mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
synced 2024-12-22 18:52:54 +07:00
7076bf4da0
Qualcomm DSPs expect ALAC and APE configs to be send for decoders, so add the API to program the respective config to the DSP. Signed-off-by: Vinod Koul <vkoul@kernel.org> Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Reviewed-by: Takashi Iwai <tiwai@suse.de> Link: https://lore.kernel.org/r/20200316055221.1944464-8-vkoul@kernel.org Signed-off-by: Mark Brown <broonie@kernel.org>
1687 lines
40 KiB
C
1687 lines
40 KiB
C
// SPDX-License-Identifier: GPL-2.0
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// Copyright (c) 2011-2017, The Linux Foundation. All rights reserved.
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// Copyright (c) 2018, Linaro Limited
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#include <linux/mutex.h>
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#include <linux/wait.h>
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#include <linux/module.h>
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#include <linux/soc/qcom/apr.h>
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#include <linux/device.h>
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#include <linux/of_platform.h>
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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 <uapi/sound/asound.h>
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#include <uapi/sound/compress_params.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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#include "q6asm.h"
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#include "q6core.h"
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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_MEDIA_FMT_MP3 0x00010BE9
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#define ASM_MEDIA_FMT_FLAC 0x00010C16
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#define ASM_MEDIA_FMT_WMA_V9 0x00010DA8
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#define ASM_MEDIA_FMT_WMA_V10 0x00010DA7
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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_MEDIA_FMT_ALAC 0x00012f31
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#define ASM_MEDIA_FMT_APE 0x00012f32
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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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#define ASM_TUN_WRITE_IO_MODE 0x0008 /* tunnel read write mode */
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#define ASM_SHIFT_GAPLESS_MODE_FLAG 31
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#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3
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struct avs_cmd_shared_mem_map_regions {
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u16 mem_pool_id;
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u16 num_regions;
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u32 property_flag;
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} __packed;
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struct avs_shared_map_region_payload {
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u32 shm_addr_lsw;
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u32 shm_addr_msw;
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u32 mem_size_bytes;
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} __packed;
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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_flac_fmt_blk_v2 {
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struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
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u16 is_stream_info_present;
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u16 num_channels;
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u16 min_blk_size;
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u16 max_blk_size;
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u16 md5_sum[8];
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u32 sample_rate;
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u32 min_frame_size;
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u32 max_frame_size;
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u16 sample_size;
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u16 reserved;
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} __packed;
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struct asm_wmastdv9_fmt_blk_v2 {
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struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
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u16 fmtag;
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u16 num_channels;
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u32 sample_rate;
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u32 bytes_per_sec;
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u16 blk_align;
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u16 bits_per_sample;
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u32 channel_mask;
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u16 enc_options;
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u16 reserved;
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} __packed;
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struct asm_wmaprov10_fmt_blk_v2 {
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struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
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u16 fmtag;
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u16 num_channels;
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u32 sample_rate;
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u32 bytes_per_sec;
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u16 blk_align;
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u16 bits_per_sample;
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u32 channel_mask;
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u16 enc_options;
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u16 advanced_enc_options1;
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u32 advanced_enc_options2;
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} __packed;
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struct asm_alac_fmt_blk_v2 {
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struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
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u32 frame_length;
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u8 compatible_version;
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u8 bit_depth;
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u8 pb;
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u8 mb;
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u8 kb;
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u8 num_channels;
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u16 max_run;
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u32 max_frame_bytes;
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u32 avg_bit_rate;
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u32 sample_rate;
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u32 channel_layout_tag;
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} __packed;
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struct asm_ape_fmt_blk_v2 {
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struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
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u16 compatible_version;
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u16 compression_level;
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u32 format_flags;
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u32 blocks_per_frame;
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u32 final_frame_blocks;
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u32 total_frames;
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u16 bits_per_sample;
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u16 num_channels;
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u32 sample_rate;
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u32 seek_table_present;
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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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};
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struct audio_port_data {
