Merge commit 'AU_LINUX_ANDROID_ICS.04.00.04.00.126' into msm-3.4
AU_LINUX_ANDROID_ICS.04.00.04.00.126 from msm-3.0.
First parent is from google/android-3.4.
* commit 'AU_LINUX_ANDROID_ICS.04.00.04.00.126': (8712 commits)
PRNG: Device tree entry for qrng device.
vidc:1080p: Set video core timeout value for Thumbnail mode
msm: sps: improve the debugging support in SPS driver
board-8064 msm: Overlap secure and non secure video firmware heaps.
msm: clock: Add handoff ops for 7x30 and copper XO clocks
msm_fb: display: Wait for external vsync before DTV IOMMU unmap
msm: Fix ciruclar dependency in debug UART settings
msm: gdsc: Add GDSC regulator driver for msm-copper
defconfig: Enable Mobicore Driver.
mobicore: Add mobicore driver.
mobicore: rename variable to lower case.
mobicore: rename folder.
mobicore: add makefiles
mobicore: initial import of kernel driver
ASoC: msm: Add SLIMBUS_2_RX CPU DAI
board-8064-gpio: Update FUNC for EPM SPI CS
msm_fb: display: Remove chicken bit config during video playback
mmc: msm_sdcc: enable the sanitize capability
msm-fb: display: lm2 writeback support on mpq platfroms
msm_fb: display: Disable LVDS phy & pll during panel off
...
Signed-off-by: Steve Muckle <smuckle@codeaurora.org>
diff --git a/include/sound/Kbuild b/include/sound/Kbuild
index 6df30ed..739f289 100644
--- a/include/sound/Kbuild
+++ b/include/sound/Kbuild
@@ -8,3 +8,4 @@
header-y += sfnt_info.h
header-y += compress_params.h
header-y += compress_offload.h
+header-y += tlv.h
diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h
new file mode 100644
index 0000000..695fea9
--- /dev/null
+++ b/include/sound/apr_audio-v2.h
@@ -0,0 +1,6172 @@
+/* Copyright (c) 2012, Code Aurora Forum. 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
+* only version 2 as published by the Free Software Foundation.
+*
+* 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.
+*/
+
+
+#ifndef _APR_AUDIO_V2_H_
+#define _APR_AUDIO_V2_H_
+
+#include <mach/qdsp6v2/apr.h>
+
+#define ADSP_ADM_VERSION 0x00070000
+
+#define ADM_CMD_SHARED_MEM_MAP_REGIONS 0x00010322
+#define ADM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010323
+#define ADM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010324
+
+#define ADM_CMD_MATRIX_MAP_ROUTINGS_V5 0x00010325
+
+/* Enumeration for an audio Rx matrix ID.*/
+#define ADM_MATRIX_ID_AUDIO_RX 0
+
+#define ADM_MATRIX_ID_AUDIO_TX 1
+
+/* Enumeration for an audio Tx matrix ID.*/
+#define ADM_MATRIX_ID_AUDIOX 1
+
+#define ADM_MAX_COPPS 5
+
+
+/* Session map node structure.
+* Immediately following this structure are num_copps
+* entries of COPP IDs. The COPP IDs are 16 bits, so
+* there might be a padding 16-bit field if num_copps
+* is odd.
+*/
+struct adm_session_map_node_v5 {
+ u16 session_id;
+/* Handle of the ASM session to be routed. Supported values: 1
+* to 8.
+*/
+
+
+ u16 num_copps;
+ /* Number of COPPs to which this session is to be routed.
+ Supported values: 0 < num_copps <= ADM_MAX_COPPS.
+ */
+} __packed;
+
+/* Payload of the #ADM_CMD_MATRIX_MAP_ROUTINGS_V5 command.
+* Immediately following this structure are num_sessions of the session map
+* node payload (adm_session_map_node_v5).
+*/
+
+struct adm_cmd_matrix_map_routings_v5 {
+ 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.
+*/
+ u32 num_sessions;
+ /* Number of sessions being updated by this command (optional).*/
+} __packed;
+
+/* This command allows a client to open a COPP/Voice Proc. TX module
+* and sets up the device session: Matrix -> COPP -> AFE on the RX
+* and AFE -> COPP -> Matrix on the TX. This enables PCM data to
+* be transferred to/from the endpoint (AFEPortID).
+*
+* @return
+* #ADM_CMDRSP_DEVICE_OPEN_V5 with the resulting status and
+* COPP ID.
+*/
+#define ADM_CMD_DEVICE_OPEN_V5 0x00010326
+
+/* Indicates that endpoint_id_2 is to be ignored.*/
+#define ADM_CMD_COPP_OPEN_END_POINT_ID_2_IGNORE 0xFFFF
+
+#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATION_RX_PATH_COPP 1
+
+#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_LIVE_COPP 2
+
+#define ADM_CMD_COPP_OPEN_MODE_OF_OPERATIONX_PATH_NON_LIVE_COPP 3
+
+/* Indicates that an audio COPP is to send/receive a mono PCM
+ * stream to/from
+ * END_POINT_ID_1.
+ */
+#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_MONO 1
+
+/* Indicates that an audio COPP is to send/receive a
+ * stereo PCM stream to/from END_POINT_ID_1.
+ */
+#define ADM_CMD_COPP_OPEN_CHANNEL_CONFIG_STEREO 2
+
+/* Sample rate is 8000 Hz.*/
+#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_8K 8000
+
+/* Sample rate is 16000 Hz.*/
+#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_16K 16000
+
+/* Sample rate is 48000 Hz.*/
+#define ADM_CMD_COPP_OPEN_SAMPLE_RATE_48K 48000
+
+/* Definition for a COPP live input flag bitmask.*/
+#define ADM_BIT_MASK_COPP_LIVE_INPUT_FLAG (0x0001U)
+
+/* Definition for a COPP live shift value bitmask.*/
+#define ADM_SHIFT_COPP_LIVE_INPUT_FLAG 0
+
+/* Definition for the COPP ID bitmask.*/
+#define ADM_BIT_MASK_COPP_ID (0x0000FFFFUL)
+
+/* Definition for the COPP ID shift value.*/
+#define ADM_SHIFT_COPP_ID 0
+
+/* Definition for the service ID bitmask.*/
+#define ADM_BIT_MASK_SERVICE_ID (0x00FF0000UL)
+
+/* Definition for the service ID shift value.*/
+#define ADM_SHIFT_SERVICE_ID 16
+
+/* Definition for the domain ID bitmask.*/
+#define ADM_BIT_MASK_DOMAIN_ID (0xFF000000UL)
+
+/* Definition for the domain ID shift value.*/
+#define ADM_SHIFT_DOMAIN_ID 24
+
+/* ADM device open command payload of the
+ #ADM_CMD_DEVICE_OPEN_V5 command.
+*/
+struct adm_cmd_device_open_v5 {
+ struct apr_hdr hdr;
+ u16 flags;
+/* Reserved for future use. Clients must set this field
+ * to zero.
+ */
+
+ 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.
+ */
+
+ 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]}.
+ */
+
+ u32 topology_id;
+ /* Audio COPP topology ID; 32-bit GUID. */
+
+ u16 dev_num_channel;
+/* Number of channels the audio COPP sends to/receives from
+ * the endpoint.
+ * Supported values: 1 to 8.
+ * 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[8];
+/* 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 relevent 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_CMD_DEVICE_CLOSE_V5 0x00010327
+
+/* Sets one or more parameters to a COPP.
+*/
+#define ADM_CMD_SET_PP_PARAMS_V5 0x00010328
+
+/* Payload of the #ADM_CMD_SET_PP_PARAMS_V5 command.
+ * If the data_payload_addr_lsw and data_payload_addr_msw element
+ * are NULL, a series of adm_param_datastructures immediately
+ * follows, whose total size is data_payload_size bytes.
+ */
+struct adm_cmd_set_pp_params_v5 {
+ struct apr_hdr hdr;
+ u32 data_payload_addr_lsw;
+ /* LSW of parameter data payload address.*/
+ u32 data_payload_addr_msw;
+ /* MSW of parameter data payload address.*/
+
+ u32 mem_map_handle;
+/* Memory map handle returned by ADM_CMD_SHARED_MEM_MAP_REGIONS
+ * command */
+/* If mem_map_handle is zero implies the message is in
+ * the payload */
+
+ u32 data_payload_size;
+/* Size in bytes of the variable payload accompanying this
+ * message or
+ * in shared memory. This is used for parsing the parameter
+ * payload.
+ */
+} __packed;
+
+/* Payload format for COPP parameter data.
+ * Immediately following this structure are param_size bytes
+ * of parameter
+ * data.
+ */
+struct adm_param_data_v5 {
+ u32 module_id;
+ /* Unique ID of the module. */
+ u32 param_id;
+ /* Unique ID of the parameter. */
+ u16 param_size;
+ /* Data size of the param_id/module_id combination.
+ This value is a
+ multiple of 4 bytes. */
+ u16 reserved;
+ /* Reserved for future enhancements.
+ * This field must be set to zero.
+ */
+} __packed;
+
+/* Returns the status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
+ */
+#define ADM_CMDRSP_DEVICE_OPEN_V5 0x00010329
+
+/* Payload of the #ADM_CMDRSP_DEVICE_OPEN_V5 message,
+ * which returns the
+ * status and COPP ID to an #ADM_CMD_DEVICE_OPEN_V5 command.
+ */
+struct adm_cmd_rsp_device_open_v5 {
+ u32 status;
+ /* Status message (error code).*/
+
+ u16 copp_id;
+ /* COPP ID: Supported values: 0 <= copp_id < ADM_MAX_COPPS*/
+
+ u16 reserved;
+ /* Reserved. This field must be set to zero.*/
+} __packed;
+
+/* This command allows a query of one COPP parameter.
+*/
+#define ADM_CMD_GET_PP_PARAMS_V5 0x0001032A
+
+/* Payload an #ADM_CMD_GET_PP_PARAMS_V5 command.
+*/
+struct adm_cmd_get_pp_params_v5 {
+ u32 data_payload_addr_lsw;
+ /* LSW of parameter data payload address.*/
+
+ u32 data_payload_addr_msw;
+ /* MSW of parameter data payload address.*/
+
+ /* If the mem_map_handle is non zero,
+ * on ACK, the ParamData payloads begin at
+ * the address specified (out-of-band).
+ */
+
+ u32 mem_map_handle;
+ /* Memory map handle returned
+ * by ADM_CMD_SHARED_MEM_MAP_REGIONS command.
+ * If the mem_map_handle is 0, it implies that
+ * the ACK's payload will contain the ParamData (in-band).
+ */
+
+ u32 module_id;
+ /* Unique ID of the module. */
+
+ u32 param_id;
+ /* Unique ID of the parameter. */
+
+ u16 param_max_size;
+ /* Maximum data size of the parameter
+ *ID/module ID combination. This
+ * field is a multiple of 4 bytes.
+ */
+ u16 reserved;
+ /* Reserved for future enhancements.
+ * This field must be set to zero.
+ */
+} __packed;
+
+/* Returns parameter values
+ * in response to an #ADM_CMD_GET_PP_PARAMS_V5 command.
+ */
+#define ADM_CMDRSP_GET_PP_PARAMS_V5 0x0001032B
+
+/* Payload of the #ADM_CMDRSP_GET_PP_PARAMS_V5 message,
+ * which returns parameter values in response
+ * to an #ADM_CMD_GET_PP_PARAMS_V5 command.
+ * Immediately following this
+ * structure is the adm_param_data_v5
+ * structure containing the pre/postprocessing
+ * parameter data. For an in-band
+ * scenario, the variable payload depends
+ * on the size of the parameter.
+*/
+struct adm_cmd_rsp_get_pp_params_v5 {
+ u32 status;
+ /* Status message (error code).*/
+} __packed;
+
+/* Allows a client to control the gains on various session-to-COPP paths.
+ */
+#define ADM_CMD_MATRIX_RAMP_GAINS_V5 0x0001032C
+
+/* Indicates that the target gain in the
+ * current adm_session_copp_gain_v5
+ * structure is to be applied to all
+ * the session-to-COPP paths that exist for
+ * the specified session.
+ */
+#define ADM_CMD_MATRIX_RAMP_GAINS_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF
+
+/* Indicates that the target gain is
+ * to be immediately applied to the
+ * specified session-to-COPP path,
+ * without a ramping fashion.
+ */
+#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_DURATION_IMMEDIATE 0x0000
+
+/* Enumeration for a linear ramping curve.*/
+#define ADM_CMD_MATRIX_RAMP_GAINS_RAMP_CURVE_LINEAR 0x0000
+
+/* Payload of the #ADM_CMD_MATRIX_RAMP_GAINS_V5 command.
+ * Immediately following this structure are num_gains of the
+ * adm_session_copp_gain_v5structure.
+ */
+struct adm_cmd_matrix_ramp_gains_v5 {
+ 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_V5 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_v5 {
+ 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 reserved_for_align;
+ /* Reserved. This field must be set to zero. */
+
+ u16 target_gain_ch_1;
+ /* Target linear gain for channel 1 in Q13 format; */
+
+ u16 target_gain_ch_2;
+ /* Target linear gain for channel 2 in Q13 format; */
+
+ u16 target_gain_ch_3;
+ /* Target linear gain for channel 3 in Q13 format; */
+
+ u16 target_gain_ch_4;
+ /* Target linear gain for channel 4 in Q13 format; */
+
+ u16 target_gain_ch_5;
+ /* Target linear gain for channel 5 in Q13 format; */
+
+ u16 target_gain_ch_6;
+ /* Target linear gain for channel 6 in Q13 format; */
+
+ u16 target_gain_ch_7;
+ /* Target linear gain for channel 7 in Q13 format; */
+
+ u16 target_gain_ch_8;
+ /* Target linear gain for channel 8 in Q13 format; */
+} __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
+
+/* Indicates that mute/unmute in the
+ * current adm_session_copp_mute_v5structure
+ * is to be applied to all the session-to-COPP
+ * paths that exist for the specified session.
+ */
+#define ADM_CMD_MATRIX_MUTE_COPP_ID_ALL_CONNECTED_COPPS 0xFFFF
+
+/* Payload of the #ADM_CMD_MATRIX_MUTE_V5 command*/
+struct adm_cmd_matrix_mute_v5 {
+ 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 8.
+ */
+
+ 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
+ */
+
+ u8 mute_flag_ch_1;
+ /* Mute flag for channel 1 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_2;
+ /* Mute flag for channel 2 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_3;
+ /* Mute flag for channel 3 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_4;
+ /* Mute flag for channel 4 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_5;
+ /* Mute flag for channel 5 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_6;
+ /* Mute flag for channel 6 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_7;
+ /* Mute flag for channel 7 is set to unmute (0) or mute (1). */
+
+ u8 mute_flag_ch_8;
+ /* Mute flag for channel 8 is set to unmute (0) or mute (1). */
+
+ u16 ramp_duration;
+/* Period (in milliseconds) over which the soft mute/unmute will be
+ * applied.
+ * Supported values: 0 (Default) to 0xFFFF
+ * The default of 0 means mute/unmute will be applied immediately.
+ */
+
+ u16 reserved_for_align;
+ /* Clients must set this field to zero.*/
+} __packed;
+
+/* Allows a client to connect the desired stream to
+ * the desired AFE port through the stream router
+ *
+ * This command allows the client to connect specified session to
+ * specified AFE port. This is used for compressed streams only
+ * opened using the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or
+ * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED command.
+ *
+ * @prerequisites
+ * Session ID and AFE Port ID must be valid.
+ * #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED or
+ * #ASM_STREAM_CMD_OPEN_READ_COMPRESSED
+ * must have been called on this session.
+ */
+
+#define ADM_CMD_CONNECT_AFE_PORT_V5 0x0001032E
+#define ADM_CMD_DISCONNECT_AFE_PORT_V5 0x0001032F
+/* Enumeration for the Rx stream router ID.*/
+#define ADM_STRTR_ID_RX 0
+/* Enumeration for the Tx stream router ID.*/
+#define ADM_STRTR_IDX 1
+
+/* Payload of the #ADM_CMD_CONNECT_AFE_PORT_V5 command.*/
+struct adm_cmd_connect_afe_port_v5 {
+ u8 mode;
+/* ID of the stream router (RX/TX). Use the
+ * ADM_STRTR_ID_RX or ADM_STRTR_IDX macros
+ * to set this field.
+ */
+
+ u8 session_id;
+ /* Session ID of the stream to connect */
+
+ u16 afe_port_id;
+ /* Port ID of the AFE port to connect to.*/
+ u32 num_channels;
+/* Number of device channels
+ * Supported values: 2(Audio Sample Packet),
+ * 8 (HBR Audio Stream Sample Packet)
+ */
+
+ u32 sampling_rate;
+/* Device sampling rate
+* Supported values: Any
+*/
+} __packed;
+
+
+/* adsp_adm_api.h */
+
+
+/* Port ID. Update afe_get_port_index
+ * when a new port is added here. */
+#define PRIMARY_I2S_RX 0 /* index = 0 */
+#define PRIMARY_I2S_TX 1 /* index = 1 */
+#define PCM_RX 2 /* index = 2 */
+#define PCM_TX 3 /* index = 3 */
+#define SECONDARY_I2S_RX 4 /* index = 4 */
+#define SECONDARY_I2S_TX 5 /* index = 5 */
+#define MI2S_RX 6 /* index = 6 */
+#define MI2S_TX 7 /* index = 7 */
+#define HDMI_RX 8 /* index = 8 */
+#define RSVD_2 9 /* index = 9 */
+#define RSVD_3 10 /* index = 10 */
+#define DIGI_MIC_TX 11 /* index = 11 */
+#define VOICE_RECORD_RX 0x8003 /* index = 12 */
+#define VOICE_RECORD_TX 0x8004 /* index = 13 */
+#define VOICE_PLAYBACK_TX 0x8005 /* index = 14 */
+
+/* Slimbus Multi channel port id pool */
+#define SLIMBUS_0_RX 0x4000 /* index = 15 */
+#define SLIMBUS_0_TX 0x4001 /* index = 16 */
+#define SLIMBUS_1_RX 0x4002 /* index = 17 */
+#define SLIMBUS_1_TX 0x4003 /* index = 18 */
+#define SLIMBUS_2_RX 0x4004
+#define SLIMBUS_2_TX 0x4005
+#define SLIMBUS_3_RX 0x4006
+#define SLIMBUS_3_TX 0x4007
+#define SLIMBUS_4_RX 0x4008
+#define SLIMBUS_4_TX 0x4009 /* index = 24 */
+#define INT_BT_SCO_RX 0x3000 /* index = 25 */
+#define INT_BT_SCO_TX 0x3001 /* index = 26 */
+#define INT_BT_A2DP_RX 0x3002 /* index = 27 */
+#define INT_FM_RX 0x3004 /* index = 28 */
+#define INT_FM_TX 0x3005 /* index = 29 */
+#define RT_PROXY_PORT_001_RX 0x2000 /* index = 30 */
+#define RT_PROXY_PORT_001_TX 0x2001 /* index = 31 */
+
+#define AFE_PORT_INVALID 0xFFFF
+#define SLIMBUS_INVALID AFE_PORT_INVALID
+
+#define AFE_PORT_CMD_START 0x000100ca
+
+#define AFE_EVENT_RTPORT_START 0
+#define AFE_EVENT_RTPORT_STOP 1
+#define AFE_EVENT_RTPORT_LOW_WM 2
+#define AFE_EVENT_RTPORT_HI_WM 3
+
+#define ADSP_AFE_VERSION 0x00200000
+
+/* Size of the range of port IDs for the audio interface. */
+#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE 0xF
+
+/* Size of the range of port IDs for internal BT-FM ports. */
+#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE 0x6
+
+/* Size of the range of port IDs for SLIMbus<sup>®
+ * </sup> multichannel
+ * ports.
+ */
+#define AFE_PORT_ID_SLIMBUS_RANGE_SIZE 0xA
+
+/* Size of the range of port IDs for real-time proxy ports. */
+#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE 0x2
+
+/* Size of the range of port IDs for pseudoports. */
+#define AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE 0x5
+
+/* Start of the range of port IDs for the audio interface. */
+#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START 0x1000
+
+/* End of the range of port IDs for the audio interface. */
+#define AFE_PORT_ID_AUDIO_IF_PORT_RANGE_END \
+ (AFE_PORT_ID_AUDIO_IF_PORT_RANGE_START +\
+ AFE_PORT_ID_AUDIO_IF_PORT_RANGE_SIZE - 1)
+
+/* Start of the range of port IDs for real-time proxy ports. */
+#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_START 0x2000
+
+/* End of the range of port IDs for real-time proxy ports. */
+#define AFE_PORT_ID_RT_PROXY_PORT_RANGE_END \
+ (AFE_PORT_ID_RT_PROXY_PORT_RANGE_START +\
+ AFE_PORT_ID_RT_PROXY_PORT_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for internal BT-FM devices. */
+#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START 0x3000
+
+/* End of the range of port IDs for internal BT-FM devices. */
+#define AFE_PORT_ID_INTERNAL_BT_FM_RANGE_END \
+ (AFE_PORT_ID_INTERNAL_BT_FM_RANGE_START +\
+ AFE_PORT_ID_INTERNAL_BT_FM_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for SLIMbus devices. */
+#define AFE_PORT_ID_SLIMBUS_RANGE_START 0x4000
+
+/* End of the range of port IDs for SLIMbus devices. */
+#define AFE_PORT_ID_SLIMBUS_RANGE_END \
+ (AFE_PORT_ID_SLIMBUS_RANGE_START +\
+ AFE_PORT_ID_SLIMBUS_RANGE_SIZE-1)
+
+/* Start of the range of port IDs for pseudoports. */
+#define AFE_PORT_ID_PSEUDOPORT_RANGE_START 0x8001
+
+/* End of the range of port IDs for pseudoports. */
+#define AFE_PORT_ID_PSEUDOPORT_RANGE_END \
+ (AFE_PORT_ID_PSEUDOPORT_RANGE_START +\
+ AFE_PORT_ID_PSEUDOPORT_RANGE_SIZE-1)
+
+#define AFE_PORT_ID_PRIMARY_MI2S_RX 0x1000
+#define AFE_PORT_ID_PRIMARY_MI2S_TX 0x1001
+#define AFE_PORT_ID_SECONDARY_MI2S_RX 0x1002
+#define AFE_PORT_ID_SECONDARY_MI2S_TX 0x1003
+#define AFE_PORT_IDERTIARY_MI2S_RX 0x1004
+#define AFE_PORT_IDERTIARY_MI2S_TX 0x1005
+#define AFE_PORT_ID_QUATERNARY_MI2S_RX 0x1006
+#define AFE_PORT_ID_QUATERNARY_MI2S_TX 0x1007
+#define AUDIO_PORT_ID_I2S_RX 0x1008
+#define AFE_PORT_ID_DIGITAL_MIC_TX 0x1009
+#define AFE_PORT_ID_PRIMARY_PCM_RX 0x100A
+#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B
+#define AFE_PORT_ID_SECONDARY_PCM_RX 0x100C
+#define AFE_PORT_ID_SECONDARY_PCM_TX 0x100D
+#define AFE_PORT_ID_MULTICHAN_HDMI_RX 0x100E
+#define AFE_PORT_ID_RT_PROXY_PORT_001_RX 0x2000
+#define AFE_PORT_ID_RT_PROXY_PORT_001_TX 0x2001
+#define AFE_PORT_ID_INTERNAL_BT_SCO_RX 0x3000
+#define AFE_PORT_ID_INTERNAL_BT_SCO_TX 0x3001
+#define AFE_PORT_ID_INTERNAL_BT_A2DP_RX 0x3002
+#define AFE_PORT_ID_INTERNAL_FM_RX 0x3004
+#define AFE_PORT_ID_INTERNAL_FM_TX 0x3005
+/* SLIMbus Rx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000
+/* SLIMbus Tx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX 0x4001
+/* SLIMbus Rx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX 0x4002
+/* SLIMbus Tx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX 0x4003
+/* SLIMbus Rx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX 0x4004
+/* SLIMbus Tx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX 0x4005
+/* SLIMbus Rx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX 0x4006
+/* SLIMbus Tx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX 0x4007
+/* SLIMbus Rx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX 0x4008
+/* SLIMbus Tx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX 0x4009
+/* SLIMbus Rx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_RX 0x4000
+/* SLIMbus Tx port on channel 0. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_0_TX 0x4001
+/* SLIMbus Rx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_RX 0x4002
+/* SLIMbus Tx port on channel 1. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_1_TX 0x4003
+/* SLIMbus Rx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_RX 0x4004
+/* SLIMbus Tx port on channel 2. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_2_TX 0x4005
+/* SLIMbus Rx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_RX 0x4006
+/* SLIMbus Tx port on channel 3. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_3_TX 0x4007
+/* SLIMbus Rx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_RX 0x4008
+/* SLIMbus Tx port on channel 4. */
+#define AFE_PORT_ID_SLIMBUS_MULTI_CHAN_4_TX 0x4009
+/* Generic pseudoport 1. */
+#define AFE_PORT_ID_PSEUDOPORT_01 0x8001
+/* Generic pseudoport 2. */
+#define AFE_PORT_ID_PSEUDOPORT_02 0x8002
+
+/* @xreflabel{hdr:AfePortIdPrimaryAuxPcmTx}
+ Primary Aux PCM Tx port ID.
+*/
+#define AFE_PORT_ID_PRIMARY_PCM_TX 0x100B
+/* Pseudoport that corresponds to the voice Rx path.
+ * For recording, the voice Rx path samples are written to this
+ * port and consumed by the audio path.
+ */
+
+#define AFE_PORT_ID_VOICE_RECORD_RX 0x8003
+
+/* Pseudoport that corresponds to the voice Tx path.
+ * For recording, the voice Tx path samples are written to this
+ * port and consumed by the audio path.
+ */
+
+#define AFE_PORT_ID_VOICE_RECORD_TX 0x8004
+/* Pseudoport that corresponds to in-call voice delivery samples.
+ * During in-call audio delivery, the audio path delivers samples
+ * to this port from where the voice path delivers them on the
+ * Rx path.
