This patch add dt bindings for Qualcomm APR (Asynchronous Packet Router) bus driver. This bus is used for communicating with DSP which provides audio and various other services to cpu. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Rob Herring <robh@kernel.org> Reviewed-by: Bjorn Andersson <bjorn.andersson@linaro.org> Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org> --- .../devicetree/bindings/soc/qcom/qcom,apr.txt | 84 ++++++++++++++++++++++ include/dt-bindings/soc/qcom,apr.h | 28 ++++++++ 2 files changed, 112 insertions(+) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 include/dt-bindings/soc/qcom,apr.h diff --git a/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt b/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt new file mode 100644 index 000000000000..bcc612cc7423 --- /dev/null +++ b/Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt @@ -0,0 +1,84 @@ +Qualcomm APR (Asynchronous Packet Router) binding + +This binding describes the Qualcomm APR. APR is a IPC protocol for +communication between Application processor and QDSP. APR is mainly +used for audio/voice services on the QDSP. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,apr-v<VERSION-NUMBER>", example "qcom,apr-v2" + +- reg + Usage: required + Value type: <u32> + Definition: Destination processor ID. + Possible values are : + 1 - APR simulator + 2 - PC + 3 - MODEM + 4 - ADSP + 5 - APPS + 6 - MODEM2 + 7 - APPS2 + += APR SERVICES +Each subnode of the APR node represents service tied to this apr. The name +of the nodes are not important. The properties of these nodes are defined +by the individual bindings for the specific service +- All APR services MUST contain the following property: + +- reg + Usage: required + Value type: <u32> + Definition: APR Service ID + Possible values are : + 3 - DSP Core Service + 4 - Audio Front End Service. + 5 - Voice Stream Manager Service. + 6 - Voice processing manager. + 7 - Audio Stream Manager Service. + 8 - Audio Device Manager Service. + 9 - Multimode voice manager. + 10 - Core voice stream. + 11 - Core voice processor. + 12 - Ultrasound stream manager. + 13 - Listen stream manager. + += EXAMPLE +The following example represents a QDSP based sound card on a MSM8996 device +which uses apr as communication between Apps and QDSP. + + apr@4 { + compatible = "qcom,apr-v2"; + reg = <APR_DOMAIN_ADSP>; + + q6core@3 { + compatible = "qcom,q6core"; + reg = <APR_SVC_ADSP_CORE>; + }; + + q6afe@4 { + compatible = "qcom,q6afe"; + reg = <APR_SVC_AFE>; + + dais { + #sound-dai-cells = <1>; + hdmi@1 { + reg = <1>; + }; + }; + }; + + q6asm@7 { + compatible = "qcom,q6asm"; + reg = <APR_SVC_ASM>; + ... + }; + + q6adm@8 { + compatible = "qcom,q6adm"; + reg = <APR_SVC_ADM>; + ... + }; + }; diff --git a/include/dt-bindings/soc/qcom,apr.h b/include/dt-bindings/soc/qcom,apr.h new file mode 100644 index 000000000000..006362400c0f --- /dev/null +++ b/include/dt-bindings/soc/qcom,apr.h @@ -0,0 +1,28 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_BINDINGS_QCOM_APR_H +#define __DT_BINDINGS_QCOM_APR_H + +/* Domain IDs */ +#define APR_DOMAIN_SIM 0x1 +#define APR_DOMAIN_PC 0x2 +#define APR_DOMAIN_MODEM 0x3 +#define APR_DOMAIN_ADSP 0x4 +#define APR_DOMAIN_APPS 0x5 +#define APR_DOMAIN_MAX 0x6 + +/* ADSP service IDs */ +#define APR_SVC_ADSP_CORE 0x3 +#define APR_SVC_AFE 0x4 +#define APR_SVC_VSM 0x5 +#define APR_SVC_VPM 0x6 +#define APR_SVC_ASM 0x7 +#define APR_SVC_ADM 0x8 +#define APR_SVC_ADSP_MVM 0x09 +#define APR_SVC_ADSP_CVS 0x0A +#define APR_SVC_ADSP_CVP 0x0B +#define APR_SVC_USM 0x0C +#define APR_SVC_LSM 0x0D +#define APR_SVC_VIDC 0x16 +#define APR_SVC_MAX 0x17 + +#endif /* __DT_BINDINGS_QCOM_APR_H */ -- 2.16.2

This patch add DT bindings for Q6CORE DSP module. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org> Reviewed-by: Rob Herring <robh@kernel.org> Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org> --- .../devicetree/bindings/sound/qcom,q6core.txt | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6core.txt diff --git a/Documentation/devicetree/bindings/sound/qcom,q6core.txt b/Documentation/devicetree/bindings/sound/qcom,q6core.txt new file mode 100644 index 000000000000..7f36ff8bec18 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6core.txt @@ -0,0 +1,21 @@ +Qualcomm ADSP Core service binding + +Q6CORE is one of the APR audio service on Q6DSP. +Please refer to qcom,apr.txt for details of the common apr service bindings +used by the apr service device. + +- but must contain the following property: + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6core-v<MAJOR-NUMBER>.<MINOR-NUMBER>". + Or "qcom,q6core" where the version number can be queried + from DSP. + example "qcom,q6core-v2.0" + += EXAMPLE +q6core@3 { + compatible = "qcom,q6core"; + reg = <APR_SVC_ADSP_CORE>; +}; -- 2.16.2

