Donate to e Foundation | Murena handsets with /e/OS | Own a part of Murena! Learn more

Commit 9ae73bba authored by Cong Tang's avatar Cong Tang
Browse files

ASoC: msm: Update Audio Header File for 32 Channels Support 



Update audio header files include new macro and structure definitions
for 32 channels support.

Change-Id: Idf5e92f7fda4b820b1a7f01001a683772281d8ba
Signed-off-by: default avatarCong Tang <congt@codeaurora.org>
parent 0b1ba172

include/sound/apr_audio-v2.h

+606 −3
Original line number Diff line number Diff line 
/* Copyright (c) 2012-2017, The Linux Foundation. All rights reserved.

/* Copyright (c) 2012-2018, The Linux Foundation. All rights reserved.

*

* This program is free software; you can redistribute it and/or modify

* it under the terms of the GNU General Public License version 2 and
@@ -214,6 +214,16 @@ struct adm_cmd_matrix_map_routings_v5 { 
*/

#define ADM_CMD_DEVICE_OPEN_V6                      0x00010356



/* This command allows a client to open a COPP/Voice Proc the

*	way as ADM_CMD_DEVICE_OPEN_V8 but supports any number channel

*	of configuration.

*

*	@return

*	#ADM_CMDRSP_DEVICE_OPEN_V8 with the resulting status and

*	COPP ID.

*/

#define ADM_CMD_DEVICE_OPEN_V8                      0x0001036A



/* Definition for a low latency stream session. */

#define ADM_LOW_LATENCY_DEVICE_SESSION			0x2000
@@ -492,6 +502,179 @@ struct adm_cmd_device_open_v6 { 
 */

} __packed;



/*  ADM device open command payload of the

*   #ADM_CMD_DEVICE_OPEN_V8 command.

*/

struct adm_cmd_device_open_v8 {

	struct apr_hdr       hdr;

	u16                  flags;

/* Bit width Native mode enabled : 11th bit of flag parameter

*  If 11th bit of flag is set then that means matrix mixer will be

*  running in native mode for bit width for this device session.

*

*  Channel Native mode enabled : 12th bit of flag parameter

*  If 12th bit of flag is set then that means matrix mixer will be

*  running in native mode for channel configuration for this device session.

*  All other bits are reserved; clients must set them to 0.

**/

	u16                  mode_of_operation;

/* Specifies whether the COPP must be opened on the Tx or Rx

 * path. Use the ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_* macros for

 * supported values and interpretation.

 * Supported values:

 * - 0x1 -- Rx path COPP

 * - 0x2 -- Tx path live COPP

 * - 0x3 -- Tx path nonlive COPP

 * Live connections cause sample discarding in the Tx device

 * matrix if the destination output ports do not pull them

 * fast enough. Nonlive connections queue the samples

 * indefinitely.

 */

	u32                  topology_id;

	/* Audio COPP topology ID; 32-bit GUID. */





	u16                  endpoint_id_1;

/* Logical and physical endpoint ID of the audio path.

 * If the ID is a voice processor Tx block, it receives near

 * samples.	Supported values: Any pseudoport, AFE Rx port,

 * or AFE Tx port For a list of valid IDs, refer to

 * @xhyperref{Q4,[Q4]}.

 * Q4 = Hexagon Multimedia: AFE Interface Specification

 */



	u16                  endpoint_id_2;

/* Logical and physical endpoint ID 2 for a voice processor

 * Tx block.

 * This is not applicable to audio COPP.

 * Supported values:

 * - AFE Rx port

 * - 0xFFFF -- Endpoint 2 is unavailable and the voice

 * processor Tx

 * block ignores this endpoint

 * When the voice processor Tx block is created on the audio

 * record path,

 * it can receive far-end samples from an AFE Rx port if the

 * voice call

 * is active. The ID of the AFE port is provided in this

 * field.

 * For a list of valid IDs, refer @xhyperref{Q4,[Q4]}.

 */



/*

 * Logical and physical endpoint ID of the audio path.

 * This indicated afe rx port in ADM loopback use cases.

 * In all other use cases this should be set to 0xffff

 */

	u16                  endpoint_id_3;

	u16                  reserved;



	u16                  dev_num_channel;

/* Number of channels the audio COPP sends to/receives from

 * the endpoint.

 * Supported values: 1 to 32.

 * The value is ignored for the voice processor Tx block,

 * where channel

 * configuration is derived from the topology ID.

