8e2febb7ec 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -2,4 +2,5 @@ obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
+obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
new file mode 100644
index 000000000000..1d7a4088a435
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -0,0 +1,400 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_platform.h>
+#include <linux/bitops.h>
+#include <linux/component.h>
+#include <linux/mutex.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/control.h>
+#include <sound/asound.h>
+#include <sound/pcm_params.h>
+#include "q6afe.h"
+#include "q6asm.h"
+#include "q6adm.h"
+#include "q6routing.h"
+
+#define DRV_NAME "q6routing-component"
+
+struct session_data {
+ int state;
+ int port_id;
+ int path_type;
+ int app_type;
+ int acdb_id;
+ int sample_rate;
+ int bits_per_sample;
+ int channels;
+ int perf_mode;
+ int numcopps;
+ int fedai_id;
+ unsigned long copp_map;
+ struct q6copp *copps[MAX_COPPS_PER_PORT];
+};
+
+struct msm_routing_data {
+ struct session_data sessions[MAX_SESSIONS];
+ struct session_data port_data[AFE_MAX_PORTS];
+ struct device *dev;
+ struct mutex lock;
+};
+
+static struct msm_routing_data *routing_data;
+
+/**
+ * q6routing_stream_open() - Register a new stream for route setup
+ *
+ * @fedai_id: Frontend dai id.
+ * @perf_mode: Performance mode.
+ * @stream_id: ASM stream id to map.
+ * @stream_type: Direction of stream
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6routing_stream_open(int fedai_id, int perf_mode,
+ int stream_id, int stream_type)
+{
+ int j, topology, num_copps = 0;
+ struct route_payload payload;
+ struct q6copp *copp;
+ int copp_idx;
+ struct session_data *session, *pdata;
+
+ if (!routing_data) {
+ pr_err("Routing driver not yet ready\n");
+ return -EINVAL;
+ }
+
+ session = &routing_data->sessions[stream_id - 1];
+ pdata = &routing_data->port_data[session->port_id];
+
+ mutex_lock(&routing_data->lock);
+ session->fedai_id = fedai_id;
+
+ session->path_type = pdata->path_type;
+ session->sample_rate = pdata->sample_rate;
+ session->channels = pdata->channels;
+ session->bits_per_sample = pdata->bits_per_sample;
+
+ payload.num_copps = 0; /* only RX needs to use payload */
+ topology = NULL_COPP_TOPOLOGY;
+ copp = 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) {
+ mutex_unlock(&routing_data->lock);
+ return -EINVAL;
+ }
+
+ copp_idx = q6adm_get_copp_id(copp);
+ set_bit(copp_idx, &session->copp_map);
+ session->copps[copp_idx] = copp;
+
+ for_each_set_bit(j, &session->copp_map, MAX_COPPS_PER_PORT) {
+ payload.port_id[num_copps] = session->port_id;
+ payload.copp_idx[num_copps] = j;
+ num_copps++;
+ }
+
+ if (num_copps) {
+ payload.num_copps = num_copps;
+ payload.session_id = stream_id;
+ q6adm_matrix_map(routing_data->dev, session->path_type,
+ payload, perf_mode);
+ }
+ mutex_unlock(&routing_data->lock);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6routing_stream_open);
+
+static struct session_data *get_session_from_id(struct msm_routing_data *data,
+ int fedai_id)
+{
+ int i;
+
+ for (i = 0; i < MAX_SESSIONS; i++) {
+ if (fedai_id == data->sessions[i].fedai_id)
+ return &data->sessions[i];
+ }
+
+ return NULL;
+}
+/**
+ * q6routing_stream_close() - Deregister a stream
+ *
+ * @fedai_id: Frontend dai id.
+ * @stream_type: Direction of stream
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+void q6routing_stream_close(int fedai_id, int stream_type)
+{
+ struct session_data *session;
+ int idx;
+
+ session = get_session_from_id(routing_data, fedai_id);
+ if (!session)
+ return;
+
+ for_each_set_bit(idx, &session->copp_map, MAX_COPPS_PER_PORT) {
+ if (session->copps[idx]) {
+ q6adm_close(routing_data->dev, session->copps[idx]);
+ session->copps[idx] = NULL;
+ }
+ }
+
+ session->fedai_id = -1;
+ session->copp_map = 0;
+}
+EXPORT_SYMBOL_GPL(q6routing_stream_close);
+
+static int msm_routing_get_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int session_id = mc->shift;
+ struct snd_soc_component *c = snd_soc_dapm_to_component(dapm);
+ struct msm_routing_data *priv = dev_get_drvdata(c->dev);
+ struct session_data *session = &priv->sessions[session_id];
+
+ if (session->port_id == mc->reg)
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_component *c = snd_soc_dapm_to_component(dapm);
+ struct msm_routing_data *data = dev_get_drvdata(c->dev);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_dapm_update *update = NULL;
+ int be_id = mc->reg;
+ int session_id = mc->shift;
+ struct session_data *session = &data->sessions[session_id];
+
+ if (ucontrol->value.integer.value[0]) {
+ session->port_id = be_id;
+ snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update);
+ } else {
+ session->port_id = -1;
+ snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update);
+ }
+
+ return 1;
+}
+
+static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
+ /* Frontend AIF */
+ SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0),
+
+ /* Mixer definitions */
+ SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0,
+ hdmi_mixer_controls,
+ ARRAY_SIZE(hdmi_mixer_controls)),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"HDMI Mixer", "MultiMedia1", "MM_DL1"},
+ {"HDMI Mixer", "MultiMedia2", "MM_DL2"},
+ {"HDMI Mixer", "MultiMedia3", "MM_DL3"},
+ {"HDMI Mixer", "MultiMedia4", "MM_DL4"},
+ {"HDMI Mixer", "MultiMedia5", "MM_DL5"},
+ {"HDMI Mixer", "MultiMedia6", "MM_DL6"},
+ {"HDMI Mixer", "MultiMedia7", "MM_DL7"},
+ {"HDMI Mixer", "MultiMedia8", "MM_DL8"},
+ {"HDMI_RX", NULL, "HDMI Mixer"},
+};
+
+static int routing_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+ struct msm_routing_data *data = dev_get_drvdata(c->dev);
+ unsigned int be_id = rtd->cpu_dai->id;
+ struct session_data *session;
+ int path_type;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ path_type = ADM_PATH_PLAYBACK;
+ else
+ path_type = ADM_PATH_LIVE_REC;
+
+ if (be_id > AFE_MAX_PORTS)
+ return -EINVAL;
+
+ session = &data->port_data[be_id];
+
+ mutex_lock(&data->lock);
+
+ session->path_type = path_type;
+ session->sample_rate = params_rate(params);
+ session->channels = params_channels(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ session->bits_per_sample = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ session->bits_per_sample = 24;
+ break;
+ default:
+ break;
+ }
+
+ mutex_unlock(&data->lock);
+ return 0;
+}
+
+static struct snd_pcm_ops q6pcm_routing_ops = {
+ .hw_params = routing_hw_params,
+};
+
+static int msm_routing_probe(struct snd_soc_component *c)
+{
+ int i;
+
+ for (i = 0; i < MAX_SESSIONS; i++)
+ routing_data->sessions[i].port_id = -1;
+
+ return 0;
+}
+
+static const struct snd_soc_component_driver msm_soc_routing_component = {
+ .ops = &q6pcm_routing_ops,
+ .probe = msm_routing_probe,
+ .name = DRV_NAME,
+ .dapm_widgets = msm_qdsp6_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(msm_qdsp6_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
+};
+
+static int q6routing_dai_bind(struct device *dev, struct device *master,
+ void *data)
+{
+ routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL);
+ if (!routing_data)
+ return -ENOMEM;
+
+ routing_data->dev = dev;
+
+ mutex_init(&routing_data->lock);
+ dev_set_drvdata(dev, routing_data);
+
+ return snd_soc_register_component(dev, &msm_soc_routing_component,
+ NULL, 0);
+}
+
+static void q6routing_dai_unbind(struct device *dev, struct device *master,
+ void *d)
+{
+ struct msm_routing_data *data = dev_get_drvdata(dev);
+
+ snd_soc_unregister_component(dev);
+
+ kfree(data);
+
+ routing_data = NULL;
+}
+
+static const struct component_ops q6routing_dai_comp_ops = {
+ .bind = q6routing_dai_bind,
+ .unbind = q6routing_dai_unbind,
+};
+
+static int q6pcm_routing_probe(struct platform_device *pdev)
+{
+ return component_add(&pdev->dev, &q6routing_dai_comp_ops);
+}
+
+static int q6pcm_routing_remove(struct platform_device *pdev)
+{
+ component_del(&pdev->dev, &q6routing_dai_comp_ops);
+ return 0;
+}
+
+static struct platform_driver q6pcm_routing_platform_driver = {
+ .driver = {
+ .name = "q6routing",
+ },
+ .probe = q6pcm_routing_probe,
+ .remove = q6pcm_routing_remove,
+};
+module_platform_driver(q6pcm_routing_platform_driver);
+
+MODULE_DESCRIPTION("Q6 Routing platform");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6routing.h b/sound/soc/qcom/qdsp6/q6routing.h
new file mode 100644
index 000000000000..35514e651130
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6routing.h
@@ -0,0 +1,9 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef _Q6_PCM_ROUTING_H
+#define _Q6_PCM_ROUTING_H
+
+int q6routing_stream_open(int fedai_id, int perf_mode,
+ int stream_id, int stream_type);
+void q6routing_stream_close(int fedai_id, int stream_type);
+
+#endif /*_Q6_PCM_ROUTING_H */
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* Applied "ASoC: qdsp6: q6routing: Add q6routing driver" to the asoc tree
2018-05-18 12:56 ` [PATCH v9 09/15] ASoC: qdsp6: q6routing: Add q6routing driver Srinivas Kandagatla
@ 2018-05-21 15:47 ` Mark Brown
0 siblings, 0 replies; 25+ messages in thread
From: Mark Brown @ 2018-05-21 15:47 UTC (permalink / raw)
To: Srinivas Kandagatla
Cc: Mark Brown, broonie, linux-arm-msm, alsa-devel, bgoswami,
mark.rutland, devicetree, rohkumar, gregkh, plai, lgirdwood,
tiwai, david.brown, robh+dt, linux-arm-kernel, spatakok,
linux-kernel, alsa-devel
The patch
ASoC: qdsp6: q6routing: Add q6routing driver
has been applied to the asoc tree at
https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
All being well this means that it will be integrated into the linux-next
tree (usually sometime in the next 24 hours) and sent to Linus during
the next merge window (or sooner if it is a bug fix), however if
problems are discovered then the patch may be dropped or reverted.
