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ASoC: qdsp6: q6asm: Add support to audio stream apis
This patch adds support to open, write and media format commands in the q6asm module. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org> Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org> Signed-off-by: Mark Brown <broonie@kernel.org>
This commit is contained in:
Srinivas Kandagatla
Mark Brown
parent
a2a5d30218
commit
68fd8480bb
@ -11,6 +11,8 @@
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#include <linux/spinlock.h>
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#include <linux/kref.h>
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#include <linux/of.h>
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#include <linux/of_platform.h>
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#include <uapi/sound/asound.h>
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#include <linux/delay.h>
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#include <linux/slab.h>
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#include <linux/mm.h>
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@ -19,10 +21,36 @@
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#include "q6dsp-errno.h"
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#include "q6dsp-common.h"
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#define ASM_STREAM_CMD_CLOSE 0x00010BCD
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#define ASM_STREAM_CMD_FLUSH 0x00010BCE
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#define ASM_SESSION_CMD_PAUSE 0x00010BD3
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#define ASM_DATA_CMD_EOS 0x00010BDB
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#define ASM_NULL_POPP_TOPOLOGY 0x00010C68
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#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
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#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
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#define ASM_STREAM_POSTPROC_TOPO_ID_NONE 0x00010C68
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#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
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#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
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#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
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#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
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#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99
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#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
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#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
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#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
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#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
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#define ASM_DATA_CMD_READ_V2 0x00010DAC
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#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
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#define ASM_STREAM_CMD_OPEN_WRITE_V3 0x00010DB3
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#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
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#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
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#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
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#define ASM_LEGACY_STREAM_SESSION 0
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/* Bit shift for the stream_perf_mode subfield. */
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#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ 29
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#define ASM_END_POINT_DEVICE_MATRIX 0
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#define ASM_DEFAULT_APP_TYPE 0
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#define ASM_SYNC_IO_MODE 0x0001
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#define ASM_ASYNC_IO_MODE 0x0002
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#define ASM_TUN_READ_IO_MODE 0x0004 /* tunnel read write mode */
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@ -46,6 +74,89 @@ struct avs_cmd_shared_mem_unmap_regions {
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u32 mem_map_handle;
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} __packed;
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struct asm_data_cmd_media_fmt_update_v2 {
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u32 fmt_blk_size;
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} __packed;
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struct asm_multi_channel_pcm_fmt_blk_v2 {
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struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
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u16 num_channels;
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u16 bits_per_sample;
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u32 sample_rate;
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u16 is_signed;
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u16 reserved;
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u8 channel_mapping[PCM_MAX_NUM_CHANNEL];
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} __packed;
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struct asm_stream_cmd_set_encdec_param {
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u32 param_id;
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u32 param_size;
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} __packed;
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struct asm_enc_cfg_blk_param_v2 {
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u32 frames_per_buf;
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u32 enc_cfg_blk_size;
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} __packed;
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struct asm_multi_channel_pcm_enc_cfg_v2 {
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struct asm_stream_cmd_set_encdec_param encdec;
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struct asm_enc_cfg_blk_param_v2 encblk;
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uint16_t num_channels;
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uint16_t bits_per_sample;
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uint32_t sample_rate;
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uint16_t is_signed;
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uint16_t reserved;
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uint8_t channel_mapping[8];
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} __packed;
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struct asm_data_cmd_read_v2 {
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u32 buf_addr_lsw;
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u32 buf_addr_msw;
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u32 mem_map_handle;
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u32 buf_size;
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u32 seq_id;
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} __packed;
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struct asm_data_cmd_read_v2_done {
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u32 status;
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u32 buf_addr_lsw;
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u32 buf_addr_msw;
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};
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struct asm_stream_cmd_open_read_v3 {
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u32 mode_flags;
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u32 src_endpointype;
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u32 preprocopo_id;
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u32 enc_cfg_id;
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u16 bits_per_sample;
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u16 reserved;