 */



	u16                  bit_width;

/* Bit width (in bits) that the audio COPP sends to/receives

 * from the

 * endpoint. The value is ignored for the voice processing

 * Tx block,

 * where the PCM width is 16 bits.

 */



	u32                  sample_rate;

/* Sampling rate at which the audio COPP/voice processor

 * Tx block

 * interfaces with the endpoint.

 * Supported values for voice processor Tx: 8000, 16000,

 * 48000 Hz

 * Supported values for audio COPP: >0 and <=192 kHz

 */



	u8                   dev_channel_mapping[32];

/* Array of channel mapping of buffers that the audio COPP

 * sends to the endpoint. Channel[i] mapping describes channel

 * I inside the buffer, where 0 < i < dev_num_channel.

 * This value is relevant only for an audio Rx COPP.

 * For the voice processor block and Tx audio block, this field

 * is set to zero and is ignored.

 */



	u16                  dev_num_channel_eid2;

/* Number of channels the audio COPP sends to/receives from

 * the endpoint.

 * Supported values: 1 to 32.

 * The value is ignored for the voice processor Tx block,

 * where channel

 * configuration is derived from the topology ID.

 */



	u16                  bit_width_eid2;

/* Bit width (in bits) that the audio COPP sends to/receives

 * from the

 * endpoint. The value is ignored for the voice processing

 * Tx block,

 * where the PCM width is 16 bits.

 */



	u32                  sample_rate_eid2;

/* Sampling rate at which the audio COPP/voice processor

 * Tx block

 * interfaces with the endpoint.

 * Supported values for voice processor Tx: 8000, 16000,

 * 48000 Hz

 * Supported values for audio COPP: >0 and <=192 kHz

 */



	u8                   dev_channel_mapping_eid2[32];

/* Array of channel mapping of buffers that the audio COPP

 * sends to the endpoint. Channel[i] mapping describes channel

 * I inside the buffer, where 0 < i < dev_num_channel.

 * This value is relevant only for an audio Rx COPP.

 * For the voice processor block and Tx audio block, this field

 * is set to zero and is ignored.

 */



	u16                  dev_num_channel_eid3;

/* Number of channels the audio COPP sends to/receives from

 * the endpoint.

 * Supported values: 1 to 32.

 * The value is ignored for the voice processor Tx block,

 * where channel

 * configuration is derived from the topology ID.

 */



	u16                  bit_width_eid3;

/* Bit width (in bits) that the audio COPP sends to/receives

 * from the

 * endpoint. The value is ignored for the voice processing

 * Tx block,

 * where the PCM width is 16 bits.

 */



	u32                  sample_rate_eid3;

/* Sampling rate at which the audio COPP/voice processor

 * Tx block

 * interfaces with the endpoint.

 * Supported values for voice processor Tx: 8000, 16000,

 * 48000 Hz

 * Supported values for audio COPP: >0 and <=192 kHz

 */



	u8                   dev_channel_mapping_eid3[32];

/* Array of channel mapping of buffers that the audio COPP

 * sends to the endpoint. Channel[i] mapping describes channel

 * I inside the buffer, where 0 < i < dev_num_channel.

 * This value is relevant only for an audio Rx COPP.

 * For the voice processor block and Tx audio block, this field

 * is set to zero and is ignored.

 */

} __packed;



/*

 *	This command allows the client to close a COPP and disconnect

 *	the device session.
@@ -661,6 +844,10 @@ struct adm_cmd_rsp_device_open_v5 { 
 */

#define ADM_CMDRSP_DEVICE_OPEN_V6                      0x00010357



/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V8 command.

 */

#define ADM_CMDRSP_DEVICE_OPEN_V8                      0x0001036B



/*  Payload of the #ADM_CMDRSP_DEVICE_OPEN_V6 message,

 *	which returns the

 *	status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V6 command
@@ -863,6 +1050,12 @@ struct audproc_enable_param_t { 
 */

#define ADM_CMD_MATRIX_RAMP_GAINS_V5                                 0x0001032C



/*

 * Allows a client to control the gains on various session-to-COPP paths.

 * Maximum support 32 channels

 */

#define ADM_CMD_MATRIX_RAMP_GAINS_V7                                 0x0001036C



/* Indicates that the target gain in the

 *	current adm_session_copp_gain_v5

 *	structure is to be applied to all
@@ -975,12 +1168,97 @@ struct adm_session_copp_gain_v5 { 
	/* Target linear gain for channel 8 in Q13 format; */

} __packed;



/*  Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V7 command.