You may get further e-mails resulting from automated or manual testing
and review of the tree, please engage with people reporting problems and
send followup patches addressing any issues that are reported if needed.
If any updates are required or you are submitting further changes they
should be sent as incremental updates against current git, existing
patches will not be replaced.
Please add any relevant lists and maintainers to the CCs when replying
to this mail.
Thanks,
Mark
>From e3a33673e845f609977970d147d5f8e094c52c73 Mon Sep 17 00:00:00 2001
From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Date: Fri, 18 May 2018 13:56:04 +0100
Subject: [PATCH] ASoC: qdsp6: q6routing: Add q6routing driver
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>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6routing.c | 400 +++++++++++++++++++++++++++++++
sound/soc/qcom/qdsp6/q6routing.h | 9 +
4 files changed, 414 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 941774abd94f..43f9ed85efa8 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -53,6 +53,9 @@ config SND_SOC_QDSP6_AFE
config SND_SOC_QDSP6_ADM
tristate
+config SND_SOC_QDSP6_ROUTING
+ tristate
+
config SND_SOC_QDSP6_ASM
tristate
@@ -63,6 +66,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_CORE
select SND_SOC_QDSP6_AFE
select SND_SOC_QDSP6_ADM
+ select SND_SOC_QDSP6_ROUTING
select SND_SOC_QDSP6_ASM
help
To add support for MSM QDSP6 Soc Audio.
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 01d9dcf3375c..0e8e2febb7ec 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -2,4 +2,5 @@ obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
+obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
new file mode 100644
index 000000000000..1d7a4088a435
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -0,0 +1,400 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_platform.h>
+#include <linux/bitops.h>
+#include <linux/component.h>
+#include <linux/mutex.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/control.h>
+#include <sound/asound.h>
+#include <sound/pcm_params.h>
+#include "q6afe.h"
+#include "q6asm.h"
+#include "q6adm.h"
+#include "q6routing.h"
+
+#define DRV_NAME "q6routing-component"
+
+struct session_data {
+ int state;
+ int port_id;
+ int path_type;
+ int app_type;
+ int acdb_id;
+ int sample_rate;
+ int bits_per_sample;
+ int channels;
+ int perf_mode;
+ int numcopps;
+ int fedai_id;
+ unsigned long copp_map;
+ struct q6copp *copps[MAX_COPPS_PER_PORT];
+};
+
+struct msm_routing_data {
+ struct session_data sessions[MAX_SESSIONS];
+ struct session_data port_data[AFE_MAX_PORTS];
+ struct device *dev;
+ struct mutex lock;
+};
+
+static struct msm_routing_data *routing_data;
+
+/**
+ * q6routing_stream_open() - Register a new stream for route setup
+ *
+ * @fedai_id: Frontend dai id.
+ * @perf_mode: Performance mode.
+ * @stream_id: ASM stream id to map.
+ * @stream_type: Direction of stream
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+int q6routing_stream_open(int fedai_id, int perf_mode,
+ int stream_id, int stream_type)
+{
+ int j, topology, num_copps = 0;
+ struct route_payload payload;
+ struct q6copp *copp;
+ int copp_idx;
+ struct session_data *session, *pdata;
+
+ if (!routing_data) {
+ pr_err("Routing driver not yet ready\n");
+ return -EINVAL;
+ }
+
+ session = &routing_data->sessions[stream_id - 1];
+ pdata = &routing_data->port_data[session->port_id];
+
+ mutex_lock(&routing_data->lock);
+ session->fedai_id = fedai_id;
+
+ session->path_type = pdata->path_type;
+ session->sample_rate = pdata->sample_rate;
+ session->channels = pdata->channels;
+ session->bits_per_sample = pdata->bits_per_sample;
+
+ payload.num_copps = 0; /* only RX needs to use payload */
+ topology = NULL_COPP_TOPOLOGY;
+ copp = 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) {
+ mutex_unlock(&routing_data->lock);
+ return -EINVAL;
+ }
+
+ copp_idx = q6adm_get_copp_id(copp);
+ set_bit(copp_idx, &session->copp_map);
+ session->copps[copp_idx] = copp;
+
+ for_each_set_bit(j, &session->copp_map, MAX_COPPS_PER_PORT) {
+ payload.port_id[num_copps] = session->port_id;
+ payload.copp_idx[num_copps] = j;
+ num_copps++;
+ }
+
+ if (num_copps) {
+ payload.num_copps = num_copps;
+ payload.session_id = stream_id;
+ q6adm_matrix_map(routing_data->dev, session->path_type,
+ payload, perf_mode);
+ }
+ mutex_unlock(&routing_data->lock);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(q6routing_stream_open);
+
+static struct session_data *get_session_from_id(struct msm_routing_data *data,
+ int fedai_id)
+{
+ int i;
+
+ for (i = 0; i < MAX_SESSIONS; i++) {
+ if (fedai_id == data->sessions[i].fedai_id)
+ return &data->sessions[i];
+ }
+
+ return NULL;
+}
+/**
+ * q6routing_stream_close() - Deregister a stream
+ *
+ * @fedai_id: Frontend dai id.
+ * @stream_type: Direction of stream
+ *
+ * Return: Will be an negative on error or a zero on success.
+ */
+void q6routing_stream_close(int fedai_id, int stream_type)
+{
+ struct session_data *session;
+ int idx;
+
+ session = get_session_from_id(routing_data, fedai_id);
+ if (!session)
+ return;
+
+ for_each_set_bit(idx, &session->copp_map, MAX_COPPS_PER_PORT) {
+ if (session->copps[idx]) {
+ q6adm_close(routing_data->dev, session->copps[idx]);
+ session->copps[idx] = NULL;
+ }
+ }
+
+ session->fedai_id = -1;
+ session->copp_map = 0;
+}
+EXPORT_SYMBOL_GPL(q6routing_stream_close);
+
+static int msm_routing_get_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ int session_id = mc->shift;
+ struct snd_soc_component *c = snd_soc_dapm_to_component(dapm);
+ struct msm_routing_data *priv = dev_get_drvdata(c->dev);
+ struct session_data *session = &priv->sessions[session_id];
+
+ if (session->port_id == mc->reg)
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+static int msm_routing_put_audio_mixer(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dapm_context *dapm =
+ snd_soc_dapm_kcontrol_dapm(kcontrol);
+ struct snd_soc_component *c = snd_soc_dapm_to_component(dapm);
+ struct msm_routing_data *data = dev_get_drvdata(c->dev);
+ struct soc_mixer_control *mc =
+ (struct soc_mixer_control *)kcontrol->private_value;
+ struct snd_soc_dapm_update *update = NULL;
+ int be_id = mc->reg;
+ int session_id = mc->shift;
+ struct session_data *session = &data->sessions[session_id];
+
+ if (ucontrol->value.integer.value[0]) {
+ session->port_id = be_id;
+ snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, update);
+ } else {
+ session->port_id = -1;
+ snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, update);
+ }
+
+ return 1;
+}
+
+static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", HDMI_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0,
+ msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
+ /* Frontend AIF */
+ SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL2", "MultiMedia2 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL3", "MultiMedia3 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL4", "MultiMedia4 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL5", "MultiMedia5 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL6", "MultiMedia6 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL7", "MultiMedia7 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("MM_DL8", "MultiMedia8 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL3", "MultiMedia3 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL4", "MultiMedia4 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL5", "MultiMedia5 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL6", "MultiMedia6 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL7", "MultiMedia7 Capture", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("MM_UL8", "MultiMedia8 Capture", 0, 0, 0, 0),
+
+ /* Mixer definitions */
+ SND_SOC_DAPM_MIXER("HDMI Mixer", SND_SOC_NOPM, 0, 0,
+ hdmi_mixer_controls,
+ ARRAY_SIZE(hdmi_mixer_controls)),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+ {"HDMI Mixer", "MultiMedia1", "MM_DL1"},
+ {"HDMI Mixer", "MultiMedia2", "MM_DL2"},
+ {"HDMI Mixer", "MultiMedia3", "MM_DL3"},
+ {"HDMI Mixer", "MultiMedia4", "MM_DL4"},
+ {"HDMI Mixer", "MultiMedia5", "MM_DL5"},
+ {"HDMI Mixer", "MultiMedia6", "MM_DL6"},
+ {"HDMI Mixer", "MultiMedia7", "MM_DL7"},
+ {"HDMI Mixer", "MultiMedia8", "MM_DL8"},
+ {"HDMI_RX", NULL, "HDMI Mixer"},
+};
+
+static int routing_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+ struct msm_routing_data *data = dev_get_drvdata(c->dev);
+ unsigned int be_id = rtd->cpu_dai->id;
+ struct session_data *session;
+ int path_type;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ path_type = ADM_PATH_PLAYBACK;
+ else
+ path_type = ADM_PATH_LIVE_REC;
+
+ if (be_id > AFE_MAX_PORTS)
+ return -EINVAL;
+
+ session = &data->port_data[be_id];
+
+ mutex_lock(&data->lock);
+
+ session->path_type = path_type;
+ session->sample_rate = params_rate(params);
+ session->channels = params_channels(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ session->bits_per_sample = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ session->bits_per_sample = 24;
+ break;
+ default:
+ break;
+ }
+
+ mutex_unlock(&data->lock);
+ return 0;
+}
+
+static struct snd_pcm_ops q6pcm_routing_ops = {
+ .hw_params = routing_hw_params,
+};
+
+static int msm_routing_probe(struct snd_soc_component *c)
+{
+ int i;
+
+ for (i = 0; i < MAX_SESSIONS; i++)
+ routing_data->sessions[i].port_id = -1;
+
+ return 0;
+}
+
+static const struct snd_soc_component_driver msm_soc_routing_component = {
+ .ops = &q6pcm_routing_ops,
+ .probe = msm_routing_probe,
+ .name = DRV_NAME,
+ .dapm_widgets = msm_qdsp6_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(msm_qdsp6_widgets),
+ .dapm_routes = intercon,
+ .num_dapm_routes = ARRAY_SIZE(intercon),
+};
+