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struct audio_buffer *buf;
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uint32_t num_periods;
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uint32_t dsp_buf;
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uint32_t mem_map_handle;
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};
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struct q6asm {
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struct apr_device *adev;
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struct device *dev;
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struct q6core_svc_api_info ainfo;
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wait_queue_head_t mem_wait;
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spinlock_t slock;
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struct audio_client *session[MAX_SESSIONS + 1];
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};
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struct audio_client {
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int session;
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q6asm_cb cb;
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void *priv;
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uint32_t io_mode;
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struct apr_device *adev;
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struct mutex cmd_lock;
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spinlock_t lock;
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struct kref refcount;
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/* idx:1 out port, 0: in port */
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struct audio_port_data port[2];
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wait_queue_head_t cmd_wait;
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struct aprv2_ibasic_rsp_result_t result;
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int perf_mode;
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int stream_id;
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struct q6asm *q6asm;
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struct device *dev;
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};
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static inline void q6asm_add_hdr(struct audio_client *ac, struct apr_hdr *hdr,
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uint32_t pkt_size, bool cmd_flg,
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uint32_t stream_id)
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{
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hdr->hdr_field = APR_SEQ_CMD_HDR_FIELD;
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hdr->src_port = ((ac->session << 8) & 0xFF00) | (stream_id);
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hdr->dest_port = ((ac->session << 8) & 0xFF00) | (stream_id);
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hdr->pkt_size = pkt_size;
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if (cmd_flg)
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hdr->token = ac->session;
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}
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static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,
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struct apr_pkt *pkt, uint32_t rsp_opcode)
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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(a->adev, pkt);
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if (rc < 0)
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goto err;
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if (rsp_opcode)
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rc = wait_event_timeout(a->mem_wait,
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(ac->result.opcode == hdr->opcode) ||
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(ac->result.opcode == rsp_opcode),
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5 * HZ);
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else
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rc = wait_event_timeout(a->mem_wait,
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(ac->result.opcode == hdr->opcode),
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5 * HZ);
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if (!rc) {
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dev_err(a->dev, "CMD timeout\n");
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rc = -ETIMEDOUT;
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} else if (ac->result.status > 0) {
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dev_err(a->dev, "DSP returned error[%x]\n",
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ac->result.status);
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rc = -EINVAL;
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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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static int __q6asm_memory_unmap(struct audio_client *ac,
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phys_addr_t buf_add, int dir)
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{
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struct avs_cmd_shared_mem_unmap_regions *mem_unmap;
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struct q6asm *a = dev_get_drvdata(ac->dev->parent);
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struct apr_pkt *pkt;
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int rc, pkt_size;
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void *p;
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if (ac->port[dir].mem_map_handle == 0) {
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dev_err(ac->dev, "invalid mem handle\n");
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return -EINVAL;
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}
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pkt_size = APR_HDR_SIZE + sizeof(*mem_unmap);
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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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mem_unmap = p + APR_HDR_SIZE;
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pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
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pkt->hdr.src_port = 0;
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pkt->hdr.dest_port = 0;
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pkt->hdr.pkt_size = pkt_size;
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pkt->hdr.token = ((ac->session << 8) | dir);
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pkt->hdr.opcode = ASM_CMD_SHARED_MEM_UNMAP_REGIONS;
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mem_unmap->mem_map_handle = ac->port[dir].mem_map_handle;
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rc = q6asm_apr_send_session_pkt(a, ac, pkt, 0);
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if (rc < 0) {
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kfree(pkt);
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return rc;
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}
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ac->port[dir].mem_map_handle = 0;
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kfree(pkt);
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return 0;
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}
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static void q6asm_audio_client_free_buf(struct audio_client *ac,
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struct audio_port_data *port)
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{
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unsigned long flags;
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spin_lock_irqsave(&ac->lock, flags);
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port->num_periods = 0;
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kfree(port->buf);
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port->buf = NULL;
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spin_unlock_irqrestore(&ac->lock, flags);
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}
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/**
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* q6asm_unmap_memory_regions() - unmap memory regions in the dsp.