 * Immediately following this structure are num_gains of the

 * adm_session_copp_gain_v5structure.

 */

struct adm_cmd_matrix_ramp_gains_v7 {

	struct apr_hdr       hdr;

	u32                  matrix_id;

/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).

 * Use the ADM_MATRIX_ID_AUDIO_RX or  ADM_MATRIX_ID_AUDIOX

 * macros to set this field.

*/



	u16                  num_gains;

	/* Number of gains being applied. */



	u16                  reserved_for_align;

	/* Reserved. This field must be set to zero.*/

} __packed;



/*  Session-to-COPP path gain structure, used by the

 *	#ADM_CMD_MATRIX_RAMP_GAINS_V7 command.

 *	This structure specifies the target

 *	gain (per channel) that must be applied

 *	to a particular session-to-COPP path in

 *	the audio matrix. The structure can

 *	also be used to apply the gain globally

 *	to all session-to-COPP paths that

 *	exist for the given session.

 *	The aDSP uses device channel mapping to

 *	determine which channel gains to

 *	use from this command. For example,

 *	if the device is configured as stereo,

 *	the aDSP uses only target_gain_ch_1 and

 *	target_gain_ch_2, and it ignores

 *	the others.

 */

struct adm_session_copp_gain_v7 {

	u16                  session_id;

/* Handle of the ASM session.

 *	Supported values: 1 to 8.

 */



	u16                  copp_id;

/* Handle of the COPP. Gain will be applied on the Session ID

 * COPP ID path.

 */



	u16                  ramp_duration;

/* Duration (in milliseconds) of the ramp over

 * which target gains are

 * to be applied. Use

 * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE

 * to indicate that gain must be applied immediately.

 */



	u16                  step_duration;

/* Duration (in milliseconds) of each step in the ramp.

 * This parameter is ignored if ramp_duration is equal to

 * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE.

 * Supported value: 1

 */



	u16                  ramp_curve;

/* Type of ramping curve.

 * Supported value: #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR

 */



	u16                  stream_type;

/* Type of stream_type.

 * Supported value: #STREAM_TYPE_ASM STREAM_TYPE_LSM

 */

	u16                  num_channels;

/* Number of channels on which gain needs to be applied.

 * Supported value: 1 to 32.

 */

	u16                  reserved_for_align;

	/* Reserved. This field must be set to zero. */

} __packed;



/* Allows to set mute/unmute on various session-to-COPP paths.

 *	For every session-to-COPP path (stream-device interconnection),

 *	mute/unmute can be set individually on the output channels.

 */

#define ADM_CMD_MATRIX_MUTE_V5                                0x0001032D



/* Allows to set mute/unmute on various session-to-COPP paths.

 *	For every session-to-COPP path (stream-device interconnection),

 *	mute/unmute can be set individually on the output channels.

 */

#define ADM_CMD_MATRIX_MUTE_V7                                0x0001036D



/* Indicates that mute/unmute in the

 *	current adm_session_copp_mute_v5structure

 *	is to be applied to all the session-to-COPP
@@ -1046,6 +1324,48 @@ struct adm_cmd_matrix_mute_v5 { 
	/* Clients must set this field to zero.*/

} __packed;



/*  Payload of the #ADM_CMD_MATRIX_MUTE_V7 command*/

struct adm_cmd_matrix_mute_v7 {

	struct apr_hdr       hdr;

	u32                  matrix_id;

/* Specifies whether the matrix ID is Audio Rx (0) or Audio Tx (1).

 * Use the ADM_MATRIX_ID_AUDIO_RX or  ADM_MATRIX_ID_AUDIOX

 * macros to set this field.

 */



	u16                  session_id;

/* Handle of the ASM session.

 * Supported values: 1 to .

 */



	u16                  copp_id;

/* Handle of the COPP.

 * Use ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS

 * to indicate that mute/unmute must be applied to

 * all the COPPs connected to session_id.

 * Supported values:

 * - 0xFFFF -- Apply mute/unmute to all connected COPPs

 * - Other values -- Valid COPP ID

 */



	u16                  ramp_duration;

/* Duration (in milliseconds) of the ramp over

 * which target gains are

 * to be applied. Use

 * #ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE

 * to indicate that gain must be applied immediately.