+static int q6routing_dai_bind(struct device *dev, struct device *master,
+ void *data)
+{
+ routing_data = kzalloc(sizeof(*routing_data), GFP_KERNEL);
+ if (!routing_data)
+ return -ENOMEM;
+
+ routing_data->dev = dev;
+
+ mutex_init(&routing_data->lock);
+ dev_set_drvdata(dev, routing_data);
+
+ return snd_soc_register_component(dev, &msm_soc_routing_component,
+ NULL, 0);
+}
+
+static void q6routing_dai_unbind(struct device *dev, struct device *master,
+ void *d)
+{
+ struct msm_routing_data *data = dev_get_drvdata(dev);
+
+ snd_soc_unregister_component(dev);
+
+ kfree(data);
+
+ routing_data = NULL;
+}
+
+static const struct component_ops q6routing_dai_comp_ops = {
+ .bind = q6routing_dai_bind,
+ .unbind = q6routing_dai_unbind,
+};
+
+static int q6pcm_routing_probe(struct platform_device *pdev)
+{
+ return component_add(&pdev->dev, &q6routing_dai_comp_ops);
+}
+
+static int q6pcm_routing_remove(struct platform_device *pdev)
+{
+ component_del(&pdev->dev, &q6routing_dai_comp_ops);
+ return 0;
+}
+
+static struct platform_driver q6pcm_routing_platform_driver = {
+ .driver = {
+ .name = "q6routing",
+ },
+ .probe = q6pcm_routing_probe,
+ .remove = q6pcm_routing_remove,
+};
+module_platform_driver(q6pcm_routing_platform_driver);
+
+MODULE_DESCRIPTION("Q6 Routing platform");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6routing.h b/sound/soc/qcom/qdsp6/q6routing.h
new file mode 100644
index 000000000000..35514e651130
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6routing.h
@@ -0,0 +1,9 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+#ifndef _Q6_PCM_ROUTING_H
+#define _Q6_PCM_ROUTING_H
+
+int q6routing_stream_open(int fedai_id, int perf_mode,
+ int stream_id, int stream_type);
+void q6routing_stream_close(int fedai_id, int stream_type);
+
+#endif /*_Q6_PCM_ROUTING_H */
--
2.17.0
^ permalink raw reply related [flat|nested] 25+ messages in thread
* [PATCH v9 10/15] ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (8 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 09/15] ASoC: qdsp6: q6routing: Add q6routing driver Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-18 12:56 ` [PATCH v9 11/15] ASoC: qdsp6: q6routing: Add support to MI2S Mixers Srinivas Kandagatla
` (4 subsequent siblings)
14 siblings, 0 replies; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
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>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.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 1d7a4088a435..43086913060a 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -241,6 +241,180 @@ 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),
@@ -264,6 +438,28 @@ 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[] = {
@@ -276,6 +472,71 @@ static const struct snd_soc_dapm_route intercon[] = {
{"HDMI Mixer", "MultiMedia7", "MM_DL7"},
{"HDMI Mixer", "MultiMedia8", "MM_DL8"},
{"HDMI_RX", NULL, "HDMI Mixer"},
+
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_0_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_0_RX", NULL, "SLIMBUS_0_RX Audio Mixer"},
+
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_1_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_1_RX", NULL, "SLIMBUS_1_RX Audio Mixer"},
+
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_2_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_2_RX", NULL, "SLIMBUS_2_RX Audio Mixer"},
+
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_3_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_3_RX", NULL, "SLIMBUS_3_RX Audio Mixer"},
+
+ {"SLIMBUS_4_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_4_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_4_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_4_RX", NULL, "SLIMBUS_4_RX Audio Mixer"},
+
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_5_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_5_RX", NULL, "SLIMBUS_5_RX Audio Mixer"},
+
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SLIMBUS_6_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"SLIMBUS_6_RX", NULL, "SLIMBUS_6_RX Audio Mixer"},
};
static int routing_hw_params(struct snd_pcm_substream *substream,
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* [PATCH v9 11/15] ASoC: qdsp6: q6routing: Add support to MI2S Mixers
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (9 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 10/15] ASoC: qdsp6: q6routing: Add support to all SLIMBus Mixers Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-18 12:56 ` [PATCH v9 12/15] ASoC: qdsp6: q6afe: Add q6afe dai driver Srinivas Kandagatla
` (3 subsequent siblings)
14 siblings, 0 replies; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
This patch add support to MI2S mixers required to select path between
ASM stream and AFE ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/qdsp6/q6routing.c | 329 +++++++++++++++++++++++++++++++++++++++
1 file changed, 329 insertions(+)
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 43086913060a..08c25c26adf4 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -241,6 +241,103 @@ static const struct snd_kcontrol_new hdmi_mixer_controls[] = {
msm_routing_put_audio_mixer),
};
+static const struct snd_kcontrol_new primary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", PRIMARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new secondary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", SECONDARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new quaternary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia5", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia6", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia7", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia8", QUATERNARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new tertiary_mi2s_rx_mixer_controls[] = {
+ SOC_SINGLE_EXT("MultiMedia1", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia2", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia3", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("MultiMedia4", TERTIARY_MI2S_RX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+
static const struct snd_kcontrol_new slimbus_rx_mixer_controls[] = {
SOC_SINGLE_EXT("MultiMedia1", SLIMBUS_0_RX,
MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
@@ -415,6 +512,126 @@ static const struct snd_kcontrol_new slimbus_6_rx_mixer_controls[] = {
msm_routing_put_audio_mixer),
};
+static const struct snd_kcontrol_new mmul1_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA1, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul2_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA2, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul3_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA3, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul4_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA4, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul5_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA5, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul6_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA6, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul7_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA7, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
+static const struct snd_kcontrol_new mmul8_mixer_controls[] = {
+ SOC_SINGLE_EXT("PRI_MI2S_TX", PRIMARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("QUAT_MI2S_TX", QUATERNARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("TERT_MI2S_TX", TERTIARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+ SOC_SINGLE_EXT("SEC_MI2S_TX", SECONDARY_MI2S_TX,
+ MSM_FRONTEND_DAI_MULTIMEDIA8, 1, 0, msm_routing_get_audio_mixer,
+ msm_routing_put_audio_mixer),
+};
+
static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
/* Frontend AIF */
SND_SOC_DAPM_AIF_IN("MM_DL1", "MultiMedia1 Playback", 0, 0, 0, 0),
@@ -460,6 +677,35 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = {
SND_SOC_DAPM_MIXER("SLIMBUS_6_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
slimbus_6_rx_mixer_controls,
ARRAY_SIZE(slimbus_6_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("PRI_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ primary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(primary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("SEC_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ secondary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(secondary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("QUAT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ quaternary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(quaternary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("TERT_MI2S_RX Audio Mixer", SND_SOC_NOPM, 0, 0,
+ tertiary_mi2s_rx_mixer_controls,
+ ARRAY_SIZE(tertiary_mi2s_rx_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia1 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul1_mixer_controls, ARRAY_SIZE(mmul1_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia2 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul2_mixer_controls, ARRAY_SIZE(mmul2_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia3 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul3_mixer_controls, ARRAY_SIZE(mmul3_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia4 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul4_mixer_controls, ARRAY_SIZE(mmul4_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia5 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul5_mixer_controls, ARRAY_SIZE(mmul5_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia6 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul6_mixer_controls, ARRAY_SIZE(mmul6_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia7 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul7_mixer_controls, ARRAY_SIZE(mmul7_mixer_controls)),
+ SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0,
+ mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)),
+
};
static const struct snd_soc_dapm_route intercon[] = {
@@ -537,6 +783,89 @@ static const struct snd_soc_dapm_route intercon[] = {
{"SLIMBUS_6_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
{"SLIMBUS_6_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
{"SLIMBUS_6_RX", NULL, "SLIMBUS_6_RX Audio Mixer"},
+
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia6", "MM_DL6"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"QUAT_MI2S_RX Audio Mixer", "MultiMedia8", "MM_DL8"},
+ {"QUAT_MI2S_RX", NULL, "QUAT_MI2S_RX Audio Mixer"},
+
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"TERT_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"TERT_MI2S_RX", NULL, "TERT_MI2S_RX Audio Mixer"},
+
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia6", "MM_DL5"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"SEC_MI2S_RX Audio Mixer", "MultiMedia8", "MM_DL7"},
+ {"SEC_MI2S_RX", NULL, "SEC_MI2S_RX Audio Mixer"},
+
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia1", "MM_DL1"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia2", "MM_DL2"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia3", "MM_DL3"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia4", "MM_DL4"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia5", "MM_DL5"},
+ {"PRI_MI2S_RX Audio Mixer", "MultiMedia7", "MM_DL7"},
+ {"PRI_MI2S_RX", NULL, "PRI_MI2S_RX Audio Mixer"},
+
+ {"MultiMedia1 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia1 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia1 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia1 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia2 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia2 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia2 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia2 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia3 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia3 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia3 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia3 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia4 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia4 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia4 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia4 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia5 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia5 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia5 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia5 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia6 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia6 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia6 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia6 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia7 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia7 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia7 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia7 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MultiMedia8 Mixer", "PRI_MI2S_TX", "PRI_MI2S_TX"},
+ {"MultiMedia8 Mixer", "SEC_MI2S_TX", "SEC_MI2S_TX"},
+ {"MultiMedia8 Mixer", "QUAT_MI2S_TX", "QUAT_MI2S_TX"},
+ {"MultiMedia8 Mixer", "TERT_MI2S_TX", "TERT_MI2S_TX"},
+
+ {"MM_UL1", NULL, "MultiMedia1 Mixer"},
+ {"MM_UL2", NULL, "MultiMedia2 Mixer"},
+ {"MM_UL3", NULL, "MultiMedia3 Mixer"},
+ {"MM_UL4", NULL, "MultiMedia4 Mixer"},
+ {"MM_UL5", NULL, "MultiMedia5 Mixer"},
+ {"MM_UL6", NULL, "MultiMedia6 Mixer"},
+ {"MM_UL7", NULL, "MultiMedia7 Mixer"},
+ {"MM_UL8", NULL, "MultiMedia8 Mixer"},
};
static int routing_hw_params(struct snd_pcm_substream *substream,
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* [PATCH v9 12/15] ASoC: qdsp6: q6afe: Add q6afe dai driver
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (10 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 11/15] ASoC: qdsp6: q6routing: Add support to MI2S Mixers Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-18 12:56 ` [PATCH v9 13/15] ASoC: qdsp6: q6asm: Add q6asm " Srinivas Kandagatla
` (2 subsequent siblings)
14 siblings, 0 replies; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
This patch adds support to q6afe backend dais driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6afe-dai.c | 748 +++++++++++++++++++++++++++++++++++++++
3 files changed, 753 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6afe-dai.c
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 43f9ed85efa8..d3523a30d942 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -50,6 +50,9 @@ config SND_SOC_QDSP6_CORE
config SND_SOC_QDSP6_AFE
tristate
+config SND_SOC_QDSP6_AFE_DAI
+ tristate
+
config SND_SOC_QDSP6_ADM
tristate
@@ -65,6 +68,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_COMMON
select SND_SOC_QDSP6_CORE
select SND_SOC_QDSP6_AFE
+ select SND_SOC_QDSP6_AFE_DAI
select SND_SOC_QDSP6_ADM
select SND_SOC_QDSP6_ROUTING
select SND_SOC_QDSP6_ASM
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 0e8e2febb7ec..bada1aa303c2 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -1,6 +1,7 @@
obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
obj-$(CONFIG_SND_SOC_QDSP6_AFE) += q6afe.o
+obj-$(CONFIG_SND_SOC_QDSP6_AFE_DAI) += q6afe-dai.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
new file mode 100644
index 000000000000..4378e29a95c5
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -0,0 +1,748 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/err.h>
+#include <linux/init.h>
+#include <linux/component.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include "q6afe.h"
+
+struct q6afe_dai_priv_data {
+ uint32_t sd_line_mask;
+};
+
+struct q6afe_dai_data {
+ struct q6afe_port *port[AFE_PORT_MAX];
+ struct q6afe_port_config port_config[AFE_PORT_MAX];
+ bool is_port_started[AFE_PORT_MAX];
+ struct q6afe_dai_priv_data priv[AFE_PORT_MAX];
+};
+
+static int q6slim_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_slim_cfg *slim = &dai_data->port_config[dai->id].slim;
+
+ slim->num_channels = params_channels(params);
+ slim->sample_rate = params_rate(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ case SNDRV_PCM_FORMAT_SPECIAL:
+ slim->bit_width = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ slim->bit_width = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ slim->bit_width = 32;
+ break;
+ default:
+ pr_err("%s: format %d\n",
+ __func__, params_format(params));
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int q6hdmi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int channels = params_channels(params);
+ struct q6afe_hdmi_cfg *hdmi = &dai_data->port_config[dai->id].hdmi;
+
+ hdmi->sample_rate = params_rate(params);
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ hdmi->bit_width = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ hdmi->bit_width = 24;
+ break;
+ }
+
+ /* HDMI spec CEA-861-E: Table 28 Audio InfoFrame Data Byte 4 */
+ switch (channels) {
+ case 2:
+ hdmi->channel_allocation = 0;
+ break;
+ case 3:
+ hdmi->channel_allocation = 0x02;
+ break;
+ case 4:
+ hdmi->channel_allocation = 0x06;
+ break;
+ case 5:
+ hdmi->channel_allocation = 0x0A;
+ break;
+ case 6:
+ hdmi->channel_allocation = 0x0B;
+ break;
+ case 7:
+ hdmi->channel_allocation = 0x12;
+ break;
+ case 8:
+ hdmi->channel_allocation = 0x13;
+ break;
+ default:
+ dev_err(dai->dev, "invalid Channels = %u\n", channels);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int q6i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_i2s_cfg *i2s = &dai_data->port_config[dai->id].i2s_cfg;
+
+ i2s->sample_rate = params_rate(params);
+ i2s->bit_width = params_width(params);
+ i2s->num_channels = params_channels(params);
+ i2s->sd_line_mask = dai_data->priv[dai->id].sd_line_mask;
+
+ return 0;
+}
+
+static int q6i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_i2s_cfg *i2s = &dai_data->port_config[dai->id].i2s_cfg;
+
+ i2s->fmt = fmt;
+
+ return 0;
+}
+
+static void q6afe_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc;
+
+ rc = q6afe_port_stop(dai_data->port[dai->id]);
+ if (rc < 0)
+ dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
+
+ dai_data->is_port_started[dai->id] = false;
+
+}
+
+static int q6afe_mi2s_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc;
+
+ if (dai_data->is_port_started[dai->id]) {
+ /* stop the port and restart with new port config */
+ rc = q6afe_port_stop(dai_data->port[dai->id]);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
+ return rc;
+ }
+ }
+
+ rc = q6afe_i2s_port_prepare(dai_data->port[dai->id],
+ &dai_data->port_config[dai->id].i2s_cfg);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to prepare AFE port %x\n", dai->id);
+ return rc;
+ }
+
+ rc = q6afe_port_start(dai_data->port[dai->id]);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to start AFE port %x\n", dai->id);
+ return rc;
+ }
+ dai_data->is_port_started[dai->id] = true;
+
+ return 0;
+}
+
+static int q6afe_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ int rc;
+
+ if (dai_data->is_port_started[dai->id]) {
+ /* stop the port and restart with new port config */
+ rc = q6afe_port_stop(dai_data->port[dai->id]);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to close AFE port (%d)\n", rc);
+ return rc;
+ }
+ }
+
+ if (dai->id == HDMI_RX)
+ q6afe_hdmi_port_prepare(dai_data->port[dai->id],
+ &dai_data->port_config[dai->id].hdmi);
+ else if (dai->id >= SLIMBUS_0_RX && dai->id <= SLIMBUS_6_TX)
+ q6afe_slim_port_prepare(dai_data->port[dai->id],
+ &dai_data->port_config[dai->id].slim);
+
+ rc = q6afe_port_start(dai_data->port[dai->id]);
+ if (rc < 0) {
+ dev_err(dai->dev, "fail to start AFE port %x\n", dai->id);