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} __packed;
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struct asm_data_cmd_write_v2 {
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u32 buf_addr_lsw;
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u32 buf_addr_msw;
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u32 mem_map_handle;
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u32 buf_size;
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u32 seq_id;
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u32 timestamp_lsw;
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u32 timestamp_msw;
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u32 flags;
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} __packed;
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struct asm_stream_cmd_open_write_v3 {
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uint32_t mode_flags;
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uint16_t sink_endpointype;
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uint16_t bits_per_sample;
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uint32_t postprocopo_id;
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uint32_t dec_fmt_id;
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} __packed;
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struct asm_session_cmd_run_v2 {
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u32 flags;
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u32 time_lsw;
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u32 time_msw;
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} __packed;
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struct audio_buffer {
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phys_addr_t phys;
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uint32_t size; /* size of buffer */
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@ -409,6 +520,149 @@ err:
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return ac;
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}
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static int32_t q6asm_stream_callback(struct apr_device *adev,
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struct apr_resp_pkt *data,
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int session_id)
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{
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struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
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struct aprv2_ibasic_rsp_result_t *result;
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struct apr_hdr *hdr = &data->hdr;
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struct audio_port_data *port;
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struct audio_client *ac;
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uint32_t client_event = 0;
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int ret = 0;
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ac = q6asm_get_audio_client(q6asm, session_id);
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if (!ac)/* Audio client might already be freed by now */
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return 0;
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result = data->payload;
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switch (hdr->opcode) {
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case APR_BASIC_RSP_RESULT:
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switch (result->opcode) {
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case ASM_SESSION_CMD_PAUSE:
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client_event = ASM_CLIENT_EVENT_CMD_PAUSE_DONE;
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break;
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case ASM_SESSION_CMD_SUSPEND:
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client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
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break;
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case ASM_DATA_CMD_EOS:
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client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
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break;
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case ASM_STREAM_CMD_FLUSH:
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client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
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break;
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case ASM_SESSION_CMD_RUN_V2:
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client_event = ASM_CLIENT_EVENT_CMD_RUN_DONE;
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break;
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case ASM_STREAM_CMD_CLOSE:
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client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
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break;
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case ASM_STREAM_CMD_FLUSH_READBUFS:
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client_event = ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE;
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break;
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case ASM_STREAM_CMD_OPEN_WRITE_V3:
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case ASM_STREAM_CMD_OPEN_READ_V3:
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case ASM_STREAM_CMD_OPEN_READWRITE_V2:
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case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
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case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
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if (result->status != 0) {
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dev_err(ac->dev,
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"cmd = 0x%x returned error = 0x%x\n",
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result->opcode, result->status);
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ac->result = *result;
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wake_up(&ac->cmd_wait);
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ret = 0;
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goto done;
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}
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break;
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default:
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dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
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result->opcode);
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break;
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}
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ac->result = *result;
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wake_up(&ac->cmd_wait);
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if (ac->cb)
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ac->cb(client_event, hdr->token,
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data->payload, ac->priv);
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ret = 0;
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goto done;
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case ASM_DATA_EVENT_WRITE_DONE_V2:
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client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
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if (ac->io_mode & ASM_SYNC_IO_MODE) {
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phys_addr_t phys;
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unsigned long flags;
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spin_lock_irqsave(&ac->lock, flags);
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port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
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if (!port->buf) {
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spin_unlock_irqrestore(&ac->lock, flags);
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ret = 0;
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goto done;
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}
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phys = port->buf[hdr->token].phys;