This patch add DT bindings for AFE (Audio Frontend) DSP module. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org> Reviewed-by: Rob Herring <robh@kernel.org> Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org> --- .../devicetree/bindings/sound/qcom,q6afe.txt | 104 +++++++++++++++++++++ include/dt-bindings/sound/qcom,q6afe.h | 31 ++++++ 2 files changed, 135 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt create mode 100644 include/dt-bindings/sound/qcom,q6afe.h diff --git a/Documentation/devicetree/bindings/sound/qcom,q6afe.txt b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt new file mode 100644 index 000000000000..14335a08b963 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6afe.txt @@ -0,0 +1,104 @@ +Qualcomm Audio Front End (Q6AFE) binding + +AFE is one of the APR audio service on Q6DSP +Please refer to qcom,apr.txt for details of the common apr service bindings +used by all apr services. Must contain the following properties. + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6afe-v<MAJOR-NUMBER>.<MINOR-NUMBER>" + Or "qcom,q6afe" where the version number can be queried + from DSP. + example "qcom,q6afe" + += AFE DAIs (Digial Audio Interface) +"dais" subnode of the AFE node. It represents afe dais, each afe dai is a +subnode of "dais" representing board specific dai setup. +"dais" node should have following properties followed by dai children. + +- #sound-dai-cells + Usage: required + Value type: <u32> + Definition: Must be 1 + +- #address-cells + Usage: required + Value type: <u32> + Definition: Must be 1 + +- #size-cells + Usage: required + Value type: <u32> + Definition: Must be 0 + +== AFE DAI is subnode of "dais" and represent a dai, it includes board specific +configuration of each dai. Must contain the following properties. + +- reg + Usage: required + Value type: <u32> + Definition: Must be dai id + +- qcom,sd-lines + Usage: required for mi2s interface + Value type: <prop-encoded-array> + Definition: Must be list of serial data lines used by this dai. + should be one or more of the 1-4 sd lines. + += EXAMPLE + +q6afe@4 { + compatible = "qcom,q6afe"; + reg = <APR_SVC_AFE>; + + dais { + #sound-dai-cells = <1>; + #address-cells = <1>; + #size-cells = <0>; + + hdmi@1 { + reg = <1>; + }; + + prim-mi2s-rx@16 { + reg = <16>; + qcom,sd-lines = <1 3>; + }; + + prim-mi2s-tx@17 { + reg = <17>; + qcom,sd-lines = <2>; + }; + + sec-mi2s-rx@18 { + reg = <18>; + qcom,sd-lines = <1 4>; + }; + + sec-mi2s-tx@19 { + reg = <19>; + qcom,sd-lines = <2>; + }; + + tert-mi2s-rx@20 { + reg = <20>; + qcom,sd-lines = <2 4>; + }; + + tert-mi2s-tx@21 { + reg = <21>; + qcom,sd-lines = <1>; + }; + + quat-mi2s-rx@22 { + reg = <22>; + qcom,sd-lines = <1>; + }; + + quat-mi2s-tx@23 { + reg = <23>; + qcom,sd-lines = <2>; + }; + }; +}; diff --git a/include/dt-bindings/sound/qcom,q6afe.h b/include/dt-bindings/sound/qcom,q6afe.h new file mode 100644 index 000000000000..e162045f5dc9 --- /dev/null +++ b/include/dt-bindings/sound/qcom,q6afe.h @@ -0,0 +1,31 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_BINDINGS_Q6_AFE_H__ +#define __DT_BINDINGS_Q6_AFE_H__ + +/* Audio Front End (AFE) virtual ports IDs */ +#define HDMI_RX 1 +#define SLIMBUS_0_RX 2 +#define SLIMBUS_0_TX 3 +#define SLIMBUS_1_RX 4 +#define SLIMBUS_1_TX 5 +#define SLIMBUS_2_RX 6 +#define SLIMBUS_2_TX 7 +#define SLIMBUS_3_RX 8 +#define SLIMBUS_3_TX 9 +#define SLIMBUS_4_RX 10 +#define SLIMBUS_4_TX 11 +#define SLIMBUS_5_RX 12 +#define SLIMBUS_5_TX 13 +#define SLIMBUS_6_RX 14 +#define SLIMBUS_6_TX 15 +#define PRIMARY_MI2S_RX 16 +#define PRIMARY_MI2S_TX 17 +#define SECONDARY_MI2S_RX 18 +#define SECONDARY_MI2S_TX 19 +#define TERTIARY_MI2S_RX 20 +#define TERTIARY_MI2S_TX 21 +#define QUATERNARY_MI2S_RX 22 +#define QUATERNARY_MI2S_TX 23 + +#endif /* __DT_BINDINGS_Q6_AFE_H__ */ + -- 2.16.2