 */



	u16                  stream_type;

/* Specify whether the stream type is connectedon the ASM or LSM

 * Supported value: 1

 */

	u16                  num_channels;

/* Number of channels on which gain needs to be applied

 * Supported value: 1 to 32

 */

} __packed;



#define ASM_PARAM_ID_AAC_STEREO_MIX_COEFF_SELECTION_FLAG_V2 (0x00010DD8)



struct asm_aac_stereo_mix_coeff_selection_param_v2 {
@@ -2646,6 +2966,8 @@ struct afe_param_id_internal_bt_fm_cfg { 


#define AFE_PORT_MAX_AUDIO_CHAN_CNT	0x8



#define AFE_PORT_MAX_AUDIO_CHAN_CNT_V2	0x20



/* Payload of the #AFE_PORT_CMD_SLIMBUS_CONFIG command's SLIMbus

 * port configuration parameter.

 */
@@ -3075,6 +3397,11 @@ struct afe_param_id_tdm_cfg { 
*/

#define AFE_API_VERSION_SLOT_MAPPING_CONFIG	0x1



/** Version information used to handle future additions to slot mapping

*	configuration support 32 channels.

*/

#define AFE_API_VERSION_SLOT_MAPPING_CONFIG_V2	0x2



/** Data align type  */

#define AFE_SLOT_MAPPING_DATA_ALIGN_MSB		0

#define AFE_SLOT_MAPPING_DATA_ALIGN_LSB		1
@@ -3118,6 +3445,49 @@ struct afe_param_id_slot_mapping_cfg { 
	@values, in byte*/

} __packed;



/* Payload of the AFE_PARAM_ID_PORT_SLOT_MAPPING_CONFIG_V2

*  command's TDM configuration parameter.

*/

struct afe_param_id_slot_mapping_cfg_v2 {

	u32	minor_version;

	/**< Minor version used for tracking TDM slot configuration.

	 * @values #AFE_API_VERSION_TDM_SLOT_CONFIG

	 */



	u16	num_channel;

	/**< number of channel of the audio sample.

	* @values 1, 2, 4, 6, 8, 16, 32 @tablebulletend

	*/



	u16	bitwidth;

	/**< Slot bit width for each channel

	* @values 16, 24, 32

	*/



	u32	data_align_type;

	/**< indicate how data packed from slot_offset for 32 slot bit width

	* in case of sample bit width is 24.

	* @values

	* #AFE_SLOT_MAPPING_DATA_ALIGN_MSB

	* #AFE_SLOT_MAPPING_DATA_ALIGN_LSB

	*/



	u16	offset[AFE_PORT_MAX_AUDIO_CHAN_CNT_V2];

	/**< Array of the slot mapping start offset in bytes for this frame.

	* The bytes is counted from 0. The 0 is mapped to the 1st byte

	* in or out of the digital serial data line this sub-frame belong to.

	* slot_offset[] setting is per-channel based.

	* The max num of channel supported is 8.

	* The valid offset value must always be continuly placed in

	* from index 0.

	* Set offset as AFE_SLOT_MAPPING_OFFSET_INVALID for not used arrays.

	* If "slot_bitwidth_per_channel" is 32 and "sample_bitwidth" is 24,

	* "data_align_type" is used to indicate how 24 bit sample data in

	* aligning with 32 bit slot width per-channel.

	* @values, in byte

	*/

} __packed;



/** ID of the parameter used by #AFE_MODULE_TDM to configure

    the customer TDM header. #AFE_PORT_CMD_SET_PARAM can use this parameter ID.

*/
@@ -3183,6 +3553,7 @@ struct afe_param_id_custom_tdm_header_cfg { 
struct afe_tdm_port_config {

	struct afe_param_id_tdm_cfg				tdm;

	struct afe_param_id_slot_mapping_cfg		slot_mapping;

	struct afe_param_id_slot_mapping_cfg_v2		slot_mapping_v2;

	struct afe_param_id_custom_tdm_header_cfg	custom_tdm_header;

} __packed;
@@ -4139,6 +4510,23 @@ struct asm_stream_pan_ctrl_params { 
	uint32_t gain[64];

} __packed;



struct adm_matrix_ramp_gains_params {

	uint16_t session_id;

	uint16_t be_id;

	uint16_t num_gains;

	uint16_t path;

	uint16_t channels;

	uint16_t gain_value[32];