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*
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* @dir: direction of audio stream
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* @ac: audio client instanace
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*
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* Return: Will be an negative value on failure or zero on success
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*/
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int q6asm_unmap_memory_regions(unsigned int dir, struct audio_client *ac)
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{
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struct audio_port_data *port;
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int cnt = 0;
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int rc = 0;
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port = &ac->port[dir];
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if (!port->buf) {
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rc = -EINVAL;
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goto err;
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}
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cnt = port->num_periods - 1;
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if (cnt >= 0) {
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rc = __q6asm_memory_unmap(ac, port->buf[dir].phys, dir);
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if (rc < 0) {
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dev_err(ac->dev, "%s: Memory_unmap_regions failed %d\n",
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__func__, rc);
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goto err;
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}
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}
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q6asm_audio_client_free_buf(ac, port);
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err:
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return rc;
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}
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EXPORT_SYMBOL_GPL(q6asm_unmap_memory_regions);
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static int __q6asm_memory_map_regions(struct audio_client *ac, int dir,
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size_t period_sz, unsigned int periods,
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bool is_contiguous)
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{
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struct avs_cmd_shared_mem_map_regions *cmd = NULL;
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struct avs_shared_map_region_payload *mregions = NULL;
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struct q6asm *a = dev_get_drvdata(ac->dev->parent);
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struct audio_port_data *port = NULL;
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struct audio_buffer *ab = NULL;
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struct apr_pkt *pkt;
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void *p;
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unsigned long flags;
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uint32_t num_regions, buf_sz;
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int rc, i, pkt_size;
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if (is_contiguous) {
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num_regions = 1;
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buf_sz = period_sz * periods;
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} else {
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buf_sz = period_sz;
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num_regions = periods;
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}