+ return rc;
+ }
+ dai_data->is_port_started[dai->id] = true;
+
+ return 0;
+}
+
+static int q6slim_set_channel_map(struct snd_soc_dai *dai,
+ unsigned int tx_num, unsigned int *tx_slot,
+ unsigned int rx_num, unsigned int *rx_slot)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id];
+ int i;
+
+ if (!rx_slot) {
+ pr_err("%s: rx slot not found\n", __func__);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < rx_num; i++) {
+ pcfg->slim.ch_mapping[i] = rx_slot[i];
+ pr_debug("%s: find number of channels[%d] ch[%d]\n",
+ __func__, i, rx_slot[i]);
+ }
+
+ pcfg->slim.num_channels = rx_num;
+
+ pr_debug("%s: SLIMBUS_%d_RX cnt[%d] ch[%d %d]\n", __func__,
+ (dai->id - SLIMBUS_0_RX) / 2, rx_num,
+ pcfg->slim.ch_mapping[0],
+ pcfg->slim.ch_mapping[1]);
+
+ return 0;
+}
+
+static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_port *port = dai_data->port[dai->id];
+
+ switch (clk_id) {
+ case LPAIF_DIG_CLK:
+ return q6afe_port_set_sysclk(port, clk_id, 0, 5, freq, dir);
+ case LPAIF_BIT_CLK:
+ case LPAIF_OSR_CLK:
+ return q6afe_port_set_sysclk(port, clk_id,
+ Q6AFE_LPASS_CLK_SRC_INTERNAL,
+ Q6AFE_LPASS_CLK_ROOT_DEFAULT,
+ freq, dir);
+ case Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT ... Q6AFE_LPASS_CLK_ID_INT_MCLK_1:
+ return q6afe_port_set_sysclk(port, clk_id,
+ Q6AFE_LPASS_CLK_ATTRIBUTE_COUPLE_NO,
+ Q6AFE_LPASS_CLK_ROOT_DEFAULT,
+ freq, dir);
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_route q6afe_dapm_routes[] = {
+ {"HDMI Playback", NULL, "HDMI_RX"},
+ {"Slimbus1 Playback", NULL, "SLIMBUS_1_RX"},
+ {"Slimbus2 Playback", NULL, "SLIMBUS_2_RX"},
+ {"Slimbus3 Playback", NULL, "SLIMBUS_3_RX"},
+ {"Slimbus4 Playback", NULL, "SLIMBUS_4_RX"},
+ {"Slimbus5 Playback", NULL, "SLIMBUS_5_RX"},
+ {"Slimbus6 Playback", NULL, "SLIMBUS_6_RX"},
+
+ {"Primary MI2S Playback", NULL, "PRI_MI2S_RX"},
+ {"Secondary MI2S Playback", NULL, "SEC_MI2S_RX"},
+ {"Tertiary MI2S Playback", NULL, "TERT_MI2S_RX"},
+ {"Quaternary MI2S Playback", NULL, "QUAT_MI2S_RX"},
+
+ {"TERT_MI2S_TX", NULL, "Tertiary MI2S Capture"},
+ {"PRI_MI2S_TX", NULL, "Primary MI2S Capture"},
+ {"SEC_MI2S_TX", NULL, "Secondary MI2S Capture"},
+ {"QUAT_MI2S_TX", NULL, "Quaternary MI2S Capture"},
+};
+
+static struct snd_soc_dai_ops q6hdmi_ops = {
+ .prepare = q6afe_dai_prepare,
+ .hw_params = q6hdmi_hw_params,
+ .shutdown = q6afe_dai_shutdown,
+};
+
+static struct snd_soc_dai_ops q6i2s_ops = {
+ .prepare = q6afe_mi2s_prepare,
+ .hw_params = q6i2s_hw_params,
+ .set_fmt = q6i2s_set_fmt,
+ .shutdown = q6afe_dai_shutdown,
+ .set_sysclk = q6afe_mi2s_set_sysclk,
+};
+
+static struct snd_soc_dai_ops q6slim_ops = {
+ .prepare = q6afe_dai_prepare,
+ .hw_params = q6slim_hw_params,
+ .shutdown = q6afe_dai_shutdown,
+ .set_channel_map = q6slim_set_channel_map,
+};
+
+static int msm_dai_q6_dai_probe(struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+ struct q6afe_port *port;
+
+ port = q6afe_port_get_from_id(dai->dev, dai->id);
+ if (IS_ERR(port)) {
+ dev_err(dai->dev, "Unable to get afe port\n");
+ return -EINVAL;
+ }
+ dai_data->port[dai->id] = port;
+
+ return 0;
+}
+
+static int msm_dai_q6_dai_remove(struct snd_soc_dai *dai)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
+
+ q6afe_port_put(dai_data->port[dai->id]);
+ dai_data->port[dai->id] = NULL;
+
+ return 0;
+}
+
+static struct snd_soc_dai_driver q6afe_dais[] = {
+ {
+ .playback = {
+ .stream_name = "HDMI Playback",
+ .rates = SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 2,
+ .channels_max = 8,
+ .rate_max = 192000,
+ .rate_min = 48000,
+ },
+ .ops = &q6hdmi_ops,
+ .id = HDMI_RX,
+ .name = "HDMI",
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .name = "SLIMBUS_0_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_0_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ .playback = {
+ .stream_name = "Slimbus Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ }, {
+ .playback = {
+ .stream_name = "Slimbus1 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_1_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_1_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus2 Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 8,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_2_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_2_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus3 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_3_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_3_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus4 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_4_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_4_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus5 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .name = "SLIMBUS_5_RX",
+ .ops = &q6slim_ops,
+ .id = SLIMBUS_5_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Slimbus6 Playback",
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .channels_min = 1,
+ .channels_max = 2,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ },
+ .ops = &q6slim_ops,
+ .name = "SLIMBUS_6_RX",
+ .id = SLIMBUS_6_RX,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Primary MI2S Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = PRIMARY_MI2S_RX,
+ .name = "PRI_MI2S_RX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .capture = {
+ .stream_name = "Primary MI2S Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = PRIMARY_MI2S_TX,
+ .name = "PRI_MI2S_TX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Secondary MI2S Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .name = "SEC_MI2S_RX",
+ .id = SECONDARY_MI2S_RX,
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .capture = {
+ .stream_name = "Secondary MI2S Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = SECONDARY_MI2S_TX,
+ .name = "SEC_MI2S_TX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Tertiary MI2S Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .name = "TERT_MI2S_RX",
+ .id = TERTIARY_MI2S_RX,
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .capture = {
+ .stream_name = "Tertiary MI2S Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = TERTIARY_MI2S_TX,
+ .name = "TERT_MI2S_TX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .playback = {
+ .stream_name = "Quaternary MI2S Playback",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .name = "QUAT_MI2S_RX",
+ .id = QUATERNARY_MI2S_RX,
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ }, {
+ .capture = {
+ .stream_name = "Quaternary MI2S Capture",
+ .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
+ SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ },
+ .id = QUATERNARY_MI2S_TX,
+ .name = "QUAT_MI2S_TX",
+ .ops = &q6i2s_ops,
+ .probe = msm_dai_q6_dai_probe,
+ .remove = msm_dai_q6_dai_remove,
+ },
+};
+
+static int q6afe_of_xlate_dai_name(struct snd_soc_component *component,
+ struct of_phandle_args *args,
+ const char **dai_name)
+{
+ int id = args->args[0];
+ int ret = -EINVAL;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(q6afe_dais); i++) {
+ if (q6afe_dais[i].id == id) {
+ *dai_name = q6afe_dais[i].name;
+ ret = 0;
+ break;
+ }
+ }
+
+ return ret;
+}
+
+static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = {
+ SND_SOC_DAPM_AIF_OUT("HDMI_RX", "HDMI Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_RX", "Slimbus Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_RX", "Slimbus1 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_RX", "Slimbus2 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_RX", "Slimbus3 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_RX", "Slimbus4 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_RX", "Slimbus5 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_RX", "Slimbus6 Playback", 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_RX", "Quaternary MI2S Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("QUAT_MI2S_TX", "Quaternary MI2S Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("TERT_MI2S_RX", "Tertiary MI2S Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("TERT_MI2S_TX", "Tertiary MI2S Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX", "Secondary MI2S Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("SEC_MI2S_TX", "Secondary MI2S Capture",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("SEC_MI2S_RX_SD1",
+ "Secondary MI2S Playback SD1",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_OUT("PRI_MI2S_RX", "Primary MI2S Playback",
+ 0, 0, 0, 0),
+ SND_SOC_DAPM_AIF_IN("PRI_MI2S_TX", "Primary MI2S Capture",
+ 0, 0, 0, 0),
+};
+
+static const struct snd_soc_component_driver q6afe_dai_component = {
+ .name = "q6afe-dai-component",
+ .dapm_widgets = q6afe_dai_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(q6afe_dai_widgets),
+ .dapm_routes = q6afe_dapm_routes,
+ .num_dapm_routes = ARRAY_SIZE(q6afe_dapm_routes),
+ .of_xlate_dai_name = q6afe_of_xlate_dai_name,
+
+};
+
+static void of_q6afe_parse_dai_data(struct device *dev,
+ struct q6afe_dai_data *data)
+{
+ struct device_node *node;
+ int ret;
+
+ for_each_child_of_node(dev->of_node, node) {
+ unsigned int lines[Q6AFE_MAX_MI2S_LINES];
+ struct q6afe_dai_priv_data *priv;
+ int id, i, num_lines;
+
+ ret = of_property_read_u32(node, "reg", &id);
+ if (ret || id > AFE_PORT_MAX) {
+ dev_err(dev, "valid dai id not found:%d\n", ret);
+ continue;
+ }
+
+ switch (id) {
+ /* MI2S specific properties */
+ case PRIMARY_MI2S_RX ... QUATERNARY_MI2S_TX:
+ priv = &data->priv[id];
+ ret = of_property_read_variable_u32_array(node,
+ "qcom,sd-lines",
+ lines, 0,
+ Q6AFE_MAX_MI2S_LINES);
+ if (ret < 0)
+ num_lines = 0;
+ else
+ num_lines = ret;
+
+ priv->sd_line_mask = 0;
+
+ for (i = 0; i < num_lines; i++)
+ priv->sd_line_mask |= BIT(lines[i]);
+
+ break;
+ default:
+ break;
+ }
+ }
+}
+
+static int q6afe_dai_bind(struct device *dev, struct device *master, void *data)