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if (lower_32_bits(phys) != result->opcode ||
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upper_32_bits(phys) != result->status) {
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dev_err(ac->dev, "Expected addr %pa\n",
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&port->buf[hdr->token].phys);
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spin_unlock_irqrestore(&ac->lock, flags);
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ret = -EINVAL;
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goto done;
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}
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spin_unlock_irqrestore(&ac->lock, flags);
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}
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break;
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case ASM_DATA_EVENT_READ_DONE_V2:
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client_event = ASM_CLIENT_EVENT_DATA_READ_DONE;
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if (ac->io_mode & ASM_SYNC_IO_MODE) {
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struct asm_data_cmd_read_v2_done *done = data->payload;
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unsigned long flags;
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phys_addr_t phys;
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spin_lock_irqsave(&ac->lock, flags);
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port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
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if (!port->buf) {
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spin_unlock_irqrestore(&ac->lock, flags);
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ret = 0;
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goto done;
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}
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phys = port->buf[hdr->token].phys;
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if (upper_32_bits(phys) != done->buf_addr_msw ||
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lower_32_bits(phys) != done->buf_addr_lsw) {
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dev_err(ac->dev, "Expected addr %pa %08x-%08x\n",
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&port->buf[hdr->token].phys,
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done->buf_addr_lsw,
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done->buf_addr_msw);
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spin_unlock_irqrestore(&ac->lock, flags);
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ret = -EINVAL;
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goto done;
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}
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spin_unlock_irqrestore(&ac->lock, flags);
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}
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break;
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}
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if (ac->cb)
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ac->cb(client_event, hdr->token, data->payload, ac->priv);
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done:
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kref_put(&ac->refcount, q6asm_audio_client_release);
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return ret;
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}
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static int q6asm_srvc_callback(struct apr_device *adev,
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struct apr_resp_pkt *data)
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{
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@ -420,6 +674,11 @@ static int q6asm_srvc_callback(struct apr_device *adev,
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struct q6asm *a;
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uint32_t sid = 0;
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uint32_t dir = 0;
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int session_id;
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session_id = (hdr->dest_port >> 8) & 0xFF;
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if (session_id)
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return q6asm_stream_callback(adev, data, session_id);
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sid = (hdr->token >> 8) & 0x0F;
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ac = q6asm_get_audio_client(q6asm, sid);
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@ -540,6 +799,547 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
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}
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EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
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static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
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{
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struct apr_hdr *hdr = &pkt->hdr;
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int rc;
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mutex_lock(&ac->cmd_lock);
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ac->result.opcode = 0;
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ac->result.status = 0;
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rc = apr_send_pkt(ac->adev, pkt);
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if (rc < 0)
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goto err;
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rc = wait_event_timeout(ac->cmd_wait,
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(ac->result.opcode == hdr->opcode), 5 * HZ);
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if (!rc) {
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dev_err(ac->dev, "CMD timeout\n");
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rc = -ETIMEDOUT;
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goto err;
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}
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if (ac->result.status > 0) {
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dev_err(ac->dev, "DSP returned error[%x]\n",
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ac->result.status);
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rc = -EINVAL;
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} else {
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rc = 0;
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}
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err:
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mutex_unlock(&ac->cmd_lock);
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return rc;
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}
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/**
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* q6asm_open_write() - Open audio client for writing
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*
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* @ac: audio client pointer
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* @format: audio sample format
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* @bits_per_sample: bits per sample
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*
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* Return: Will be an negative value on error or zero on success
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*/
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int q6asm_open_write(struct audio_client *ac, uint32_t format,
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uint16_t bits_per_sample)
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{
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struct asm_stream_cmd_open_write_v3 *open;
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struct apr_pkt *pkt;
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void *p;
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int rc, pkt_size;
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pkt_size = APR_HDR_SIZE + sizeof(*open);
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p = kzalloc(pkt_size, GFP_KERNEL);