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/* DSP expects size should be aligned to 4K */
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buf_sz = ALIGN(buf_sz, 4096);
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pkt_size = APR_HDR_SIZE + sizeof(*cmd) +
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(sizeof(*mregions) * num_regions);
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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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cmd = p + APR_HDR_SIZE;
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mregions = p + APR_HDR_SIZE + sizeof(*cmd);
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pkt->hdr.hdr_field = APR_SEQ_CMD_HDR_FIELD;
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pkt->hdr.src_port = 0;
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pkt->hdr.dest_port = 0;
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pkt->hdr.pkt_size = pkt_size;
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pkt->hdr.token = ((ac->session << 8) | dir);
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pkt->hdr.opcode = ASM_CMD_SHARED_MEM_MAP_REGIONS;
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cmd->mem_pool_id = ADSP_MEMORY_MAP_SHMEM8_4K_POOL;
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cmd->num_regions = num_regions;
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cmd->property_flag = 0x00;
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spin_lock_irqsave(&ac->lock, flags);
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port = &ac->port[dir];
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for (i = 0; i < num_regions; i++) {
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ab = &port->buf[i];
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mregions->shm_addr_lsw = lower_32_bits(ab->phys);
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mregions->shm_addr_msw = upper_32_bits(ab->phys);
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mregions->mem_size_bytes = buf_sz;
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++mregions;
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}
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
|
|
rc = q6asm_apr_send_session_pkt(a, ac, pkt,
|
|
ASM_CMDRSP_SHARED_MEM_MAP_REGIONS);
|
|
|
|
kfree(pkt);
|
|
|
|
return rc;
|
|
}
|
|
|
|
/**
|
|
* q6asm_map_memory_regions() - map memory regions in the dsp.
|
|
*
|
|
* @dir: direction of audio stream
|
|
* @ac: audio client instanace
|
|
* @phys: physcial address that needs mapping.
|
|
* @period_sz: audio period size
|
|
* @periods: number of periods
|
|
*
|
|
* Return: Will be an negative value on failure or zero on success
|
|
*/
|
|
int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac,
|
|
phys_addr_t phys,
|
|
size_t period_sz, unsigned int periods)
|
|
{
|
|
struct audio_buffer *buf;
|
|
unsigned long flags;
|
|
int cnt;
|
|
int rc;
|
|
|
|
spin_lock_irqsave(&ac->lock, flags);
|
|
if (ac->port[dir].buf) {
|
|
dev_err(ac->dev, "Buffer already allocated\n");
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
return 0;
|
|
}
|
|
|
|
buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC);
|
|
if (!buf) {
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
return -ENOMEM;
|
|
}
|
|
|
|
|
|
ac->port[dir].buf = buf;
|
|
|
|
buf[0].phys = phys;
|
|
buf[0].size = period_sz;
|
|
|
|
for (cnt = 1; cnt < periods; cnt++) {
|
|
if (period_sz > 0) {
|
|
buf[cnt].phys = buf[0].phys + (cnt * period_sz);
|
|
buf[cnt].size = period_sz;
|
|
}
|
|
}
|
|
ac->port[dir].num_periods = periods;
|
|
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
|
|
rc = __q6asm_memory_map_regions(ac, dir, period_sz, periods, 1);
|
|
if (rc < 0) {
|
|
dev_err(ac->dev, "Memory_map_regions failed\n");
|
|
q6asm_audio_client_free_buf(ac, &ac->port[dir]);
|
|
}
|
|
|
|
return rc;
|
|
}
|
|