+{
+ struct q6afe_dai_data *dai_data;
+
+ dai_data = kzalloc(sizeof(*dai_data), GFP_KERNEL);
+ if (!dai_data)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, dai_data);
+
+ of_q6afe_parse_dai_data(dev, dai_data);
+
+ return snd_soc_register_component(dev, &q6afe_dai_component,
+ q6afe_dais, ARRAY_SIZE(q6afe_dais));
+}
+
+static void q6afe_dai_unbind(struct device *dev, struct device *master,
+ void *data)
+{
+ struct q6afe_dai_data *dai_data = dev_get_drvdata(dev);
+
+ snd_soc_unregister_component(dev);
+ kfree(dai_data);
+}
+
+static const struct component_ops q6afe_dai_comp_ops = {
+ .bind = q6afe_dai_bind,
+ .unbind = q6afe_dai_unbind,
+};
+
+static int q6afe_dai_dev_probe(struct platform_device *pdev)
+{
+ return component_add(&pdev->dev, &q6afe_dai_comp_ops);
+}
+
+static int q6afe_dai_dev_remove(struct platform_device *pdev)
+{
+ component_del(&pdev->dev, &q6afe_dai_comp_ops);
+ return 0;
+}
+
+static struct platform_driver q6afe_dai_platform_driver = {
+ .driver = {
+ .name = "q6afe-dai",
+ },
+ .probe = q6afe_dai_dev_probe,
+ .remove = q6afe_dai_dev_remove,
+};
+module_platform_driver(q6afe_dai_platform_driver);
+
+MODULE_DESCRIPTION("Q6 Audio Fronend dai driver");
+MODULE_LICENSE("GPL v2");
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* [PATCH v9 13/15] ASoC: qdsp6: q6asm: Add q6asm dai driver
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (11 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 12/15] ASoC: qdsp6: q6afe: Add q6afe dai driver Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-18 12:56 ` [PATCH v9 14/15] ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings Srinivas Kandagatla
2018-05-18 12:56 ` [PATCH v9 15/15] ASoC: qcom: apq8096: Add db820c machine driver Srinivas Kandagatla
14 siblings, 0 replies; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
This patch adds support to q6asm dai driver which configures Q6ASM streams
to pass pcm data.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/Kconfig | 4 +
sound/soc/qcom/qdsp6/Makefile | 1 +
sound/soc/qcom/qdsp6/q6asm-dai.c | 624 +++++++++++++++++++++++++++++++++++++++
3 files changed, 629 insertions(+)
create mode 100644 sound/soc/qcom/qdsp6/q6asm-dai.c
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index d3523a30d942..85bb7dd11fd9 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -62,6 +62,9 @@ config SND_SOC_QDSP6_ROUTING
config SND_SOC_QDSP6_ASM
tristate
+config SND_SOC_QDSP6_ASM_DAI
+ tristate
+
config SND_SOC_QDSP6
tristate "SoC ALSA audio driver for QDSP6"
depends on QCOM_APR && HAS_DMA
@@ -72,6 +75,7 @@ config SND_SOC_QDSP6
select SND_SOC_QDSP6_ADM
select SND_SOC_QDSP6_ROUTING
select SND_SOC_QDSP6_ASM
+ select SND_SOC_QDSP6_ASM_DAI
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 bada1aa303c2..c33b3cacbea1 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -5,3 +5,4 @@ obj-$(CONFIG_SND_SOC_QDSP6_AFE_DAI) += q6afe-dai.o
obj-$(CONFIG_SND_SOC_QDSP6_ADM) += q6adm.o
obj-$(CONFIG_SND_SOC_QDSP6_ROUTING) += q6routing.o
obj-$(CONFIG_SND_SOC_QDSP6_ASM) += q6asm.o
+obj-$(CONFIG_SND_SOC_QDSP6_ASM_DAI) += q6asm-dai.o
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
new file mode 100644
index 000000000000..349c6a883c63
--- /dev/null
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -0,0 +1,624 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/component.h>
+#include <sound/soc.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/of_device.h>
+#include <sound/pcm_params.h>
+#include "q6asm.h"
+#include "q6routing.h"
+#include "q6dsp-errno.h"
+
+#define DRV_NAME "q6asm-fe-dai"
+
+#define PLAYBACK_MIN_NUM_PERIODS 2
+#define PLAYBACK_MAX_NUM_PERIODS 8
+#define PLAYBACK_MAX_PERIOD_SIZE 65536
+#define PLAYBACK_MIN_PERIOD_SIZE 128
+#define CAPTURE_MIN_NUM_PERIODS 2
+#define CAPTURE_MAX_NUM_PERIODS 8
+#define CAPTURE_MAX_PERIOD_SIZE 4096
+#define CAPTURE_MIN_PERIOD_SIZE 320
+#define SID_MASK_DEFAULT 0xF
+
+enum stream_state {
+ Q6ASM_STREAM_IDLE = 0,
+ Q6ASM_STREAM_STOPPED,
+ Q6ASM_STREAM_RUNNING,
+};
+
+struct q6asm_dai_rtd {
+ struct snd_pcm_substream *substream;
+ phys_addr_t phys;
+ unsigned int pcm_size;
+ unsigned int pcm_count;
+ unsigned int pcm_irq_pos; /* IRQ position */
+ unsigned int periods;
+ uint16_t bits_per_sample;
+ uint16_t source; /* Encoding source bit mask */
+ struct audio_client *audio_client;
+ uint16_t session_id;
+ enum stream_state state;
+};
+
+struct q6asm_dai_data {
+ long long int sid;
+};
+
+static struct snd_pcm_hardware q6asm_dai_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 4,
+ .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS *
+ CAPTURE_MAX_PERIOD_SIZE,
+ .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE,
+ .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE,
+ .periods_min = CAPTURE_MIN_NUM_PERIODS,
+ .periods_max = CAPTURE_MAX_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE),
+ .rates = SNDRV_PCM_RATE_8000_192000,
+ .rate_min = 8000,
+ .rate_max = 192000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = (PLAYBACK_MAX_NUM_PERIODS *
+ PLAYBACK_MAX_PERIOD_SIZE),
+ .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE,
+ .periods_min = PLAYBACK_MIN_NUM_PERIODS,
+ .periods_max = PLAYBACK_MAX_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+#define Q6ASM_FEDAI_DRIVER(num) { \
+ .playback = { \
+ .stream_name = "MultiMedia"#num" Playback", \
+ .rates = (SNDRV_PCM_RATE_8000_192000| \
+ SNDRV_PCM_RATE_KNOT), \
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE), \
+ .channels_min = 1, \
+ .channels_max = 8, \
+ .rate_min = 8000, \
+ .rate_max = 192000, \
+ }, \
+ .capture = { \
+ .stream_name = "MultiMedia"#num" Capture", \
+ .rates = (SNDRV_PCM_RATE_8000_48000| \
+ SNDRV_PCM_RATE_KNOT), \
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S24_LE), \
+ .channels_min = 1, \
+ .channels_max = 4, \
+ .rate_min = 8000, \
+ .rate_max = 48000, \
+ }, \
+ .name = "MultiMedia"#num, \
+ .probe = fe_dai_probe, \
+ .id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \
+ }
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
+ 88200, 96000, 176400, 192000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void event_handler(uint32_t opcode, uint32_t token,
+ uint32_t *payload, void *priv)
+{
+ struct q6asm_dai_rtd *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+
+ switch (opcode) {
+ case ASM_CLIENT_EVENT_CMD_RUN_DONE:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ q6asm_write_async(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ break;
+ case ASM_CLIENT_EVENT_CMD_EOS_DONE:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ break;
+ case ASM_CLIENT_EVENT_DATA_WRITE_DONE: {
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ snd_pcm_period_elapsed(substream);
+ if (prtd->state == Q6ASM_STREAM_RUNNING)
+ q6asm_write_async(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+
+ break;
+ }
+ case ASM_CLIENT_EVENT_DATA_READ_DONE:
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ snd_pcm_period_elapsed(substream);
+ if (prtd->state == Q6ASM_STREAM_RUNNING)
+ q6asm_read(prtd->audio_client);
+
+ break;
+ default:
+ break;
+ }
+}
+
+static int q6asm_dai_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+ struct snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME);
+ struct q6asm_dai_data *pdata;
+ int ret, i;
+
+ pdata = snd_soc_component_get_drvdata(c);
+ if (!pdata)
+ return -EINVAL;
+
+ if (!prtd || !prtd->audio_client) {
+ pr_err("%s: private data null or audio client freed\n",
+ __func__);
+ return -EINVAL;
+ }
+
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ if (prtd->state) {
+ /* clear the previous setup if any */
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_unmap_memory_regions(substream->stream,
+ prtd->audio_client);
+ q6routing_stream_close(soc_prtd->dai_link->id,
+ substream->stream);
+ }
+
+ ret = q6asm_map_memory_regions(substream->stream, prtd->audio_client,
+ prtd->phys,
+ (prtd->pcm_size / prtd->periods),
+ prtd->periods);
+
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
+ ret);
+ return -ENOMEM;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = q6asm_open_write(prtd->audio_client, FORMAT_LINEAR_PCM,
+ prtd->bits_per_sample);
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = q6asm_open_read(prtd->audio_client, FORMAT_LINEAR_PCM,
+ prtd->bits_per_sample);
+ }
+
+ if (ret < 0) {
+ pr_err("%s: q6asm_open_write failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ return -ENOMEM;
+ }
+
+ prtd->session_id = q6asm_get_session_id(prtd->audio_client);
+ ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
+ prtd->session_id, substream->stream);
+ if (ret) {
+ pr_err("%s: stream reg failed ret:%d\n", __func__, ret);
+ return ret;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = q6asm_media_format_block_multi_ch_pcm(
+ prtd->audio_client, runtime->rate,
+ runtime->channels, NULL,
+ prtd->bits_per_sample);
+ } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = q6asm_enc_cfg_blk_pcm_format_support(prtd->audio_client,
+ runtime->rate, runtime->channels,
+ prtd->bits_per_sample);
+
+ /* Queue the buffers */
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read(prtd->audio_client);
+
+ }
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+
+ prtd->state = Q6ASM_STREAM_RUNNING;
+
+ return 0;
+}
+
+static int q6asm_dai_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ prtd->state = Q6ASM_STREAM_STOPPED;
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int q6asm_dai_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct snd_soc_dai *cpu_dai = soc_prtd->cpu_dai;
+ struct snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME);
+ struct q6asm_dai_rtd *prtd;
+ struct q6asm_dai_data *pdata;
+ struct device *dev = c->dev;
+ int ret = 0;
+ int stream_id;
+
+ stream_id = cpu_dai->driver->id;