This patch add DT bindings for ASM (Audio Stream Manager) DSP module. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org> Reviewed-by: Rob Herring <robh@kernel.org> Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org> --- .../devicetree/bindings/sound/qcom,q6asm.txt | 33 ++++++++++++++++++++++ include/dt-bindings/sound/qcom,q6asm.h | 22 +++++++++++++++ 2 files changed, 55 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 include/dt-bindings/sound/qcom,q6asm.h diff --git a/Documentation/devicetree/bindings/sound/qcom,q6asm.txt b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt new file mode 100644 index 000000000000..2178eb91146f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,q6asm.txt @@ -0,0 +1,33 @@ +Qualcomm Audio Stream Manager (Q6ASM) binding + +Q6ASM is one of the APR audio service on Q6DSP. +Please refer to qcom,apr.txt for details of the common apr service bindings +used by the apr service device. + +- but must contain the following property: + +- compatible: + Usage: required + Value type: <stringlist> + Definition: must be "qcom,q6asm-v<MAJOR-NUMBER>.<MINOR-NUMBER>". + Or "qcom,q6asm" where the version number can be queried + from DSP. + example "qcom,q6asm-v2.0" + += ASM DAIs (Digial Audio Interface) +"dais" subnode of the ASM node represents dai specific configuration + +- #sound-dai-cells + Usage: required + Value type: <u32> + Definition: Must be 1 + += EXAMPLE + +q6asm@7 { + compatible = "qcom,q6asm"; + reg = <APR_SVC_ASM>; + q6asmdai: dais { + #sound-dai-cells = <1>; + }; +}; diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h new file mode 100644 index 000000000000..1eb77d87c2e8 --- /dev/null +++ b/include/dt-bindings/sound/qcom,q6asm.h @@ -0,0 +1,22 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_BINDINGS_Q6_ASM_H__ +#define __DT_BINDINGS_Q6_ASM_H__ + +#define MSM_FRONTEND_DAI_MULTIMEDIA1 0 +#define MSM_FRONTEND_DAI_MULTIMEDIA2 1 +#define MSM_FRONTEND_DAI_MULTIMEDIA3 2 +#define MSM_FRONTEND_DAI_MULTIMEDIA4 3 +#define MSM_FRONTEND_DAI_MULTIMEDIA5 4 +#define MSM_FRONTEND_DAI_MULTIMEDIA6 5 +#define MSM_FRONTEND_DAI_MULTIMEDIA7 6 +#define MSM_FRONTEND_DAI_MULTIMEDIA8 7 +#define MSM_FRONTEND_DAI_MULTIMEDIA9 8 +#define MSM_FRONTEND_DAI_MULTIMEDIA10 9 +#define MSM_FRONTEND_DAI_MULTIMEDIA11 10 +#define MSM_FRONTEND_DAI_MULTIMEDIA12 11 +#define MSM_FRONTEND_DAI_MULTIMEDIA13 12 +#define MSM_FRONTEND_DAI_MULTIMEDIA14 13 +#define MSM_FRONTEND_DAI_MULTIMEDIA15 14 +#define MSM_FRONTEND_DAI_MULTIMEDIA16 15 + +#endif /* __DT_BINDINGS_Q6_ASM_H__ */ -- 2.16.2

This patch adds support to core apr service, which is used to query status of other static and dynamic services on the dsp. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org> --- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6core.c | 380 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6core.h | 15 ++ 4 files changed, 400 insertions(+) create mode 100644 sound/soc/qcom/qdsp6/q6core.c create mode 100644 sound/soc/qcom/qdsp6/q6core.h diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index b44a9fcd7ed3..37ee0d958145 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -44,10 +44,14 @@ config SND_SOC_APQ8016_SBC config SND_SOC_QDSP6_COMMON tristate +config SND_SOC_QDSP6_CORE + 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 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 accebdb49306..03b8e89c9731 100644 --- a/sound/soc/qcom/qdsp6/Makefile +++ b/sound/soc/qcom/qdsp6/Makefile @@ -1 +1,2 @@ obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += q6dsp-common.o +obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c new file mode 100644 index 000000000000..06f03a5fe9bd --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -0,0 +1,380 @@ +// 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/module.h> +#include <linux/sched.