EXPORT_SYMBOL_GPL(q6asm_map_memory_regions);
|
|
|
|
static void q6asm_audio_client_release(struct kref *ref)
|
|
{
|
|
struct audio_client *ac;
|
|
struct q6asm *a;
|
|
unsigned long flags;
|
|
|
|
ac = container_of(ref, struct audio_client, refcount);
|
|
a = ac->q6asm;
|
|
|
|
spin_lock_irqsave(&a->slock, flags);
|
|
a->session[ac->session] = NULL;
|
|
spin_unlock_irqrestore(&a->slock, flags);
|
|
|
|
kfree(ac);
|
|
}
|
|
|
|
/**
|
|
* q6asm_audio_client_free() - Freee allocated audio client
|
|
*
|
|
* @ac: audio client to free
|
|
*/
|
|
void q6asm_audio_client_free(struct audio_client *ac)
|
|
{
|
|
kref_put(&ac->refcount, q6asm_audio_client_release);
|
|
}
|
|
EXPORT_SYMBOL_GPL(q6asm_audio_client_free);
|
|
|
|
static struct audio_client *q6asm_get_audio_client(struct q6asm *a,
|
|
int session_id)
|
|
{
|
|
struct audio_client *ac = NULL;
|
|
unsigned long flags;
|
|
|
|
spin_lock_irqsave(&a->slock, flags);
|
|
if ((session_id <= 0) || (session_id > MAX_SESSIONS)) {
|
|
dev_err(a->dev, "invalid session: %d\n", session_id);
|
|
goto err;
|
|
}
|
|
|
|
/* check for valid session */
|
|
if (!a->session[session_id])
|
|
goto err;
|
|
else if (a->session[session_id]->session != session_id)
|
|
goto err;
|
|
|
|
ac = a->session[session_id];
|
|
kref_get(&ac->refcount);
|
|
err:
|
|
spin_unlock_irqrestore(&a->slock, flags);
|
|
return ac;
|
|
}
|
|
|
|
static int32_t q6asm_stream_callback(struct apr_device *adev,
|
|
struct apr_resp_pkt *data,
|
|
int session_id)
|
|
{
|
|
struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
|
|
struct aprv2_ibasic_rsp_result_t *result;
|
|
struct apr_hdr *hdr = &data->hdr;
|
|
struct audio_port_data *port;
|
|
struct audio_client *ac;
|
|
uint32_t client_event = 0;
|
|
int ret = 0;
|
|
|
|
ac = q6asm_get_audio_client(q6asm, session_id);
|
|
if (!ac)/* Audio client might already be freed by now */
|
|
return 0;
|
|
|
|
result = data->payload;
|
|
|
|
switch (hdr->opcode) {
|
|
case APR_BASIC_RSP_RESULT:
|
|
switch (result->opcode) {
|
|
case ASM_SESSION_CMD_PAUSE:
|
|
client_event = ASM_CLIENT_EVENT_CMD_PAUSE_DONE;
|
|
break;
|
|
case ASM_SESSION_CMD_SUSPEND:
|
|
client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
|
|
break;
|
|
case ASM_DATA_CMD_EOS:
|
|
client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
|
|
break;
|
|
case ASM_STREAM_CMD_FLUSH:
|
|
client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
|
|
break;
|
|
case ASM_SESSION_CMD_RUN_V2:
|
|
client_event = ASM_CLIENT_EVENT_CMD_RUN_DONE;
|
|
break;
|
|
case ASM_STREAM_CMD_CLOSE:
|
|
client_event = ASM_CLIENT_EVENT_CMD_CLOSE_DONE;
|
|
break;
|
|
case ASM_STREAM_CMD_FLUSH_READBUFS:
|
|
client_event = ASM_CLIENT_EVENT_CMD_OUT_FLUSH_DONE;
|
|
break;
|
|
case ASM_STREAM_CMD_OPEN_WRITE_V3:
|
|
case ASM_STREAM_CMD_OPEN_READ_V3:
|
|
case ASM_STREAM_CMD_OPEN_READWRITE_V2:
|
|
case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
|
|
case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
|
|
if (result->status != 0) {
|
|
dev_err(ac->dev,
|
|
"cmd = 0x%x returned error = 0x%x\n",
|
|
result->opcode, result->status);
|
|
ac->result = *result;
|
|
wake_up(&ac->cmd_wait);
|
|
ret = 0;
|
|
goto done;
|
|
}
|
|
break;
|
|
default:
|
|
dev_err(ac->dev, "command[0x%x] not expecting rsp\n",
|
|
result->opcode);
|
|
break;
|
|
}
|
|
|
|
ac->result = *result;
|
|
wake_up(&ac->cmd_wait);
|
|
|
|
if (ac->cb)
|
|
ac->cb(client_event, hdr->token,
|
|
data->payload, ac->priv);
|
|
|
|
ret = 0;
|
|
goto done;
|
|
|
|
case ASM_DATA_EVENT_WRITE_DONE_V2:
|
|
client_event = ASM_CLIENT_EVENT_DATA_WRITE_DONE;
|
|
if (ac->io_mode & ASM_SYNC_IO_MODE) {
|
|
phys_addr_t phys;
|
|
unsigned long flags;
|
|
|
|
spin_lock_irqsave(&ac->lock, flags);
|
|
|
|
port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
|
|
|
|
if (!port->buf) {
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
ret = 0;
|
|
goto done;
|
|
}
|
|
|
|
phys = port->buf[hdr->token].phys;
