-/* 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
*/
#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
*/
} __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.
*/
#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
*/
#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
/* 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
/* 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 {
#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.
*/
*/
#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
@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.
*/
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;
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
/* Rear right of center. */
#define PCM_CHANNEL_RRC 16
-#define PCM_FORMAT_MAX_NUM_CHANNEL 8
+/* 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_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
*/
} __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.
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. */
/*
* 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.
*/
#define ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP 0x00010D82
-/* Maximum number of decoder output channels.*/
+/* 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
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.*/
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