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if (!p)
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return -ENOMEM;
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pkt = p;
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open = p + APR_HDR_SIZE;
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q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
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pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
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open->mode_flags = 0x00;
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open->mode_flags |= ASM_LEGACY_STREAM_SESSION;
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/* source endpoint : matrix */
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open->sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
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open->bits_per_sample = bits_per_sample;
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open->postprocopo_id = ASM_NULL_POPP_TOPOLOGY;
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switch (format) {
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case FORMAT_LINEAR_PCM:
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open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
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break;
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default:
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dev_err(ac->dev, "Invalid format 0x%x\n", format);
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rc = -EINVAL;
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goto err;
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}
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rc = q6asm_ac_send_cmd_sync(ac, pkt);
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if (rc < 0)
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goto err;
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ac->io_mode |= ASM_TUN_WRITE_IO_MODE;
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err:
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kfree(pkt);
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return rc;
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}
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EXPORT_SYMBOL_GPL(q6asm_open_write);
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static int __q6asm_run(struct audio_client *ac, uint32_t flags,
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uint32_t msw_ts, uint32_t lsw_ts, bool wait)
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{
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struct asm_session_cmd_run_v2 *run;
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struct apr_pkt *pkt;
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int pkt_size, rc;
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void *p;
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pkt_size = APR_HDR_SIZE + sizeof(*run);
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p = kzalloc(pkt_size, GFP_ATOMIC);
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if (!p)
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return -ENOMEM;
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pkt = p;
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run = p + APR_HDR_SIZE;
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q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
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pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2;
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run->flags = flags;
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run->time_lsw = lsw_ts;
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run->time_msw = msw_ts;
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if (wait) {
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rc = q6asm_ac_send_cmd_sync(ac, pkt);
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} else {
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rc = apr_send_pkt(ac->adev, pkt);
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if (rc == pkt_size)
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rc = 0;
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}
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kfree(pkt);
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return rc;
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}
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/**
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* q6asm_run() - start the audio client
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*
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* @ac: audio client pointer
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* @flags: flags associated with write
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* @msw_ts: timestamp msw
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* @lsw_ts: timestamp lsw
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*
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* Return: Will be an negative value on error or zero on success
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*/
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int q6asm_run(struct audio_client *ac, uint32_t flags,
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uint32_t msw_ts, uint32_t lsw_ts)
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{
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return __q6asm_run(ac, flags, msw_ts, lsw_ts, true);
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}
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EXPORT_SYMBOL_GPL(q6asm_run);
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/**
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* q6asm_run_nowait() - start the audio client withou blocking
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*
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||||
* @ac: audio client pointer
|
||||
* @flags: flags associated with write
|
||||
* @msw_ts: timestamp msw
|
||||
* @lsw_ts: timestamp lsw
|
||||
*
|
||||
* Return: Will be an negative value on error or zero on success
|
||||
*/
|
||||
int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
|
||||
uint32_t msw_ts, uint32_t lsw_ts)
|
||||
{
|
||||
return __q6asm_run(ac, flags, msw_ts, lsw_ts, false);
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(q6asm_run_nowait);
|
||||
|
||||
/**
|
||||
* q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration
|
||||
*
|
||||
* @ac: audio client pointer
|
||||
* @rate: audio sample rate
|
||||
* @channels: number of audio channels.
|
||||
* @channel_map: channel map pointer
|
||||
* @bits_per_sample: bits per sample
|
||||
*
|
||||
* Return: Will be an negative value on error or zero on success
|
||||
*/
|
||||
int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
|
||||
uint32_t rate, uint32_t channels,
|
||||
u8 channel_map[PCM_MAX_NUM_CHANNEL],
|
||||
uint16_t bits_per_sample)
|
||||
{
|
||||
struct asm_multi_channel_pcm_fmt_blk_v2 *fmt;
|
||||
struct apr_pkt *pkt;
|
||||
u8 *channel_mapping;
|
||||
void *p;
|
||||
int rc, pkt_size;
|
||||
|
||||
pkt_size = APR_HDR_SIZE + sizeof(*fmt);
|
||||
p = kzalloc(pkt_size, GFP_KERNEL);
|
||||
if (!p)
|
||||
return -ENOMEM;
|
||||
|
||||
pkt = p;
|
||||
fmt = p + APR_HDR_SIZE;
|
||||
|
||||
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
|
||||
|
||||
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
|
||||
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
|
||||
fmt->num_channels = channels;
|
||||
fmt->bits_per_sample = bits_per_sample;
|
||||
fmt->sample_rate = rate;
|
||||
fmt->is_signed = 1;
|
||||
|
||||
channel_mapping = fmt->channel_mapping;
|
||||
|
||||
if (channel_map) {
|
||||
memcpy(channel_mapping, channel_map, PCM_MAX_NUM_CHANNEL);
|
||||
} else {
|
||||
if (q6dsp_map_channels(channel_mapping, channels)) {
|
||||
dev_err(ac->dev, " map channels failed %d\n", channels);
|
||||
rc = -EINVAL;
|
||||
goto err;
|
||||
}
|
||||
}
|
||||
|
||||
rc = q6asm_ac_send_cmd_sync(ac, pkt);
|
||||
|
||||
err:
|
||||
kfree(pkt);
|
||||
return rc;
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
|
||||
|
||||
/**
|
||||
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
|
||||
*
|
||||
* @ac: audio client pointer
|
||||
* @rate: audio sample rate
|
||||
* @channels: number of audio channels.
