linux_dsm_epyc7002/sound/soc/mid-x86/sst_dsp.h
Vinod Koul c514a9119a ASoC: mid-x86 - add support for compressed streams
Signed-off-by: Namarta Kohli <namartax.kohli@intel.com>
Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-20 20:50:39 +01:00

135 lines
3.8 KiB
C

#ifndef __SST_DSP_H__
#define __SST_DSP_H__
/*
* sst_dsp.h - Intel SST Driver for audio engine
*
* Copyright (C) 2008-12 Intel Corporation
* Authors: Vinod Koul <vinod.koul@linux.intel.com>
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; version 2 of the License.
*
* This program is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
enum sst_codec_types {
/* AUDIO/MUSIC CODEC Type Definitions */
SST_CODEC_TYPE_UNKNOWN = 0,
SST_CODEC_TYPE_PCM, /* Pass through Audio codec */
SST_CODEC_TYPE_MP3,
SST_CODEC_TYPE_MP24,
SST_CODEC_TYPE_AAC,
SST_CODEC_TYPE_AACP,
SST_CODEC_TYPE_eAACP,
};
enum stream_type {
SST_STREAM_TYPE_NONE = 0,
SST_STREAM_TYPE_MUSIC = 1,
};
struct snd_pcm_params {
u16 codec; /* codec type */
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
u32 reserved; /* Bitrate in bits per second */
u32 sfreq; /* Sampling rate in Hz */
u8 use_offload_path;
u8 reserved2;
u16 reserved3;
u8 channel_map[8];
} __packed;
/* MP3 Music Parameters Message */
struct snd_mp3_params {
u16 codec;
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
u8 crc_check; /* crc_check - disable (0) or enable (1) */
u8 reserved1; /* unused*/
u16 reserved2; /* Unused */
} __packed;
#define AAC_BIT_STREAM_ADTS 0
#define AAC_BIT_STREAM_ADIF 1
#define AAC_BIT_STREAM_RAW 2
/* AAC Music Parameters Message */
struct snd_aac_params {
u16 codec;
u8 num_chan; /* 1=Mono, 2=Stereo*/
u8 pcm_wd_sz; /* 16/24 - bit*/
u8 bdownsample; /*SBR downsampling 0 - disable 1 -enabled AAC+ only */
u8 bs_format; /* input bit stream format adts=0, adif=1, raw=2 */
u16 reser2;
u32 externalsr; /*sampling rate of basic AAC raw bit stream*/
u8 sbr_signalling;/*disable/enable/set automode the SBR tool.AAC+*/
u8 reser1;
u16 reser3;
} __packed;
/* WMA Music Parameters Message */
struct snd_wma_params {
u16 codec;
u8 num_chan; /* 1=Mono, 2=Stereo */
u8 pcm_wd_sz; /* 16/24 - bit*/
u32 brate; /* Use the hard coded value. */
u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */
u32 channel_mask; /* Channel Mask */
u16 format_tag; /* Format Tag */
u16 block_align; /* packet size */
u16 wma_encode_opt;/* Encoder option */
u8 op_align; /* op align 0- 16 bit, 1- MSB, 2 LSB */
u8 reserved; /* reserved */
} __packed;
/* Codec params struture */
union snd_sst_codec_params {
struct snd_pcm_params pcm_params;
struct snd_mp3_params mp3_params;
struct snd_aac_params aac_params;
struct snd_wma_params wma_params;
} __packed;
/* Address and size info of a frame buffer */
struct sst_address_info {
u32 addr; /* Address at IA */
u32 size; /* Size of the buffer */
};
struct snd_sst_alloc_params_ext {
struct sst_address_info ring_buf_info[8];
u8 sg_count;
u8 reserved;
u16 reserved2;
u32 frag_size; /*Number of samples after which period elapsed
message is sent valid only if path = 0*/
} __packed;
struct snd_sst_stream_params {
union snd_sst_codec_params uc;
} __packed;
struct snd_sst_params {
u32 stream_id;
u8 codec;
u8 ops;
u8 stream_type;
u8 device_type;
struct snd_sst_stream_params sparams;
struct snd_sst_alloc_params_ext aparams;
};
#endif /* __SST_DSP_H__ */