} __packed;



struct adm_matrix_mute_params {

	uint16_t session_id;

	uint16_t be_id;

	uint16_t channels;

	uint16_t path;

	uint8_t mute_flag[32];

} __packed;



#define ASM_END_POINT_DEVICE_MATRIX     0



#define PCM_CHANNEL_NULL 0
@@ -4191,14 +4579,78 @@ struct asm_stream_pan_ctrl_params { 
/* Rear right of center. */

#define PCM_CHANNEL_RRC  16



/* Second low frequency channel. */

#define PCM_CHANNEL_LFE2 17



/* Side left channel. */

#define PCM_CHANNEL_SL   18



/* Side right channel. */

#define PCM_CHANNEL_SR   19



/* Top front left channel. */

#define PCM_CHANNEL_TFL  20



/* Left vertical height channel. */

#define PCM_CHANNEL_LVH  20



/* Top front right channel. */

#define PCM_CHANNEL_TFR  21



/* Right vertical height channel. */

#define PCM_CHANNEL_RVH  21



/* Top center channel. */

#define PCM_CHANNEL_TC   22



/* Top back left channel. */

#define PCM_CHANNEL_TBL  23



/* Top back right channel. */

#define PCM_CHANNEL_TBR  24



/* Top side left channel. */

#define PCM_CHANNEL_TSL  25



/* Top side right channel. */

#define PCM_CHANNEL_TSR  26



/* Top back center channel. */

#define PCM_CHANNEL_TBC  27



/* Bottom front center channel. */

#define PCM_CHANNEL_BFC  28



/* Bottom front left channel. */

#define PCM_CHANNEL_BFL  29



/* Bottom front right channel. */

#define PCM_CHANNEL_BFR  30



/* Left wide channel. */

#define PCM_CHANNEL_LW   31



/* Right wide channel. */

#define PCM_CHANNEL_RW   32



/* Left side direct channel. */

#define PCM_CHANNEL_LSD  33



/* Right side direct channel. */

#define PCM_CHANNEL_RSD  34



#define PCM_FORMAT_MAX_NUM_CHANNEL 8



#define PCM_FORMAT_MAX_NUM_CHANNEL_V2  32



#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5



#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 0x00010DDC



#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 0x0001320C



#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5 0x00013222



#define ASM_MEDIA_FMT_EVRCB_FS 0x00010BEF



#define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0
@@ -4413,6 +4865,56 @@ struct asm_multi_channel_pcm_fmt_blk_v4 { 
 */

} __packed;



struct asm_multi_channel_pcm_fmt_blk_v5 {

	uint16_t                num_channels;

/*

 * Number of channels

 * Supported values: 1 to 32

 */



	uint16_t                bits_per_sample;

/*

 * Number of bits per sample per channel

 * Supported values: 16, 24, 32

 */



	uint32_t                sample_rate;

/*

 * Number of samples per second

 * Supported values: 2000 to 48000, 96000,192000 Hz

 */



	uint16_t                is_signed;

/* Flag that indicates that PCM samples are signed (1) */



	uint16_t                sample_word_size;

/*

 * Size in bits of the word that holds a sample of a channel.

 * Supported values: 12,24,32

 */

	uint16_t                endianness;

/*

 * Flag to indicate the endianness of the pcm sample

 * Supported values: 0 - Little endian (all other formats)

 *                   1 - Big endian (AIFF)

 */

	uint16_t                mode;

/*

 * Mode to provide additional info about the pcm input data.

 * Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,

 *                       Q31 for unpacked 24b or 32b)

 *                  15 - for 16 bit

 *                  23 - for 24b packed or 8.24 format

 *                  31 - for 24b unpacked or 32bit

 */



	uint8_t                 channel_mapping[PCM_FORMAT_MAX_NUM_CHANNEL_V2];

/*

 * Each element, i, in the array describes channel i inside the buffer where

 * 0 <= i < num_channels. Unused channels are set to 0.

 */

} __packed;



/*

 * Payload of the multichannel PCM configuration parameters in

 * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format.
@@ -4433,6 +4935,16 @@ struct asm_multi_channel_pcm_fmt_blk_param_v4 { 
	struct asm_multi_channel_pcm_fmt_blk_v4 param;

} __packed;



/*

 * Payload of the multichannel PCM configuration parameters in

 * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5 media format.