|
||||
* @bits_per_sample: bits per sample
|
||||
*
|
||||
* Return: Will be an negative value on error or zero on success
|
||||
*/
|
||||
int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
|
||||
uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
|
||||
{
|
||||
struct asm_multi_channel_pcm_enc_cfg_v2 *enc_cfg;
|
||||
struct apr_pkt *pkt;
|
||||
u8 *channel_mapping;
|
||||
u32 frames_per_buf = 0;
|
||||
int pkt_size, rc;
|
||||
void *p;
|
||||
|
||||
pkt_size = APR_HDR_SIZE + sizeof(*enc_cfg);
|
||||
p = kzalloc(pkt_size, GFP_KERNEL);
|
||||
if (!p)
|
||||
return -ENOMEM;
|
||||
|
||||
pkt = p;
|
||||
enc_cfg = p + APR_HDR_SIZE;
|
||||
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
|
||||
|
||||
pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
|
||||
enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
|
||||
enc_cfg->encdec.param_size = sizeof(*enc_cfg) - sizeof(enc_cfg->encdec);
|
||||
enc_cfg->encblk.frames_per_buf = frames_per_buf;
|
||||
enc_cfg->encblk.enc_cfg_blk_size = enc_cfg->encdec.param_size -
|
||||
sizeof(struct asm_enc_cfg_blk_param_v2);
|
||||
|
||||
enc_cfg->num_channels = channels;
|
||||
enc_cfg->bits_per_sample = bits_per_sample;
|
||||
enc_cfg->sample_rate = rate;
|
||||
enc_cfg->is_signed = 1;
|
||||
channel_mapping = enc_cfg->channel_mapping;
|
||||
|
||||
if (q6dsp_map_channels(channel_mapping, channels)) {
|
||||
rc = -EINVAL;
|
||||
goto err;
|
||||
}
|
||||
|
||||
rc = q6asm_ac_send_cmd_sync(ac, pkt);
|
||||
err:
|
||||
kfree(pkt);
|
||||
return rc;
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
|
||||
|
||||
/**
|
||||
* q6asm_read() - read data of period size from audio client
|
||||
*
|
||||
* @ac: audio client pointer
|
||||
*
|
||||
* Return: Will be an negative value on error or zero on success
|
||||
*/
|
||||
int q6asm_read(struct audio_client *ac)
|
||||
{
|
||||
struct asm_data_cmd_read_v2 *read;
|
||||
struct audio_port_data *port;
|
||||
struct audio_buffer *ab;
|
||||
struct apr_pkt *pkt;
|
||||
unsigned long flags;
|
||||
int pkt_size;
|
||||
int rc = 0;
|
||||
void *p;
|
||||
|
||||
pkt_size = APR_HDR_SIZE + sizeof(*read);
|
||||
p = kzalloc(pkt_size, GFP_ATOMIC);
|
||||
if (!p)
|
||||
return -ENOMEM;
|
||||
|
||||
pkt = p;
|
||||
read = p + APR_HDR_SIZE;
|
||||
|
||||
spin_lock_irqsave(&ac->lock, flags);
|
||||
port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
|
||||
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
|
||||
ab = &port->buf[port->dsp_buf];
|
||||
pkt->hdr.opcode = ASM_DATA_CMD_READ_V2;
|
||||
read->buf_addr_lsw = lower_32_bits(ab->phys);
|
||||
read->buf_addr_msw = upper_32_bits(ab->phys);
|
||||
read->mem_map_handle = port->mem_map_handle;
|
||||
|
||||
read->buf_size = ab->size;
|
||||
read->seq_id = port->dsp_buf;
|
||||
pkt->hdr.token = port->dsp_buf;
|
||||
|
||||
port->dsp_buf++;
|
||||
|
||||
if (port->dsp_buf >= port->num_periods)
|
||||
port->dsp_buf = 0;
|
||||
|
||||
spin_unlock_irqrestore(&ac->lock, flags);
|
||||
rc = apr_send_pkt(ac->adev, pkt);
|
||||
if (rc == pkt_size)
|
||||
rc = 0;
|
||||
else
|
||||
pr_err("read op[0x%x]rc[%d]\n", pkt->hdr.opcode, rc);
|
||||
|
||||
kfree(pkt);
|
||||
return rc;
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(q6asm_read);
|
||||
|
||||