+ */
+#define AFE_PORT_ID_VOICE_PLAYBACK_TX 0x8005
+#define AFE_PORT_ID_INVALID 0xFFFF
+
+#define AAC_ENC_MODE_AAC_LC 0x02
+#define AAC_ENC_MODE_AAC_P 0x05
+#define AAC_ENC_MODE_EAAC_P 0x1D
+
+#define AFE_PSEUDOPORT_CMD_START 0x000100cf
+struct afe_pseudoport_start_command {
+ struct apr_hdr hdr;
+ u16 port_id; /* Pseudo Port 1 = 0x8000 */
+ /* Pseudo Port 2 = 0x8001 */
+ /* Pseudo Port 3 = 0x8002 */
+ u16 timing; /* FTRT = 0 , AVTimer = 1, */
+} __packed;
+
+#define AFE_PSEUDOPORT_CMD_STOP 0x000100d0
+struct afe_pseudoport_stop_command {
+ struct apr_hdr hdr;
+ u16 port_id; /* Pseudo Port 1 = 0x8000 */
+ /* Pseudo Port 2 = 0x8001 */
+ /* Pseudo Port 3 = 0x8002 */
+ u16 reserved;
+} __packed;
+
+
+#define AFE_MODULE_SIDETONE_IIR_FILTER 0x00010202
+#define AFE_PARAM_ID_ENABLE 0x00010203
+
+/* Payload of the #AFE_PARAM_ID_ENABLE
+ * parameter, which enables or
+ * disables any module.
+ * The fixed size of this structure is four bytes.
+ */
+
+struct afe_mod_enable_param {
+ u16 enable;
+ /* Enables (1) or disables (0) the module. */
+
+ u16 reserved;
+ /* This field must be set to zero.
+ */
+} __packed;
+
+/* ID of the configuration parameter used by the
+ * #AFE_MODULE_SIDETONE_IIR_FILTER module.
+ */
+#define AFE_PARAM_ID_SIDETONE_IIR_FILTER_CONFIG 0x00010204
+
+struct afe_sidetone_iir_filter_config_params {
+ u16 num_biquad_stages;
+/* Number of stages.
+ * Supported values: Minimum of 5 and maximum of 10
+ */
+
+ u16 pregain;
+/* Pregain for the compensating filter response.
+ * Supported values: Any number in Q13 format
+ */
+} __packed;
+
+#define AFE_MODULE_LOOPBACK 0x00010205
+#define AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH 0x00010206
+
+/* Payload of the #AFE_PARAM_ID_LOOPBACK_GAIN_PER_PATH parameter,
+ * which gets/sets loopback gain of a port to an Rx port.
+ * The Tx port ID of the loopback is part of the set_param command.
+ */
+
+/* Payload of the #AFE_PORT_CMD_SET_PARAM_V2 command's
+ * configuration/calibration settings for the AFE port.
+ */
+struct afe_port_cmd_set_param_v2 {
+ u16 port_id;
+/* Port interface and direction (Rx or Tx) to start.
+ */
+
+ u16 payload_size;
+/* Actual size of the payload in bytes.
+ * This is used for parsing the parameter payload.
+ * Supported values: > 0
+ */
+
+u32 payload_address_lsw;
+/* LSW of 64 bit Payload address.
+ * Address should be 32-byte,
+ * 4kbyte aligned and must be contiguous memory.
+ */
+
+u32 payload_address_msw;
+/* MSW of 64 bit Payload address.
+ * In case of 32-bit shared memory address,
+ * this field must be set to zero.
+ * In case of 36-bit shared memory address,
+ * bit-4 to bit-31 must be set to zero.
+ * Address should be 32-byte, 4kbyte aligned
+ * and must be contiguous memory.
+ */
+
+u32 mem_map_handle;
+/* Memory map handle returned by
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands.
+ * Supported Values:
+ * - NULL -- Message. The parameter data is in-band.
+ * - Non-NULL -- The parameter data is Out-band.Pointer to
+ * the physical address
+ * in shared memory of the payload data.
+ * An optional field is available if parameter
+ * data is in-band:
+ * afe_param_data_v2 param_data[...].
+ * For detailed payload content, see the
+ * afe_port_param_data_v2 structure.
+ */
+} __packed;
+
+#define AFE_PORT_CMD_SET_PARAM_V2 0x000100EF
+
+struct afe_port_param_data_v2 {
+ u32 module_id;
+/* ID of the module to be configured.
+ * Supported values: Valid module ID
+ */
+
+u32 param_id;
+/* ID of the parameter corresponding to the supported parameters
+ * for the module ID.
+ * Supported values: Valid parameter ID
+ */
+
+u16 param_size;
+/* Actual size of the data for the
+ * module_id/param_id pair. The size is a
+ * multiple of four bytes.
+ * Supported values: > 0
+ */
+
+u16 reserved;
+/* This field must be set to zero.
+ */
+} __packed;
+
+struct afe_loopback_gain_per_path_param {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ u16 rx_port_id;
+/* Rx port of the loopback. */
+
+u16 gain;
+/* Loopback gain per path of the port.
+ * Supported values: Any number in Q13 format
+ */
+} __packed;
+
+/* Parameter ID used to configure and enable/disable the
+ * loopback path. The difference with respect to the existing
+ * API, AFE_PORT_CMD_LOOPBACK, is that it allows Rx port to be
+ * configured as source port in loopback path. Port-id in
+ * AFE_PORT_CMD_SET_PARAM cmd is the source port whcih can be
+ * Tx or Rx port. In addition, we can configure the type of
+ * routing mode to handle different use cases.
+ */
+#define AFE_PARAM_ID_LOOPBACK_CONFIG 0x0001020B
+#define AFE_API_VERSION_LOOPBACK_CONFIG 0x1
+
+enum afe_loopback_routing_mode {
+ LB_MODE_DEFAULT = 1,
+ /* Regular loopback from source to destination port */
+ LB_MODE_SIDETONE,
+ /* Sidetone feed from Tx source to Rx destination port */
+ LB_MODE_EC_REF_VOICE_AUDIO,
+ /* Echo canceller reference, voice + audio + DTMF */
+ LB_MODE_EC_REF_VOICE
+ /* Echo canceller reference, voice alone */
+} __packed;
+
+/* Payload of the #AFE_PARAM_ID_LOOPBACK_CONFIG ,
+ * which enables/disables one AFE loopback.
+ */
+struct afe_loopback_cfg_v1 {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ u32 loopback_cfg_minor_version;
+/* Minor version used for tracking the version of the RMC module
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_LOOPBACK_CONFIG
+ */
+ u16 dst_port_id;
+ /* Destination Port Id. */
+ u16 routing_mode;
+/* Specifies data path type from src to dest port.
+ * Supported values:
+ * #LB_MODE_DEFAULT
+ * #LB_MODE_SIDETONE
+ * #LB_MODE_EC_REF_VOICE_AUDIO
+ * #LB_MODE_EC_REF_VOICE_A
+ * #LB_MODE_EC_REF_VOICE
+ */
+
+ u16 enable;
+/* Specifies whether to enable (1) or
+ * disable (0) an AFE loopback.
+ */
+ u16 reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.
+ */
+
+} __packed;
+
+#define AFE_MODULE_SPEAKER_PROTECTION 0x00010209
+#define AFE_PARAM_ID_SPKR_PROT_CONFIG 0x0001020a
+#define AFE_API_VERSION_SPKR_PROT_CONFIG 0x1
+#define AFE_SPKR_PROT_EXCURSIONF_LEN 512
+struct afe_spkr_prot_cfg_param_v1 {
+ u32 spkr_prot_minor_version;
+/*
+ * Minor version used for tracking the version of the
+ * speaker protection module configuration interface.
+ * Supported values: #AFE_API_VERSION_SPKR_PROT_CONFIG
+ */
+
+int16_t win_size;
+/* Analysis and synthesis window size (nWinSize).
+ * Supported values: 1024, 512, 256 samples
+ */
+
+int16_t margin;
+/* Allowable margin for excursion prediction,
+ * in L16Q15 format. This is a
+ * control parameter to allow
+ * for overestimation of peak excursion.
+ */
+
+int16_t spkr_exc_limit;
+/* Speaker excursion limit, in L16Q15 format.*/
+
+int16_t spkr_resonance_freq;
+/* Resonance frequency of the speaker; used
+ * to define a frequency range
+ * for signal modification.
+ *
+ * Supported values: 0 to 2000 Hz */
+
+int16_t limhresh;
+/* Threshold of the hard limiter; used to
+ * prevent overshooting beyond a
+ * signal level that was set by the limiter
+ * prior to speaker protection.
+ * Supported values: 0 to 32767
+ */
+
+int16_t hpf_cut_off_freq;
+/* High pass filter cutoff frequency.
+ * Supported values: 100, 200, 300 Hz
+ */
+
+int16_t hpf_enable;
+/* Specifies whether the high pass filter
+ * is enabled (0) or disabled (1).
+ */
+
+int16_t reserved;
+/* This field must be set to zero. */
+
+int32_t amp_gain;
+/* Amplifier gain in L32Q15 format.
+ * This is the RMS voltage at the
+ * loudspeaker when a 0dBFS tone
+ * is played in the digital domain.
+ */
+
+int16_t excursionf[AFE_SPKR_PROT_EXCURSIONF_LEN];
+/* Array of the excursion transfer function.
+ * The peak excursion of the
+ * loudspeaker diaphragm is
+ * measured in millimeters for 1 Vrms Sine
+ * tone at all FFT bin frequencies.
+ * Supported values: Q15 format
+ */
+} __packed;
+
+
+#define AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER 0x000100E0
+
+/* Payload of the #AFE_SERVICE_CMD_REGISTER_RT_PORT_DRIVER
+ * command, which registers a real-time port driver
+ * with the AFE service.
+ */
+struct afe_service_cmd_register_rt_port_driver {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Port ID with which the real-time driver exchanges data
+ * (registers for events).
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+ u16 reserved;
+ /* This field must be set to zero. */
+} __packed;
+
+#define AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER 0x000100E1
+
+/* Payload of the #AFE_SERVICE_CMD_UNREGISTER_RT_PORT_DRIVER
+ * command, which unregisters a real-time port driver from
+ * the AFE service.
+ */
+struct afe_service_cmd_unregister_rt_port_driver {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Port ID from which the real-time
+ * driver unregisters for events.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+ u16 reserved;
+ /* This field must be set to zero. */
+} __packed;
+
+#define AFE_EVENT_RT_PROXY_PORT_STATUS 0x00010105
+#define AFE_EVENTYPE_RT_PROXY_PORT_START 0
+#define AFE_EVENTYPE_RT_PROXY_PORT_STOP 1
+#define AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK 2
+#define AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK 3
+#define AFE_EVENTYPE_RT_PROXY_PORT_INVALID 0xFFFF
+
+/* Payload of the #AFE_EVENT_RT_PROXY_PORT_STATUS
+ * message, which sends an event from the AFE service
+ * to a registered client.
+ */
+struct afe_event_rt_proxy_port_status {
+ u16 port_id;
+/* Port ID to which the event is sent.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+ u16 eventype;
+/* Type of event.
+ * Supported values:
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_START
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_STOP
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_LOW_WATER_MARK
+ * - #AFE_EVENTYPE_RT_PROXY_PORT_HIGH_WATER_MARK
+ */
+} __packed;
+
+#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_WRITE_V2 0x000100ED
+
+struct afe_port_data_cmd_rt_proxy_port_write_v2 {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Tx (mic) proxy port ID with which the real-time
+ * driver exchanges data.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ */
+
+ u16 reserved;
+ /* This field must be set to zero. */
+
+ u32 buffer_address_lsw;
+/* LSW Address of the buffer containing the
+ * data from the real-time source
+ * device on a client.
+ */
+
+ u32 buffer_address_msw;
+/* MSW Address of the buffer containing the
+ * data from the real-time source
+ * device on a client.
+ */
+
+ u32 mem_map_handle;
+/* A memory map handle encapsulating shared memory
+ * attributes is returned if
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS
+ * command is successful.
+ * Supported Values:
+ * - Any 32 bit value
+ */
+
+ u32 available_bytes;
+/* Number of valid bytes available
+ * in the buffer (including all
+ * channels: number of bytes per
+ * channel = availableBytesumChannels).
+ * Supported values: > 0
+ *
+ * This field must be equal to the frame
+ * size specified in the #AFE_PORT_AUDIO_IF_CONFIG
+ * command that was sent to configure this
+ * port.
+ */
+} __packed;
+
+#define AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 0x000100EE
+
+/* Payload of the
+ * #AFE_PORT_DATA_CMD_RT_PROXY_PORT_READ_V2 command, which
+ * delivers an empty buffer to the AFE service. On
+ * acknowledgment, data is filled in the buffer.
+ */
+struct afe_port_data_cmd_rt_proxy_port_read_v2 {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Rx proxy port ID with which the real-time
+ * driver exchanges data.
+ * Supported values: #AFE_PORT_ID_RT_PROXY_PORT_RANGE_START to
+ * #AFE_PORT_ID_RT_PROXY_PORT_RANGE_END
+ * (This must be an Rx (speaker) port.)
+ */
+
+ u16 reserved;
+ /* This field must be set to zero. */
+
+ u32 buffer_address_lsw;
+/* LSW Address of the buffer containing the data sent from the AFE
+ * service to a real-time sink device on the client.
+ */
+
+
+ u32 buffer_address_msw;
+/* MSW Address of the buffer containing the data sent from the AFE
+ * service to a real-time sink device on the client.
+ */
+
+ u32 mem_map_handle;
+/* A memory map handle encapsulating shared memory attributes is
+ * returned if AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is
+ * successful.
+ * Supported Values:
+ * - Any 32 bit value
+ */
+
+ u32 available_bytes;
+/* Number of valid bytes available in the buffer (including all
+ * channels).
+ * Supported values: > 0
+ * This field must be equal to the frame size specified in the
+ * #AFE_PORT_AUDIO_IF_CONFIG command that was sent to configure
+ * this port.
+ */
+} __packed;
+
+/* This module ID is related to device configuring like I2S,PCM,
+ * HDMI, SLIMBus etc. This module supports follwing parameter ids.
+ * - #AFE_PARAM_ID_I2S_CONFIG
+ * - #AFE_PARAM_ID_PCM_CONFIG
+ * - #AFE_PARAM_ID_DIGI_MIC_CONFIG
+ * - #AFE_PARAM_ID_HDMI_CONFIG
+ * - #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
+ * - #AFE_PARAM_ID_SLIMBUS_CONFIG
+ * - #AFE_PARAM_ID_RT_PROXY_CONFIG
+ */
+
+#define AFE_MODULE_AUDIO_DEV_INTERFACE 0x0001020C
+#define AFE_PORT_SAMPLE_RATE_8K 8000
+#define AFE_PORT_SAMPLE_RATE_16K 16000
+#define AFE_PORT_SAMPLE_RATE_48K 48000
+#define AFE_PORT_SAMPLE_RATE_96K 96000
+#define AFE_PORT_SAMPLE_RATE_192K 192000
+#define AFE_LINEAR_PCM_DATA 0x0
+#define AFE_NON_LINEAR_DATA 0x1
+#define AFE_LINEAR_PCM_DATA_PACKED_60958 0x2
+#define AFE_NON_LINEAR_DATA_PACKED_60958 0x3
+
+/* This param id is used to configure I2S interface */
+#define AFE_PARAM_ID_I2S_CONFIG 0x0001020D
+#define AFE_API_VERSION_I2S_CONFIG 0x1
+/* Enumeration for setting the I2S configuration
+ * channel_mode parameter to
+ * serial data wire number 1-3 (SD3).
+ */
+#define AFE_PORT_I2S_SD0 0x1
+#define AFE_PORT_I2S_SD1 0x2
+#define AFE_PORT_I2S_SD2 0x3
+#define AFE_PORT_I2S_SD3 0x4
+#define AFE_PORT_I2S_QUAD01 0x5
+#define AFE_PORT_I2S_QUAD23 0x6
+#define AFE_PORT_I2S_6CHS 0x7
+#define AFE_PORT_I2S_8CHS 0x8
+#define AFE_PORT_I2S_MONO 0x0
+#define AFE_PORT_I2S_STEREO 0x1
+#define AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL 0x0
+#define AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL 0x1
+
+/* Payload of the #AFE_PARAM_ID_I2S_CONFIG
+ * command's (I2S configuration
+ * parameter).
+ */
+struct afe_param_id_i2s_cfg {
+ u32 i2s_cfg_minor_version;
+/* Minor version used for tracking the version of the I2S
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_I2S_CONFIG
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+
+ u16 channel_mode;
+/* I2S lines and multichannel operation.
+ * Supported values:
+ * - #AFE_PORT_I2S_SD0
+ * - #AFE_PORT_I2S_SD1
+ * - #AFE_PORT_I2S_SD2
+ * - #AFE_PORT_I2S_SD3
+ * - #AFE_PORT_I2S_QUAD01
+ * - #AFE_PORT_I2S_QUAD23
+ * - #AFE_PORT_I2S_6CHS
+ * - #AFE_PORT_I2S_8CHS
+ */
+
+ u16 mono_stereo;
+/* Specifies mono or stereo. This applies only when
+ * a single I2S line is used.
+ * Supported values:
+ * - #AFE_PORT_I2S_MONO
+ * - #AFE_PORT_I2S_STEREO
+ */
+
+ u16 ws_src;
+/* Word select source: internal or external.
+ * Supported values:
+ * - #AFE_PORT_CONFIG_I2S_WS_SRC_EXTERNAL
+ * - #AFE_PORT_CONFIG_I2S_WS_SRC_INTERNAL
+ */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_192K
+ */
+
+ u16 data_format;
+/* data format
+ * Supported values:
+ * - #LINEAR_PCM_DATA
+ * - #NON_LINEAR_DATA
+ * - #LINEAR_PCM_DATA_PACKED_IN_60958
+ * - #NON_LINEAR_DATA_PACKED_IN_60958
+ */
+ u16 reserved;
+ /* This field must be set to zero. */
+} __packed;
+
+/*
+ * This param id is used to configure PCM interface
+ */
+#define AFE_PARAM_ID_PCM_CONFIG 0x0001020E
+#define AFE_API_VERSION_PCM_CONFIG 0x1
+/* Enumeration for the auxiliary PCM synchronization signal
+ * provided by an external source.
+ */
+
+#define AFE_PORT_PCM_SYNC_SRC_EXTERNAL 0x0
+/* Enumeration for the auxiliary PCM synchronization signal
+ * provided by an internal source.
+ */
+#define AFE_PORT_PCM_SYNC_SRC_INTERNAL 0x1
+/* Enumeration for the PCM configuration aux_mode parameter,
+ * which configures the auxiliary PCM interface to use
+ * short synchronization.
+ */
+#define AFE_PORT_PCM_AUX_MODE_PCM 0x0
+/*
+ * Enumeration for the PCM configuration aux_mode parameter,
+ * which configures the auxiliary PCM interface to use long
+ * synchronization.
+ */
+#define AFE_PORT_PCM_AUX_MODE_AUX 0x1
+/*
+ * Enumeration for setting the PCM configuration frame to 8.
+ */
+#define AFE_PORT_PCM_BITS_PER_FRAME_8 0x0
+/*
+ * Enumeration for setting the PCM configuration frame to 16.
+ */
+#define AFE_PORT_PCM_BITS_PER_FRAME_16 0x1
+
+/* Enumeration for setting the PCM configuration frame to 32.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_32 0x2
+
+/* Enumeration for setting the PCM configuration frame to 64.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_64 0x3
+
+/* Enumeration for setting the PCM configuration frame to 128.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_128 0x4
+
+/* Enumeration for setting the PCM configuration frame to 256.*/
+#define AFE_PORT_PCM_BITS_PER_FRAME_256 0x5
+
+/* Enumeration for setting the PCM configuration
+ * quantype parameter to A-law with no padding.
+ */
+#define AFE_PORT_PCM_ALAW_NOPADDING 0x0
+
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to mu-law with no padding.
+ */
+#define AFE_PORT_PCM_MULAW_NOPADDING 0x1
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to linear with no padding.
+ */
+#define AFE_PORT_PCM_LINEAR_NOPADDING 0x2
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to A-law with padding.
+ */
+#define AFE_PORT_PCM_ALAW_PADDING 0x3
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to mu-law with padding.
+ */
+#define AFE_PORT_PCM_MULAW_PADDING 0x4
+/* Enumeration for setting the PCM configuration quantype
+ * parameter to linear with padding.
+ */
+#define AFE_PORT_PCM_LINEAR_PADDING 0x5
+/* Enumeration for disabling the PCM configuration
+ * ctrl_data_out_enable parameter.
+ * The PCM block is the only master.
+ */
+#define AFE_PORT_PCM_CTRL_DATA_OE_DISABLE 0x0
+/*
+ * Enumeration for enabling the PCM configuration
+ * ctrl_data_out_enable parameter. The PCM block shares
+ * the signal with other masters.
+ */
+#define AFE_PORT_PCM_CTRL_DATA_OE_ENABLE 0x1
+
+/* Payload of the #AFE_PARAM_ID_PCM_CONFIG command's
+ * (PCM configuration parameter).
+ */
+
+struct afe_param_id_pcm_cfg {
+ u32 pcm_cfg_minor_version;
+/* Minor version used for tracking the version of the AUX PCM
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_PCM_CONFIG
+ */
+
+ u16 aux_mode;
+/* PCM synchronization setting.
+ * Supported values:
+ * - #AFE_PORT_PCM_AUX_MODE_PCM
+ * - #AFE_PORT_PCM_AUX_MODE_AUX
+ */
+
+ u16 sync_src;
+/* Synchronization source.
+ * Supported values:
+ * - #AFE_PORT_PCM_SYNC_SRC_EXTERNAL
+ * - #AFE_PORT_PCM_SYNC_SRC_INTERNAL
+ */
+
+ u16 frame_setting;
+/* Number of bits per frame.
+ * Supported values:
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_8
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_16
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_32
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_64
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_128
+ * - #AFE_PORT_PCM_BITS_PER_FRAME_256
+ */
+
+ u16 quantype;
+/* PCM quantization type.
+ * Supported values:
+ * - #AFE_PORT_PCM_ALAW_NOPADDING
+ * - #AFE_PORT_PCM_MULAW_NOPADDING
+ * - #AFE_PORT_PCM_LINEAR_NOPADDING
+ * - #AFE_PORT_PCM_ALAW_PADDING
+ * - #AFE_PORT_PCM_MULAW_PADDING
+ * - #AFE_PORT_PCM_LINEAR_PADDING
+ */
+
+ u16 ctrl_data_out_enable;
+/* Specifies whether the PCM block shares the data-out
+ * signal to the drive with other masters.
+ * Supported values:
+ * - #AFE_PORT_PCM_CTRL_DATA_OE_DISABLE
+ * - #AFE_PORT_PCM_CTRL_DATA_OE_ENABLE
+ */
+ u16 reserved;
+ /* This field must be set to zero. */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+ u16 num_channels;
+/* Number of channels.
+ * Supported values: 1 to 4
+ */
+
+ u16 slot_number_mapping[4];
+/* Specifies the slot number for the each channel in
+ * multi channel scenario.
+ * Supported values: 1 to 32
+ */
+} __packed;
+
+/*
+ * This param id is used to configure DIGI MIC interface
+ */
+#define AFE_PARAM_ID_DIGI_MIC_CONFIG 0x0001020F
+/* This version information is used to handle the new
+ * additions to the config interface in future in backward
+ * compatible manner.
+ */
+#define AFE_API_VERSION_DIGI_MIC_CONFIG 0x1
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to left 0.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_LEFT0 0x1
+
+/*Enumeration for setting the digital mic configuration
+ * channel_mode parameter to right 0.
+ */
+
+
+#define AFE_PORT_DIGI_MIC_MODE_RIGHT0 0x2
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to left 1.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_LEFT1 0x3
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to right 1.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_RIGHT1 0x4
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to stereo 0.
+ */
+#define AFE_PORT_DIGI_MIC_MODE_STEREO0 0x5
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to stereo 1.
+ */
+
+
+#define AFE_PORT_DIGI_MIC_MODE_STEREO1 0x6
+
+/* Enumeration for setting the digital mic configuration
+ * channel_mode parameter to quad.
+ */
+
+#define AFE_PORT_DIGI_MIC_MODE_QUAD 0x7
+
+/* Payload of the #AFE_PARAM_ID_DIGI_MIC_CONFIG command's
+ * (DIGI MIC configuration
+ * parameter).
+ */
+struct afe_param_id_digi_mic_cfg {
+ u32 digi_mic_cfg_minor_version;
+/* Minor version used for tracking the version of the DIGI Mic
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_DIGI_MIC_CONFIG
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+ u16 channel_mode;
+/* Digital mic and multichannel operation.
+ * Supported values:
+ * - #AFE_PORT_DIGI_MIC_MODE_LEFT0
+ * - #AFE_PORT_DIGI_MIC_MODE_RIGHT0
+ * - #AFE_PORT_DIGI_MIC_MODE_LEFT1
+ * - #AFE_PORT_DIGI_MIC_MODE_RIGHT1
+ * - #AFE_PORT_DIGI_MIC_MODE_STEREO0
+ * - #AFE_PORT_DIGI_MIC_MODE_STEREO1
+ * - #AFE_PORT_DIGI_MIC_MODE_QUAD
+ */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ */
+} __packed;
+
+/*
+* This param id is used to configure HDMI interface
+*/
+#define AFE_PARAM_ID_HDMI_CONFIG 0x00010210
+
+/* This version information is used to handle the new
+* additions to the config interface in future in backward
+* compatible manner.
+*/
+#define AFE_API_VERSION_HDMI_CONFIG 0x1
+
+/* Payload of the #AFE_PARAM_ID_HDMI_CONFIG command,
+ * which configures a multichannel HDMI audio interface.
+ */
+struct afe_param_id_hdmi_multi_chan_audio_cfg {
+ u32 hdmi_cfg_minor_version;
+/* Minor version used for tracking the version of the HDMI
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_HDMI_CONFIG
+ */
+
+u16 dataype;
+/* data type
+ * Supported values:
+ * - #LINEAR_PCM_DATA
+ * - #NON_LINEAR_DATA
+ * - #LINEAR_PCM_DATA_PACKED_IN_60958
+ * - #NON_LINEAR_DATA_PACKED_IN_60958
+ */
+
+u16 channel_allocation;
+/* HDMI channel allocation information for programming an HDMI
+ * frame. The default is 0 (Stereo).
+ *
+ * This information is defined in the HDMI standard, CEA 861-D
+ * (refer to @xhyperref{S1,[S1]}). The number of channels is also
+ * inferred from this parameter.
+*/
+
+
+u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - 22050, 44100, 176400 for compressed streams
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+ u16 reserved;
+ /* This field must be set to zero. */
+} __packed;
+
+/*
+* This param id is used to configure BT or FM(RIVA) interface
+*/
+#define AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG 0x00010211
+
+/* This version information is used to handle the new
+* additions to the config interface in future in backward
+* compatible manner.
+*/
+#define AFE_API_VERSION_INTERNAL_BT_FM_CONFIG 0x1
+
+/* Payload of the #AFE_PARAM_ID_INTERNAL_BT_FM_CONFIG
+ * command's BT voice/BT audio/FM configuration parameter.
+ */
+struct afe_param_id_internal_bt_fm_cfg {
+ u32 bt_fm_cfg_minor_version;
+/* Minor version used for tracking the version of the BT and FM
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_INTERNAL_BT_FM_CONFIG
+ */
+
+ u16 num_channels;
+/* Number of channels.
