Loading 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 Loading Loading @@ -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 Loading Loading @@ -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. Loading Loading @@ -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 Loading Loading @@ -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 Loading Loading @@ -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 Loading Loading @@ -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 { Loading Loading @@ -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. */ Loading Loading @@ -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 Loading Loading @@ -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. */ Loading Loading @@ -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; Loading Loading @@ -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 Loading Loading @@ -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 Loading Loading @@ -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. Loading @@ -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. */ Loading Loading @@ -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. Loading Loading @@ -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 Loading @@ -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.*/ Loading Loading
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 Loading Loading @@ -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 Loading Loading @@ -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. Loading Loading @@ -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 Loading Loading @@ -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 Loading Loading @@ -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 Loading Loading @@ -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 { Loading Loading @@ -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. */ Loading Loading @@ -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 Loading Loading @@ -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. */ Loading Loading @@ -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; Loading Loading @@ -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 Loading Loading @@ -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 Loading Loading @@ -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. Loading @@ -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. */ Loading Loading @@ -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. Loading Loading @@ -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 Loading @@ -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.*/ Loading