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/jiffies.h> +#include <linux/wait.h> +#include <linux/soc/qcom/apr.h> +#include "q6core.h" +#include "q6dsp-errno.h" + +#define ADSP_STATE_READY_TIMEOUT_MS 3000 +#define Q6_READY_TIMEOUT_MS 100 +#define AVCS_CMD_ADSP_EVENT_GET_STATE 0x0001290C +#define AVCS_CMDRSP_ADSP_EVENT_GET_STATE 0x0001290D +#define AVCS_GET_VERSIONS 0x00012905 +#define AVCS_GET_VERSIONS_RSP 0x00012906 +#define AVCS_CMD_GET_FWK_VERSION 0x001292c +#define AVCS_CMDRSP_GET_FWK_VERSION 0x001292d + +struct avcs_svc_info { + uint32_t service_id; + uint32_t version; +} __packed; + +struct avcs_cmdrsp_get_version { + uint32_t build_id; + uint32_t num_services; + struct avcs_svc_info svc_api_info[]; +} __packed; + +/* for ADSP2.8 and above */ +struct avcs_svc_api_info { + uint32_t service_id; + uint32_t api_version; + uint32_t api_branch_version; +} __packed; + +struct avcs_cmdrsp_get_fwk_version { + uint32_t build_major_version; + uint32_t build_minor_version; + uint32_t build_branch_version; + uint32_t build_subbranch_version; + uint32_t num_services; + struct avcs_svc_api_info svc_api_info[]; +} __packed; + +struct q6core { + struct apr_device *adev; + wait_queue_head_t wait; + uint32_t avcs_state; + struct mutex lock; + bool resp_received; + uint32_t num_services; + struct avcs_cmdrsp_get_fwk_version *fwk_version; + struct avcs_cmdrsp_get_version *svc_version; + bool fwk_version_supported; + bool get_state_supported; + bool get_version_supported; + bool is_version_requested; +}; + +static struct q6core *g_core; + +static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) +{ + struct q6core *core = dev_get_drvdata(&adev->dev); + struct aprv2_ibasic_rsp_result_t *result; + struct apr_hdr *hdr = &data->hdr; + + result = data->payload; + switch (hdr->opcode) { + case APR_BASIC_RSP_RESULT:{ + result = data->payload; + switch (result->opcode) { + case AVCS_GET_VERSIONS: + if (result->status == ADSP_EUNSUPPORTED) + core->get_version_supported = false; + core->resp_received = true; + break; + case AVCS_CMD_GET_FWK_VERSION: + if (result->status == ADSP_EUNSUPPORTED) + core->fwk_version_supported = false; + core->resp_received = true; + break; + case AVCS_CMD_ADSP_EVENT_GET_STATE: + if (result->status == ADSP_EUNSUPPORTED) + core->get_state_supported = false; + core->resp_received = true; + break; + } + break; + } + case AVCS_CMDRSP_GET_FWK_VERSION: { + struct avcs_cmdrsp_get_fwk_version *fwk; + int bytes; + + fwk = data->payload; + bytes = sizeof(*fwk) + fwk->num_services * + sizeof(fwk->svc_api_info[0]); + + core->fwk_version = kzalloc(bytes, GFP_ATOMIC); + if (!core->fwk_version) + return -ENOMEM; + + memcpy(core->fwk_version, data->payload, bytes); + + core->fwk_version_supported = true; + core->resp_received = true; + + break; + } + case AVCS_GET_VERSIONS_RSP: { + struct avcs_cmdrsp_get_version *v; + int len; + + v = data->payload; + + len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]); + + core->svc_version = kzalloc(len, GFP_ATOMIC); + if (!core->svc_version) + return -ENOMEM; + + memcpy(core->svc_version, data->payload, len); + + core->get_version_supported = true; + core->resp_received = true; + + break; + } + case AVCS_CMDRSP_ADSP_EVENT_GET_STATE: + core->get_state_supported = true; + core->avcs_state = result->opcode; + + core->resp_received = true; + break; + default: + dev_err(&adev->dev, "Message id from adsp core svc: 0x%x\n", + hdr->opcode); + break; + } + + if (core->resp_received) + wake_up(&core->wait); + + return 0; +} + +static int q6core_get_fwk_versions(struct q6core *core) +{ + struct apr_device *adev = core->adev; + struct apr_pkt pkt; + int rc; + + 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 = APR_HDR_SIZE; + pkt.hdr.opcode = AVCS_CMD_GET_FWK_VERSION; + + rc = apr_send_pkt(adev, &pkt); + if (rc < 0) + return rc; + + rc = wait_event_timeout(core->wait, (core->resp_received), + msecs_to_jiffies(Q6_READY_TIMEOUT_MS)); + if (rc > 0 && core->resp_received) { + core->resp_received = false; + + if (!core->fwk_version_supported) + return -ENOTSUPP; + else + return 0; + } + + + return rc; +} + +static int q6core_get_svc_versions(struct q6core *core) +{ + struct apr_device *adev = core->adev; + struct apr_pkt pkt; + int rc; + + 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 = APR_HDR_SIZE; + pkt.hdr.opcode = AVCS_GET_VERSIONS; + + rc = apr_send_pkt(adev, &pkt); + if (rc < 0) + return rc; + + rc = wait_event_timeout(core->wait, (core->resp_received), + msecs_to_jiffies(Q6_READY_TIMEOUT_MS)); + if (rc > 0 && core->resp_received) { + core->resp_received = false; + return 0; + } + + return rc; +} + +static bool __q6core_is_adsp_ready(struct q6core *core) +{ + struct apr_device *adev = core->adev; + struct apr_pkt pkt; + int rc; + + core->get_state_supported = false; + + 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 = APR_HDR_SIZE; + pkt.hdr.opcode = AVCS_CMD_ADSP_EVENT_GET_STATE; + + rc = apr_send_pkt(adev, &pkt); + if (rc < 0) + return false; + + rc = wait_event_timeout(core->wait, (core->resp_received), + msecs_to_jiffies(Q6_READY_TIMEOUT_MS)); + if (rc > 0 && core->resp_received) { + core->resp_received = false; + + if (core->avcs_state) + return true; + } + + /* assume that the adsp is up if we not support this command */ + if (!core->get_state_supported) + return true; + + return false; +} + +/** + * q6core_get_svc_api_info() - Get version number of a service. + * + * @svc_id: service id of the service. + * @ainfo: Valid struct pointer to fill svc api information. + * + * Return: zero on success and error code on failure or unsupported + */ +int q6core_get_svc_api_info(int svc_id, struct q6core_svc_api_info *ainfo) +{ + int i; + int ret = -ENOTSUPP; + + if (!g_core || !ainfo) + return 0; + + mutex_lock(&g_core->lock); + if (!g_core->is_version_requested) { + if (q6core_get_fwk_versions(g_core) == -ENOTSUPP) + q6core_get_svc_versions(g_core); + g_core->is_version_requested = true; + } + + if (g_core->fwk_version_supported) { + for (i = 0; i < g_core->fwk_version->num_services; i++) { + struct avcs_svc_api_info *info; + + info = &g_core->fwk_version->svc_api_info[i]; + if (svc_id != info->service_id) + continue; + + ainfo->api_version = info->api_version; + ainfo->api_branch_version = info->api_branch_version; + ret = 0; + break; + } + } else if (g_core->get_version_supported) { + for (i = 0; i < g_core->svc_version->num_services; i++) { + struct avcs_svc_info *info; + + info = &g_core->svc_version->svc_api_info[i]; + if (svc_id != info->service_id) + continue; + + ainfo->api_version = info->version; + ainfo->api_branch_version = 0; + ret = 0; + break; + } + } + + mutex_unlock(&g_core->lock); + + return ret; +} +EXPORT_SYMBOL_GPL(q6core_get_svc_api_info); + +/** + * q6core_is_adsp_ready() - Get status of adsp + * + * Return: Will be an true if adsp is ready and false if not. + */ +bool q6core_is_adsp_ready(void) +{ + unsigned long timeout; + bool ret = false; + + if (!g_core) + return false; + + mutex_lock(&g_core->lock); + timeout = jiffies + msecs_to_jiffies(ADSP_STATE_READY_TIMEOUT_MS); + for (;;) { + if (__q6core_is_adsp_ready(g_core)) { + ret = true; + break; + } + + if (!time_after(timeout, jiffies)) { + ret = false; + break; + } + } + + mutex_unlock(&g_core->lock); + return ret; +} +EXPORT_SYMBOL_GPL(q6core_is_adsp_ready); + +static int q6core_probe(struct apr_device *adev) +{ + g_core = kzalloc(sizeof(*g_core), GFP_KERNEL); + if (!g_core) + return -ENOMEM; + + dev_set_drvdata(&adev->dev, g_core); + + mutex_init(&g_core->lock); + g_core->adev = adev; + init_waitqueue_head(&g_core->wait); + return 0; +} + +static int q6core_exit(struct apr_device *adev) +{ + struct q6core *core = dev_get_drvdata(&adev->dev); + + if (core->fwk_version_supported) + kfree(core->fwk_version); + if (core->get_version_supported) + kfree(core->svc_version); + + g_core = NULL; + kfree(core); + + return 0; +} + +static const struct of_device_id q6core_device_id[] = { + { .compatible = "qcom,q6core" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6core_device_id); + +static struct apr_driver qcom_q6core_driver = { + .probe = q6core_probe, + .remove = q6core_exit, + .callback = q6core_callback, + .driver = { + .name = "qcom-q6core", + .of_match_table = of_match_ptr(q6core_device_id), + }, +}; + +module_apr_driver(qcom_q6core_driver); +MODULE_DESCRIPTION("q6 core"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/qdsp6/q6core.h b/sound/soc/qcom/qdsp6/q6core.h new file mode 100644 index 000000000000..4105b1d730be --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6core.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0 */ + +#ifndef __Q6CORE_H__ +#define __Q6CORE_H__ + +struct q6core_svc_api_info { + uint32_t service_id; + uint32_t api_version; + uint32_t api_branch_version; +}; + +bool q6core_is_adsp_ready(void); +int q6core_get_svc_api_info(int svc_id, struct q6core_svc_api_info *ainfo); + +#endif /* __Q6CORE_H__ */ -- 2.16.2