+ * Supported values: 1 to 2
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K (only for BTSCO)
+ * - #AFE_PORT_SAMPLE_RATE_16K (only for BTSCO)
+ * - #AFE_PORT_SAMPLE_RATE_48K (FM and A2DP)
+ */
+} __packed;
+
+/* This param id is used to configure SLIMBUS interface using
+ * shared channel approach.
+ */
+
+
+#define AFE_PARAM_ID_SLIMBUS_CONFIG 0x00010212
+
+/* This version information is used to handle the new
+* additions to the config interface in future in backward
+* compatible manner.
+*/
+#define AFE_API_VERSION_SLIMBUS_CONFIG 0x1
+
+/* Enumeration for setting SLIMbus device ID 1.
+*/
+#define AFE_SLIMBUS_DEVICE_1 0x0
+
+/* Enumeration for setting SLIMbus device ID 2.
+*/
+#define AFE_SLIMBUS_DEVICE_2 0x1
+
+/* Enumeration for setting the SLIMbus data formats.
+*/
+#define AFE_SB_DATA_FORMAT_NOT_INDICATED 0x0
+
+/* Enumeration for setting the maximum number of streams per
+ * device.
+ */
+
+#define AFE_PORT_MAX_AUDIO_CHAN_CNT 0x8
+
+/* Payload of the #AFE_PORT_CMD_SLIMBUS_CONFIG command's SLIMbus
+ * port configuration parameter.
+ */
+
+struct afe_param_id_slimbus_cfg {
+ u32 sb_cfg_minor_version;
+/* Minor version used for tracking the version of the SLIMBUS
+ * configuration interface.
+ * Supported values: #AFE_API_VERSION_SLIMBUS_CONFIG
+ */
+
+ u16 slimbus_dev_id;
+/* SLIMbus hardware device ID, which is required to handle
+ * multiple SLIMbus hardware blocks.
+ * Supported values: - #AFE_SLIMBUS_DEVICE_1 - #AFE_SLIMBUS_DEVICE_2
+ */
+
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16, 24
+ */
+
+ u16 data_format;
+/* Data format supported by the SLIMbus hardware. The default is
+ * 0 (#AFE_SB_DATA_FORMAT_NOT_INDICATED), which indicates the
+ * hardware does not perform any format conversions before the data
+ * transfer.
+ */
+
+
+ u16 num_channels;
+/* Number of channels.
+ * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+ */
+
+ u8 shared_ch_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+/* Mapping of shared channel IDs (128 to 255) to which the
+ * master port is to be connected.
+ * Shared_channel_mapping[i] represents the shared channel assigned
+ * for audio channel i in multichannel audio data.
+ */
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ * - #AFE_PORT_SAMPLE_RATE_96K
+ * - #AFE_PORT_SAMPLE_RATE_192K
+ */
+} __packed;
+
+/*
+* This param id is used to configure Real Time Proxy interface.
+*/
+#define AFE_PARAM_ID_RT_PROXY_CONFIG 0x00010213
+
+/* This version information is used to handle the new
+* additions to the config interface in future in backward
+* compatible manner.
+*/
+#define AFE_API_VERSION_RT_PROXY_CONFIG 0x1
+
+/* Payload of the #AFE_PARAM_ID_RT_PROXY_CONFIG
+ * command (real-time proxy port configuration parameter).
+ */
+struct afe_param_id_rt_proxy_port_cfg {
+ u32 rt_proxy_cfg_minor_version;
+/* Minor version used for tracking the version of rt-proxy
+ * config interface.
+ */
+
+ u16 bit_width;
+/* Bit width of the sample.
+ * Supported values: 16
+ */
+
+ u16 interleaved;
+/* Specifies whether the data exchanged between the AFE
+ * interface and real-time port is interleaved.
+ * Supported values: - 0 -- Non-interleaved (samples from each
+ * channel are contiguous in the buffer) - 1 -- Interleaved
+ * (corresponding samples from each input channel are interleaved
+ * within the buffer)
+ */
+
+
+ u16 frame_size;
+ /* Size of the frames that are used for PCM exchanges with this
+ * port.
+ * Supported values: > 0, in bytes
+ * For example, 5 ms buffers of 16 bits and 16 kHz stereo samples
+ * is 5 ms * 16 samples/ms * 2 bytes/sample * 2 channels = 320
+ * bytes.
+ */
+ u16 jitter_allowance;
+/* Configures the amount of jitter that the port will allow.
+ * Supported values: > 0
+ * For example, if +/-10 ms of jitter is anticipated in the timing
+ * of sending frames to the port, and the configuration is 16 kHz
+ * mono with 16-bit samples, this field is 10 ms * 16 samples/ms * 2
+ * bytes/sample = 320.
+ */
+
+ u16 low_water_mark;
+/* Low watermark in bytes (including all channels).
+ * Supported values:
+ * - 0 -- Do not send any low watermark events
+ * - > 0 -- Low watermark for triggering an event
+ * If the number of bytes in an internal circular buffer is lower
+ * than this low_water_mark parameter, a LOW_WATER_MARK event is
+ * sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
+ * event).
+ * Use of watermark events is optional for debugging purposes.
+ */
+
+ u16 high_water_mark;
+/* High watermark in bytes (including all channels).
+ * Supported values:
+ * - 0 -- Do not send any high watermark events
+ * - > 0 -- High watermark for triggering an event
+ * If the number of bytes in an internal circular buffer exceeds
+ * TOTAL_CIRC_BUF_SIZE minus high_water_mark, a high watermark event
+ * is sent to applications (via the #AFE_EVENT_RT_PROXY_PORT_STATUS
+ * event).
+ * The use of watermark events is optional and for debugging
+ * purposes.
+ */
+
+
+ u32 sample_rate;
+/* Sampling rate of the port.
+ * Supported values:
+ * - #AFE_PORT_SAMPLE_RATE_8K
+ * - #AFE_PORT_SAMPLE_RATE_16K
+ * - #AFE_PORT_SAMPLE_RATE_48K
+ */
+
+ u16 num_channels;
+/* Number of channels.
+ * Supported values: 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+ */
+
+ u16 reserved;
+ /* For 32 bit alignment. */
+} __packed;
+
+union afe_port_config {
+ struct afe_param_id_pcm_cfg pcm;
+ struct afe_param_id_i2s_cfg i2s;
+ struct afe_param_id_hdmi_multi_chan_audio_cfg hdmi_multi_ch;
+ struct afe_param_id_slimbus_cfg slim_sch;
+ struct afe_param_id_rt_proxy_port_cfg rtproxy;
+} __packed;
+
+struct afe_audioif_config_command {
+ struct apr_hdr hdr;
+ struct afe_port_cmd_set_param_v2 param;
+ struct afe_port_param_data_v2 pdata;
+ union afe_port_config port;
+} __packed;
+
+#define AFE_PORT_CMD_DEVICE_START 0x000100E5
+
+/* Payload of the #AFE_PORT_CMD_DEVICE_START.*/
+struct afe_port_cmd_device_start {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Port interface and direction (Rx or Tx) to start. An even
+ * number represents the Rx direction, and an odd number represents
+ * the Tx direction.
+ */
+
+
+ u16 reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.*/
+
+} __packed;
+
+#define AFE_PORT_CMD_DEVICE_STOP 0x000100E6
+
+/* Payload of the #AFE_PORT_CMD_DEVICE_STOP.
+*/
+struct afe_port_cmd_device_stop {
+ struct apr_hdr hdr;
+ u16 port_id;
+/* Port interface and direction (Rx or Tx) to start. An even
+ * number represents the Rx direction, and an odd number represents
+ * the Tx direction.
+ */
+
+ u16 reserved;
+/* Reserved for 32-bit alignment. This field must be set to 0.*/
+} __packed;
+
+#define AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS 0x000100EA
+
+/* Memory map regions command payload used by the
+ * #AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS .
+ * This structure allows clients to map multiple shared memory
+ * regions in a single command. Following this structure are
+ * num_regions of afe_service_shared_map_region_payload.
+ */
+struct afe_service_cmd_shared_mem_map_regions {
+ struct apr_hdr hdr;
+u16 mem_pool_id;
+/* Type of memory on which this memory region is mapped.
+ * Supported values:
+ * - #ADSP_MEMORY_MAP_EBI_POOL
+ * - #ADSP_MEMORY_MAP_SMI_POOL
+ * - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL
+ * - Other values are reserved
+ *
+ * The memory pool ID implicitly defines the characteristics of the
+ * memory. Characteristics may include alignment type, permissions,
+ * etc.
+ *
+ * ADSP_MEMORY_MAP_EBI_POOL is External Buffer Interface type memory
+ * ADSP_MEMORY_MAP_SMI_POOL is Shared Memory Interface type memory
+ * ADSP_MEMORY_MAP_SHMEM8_4K_POOL is shared memory, byte
+ * addressable, and 4 KB aligned.
+ */
+
+
+ u16 num_regions;
+/* Number of regions to map.
+ * Supported values:
+ * - Any value greater than zero
+ */
+
+ u32 property_flag;
+/* Configures one common property for all the regions in the
+ * payload.
+ *
+ * Supported values: - 0x00000000 to 0x00000001
+ *
+ * b0 - bit 0 indicates physical or virtual mapping 0 Shared memory
+ * address provided in afe_service_shared_map_region_payloadis a
+ * physical address. The shared memory needs to be mapped( hardware
+ * TLB entry) and a software entry needs to be added for internal
+ * book keeping.
+ *
+ * 1 Shared memory address provided in
+ * afe_service_shared_map_region_payloadis a virtual address. The
+ * shared memory must not be mapped (since hardware TLB entry is
+ * already available) but a software entry needs to be added for
+ * internal book keeping. This can be useful if two services with in
+ * ADSP is communicating via APR. They can now directly communicate
+ * via the Virtual address instead of Physical address. The virtual
+ * regions must be contiguous. num_regions must be 1 in this case.
+ *
+ * b31-b1 - reserved bits. must be set to zero
+ */
+
+
+} __packed;
+/* Map region payload used by the
+ * afe_service_shared_map_region_payloadstructure.
+ */
+struct afe_service_shared_map_region_payload {
+ u32 shm_addr_lsw;
+/* least significant word of starting address in the memory
+ * region to map. It must be contiguous memory, and it must be 4 KB
+ * aligned.
+ * Supported values: - Any 32 bit value
+ */
+
+
+ u32 shm_addr_msw;
+/* most significant word of startng address in the memory region
+ * to map. For 32 bit shared memory address, this field must be set
+ * to zero. For 36 bit shared memory address, bit31 to bit 4 must be
+ * set to zero
+ *
+ * Supported values: - For 32 bit shared memory address, this field
+ * must be set to zero. - For 36 bit shared memory address, bit31 to
+ * bit 4 must be set to zero - For 64 bit shared memory address, any
+ * 32 bit value
+ */
+
+
+ u32 mem_size_bytes;
+/* Number of bytes in the region. The aDSP will always map the
+ * regions as virtual contiguous memory, but the memory size must be
+ * in multiples of 4 KB to avoid gaps in the virtually contiguous
+ * mapped memory.
+ *
+ * Supported values: - multiples of 4KB
+ */
+
+} __packed;
+
+#define AFE_SERVICE_CMDRSP_SHARED_MEM_MAP_REGIONS 0x000100EB
+struct afe_service_cmdrsp_shared_mem_map_regions {
+ u32 mem_map_handle;
+/* A memory map handle encapsulating shared memory attributes is
+ * returned iff AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS command is
+ * successful. In the case of failure , a generic APR error response
+ * is returned to the client.
+ *
+ * Supported Values: - Any 32 bit value
+ */
+
+} __packed;
+#define AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS 0x000100EC
+/* Memory unmap regions command payload used by the
+ * #AFE_SERVICE_CMD_SHARED_MEM_UNMAP_REGIONS
+ *
+ * This structure allows clients to unmap multiple shared memory
+ * regions in a single command.
+ */
+
+
+struct afe_service_cmd_shared_mem_unmap_regions {
+ struct apr_hdr hdr;
+u32 mem_map_handle;
+/* memory map handle returned by
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands
+ *
+ * Supported Values:
+ * - Any 32 bit value
+ */
+} __packed;
+
+#define AFE_PORT_CMD_GET_PARAM_V2 0x000100F0
+
+/* Payload of the #AFE_PORT_CMD_GET_PARAM_V2 command,
+ * which queries for one post/preprocessing parameter of a
+ * stream.
+ */
+struct afe_port_cmd_get_param_v2 {
+
+ struct apr_hdr hdr;
+u16 port_id;
+/* Port interface and direction (Rx or Tx) to start. */
+
+ u16 payload_size;
+/* Maximum data size of the parameter ID/module ID combination.
+ * This is a multiple of four bytes
+ * Supported values: > 0
+ */
+
+ u32 payload_address_lsw;
+/* LSW of 64 bit Payload address. Address should be 32-byte,
+ * 4kbyte aligned and must be contig memory.
+ */
+
+
+ u32 payload_address_msw;
+/* MSW of 64 bit Payload address. In case of 32-bit shared
+ * memory address, this field must be set to zero. In case of 36-bit
+ * shared memory address, bit-4 to bit-31 must be set to zero.
+ * Address should be 32-byte, 4kbyte aligned and must be contiguous
+ * memory.
+ */
+
+ u32 mem_map_handle;
+/* Memory map handle returned by
+ * AFE_SERVICE_CMD_SHARED_MEM_MAP_REGIONS commands.
+ * Supported Values: - NULL -- Message. The parameter data is
+ * in-band. - Non-NULL -- The parameter data is Out-band.Pointer to
+ * - the physical address in shared memory of the payload data.
+ * For detailed payload content, see the afe_port_param_data_v2
+ * structure
+ */
+
+
+ u32 module_id;
+/* ID of the module to be queried.
+ * Supported values: Valid module ID
+ */
+
+ u32 param_id;
+/* ID of the parameter to be queried.
+ * Supported values: Valid parameter ID
+ */
+} __packed;
+
+#define AFE_PORT_CMDRSP_GET_PARAM_V2 0x00010106
+
+/* Payload of the #AFE_PORT_CMDRSP_GET_PARAM_V2 message, which
+ * responds to an #AFE_PORT_CMD_GET_PARAM_V2 command.
+ *
+ * Immediately following this structure is the parameters structure
+ * (afe_port_param_data) containing the response(acknowledgment)
+ * parameter payload. This payload is included for an in-band
+ * scenario. For an address/shared memory-based set parameter, this
+ * payload is not needed.
+ */
+
+
+struct afe_port_cmdrsp_get_param_v2 {
+ u32 status;
+} __packed;
+
+/* adsp_afe_service_commands.h */
+
+#define ADSP_MEMORY_MAP_EBI_POOL 0
+
+#define ADSP_MEMORY_MAP_SMI_POOL 1
+#define ADSP_MEMORY_MAP_IMEM_POOL 2
+#define ADSP_MEMORY_MAP_SHMEM8_4K_POOL 3
+/*
+* Definition of virtual memory flag
+*/
+#define ADSP_MEMORY_MAP_VIRTUAL_MEMORY 1
+
+/*
+* Definition of physical memory flag
+*/
+#define ADSP_MEMORY_MAP_PHYSICAL_MEMORY 0
+
+
+#define DEFAULT_COPP_TOPOLOGY 0x00010be3
+#define DEFAULT_POPP_TOPOLOGY 0x00010be4
+#define VPM_TX_SM_ECNS_COPP_TOPOLOGY 0x00010F71
+#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY 0x00010F72
+#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY 0x00010F75
+
+/* Memory map regions command payload used by the
+ * #ASM_CMD_SHARED_MEM_MAP_REGIONS ,#ADM_CMD_SHARED_MEM_MAP_REGIONS
+ * commands.
+ *
+ * This structure allows clients to map multiple shared memory
+ * regions in a single command. Following this structure are
+ * num_regions of avs_shared_map_region_payload.
+ */
+
+
+struct avs_cmd_shared_mem_map_regions {
+ struct apr_hdr hdr;
+ u16 mem_pool_id;
+/* Type of memory on which this memory region is mapped.
+ *
+ * Supported values: - #ADSP_MEMORY_MAP_EBI_POOL -
+ * #ADSP_MEMORY_MAP_SMI_POOL - #ADSP_MEMORY_MAP_IMEM_POOL
+ * (unsupported) - #ADSP_MEMORY_MAP_SHMEM8_4K_POOL - Other values
+ * are reserved
+ *
+ * The memory ID implicitly defines the characteristics of the
+ * memory. Characteristics may include alignment type, permissions,
+ * etc.
+ *
+ * SHMEM8_4K is shared memory, byte addressable, and 4 KB aligned.
+ */
+
+
+ u16 num_regions;
+ /* Number of regions to map.*/
+
+ u32 property_flag;
+/* Configures one common property for all the regions in the
+ * payload. No two regions in the same memory map regions cmd can
+ * have differnt property. Supported values: - 0x00000000 to
+ * 0x00000001
+ *
+ * b0 - bit 0 indicates physical or virtual mapping 0 shared memory
+ * address provided in avs_shared_map_regions_payload is physical
+ * address. The shared memory needs to be mapped( hardware TLB
+ * entry)
+ *
+ * and a software entry needs to be added for internal book keeping.
+ *
+ * 1 Shared memory address provided in MayPayload[usRegions] is
+ * virtual address. The shared memory must not be mapped (since
+ * hardware TLB entry is already available) but a software entry
+ * needs to be added for internal book keeping. This can be useful
+ * if two services with in ADSP is communicating via APR. They can
+ * now directly communicate via the Virtual address instead of
+ * Physical address. The virtual regions must be contiguous.
+ *
+ * b31-b1 - reserved bits. must be set to zero
+ */
+
+} __packed;
+
+struct avs_shared_map_region_payload {
+ u32 shm_addr_lsw;
+/* least significant word of shared memory address of the memory
+ * region to map. It must be contiguous memory, and it must be 4 KB
+ * aligned.
+ */
+
+ u32 shm_addr_msw;
+/* most significant word of shared memory address of the memory
+ * region to map. For 32 bit shared memory address, this field must
+ * tbe set to zero. For 36 bit shared memory address, bit31 to bit 4
+ * must be set to zero
+ */
+
+ u32 mem_size_bytes;
+/* Number of bytes in the region.
+ *
+ * The aDSP will always map the regions as virtual contiguous
+ * memory, but the memory size must be in multiples of 4 KB to avoid
+ * gaps in the virtually contiguous mapped memory.
+ */
+
+} __packed;
+
+struct avs_cmd_shared_mem_unmap_regions {
+ struct apr_hdr hdr;
+ u32 mem_map_handle;
+/* memory map handle returned by ASM_CMD_SHARED_MEM_MAP_REGIONS
+ * , ADM_CMD_SHARED_MEM_MAP_REGIONS, commands
+ */
+
+} __packed;
+
+/* Memory map command response payload used by the
+ * #ASM_CMDRSP_SHARED_MEM_MAP_REGIONS
+ * ,#ADM_CMDRSP_SHARED_MEM_MAP_REGIONS
+ */
+
+
+struct avs_cmdrsp_shared_mem_map_regions {
+ u32 mem_map_handle;
+/* A memory map handle encapsulating shared memory attributes is
+ * returned
+ */
+
+} __packed;
+
+/*adsp_audio_memmap_api.h*/
+
+/* ASM related data structures */
+struct asm_wma_cfg {
+ u16 format_tag;
+ u16 ch_cfg;
+ u32 sample_rate;
+ u32 avg_bytes_per_sec;
+ u16 block_align;
+ u16 valid_bits_per_sample;
+ u32 ch_mask;
+ u16 encode_opt;
+ u16 adv_encode_opt;
+ u32 adv_encode_opt2;
+ u32 drc_peak_ref;
+ u32 drc_peak_target;
+ u32 drc_ave_ref;
+ u32 drc_ave_target;
+} __packed;
+
+struct asm_wmapro_cfg {
+ u16 format_tag;
+ u16 ch_cfg;
+ u32 sample_rate;
+ u32 avg_bytes_per_sec;
+ u16 block_align;
+ u16 valid_bits_per_sample;
+ u32 ch_mask;
+ u16 encode_opt;
+ u16 adv_encode_opt;
+ u32 adv_encode_opt2;
+ u32 drc_peak_ref;
+ u32 drc_peak_target;
+ u32 drc_ave_ref;
+ u32 drc_ave_target;
+} __packed;
+
+struct asm_aac_cfg {
+ u16 format;
+ u16 aot;
+ u16 ep_config;
+ u16 section_data_resilience;
+ u16 scalefactor_data_resilience;
+ u16 spectral_data_resilience;
+ u16 ch_cfg;
+ u16 reserved;
+ u32 sample_rate;
+} __packed;
+
+struct asm_softpause_params {
+ u32 enable;
+ u32 period;
+ u32 step;
+ u32 rampingcurve;
+} __packed;
+
+struct asm_softvolume_params {
+ u32 period;
+ u32 step;
+ u32 rampingcurve;
+} __packed;
+
+#define ASM_END_POINT_DEVICE_MATRIX 0
+/* Front left channel. */
+#define PCM_CHANNEL_FL 1
+
+/* Front right channel. */
+#define PCM_CHANNEL_FR 2
+
+/* Front center channel. */
+#define PCM_CHANNEL_FC 3
+
+/* Left surround channel.*/
+#define PCM_CHANNEL_LS 4
+
+/* Right surround channel.*/
+#define PCM_CHANNEL_RS 5
+
+/* Low frequency effect channel. */
+#define PCM_CHANNEL_LFE 6
+
+/* Center surround channel; Rear center channel. */
+#define PCM_CHANNEL_CS 7
+
+/* Left back channel; Rear left channel. */
+#define PCM_CHANNEL_LB 8
+
+/* Right back channel; Rear right channel. */
+#define PCM_CHANNEL_RB 9
+
+/* Top surround channel. */
+#define PCM_CHANNELS 10
+
+/* Center vertical height channel.*/
+#define PCM_CHANNEL_CVH 11
+
+/* Mono surround channel.*/
+#define PCM_CHANNEL_MS 12
+
+/* Front left of center. */
+#define PCM_CHANNEL_FLC 13
+
+/* Front right of center. */
+#define PCM_CHANNEL_FRC 14
+
+/* Rear left of center. */
+#define PCM_CHANNEL_RLC 15
+
+/* Rear right of center. */
+#define PCM_CHANNEL_RRC 16
+
+#define PCM_FORMAT_MAX_NUM_CHANNEL 8
+
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
+
+#define ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT 0x00010BE4
+
+#define ASM_MEDIA_FMT_EVRCB_FS 0x00010BEF
+
+#define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0
+
+#define ASM_MAX_EQ_BANDS 12
+
+#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
+
+struct asm_data_cmd_media_fmt_update_v2 {
+u32 fmt_blk_size;
+ /* Media format block size in bytes.*/
+} __packed;
+
+struct asm_multi_channel_pcm_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+
+ u16 num_channels;
+ /* Number of channels. Supported values: 1 to 8 */
+ u16 bits_per_sample;
+/* Number of bits per sample per channel. * Supported values:
+ * 16, 24 * When used for playback, the client must send 24-bit
+ * samples packed in 32-bit words. The 24-bit samples must be placed
+ * in the most significant 24 bits of the 32-bit word. When used for
+ * recording, the aDSP sends 24-bit samples packed in 32-bit words.
+ * The 24-bit samples are placed in the most significant 24 bits of
+ * the 32-bit word.
+ */
+
+
+ u32 sample_rate;
+/* Number of samples per second (in Hertz).
+ * Supported values: 2000 to 48000
+ */
+
+ u16 is_signed;
+ /* Flag that indicates the samples are signed (1). */
+
+ u16 reserved;
+ /* reserved field for 32 bit alignment. must be set to zero. */
+
+ u8 channel_mapping[8];
+/* Channel array of size 8.
+ * Supported values:
+ * - #PCM_CHANNEL_L
+ * - #PCM_CHANNEL_R
+ * - #PCM_CHANNEL_C
+ * - #PCM_CHANNEL_LS
+ * - #PCM_CHANNEL_RS
+ * - #PCM_CHANNEL_LFE
+ * - #PCM_CHANNEL_CS
+ * - #PCM_CHANNEL_LB
+ * - #PCM_CHANNEL_RB
+ * - #PCM_CHANNELS
+ * - #PCM_CHANNEL_CVH
+ * - #PCM_CHANNEL_MS
+ * - #PCM_CHANNEL_FLC
+ * - #PCM_CHANNEL_FRC
+ * - #PCM_CHANNEL_RLC
+ * - #PCM_CHANNEL_RRC
+ *
+ * Channel[i] mapping describes channel I. Each element i of the
+ * array describes channel I inside the buffer where 0 @le I <
+ * num_channels. An unused channel is set to zero.
+ */
+} __packed;
+
+struct asm_stream_cmd_set_encdec_param {
+ u32 param_id;
+ /* ID of the parameter. */
+
+ u32 param_size;
+/* Data size of this parameter, in bytes. The size is a multiple
+ * of 4 bytes.
+ */
+
+} __packed;
+
+struct asm_enc_cfg_blk_param_v2 {
+ u32 frames_per_buf;
+/* Number of encoded frames to pack into each buffer.
+ *
+ * @note1hang This is only guidance information for the aDSP. The
+ * number of encoded frames put into each buffer (specified by the
+ * client) is less than or equal to this number.
+ */
+
+ u32 enc_cfg_blk_size;
+/* Size in bytes of the encoder configuration block that follows
+ * this member.
+ */
+
+} __packed;
+
+/* @brief Multichannel PCM encoder configuration structure used
+ * in the #ASM_STREAM_CMD_OPEN_READ_V2 command.
+ */
+
+struct asm_multi_channel_pcm_enc_cfg_v2 {
+ 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.
+ *
+ * Supported values: - 0 -- Native mode - 1 -- 8 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.
+ * Supported values: 16, 24
+ */
+
+ uint32_t sample_rate;
+/*< Number of samples per second (in Hertz).
+ *
+ * Supported values: 0, 8000 to 48000 A value of 0 indicates the
+ * native sampling rate. Encoding is performed at the input sampling
+ * rate.
+ */
+
+ uint16_t is_signed;
+/*< Specifies whether the samples are signed (1). Currently,
+ * only signed samples are supported.
+ */
+
+ uint16_t reserved;
+/*< reserved field for 32 bit alignment. must be set to zero.*/
+
+
+ uint8_t channel_mapping[8];
+} __packed;
+
+#define ASM_MEDIA_FMT_MP3 0x00010BE9
+#define ASM_MEDIA_FMT_AAC_V2 0x00010DA6
+
+/* @xreflabel
+ * {hdr:AsmMediaFmtDolbyAac} Media format ID for the
+ * Dolby AAC decoder. This format ID is be used if the client wants
+ * to use the Dolby AAC decoder to decode MPEG2 and MPEG4 AAC
+ * contents.