|
|
|
|
if (lower_32_bits(phys) != result->opcode ||
|
|
upper_32_bits(phys) != result->status) {
|
|
dev_err(ac->dev, "Expected addr %pa\n",
|
|
&port->buf[hdr->token].phys);
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
ret = -EINVAL;
|
|
goto done;
|
|
}
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
}
|
|
break;
|
|
case ASM_DATA_EVENT_READ_DONE_V2:
|
|
client_event = ASM_CLIENT_EVENT_DATA_READ_DONE;
|
|
if (ac->io_mode & ASM_SYNC_IO_MODE) {
|
|
struct asm_data_cmd_read_v2_done *done = data->payload;
|
|
unsigned long flags;
|
|
phys_addr_t phys;
|
|
|
|
spin_lock_irqsave(&ac->lock, flags);
|
|
port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
|
|
if (!port->buf) {
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
ret = 0;
|
|
goto done;
|
|
}
|
|
|
|
phys = port->buf[hdr->token].phys;
|
|
|
|
if (upper_32_bits(phys) != done->buf_addr_msw ||
|
|
lower_32_bits(phys) != done->buf_addr_lsw) {
|
|
dev_err(ac->dev, "Expected addr %pa %08x-%08x\n",
|
|
&port->buf[hdr->token].phys,
|
|
done->buf_addr_lsw,
|
|
done->buf_addr_msw);
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
ret = -EINVAL;
|
|
goto done;
|
|
}
|
|
spin_unlock_irqrestore(&ac->lock, flags);
|
|
}
|
|
|
|
break;
|
|
}
|
|
|
|
if (ac->cb)
|
|
ac->cb(client_event, hdr->token, data->payload, ac->priv);
|
|
|
|
done:
|
|
kref_put(&ac->refcount, q6asm_audio_client_release);
|
|
return ret;
|
|
}
|
|
|
|
static int q6asm_srvc_callback(struct apr_device *adev,
|
|
struct apr_resp_pkt *data)
|
|
{
|
|
struct q6asm *q6asm = dev_get_drvdata(&adev->dev);
|
|
struct aprv2_ibasic_rsp_result_t *result;
|
|
struct audio_port_data *port;
|
|
struct audio_client *ac = NULL;
|
|
struct apr_hdr *hdr = &data->hdr;
|
|
struct q6asm *a;
|
|
uint32_t sid = 0;
|
|
uint32_t dir = 0;
|
|
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);
|
|
if (!ac) {
|
|
dev_err(&adev->dev, "Audio Client not active\n");
|
|
return 0;
|
|
}
|
|
|
|
a = dev_get_drvdata(ac->dev->parent);
|
|
dir = (hdr->token & 0x0F);
|
|
port = &ac->port[dir];
|
|
result = data->payload;
|
|
|
|
switch (hdr->opcode) {
|
|
case APR_BASIC_RSP_RESULT:
|
|
switch (result->opcode) {
|
|
case ASM_CMD_SHARED_MEM_MAP_REGIONS:
|
|
case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
|
|
ac->result = *result;
|
|
wake_up(&a->mem_wait);
|
|
break;
|
|
default:
|
|
dev_err(&adev->dev, "command[0x%x] not expecting rsp\n",
|
|
result->opcode);
|
|
break;
|
|
}
|
|
goto done;
|
|
case ASM_CMDRSP_SHARED_MEM_MAP_REGIONS:
|
|
ac->result.status = 0;
|
|
ac->result.opcode = hdr->opcode;
|
|
port->mem_map_handle = result->opcode;
|
|
wake_up(&a->mem_wait);
|
|
break;
|
|
case ASM_CMD_SHARED_MEM_UNMAP_REGIONS:
|
|
ac->result.opcode = hdr->opcode;
|
|
ac->result.status = 0;
|
|
port->mem_map_handle = 0;
|
|
wake_up(&a->mem_wait);
|
|
break;
|
|
default:
|
|
dev_dbg(&adev->dev, "command[0x%x]success [0x%x]\n",
|
|
result->opcode, result->status);
|
|
break;
|
|
}
|
|
|
|
if (ac->cb)
|
|
ac->cb(hdr->opcode, hdr->token, data->payload, ac->priv);
|
|
|
|
done:
|
|
kref_put(&ac->refcount, q6asm_audio_client_release);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* q6asm_get_session_id() - get session id for audio client
|
|
*
|
|
* @c: audio client pointer
|
|
*
|
|
* Return: Will be an session id of the audio client.
|
|
*/
|
|
int q6asm_get_session_id(struct audio_client *c)
|
|
{
|
|
return c->session;
|
|
}
|
|
EXPORT_SYMBOL_GPL(q6asm_get_session_id);
|
|
|
|
/**
|
|
* q6asm_audio_client_alloc() - Allocate a new audio client
|
|
*
|
|
* @dev: Pointer to asm child device.
|
|
* @cb: event callback.
|
|
* @priv: private data associated with this client.
|
|
* @stream_id: stream id
|
|
* @perf_mode: performace mode for this client
|
|
*
|
|
* Return: Will be an error pointer on error or a valid audio client
|
|
* on success.
|
|
*/
|
|
struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
|
|