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 ec35ed1ffede..1de5c941b11e 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -27,9 +27,45 @@ #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 { @@ -110,6 +191,20 @@ struct afe_port_map { static struct afe_port_map port_maps[AFE_PORT_MAX] = { [HDMI_RX] = { AFE_PORT_ID_MULTICHAN_HDMI_RX, HDMI_RX, 1, 1}, + [SLIMBUS_0_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX, + SLIMBUS_0_RX, 1, 1}, + [SLIMBUS_1_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX, + SLIMBUS_1_RX, 1, 1}, + [SLIMBUS_2_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX, + SLIMBUS_2_RX, 1, 1}, + [SLIMBUS_3_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX, + SLIMBUS_3_RX, 1, 1}, + [SLIMBUS_4_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX, + SLIMBUS_4_RX, 1, 1}, + [SLIMBUS_5_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_5_RX, + SLIMBUS_5_RX, 1, 1}, + [SLIMBUS_6_RX] = { AFE_PORT_ID_SLIMBUS_MULTI_CHAN_6_RX, + SLIMBUS_6_RX, 1, 1}, }; static struct q6afe_port *afe_find_port(struct q6afe *afe, int token) @@ -323,6 +418,31 @@ 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. * @@ -426,6 +546,15 @@ 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

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 | 164 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.h | 131 ++++++++++++++++++++++++++++++++++ 2 files changed, 295 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6afe.c b/sound/soc/qcom/qdsp6/q6afe.c index 6808b7bae5d5..11525a25db13 100644 --- a/sound/soc/qcom/qdsp6/q6afe.c +++ b/sound/soc/qcom/qdsp6/q6afe.c @@ -33,6 +33,9 @@ #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 @@ -66,6 +69,11 @@ #define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E #define AFE_API_VERSION_SLIMBUS_CONFIG 0x1 +/* Clock set API version */ +#define AFE_API_VERSION_CLOCK_SET 1 +#define Q6AFE_LPASS_CLK_CONFIG_API_VERSION 0x1 +#define AFE_MODULE_CLOCK_SET 0x0001028F +#define AFE_PARAM_ID_CLOCK_SET 0x00010290 /* SLIMbus Rx port on channel 0. */ #define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000 @@ -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; @@ -387,6 +429,54 @@ 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 psize) { @@ -436,6 +526,80 @@ 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, + 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, + 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

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 | 215 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 15 +++ 4 files changed, 235 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..09f62e90ab00 --- /dev/null +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -0,0 +1,215 @@ +// 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/of.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/mm.h> +#include "q6asm.h" +#include "q6core.h" +#include "q6dsp-errno.h" +#include "q6dsp-common.h" + +#define ASM_SYNC_IO_MODE 0x0001 +#define ASM_ASYNC_IO_MODE 0x0002 +#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */ +#define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */ + +struct audio_client { + int session; + q6asm_cb cb; + void *priv; + uint32_t io_mode; + struct apr_device *adev; + struct mutex lock; + spinlock_t buf_lock; + wait_queue_head_t cmd_wait; + struct aprv2_ibasic_rsp_result_t result; + int perf_mode; + int stream_id; + struct device *dev; +}; + +struct q6asm { + struct apr_device *adev; + struct device *dev; + struct q6core_svc_api_info ainfo; + wait_queue_head_t mem_wait; + struct platform_device *pcmdev; + struct audio_client *session[MAX_SESSIONS + 1]; + struct platform_device *pdev_dais; +}; + +/** + * q6asm_audio_client_free() - Freee allocated audio client + * + * @ac: audio client to free + */ +void q6asm_audio_client_free(struct audio_client *ac) +{ + struct q6asm *a = dev_get_drvdata(ac->dev->parent); + + a->session[ac->session] = NULL; + kfree(ac); +} +EXPORT_SYMBOL_GPL(q6asm_audio_client_free); + +static struct audio_client *q6asm_get_audio_client(struct q6asm *a, + int session_id) +{ + if ((session_id <= 0) || (session_id > MAX_SESSIONS)) { + dev_err(a->dev, "invalid session: %d\n", session_id); + return NULL; + } + + if (!a->session[session_id]) { + dev_err(a->dev, "session not active: %d\n", session_id); + return NULL; + } + + return a->session[session_id]; +} + +static int q6asm_srvc_callback(struct apr_device *adev, + struct apr_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); + + 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; + + if (stream_id + 1 > MAX_SESSIONS) + return ERR_PTR(-EINVAL); + + ac = kzalloc(sizeof(*ac), GFP_KERNEL); + if (!ac) + return ERR_PTR(-ENOMEM); + + a->session[stream_id + 1] = ac; + ac->session = stream_id + 1; + ac->cb = cb; + ac->dev = dev; + ac->priv = priv; + ac->io_mode = ASM_SYNC_IO_MODE; + ac->perf_mode = perf_mode; + /* DSP expects stream id from 1 */ + ac->stream_id = 1; + ac->adev = a->adev; + + init_waitqueue_head(&ac->cmd_wait); + mutex_init(&ac->lock); + spin_lock_init(&ac->buf_lock); + + return ac; +} +EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc); + + +static int q6asm_probe(struct apr_device *adev) +{ + struct device *dev = &adev->dev; + struct device_node *dais_np; + struct q6asm *q6asm; + + q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL); + if (!q6asm) + return -ENOMEM; + + q6core_get_svc_api_info(adev->svc_id, &q6asm->ainfo); + + q6asm->dev = dev; + q6asm->adev = adev; + init_waitqueue_head(&q6asm->mem_wait); + dev_set_drvdata(dev, q6asm); + + dais_np = of_get_child_by_name(dev->of_node, "dais"); + if (dais_np) { + q6asm->pdev_dais = of_platform_device_create(dais_np, + "q6asm-dai", dev); + of_node_put(dais_np); + } + + return 0; +} + +static int q6asm_remove(struct apr_device *adev) +{ + struct q6asm *q6asm = dev_get_drvdata(&adev->dev); + + if (q6asm->pdev_dais) + of_platform_device_destroy(&q6asm->pdev_dais->dev, NULL); + + return 0; +} +static const struct of_device_id q6asm_device_id[] = { + { .compatible = "qcom,q6asm" }, + {}, +}; +MODULE_DEVICE_TABLE(of, q6asm_device_id); + +static struct apr_driver qcom_q6asm_driver = { + .probe = q6asm_probe, + .remove = q6asm_remove, + .callback = q6asm_srvc_callback, + .driver = { + .name = "qcom-q6asm", + .of_match_table = of_match_ptr(q6asm_device_id), + }, +}; + +module_apr_driver(qcom_q6asm_driver); +MODULE_DESCRIPTION("Q6 Audio Stream Manager driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h new file mode 100644 index 000000000000..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

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 | 834 ++++++++++++++++++++++++++++++++++++++++++- sound/soc/qcom/qdsp6/q6asm.h | 49 +++ 2 files changed, 882 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 044ad57d15a0..4d70b2267ce2 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -10,6 +10,8 @@ #include <linux/of_platform.h> #include <linux/spinlock.h> #include <linux/of.h> +#include <linux/of_platform.h> +#include <uapi/sound/asound.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/mm.h> @@ -18,10 +20,36 @@ #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 */ @@ -45,6 +73,89 @@ 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 used; @@ -85,6 +196,22 @@ struct q6asm { struct platform_device *pdev_dais; }; +static bool q6asm_is_valid_audio_client(struct audio_client *ac) +{ + struct q6asm *a = dev_get_drvdata(ac->dev->parent); + int n; + + if (!ac) + return false; + + for (n = 1; n <= MAX_SESSIONS; n++) { + if (a->session[n] == ac) + return true; + } + + return false; +} + 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) @@ -388,6 +515,150 @@ static struct audio_client *q6asm_get_audio_client(struct q6asm *a, return a->session[session_id]; } +static int32_t q6asm_stream_callback(struct apr_device *adev, + struct apr_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 token; + uint32_t client_event = 0; + + ac = q6asm_get_audio_client(q6asm, session_id); + if (!ac)/* Audio client might already be freed by now */ + return 0; + + if (!q6asm_is_valid_audio_client(ac)) + return -EINVAL; + + result = data->payload; + + switch (hdr->opcode) { + case APR_BASIC_RSP_RESULT: + token = hdr->token; + switch (result->opcode) { + case ASM_SESSION_CMD_PAUSE: + client_event = ASM_CLIENT_EVENT_CMD_PAUSE_DONE; + break; + case ASM_SESSION_CMD_SUSPEND: + client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE; + break; + case ASM_DATA_CMD_EOS: + client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE; + break; + 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); + return 0; + } + break; + default: + dev_err(ac->dev, "command[0x%x] not expecting rsp\n", + result->opcode); + break; + } + + ac->result = *result; + wake_up(&ac->cmd_wait); + + if (ac->cb) + ac->cb(client_event, hdr->token, + data->payload, ac->priv); + + return 0; + + case ASM_DATA_EVENT_WRITE_DONE_V2: + port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK]; + + client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE; + + if (ac->io_mode & ASM_SYNC_IO_MODE) { + phys_addr_t phys; + unsigned long flags; + + spin_lock_irqsave(&ac->buf_lock, flags); + if (!port->buf) { + spin_unlock_irqrestore(&ac->buf_lock, flags); + return 0; + } + + phys = port->buf[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->buf_lock, flags); + return -EINVAL; + } + token = hdr->token; + port->buf[token].used = 1; + spin_unlock_irqrestore(&ac->buf_lock, flags); + } + break; + case ASM_DATA_EVENT_READ_DONE_V2: + port = &ac->port[SNDRV_PCM_STREAM_CAPTURE]; + client_event = ASM_CLIENT_EVENT_DATA_READ_DONE; + + if (ac->io_mode & ASM_SYNC_IO_MODE) { + struct asm_data_cmd_read_v2_done *done = data->payload; + unsigned long flags; + phys_addr_t phys; + + spin_lock_irqsave(&ac->buf_lock, flags); + if (!port->buf) { + spin_unlock_irqrestore(&ac->buf_lock, flags); + return 0; + } + + phys = port->buf[hdr->token].phys; + token = hdr->token; + port->buf[token].used = 0; + + if (upper_32_bits(phys) != done->buf_addr_msw || + lower_32_bits(phys) != done->buf_addr_lsw) { + dev_err(ac->dev, "Expected addr %pa %08x-%08x\n", + &port->buf[hdr->token].phys, + done->buf_addr_lsw, + done->buf_addr_msw); + spin_unlock_irqrestore(&ac->buf_lock, flags); + return -EINVAL; + } + spin_unlock_irqrestore(&ac->buf_lock, flags); + } + + break; + } + + if (ac->cb) + ac->cb(client_event, hdr->token, data->payload, ac->priv); + + return 0; +} + static int q6asm_srvc_callback(struct apr_device *adev, struct apr_resp_pkt *data) { @@ -399,6 +670,11 @@ 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); @@ -508,6 +784,562 @@ 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->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->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; + int pkt_size; + int rc = 0; + void *p; + + if (!(ac->io_mode & ASM_SYNC_IO_MODE)) + return 0; + + pkt_size = APR_HDR_SIZE + sizeof(*read); + p = kzalloc(pkt_size, GFP_ATOMIC); + if (!p) + return -ENOMEM; + + pkt = p; + read = p + APR_HDR_SIZE; + + 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; + + 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 + * @flags: flags associated with write + * + * Return: Will be an negative value on error or zero on success + */ +int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts, + uint32_t lsw_ts, uint32_t flags) +{ + struct asm_data_cmd_write_v2 *write; + struct audio_port_data *port; + struct audio_buffer *ab; + 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; + + if (!(ac->io_mode & ASM_SYNC_IO_MODE)) + return 0; + + 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 (flags == NO_TIMESTAMP) + write->flags = (flags & 0x800000FF); + else + write->flags = (0x80000000 | flags); + + port->dsp_buf++; + + if (port->dsp_buf >= port->num_periods) + port->dsp_buf = 0; + + rc = apr_send_pkt(ac->adev, 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; + int loopcnt = 0; + int cnt = 0; + int used; + + if (!(ac->io_mode & ASM_SYNC_IO_MODE)) + return; + + used = (ac->io_mode & ASM_TUN_WRITE_IO_MODE ? 1 : 0); + spin_lock_irqsave(&ac->buf_lock, flags); + for (loopcnt = 0; loopcnt <= SNDRV_PCM_STREAM_CAPTURE; loopcnt++) { + port = &ac->port[loopcnt]; + cnt = port->num_periods - 1; + port->dsp_buf = 0; + while (cnt >= 0) { + if (!port->buf) + continue; + port->buf[cnt].used = used; + cnt--; + } + } + spin_unlock_irqrestore(&ac->buf_lock, flags); +} + +static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait) +{ + int stream_id = ac->stream_id; + struct apr_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

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 f7ee9a1a29b2..97204f2d1894 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -233,6 +233,180 @@ 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), @@ -256,6 +430,28 @@ 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[] = { @@ -268,6 +464,71 @@ 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

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

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

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

Add myself as co-maintainer of qcom audio drivers Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 20032feaa9e7..3df5f6226827 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -11522,6 +11522,7 @@ F: drivers/crypto/qat/ QCOM AUDIO (ASoC) DRIVERS M: Patrick Lai <plai@codeaurora.org> M: Banajit Goswami <bgoswami@codeaurora.org> +M: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/qcom/ -- 2.16.2