+ */
+
+#define ASM_MEDIA_FMT_DOLBY_AAC 0x00010D86
+
+/* Enumeration for the audio data transport stream AAC format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS 0
+
+/* Enumeration for low overhead audio stream AAC format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS 1
+
+/* Enumeration for the audio data interchange format
+ * AAC format.
+ */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF 2
+
+/* Enumeration for the raw AAC format. */
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW 3
+
+#define ASM_MEDIA_FMT_AAC_AOT_LC 2
+#define ASM_MEDIA_FMT_AAC_AOT_SBR 5
+#define ASM_MEDIA_FMT_AAC_AOT_PS 29
+#define ASM_MEDIA_FMT_AAC_AOT_BSAC 22
+
+struct asm_aac_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+
+ u16 aac_fmt_flag;
+/* Bitstream format option.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
+ */
+
+ u16 audio_objype;
+/* Audio Object Type (AOT) present in the AAC stream.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_AOT_LC
+ * - #ASM_MEDIA_FMT_AAC_AOT_SBR
+ * - #ASM_MEDIA_FMT_AAC_AOT_BSAC
+ * - #ASM_MEDIA_FMT_AAC_AOT_PS
+ * - Otherwise -- Not supported
+ */
+
+ u16 channel_config;
+/* Number of channels present in the AAC stream.
+ * Supported values:
+ * - 1 -- Mono
+ * - 2 -- Stereo
+ * - 6 -- 5.1 content
+ */
+
+ u16 reserved;
+ /* Reserved. Clients must set this field to zero. */
+
+ u16 total_size_of_PCE_bits;
+/* greater or equal to zero. * -In case of RAW formats and
+ * channel config = 0 (PCE), client can send * the bit stream
+ * containing PCE immediately following this structure * (in-band).
+ * -This number does not include bits included for 32 bit alignment.
+ * -If zero, then the PCE info is assumed to be available in the
+ * audio -bit stream & not in-band.
+ */
+
+ u32 sample_rate;
+/* Number of samples per second (in Hertz).
+ *
+ * Supported values: 8000, 11025, 12000, 16000, 22050, 24000, 32000,
+ * 44100, 48000
+ *
+ * This field must be equal to the sample rate of the AAC-LC
+ * decoder's output. - For MP4 or 3GP containers, this is indicated
+ * by the samplingFrequencyIndex field in the AudioSpecificConfig
+ * element. - For ADTS format, this is indicated by the
+ * samplingFrequencyIndex in the ADTS fixed header. - For ADIF
+ * format, this is indicated by the samplingFrequencyIndex in the
+ * program_config_element present in the ADIF header.
+ */
+
+} __packed;
+
+struct asm_aac_enc_cfg_v2 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+
+ u32 bit_rate;
+ /* Encoding rate in bits per second. */
+ u32 enc_mode;
+/* Encoding mode.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_AOT_LC
+ * - #ASM_MEDIA_FMT_AAC_AOT_SBR
+ * - #ASM_MEDIA_FMT_AAC_AOT_PS
+ */
+ u16 aac_fmt_flag;
+/* AAC format flag.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
+ * - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
+ */
+ u16 channel_cfg;
+/* Number of channels to encode.
+ * Supported values:
+ * - 0 -- Native mode
+ * - 1 -- Mono
+ * - 2 -- Stereo
+ * - Other values are not supported.
+ * @note1hang The eAAC+ encoder mode supports only stereo.
+ * Native mode indicates that encoding must be performed with the
+ * number of channels at the input.
+ * The number of channels must not change during encoding.
+ */
+
+ u32 sample_rate;
+/* Number of samples per second.
+ * Supported values: - 0 -- Native mode - For other values,
+ * Native mode indicates that encoding must be performed with the
+ * sampling rate at the input.
+ * The sampling rate must not change during encoding.
+ */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_AMRNB_FS 0x00010BEB
+
+/* Enumeration for 4.75 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR475 0
+
+/* Enumeration for 5.15 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MR515 1
+
+/* Enumeration for 5.90 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR59 2
+
+/* Enumeration for 6.70 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR67 3
+
+/* Enumeration for 7.40 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR74 4
+
+/* Enumeration for 7.95 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR795 5
+
+/* Enumeration for 10.20 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR102 6
+
+/* Enumeration for 12.20 kbps AMR-NB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_MMR122 7
+
+/* Enumeration for AMR-NB Discontinuous Transmission mode off. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF 0
+
+/* Enumeration for AMR-NB DTX mode VAD1. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1 1
+
+/* Enumeration for AMR-NB DTX mode VAD2. */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD2 2
+
+/* Enumeration for AMR-NB DTX mode auto.
+ */
+#define ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_AUTO 3
+
+struct asm_amrnb_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+
+ u16 enc_mode;
+/* AMR-NB encoding rate.
+ * Supported values:
+ * Use the ASM_MEDIA_FMT_AMRNB_FS_ENCODE_MODE_*
+ * macros
+ */
+
+ u16 dtx_mode;
+/* Specifies whether DTX mode is disabled or enabled.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1
+ */
+} __packed;
+
+#define ASM_MEDIA_FMT_AMRWB_FS 0x00010BEC
+
+/* Enumeration for 6.6 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR66 0
+
+/* Enumeration for 8.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR885 1
+
+/* Enumeration for 12.65 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1265 2
+
+/* Enumeration for 14.25 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1425 3
+
+/* Enumeration for 15.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1585 4
+
+/* Enumeration for 18.25 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1825 5
+
+/* Enumeration for 19.85 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR1985 6
+
+/* Enumeration for 23.05 kbps AMR-WB Encoding mode. */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2305 7
+
+/* Enumeration for 23.85 kbps AMR-WB Encoding mode.
+ */
+#define ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_MR2385 8
+
+struct asm_amrwb_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+
+ u16 enc_mode;
+/* AMR-WB encoding rate.
+ * Suupported values:
+ * Use the ASM_MEDIA_FMT_AMRWB_FS_ENCODE_MODE_*
+ * macros
+ */
+
+ u16 dtx_mode;
+/* Specifies whether DTX mode is disabled or enabled.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_OFF
+ * - #ASM_MEDIA_FMT_AMRNB_FS_DTX_MODE_VAD1
+ */
+} __packed;
+
+#define ASM_MEDIA_FMT_V13K_FS 0x00010BED
+
+/* Enumeration for 14.4 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440 0
+
+/* Enumeration for 12.2 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220 1
+
+/* Enumeration for 11.2 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120 2
+
+/* Enumeration for 9.0 kbps V13K Encoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90 3
+
+/* Enumeration for 7.2 kbps V13K eEncoding mode. */
+#define ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720 4
+
+/* Enumeration for 1/8 vocoder rate.*/
+#define ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE 1
+
+/* Enumeration for 1/4 vocoder rate. */
+#define ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE 2
+
+/* Enumeration for 1/2 vocoder rate. */
+#define ASM_MEDIA_FMT_VOC_HALF_RATE 3
+
+/* Enumeration for full vocoder rate.
+ */
+#define ASM_MEDIA_FMT_VOC_FULL_RATE 4
+
+struct asm_v13k_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u16 max_rate;
+/* Maximum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+ u16 min_rate;
+/* Minimum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+ u16 reduced_rate_cmd;
+/* Reduced rate command, used to change
+ * the average bitrate of the V13K
+ * vocoder.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1440 (Default)
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1220
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR1120
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR90
+ * - #ASM_MEDIA_FMT_V13K_FS_ENCODE_MODE_MR720
+ */
+
+ u16 rate_mod_cmd;
+/* Rate modulation command. Default = 0.
+ *- If bit 0=1, rate control is enabled.
+ *- If bit 1=1, the maximum number of consecutive full rate
+ * frames is limited with numbers supplied in
+ * bits 2 to 10.
+ *- If bit 1=0, the minimum number of non-full rate frames
+ * in between two full rate frames is forced to
+ * the number supplied in bits 2 to 10. In both cases, if necessary,
+ * half rate is used to substitute full rate. - Bits 15 to 10 are
+ * reserved and must all be set to zero.
+ */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_EVRC_FS 0x00010BEE
+
+/* EVRC encoder configuration structure used in the
+ * #ASM_STREAM_CMD_OPEN_READ_V2 command.
+ */
+struct asm_evrc_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u16 max_rate;
+/* Maximum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+ u16 min_rate;
+/* Minimum allowed encoder frame rate.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_VOC_ONE_EIGHTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_ONE_FOURTH_RATE
+ * - #ASM_MEDIA_FMT_VOC_HALF_RATE
+ * - #ASM_MEDIA_FMT_VOC_FULL_RATE
+ */
+
+ u16 rate_mod_cmd;
+/* Rate modulation command. Default: 0.
+ * - If bit 0=1, rate control is enabled.
+ * - If bit 1=1, the maximum number of consecutive full rate frames
+ * is limited with numbers supplied in bits 2 to 10.
+ *
+ * - If bit 1=0, the minimum number of non-full rate frames in
+ * between two full rate frames is forced to the number supplied in
+ * bits 2 to 10. In both cases, if necessary, half rate is used to
+ * substitute full rate.
+ *
+ * - Bits 15 to 10 are reserved and must all be set to zero.
+ */
+
+ u16 reserved;
+ /* Reserved. Clients must set this field to zero. */
+} __packed;
+
+#define ASM_MEDIA_FMT_WMA_V10PRO_V2 0x00010DA7
+
+struct asm_wmaprov10_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+ u16 fmtag;
+/* WMA format type.
+ * Supported values:
+ * - 0x162 -- WMA 9 Pro
+ * - 0x163 -- WMA 9 Pro Lossless
+ * - 0x166 -- WMA 10 Pro
+ * - 0x167 -- WMA 10 Pro Lossless
+ */
+
+ u16 num_channels;
+/* Number of channels encoded in the input stream.
+ * Supported values: 1 to 8
+ */
+
+ u32 sample_rate;
+/* Number of samples per second (in Hertz).
+ * Supported values: 11025, 16000, 22050, 32000, 44100, 48000,
+ * 88200, 96000
+ */
+
+ u32 avg_bytes_per_sec;
+/* Bitrate expressed as the average bytes per second.
+ * Supported values: 2000 to 96000
+ */
+
+ u16 blk_align;
+/* Size of the bitstream packet size in bytes. WMA Pro files
+ * have a payload of one block per bitstream packet.
+ * Supported values: @le 13376
+ */
+
+ u16 bits_per_sample;
+/* Number of bits per sample in the encoded WMA stream.
+ * Supported values: 16, 24
+ */
+
+ u32 channel_mask;
+/* Bit-packed double word (32-bits) that indicates the
+ * recommended speaker positions for each source channel.
+ */
+
+ u16 enc_options;
+/* Bit-packed word with values that indicate whether certain
+ * features of the bitstream are used.
+ * Supported values: - 0x0001 -- ENCOPT3_PURE_LOSSLESS - 0x0006 --
+ * ENCOPT3_FRM_SIZE_MOD - 0x0038 -- ENCOPT3_SUBFRM_DIV - 0x0040 --
+ * ENCOPT3_WRITE_FRAMESIZE_IN_HDR - 0x0080 --
+ * ENCOPT3_GENERATE_DRC_PARAMS - 0x0100 -- ENCOPT3_RTMBITS
+ */
+
+
+ u16 usAdvancedEncodeOpt;
+ /* Advanced encoding option. */
+
+ u32 advanced_enc_options2;
+ /* Advanced encoding option 2. */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_WMA_V9_V2 0x00010DA8
+struct asm_wmastdv9_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+ u16 fmtag;
+/* WMA format tag.
+ * Supported values: 0x161 (WMA 9 standard)
+ */
+
+ u16 num_channels;
+/* Number of channels in the stream.
+ * Supported values: 1, 2
+ */
+
+ u32 sample_rate;
+/* Number of samples per second (in Hertz).
+ * Supported values: 48000
+ */
+
+ u32 avg_bytes_per_sec;
+ /* Bitrate expressed as the average bytes per second. */
+
+ u16 blk_align;
+/* Block align. All WMA files with a maximum packet size of
+ * 13376 are supported.
+ */
+
+
+ u16 bits_per_sample;
+/* Number of bits per sample in the output.
+ * Supported values: 16
+ */
+
+ u32 channel_mask;
+/* Channel mask.
+ * Supported values:
+ * - 3 -- Stereo (front left/front right)
+ * - 4 -- Mono (center)
+ */
+
+ u16 enc_options;
+ /* Options used during encoding. */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_WMA_V8 0x00010D91
+
+struct asm_wmastdv8_enc_cfg {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u32 bit_rate;
+ /* Encoding rate in bits per second. */
+
+ u32 sample_rate;
+/* Number of samples per second.
+ *
+ * Supported values:
+ * - 0 -- Native mode
+ * - Other Supported values are 22050, 32000, 44100, and 48000.
+ *
+ * Native mode indicates that encoding must be performed with the
+ * sampling rate at the input.
+ * The sampling rate must not change during encoding.
+ */
+
+ u16 channel_cfg;
+/* Number of channels to encode.
+ * Supported values:
+ * - 0 -- Native mode
+ * - 1 -- Mono
+ * - 2 -- Stereo
+ * - Other values are not supported.
+ *
+ * Native mode indicates that encoding must be performed with the
+ * number of channels at the input.
+ * The number of channels must not change during encoding.
+ */
+
+ u16 reserved;
+ /* Reserved. Clients must set this field to zero.*/
+ } __packed;
+
+#define ASM_MEDIA_FMT_AMR_WB_PLUS_V2 0x00010DA9
+
+struct asm_amrwbplus_fmt_blk_v2 {
+ struct apr_hdr hdr;
+ struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+ u32 amr_frame_fmt;
+/* AMR frame format.
+ * Supported values:
+ * - 6 -- Transport Interface Format (TIF)
+ * - Any other value -- File storage format (FSF)
+ *
+ * TIF stream contains 2-byte header for each frame within the
+ * superframe. FSF stream contains one 2-byte header per superframe.
+ */
+
+} __packed;
+
+#define ASM_MEDIA_FMT_AC3_DEC 0x00010BF6
+#define ASM_MEDIA_FMT_EAC3_DEC 0x00010C3C
+#define ASM_MEDIA_FMT_DTS 0x00010D88
+
+/* Media format ID for adaptive transform acoustic coding. This
+ * ID is used by the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED command
+ * only.
+ */
+
+#define ASM_MEDIA_FMT_ATRAC 0x00010D89
+
+/* Media format ID for metadata-enhanced audio transmission.
+ * This ID is used by the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED
+ * command only.
+ */
+
+#define ASM_MEDIA_FMT_MAT 0x00010D8A
+
+/* adsp_media_fmt.h */
+
+#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
+
+struct asm_data_cmd_write_v2 {
+ struct apr_hdr hdr;
+ u32 buf_addr_lsw;
+/* The 64 bit address msw-lsw should be a valid, mapped address.
+ * 64 bit address should be a multiple of 32 bytes
+ */
+
+ u32 buf_addr_msw;
+/* The 64 bit address msw-lsw should be a valid, mapped address.
+ * 64 bit address should be a multiple of 32 bytes.
+ * -Address of the buffer containing the data to be decoded.
+ * The buffer should be aligned to a 32 byte boundary.
+ * -In the case of 32 bit Shared memory address, msw field must
+ * -be set to zero.
+ * -In the case of 36 bit shared memory address, bit 31 to bit 4
+ * -of msw must be set to zero.
+ */
+ u32 mem_map_handle;
+/* memory map handle returned by DSP through
+ * ASM_CMD_SHARED_MEM_MAP_REGIONS command
+ */
+ u32 buf_size;
+/* Number of valid bytes available in the buffer for decoding. The
+ * first byte starts at buf_addr.
+ */
+
+ u32 seq_id;
+ /* Optional buffer sequence ID. */
+
+ u32 timestamp_lsw;
+/* Lower 32 bits of the 64-bit session time in microseconds of the
+ * first buffer sample.
+ */
+
+ u32 timestamp_msw;
+/* Upper 32 bits of the 64-bit session time in microseconds of the
+ * first buffer sample.
+ */
+
+ u32 flags;
+/* Bitfield of flags.
+ * Supported values for bit 31:
+ * - 1 -- Valid timestamp.
+ * - 0 -- Invalid timestamp.
+ * - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG as the bitmask and
+ * #ASM_SHIFTIMESTAMP_VALID_FLAG as the shift value to set this bit.
+ * Supported values for bit 30:
+ * - 1 -- Last buffer.
+ * - 0 -- Not the last buffer.
+ *
+ * Supported values for bit 29:
+ * - 1 -- Continue the timestamp from the previous buffer.
+ * - 0 -- Timestamp of the current buffer is not related
+ * to the timestamp of the previous buffer.
+ * - Use #ASM_BIT_MASKS_CONTINUE_FLAG and #ASM_SHIFTS_CONTINUE_FLAG
+ * to set this bit.
+ *
+ * Supported values for bit 4:
+ * - 1 -- End of the frame.
+ * - 0 -- Not the end of frame, or this information is not known.
+ * - Use #ASM_BIT_MASK_EOF_FLAG as the bitmask and #ASM_SHIFT_EOF_FLAG
+ * as the shift value to set this bit.
+ *
+ * All other bits are reserved and must be set to 0.
+ *
+ * If bit 31=0 and bit 29=1: The timestamp of the first sample in
+ * this buffer continues from the timestamp of the last sample in
+ * the previous buffer. If there is no previous buffer (i.e., this
+ * is the first buffer sent after opening the stream or after a
+ * flush operation), or if the previous buffer does not have a valid
+ * timestamp, the samples in the current buffer also do not have a
+ * valid timestamp. They are played out as soon as possible.
+ *
+ *
+ * If bit 31=0 and bit 29=0: No timestamp is associated with the
+ * first sample in this buffer. The samples are played out as soon
+ * as possible.
+ *
+ *
+ * If bit 31=1 and bit 29 is ignored: The timestamp specified in
+ * this payload is honored.
+ *
+ *
+ * If bit 30=0: Not the last buffer in the stream. This is useful
+ * in removing trailing samples.
+ *
+ *
+ * For bit 4: The client can set this flag for every buffer sent in
+ * which the last byte is the end of a frame. If this flag is set,
+ * the buffer can contain data from multiple frames, but it should
+ * always end at a frame boundary. Restrictions allow the aDSP to
+ * detect an end of frame without requiring additional processing.
+ */
+
+} __packed;
+
+#define ASM_DATA_CMD_READ_V2 0x00010DAC
+
+struct asm_data_cmd_read_v2 {
+ struct apr_hdr hdr;
+ u32 buf_addr_lsw;
+/* the 64 bit address msw-lsw should be a valid mapped address
+ * and should be a multiple of 32 bytes
+ */
+
+
+ u32 buf_addr_msw;
+/* the 64 bit address msw-lsw should be a valid mapped address
+ * and should be a multiple of 32 bytes.
+* - Address of the buffer where the DSP puts the encoded data,
+* potentially, at an offset specified by the uOffset field in
+* ASM_DATA_EVENT_READ_DONE structure. The buffer should be aligned
+* to a 32 byte boundary.
+*- In the case of 32 bit Shared memory address, msw field must
+*- be set to zero.
+*- In the case of 36 bit shared memory address, bit 31 to bit
+*- 4 of msw must be set to zero.
+*/
+ u32 mem_map_handle;
+/* memory map handle returned by DSP through
+ * ASM_CMD_SHARED_MEM_MAP_REGIONS command.
+ */
+
+ u32 buf_size;
+/* Number of bytes available for the aDSP to write. The aDSP
+ * starts writing from buf_addr.
+ */
+
+ u32 seq_id;
+ /* Optional buffer sequence ID.
+ */
+} __packed;
+
+#define ASM_DATA_CMD_EOS 0x00010BDB
+#define ASM_DATA_EVENT_RENDERED_EOS 0x00010C1C
+#define ASM_DATA_EVENT_EOS 0x00010BDD
+
+#define ASM_DATA_EVENT_WRITE_DONE_V2 0x00010D99
+struct asm_data_event_write_done_v2 {
+ u32 buf_addr_lsw;
+ /* lsw of the 64 bit address */
+ u32 buf_addr_msw;
+ /* msw of the 64 bit address. address given by the client in
+ * ASM_DATA_CMD_WRITE_V2 command.
+ */
+ u32 mem_map_handle;
+ /* memory map handle in the ASM_DATA_CMD_WRITE_V2 */
+
+ u32 status;
+/* Status message (error code) that indicates whether the
+ * referenced buffer has been successfully consumed.
+ * Supported values: Refer to @xhyperref{Q3,[Q3]}
+ */
+} __packed;
+
+#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
+
+/* Definition of the frame metadata flag bitmask.*/
+#define ASM_BIT_MASK_FRAME_METADATA_FLAG (0x40000000UL)
+
+/* Definition of the frame metadata flag shift value. */
+#define ASM_SHIFT_FRAME_METADATA_FLAG 30
+
+struct asm_data_event_read_done_v2 {
+ u32 status;
+/* Status message (error code).
+ * Supported values: Refer to @xhyperref{Q3,[Q3]}
+ */
+
+u32 buf_addr_lsw;
+/* 64 bit address msw-lsw is a valid, mapped address. 64 bit
+ * address is a multiple of 32 bytes.
+ */
+
+u32 buf_addr_msw;
+/* 64 bit address msw-lsw is a valid, mapped address. 64 bit
+* address is a multiple of 32 bytes.
+*
+* -Same address provided by the client in ASM_DATA_CMD_READ_V2
+* -In the case of 32 bit Shared memory address, msw field is set to
+* zero.
+* -In the case of 36 bit shared memory address, bit 31 to bit 4
+* -of msw is set to zero.
+*/
+
+u32 mem_map_handle;
+/* memory map handle in the ASM_DATA_CMD_READ_V2 */
+
+u32 enc_framesotal_size;
+/* Total size of the encoded frames in bytes.
+ * Supported values: >0
+ */
+
+u32 offset;
+/* Offset (from buf_addr) to the first byte of the first encoded
+ * frame. All encoded frames are consecutive, starting from this
+ * offset.
+ * Supported values: > 0
+ */
+
+u32 timestamp_lsw;
+/* Lower 32 bits of the 64-bit session time in microseconds of
+ * the first sample in the buffer. If Bit 5 of mode_flags flag of
+ * ASM_STREAM_CMD_OPEN_READ_V2 is 1 then the 64 bit timestamp is
+ * absolute capture time otherwise it is relative session time. The
+ * absolute timestamp doesnt reset unless the system is reset.
+ */
+
+
+u32 timestamp_msw;
+/* Upper 32 bits of the 64-bit session time in microseconds of
+ * the first sample in the buffer.
+ */
+
+
+u32 flags;
+/* Bitfield of flags. Bit 30 indicates whether frame metadata is
+ * present. If frame metadata is present, num_frames consecutive
+ * instances of @xhyperref{hdr:FrameMetaData,Frame metadata} start
+ * at the buffer address.
+ * Supported values for bit 31:
+ * - 1 -- Timestamp is valid.
+ * - 0 -- Timestamp is invalid.
+ * - Use #ASM_BIT_MASKIMESTAMP_VALID_FLAG and
+ * #ASM_SHIFTIMESTAMP_VALID_FLAG to set this bit.
+ *
+ * Supported values for bit 30:
+ * - 1 -- Frame metadata is present.
+ * - 0 -- Frame metadata is absent.
+ * - Use #ASM_BIT_MASK_FRAME_METADATA_FLAG and
+ * #ASM_SHIFT_FRAME_METADATA_FLAG to set this bit.
+ *
+ * All other bits are reserved; the aDSP sets them to 0.
+ */
+
+u32 num_frames;
+/* Number of encoded frames in the buffer. */
+
+u32 seq_id;
+/* Optional buffer sequence ID. */
+} __packed;
+
+struct asm_data_read_buf_metadata_v2 {
+ u32 offset;
+/* Offset from buf_addr in #ASM_DATA_EVENT_READ_DONE_PAYLOAD to
+ * the frame associated with this metadata.
+ * Supported values: > 0
+ */
+
+u32 frm_size;
+/* Size of the encoded frame in bytes.
+ * Supported values: > 0
+ */
+
+u32 num_encoded_pcm_samples;
+/* Number of encoded PCM samples (per channel) in the frame
+ * associated with this metadata.
+ * Supported values: > 0
+ */
+
+u32 timestamp_lsw;
+/* Lower 32 bits of the 64-bit session time in microseconds of the
+ * first sample for this frame.
+ * If Bit 5 of mode_flags flag of ASM_STREAM_CMD_OPEN_READ_V2 is 1
+ * then the 64 bit timestamp is absolute capture time otherwise it
+ * is relative session time. The absolute timestamp doesnt reset
+ * unless the system is reset.
+ */
+
+
+u32 timestamp_msw;
+/* Lower 32 bits of the 64-bit session time in microseconds of the
+ * first sample for this frame.
+ */
+
+u32 flags;
+/* Frame flags.
+ * Supported values for bit 31:
+ * - 1 -- Time stamp is valid
+ * - 0 -- Time stamp is not valid
+ * - All other bits are reserved; the aDSP sets them to 0.
+*/
+} __packed;
+
+/* Notifies the client of a change in the data sampling rate or
+ * Channel mode. This event is raised by the decoder service. The
+ * event is enabled through the mode flags of
+ * #ASM_STREAM_CMD_OPEN_WRITE_V2 or
+ * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
+ * in the output sampling frequency or the number/positioning of
+ * output channels, or if it is the first frame decoded.The new
+ * sampling frequency or the new channel configuration is
+ * communicated back to the client asynchronously.
+ */
+
+#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY 0x00010C65
+
+/* Payload of the #ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY event.
+ * This event is raised when the following conditions are both true:
+ * - The event is enabled through the mode_flags of
+ * #ASM_STREAM_CMD_OPEN_WRITE_V2 or
+ * #ASM_STREAM_CMD_OPEN_READWRITE_V2. - The decoder detects a change
+ * in either the output sampling frequency or the number/positioning
+ * of output channels, or if it is the first frame decoded.
+ * This event is not raised (even if enabled) if the decoder is
+ * MIDI, because
+ */
+
+
+struct asm_data_event_sr_cm_change_notify {
+ u32 sample_rate;
+/* New sampling rate (in Hertz) after detecting a change in the
+ * bitstream.
+ * Supported values: 2000 to 48000
+ */
+
+ u16 num_channels;
+/* New number of channels after detecting a change in the
+ * bitstream.
+ * Supported values: 1 to 8
+ */
+
+
+ u16 reserved;
+ /* Reserved for future use. This field must be set to 0.*/
+
+ u8 channel_mapping[8];
+
+} __packed;
+
+/* Notifies the client of a data sampling rate or channel mode
+ * change. This event is raised by the encoder service.