+
+ pdata = snd_soc_component_get_drvdata(c);
+ if (!pdata) {
+ pr_err("Drv data not found ..\n");
+ return -EINVAL;
+ }
+
+ prtd = kzalloc(sizeof(struct q6asm_dai_rtd), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(dev,
+ (q6asm_cb)event_handler, prtd, stream_id,
+ LEGACY_PCM_MODE);
+ if (!prtd->audio_client) {
+ pr_info("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ runtime->hw = q6asm_dai_hardware_playback;
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ runtime->hw = q6asm_dai_hardware_capture;
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_integer failed\n");
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
+ PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes min max ret = %d\n",
+ ret);
+ }
+ }
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (ret < 0) {
+ pr_err("constraint for period bytes step ret = %d\n",
+ ret);
+ }
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes step ret = %d\n",
+ ret);
+ }
+
+ runtime->private_data = prtd;
+
+ snd_soc_set_runtime_hwparams(substream, &q6asm_dai_hardware_playback);
+
+ runtime->dma_bytes = q6asm_dai_hardware_playback.buffer_bytes_max;
+
+
+ if (pdata->sid < 0)
+ prtd->phys = substream->dma_buffer.addr;
+ else
+ prtd->phys = substream->dma_buffer.addr | (pdata->sid << 32);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+ return 0;
+}
+
+static int q6asm_dai_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ if (prtd->audio_client) {
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_unmap_memory_regions(substream->stream,
+ prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ }
+ q6routing_stream_close(soc_prtd->dai_link->id,
+ substream->stream);
+ kfree(prtd);
+ return 0;
+}
+
+static snd_pcm_uframes_t q6asm_dai_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int q6asm_dai_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct snd_soc_component *c = snd_soc_rtdcom_lookup(soc_prtd, DRV_NAME);
+ struct device *dev = c->dev;
+
+ return dma_mmap_coherent(dev, vma,
+ runtime->dma_area, runtime->dma_addr,
+ runtime->dma_bytes);
+}
+
+static int q6asm_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct q6asm_dai_rtd *prtd = runtime->private_data;
+
+ prtd->pcm_size = params_buffer_bytes(params);
+ prtd->periods = params_periods(params);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ prtd->bits_per_sample = 16;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ prtd->bits_per_sample = 24;
+ break;
+ }
+
+ return 0;
+}
+
+static struct snd_pcm_ops q6asm_dai_ops = {
+ .open = q6asm_dai_open,
+ .hw_params = q6asm_dai_hw_params,
+ .close = q6asm_dai_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = q6asm_dai_prepare,
+ .trigger = q6asm_dai_trigger,
+ .pointer = q6asm_dai_pointer,
+ .mmap = q6asm_dai_mmap,
+};
+
+static int q6asm_dai_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm_substream *psubstream, *csubstream;
+ struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME);
+ struct snd_pcm *pcm = rtd->pcm;
+ struct device *dev;
+ int size, ret;
+
+ dev = c->dev;
+ size = q6asm_dai_hardware_playback.buffer_bytes_max;
+ psubstream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (psubstream) {
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+ &psubstream->dma_buffer);
+ if (ret) {
+ dev_err(dev, "Cannot allocate buffer(s)\n");
+ return ret;
+ }
+ }
+
+ csubstream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ if (csubstream) {
+ ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size,
+ &csubstream->dma_buffer);
+ if (ret) {
+ dev_err(dev, "Cannot allocate buffer(s)\n");
+ if (psubstream)
+ snd_dma_free_pages(&psubstream->dma_buffer);
+ return ret;
+ }
+ }
+
+ return ret;
+}
+
+static void q6asm_dai_pcm_free(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(pcm->streams); i++) {
+ substream = pcm->streams[i].substream;
+ if (substream) {
+ snd_dma_free_pages(&substream->dma_buffer);
+ substream->dma_buffer.area = NULL;
+ substream->dma_buffer.addr = 0;
+ }
+ }
+}
+
+static const struct snd_soc_dapm_route afe_pcm_routes[] = {
+ {"MM_DL1", NULL, "MultiMedia1 Playback" },
+ {"MM_DL2", NULL, "MultiMedia2 Playback" },
+ {"MM_DL3", NULL, "MultiMedia3 Playback" },
+ {"MM_DL4", NULL, "MultiMedia4 Playback" },
+ {"MM_DL5", NULL, "MultiMedia5 Playback" },
+ {"MM_DL6", NULL, "MultiMedia6 Playback" },
+ {"MM_DL7", NULL, "MultiMedia7 Playback" },
+ {"MM_DL7", NULL, "MultiMedia8 Playback" },
+ {"MultiMedia1 Capture", NULL, "MM_UL1"},
+ {"MultiMedia2 Capture", NULL, "MM_UL2"},
+ {"MultiMedia3 Capture", NULL, "MM_UL3"},
+ {"MultiMedia4 Capture", NULL, "MM_UL4"},
+ {"MultiMedia5 Capture", NULL, "MM_UL5"},
+ {"MultiMedia6 Capture", NULL, "MM_UL6"},
+ {"MultiMedia7 Capture", NULL, "MM_UL7"},
+ {"MultiMedia8 Capture", NULL, "MM_UL8"},
+
+};
+
+static int fe_dai_probe(struct snd_soc_dai *dai)
+{
+ struct snd_soc_dapm_context *dapm;
+
+ dapm = snd_soc_component_get_dapm(dai->component);
+ snd_soc_dapm_add_routes(dapm, afe_pcm_routes,
+ ARRAY_SIZE(afe_pcm_routes));
+
+ return 0;
+}
+
+
+static const struct snd_soc_component_driver q6asm_fe_dai_component = {
+ .name = DRV_NAME,
+ .ops = &q6asm_dai_ops,
+ .pcm_new = q6asm_dai_pcm_new,
+ .pcm_free = q6asm_dai_pcm_free,
+
+};
+
+static struct snd_soc_dai_driver q6asm_fe_dais[] = {
+ Q6ASM_FEDAI_DRIVER(1),
+ Q6ASM_FEDAI_DRIVER(2),
+ Q6ASM_FEDAI_DRIVER(3),
+ Q6ASM_FEDAI_DRIVER(4),
+ Q6ASM_FEDAI_DRIVER(5),
+ Q6ASM_FEDAI_DRIVER(6),
+ Q6ASM_FEDAI_DRIVER(7),
+ Q6ASM_FEDAI_DRIVER(8),
+};
+
+static int q6asm_dai_bind(struct device *dev, struct device *master, void *data)
+{
+ struct device_node *node = dev->of_node;
+ struct of_phandle_args args;
+ struct q6asm_dai_data *pdata;
+ int rc;
+
+ pdata = kzalloc(sizeof(struct q6asm_dai_data), GFP_KERNEL);
+ if (!pdata)
+ return -ENOMEM;
+
+ rc = of_parse_phandle_with_fixed_args(node, "iommus", 1, 0, &args);
+ if (rc < 0)
+ pdata->sid = -1;
+ else
+ pdata->sid = args.args[0] & SID_MASK_DEFAULT;
+
+ dev_set_drvdata(dev, pdata);
+
+ return snd_soc_register_component(dev, &q6asm_fe_dai_component,
+ q6asm_fe_dais,
+ ARRAY_SIZE(q6asm_fe_dais));
+}
+static void q6asm_dai_unbind(struct device *dev, struct device *master,
+ void *data)
+{
+ struct q6asm_dai_data *pdata = dev_get_drvdata(dev);
+
+ snd_soc_unregister_component(dev);
+
+ kfree(pdata);
+
+}
+
+static const struct component_ops q6asm_dai_comp_ops = {
+ .bind = q6asm_dai_bind,
+ .unbind = q6asm_dai_unbind,
+};
+
+static int q6asm_dai_probe(struct platform_device *pdev)
+{
+ return component_add(&pdev->dev, &q6asm_dai_comp_ops);
+}
+
+static int q6asm_dai_dev_remove(struct platform_device *pdev)
+{
+ component_del(&pdev->dev, &q6asm_dai_comp_ops);
+ return 0;
+}
+
+static struct platform_driver q6asm_dai_platform_driver = {
+ .driver = {
+ .name = "q6asm-dai",
+ },
+ .probe = q6asm_dai_probe,
+ .remove = q6asm_dai_dev_remove,
+};
+module_platform_driver(q6asm_dai_platform_driver);
+
+MODULE_DESCRIPTION("Q6ASM dai driver");
+MODULE_LICENSE("GPL v2");
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* [PATCH v9 14/15] ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (12 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 13/15] ASoC: qdsp6: q6asm: Add q6asm " Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
2018-05-18 12:56 ` [PATCH v9 15/15] ASoC: qcom: apq8096: Add db820c machine driver Srinivas Kandagatla
14 siblings, 0 replies; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
Add devicetree bindings documentation file for Qualcomm apq8096 sound card.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Rob Herring <robh@kernel.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
.../devicetree/bindings/sound/qcom,apq8096.txt | 109 +++++++++++++++++++++
1 file changed, 109 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8096.txt
diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8096.txt b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt
new file mode 100644
index 000000000000..aa54e49fc8a2
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/qcom,apq8096.txt
@@ -0,0 +1,109 @@
+* Qualcomm Technologies APQ8096 ASoC sound card driver
+
+This binding describes the APQ8096 sound card, which uses qdsp for audio.
+
+- compatible:
+ Usage: required
+ Value type: <stringlist>
+ Definition: must be "qcom,apq8096-sndcard"
+
+- qcom,audio-routing:
+ Usage: Optional
+ Value type: <stringlist>
+ Definition: A list of the connections between audio components.
+ Each entry is a pair of strings, the first being the
+ connection's sink, the second being the connection's
+ source. Valid names could be power supplies, MicBias
+ of codec and the jacks on the board:
+ Valid names include:
+
+ Board Connectors:
+ "Headphone Left"
+ "Headphone Right"
+ "Earphone"
+ "Line Out1"
+ "Line Out2"
+ "Line Out3"
+ "Line Out4"
+ "Analog Mic1"
+ "Analog Mic2"
+ "Analog Mic3"
+ "Analog Mic4"
+ "Analog Mic5"
+ "Analog Mic6"
+ "Digital Mic2"
+ "Digital Mic3"
+
+ Audio pins and MicBias on WCD9335 Codec:
+ "MIC_BIAS1
+ "MIC_BIAS2"
+ "MIC_BIAS3"
+ "MIC_BIAS4"
+ "AMIC1"
+ "AMIC2"
+ "AMIC3"
+ "AMIC4"
+ "AMIC5"
+ "AMIC6"
+ "AMIC6"
+ "DMIC1"
+ "DMIC2"
+ "DMIC3"
+= dailinks
+Each subnode of sndcard represents either a dailink, and subnodes of each
+dailinks would be cpu/codec/platform dais.
+
+- link-name:
+ Usage: required
+ Value type: <string>
+ Definition: User friendly name for dai link
+
+= CPU, PLATFORM, CODEC dais subnodes
+- cpu:
+ Usage: required
+ Value type: <subnode>
+ Definition: cpu dai sub-node
+
+- codec:
+ Usage: Optional
+ Value type: <subnode>
+ Definition: codec dai sub-node
+
+- platform:
+ Usage: Optional
+ Value type: <subnode>
+ Definition: platform dai sub-node
+
+- sound-dai:
+ Usage: required
+ Value type: <phandle with arguments>
+ Definition: dai phandle/s and port of CPU/CODEC/PLATFORM node.