 */

struct asm_multi_channel_pcm_fmt_blk_param_v5 {

	struct apr_hdr hdr;

	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;

	struct asm_multi_channel_pcm_fmt_blk_v5 param;

} __packed;



struct asm_stream_cmd_set_encdec_param {

	u32                  param_id;

	/* ID of the parameter. */
@@ -4468,6 +4980,78 @@ struct asm_dec_ddp_endp_param_v2 { 
	int endp_param_value;

} __packed;



/*

 * Payload of the multichannel PCM encoder configuration parameters in

 * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5 media format.

 */

struct asm_multi_channel_pcm_enc_cfg_v5 {

	struct apr_hdr hdr;

	struct asm_stream_cmd_set_encdec_param encdec;

	struct asm_enc_cfg_blk_param_v2 encblk;

	uint16_t num_channels;

	/*

	 * Number of PCM channels.

	 * @values

	 * - 0 -- Native mode

	 * - 1 -- 8 channels

	 * Native mode indicates that encoding must be performed with the number

	 * of channels at the input.

	 */

	uint16_t  bits_per_sample;

	/*

	 * Number of bits per sample per channel.

	 * @values 16, 24

	 */

	uint32_t  sample_rate;

	/*

	 * Number of samples per second.

	 * @values 0, 8000 to 48000 Hz

	 * A value of 0 indicates the native sampling rate. Encoding is

	 * performed at the input sampling rate.

	 */

	uint16_t  is_signed;

	/*

	 * Flag that indicates the PCM samples are signed (1). Currently, only

	 * signed PCM samples are supported.

	 */

	uint16_t    sample_word_size;

	/*

	 * The size in bits of the word that holds a sample of a channel.

	 * @values 16, 24, 32

	 * 16-bit samples are always placed in 16-bit words:

	 * sample_word_size = 1.

	 * 24-bit samples can be placed in 32-bit words or in consecutive

	 * 24-bit words.

	 * - If sample_word_size = 32, 24-bit samples are placed in the

	 * most significant 24 bits of a 32-bit word.

	 * - If sample_word_size = 24, 24-bit samples are placed in

	 * 24-bit words. @tablebulletend

	 */

	uint16_t                endianness;

	/*

	 * Flag to indicate the endianness of the pcm sample

	 * Supported values: 0 - Little endian (all other formats)

	 *                   1 - Big endian (AIFF)

	 */

	uint16_t                mode;

	/*

	 * Mode to provide additional info about the pcm input data.

	 * Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,

	 *                       Q31 for unpacked 24b or 32b)

	 *                  15 - for 16 bit

	 *                  23 - for 24b packed or 8.24 format

	 */

	uint8_t   channel_mapping[PCM_FORMAT_MAX_NUM_CHANNEL_V2];

	/*

	 * Channel mapping array expected at the encoder output.

	 *  Channel[i] mapping describes channel i inside the buffer, where

	 *  0 @le i < num_channels. All valid used channels must be present at

	 *  the beginning of the array.

	 * If Native mode is set for the channels, this field is ignored.

	 * @values See Section @xref{dox:PcmChannelDefs}

	 */

} __packed;



/*

 * Payload of the multichannel PCM encoder configuration parameters in

 * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format.
@@ -7019,6 +7603,8 @@ struct asm_ac3_generic_param { 
/* Maximum number of decoder output channels.*/

#define MAX_CHAN_MAP_CHANNELS  16



#define MAX_CHAN_MAP_CHANNELS_V2 32



/* Structure for decoder output channel mapping. */



/* Payload of the #ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP parameter in the
@@ -7038,6 +7624,23 @@ struct asm_dec_out_chan_map_param { 
	u8                  channel_mapping[MAX_CHAN_MAP_CHANNELS];

} __packed;



/* Payload of the #ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP parameter in the

 * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.

 */

struct asm_dec_out_chan_map_param_v2 {

	struct apr_hdr hdr;

	struct asm_stream_cmd_set_encdec_param  encdec;

	u32                 num_channels;

/* Number of decoder output channels.

 * Supported values: 0 to #MAX_CHAN_MAP_CHANNELS_V2

 *

 * A value of 0 indicates native channel mapping, which is valid

 * only for NT mode. This means the output of the decoder is to be

 * preserved as is.

 */

	u8                  channel_mapping[MAX_CHAN_MAP_CHANNELS_V2];

} __packed;



#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED  0x00010D84



/* Bitmask for the IEC 61937 enable flag.*/