+ * This event is raised when :
+ * - Native mode encoding was requested in the encoder
+ * configuration (i.e., the channel number was 0), the sample rate
+ * was 0, or both were 0.
+ *
+ * - The input data frame at the encoder is the first one, or the
+ * sampling rate/channel mode is different from the previous input
+ * data frame.
+ *
+ */
+#define ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY 0x00010BDE
+
+struct asm_data_event_enc_sr_cm_change_notify {
+ u32 sample_rate;
+/* New sampling rate (in Hertz) after detecting a change in the
+ * input data.
+ * Supported values: 2000 to 48000
+ */
+
+
+ u16 num_channels;
+/* New number of channels after detecting a change in the input
+ * data. Supported values: 1 to 8
+ */
+
+
+ u16 bits_per_sample;
+/* New bits per sample after detecting a change in the input
+ * data.
+ * Supported values: 16, 24
+ */
+
+
+ u8 channel_mapping[8];
+
+} __packed;
+#define ASM_DATA_CMD_IEC_60958_FRAME_RATE 0x00010D87
+
+
+/* Payload of the #ASM_DATA_CMD_IEC_60958_FRAME_RATE command,
+ * which is used to indicate the IEC 60958 frame rate of a given
+ * packetized audio stream.
+ */
+
+struct asm_data_cmd_iec_60958_frame_rate {
+ u32 frame_rate;
+/* IEC 60958 frame rate of the incoming IEC 61937 packetized stream.
+ * Supported values: Any valid frame rate
+ */
+} __packed;
+
+/* adsp_asm_data_commands.h*/
+#define ASM_SVC_CMD_GET_STREAM_HANDLES 0x00010C0B
+
+#define ASM_SVC_CMDRSP_GET_STREAM_HANDLES 0x00010C1B
+
+/* Definition of the stream ID bitmask.*/
+#define ASM_BIT_MASK_STREAM_ID (0x000000FFUL)
+
+/* Definition of the stream ID shift value.*/
+#define ASM_SHIFT_STREAM_ID 0
+
+/* Definition of the session ID bitmask.*/
+#define ASM_BIT_MASK_SESSION_ID (0x0000FF00UL)
+
+/* Definition of the session ID shift value.*/
+#define ASM_SHIFT_SESSION_ID 8
+
+/* Definition of the service ID bitmask.*/
+#define ASM_BIT_MASK_SERVICE_ID (0x00FF0000UL)
+
+/* Definition of the service ID shift value.*/
+#define ASM_SHIFT_SERVICE_ID 16
+
+/* Definition of the domain ID bitmask.*/
+#define ASM_BIT_MASK_DOMAIN_ID (0xFF000000UL)
+
+/* Definition of the domain ID shift value.*/
+#define ASM_SHIFT_DOMAIN_ID 24
+
+/* Payload of the #ASM_SVC_CMDRSP_GET_STREAM_HANDLES message,
+ * which returns a list of currently active stream handles.
+ * Immediately following this structure are num_handles of uint32
+ * stream handles.
+ */
+
+
+struct asm_svc_cmdrsp_get_stream_handles {
+ u32 num_handles;
+ /* Number of active stream handles. */
+} __packed;
+
+#define ASM_CMD_SHARED_MEM_MAP_REGIONS 0x00010D92
+#define ASM_CMDRSP_SHARED_MEM_MAP_REGIONS 0x00010D93
+#define ASM_CMD_SHARED_MEM_UNMAP_REGIONS 0x00010D94
+
+/* adsp_asm_service_commands.h */
+
+#define ASM_MAX_SESSION_ID (8)
+
+/* Maximum number of sessions.*/
+#define ASM_MAX_NUM_SESSIONS ASM_MAX_SESSION_ID
+
+/* Maximum number of streams per session.*/
+#define ASM_MAX_STREAMS_PER_SESSION (8)
+#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE 0
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME 1
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME 2
+#define ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY 3
+
+#define ASM_BIT_MASK_RUN_STARTIME (0x00000003UL)
+
+/* Bit shift value used to specify the start time for the
+ * ASM_SESSION_CMD_RUN_V2 command.
+ */
+#define ASM_SHIFT_RUN_STARTIME 0
+struct asm_session_cmd_run_v2 {
+ struct apr_hdr hdr;
+ u32 flags;
+/* Specifies whether to run immediately or at a specific
+ * rendering time or with a specified delay. Run with delay is
+ * useful for delaying in case of ASM loopback opened through
+ * ASM_STREAM_CMD_OPEN_LOOPBACK_V2. Use #ASM_BIT_MASK_RUN_STARTIME
+ * and #ASM_SHIFT_RUN_STARTIME to set this 2-bit flag.
+ *
+ *
+ *Bits 0 and 1 can take one of four possible values:
+ *
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_IMMEDIATE
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_ABSOLUTEIME
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_AT_RELATIVEIME
+ *- #ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY
+ *
+ *All other bits are reserved; clients must set them to zero.
+ */
+
+ u32 time_lsw;
+/* Lower 32 bits of the time in microseconds used to align the
+ * session origin time. When bits 0-1 of flags is
+ * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time lsw is the lsw of
+ * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY,
+ * maximum value of the 64 bit delay is 150 ms.
+ */
+
+ u32 time_msw;
+/* Upper 32 bits of the time in microseconds used to align the
+ * session origin time. When bits 0-1 of flags is
+ * ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY, time msw is the msw of
+ * the delay in us. For ASM_SESSION_CMD_RUN_START_RUN_WITH_DELAY,
+ * maximum value of the 64 bit delay is 150 ms.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMD_PAUSE 0x00010BD3
+#define ASM_SESSION_CMD_GET_SESSIONTIME_V3 0x00010D9D
+#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5
+
+struct asm_session_cmd_rgstr_rx_underflow {
+ struct apr_hdr hdr;
+ u16 enable_flag;
+/* Specifies whether a client is to receive events when an Rx
+ * session underflows.
+ * Supported values:
+ * - 0 -- Do not send underflow events
+ * - 1 -- Send underflow events
+ */
+ u16 reserved;
+ /* Reserved. This field must be set to zero.*/
+} __packed;
+
+#define ASM_SESSION_CMD_REGISTER_FORX_OVERFLOW_EVENTS 0x00010BD6
+
+struct asm_session_cmd_regx_overflow {
+ struct apr_hdr hdr;
+ u16 enable_flag;
+/* Specifies whether a client is to receive events when a Tx
+* session overflows.
+ * Supported values:
+ * - 0 -- Do not send overflow events
+ * - 1 -- Send overflow events
+ */
+
+ u16 reserved;
+ /* Reserved. This field must be set to zero.*/
+} __packed;
+
+#define ASM_SESSION_EVENT_RX_UNDERFLOW 0x00010C17
+#define ASM_SESSION_EVENTX_OVERFLOW 0x00010C18
+#define ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3 0x00010D9E
+
+struct asm_session_cmdrsp_get_sessiontime_v3 {
+ u32 status;
+ /* Status message (error code).
+ * Supported values: Refer to @xhyperref{Q3,[Q3]}
+ */
+
+ u32 sessiontime_lsw;
+ /* Lower 32 bits of the current session time in microseconds.*/
+
+ u32 sessiontime_msw;
+ /* Upper 32 bits of the current session time in microseconds.*/
+
+ u32 absolutetime_lsw;
+/* Lower 32 bits in micro seconds of the absolute time at which
+ * the * sample corresponding to the above session time gets
+ * rendered * to hardware. This absolute time may be slightly in the
+ * future or past.
+ */
+
+
+ u32 absolutetime_msw;
+/* Upper 32 bits in micro seconds of the absolute time at which
+ * the * sample corresponding to the above session time gets
+ * rendered to * hardware. This absolute time may be slightly in the
+ * future or past.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMD_ADJUST_SESSION_CLOCK_V2 0x00010D9F
+
+struct asm_session_cmd_adjust_session_clock_v2 {
+ struct apr_hdr hdr;
+u32 adjustime_lsw;
+/* Lower 32 bits of the signed 64-bit quantity that specifies the
+ * adjustment time in microseconds to the session clock.
+ *
+ * Positive values indicate advancement of the session clock.
+ * Negative values indicate delay of the session clock.
+ */
+
+
+ u32 adjustime_msw;
+/* Upper 32 bits of the signed 64-bit quantity that specifies
+ * the adjustment time in microseconds to the session clock.
+ * Positive values indicate advancement of the session clock.
+ * Negative values indicate delay of the session clock.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMDRSP_ADJUST_SESSION_CLOCK_V2 0x00010DA0
+
+struct asm_session_cmdrsp_adjust_session_clock_v2 {
+ u32 status;
+/* Status message (error code).
+ * Supported values: Refer to @xhyperref{Q3,[Q3]}
+ * An error means the session clock is not adjusted. In this case,
+ * the next two fields are irrelevant.
+ */
+
+
+ u32 actual_adjustime_lsw;
+/* Lower 32 bits of the signed 64-bit quantity that specifies
+ * the actual adjustment in microseconds performed by the aDSP.
+ * A positive value indicates advancement of the session clock. A
+ * negative value indicates delay of the session clock.
+ */
+
+
+ u32 actual_adjustime_msw;
+/* Upper 32 bits of the signed 64-bit quantity that specifies
+ * the actual adjustment in microseconds performed by the aDSP.
+ * A positive value indicates advancement of the session clock. A
+ * negative value indicates delay of the session clock.
+ */
+
+
+ u32 cmd_latency_lsw;
+/* Lower 32 bits of the unsigned 64-bit quantity that specifies
+ * the amount of time in microseconds taken to perform the session
+ * clock adjustment.
+ */
+
+
+ u32 cmd_latency_msw;
+/* Upper 32 bits of the unsigned 64-bit quantity that specifies
+ * the amount of time in microseconds taken to perform the session
+ * clock adjustment.
+ */
+
+} __packed;
+
+#define ASM_SESSION_CMD_GET_PATH_DELAY_V2 0x00010DAF
+#define ASM_SESSION_CMDRSP_GET_PATH_DELAY_V2 0x00010DB0
+
+struct asm_session_cmdrsp_get_path_delay_v2 {
+ u32 status;
+/* Status message (error code). Whether this get delay operation
+ * is successful or not. Delay value is valid only if status is
+ * success.
+ * Supported values: Refer to @xhyperref{Q5,[Q5]}
+ */
+
+ u32 audio_delay_lsw;
+ /* Upper 32 bits of the aDSP delay in microseconds. */
+
+ u32 audio_delay_msw;
+ /* Lower 32 bits of the aDSP delay in microseconds. */
+
+} __packed;
+
+/* adsp_asm_session_command.h*/
+#define ASM_STREAM_CMD_OPEN_WRITE_V2 0x00010D8F
+
+struct asm_stream_cmd_open_write_v2 {
+ struct apr_hdr hdr;
+ uint32_t mode_flags;
+/* Mode flags that configure the stream to notify the client
+ * whenever it detects an SR/CM change at the input to its POPP.
+ * Supported values for bits 0 to 1:
+ * - Reserved; clients must set them to zero.
+ * Supported values for bit 2:
+ * - 0 -- SR/CM change notification event is disabled.
+ * - 1 -- SR/CM change notification event is enabled.
+ * - Use #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and
+ * #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or get this bit.
+ *
+ * Supported values for bit 31:
+ * - 0 -- Stream to be opened in on-Gapless mode.
+ * - 1 -- Stream to be opened in Gapless mode. In Gapless mode,
+ * successive streams must be opened with same session ID but
+ * different stream IDs.
+ *
+ * - Use #ASM_BIT_MASK_GAPLESS_MODE_FLAG and
+ * #ASM_SHIFT_GAPLESS_MODE_FLAG to set or get this bit.
+ *
+ *
+ * @note1hang MIDI and DTMF streams cannot be opened in Gapless mode.
+ */
+
+ uint16_t sink_endpointype;
+/*< Sink point type.
+ * Supported values:
+ * - 0 -- Device matrix
+ * - Other values are reserved.
+ *
+ * The device matrix is the gateway to the hardware ports.
+ */
+
+ uint16_t bits_per_sample;
+/*< Number of bits per sample processed by ASM modules.
+ * Supported values: 16 and 24 bits per sample
+ */
+
+ uint32_t postprocopo_id;
+/*< Specifies the topology (order of processing) of
+ * postprocessing algorithms. <i>None</i> means no postprocessing.
+ * Supported values:
+ * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT
+ * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL
+ * - #ASM_STREAM_POSTPROCOPO_ID_NONE
+ *
+ * This field can also be enabled through SetParams flags.
+ */
+
+ uint32_t dec_fmt_id;
+/*< Configuration ID of the decoder media format.
+ *
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_ADPCM
+ * - #ASM_MEDIA_FMT_MP3
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_DOLBY_AAC
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_WMA_V10PRO_V2
+ * - #ASM_MEDIA_FMT_WMA_V9_V2
+ * - #ASM_MEDIA_FMT_AC3_DEC
+ * - #ASM_MEDIA_FMT_EAC3_DEC
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_FR_FS
+ * - #ASM_MEDIA_FMT_VORBIS
+ * - #ASM_MEDIA_FMT_FLAC
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ */
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_READ_V2 0x00010D8C
+/* Definition of the timestamp type flag bitmask */
+#define ASM_BIT_MASKIMESTAMPYPE_FLAG (0x00000020UL)
+
+/* Definition of the timestamp type flag shift value. */
+#define ASM_SHIFTIMESTAMPYPE_FLAG 5
+
+/* Relative timestamp is identified by this value.*/
+#define ASM_RELATIVEIMESTAMP 0
+
+/* Absolute timestamp is identified by this value.*/
+#define ASM_ABSOLUTEIMESTAMP 1
+
+
+struct asm_stream_cmd_open_read_v2 {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode flags that indicate whether meta information per encoded
+ * frame is to be provided.
+ * Supported values for bit 4:
+ *
+ * - 0 -- Return data buffer contains all encoded frames only; it
+ * does not contain frame metadata.
+ *
+ * - 1 -- Return data buffer contains an array of metadata and
+ * encoded frames.
+ *
+ * - Use #ASM_BIT_MASK_META_INFO_FLAG as the bitmask and
+ * #ASM_SHIFT_META_INFO_FLAG as the shift value for this bit.
+ *
+ *
+ * Supported values for bit 5:
+ *
+ * - ASM_RELATIVEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will have
+ * - relative time-stamp.
+ * - ASM_ABSOLUTEIMESTAMP -- ASM_DATA_EVENT_READ_DONE_V2 will
+ * - have absolute time-stamp.
+ *
+ * - Use #ASM_BIT_MASKIMESTAMPYPE_FLAG as the bitmask and
+ * #ASM_SHIFTIMESTAMPYPE_FLAG as the shift value for this bit.
+ *
+ * All other bits are reserved; clients must set them to zero.
+ */
+
+ u32 src_endpointype;
+/* Specifies the endpoint providing the input samples.
+ * Supported values:
+ * - 0 -- Device matrix
+ * - All other values are reserved; clients must set them to zero.
+ * Otherwise, an error is returned.
+ * The device matrix is the gateway from the tunneled Tx ports.
+ */
+
+ u32 preprocopo_id;
+/* Specifies the topology (order of processing) of preprocessing
+ * algorithms. <i>None</i> means no preprocessing.
+ * Supported values:
+ * - #ASM_STREAM_PREPROCOPO_ID_DEFAULT
+ * - #ASM_STREAM_PREPROCOPO_ID_NONE
+ *
+ * This field can also be enabled through SetParams flags.
+ */
+
+ u32 enc_cfg_id;
+/* Media configuration ID for encoded output.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ * - #ASM_MEDIA_FMT_WMA_V8
+ */
+
+ u16 bits_per_sample;
+/* Number of bits per sample processed by ASM modules.
+ * Supported values: 16 and 24 bits per sample
+ */
+
+ u16 reserved;
+/* Reserved for future use. This field must be set to zero.*/
+} __packed;
+
+#define ASM_POPP_OUTPUT_SR_NATIVE_RATE 0
+
+/* Enumeration for the maximum sampling rate at the POPP output.*/
+#define ASM_POPP_OUTPUT_SR_MAX_RATE 48000
+
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+#define ASM_STREAM_CMD_OPEN_READ_V2 0x00010D8C
+
+struct asm_stream_cmd_open_readwrite_v2 {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode flags.
+ * Supported values for bit 2:
+ * - 0 -- SR/CM change notification event is disabled.
+ * - 1 -- SR/CM change notification event is enabled. Use
+ * #ASM_BIT_MASK_SR_CM_CHANGE_NOTIFY_FLAG and
+ * #ASM_SHIFT_SR_CM_CHANGE_NOTIFY_FLAG to set or
+ * getting this flag.
+ *
+ * Supported values for bit 4:
+ * - 0 -- Return read data buffer contains all encoded frames only; it
+ * does not contain frame metadata.
+ * - 1 -- Return read data buffer contains an array of metadata and
+ * encoded frames.
+ *
+ * All other bits are reserved; clients must set them to zero.
+ */
+
+ u32 postprocopo_id;
+/* Specifies the topology (order of processing) of postprocessing
+ * algorithms. <i>None</i> means no postprocessing.
+ *
+ * Supported values:
+ * - #ASM_STREAM_POSTPROCOPO_ID_DEFAULT
+ * - #ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL
+ * - #ASM_STREAM_POSTPROCOPO_ID_NONE
+ */
+
+ u32 dec_fmt_id;
+/* Specifies the media type of the input data. PCM indicates that
+ * no decoding must be performed, e.g., this is an NT encoder
+ * session.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_ADPCM
+ * - #ASM_MEDIA_FMT_MP3
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_DOLBY_AAC
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_WMA_V10PRO_V2
+ * - #ASM_MEDIA_FMT_WMA_V9_V2
+ * - #ASM_MEDIA_FMT_AMR_WB_PLUS_V2
+ * - #ASM_MEDIA_FMT_AC3_DEC
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ */
+
+ u32 enc_cfg_id;
+/* Specifies the media type for the output of the stream. PCM
+ * indicates that no encoding must be performed, e.g., this is an NT
+ * decoder session.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2
+ * - #ASM_MEDIA_FMT_AAC_V2
+ * - #ASM_MEDIA_FMT_AMRNB_FS
+ * - #ASM_MEDIA_FMT_AMRWB_FS
+ * - #ASM_MEDIA_FMT_V13K_FS
+ * - #ASM_MEDIA_FMT_EVRC_FS
+ * - #ASM_MEDIA_FMT_EVRCB_FS
+ * - #ASM_MEDIA_FMT_EVRCWB_FS
+ * - #ASM_MEDIA_FMT_SBC
+ * - #ASM_MEDIA_FMT_G711_ALAW_FS
+ * - #ASM_MEDIA_FMT_G711_MLAW_FS
+ * - #ASM_MEDIA_FMT_G729A_FS
+ * - #ASM_MEDIA_FMT_EXAMPLE
+ * - #ASM_MEDIA_FMT_WMA_V8
+ */
+
+ u16 bits_per_sample;
+/* Number of bits per sample processed by ASM modules.
+ * Supported values: 16 and 24 bits per sample
+ */
+
+ u16 reserved;
+/* Reserved for future use. This field must be set to zero.*/
+
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_LOOPBACK_V2 0x00010D8E
+struct asm_stream_cmd_open_loopback_v2 {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode flags.
+ * Bit 0-31: reserved; client should set these bits to 0
+ */
+ u16 src_endpointype;
+ /* Endpoint type. 0 = Tx Matrix */
+ u16 sink_endpointype;
+ /* Endpoint type. 0 = Rx Matrix */
+ u32 postprocopo_id;
+/* Postprocessor topology ID. Specifies the topology of
+ * postprocessing algorithms.
+ */
+
+ u16 bits_per_sample;
+/* The number of bits per sample processed by ASM modules
+ * Supported values: 16 and 24 bits per sample
+ */
+ u16 reserved;
+/* Reserved for future use. This field must be set to zero. */
+} __packed;
+
+#define ASM_STREAM_CMD_CLOSE 0x00010BCD
+#define ASM_STREAM_CMD_FLUSH 0x00010BCE
+
+
+#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
+#define ASM_STREAM_CMD_SET_PP_PARAMS_V2 0x00010DA1
+
+struct asm_stream_cmd_set_pp_params_v2 {
+ u32 data_payload_addr_lsw;
+/* LSW of parameter data payload address. Supported values: any. */
+ u32 data_payload_addr_msw;
+/* MSW of Parameter data payload address. Supported values: any.
+ * - Must be set to zero for in-band data.
+ * - In the case of 32 bit Shared memory address, msw field must be
+ * - set to zero.
+ * - In the case of 36 bit shared memory address, bit 31 to bit 4 of
+ * msw
+ *
+ * - must be set to zero.
+ */
+ u32 mem_map_handle;
+/* Supported Values: Any.
+* memory map handle returned by DSP through
+* ASM_CMD_SHARED_MEM_MAP_REGIONS
+* command.
+* if mmhandle is NULL, the ParamData payloads are within the
+* message payload (in-band).
+* If mmhandle is non-NULL, the ParamData payloads begin at the
+* address specified in the address msw and lsw (out-of-band).
+*/
+
+ u32 data_payload_size;
+/* Size in bytes of the variable payload accompanying the
+message, or in shared memory. This field is used for parsing the
+parameter payload. */
+
+} __packed;
+
+
+struct asm_stream_param_data_v2 {
+ u32 module_id;
+ /* Unique module ID. */
+
+ u32 param_id;
+ /* Unique parameter ID. */
+
+ u16 param_size;
+/* Data size of the param_id/module_id combination. This is
+ * a multiple of 4 bytes.
+ */
+
+ u16 reserved;
+/* Reserved for future enhancements. This field must be set to
+ * zero.
+ */
+
+} __packed;
+
+#define ASM_STREAM_CMD_GET_PP_PARAMS_V2 0x00010DA2
+
+struct asm_stream_cmd_get_pp_params_v2 {
+ u32 data_payload_addr_lsw;
+ /* LSW of the parameter data payload address. */
+ u32 data_payload_addr_msw;
+/* MSW of the parameter data payload address.
+ * - Size of the shared memory, if specified, shall be large enough
+ * to contain the whole ParamData payload, including Module ID,
+ * Param ID, Param Size, and Param Values
+ * - Must be set to zero for in-band data
+ * - In the case of 32 bit Shared memory address, msw field must be
+ * set to zero.
+ * - In the case of 36 bit shared memory address, bit 31 to bit 4 of
+ * msw must be set to zero.
+ */
+
+ u32 mem_map_handle;
+/* Supported Values: Any.
+* memory map handle returned by DSP through ASM_CMD_SHARED_MEM_MAP_REGIONS
+* command.
+* if mmhandle is NULL, the ParamData payloads in the ACK are within the
+* message payload (in-band).
+* If mmhandle is non-NULL, the ParamData payloads in the ACK begin at the
+* address specified in the address msw and lsw.
+* (out-of-band).
+*/
+
+ u32 module_id;
+ /* Unique module ID. */
+
+ u32 param_id;
+ /* Unique parameter ID. */
+
+ u16 param_max_size;
+/* Maximum data size of the module_id/param_id combination. This
+ * is a multiple of 4 bytes.
+ */
+
+
+ u16 reserved;
+/* Reserved for backward compatibility. Clients must set this
+* field to zero.
+*/
+
+} __packed;
+
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
+
+#define ASM_PARAM_ID_ENCDEC_BITRATE 0x00010C13
+
+struct asm_bitrate_param {
+ u32 bitrate;
+/* Maximum supported bitrate. Only the AAC encoder is supported.*/
+
+} __packed;
+
+#define ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2 0x00010DA3
+#define ASM_PARAM_ID_AAC_SBR_PS_FLAG 0x00010C63
+
+/* Flag to turn off both SBR and PS processing, if they are
+ * present in the bitstream.
+ */
+
+#define ASM_AAC_SBR_OFF_PS_OFF (2)
+
+/* Flag to turn on SBR but turn off PS processing,if they are
+ * present in the bitstream.
+ */
+
+#define ASM_AAC_SBR_ON_PS_OFF (1)
+
+/* Flag to turn on both SBR and PS processing, if they are
+ * present in the bitstream (default behavior).
+ */
+
+
+#define ASM_AAC_SBR_ON_PS_ON (0)
+
+/* Structure for an AAC SBR PS processing flag. */
+
+/* Payload of the #ASM_PARAM_ID_AAC_SBR_PS_FLAG parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+struct asm_aac_sbr_ps_flag_param {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+
+ u32 sbr_ps_flag;
+/* Control parameter to enable or disable SBR/PS processing in
+ * the AAC bitstream. Use the following macros to set this field:
+ * - #ASM_AAC_SBR_OFF_PS_OFF -- Turn off both SBR and PS
+ * processing, if they are present in the bitstream.
+ * - #ASM_AAC_SBR_ON_PS_OFF -- Turn on SBR processing, but not PS
+ * processing, if they are present in the bitstream.
+ * - #ASM_AAC_SBR_ON_PS_ON -- Turn on both SBR and PS processing,
+ * if they are present in the bitstream (default behavior).
+ * - All other values are invalid.
+ * Changes are applied to the next decoded frame.
+ */
+} __packed;
+
+#define ASM_PARAM_ID_AAC_DUAL_MONO_MAPPING 0x00010C64
+
+/* First single channel element in a dual mono bitstream.*/
+#define ASM_AAC_DUAL_MONO_MAP_SCE_1 (1)
+
+/* Second single channel element in a dual mono bitstream.*/
+#define ASM_AAC_DUAL_MONO_MAP_SCE_2 (2)
+
+/* Structure for AAC decoder dual mono channel mapping. */
+
+
+struct asm_aac_dual_mono_mapping_param {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u16 left_channel_sce;
+ u16 right_channel_sce;
+
+} __packed;
+
+#define ASM_STREAM_CMDRSP_GET_PP_PARAMS_V2 0x00010DA4
+
+struct asm_stream_cmdrsp_get_pp_params_v2 {
+ u32 status;
+} __packed;
+
+#define ASM_PARAM_ID_AC3_KARAOKE_MODE 0x00010D73
+
+/* Enumeration for both vocals in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_NO_VOCAL (0)
+
+/* Enumeration for only the left vocal in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_LEFT_VOCAL (1)
+
+/* Enumeration for only the right vocal in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_RIGHT_VOCAL (2)
+
+/* Enumeration for both vocal channels in a karaoke stream.*/
+#define AC3_KARAOKE_MODE_BOTH_VOCAL (3)
+#define ASM_PARAM_ID_AC3_DRC_MODE 0x00010D74
+/* Enumeration for the Custom Analog mode.*/
+#define AC3_DRC_MODE_CUSTOM_ANALOG (0)
+
+/* Enumeration for the Custom Digital mode.*/
+#define AC3_DRC_MODE_CUSTOM_DIGITAL (1)
+/* Enumeration for the Line Out mode (light compression).*/
+#define AC3_DRC_MODE_LINE_OUT (2)
+
+/* Enumeration for the RF remodulation mode (heavy compression).*/
+#define AC3_DRC_MODE_RF_REMOD (3)
+#define ASM_PARAM_ID_AC3_DUAL_MONO_MODE 0x00010D75
+
+/* Enumeration for playing dual mono in stereo mode.*/
+#define AC3_DUAL_MONO_MODE_STEREO (0)
+
+/* Enumeration for playing left mono.*/
+#define AC3_DUAL_MONO_MODE_LEFT_MONO (1)
+
+/* Enumeration for playing right mono.*/
+#define AC3_DUAL_MONO_MODE_RIGHT_MONO (2)
+
+/* Enumeration for mixing both dual mono channels and playing them.*/
+#define AC3_DUAL_MONO_MODE_MIXED_MONO (3)
+#define ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE 0x00010D76
+
+/* Enumeration for using the Downmix mode indicated in the bitstream. */
+
+#define AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT (0)
+
+/* Enumeration for Surround Compatible mode (preserves the
+ * surround information).