+
+Example:
+
+audio {
+ compatible = "qcom,apq8096-sndcard";
+ qcom,model = "DB820c";
+
+ mm1-dai-link {
+ link-name = "MultiMedia1";
+ cpu {
+ sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>;
+ };
+ };
+
+ hdmi-dai-link {
+ link-name = "HDMI Playback";
+ cpu {
+ sound-dai = <&q6afe HDMI_RX>;
+ };
+
+ platform {
+ sound-dai = <&q6adm>;
+ };
+
+ codec {
+ sound-dai = <&hdmi 0>;
+ };
+ };
+};
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread* [PATCH v9 15/15] ASoC: qcom: apq8096: Add db820c machine driver
2018-05-18 12:55 [PATCH v9 00/15] ASoC: qcom: Add support to QDSP based Audio Srinivas Kandagatla
` (13 preceding siblings ...)
2018-05-18 12:56 ` [PATCH v9 14/15] ASoC: qdsp6: dt-bindings: Add apq8096 machine bindings Srinivas Kandagatla
@ 2018-05-18 12:56 ` Srinivas Kandagatla
14 siblings, 0 replies; 25+ messages in thread
From: Srinivas Kandagatla @ 2018-05-18 12:56 UTC (permalink / raw)
To: broonie, linux-arm-msm, alsa-devel, bgoswami
Cc: robh+dt, gregkh, david.brown, mark.rutland, lgirdwood, plai,
tiwai, perex, devicetree, linux-kernel, linux-arm-kernel,
rohkumar, spatakok, Srinivas Kandagatla
This patch adds support to DB820c machine driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
---
sound/soc/qcom/Kconfig | 9 ++
sound/soc/qcom/Makefile | 2 +
sound/soc/qcom/apq8096.c | 255 +++++++++++++++++++++++++++++++++++++++++++++++
3 files changed, 266 insertions(+)
create mode 100644 sound/soc/qcom/apq8096.c
diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig
index 85bb7dd11fd9..87838fa27997 100644
--- a/sound/soc/qcom/Kconfig
+++ b/sound/soc/qcom/Kconfig
@@ -81,3 +81,12 @@ config SND_SOC_QDSP6
This will enable sound soc platform specific
audio drivers. This includes q6asm, q6adm,
q6afe interfaces to DSP using apr.
+
+config SND_SOC_MSM8996
+ tristate "SoC Machine driver for MSM8996 and APQ8096 boards"
+ depends on QCOM_APR
+ select SND_SOC_QDSP6
+ help
+ Support for Qualcomm Technologies LPASS audio block in
+ APQ8096 SoC-based systems.
+ Say Y if you want to use audio device on this SoCs
diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile
index 0276717917c0..206945bb9ba1 100644
--- a/sound/soc/qcom/Makefile
+++ b/sound/soc/qcom/Makefile
@@ -13,9 +13,11 @@ obj-$(CONFIG_SND_SOC_LPASS_APQ8016) += snd-soc-lpass-apq8016.o
# Machine
snd-soc-storm-objs := storm.o
snd-soc-apq8016-sbc-objs := apq8016_sbc.o
+snd-soc-apq8096-objs := apq8096.o
obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o
obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o
+obj-$(CONFIG_SND_SOC_MSM8996) += snd-soc-apq8096.o
#DSP lib
obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
new file mode 100644
index 000000000000..561cd429e6f2
--- /dev/null
+++ b/sound/soc/qcom/apq8096.c
@@ -0,0 +1,255 @@
+// SPDX-License-Identifier: GPL-2.0
+// Copyright (c) 2018, Linaro Limited
+
+#include <linux/soc/qcom/apr.h>
+#include <linux/module.h>
+#include <linux/component.h>
+#include <linux/platform_device.h>
+#include <linux/of_device.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+
+static int apq8096_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ return 0;
+}
+
+static int apq8096_sbc_parse_of(struct snd_soc_card *card)
+{
+ struct device_node *np;
+ struct device_node *codec = NULL;
+ struct device_node *platform = NULL;
+ struct device_node *cpu = NULL;
+ struct device *dev = card->dev;
+ struct snd_soc_dai_link *link;
+ int ret, num_links;
+
+ ret = snd_soc_of_parse_card_name(card, "qcom,model");
+ if (ret) {
+ dev_err(dev, "Error parsing card name: %d\n", ret);
+ return ret;
+ }
+
+ /* DAPM routes */
+ if (of_property_read_bool(dev->of_node, "qcom,audio-routing")) {
+ ret = snd_soc_of_parse_audio_routing(card,
+ "qcom,audio-routing");
+ if (ret)
+ return ret;
+ }
+
+ /* Populate links */
+ num_links = of_get_child_count(dev->of_node);
+
+ /* Allocate the DAI link array */
+ card->dai_link = kcalloc(num_links, sizeof(*link), GFP_KERNEL);
+ if (!card->dai_link)
+ return -ENOMEM;
+
+ card->num_links = num_links;
+ link = card->dai_link;
+
+ for_each_child_of_node(dev->of_node, np) {
+ cpu = of_get_child_by_name(np, "cpu");
+ if (!cpu) {
+ dev_err(dev, "Can't find cpu DT node\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ 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");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ ret = snd_soc_of_get_dai_name(cpu, &link->cpu_dai_name);
+ if (ret) {
+ dev_err(card->dev, "error getting cpu dai name\n");
+ goto err;
+ }
+
+ platform = of_get_child_by_name(np, "platform");
+ codec = of_get_child_by_name(np, "codec");
+ if (codec && platform) {
+ link->platform_of_node = of_parse_phandle(platform,
+ "sound-dai",
+ 0);
+ if (!link->platform_of_node) {
+ dev_err(card->dev, "platform dai not found\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ ret = snd_soc_of_get_dai_link_codecs(dev, codec, link);
+ if (ret < 0) {
+ dev_err(card->dev, "codec dai not found\n");
+ goto err;
+ }
+ link->no_pcm = 1;
+ link->ignore_pmdown_time = 1;
+ link->be_hw_params_fixup = apq8096_be_hw_params_fixup;
+ } else {
+ link->platform_of_node = link->cpu_of_node;
+ link->codec_dai_name = "snd-soc-dummy-dai";
+ link->codec_name = "snd-soc-dummy";
+ link->dynamic = 1;
+ }
+
+ link->ignore_suspend = 1;
+ ret = of_property_read_string(np, "link-name", &link->name);
+ if (ret) {
+ dev_err(card->dev, "error getting codec dai_link name\n");
+ goto err;
+ }
+
+ link->dpcm_playback = 1;
+ link->dpcm_capture = 1;
+ link->stream_name = link->name;
+ link++;
+ }
+
+ return 0;
+err:
+ of_node_put(cpu);
+ of_node_put(codec);
+ of_node_put(platform);
+ kfree(card->dai_link);
+ return ret;
+}
+
+static int apq8096_bind(struct device *dev)
+{
+ struct snd_soc_card *card;
+ int ret;
+
+ card = kzalloc(sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
+ component_bind_all(dev, card);
+ card->dev = dev;
+ ret = apq8096_sbc_parse_of(card);
+ if (ret) {
+ dev_err(dev, "Error parsing OF data\n");
+ goto err;
+ }
+
+ ret = snd_soc_register_card(card);
+ if (ret)
+ goto err;
+
+ return 0;
+
+err:
+ component_unbind_all(dev, card);
+ kfree(card);
+ return ret;
+}
+
+static void apq8096_unbind(struct device *dev)
+{
+ struct snd_soc_card *card = dev_get_drvdata(dev);
+
+ snd_soc_unregister_card(card);
+ component_unbind_all(dev, card);
+ kfree(card->dai_link);
+ kfree(card);
+}
+
+static const struct component_master_ops apq8096_ops = {
+ .bind = apq8096_bind,
+ .unbind = apq8096_unbind,
+};
+
+static int apq8016_compare_of(struct device *dev, void *data)
+{
+ return dev->of_node == data;
+}
+
+static void apq8016_release_of(struct device *dev, void *data)
+{
+ of_node_put(data);
+}
+
+static int add_audio_components(struct device *dev,
+ struct component_match **matchptr)
+{
+ struct device_node *np, *platform, *cpu, *node, *dai_node;
+
+ node = dev->of_node;
+
+ for_each_child_of_node(node, np) {
+ cpu = of_get_child_by_name(np, "cpu");
+ if (cpu) {
+ dai_node = of_parse_phandle(cpu, "sound-dai", 0);
+ of_node_get(dai_node);
+ component_match_add_release(dev, matchptr,
+ apq8016_release_of,
+ apq8016_compare_of,
+ dai_node);
+ }
+
+ platform = of_get_child_by_name(np, "platform");
+ if (platform) {
+ dai_node = of_parse_phandle(platform, "sound-dai", 0);
+ component_match_add_release(dev, matchptr,
+ apq8016_release_of,
+ apq8016_compare_of,
+ dai_node);
+ }
+ }
+
+ return 0;
+}
+
+static int apq8096_platform_probe(struct platform_device *pdev)
+{
+ struct component_match *match = NULL;
+ int ret;
+
+ ret = add_audio_components(&pdev->dev, &match);
+ if (ret)
+ return ret;
+
+ return component_master_add_with_match(&pdev->dev, &apq8096_ops, match);
+}
+
+static int apq8096_platform_remove(struct platform_device *pdev)
+{
+ component_master_del(&pdev->dev, &apq8096_ops);
+
+ return 0;
+}
+
+static const struct of_device_id msm_snd_apq8096_dt_match[] = {
+ {.compatible = "qcom,apq8096-sndcard"},
+ {}
+};
+
+MODULE_DEVICE_TABLE(of, msm_snd_apq8096_dt_match);
+
+static struct platform_driver msm_snd_apq8096_driver = {
+ .probe = apq8096_platform_probe,
+ .remove = apq8096_platform_remove,
+ .driver = {
+ .name = "msm-snd-apq8096",
+ .owner = THIS_MODULE,
+ .of_match_table = msm_snd_apq8096_dt_match,
+ },
+};
+module_platform_driver(msm_snd_apq8096_driver);
+MODULE_AUTHOR("Srinivas Kandagatla <srinivas.kandagatla@linaro.org");
+MODULE_DESCRIPTION("APQ8096 ASoC Machine Driver");
+MODULE_LICENSE("GPL v2");
--
2.16.2
^ permalink raw reply related [flat|nested] 25+ messages in thread