static int __q6asm_open_read(struct audio_client *ac,
|
||||
uint32_t format, uint16_t bits_per_sample)
|
||||
{
|
||||
struct asm_stream_cmd_open_read_v3 *open;
|
||||
struct apr_pkt *pkt;
|
||||
int pkt_size, rc;
|
||||
void *p;
|
||||
|
||||
pkt_size = APR_HDR_SIZE + sizeof(*open);
|
||||
p = kzalloc(pkt_size, GFP_KERNEL);
|
||||
if (!p)
|
||||
return -ENOMEM;
|
||||
|
||||
pkt = p;
|
||||
open = p + APR_HDR_SIZE;
|
||||
|
||||
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
|
||||
pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
|
||||
/* Stream prio : High, provide meta info with encoded frames */
|
||||
open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
|
||||
|
||||
open->preprocopo_id = ASM_STREAM_POSTPROC_TOPO_ID_NONE;
|
||||
open->bits_per_sample = bits_per_sample;
|
||||
open->mode_flags = 0x0;
|
||||
|
||||
open->mode_flags |= ASM_LEGACY_STREAM_SESSION <<
|
||||
ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
|
||||
|
||||
switch (format) {
|
||||
case FORMAT_LINEAR_PCM:
|
||||
open->mode_flags |= 0x00;
|
||||
open->enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
|
||||
break;
|
||||
default:
|
||||
pr_err("Invalid format[%d]\n", format);
|
||||
}
|
||||
|
||||
rc = q6asm_ac_send_cmd_sync(ac, pkt);
|
||||
|
||||
kfree(pkt);
|
||||
return rc;
|
||||
}
|
||||
|
||||
/**
|
||||
* q6asm_open_read() - Open audio client for reading
|
||||
*
|
||||
* @ac: audio client pointer
|
||||
* @format: audio sample format
|
||||
* @bits_per_sample: bits per sample
|
||||
*
|
||||
* Return: Will be an negative value on error or zero on success
|
||||
*/
|
||||
int q6asm_open_read(struct audio_client *ac, uint32_t format,
|
||||
uint16_t bits_per_sample)
|
||||
{
|
||||
return __q6asm_open_read(ac, format, bits_per_sample);
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(q6asm_open_read);
|
||||
|
||||
/**
|
||||
* q6asm_write_async() - non blocking write
|
||||
*
|
||||
* @ac: audio client pointer
|
||||
* @len: lenght in bytes
|
||||
* @msw_ts: timestamp msw
|
||||
* @lsw_ts: timestamp lsw
|
||||
* @wflags: flags associated with write
|
||||
*
|
||||
* Return: Will be an negative value on error or zero on success
|
||||
*/
|
||||
int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
|
||||
uint32_t lsw_ts, uint32_t wflags)
|
||||
{
|
||||
struct asm_data_cmd_write_v2 *write;
|
||||
struct audio_port_data *port;
|
||||
struct audio_buffer *ab;
|
||||
unsigned long flags;
|
||||
struct apr_pkt *pkt;
|
||||
int pkt_size;
|
||||
int rc = 0;
|
||||
void *p;
|
||||
|
||||
pkt_size = APR_HDR_SIZE + sizeof(*write);
|
||||
p = kzalloc(pkt_size, GFP_ATOMIC);
|
||||
if (!p)
|
||||
return -ENOMEM;
|
||||
|
||||
pkt = p;
|
||||
write = p + APR_HDR_SIZE;
|
||||
|
||||
spin_lock_irqsave(&ac->lock, flags);
|
||||
port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
|
||||
q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
|
||||
|
||||
ab = &port->buf[port->dsp_buf];
|
||||
pkt->hdr.token = port->dsp_buf;
|
||||
pkt->hdr.opcode = ASM_DATA_CMD_WRITE_V2;
|
||||
write->buf_addr_lsw = lower_32_bits(ab->phys);
|
||||
write->buf_addr_msw = upper_32_bits(ab->phys);
|
||||
write->buf_size = len;
|
||||
write->seq_id = port->dsp_buf;
|
||||
write->timestamp_lsw = lsw_ts;
|
||||