+ */
+
+#define AC3_STEREO_DOWNMIX_MODE_LT_RT (1)
+/* Enumeration for Mono Compatible mode (if the output is to be
+ * further downmixed to mono).
+ */
+
+#define AC3_STEREO_DOWNMIX_MODE_LO_RO (2)
+
+/* ID of the AC3 PCM scale factor parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+#define ASM_PARAM_ID_AC3_PCM_SCALEFACTOR 0x00010D78
+
+/* ID of the AC3 DRC boost scale factor parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+#define ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR 0x00010D79
+
+/* ID of the AC3 DRC cut scale factor parameter in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+#define ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR 0x00010D7A
+
+/* Structure for AC3 Generic Parameter. */
+
+/* Payload of the AC3 parameters in the
+ * #ASM_STREAM_CMD_SET_ENCDEC_PARAM command.
+ */
+struct asm_ac3_generic_param {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u32 generic_parameter;
+/* AC3 generic parameter. Select from one of the following
+ * possible values.
+ *
+ * For #ASM_PARAM_ID_AC3_KARAOKE_MODE, supported values are:
+ * - AC3_KARAOKE_MODE_NO_VOCAL
+ * - AC3_KARAOKE_MODE_LEFT_VOCAL
+ * - AC3_KARAOKE_MODE_RIGHT_VOCAL
+ * - AC3_KARAOKE_MODE_BOTH_VOCAL
+ *
+ * For #ASM_PARAM_ID_AC3_DRC_MODE, supported values are:
+ * - AC3_DRC_MODE_CUSTOM_ANALOG
+ * - AC3_DRC_MODE_CUSTOM_DIGITAL
+ * - AC3_DRC_MODE_LINE_OUT
+ * - AC3_DRC_MODE_RF_REMOD
+ *
+ * For #ASM_PARAM_ID_AC3_DUAL_MONO_MODE, supported values are:
+ * - AC3_DUAL_MONO_MODE_STEREO
+ * - AC3_DUAL_MONO_MODE_LEFT_MONO
+ * - AC3_DUAL_MONO_MODE_RIGHT_MONO
+ * - AC3_DUAL_MONO_MODE_MIXED_MONO
+ *
+ * For #ASM_PARAM_ID_AC3_STEREO_DOWNMIX_MODE, supported values are:
+ * - AC3_STEREO_DOWNMIX_MODE_AUTO_DETECT
+ * - AC3_STEREO_DOWNMIX_MODE_LT_RT
+ * - AC3_STEREO_DOWNMIX_MODE_LO_RO
+ *
+ * For #ASM_PARAM_ID_AC3_PCM_SCALEFACTOR, supported values are
+ * 0 to 1 in Q31 format.
+ *
+ * For #ASM_PARAM_ID_AC3_DRC_BOOST_SCALEFACTOR, supported values are
+ * 0 to 1 in Q31 format.
+ *
+ * For #ASM_PARAM_ID_AC3_DRC_CUT_SCALEFACTOR, supported values are
+ * 0 to 1 in Q31 format.
+ */
+} __packed;
+
+/* Enumeration for Raw mode (no downmixing), which specifies
+ * that all channels in the bitstream are to be played out as is
+ * without any downmixing. (Default)
+ */
+
+#define WMAPRO_CHANNEL_MASK_RAW (-1)
+
+/* Enumeration for setting the channel mask to 0. The 7.1 mode
+ * (Home Theater) is assigned.
+ */
+
+
+#define WMAPRO_CHANNEL_MASK_ZERO 0x0000
+
+/* Speaker layout mask for one channel (Home Theater, mono).
+ * - Speaker front center
+ */
+#define WMAPRO_CHANNEL_MASK_1_C 0x0004
+
+/* Speaker layout mask for two channels (Home Theater, stereo).
+ * - Speaker front left
+ * - Speaker front right
+ */
+#define WMAPRO_CHANNEL_MASK_2_L_R 0x0003
+
+/* Speaker layout mask for three channels (Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ */
+#define WMAPRO_CHANNEL_MASK_3_L_C_R 0x0007
+
+/* Speaker layout mask for two channels (stereo).
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_2_Bl_Br 0x0030
+
+/* Speaker layout mask for four channels.
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker back left
+ * - Speaker back right
+*/
+#define WMAPRO_CHANNEL_MASK_4_L_R_Bl_Br 0x0033
+
+/* Speaker layout mask for four channels (Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back center
+*/
+#define WMAPRO_CHANNEL_MASK_4_L_R_C_Bc_HT 0x0107
+/* Speaker layout mask for five channels.
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_5_L_C_R_Bl_Br 0x0037
+
+/* Speaker layout mask for five channels (5 mode, Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker side left
+ * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_5_L_C_R_Sl_Sr_HT 0x0607
+/* Speaker layout mask for six channels (5.1 mode).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker back left
+ * - Speaker back right
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_SLF 0x003F
+/* Speaker layout mask for six channels (5.1 mode, Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker side left
+ * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_SLF_HT 0x060F
+/* Speaker layout mask for six channels (5.1 mode, no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker back center
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Bl_Br_Bc 0x0137
+/* Speaker layout mask for six channels (5.1 mode, Home Theater,
+ * no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back center
+ * - Speaker side left
+ * - Speaker side right
+ */
+#define WMAPRO_CHANNEL_MASK_5DOT1_L_C_R_Sl_Sr_Bc_HT 0x0707
+
+/* Speaker layout mask for seven channels (6.1 mode).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker back center
+ */
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_Bc_SLF 0x013F
+
+/* Speaker layout mask for seven channels (6.1 mode, Home
+ * Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker low frequency
+ * - Speaker back center
+ * - Speaker side left
+ * - Speaker side right
+*/
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_Bc_SLF_HT 0x070F
+
+/* Speaker layout mask for seven channels (6.1 mode, no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker front left of center
+ * - Speaker front right of center
+*/
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Bl_Br_SFLOC_SFROC 0x00F7
+
+/* Speaker layout mask for seven channels (6.1 mode, Home
+ * Theater, no LFE).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker side left
+ * - Speaker side right
+ * - Speaker front left of center
+ * - Speaker front right of center
+*/
+#define WMAPRO_CHANNEL_MASK_6DOT1_L_C_R_Sl_Sr_SFLOC_SFROC_HT 0x0637
+
+/* Speaker layout mask for eight channels (7.1 mode).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker back left
+ * - Speaker back right
+ * - Speaker low frequency
+ * - Speaker front left of center
+ * - Speaker front right of center
+ */
+#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Bl_Br_SLF_SFLOC_SFROC \
+ 0x00FF
+
+/* Speaker layout mask for eight channels (7.1 mode, Home Theater).
+ * - Speaker front left
+ * - Speaker front right
+ * - Speaker front center
+ * - Speaker side left
+ * - Speaker side right
+ * - Speaker low frequency
+ * - Speaker front left of center
+ * - Speaker front right of center
+ *
+*/
+#define WMAPRO_CHANNEL_MASK_7DOT1_L_C_R_Sl_Sr_SLF_SFLOC_SFROC_HT \
+ 0x063F
+
+#define ASM_PARAM_ID_DEC_OUTPUT_CHAN_MAP 0x00010D82
+
+/* Maximum number of decoder output channels.*/
+#define MAX_CHAN_MAP_CHANNELS 16
+
+/* Structure for decoder output channel mapping. */
+
+/* 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 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ u32 num_channels;
+/* Number of decoder output channels.
+ * Supported values: 0 to #MAX_CHAN_MAP_CHANNELS
+ *
+ * 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];
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED 0x00010D84
+
+/* Bitmask for the IEC 61937 enable flag.*/
+#define ASM_BIT_MASK_IEC_61937_STREAM_FLAG (0x00000001UL)
+
+/* Shift value for the IEC 61937 enable flag.*/
+#define ASM_SHIFT_IEC_61937_STREAM_FLAG 0
+
+/* Bitmask for the IEC 60958 enable flag.*/
+#define ASM_BIT_MASK_IEC_60958_STREAM_FLAG (0x00000002UL)
+
+/* Shift value for the IEC 60958 enable flag.*/
+#define ASM_SHIFT_IEC_60958_STREAM_FLAG 1
+
+/* Payload format for open write compressed comand */
+
+/* Payload format for the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED
+ * comand, which opens a stream for a given session ID and stream ID
+ * to be rendered in the compressed format.
+ */
+
+struct asm_stream_cmd_open_write_compressed {
+ struct apr_hdr hdr;
+ u32 flags;
+/* Mode flags that configure the stream for a specific format.
+ * Supported values:
+ * - Bit 0 -- IEC 61937 compatibility
+ * - 0 -- Stream is not in IEC 61937 format
+ * - 1 -- Stream is in IEC 61937 format
+ * - Bit 1 -- IEC 60958 compatibility
+ * - 0 -- Stream is not in IEC 60958 format
+ * - 1 -- Stream is in IEC 60958 format
+ * - Bits 2 to 31 -- 0 (Reserved)
+ *
+ * For the same stream, bit 0 cannot be set to 0 and bit 1 cannot
+ * be set to 1. A compressed stream connot have IEC 60958
+ * packetization applied without IEC 61937 packetization.
+ * @note1hang Currently, IEC 60958 packetized input streams are not
+ * supported.
+ */
+
+
+ u32 fmt_id;
+/* Specifies the media type of the HDMI stream to be opened.
+ * Supported values:
+ * - #ASM_MEDIA_FMT_AC3_DEC
+ * - #ASM_MEDIA_FMT_EAC3_DEC
+ * - #ASM_MEDIA_FMT_DTS
+ * - #ASM_MEDIA_FMT_ATRAC
+ * - #ASM_MEDIA_FMT_MAT
+ *
+ * @note1hang This field must be set to a valid media type even if
+ * IEC 61937 packetization is not performed by the aDSP.
+ */
+
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_READ_COMPRESSED 0x00010D95
+
+struct asm_stream_cmd_open_read_compressed {
+ struct apr_hdr hdr;
+ u32 mode_flags;
+/* Mode flags that indicate whether meta information per encoded
+ * frame is to be provided.
+ * Supported values for bit 4:
+ * - 0 -- Return data buffer contains all encoded frames only; it does
+ * not contain frame metadata.
+ * - 1 -- Return data buffer contains an array of metadata and encoded
+ * frames.
+ * - Use #ASM_BIT_MASK_META_INFO_FLAG to set the bitmask and
+ * #ASM_SHIFT_META_INFO_FLAG to set the shift value for this bit.
+ * All other bits are reserved; clients must set them to zero.
+ */
+
+ u32 frames_per_buf;
+/* Indicates the number of frames that need to be returned per
+ * read buffer
+ * Supported values: should be greater than 0
+ */
+
+} __packed;
+
+/* adsp_asm_stream_commands.h*/
+
+
+/* adsp_asm_api.h (no changes)*/
+#define ASM_STREAM_POSTPROCOPO_ID_DEFAULT \
+ 0x00010BE4
+#define ASM_STREAM_POSTPROCOPO_ID_PEAKMETER \
+ 0x00010D83
+#define ASM_STREAM_POSTPROCOPO_ID_NONE \
+ 0x00010C68
+#define ASM_STREAM_POSTPROCOPO_ID_MCH_PEAK_VOL \
+ 0x00010D8B
+#define ASM_STREAM_PREPROCOPO_ID_DEFAULT \
+ ASM_STREAM_POSTPROCOPO_ID_DEFAULT
+#define ASM_STREAM_PREPROCOPO_ID_NONE \
+ ASM_STREAM_POSTPROCOPO_ID_NONE
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_NONE_AUDIO_COPP \
+ 0x00010312
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP \
+ 0x00010313
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP \
+ 0x00010314
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP\
+ 0x00010704
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MONO_AUDIO_COPP_MBDRCV2\
+ 0x0001070D
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_AUDIO_COPP_MBDRCV2\
+ 0x0001070E
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_STEREO_IIR_AUDIO_COPP_MBDRCV2\
+ 0x0001070F
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_SPEAKER_MCH_PEAK_VOL \
+ 0x0001031B
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_MONO_AUDIO_COPP 0x00010315
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_MIC_STEREO_AUDIO_COPP 0x00010316
+#define AUDPROC_COPPOPOLOGY_ID_MCHAN_IIR_AUDIO 0x00010715
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_DEFAULT_AUDIO_COPP 0x00010BE3
+#define ADM_CMD_COPP_OPENOPOLOGY_ID_PEAKMETER_AUDIO_COPP 0x00010317
+#define AUDPROC_MODULE_ID_AIG 0x00010716
+#define AUDPROC_PARAM_ID_AIG_ENABLE 0x00010717
+#define AUDPROC_PARAM_ID_AIG_CONFIG 0x00010718
+
+struct Audio_AigParam {
+ uint16_t mode;
+/*< Mode word for enabling AIG/SIG mode .
+ * Byte offset: 0
+ */
+ int16_t staticGainL16Q12;
+/*< Static input gain when aigMode is set to 1.
+ * Byte offset: 2
+ */
+ int16_t initialGainDBL16Q7;
+/*<Initial value that the adaptive gain update starts from dB
+ * Q7 Byte offset: 4
+ */
+ int16_t idealRMSDBL16Q7;
+/*<Average RMS level that AIG attempts to achieve Q8.7
+ * Byte offset: 6
+ */
+ int32_t noiseGateL32;
+/*Threshold below which signal is considered as noise and AIG
+ * Byte offset: 8
+ */
+ int32_t minGainL32Q15;
+/*Minimum gain that can be provided by AIG Q16.15
+ * Byte offset: 12
+ */
+ int32_t maxGainL32Q15;
+/*Maximum gain that can be provided by AIG Q16.15
+ * Byte offset: 16
+ */
+ uint32_t gainAtRtUL32Q31;
+/*Attack/release time for AIG update Q1.31
+ * Byte offset: 20
+ */
+ uint32_t longGainAtRtUL32Q31;
+/*Long attack/release time while updating gain for
+ * noise/silence Q1.31 Byte offset: 24
+ */
+
+ uint32_t rmsTavUL32Q32;
+/* RMS smoothing time constant used for long-term RMS estimate
+ * Q0.32 Byte offset: 28
+ */
+
+ uint32_t gainUpdateStartTimMsUL32Q0;
+/* The waiting time before which AIG starts to apply adaptive
+ * gain update Q32.0 Byte offset: 32
+ */
+
+} __packed;
+
+
+#define ADM_MODULE_ID_EANS 0x00010C4A
+#define ADM_PARAM_ID_EANS_ENABLE 0x00010C4B
+#define ADM_PARAM_ID_EANS_PARAMS 0x00010C4C
+
+struct adm_eans_enable {
+
+ uint32_t enable_flag;
+/*< Specifies whether EANS is disabled (0) or enabled
+ * (nonzero).
+ * This is supported only for sampling rates of 8, 12, 16, 24, 32,
+ * and 48 kHz. It is not supported for sampling rates of 11.025,
+ * 22.05, or 44.1 kHz.
+ */
+
+} __packed;
+
+
+struct adm_eans_params {
+ int16_t eans_mode;
+/*< Mode word for enabling/disabling submodules.
+ * Byte offset: 0
+ */
+
+ int16_t eans_input_gain;
+/*< Q2.13 input gain to the EANS module.
+ * Byte offset: 2
+ */
+
+ int16_t eans_output_gain;
+/*< Q2.13 output gain to the EANS module.
+ * Byte offset: 4
+ */
+
+ int16_t eansarget_ns;
+/*< Target noise suppression level in dB.
+ * Byte offset: 6
+ */
+
+ int16_t eans_s_alpha;
+/*< Q3.12 over-subtraction factor for stationary noise
+ * suppression.
+ * Byte offset: 8
+ */
+
+ int16_t eans_n_alpha;
+/* < Q3.12 over-subtraction factor for nonstationary noise
+ * suppression.
+ * Byte offset: 10
+ */
+
+ int16_t eans_n_alphamax;
+/*< Q3.12 maximum over-subtraction factor for nonstationary
+ * noise suppression.
+ * Byte offset: 12
+ */
+ int16_t eans_e_alpha;
+/*< Q15 scaling factor for excess noise suppression.
+ * Byte offset: 14
+ */
+
+ int16_t eans_ns_snrmax;
+/*< Upper boundary in dB for SNR estimation.
+ * Byte offset: 16
+ */
+
+ int16_t eans_sns_block;
+/*< Quarter block size for stationary noise suppression.
+ * Byte offset: 18
+ */
+
+ int16_t eans_ns_i;
+/*< Initialization block size for noise suppression.
+ * Byte offset: 20
+ */
+ int16_t eans_np_scale;
+/*< Power scale factor for nonstationary noise update.
+ * Byte offset: 22
+ */
+
+ int16_t eans_n_lambda;
+/*< Smoothing factor for higher level nonstationary noise
+ * update.
+ * Byte offset: 24
+ */
+
+ int16_t eans_n_lambdaf;
+/*< Medium averaging factor for noise update.
+ * Byte offset: 26
+ */
+
+ int16_t eans_gs_bias;
+/*< Bias factor in dB for gain calculation.
+ * Byte offset: 28
+ */
+
+ int16_t eans_gs_max;
+/*< SNR lower boundary in dB for aggressive gain calculation.
+ * Byte offset: 30
+ */
+
+ int16_t eans_s_alpha_hb;
+/*< Q3.12 over-subtraction factor for high-band stationary
+ * noise suppression.
+ * Byte offset: 32
+ */
+
+ int16_t eans_n_alphamax_hb;
+/*< Q3.12 maximum over-subtraction factor for high-band
+ * nonstationary noise suppression.
+ * Byte offset: 34
+ */
+
+ int16_t eans_e_alpha_hb;
+/*< Q15 scaling factor for high-band excess noise suppression.
+ * Byte offset: 36
+ */
+
+ int16_t eans_n_lambda0;
+/*< Smoothing factor for nonstationary noise update during
+ * speech activity.
+ * Byte offset: 38
+ */
+
+ int16_t thresh;
+/*< Threshold for generating a binary VAD decision.
+ * Byte offset: 40
+ */
+
+ int16_t pwr_scale;
+/*< Indirect lower boundary of the noise level estimate.
+ * Byte offset: 42
+ */
+
+ int16_t hangover_max;
+/*< Avoids mid-speech clipping and reliably detects weak speech
+ * bursts at the end of speech activity.
+ * Byte offset: 44
+ */
+
+ int16_t alpha_snr;
+/*< Controls responsiveness of the VAD.
+ * Byte offset: 46
+ */
+
+ int16_t snr_diff_max;
+/*< Maximum SNR difference. Decreasing this parameter value may
+ * help in making correct decisions during abrupt changes; however,
+ * decreasing too much may increase false alarms during long
+ * pauses/silences.
+ * Byte offset: 48
+ */
+
+ int16_t snr_diff_min;
+/*< Minimum SNR difference. Decreasing this parameter value may
+ * help in making correct decisions during abrupt changes; however,
+ * decreasing too much may increase false alarms during long
+ * pauses/silences.
+ * Byte offset: 50
+ */
+
+ int16_t init_length;
+/*< Defines the number of frames for which a noise level
+ * estimate is set to a fixed value.
+ * Byte offset: 52
+ */
+
+ int16_t max_val;
+/*< Defines the upper limit of the noise level.
+ * Byte offset: 54
+ */
+
+ int16_t init_bound;
+/*< Defines the initial bounding value for the noise level
+ * estimate. This is used during the initial segment defined by the
+ * init_length parameter.
+ * Byte offset: 56
+ */
+
+ int16_t reset_bound;
+/*< Reset boundary for noise tracking.
+ * Byte offset: 58
+ */
+
+ int16_t avar_scale;
+/*< Defines the bias factor in noise estimation.
+ * Byte offset: 60
+ */
+
+ int16_t sub_nc;
+/*< Defines the window length for noise estimation.
+ * Byte offset: 62
+ */
+
+ int16_t spow_min;
+/*< Defines the minimum signal power required to update the
+ * boundaries for the noise floor estimate.
+ * Byte offset: 64
+ */
+
+ int16_t eans_gs_fast;
+/*< Fast smoothing factor for postprocessor gain.
+ * Byte offset: 66
+ */
+
+ int16_t eans_gs_med;
+/*< Medium smoothing factor for postprocessor gain.
+ * Byte offset: 68
+ */
+
+ int16_t eans_gs_slow;
+/*< Slow smoothing factor for postprocessor gain.
+ * Byte offset: 70
+ */
+
+ int16_t eans_swb_salpha;
+/*< Q3.12 super wideband aggressiveness factor for stationary
+ * noise suppression.
+ * Byte offset: 72
+ */
+
+ int16_t eans_swb_nalpha;
+/*< Q3.12 super wideband aggressiveness factor for
+ * nonstationary noise suppression.
+ * Byte offset: 74
+ */
+} __packed;
+#define ADM_MODULE_IDX_MIC_GAIN_CTRL 0x00010C35
+
+/* @addtogroup audio_pp_param_ids
+ * ID of the Tx mic gain control parameter used by the
+ * #ADM_MODULE_IDX_MIC_GAIN_CTRL module.
+ * @messagepayload
+ * @structure{admx_mic_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_MIC_GAIN.tex}
+ */
+#define ADM_PARAM_IDX_MIC_GAIN 0x00010C36
+
+/* Structure for a Tx mic gain parameter for the mic gain
+ * control module.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_IDX_MIC_GAIN parameter in the
+ * Tx Mic Gain Control module.
+ */
+struct admx_mic_gain {
+ uint16_t tx_mic_gain;
+ /*< Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero. */
+} __packed;
+
+/* end_addtogroup audio_pp_param_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the Rx Codec Gain Control module.
+ *
+ * This module supports the following parameter ID:
+ * - #ADM_PARAM_ID_RX_CODEC_GAIN
+ */
+#define ADM_MODULE_ID_RX_CODEC_GAIN_CTRL 0x00010C37
+
+/* @addtogroup audio_pp_param_ids
+ * ID of the Rx codec gain control parameter used by the
+ * #ADM_MODULE_ID_RX_CODEC_GAIN_CTRL module.
+ *
+ * @messagepayload
+ * @structure{adm_rx_codec_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_RX_CODEC_GAIN.tex}
+*/
+#define ADM_PARAM_ID_RX_CODEC_GAIN 0x00010C38
+
+/* Structure for the Rx common codec gain control module. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_RX_CODEC_GAIN parameter
+ * in the Rx Codec Gain Control module.
+ */
+
+
+struct adm_rx_codec_gain {
+ uint16_t rx_codec_gain;
+ /*< Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+/* end_addtogroup audio_pp_param_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the HPF Tuning Filter module on the Tx path.
+ * This module supports the following parameter IDs:
+ * - #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG
+ * - #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN
+ * - #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS
+ */
+#define ADM_MODULE_ID_HPF_IIRX_FILTER 0x00010C3D
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the Tx HPF IIR filter enable parameter used by the
+ * #ADM_MODULE_ID_HPF_IIRX_FILTER module.
+ * @parspace Message payload
+ * @structure{adm_hpfx_iir_filter_enable_cfg}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG.tex}
+ */
+#define ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG 0x00010C3E
+
+/* ID of the Tx HPF IIR filter pregain parameter used by the
+ * #ADM_MODULE_ID_HPF_IIRX_FILTER module.
+ * @parspace Message payload
+ * @structure{adm_hpfx_iir_filter_pre_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN.tex}
+ */
+#define ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN 0x00010C3F
+
+/* ID of the Tx HPF IIR filter configuration parameters used by the
+ * #ADM_MODULE_ID_HPF_IIRX_FILTER module.
+ * @parspace Message payload
+ * @structure{adm_hpfx_iir_filter_cfg_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PA
+ * RAMS.tex}
+ */
+#define ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS 0x00010C40
+
+/* Structure for enabling a configuration parameter for
+ * the HPF IIR tuning filter module on the Tx path.
+ */
+
+/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_ENABLE_CONFIG
+ * parameter in the Tx path HPF Tuning Filter module.
+ */
+struct adm_hpfx_iir_filter_enable_cfg {
+ uint32_t enable_flag;
+/*< Specifies whether the HPF tuning filter is disabled (0) or
+ * enabled (nonzero).
+ */
+} __packed;
+
+
+/* Structure for the pregain parameter for the HPF
+ IIR tuning filter module on the Tx path. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_PRE_GAIN parameter
+ * in the Tx path HPF Tuning Filter module.
+ */
+struct adm_hpfx_iir_filter_pre_gain {
+ uint16_t pre_gain;
+ /*< Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+
+/* Structure for the configuration parameter for the
+ HPF IIR tuning filter module on the Tx path. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_HPF_IIRX_FILTER_CONFIG_PARAMS
+ * parameters in the Tx path HPF Tuning Filter module. \n
+ * \n
+ * This structure is followed by tuning filter coefficients as follows: \n
+ * - Sequence of int32_t FilterCoeffs.
+ * Each band has five coefficients, each in int32_t format in the order of
+ * b0, b1, b2, a1, a2.
+ * - Sequence of int16_t NumShiftFactor.
+ * One int16_t per band. The numerator shift factor is related to the Q
+ * factor of the filter coefficients.
+ * - Sequence of uint16_t PanSetting.
+ * One uint16_t for each band to indicate application of the filter to
+ * left (0), right (1), or both (2) channels.
+ */
+struct adm_hpfx_iir_filter_cfg_params {
+ uint16_t num_biquad_stages;
+/*< Number of bands.
+ * Supported values: 0 to 20
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @addtogroup audio_pp_module_ids */
+/* ID of the Tx path IIR Tuning Filter module.