void *priv, int stream_id,
|
|
int perf_mode)
|
|
{
|
|
struct q6asm *a = dev_get_drvdata(dev->parent);
|
|
struct audio_client *ac;
|
|
unsigned long flags;
|
|
|
|
ac = q6asm_get_audio_client(a, stream_id + 1);
|
|
if (ac) {
|
|
dev_err(dev, "Audio Client already active\n");
|
|
return ac;
|
|
}
|
|
|
|
ac = kzalloc(sizeof(*ac), GFP_KERNEL);
|
|
if (!ac)
|
|
return ERR_PTR(-ENOMEM);
|
|
|
|
spin_lock_irqsave(&a->slock, flags);
|
|
a->session[stream_id + 1] = ac;
|
|
spin_unlock_irqrestore(&a->slock, flags);
|
|
ac->session = stream_id + 1;
|
|
ac->cb = cb;
|
|
ac->dev = dev;
|
|
ac->q6asm = a;
|
|
ac->priv = priv;
|
|
ac->io_mode = ASM_SYNC_IO_MODE;
|
|
ac->perf_mode = perf_mode;
|
|
/* DSP expects stream id from 1 */
|
|
ac->stream_id = 1;
|
|
ac->adev = a->adev;
|
|
kref_init(&ac->refcount);
|
|
|
|
init_waitqueue_head(&ac->cmd_wait);
|
|
mutex_init(&ac->cmd_lock);
|
|
spin_lock_init(&ac->lock);
|
|
|
|
return ac;
|
|
}
|
|
EXPORT_SYMBOL_GPL(q6asm_audio_client_alloc);
|
|
|
|
static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
|
|
{
|
|
struct apr_hdr *hdr = &pkt->hdr;
|
|
int rc;
|
|
|
|
mutex_lock(&ac->cmd_lock);
|
|
ac->result.opcode = 0;
|
|
ac->result.status = 0;
|
|
|
|
rc = apr_send_pkt(ac->adev, pkt);
|
|
if (rc < 0)
|
|
goto err;
|
|
|
|
rc = wait_event_timeout(ac->cmd_wait,
|
|
(ac->result.opcode == hdr->opcode), 5 * HZ);
|
|
if (!rc) {
|
|
dev_err(ac->dev, "CMD timeout\n");
|
|
rc = -ETIMEDOUT;
|
|
goto err;
|
|
}
|
|
|
|
if (ac->result.status > 0) {
|
|
dev_err(ac->dev, "DSP returned error[%x]\n",
|
|
ac->result.status);
|
|
rc = -EINVAL;
|
|
} else {
|
|
rc = 0;
|
|
}
|
|
|
|
|
|
err:
|
|
mutex_unlock(&ac->cmd_lock);
|
|
return rc;
|
|
}
|
|
|
|
/**
|
|
* q6asm_open_write() - Open audio client for writing
|
|
*
|
|
* @ac: audio client pointer
|
|
* @format: audio sample format
|
|
* @bits_per_sample: bits per sample
|
|
*
|
|
* Return: Will be an negative value on error or zero on success
|
|
*/
|
|
int q6asm_open_write(struct audio_client *ac, uint32_t format,
|
|
u32 codec_profile, 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 SND_AUDIOCODEC_MP3:
|
|
open->dec_fmt_id = ASM_MEDIA_FMT_MP3;
|
|
break;
|
|
case FORMAT_LINEAR_PCM:
|
|
open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
|
|
break;
|
|
case SND_AUDIOCODEC_FLAC:
|
|
open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
|
|
break;
|
|
case SND_AUDIOCODEC_WMA:
|
|
switch (codec_profile) {
|
|
case SND_AUDIOPROFILE_WMA9:
|
|
open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
|
|
break;
|
|
case SND_AUDIOPROFILE_WMA10:
|
|
case SND_AUDIOPROFILE_WMA9_PRO:
|
|
case SND_AUDIOPROFILE_WMA9_LOSSLESS:
|
|
case SND_AUDIOPROFILE_WMA10_LOSSLESS:
|
|
open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
|
|
break;
|
|
default:
|
|
dev_err(ac->dev, "Invalid codec profile 0x%x\n",
|
|
codec_profile);
|
|
rc = -EINVAL;
|
|
goto err;
|
|
}
|
|
break;
|
|
case SND_AUDIOCODEC_ALAC:
|
|
open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
|
|
break;
|
|
case SND_AUDIOCODEC_APE:
|
|
open->dec_fmt_id = ASM_MEDIA_FMT_APE;
|
|
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);
|
|
|
|
|
|
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
|
|
struct q6asm_flac_cfg *cfg)
|
|
{
|
|
struct asm_flac_fmt_blk_v2 *fmt;
|
|
struct apr_pkt *pkt;
|
|
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->is_stream_info_present = cfg->stream_info_present;
|
|
fmt->num_channels = cfg->ch_cfg;
|
|
fmt->min_blk_size = cfg->min_blk_size;
|
|
fmt->max_blk_size = cfg->max_blk_size;
|
|
fmt->sample_rate = cfg->sample_rate;
|
|
fmt->min_frame_size = cfg->min_frame_size;
|
|
fmt->max_frame_size = cfg->max_frame_size;
|
|
fmt->sample_size = cfg->sample_size;
|
|
|
|
rc = q6asm_ac_send_cmd_sync(ac, pkt);
|
|
kfree(pkt);
|
|
|
|
return rc;
|
|
}
|
|
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
|
|
|
|
int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
|