write->timestamp_msw = msw_ts;
|
||||
write->mem_map_handle =
|
||||
ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle;
|
||||
|
||||
if (wflags == NO_TIMESTAMP)
|
||||
write->flags = (wflags & 0x800000FF);
|
||||
else
|
||||
write->flags = (0x80000000 | wflags);
|
||||
|
||||
port->dsp_buf++;
|
||||
|
||||
if (port->dsp_buf >= port->num_periods)
|
||||
port->dsp_buf = 0;
|
||||
|
||||
spin_unlock_irqrestore(&ac->lock, flags);
|
||||
rc = apr_send_pkt(ac->adev, pkt);
|
||||
if (rc == pkt_size)
|
||||
rc = 0;
|
||||
|
||||
kfree(pkt);
|
||||
return rc;
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(q6asm_write_async);
|
||||
|
||||
static void q6asm_reset_buf_state(struct audio_client *ac)
|
||||
{
|
||||
struct audio_port_data *port = NULL;
|
||||
unsigned long flags;
|
||||
|
||||
spin_lock_irqsave(&ac->lock, flags);
|
||||
port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
|
||||
port->dsp_buf = 0;
|
||||
port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
|
||||
port->dsp_buf = 0;
|
||||
spin_unlock_irqrestore(&ac->lock, flags);
|
||||
}
|
||||
|
||||
static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
|
||||
{
|
||||
int stream_id = ac->stream_id;
|
||||
struct apr_pkt pkt;
|
||||
int rc;
|
||||
|
||||
q6asm_add_hdr(ac, &pkt.hdr, APR_HDR_SIZE, true, stream_id);
|
||||
|
||||
switch (cmd) {
|
||||
case CMD_PAUSE:
|
||||
pkt.hdr.opcode = ASM_SESSION_CMD_PAUSE;
|
||||
break;
|
||||
case CMD_SUSPEND:
|
||||
pkt.hdr.opcode = ASM_SESSION_CMD_SUSPEND;
|
||||
break;
|
||||
case CMD_FLUSH:
|
||||
pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH;
|
||||
break;
|
||||
case CMD_OUT_FLUSH:
|
||||
pkt.hdr.opcode = ASM_STREAM_CMD_FLUSH_READBUFS;
|
||||
break;
|
||||
case CMD_EOS:
|
||||
pkt.hdr.opcode = ASM_DATA_CMD_EOS;
|
||||
break;
|
||||
case CMD_CLOSE:
|
||||
pkt.hdr.opcode = ASM_STREAM_CMD_CLOSE;
|
||||
break;
|
||||
default:
|
||||
return -EINVAL;
|
||||
}
|
||||
|
||||
if (wait)
|
||||
rc = q6asm_ac_send_cmd_sync(ac, &pkt);
|
||||
else
|
||||
return apr_send_pkt(ac->adev, &pkt);
|
||||
|
||||
if (rc < 0)
|
||||
return rc;
|
||||
|
||||
if (cmd == CMD_FLUSH)
|
||||
q6asm_reset_buf_state(ac);
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/**
|
||||
* q6asm_cmd() - run cmd on audio client
|
||||
*
|
||||
* @ac: audio client pointer
|
||||
* @cmd: command to run on audio client.
|
||||
*
|
||||
* Return: Will be an negative value on error or zero on success
|
||||
*/
|
||||
int q6asm_cmd(struct audio_client *ac, int cmd)
|
||||
{
|
||||
return __q6asm_cmd(ac, cmd, true);
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(q6asm_cmd);
|
||||
|
||||
/**
|
||||
* q6asm_cmd_nowait() - non blocking, run cmd on audio client
|
||||
*
|
||||
* @ac: audio client pointer
|
||||
* @cmd: command to run on audio client.
|
||||
*
|
||||
* Return: Will be an negative value on error or zero on success
|
||||
*/
|
||||
int q6asm_cmd_nowait(struct audio_client *ac, int cmd)
|
||||
{
|
||||
return __q6asm_cmd(ac, cmd, false);
|
||||
}
|
||||
EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
|
||||
|
||||
static int q6asm_probe(struct apr_device *adev)
|
||||
{
|
||||
|
||||
@ -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,
|
||||
|
||||
Loading…
Reference in New Issue
Block a user