+ * This module supports the following parameter IDs:
+ * - #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG
+ */
+#define ADM_MODULE_IDX_IIR_FILTER 0x00010C41
+
+/* ID of the Rx path IIR Tuning Filter module for the left channel.
+ * The parameter IDs of the IIR tuning filter module
+ * (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the left IIR Rx tuning
+ * filter.
+ *
+ * Pan parameters are not required for this per-channel IIR filter; the pan
+ * parameters are ignored by this module.
+ */
+#define ADM_MODULE_ID_LEFT_IIRUNING_FILTER 0x00010705
+
+/* ID of the the Rx path IIR Tuning Filter module for the right
+ * channel.
+ * The parameter IDs of the IIR tuning filter module
+ * (#ASM_MODULE_ID_IIRUNING_FILTER) are used for the right IIR Rx
+ * tuning filter.
+ *
+ * Pan parameters are not required for this per-channel IIR filter;
+ * the pan parameters are ignored by this module.
+ */
+#define ADM_MODULE_ID_RIGHT_IIRUNING_FILTER 0x00010706
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @addtogroup audio_pp_param_ids */
+
+/* ID of the Tx IIR filter enable parameter used by the
+ * #ADM_MODULE_IDX_IIR_FILTER module.
+ * @parspace Message payload
+ * @structure{admx_iir_filter_enable_cfg}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG.tex}
+ */
+#define ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG 0x00010C42
+
+/* ID of the Tx IIR filter pregain parameter used by the
+ * #ADM_MODULE_IDX_IIR_FILTER module.
+ * @parspace Message payload
+ * @structure{admx_iir_filter_pre_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN.tex}
+ */
+#define ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN 0x00010C43
+
+/* ID of the Tx IIR filter configuration parameters used by the
+ * #ADM_MODULE_IDX_IIR_FILTER module.
+ * @parspace Message payload
+ * @structure{admx_iir_filter_cfg_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS.tex}
+ */
+#define ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS 0x00010C44
+
+/* Structure for enabling the configuration parameter for the
+ * IIR filter module on the Tx path.
+ */
+
+/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_ENABLE_CONFIG
+ * parameter in the Tx Path IIR Tuning Filter module.
+ */
+
+struct admx_iir_filter_enable_cfg {
+ uint32_t enable_flag;
+/*< Specifies whether the IIR tuning filter is disabled (0) or
+ * enabled (nonzero).
+ */
+
+} __packed;
+
+
+/* Structure for the pregain parameter for the
+ * IIR filter module on the Tx path.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_PRE_GAIN
+ * parameter in the Tx Path IIR Tuning Filter module.
+ */
+
+struct admx_iir_filter_pre_gain {
+ uint16_t pre_gain;
+ /*< Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+
+/* Structure for the configuration parameter for the
+ * IIR filter module on the Tx path.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_IDX_IIR_FILTER_CONFIG_PARAMS
+ * parameter in the Tx Path IIR Tuning Filter module. \n
+ * \n
+ * This structure is followed by the HPF IIR filter coefficients on
+ * the Tx path as follows: \n
+ * - Sequence of int32_t ulFilterCoeffs. Each band has five
+ * coefficients, each in int32_t format in the order of b0, b1, b2,
+ * a1, a2.
+ * - Sequence of int16_t sNumShiftFactor. One int16_t per band. The
+ * numerator shift factor is related to the Q factor of the filter
+ * coefficients.
+ * - Sequence of uint16_t usPanSetting. One uint16_t for each band
+ * to indicate if the filter is applied to left (0), right (1), or
+ * both (2) channels.
+ */
+struct admx_iir_filter_cfg_params {
+ uint16_t num_biquad_stages;
+/*< Number of bands.
+ * Supported values: 0 to 20
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the QEnsemble module.
+ * This module supports the following parameter IDs:
+ * - #ADM_PARAM_ID_QENSEMBLE_ENABLE
+ * - #ADM_PARAM_ID_QENSEMBLE_BACKGAIN
+ * - #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE
+ */
+#define ADM_MODULE_ID_QENSEMBLE 0x00010C59
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the QEnsemble enable parameter used by the
+ * #ADM_MODULE_ID_QENSEMBLE module.
+ * @messagepayload
+ * @structure{adm_qensemble_enable}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_ENABLE.tex}
+ */
+#define ADM_PARAM_ID_QENSEMBLE_ENABLE 0x00010C60
+
+/* ID of the QEnsemble back gain parameter used by the
+ * #ADM_MODULE_ID_QENSEMBLE module.
+ * @messagepayload
+ * @structure{adm_qensemble_param_backgain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_BACKGAIN.tex}
+ */
+#define ADM_PARAM_ID_QENSEMBLE_BACKGAIN 0x00010C61
+
+/* ID of the QEnsemble new angle parameter used by the
+ * #ADM_MODULE_ID_QENSEMBLE module.
+ * @messagepayload
+ * @structure{adm_qensemble_param_set_new_angle}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE.tex}
+ */
+#define ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE 0x00010C62
+
+/* Structure for enabling the configuration parameter for the
+ * QEnsemble module.
+ */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_ENABLE
+ * parameter used by the QEnsemble module.
+ */
+struct adm_qensemble_enable {
+ uint32_t enable_flag;
+/*< Specifies whether the QEnsemble module is disabled (0) or enabled
+ * (nonzero).
+ */
+} __packed;
+
+
+/* Structure for the background gain for the QEnsemble module. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_BACKGAIN
+ * parameter used by
+ * the QEnsemble module.
+ */
+struct adm_qensemble_param_backgain {
+ int16_t back_gain;
+/*< Linear gain in Q15 format.
+ * Supported values: 0 to 32767
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+/* Structure for setting a new angle for the QEnsemble module. */
+
+
+/* @brief Payload of the #ADM_PARAM_ID_QENSEMBLE_SET_NEW_ANGLE
+ * parameter used
+ * by the QEnsemble module.
+ */
+struct adm_qensemble_param_set_new_angle {
+ int16_t new_angle;
+/*< New angle in degrees.
+ * Supported values: 0 to 359
+ */
+
+ int16_t time_ms;
+/*< Transition time in milliseconds to set the new angle.
+ * Supported values: 0 to 32767
+ */
+} __packed;
+
+/* end_addtogroup audio_pp_module_ids */
+
+/* @ingroup audio_pp_module_ids
+ * ID of the Volume Control module pre/postprocessing block.
+ * This module supports the following parameter IDs:
+ * - #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
+ * - #ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN
+ * - #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG
+ * - #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+ * - #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS
+ * - #ASM_PARAM_ID_MULTICHANNEL_GAIN
+ * - #ASM_PARAM_ID_MULTICHANNEL_MUTE
+ */
+#define ASM_MODULE_ID_VOL_CTRL 0x00010BFE
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the master gain parameter used by the #ASM_MODULE_ID_VOL_CTRL
+ * module.
+ * @messagepayload
+ * @structure{asm_volume_ctrl_master_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN.tex}
+ */
+#define ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN 0x00010BFF
+
+/* ID of the left/right channel gain parameter used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ * @messagepayload
+ * @structure{asm_volume_ctrl_lr_chan_gain}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN.tex}
+ */
+#define ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN 0x00010C00
+
+/* ID of the mute configuration parameter used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ * @messagepayload
+ * @structure{asm_volume_ctrl_mute_config}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG.tex}
+ */
+#define ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG 0x00010C01
+
+/* ID of the soft stepping volume parameters used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ * @messagepayload
+ * @structure{asm_soft_step_volume_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMET
+ * ERS.tex}
+ */
+#define ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS 0x00010C29
+
+/* ID of the soft pause parameters used by the #ASM_MODULE_ID_VOL_CTRL
+ * module.
+ */
+#define ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS 0x00010D6A
+
+/* ID of the multiple-channel volume control parameters used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ */
+#define ASM_PARAM_ID_MULTICHANNEL_GAIN 0x00010713
+
+/* ID of the multiple-channel mute configuration parameters used by the
+ * #ASM_MODULE_ID_VOL_CTRL module.
+ */
+
+#define ASM_PARAM_ID_MULTICHANNEL_MUTE 0x00010714
+
+/* Structure for the master gain parameter for a volume control
+ * module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MASTER_GAIN
+ * parameter used by the Volume Control module.
+ */
+
+
+
+struct asm_volume_ctrl_master_gain {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint16_t master_gain;
+ /*< Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.
+ */
+} __packed;
+
+
+/* Structure for the left/right channel gain parameter for a
+ * volume control module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_LR_CHANNEL_GAIN
+ * parameters used by the Volume Control module.
+ */
+
+
+
+struct asm_volume_ctrl_lr_chan_gain {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+
+ uint16_t l_chan_gain;
+ /*< Linear gain in Q13 format for the left channel. */
+
+ uint16_t r_chan_gain;
+ /*< Linear gain in Q13 format for the right channel.*/
+} __packed;
+
+
+/* Structure for the mute configuration parameter for a
+ volume control module. */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_VOL_CTRL_MUTE_CONFIG
+ * parameter used by the Volume Control module.
+ */
+
+
+struct asm_volume_ctrl_mute_config {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t mute_flag;
+/*< Specifies whether mute is disabled (0) or enabled (nonzero).*/
+
+} __packed;
+
+/*
+ * Supported parameters for a soft stepping linear ramping curve.
+ */
+#define ASM_PARAM_SVC_RAMPINGCURVE_LINEAR 0
+
+/*
+ * Exponential ramping curve.
+ */
+#define ASM_PARAM_SVC_RAMPINGCURVE_EXP 1
+
+/*
+ * Logarithmic ramping curve.
+ */
+#define ASM_PARAM_SVC_RAMPINGCURVE_LOG 2
+
+/* Structure for holding soft stepping volume parameters. */
+
+
+/* Payload of the #ASM_PARAM_ID_SOFT_VOL_STEPPING_PARAMETERS
+ * parameters used by the Volume Control module.
+ */
+struct asm_soft_step_volume_params {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t period;
+/*< Period in milliseconds.
+ * Supported values: 0 to 15000
+ */
+
+ uint32_t step;
+/*< Step in microseconds.
+ * Supported values: 0 to 15000000
+ */
+
+ uint32_t ramping_curve;
+/*< Ramping curve type.
+ * Supported values:
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
+ */
+} __packed;
+
+
+/* Structure for holding soft pause parameters. */
+
+
+/* Payload of the #ASM_PARAM_ID_SOFT_PAUSE_PARAMETERS
+ * parameters used by the Volume Control module.
+ */
+
+
+struct asm_soft_pause_params {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t enable_flag;
+/*< Specifies whether soft pause is disabled (0) or enabled
+ * (nonzero).
+ */
+
+
+
+ uint32_t period;
+/*< Period in milliseconds.
+ * Supported values: 0 to 15000
+ */
+
+ uint32_t step;
+/*< Step in microseconds.
+ * Supported values: 0 to 15000000
+ */
+
+ uint32_t ramping_curve;
+/*< Ramping curve.
+ * Supported values:
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LINEAR
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_EXP
+ * - #ASM_PARAM_SVC_RAMPINGCURVE_LOG
+ */
+} __packed;
+
+
+/* Maximum number of channels.*/
+#define VOLUME_CONTROL_MAX_CHANNELS 8
+
+/* Structure for holding one channel type - gain pair. */
+
+
+/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN channel
+ * type/gain pairs used by the Volume Control module. \n \n This
+ * structure immediately follows the
+ * asm_volume_ctrl_multichannel_gain structure.
+ */
+
+
+struct asm_volume_ctrl_channelype_gain_pair {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint8_t channelype;
+/*< Channel type for which the gain setting is to be applied.
+ * Supported values:
+ * - #PCM_CHANNEL_L
+ * - #PCM_CHANNEL_R
+ * - #PCM_CHANNEL_C
+ * - #PCM_CHANNEL_LS
+ * - #PCM_CHANNEL_RS
+ * - #PCM_CHANNEL_LFE
+ * - #PCM_CHANNEL_CS
+ * - #PCM_CHANNEL_LB
+ * - #PCM_CHANNEL_RB
+ * - #PCM_CHANNELS
+ * - #PCM_CHANNEL_CVH
+ * - #PCM_CHANNEL_MS
+ * - #PCM_CHANNEL_FLC
+ * - #PCM_CHANNEL_FRC
+ * - #PCM_CHANNEL_RLC
+ * - #PCM_CHANNEL_RRC
+ */
+
+ uint8_t reserved1;
+ /*< Clients must set this field to zero. */
+
+ uint8_t reserved2;
+ /*< Clients must set this field to zero. */
+
+ uint8_t reserved3;
+ /*< Clients must set this field to zero. */
+
+ uint32_t gain;
+/*< Gain value for this channel in Q28 format.
+ * Supported values: Any
+ */
+} __packed;
+
+
+/* Structure for the multichannel gain command */
+
+
+/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_GAIN
+ * parameters used by the Volume Control module.
+ */
+
+
+struct asm_volume_ctrl_multichannel_gain {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t num_channels;
+/*< Number of channels for which gain values are provided. Any
+ * channels present in the data for which gain is not provided are
+ * set to unity gain.
+ * Supported values: 1 to 8
+ */
+
+
+ struct asm_volume_ctrl_channelype_gain_pair
+ gain_data[VOLUME_CONTROL_MAX_CHANNELS];
+ /*< Array of channel type/gain pairs.*/
+} __packed;
+
+
+/* Structure for holding one channel type - mute pair. */
+
+
+/* Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE channel
+ * type/mute setting pairs used by the Volume Control module. \n \n
+ * This structure immediately follows the
+ * asm_volume_ctrl_multichannel_mute structure.
+ */
+
+
+struct asm_volume_ctrl_channelype_mute_pair {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint8_t channelype;
+/*< Channel type for which the mute setting is to be applied.
+ * Supported values:
+ * - #PCM_CHANNEL_L
+ * - #PCM_CHANNEL_R
+ * - #PCM_CHANNEL_C
+ * - #PCM_CHANNEL_LS
+ * - #PCM_CHANNEL_RS
+ * - #PCM_CHANNEL_LFE
+ * - #PCM_CHANNEL_CS
+ * - #PCM_CHANNEL_LB
+ * - #PCM_CHANNEL_RB
+ * - #PCM_CHANNELS
+ * - #PCM_CHANNEL_CVH
+ * - #PCM_CHANNEL_MS
+ * - #PCM_CHANNEL_FLC
+ * - #PCM_CHANNEL_FRC
+ * - #PCM_CHANNEL_RLC
+ * - #PCM_CHANNEL_RRC
+ */
+
+ uint8_t reserved1;
+ /*< Clients must set this field to zero. */
+
+ uint8_t reserved2;
+ /*< Clients must set this field to zero. */
+
+ uint8_t reserved3;
+ /*< Clients must set this field to zero. */
+
+ uint32_t mute;
+/*< Mute setting for this channel.
+ * Supported values:
+ * - 0 = Unmute
+ * - Nonzero = Mute
+ */
+} __packed;
+
+
+/* Structure for the multichannel mute command */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_MULTICHANNEL_MUTE
+ * parameters used by the Volume Control module.
+ */
+
+
+struct asm_volume_ctrl_multichannel_mute {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t num_channels;
+/*< Number of channels for which mute configuration is
+ * provided. Any channels present in the data for which mute
+ * configuration is not provided are set to unmute.
+ * Supported values: 1 to 8
+ */
+
+struct asm_volume_ctrl_channelype_mute_pair
+ mute_data[VOLUME_CONTROL_MAX_CHANNELS];
+ /*< Array of channel type/mute setting pairs.*/
+} __packed;
+/* end_addtogroup audio_pp_param_ids */
+
+/* audio_pp_module_ids
+ * ID of the IIR Tuning Filter module.
+ * This module supports the following parameter IDs:
+ * - #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
+ * - #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
+ * - #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
+ */
+#define ASM_MODULE_ID_IIRUNING_FILTER 0x00010C02
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the IIR tuning filter enable parameter used by the
+ * #ASM_MODULE_ID_IIRUNING_FILTER module.
+ * @messagepayload
+ * @structure{asm_iiruning_filter_enable}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CO
+ * NFIG.tex}
+ */
+#define ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG 0x00010C03
+
+/* ID of the IIR tuning filter pregain parameter used by the
+ * #ASM_MODULE_ID_IIRUNING_FILTER module.
+ */
+#define ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN 0x00010C04
+
+/* ID of the IIR tuning filter configuration parameters used by the
+ * #ASM_MODULE_ID_IIRUNING_FILTER module.
+ */
+#define ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS 0x00010C05
+
+/* Structure for an enable configuration parameter for an
+ * IIR tuning filter module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_ENABLE_CONFIG
+ * parameter used by the IIR Tuning Filter module.
+ */
+struct asm_iiruning_filter_enable {
+ uint32_t enable_flag;
+/*< Specifies whether the IIR tuning filter is disabled (0) or
+ * enabled (1).
+ */
+} __packed;
+
+/* Structure for the pregain parameter for an IIR tuning filter module. */
+
+
+/* Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_PRE_GAIN
+ * parameters used by the IIR Tuning Filter module.
+ */
+struct asm_iiruning_filter_pregain {
+ uint16_t pregain;
+ /*< Linear gain in Q13 format. */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+/* Structure for the configuration parameter for an IIR tuning filter
+ * module.
+ */
+
+
+/* @brief Payload of the #ASM_PARAM_ID_IIRUNING_FILTER_CONFIG_PARAMS
+ * parameters used by the IIR Tuning Filter module. \n
+ * \n
+ * This structure is followed by the IIR filter coefficients: \n
+ * - Sequence of int32_t FilterCoeffs \n
+ * Five coefficients for each band. Each coefficient is in int32_t format, in
+ * the order of b0, b1, b2, a1, a2.
+ * - Sequence of int16_t NumShiftFactor \n
+ * One int16_t per band. The numerator shift factor is related to the Q
+ * factor of the filter coefficients.
+ * - Sequence of uint16_t PanSetting \n
+ * One uint16_t per band, indicating if the filter is applied to left (0),
+ * right (1), or both (2) channels.
+ */
+struct asm_iir_filter_config_params {
+ uint16_t num_biquad_stages;
+/*< Number of bands.
+ * Supported values: 0 to 20
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero.*/
+} __packed;
+
+/* audio_pp_module_ids
+ * ID of the Multiband Dynamic Range Control (MBDRC) module on the Tx/Rx
+ * paths.
+ * This module supports the following parameter IDs:
+ * - #ASM_PARAM_ID_MBDRC_ENABLE
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
+ */
+#define ASM_MODULE_ID_MBDRC 0x00010C06
+
+/* audio_pp_param_ids */
+/* ID of the MBDRC enable parameter used by the #ASM_MODULE_ID_MBDRC module.
+ * @messagepayload
+ * @structure{asm_mbdrc_enable}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_ENABLE.tex}
+ */
+#define ASM_PARAM_ID_MBDRC_ENABLE 0x00010C07
+
+/* ID of the MBDRC configuration parameters used by the
+ * #ASM_MODULE_ID_MBDRC module.
+ * @messagepayload
+ * @structure{asm_mbdrc_config_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.tex}
+ *
+ * @parspace Sub-band DRC configuration parameters
+ * @structure{asm_subband_drc_config_params}
+ * @tablespace
+ * @inputtable{Audio_Postproc_ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_subband_DRC.tex}
+ *
+ * @keep{6}
+ * To obtain legacy ADRC from MBDRC, use the calibration tool to:
+ *
+ * - Enable MBDRC (EnableFlag = TRUE)
+ * - Set number of bands to 1 (uiNumBands = 1)
+ * - Enable the first MBDRC band (DrcMode[0] = DRC_ENABLED = 1)
+ * - Clear the first band mute flag (MuteFlag[0] = 0)
+ * - Set the first band makeup gain to unity (compMakeUpGain[0] = 0x2000)
+ * - Use the legacy ADRC parameters to calibrate the rest of the MBDRC
+ * parameters.
+ */
+#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS 0x00010C08
+
+/* end_addtogroup audio_pp_param_ids */
+
+/* audio_pp_module_ids
+ * ID of the MMBDRC module version 2 pre/postprocessing block.
+ * This module differs from the original MBDRC (#ASM_MODULE_ID_MBDRC) in
+ * the length of the filters used in each sub-band.
+ * This module supports the following parameter ID:
+ * - #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2
+ */
+#define ASM_MODULE_ID_MBDRCV2 0x0001070B
+
+/* @addtogroup audio_pp_param_ids */
+/* ID of the configuration parameters used by the
+ * #ASM_MODULE_ID_MBDRCV2 module for the improved filter structure
+ * of the MBDRC v2 pre/postprocessing block.
+ * The update to this configuration structure from the original
+ * MBDRC is the number of filter coefficients in the filter
+ * structure. The sequence for is as follows:
+ * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
+ * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
+ * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
+ * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t
+ * padding
+ * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags +
+ * uint16_t padding
+ * This block uses the same parameter structure as
+ * #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS.
+ */
+#define ASM_PARAM_ID_MBDRC_CONFIG_PARAMS_IMPROVED_FILTBANK_V2 \
+ 0x0001070C
+
+/* Structure for the enable parameter for an MBDRC module. */
+
+
+/* Payload of the #ASM_PARAM_ID_MBDRC_ENABLE parameter used by the
+ * MBDRC module.
+ */
+struct asm_mbdrc_enable {
+ uint32_t enable_flag;
+/*< Specifies whether MBDRC is disabled (0) or enabled (nonzero).*/
+} __packed;
+
+/* Structure for the configuration parameters for an MBDRC module. */
+
+
+/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS
+ * parameters used by the MBDRC module. \n \n Following this
+ * structure is the payload for sub-band DRC configuration
+ * parameters (asm_subband_drc_config_params). This sub-band
+ * structure must be repeated for each band.
+ */
+
+
+struct asm_mbdrc_config_params {
+ uint16_t num_bands;
+/*< Number of bands.
+ * Supported values: 1 to 5
+ */
+
+ int16_t limiterhreshold;
+/*< Threshold in decibels for the limiter output.
+ * Supported values: -72 to 18 \n
+ * Recommended value: 3994 (-0.22 db in Q3.12 format)
+ */
+
+ int16_t limiter_makeup_gain;
+/*< Makeup gain in decibels for the limiter output.
+ * Supported values: -42 to 42 \n
+ * Recommended value: 256 (0 dB in Q7.8 format)
+ */
+
+ int16_t limiter_gc;
+/*< Limiter gain recovery coefficient.
+ * Supported values: 0.5 to 0.99 \n
+ * Recommended value: 32440 (0.99 in Q15 format)
+ */
+
+ int16_t limiter_delay;
+/*< Limiter delay in samples.
+ * Supported values: 0 to 10 \n
+ * Recommended value: 262 (0.008 samples in Q15 format)
+ */
+
+ int16_t limiter_max_wait;
+/*< Maximum limiter waiting time in samples.
+ * Supported values: 0 to 10 \n
+ * Recommended value: 262 (0.008 samples in Q15 format)
+ */
+} __packed;
+
+/* DRC configuration structure for each sub-band of an MBDRC module. */
+
+
+/* Payload of the #ASM_PARAM_ID_MBDRC_CONFIG_PARAMS DRC
+ * configuration parameters for each sub-band in the MBDRC module.
+ * After this DRC structure is configured for valid bands, the next
+ * MBDRC setparams expects the sequence of sub-band MBDRC filter
+ * coefficients (the length depends on the number of bands) plus the
+ * mute flag for that band plus uint16_t padding.
+ *
+ * @keep{10}
+ * The filter coefficient and mute flag are of type int16_t:
+ * - FIR coefficient = int16_t firFilter
+ * - Mute flag = int16_t fMuteFlag
+ *
+ * The sequence is as follows:
+ * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
+ * - 2 bands = 97 FIR coefficients + 2 mute flags + uint16_t padding
+ * - 3 bands = 97+33 FIR coefficients + 3 mute flags + uint16_t padding
+ * - 4 bands = 97+33+33 FIR coefficients + 4 mute flags + uint16_t padding
+ * - 5 bands = 97+33+33+33 FIR coefficients + 5 mute flags + uint16_t padding
+ *
+ * For improved filterbank, the sequence is as follows:
+ * - 1 band = 0 FIR coefficient + 1 mute flag + uint16_t padding
+ * - 2 bands = 141 FIR coefficients + 2 mute flags + uint16_t padding
+ * - 3 bands = 141+81 FIR coefficients + 3 mute flags + uint16_t padding
+ * - 4 bands = 141+81+61 FIR coefficients + 4 mute flags + uint16_t padding
+ * - 5 bands = 141+81+61+61 FIR coefficients + 5 mute flags + uint16_t padding
+ */
+struct asm_subband_drc_config_params {
+ int16_t drc_stereo_linked_flag;
+/*< Specifies whether all stereo channels have the same applied
+ * dynamics (1) or if they process their dynamics independently (0).
+ * Supported values:
+ * - 0 -- Not linked
+ * - 1 -- Linked
+ */
+
+ int16_t drc_mode;
+/*< Specifies whether DRC mode is bypassed for sub-bands.
+ * Supported values:
+ * - 0 -- Disabled
+ * - 1 -- Enabled
+ */
+
+ int16_t drc_down_sample_level;
+/*< DRC down sample level.
+ * Supported values: @ge 1
+ */
+
+ int16_t drc_delay;
+/*< DRC delay in samples.
+ * Supported values: 0 to 1200
+ */
+
+ uint16_t drc_rmsime_avg_const;
+/*< RMS signal energy time-averaging constant.
+ * Supported values: 0 to 2^16-1
+ */
+
+ uint16_t drc_makeup_gain;
+/*< DRC makeup gain in decibels.
+ * Supported values: 258 to 64917
+ */
+ /* Down expander settings */
+ int16_t down_expdrhreshold;
+/*< Down expander threshold.
+ * Supported Q7 format values: 1320 to up_cmpsrhreshold
+ */
+
+ int16_t down_expdr_slope;
+/*< Down expander slope.
+ * Supported Q8 format values: -32768 to 0.
+ */
+
+ uint32_t down_expdr_attack;
+/*< Down expander attack constant.
+ * Supported Q31 format values: 196844 to 2^31.
+ */
+
+ uint32_t down_expdr_release;
+/*< Down expander release constant.
+ * Supported Q31 format values: 19685 to 2^31
+ */
+
+ uint16_t down_expdr_hysteresis;
+/*< Down expander hysteresis constant.
+ * Supported Q14 format values: 1 to 32690
+ */
+
+ uint16_t reserved;
+ /*< Clients must set this field to zero. */
+
+ int32_t down_expdr_min_gain_db;
+/*< Down expander minimum gain.
+ * Supported Q23 format values: -805306368 to 0.
+ */
+
+ /* Up compressor settings */
+
+ int16_t up_cmpsrhreshold;
+/*< Up compressor threshold.
+ * Supported Q7 format values: down_expdrhreshold to
+ * down_cmpsrhreshold.