|
struct q6asm_wma_cfg *cfg)
|
|
{
|
|
struct asm_wmastdv9_fmt_blk_v2 *fmt;
|
|
struct apr_pkt *pkt;
|
|
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->fmtag = cfg->fmtag;
|
|
fmt->num_channels = cfg->num_channels;
|
|
fmt->sample_rate = cfg->sample_rate;
|
|
fmt->bytes_per_sec = cfg->bytes_per_sec;
|
|
fmt->blk_align = cfg->block_align;
|
|
fmt->bits_per_sample = cfg->bits_per_sample;
|
|
fmt->channel_mask = cfg->channel_mask;
|
|
fmt->enc_options = cfg->enc_options;
|
|
fmt->reserved = 0;
|
|
|
|
rc = q6asm_ac_send_cmd_sync(ac, pkt);
|
|
kfree(pkt);
|
|
|
|
return rc;
|
|
}
|
|
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
|
|
|
|
int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
|
|
struct q6asm_wma_cfg *cfg)
|
|
{
|
|
struct asm_wmaprov10_fmt_blk_v2 *fmt;
|
|
struct apr_pkt *pkt;
|
|
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->fmtag = cfg->fmtag;
|
|
fmt->num_channels = cfg->num_channels;
|
|
fmt->sample_rate = cfg->sample_rate;
|
|
fmt->bytes_per_sec = cfg->bytes_per_sec;
|
|
fmt->blk_align = cfg->block_align;
|
|
fmt->bits_per_sample = cfg->bits_per_sample;
|
|
fmt->channel_mask = cfg->channel_mask;
|
|
fmt->enc_options = cfg->enc_options;
|
|
fmt->advanced_enc_options1 = cfg->adv_enc_options;
|
|
fmt->advanced_enc_options2 = cfg->adv_enc_options2;
|
|
|
|
rc = q6asm_ac_send_cmd_sync(ac, pkt);
|
|
kfree(pkt);
|
|
|
|
return rc;
|
|
}
|
|
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
|
|
|
|
int q6asm_stream_media_format_block_alac(struct audio_client *ac,
|
|
struct q6asm_alac_cfg *cfg)
|
|
{
|
|
struct asm_alac_fmt_blk_v2 *fmt;
|
|
struct apr_pkt *pkt;
|
|
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->frame_length = cfg->frame_length;
|
|
fmt->compatible_version = cfg->compatible_version;
|
|
fmt->bit_depth = cfg->bit_depth;
|
|
fmt->num_channels = cfg->num_channels;
|
|
fmt->max_run = cfg->max_run;
|
|
fmt->max_frame_bytes = cfg->max_frame_bytes;
|
|
fmt->avg_bit_rate = cfg->avg_bit_rate;
|
|
fmt->sample_rate = cfg->sample_rate;
|
|
fmt->channel_layout_tag = cfg->channel_layout_tag;
|
|
fmt->pb = cfg->pb;
|
|
fmt->mb = cfg->mb;
|
|
fmt->kb = cfg->kb;
|
|
|
|
rc = q6asm_ac_send_cmd_sync(ac, pkt);
|
|
kfree(pkt);
|
|
|
|
return rc;
|
|
}
|
|
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
|
|
|
|
int q6asm_stream_media_format_block_ape(struct audio_client *ac,
|
|
struct q6asm_ape_cfg *cfg)
|
|
{
|
|
struct asm_ape_fmt_blk_v2 *fmt;
|
|
struct apr_pkt *pkt;
|
|
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->compatible_version = cfg->compatible_version;
|
|
fmt->compression_level = cfg->compression_level;
|
|
fmt->format_flags = cfg->format_flags;
|
|
fmt->blocks_per_frame = cfg->blocks_per_frame;
|
|
fmt->final_frame_blocks = cfg->final_frame_blocks;
|
|
fmt->total_frames = cfg->total_frames;
|
|
fmt->bits_per_sample = cfg->bits_per_sample;
|
|
fmt->num_channels = cfg->num_channels;
|
|
fmt->sample_rate = cfg->sample_rate;
|
|
fmt->seek_table_present = cfg->seek_table_present;
|
|
|
|
rc = q6asm_ac_send_cmd_sync(ac, pkt);
|
|
kfree(pkt);
|
|
|
|
return rc;
|
|
}
|
|
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
|
|
|
|
/**
|
|
* 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: length 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)
|
|
{
|
|
struct device *dev = &adev->dev;
|
|
struct q6asm *q6asm;
|
|
|
|
q6asm = devm_kzalloc(dev, sizeof(*q6asm), GFP_KERNEL);
|
|
if (!q6asm)
|
|
return -ENOMEM;
|
|
|
|
q6core_get_svc_api_info(adev->svc_id, &q6asm->ainfo);
|
|
|
|
q6asm->dev = dev;
|
|
q6asm->adev = adev;
|
|
init_waitqueue_head(&q6asm->mem_wait);
|
|
spin_lock_init(&q6asm->slock);
|
|
dev_set_drvdata(dev, q6asm);
|
|
|
|
return of_platform_populate(dev->of_node, NULL, NULL, dev);
|
|
}
|
|
|
|
static int q6asm_remove(struct apr_device *adev)
|
|
{
|
|
of_platform_depopulate(&adev->dev);
|
|
|
|
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");
|