+ */
+
+ uint16_t up_cmpsr_slope;
+/*< Up compressor slope.
+ * Supported Q16 format values: 0 to 64881.
+ */
+
+ uint32_t up_cmpsr_attack;
+/*< Up compressor attack constant.
+ * Supported Q31 format values: 196844 to 2^31.
+ */
+
+ uint32_t up_cmpsr_release;
+/*< Up compressor release constant.
+ * Supported Q31 format values: 19685 to 2^31.
+ */
+
+ uint16_t up_cmpsr_hysteresis;
+/*< Up compressor hysteresis constant.
+ * Supported Q14 format values: 1 to 32690.
+ */
+
+ /* Down compressor settings */
+
+ int16_t down_cmpsrhreshold;
+/*< Down compressor threshold.
+ * Supported Q7 format values: up_cmpsrhreshold to 11560.
+ */
+
+ uint16_t down_cmpsr_slope;
+/*< Down compressor slope.
+ * Supported Q16 format values: 0 to 64881.
+ */
+
+ uint16_t reserved1;
+/*< Clients must set this field to zero. */
+
+ uint32_t down_cmpsr_attack;
+/*< Down compressor attack constant.
+ * Supported Q31 format values: 196844 to 2^31.
+ */
+
+ uint32_t down_cmpsr_release;
+/*< Down compressor release constant.
+ * Supported Q31 format values: 19685 to 2^31.
+ */
+
+ uint16_t down_cmpsr_hysteresis;
+/*< Down compressor hysteresis constant.
+ * Supported Q14 values: 1 to 32690.
+ */
+
+ uint16_t reserved2;
+/*< Clients must set this field to zero.*/
+} __packed;
+
+#define ASM_MODULE_ID_EQUALIZER 0x00010C27
+#define ASM_PARAM_ID_EQUALIZER_PARAMETERS 0x00010C28
+
+#define ASM_MAX_EQ_BANDS 12
+
+struct asm_eq_per_band_params {
+ uint32_t band_idx;
+/*< Band index.
+ * Supported values: 0 to 11
+ */
+
+ uint32_t filterype;
+/*< Type of filter.
+ * Supported values:
+ * - #ASM_PARAM_EQYPE_NONE
+ * - #ASM_PARAM_EQ_BASS_BOOST
+ * - #ASM_PARAM_EQ_BASS_CUT
+ * - #ASM_PARAM_EQREBLE_BOOST
+ * - #ASM_PARAM_EQREBLE_CUT
+ * - #ASM_PARAM_EQ_BAND_BOOST
+ * - #ASM_PARAM_EQ_BAND_CUT
+ */
+
+ uint32_t center_freq_hz;
+ /*< Filter band center frequency in Hertz. */
+
+ int32_t filter_gain;
+/*< Filter band initial gain.
+ * Supported values: +12 to -12 dB in 1 dB increments
+ */
+
+ int32_t q_factor;
+/*< Filter band quality factor expressed as a Q8 number, i.e., a
+ * fixed-point number with q factor of 8. For example, 3000/(2^8).
+ */
+} __packed;
+
+struct asm_eq_params {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_pp_params_v2 param;
+ struct asm_stream_param_data_v2 data;
+ uint32_t enable_flag;
+/*< Specifies whether the equalizer module is disabled (0) or enabled
+ * (nonzero).
+ */
+
+ uint32_t num_bands;
+/*< Number of bands.
+ * Supported values: 1 to 12
+ */
+ struct asm_eq_per_band_params eq_bands[ASM_MAX_EQ_BANDS];
+
+} __packed;
+
+/* No equalizer effect.*/
+#define ASM_PARAM_EQYPE_NONE 0
+
+/* Bass boost equalizer effect.*/
+#define ASM_PARAM_EQ_BASS_BOOST 1
+
+/*Bass cut equalizer effect.*/
+#define ASM_PARAM_EQ_BASS_CUT 2
+
+/* Treble boost equalizer effect */
+#define ASM_PARAM_EQREBLE_BOOST 3
+
+/* Treble cut equalizer effect.*/
+#define ASM_PARAM_EQREBLE_CUT 4
+
+/* Band boost equalizer effect.*/
+#define ASM_PARAM_EQ_BAND_BOOST 5
+
+/* Band cut equalizer effect.*/
+#define ASM_PARAM_EQ_BAND_CUT 6
+
+
+/* ERROR CODES */
+/* Success. The operation completed with no errors. */
+#define ADSP_EOK 0x00000000
+/* General failure. */
+#define ADSP_EFAILED 0x00000001
+/* Bad operation parameter. */
+#define ADSP_EBADPARAM 0x00000002
+/* Unsupported routine or operation. */
+#define ADSP_EUNSUPPORTED 0x00000003
+/* Unsupported version. */
+#define ADSP_EVERSION 0x00000004
+/* Unexpected problem encountered. */
+#define ADSP_EUNEXPECTED 0x00000005
+/* Unhandled problem occurred. */
+#define ADSP_EPANIC 0x00000006
+/* Unable to allocate resource. */
+#define ADSP_ENORESOURCE 0x00000007
+/* Invalid handle. */
+#define ADSP_EHANDLE 0x00000008
+/* Operation is already processed. */
+#define ADSP_EALREADY 0x00000009
+/* Operation is not ready to be processed. */
+#define ADSP_ENOTREADY 0x0000000A
+/* Operation is pending completion. */
+#define ADSP_EPENDING 0x0000000B
+/* Operation could not be accepted or processed. */
+#define ADSP_EBUSY 0x0000000C
+/* Operation aborted due to an error. */
+#define ADSP_EABORTED 0x0000000D
+/* Operation preempted by a higher priority. */
+#define ADSP_EPREEMPTED 0x0000000E
+/* Operation requests intervention to complete. */
+#define ADSP_ECONTINUE 0x0000000F
+/* Operation requests immediate intervention to complete. */
+#define ADSP_EIMMEDIATE 0x00000010
+/* Operation is not implemented. */
+#define ADSP_ENOTIMPL 0x00000011
+/* Operation needs more data or resources. */
+#define ADSP_ENEEDMORE 0x00000012
+/* Operation does not have memory. */
+#define ADSP_ENOMEMORY 0x00000014
+/* Item does not exist. */
+#define ADSP_ENOTEXIST 0x00000015
+/* Operation is finished. */
+#define ADSP_ETERMINATED 0x00011174
+
+/*bharath, adsp_error_codes.h */
+
+#endif /*_APR_AUDIO_V2_H_ */
diff --git a/include/sound/apr_audio.h b/include/sound/apr_audio.h
new file mode 100644
index 0000000..96795a3
--- /dev/null
+++ b/include/sound/apr_audio.h
@@ -0,0 +1,1535 @@
+/*
+ *
+ * Copyright (c) 2010-2012, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ *
+ */
+
+#ifndef _APR_AUDIO_H_
+#define _APR_AUDIO_H_
+
+/* ASM opcodes without APR payloads*/
+#include <mach/qdsp6v2/apr.h>
+
+/*
+ * Audio Front End (AFE)
+ */
+
+/* Port ID. Update afe_get_port_index when a new port is added here. */
+#define PRIMARY_I2S_RX 0 /* index = 0 */
+#define PRIMARY_I2S_TX 1 /* index = 1 */
+#define PCM_RX 2 /* index = 2 */
+#define PCM_TX 3 /* index = 3 */
+#define SECONDARY_I2S_RX 4 /* index = 4 */
+#define SECONDARY_I2S_TX 5 /* index = 5 */
+#define MI2S_RX 6 /* index = 6 */
+#define MI2S_TX 7 /* index = 7 */
+#define HDMI_RX 8 /* index = 8 */
+#define RSVD_2 9 /* index = 9 */
+#define RSVD_3 10 /* index = 10 */
+#define DIGI_MIC_TX 11 /* index = 11 */
+#define VOICE_RECORD_RX 0x8003 /* index = 12 */
+#define VOICE_RECORD_TX 0x8004 /* index = 13 */
+#define VOICE_PLAYBACK_TX 0x8005 /* index = 14 */
+
+/* Slimbus Multi channel port id pool */
+#define SLIMBUS_0_RX 0x4000 /* index = 15 */
+#define SLIMBUS_0_TX 0x4001 /* index = 16 */
+#define SLIMBUS_1_RX 0x4002 /* index = 17 */
+#define SLIMBUS_1_TX 0x4003 /* index = 18 */
+#define SLIMBUS_2_RX 0x4004
+#define SLIMBUS_2_TX 0x4005
+#define SLIMBUS_3_RX 0x4006
+#define SLIMBUS_3_TX 0x4007
+#define SLIMBUS_4_RX 0x4008
+#define SLIMBUS_4_TX 0x4009 /* index = 24 */
+
+#define INT_BT_SCO_RX 0x3000 /* index = 25 */
+#define INT_BT_SCO_TX 0x3001 /* index = 26 */
+#define INT_BT_A2DP_RX 0x3002 /* index = 27 */
+#define INT_FM_RX 0x3004 /* index = 28 */
+#define INT_FM_TX 0x3005 /* index = 29 */
+#define RT_PROXY_PORT_001_RX 0x2000 /* index = 30 */
+#define RT_PROXY_PORT_001_TX 0x2001 /* index = 31 */
+
+#define AFE_PORT_INVALID 0xFFFF
+#define SLIMBUS_EXTPROC_RX AFE_PORT_INVALID
+
+#define AFE_PORT_CMD_START 0x000100ca
+
+#define AFE_EVENT_RTPORT_START 0
+#define AFE_EVENT_RTPORT_STOP 1
+#define AFE_EVENT_RTPORT_LOW_WM 2
+#define AFE_EVENT_RTPORT_HI_WM 3
+
+struct afe_port_start_command {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 gain; /* Q13 */
+ u32 sample_rate; /* 8 , 16, 48khz */
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_STOP 0x000100cb
+struct afe_port_stop_command {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 reserved;
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_APPLY_GAIN 0x000100cc
+struct afe_port_gain_command {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 gain;/* Q13 */
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_SIDETONE_CTL 0x000100cd
+struct afe_port_sidetone_command {
+ struct apr_hdr hdr;
+ u16 rx_port_id; /* Primary i2s tx = 1 */
+ /* PCM tx = 3 */
+ /* Secondary i2s tx = 5 */
+ /* Mi2s tx = 7 */
+ /* Digital mic tx = 11 */
+ u16 tx_port_id; /* Primary i2s rx = 0 */
+ /* PCM rx = 2 */
+ /* Secondary i2s rx = 4 */
+ /* Mi2S rx = 6 */
+ /* HDMI rx = 8 */
+ u16 gain; /* Q13 */
+ u16 enable; /* 1 = enable, 0 = disable */
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_LOOPBACK 0x000100ce
+struct afe_loopback_command {
+ struct apr_hdr hdr;
+ u16 tx_port_id; /* Primary i2s rx = 0 */
+ /* PCM rx = 2 */
+ /* Secondary i2s rx = 4 */
+ /* Mi2S rx = 6 */
+ /* HDMI rx = 8 */
+ u16 rx_port_id; /* Primary i2s tx = 1 */
+ /* PCM tx = 3 */
+ /* Secondary i2s tx = 5 */
+ /* Mi2s tx = 7 */
+ /* Digital mic tx = 11 */
+ u16 mode; /* Default -1, DSP will conver
+ the tx to rx format */
+ u16 enable; /* 1 = enable, 0 = disable */
+} __attribute__ ((packed));
+
+#define AFE_PSEUDOPORT_CMD_START 0x000100cf
+struct afe_pseudoport_start_command {
+ struct apr_hdr hdr;
+ u16 port_id; /* Pseudo Port 1 = 0x8000 */
+ /* Pseudo Port 2 = 0x8001 */
+ /* Pseudo Port 3 = 0x8002 */
+ u16 timing; /* FTRT = 0 , AVTimer = 1, */
+} __attribute__ ((packed));
+
+#define AFE_PSEUDOPORT_CMD_STOP 0x000100d0
+struct afe_pseudoport_stop_command {
+ struct apr_hdr hdr;
+ u16 port_id; /* Pseudo Port 1 = 0x8000 */
+ /* Pseudo Port 2 = 0x8001 */
+ /* Pseudo Port 3 = 0x8002 */
+ u16 reserved;
+} __attribute__ ((packed));
+
+#define AFE_CMD_GET_ACTIVE_PORTS 0x000100d1
+
+
+#define AFE_CMD_GET_ACTIVE_HANDLES_FOR_PORT 0x000100d2
+struct afe_get_active_handles_command {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 reserved;
+} __attribute__ ((packed));
+
+#define AFE_PCM_CFG_MODE_PCM 0x0
+#define AFE_PCM_CFG_MODE_AUX 0x1
+#define AFE_PCM_CFG_SYNC_EXT 0x0
+#define AFE_PCM_CFG_SYNC_INT 0x1
+#define AFE_PCM_CFG_FRM_8BPF 0x0
+#define AFE_PCM_CFG_FRM_16BPF 0x1
+#define AFE_PCM_CFG_FRM_32BPF 0x2
+#define AFE_PCM_CFG_FRM_64BPF 0x3
+#define AFE_PCM_CFG_FRM_128BPF 0x4
+#define AFE_PCM_CFG_FRM_256BPF 0x5
+#define AFE_PCM_CFG_QUANT_ALAW_NOPAD 0x0
+#define AFE_PCM_CFG_QUANT_MULAW_NOPAD 0x1
+#define AFE_PCM_CFG_QUANT_LINEAR_NOPAD 0x2
+#define AFE_PCM_CFG_QUANT_ALAW_PAD 0x3
+#define AFE_PCM_CFG_QUANT_MULAW_PAD 0x4
+#define AFE_PCM_CFG_QUANT_LINEAR_PAD 0x5
+#define AFE_PCM_CFG_CDATAOE_MASTER 0x0
+#define AFE_PCM_CFG_CDATAOE_SHARE 0x1
+
+struct afe_port_pcm_cfg {
+ u16 mode; /* PCM (short sync) = 0, AUXPCM (long sync) = 1 */
+ u16 sync; /* external = 0 , internal = 1 */
+ u16 frame; /* 8 bpf = 0 */
+ /* 16 bpf = 1 */
+ /* 32 bpf = 2 */
+ /* 64 bpf = 3 */
+ /* 128 bpf = 4 */
+ /* 256 bpf = 5 */
+ u16 quant;
+ u16 slot; /* Slot for PCM stream , 0 - 31 */
+ u16 data; /* 0, PCM block is the only master */
+ /* 1, PCM block is shares to driver data out signal */
+ /* other master */
+ u16 reserved;
+} __attribute__ ((packed));
+
+enum {
+ AFE_I2S_SD0 = 1,
+ AFE_I2S_SD1,
+ AFE_I2S_SD2,
+ AFE_I2S_SD3,
+ AFE_I2S_QUAD01,
+ AFE_I2S_QUAD23,
+ AFE_I2S_6CHS,
+ AFE_I2S_8CHS,
+};
+
+#define AFE_MI2S_MONO 0
+#define AFE_MI2S_STEREO 3
+#define AFE_MI2S_4CHANNELS 4
+#define AFE_MI2S_6CHANNELS 6
+#define AFE_MI2S_8CHANNELS 8
+
+struct afe_port_mi2s_cfg {
+ u16 bitwidth; /* 16,24,32 */
+ u16 line; /* Called ChannelMode in documentation */
+ /* i2s_sd0 = 1 */
+ /* i2s_sd1 = 2 */
+ /* i2s_sd2 = 3 */
+ /* i2s_sd3 = 4 */
+ /* i2s_quad01 = 5 */
+ /* i2s_quad23 = 6 */
+ /* i2s_6chs = 7 */
+ /* i2s_8chs = 8 */
+ u16 channel; /* Called MonoStereo in documentation */
+ /* i2s mono = 0 */
+ /* i2s mono right = 1 */
+ /* i2s mono left = 2 */
+ /* i2s stereo = 3 */
+ u16 ws; /* 0, word select signal from external source */
+ /* 1, word select signal from internal source */
+ u16 format; /* don't touch this field if it is not for */
+ /* AFE_PORT_CMD_I2S_CONFIG opcode */
+} __attribute__ ((packed));
+
+struct afe_port_hdmi_cfg {
+ u16 bitwidth; /* 16,24,32 */
+ u16 channel_mode; /* HDMI Stereo = 0 */
+ /* HDMI_3Point1 (4-ch) = 1 */
+ /* HDMI_5Point1 (6-ch) = 2 */
+ /* HDMI_6Point1 (8-ch) = 3 */
+ u16 data_type; /* HDMI_Linear = 0 */
+ /* HDMI_non_Linear = 1 */
+} __attribute__ ((packed));
+
+
+struct afe_port_hdmi_multi_ch_cfg {
+ u16 data_type; /* HDMI_Linear = 0 */
+ /* HDMI_non_Linear = 1 */
+ u16 channel_allocation; /* The default is 0 (Stereo) */
+ u16 reserved; /* must be set to 0 */
+} __packed;
+
+
+/* Slimbus Device Ids */
+#define AFE_SLIMBUS_DEVICE_1 0x0
+#define AFE_SLIMBUS_DEVICE_2 0x1
+#define AFE_PORT_MAX_AUDIO_CHAN_CNT 16
+
+struct afe_port_slimbus_cfg {
+ u16 slimbus_dev_id; /* SLIMBUS Device id.*/
+
+ u16 slave_dev_pgd_la; /* Slave ported generic device
+ * logical address.
+ */
+ u16 slave_dev_intfdev_la; /* Slave interface device logical
+ * address.
+ */
+ u16 bit_width; /** bit width of the samples, 16, 24.*/
+
+ u16 data_format; /** data format.*/
+
+ u16 num_channels; /** Number of channels.*/
+
+ /** Slave port mapping for respective channels.*/
+ u16 slave_port_mapping[AFE_PORT_MAX_AUDIO_CHAN_CNT];
+
+ u16 reserved;
+} __packed;
+
+struct afe_port_slimbus_sch_cfg {
+ u16 slimbus_dev_id; /* SLIMBUS Device id.*/
+ u16 bit_width; /** bit width of the samples, 16, 24.*/
+ u16 data_format; /** data format.*/
+ u16 num_channels; /** Number of channels.*/
+ u16 reserved;
+ /** Slave channel mapping for respective channels.*/
+ u8 slave_ch_mapping[8];
+} __packed;
+
+struct afe_port_rtproxy_cfg {
+ u16 bitwidth; /* 16,24,32 */
+ u16 interleaved; /* interleaved = 1 */
+ /* Noninterleaved = 0 */
+ u16 frame_sz; /* 5ms buffers = 160bytes */
+ u16 jitter; /* 10ms of jitter = 320 */
+ u16 lw_mark; /* Low watermark in bytes for triggering event*/
+ u16 hw_mark; /* High watermark bytes for triggering event*/
+ u16 rsvd;
+ int num_ch; /* 1 to 8 */
+} __packed;
+
+#define AFE_PORT_AUDIO_IF_CONFIG 0x000100d3
+#define AFE_PORT_AUDIO_SLIM_SCH_CONFIG 0x000100e4
+#define AFE_PORT_MULTI_CHAN_HDMI_AUDIO_IF_CONFIG 0x000100D9
+#define AFE_PORT_CMD_I2S_CONFIG 0x000100E7
+
+union afe_port_config {
+ struct afe_port_pcm_cfg pcm;
+ struct afe_port_mi2s_cfg mi2s;
+ struct afe_port_hdmi_cfg hdmi;
+ struct afe_port_hdmi_multi_ch_cfg hdmi_multi_ch;
+ struct afe_port_slimbus_cfg slimbus;
+ struct afe_port_slimbus_sch_cfg slim_sch;
+ struct afe_port_rtproxy_cfg rtproxy;
+} __attribute__((packed));
+
+struct afe_audioif_config_command {
+ struct apr_hdr hdr;
+ u16 port_id;
+ union afe_port_config port;
+} __attribute__ ((packed));
+
+#define AFE_TEST_CODEC_LOOPBACK_CTL 0x000100d5
+struct afe_codec_loopback_command {
+ u16 port_inf; /* Primary i2s = 0 */
+ /* PCM = 2 */
+ /* Secondary i2s = 4 */
+ /* Mi2s = 6 */
+ u16 enable; /* 0, disable. 1, enable */
+} __attribute__ ((packed));
+
+
+#define AFE_PARAM_ID_SIDETONE_GAIN 0x00010300
+struct afe_param_sidetone_gain {
+ u16 gain;
+ u16 reserved;
+} __attribute__ ((packed));
+
+#define AFE_PARAM_ID_SAMPLING_RATE 0x00010301
+struct afe_param_sampling_rate {
+ u32 sampling_rate;
+} __attribute__ ((packed));
+
+
+#define AFE_PARAM_ID_CHANNELS 0x00010302
+struct afe_param_channels {
+ u16 channels;
+ u16 reserved;
+} __attribute__ ((packed));
+
+
+#define AFE_PARAM_ID_LOOPBACK_GAIN 0x00010303
+struct afe_param_loopback_gain {
+ u16 gain;
+ u16 reserved;
+} __attribute__ ((packed));
+
+/* Parameter ID used to configure and enable/disable the loopback path. The
+ * difference with respect to the existing API, AFE_PORT_CMD_LOOPBACK, is that
+ * it allows Rx port to be configured as source port in loopback path. Port-id
+ * in AFE_PORT_CMD_SET_PARAM cmd is the source port whcih can be Tx or Rx port.
+ * In addition, we can configure the type of routing mode to handle different
+ * use cases.
+*/
+enum {
+ /* Regular loopback from source to destination port */
+ LB_MODE_DEFAULT = 1,
+ /* Sidetone feed from Tx source to Rx destination port */
+ LB_MODE_SIDETONE,
+ /* Echo canceller reference, voice + audio + DTMF */
+ LB_MODE_EC_REF_VOICE_AUDIO,
+ /* Echo canceller reference, voice alone */
+ LB_MODE_EC_REF_VOICE
+};
+
+#define AFE_PARAM_ID_LOOPBACK_CONFIG 0x0001020B
+#define AFE_API_VERSION_LOOPBACK_CONFIG 0x1
+struct afe_param_loopback_cfg {
+ /* Minor version used for tracking the version of the configuration
+ * interface.
+ */
+ uint32_t loopback_cfg_minor_version;
+
+ /* Destination Port Id. */
+ uint16_t dst_port_id;
+
+ /* Specifies data path type from src to dest port. Supported values:
+ * LB_MODE_DEFAULT
+ * LB_MODE_SIDETONE
+ * LB_MODE_EC_REF_VOICE_AUDIO
+ * LB_MODE_EC_REF_VOICE
+ */
+ uint16_t routing_mode;
+
+ /* Specifies whether to enable (1) or disable (0) an AFE loopback. */
+ uint16_t enable;
+
+ /* Reserved for 32-bit alignment. This field must be set to 0. */
+ uint16_t reserved;
+} __packed;
+
+#define AFE_MODULE_ID_PORT_INFO 0x00010200
+/* Module ID for the loopback-related parameters. */
+#define AFE_MODULE_LOOPBACK 0x00010205
+struct afe_param_payload {
+ u32 module_id;
+ u32 param_id;
+ u16 param_size;
+ u16 reserved;
+ union {
+ struct afe_param_sidetone_gain sidetone_gain;
+ struct afe_param_sampling_rate sampling_rate;
+ struct afe_param_channels channels;
+ struct afe_param_loopback_gain loopback_gain;
+ struct afe_param_loopback_cfg loopback_cfg;
+ } __attribute__((packed)) param;
+} __attribute__ ((packed));
+
+#define AFE_PORT_CMD_SET_PARAM 0x000100dc
+
+struct afe_port_cmd_set_param {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 payload_size;
+ u32 payload_address;
+ struct afe_param_payload payload;
+} __attribute__ ((packed));
+
+struct afe_port_cmd_set_param_no_payload {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 payload_size;
+ u32 payload_address;
+} __packed;
+
+#define AFE_EVENT_GET_ACTIVE_PORTS 0x00010100
+struct afe_get_active_ports_rsp {
+ u16 num_ports;
+ u16 port_id;
+} __attribute__ ((packed));
+
+
+#define AFE_EVENT_GET_ACTIVE_HANDLES 0x00010102
+struct afe_get_active_handles_rsp {
+ u16 port_id;
+ u16 num_handles;
+ u16 mode; /* 0, voice rx */
+ /* 1, voice tx */
+ /* 2, audio rx */
+ /* 3, audio tx */
+ u16 handle;
+} __attribute__ ((packed));
+
+#define AFE_SERVICE_CMD_MEMORY_MAP 0x000100DE
+struct afe_cmd_memory_map {
+ struct apr_hdr hdr;
+ u32 phy_addr;
+ u32 mem_sz;
+ u16 mem_id;
+ u16 rsvd;
+} __packed;
+
+#define AFE_SERVICE_CMD_MEMORY_UNMAP 0x000100DF
+struct afe_cmd_memory_unmap {
+ struct apr_hdr hdr;
+ u32 phy_addr;
+} __packed;
+
+#define AFE_SERVICE_CMD_REG_RTPORT 0x000100E0
+struct afe_cmd_reg_rtport {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 rsvd;
+} __packed;
+
+#define AFE_SERVICE_CMD_UNREG_RTPORT 0x000100E1
+struct afe_cmd_unreg_rtport {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 rsvd;
+} __packed;
+
+#define AFE_SERVICE_CMD_RTPORT_WR 0x000100E2
+struct afe_cmd_rtport_wr {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 rsvd;
+ u32 buf_addr;
+ u32 bytes_avail;
+} __packed;
+
+#define AFE_SERVICE_CMD_RTPORT_RD 0x000100E3
+struct afe_cmd_rtport_rd {
+ struct apr_hdr hdr;
+ u16 port_id;
+ u16 rsvd;
+ u32 buf_addr;
+ u32 bytes_avail;
+} __packed;
+
+#define AFE_EVENT_RT_PROXY_PORT_STATUS 0x00010105
+
+#define ADM_MAX_COPPS 5
+
+#define ADM_SERVICE_CMD_GET_COPP_HANDLES 0x00010300
+struct adm_get_copp_handles_command {
+ struct apr_hdr hdr;
+} __attribute__ ((packed));
+
+#define ADM_CMD_MATRIX_MAP_ROUTINGS 0x00010301
+struct adm_routings_session {
+ u16 id;
+ u16 num_copps;
+ u16 copp_id[ADM_MAX_COPPS+1]; /*Padding if numCopps is odd */
+} __packed;
+
+struct adm_routings_command {
+ struct apr_hdr hdr;
+ u32 path; /* 0 = Rx, 1 Tx */
+ u32 num_sessions;
+ struct adm_routings_session session[8];
+} __attribute__ ((packed));
+
+
+#define ADM_CMD_MATRIX_RAMP_GAINS 0x00010302
+struct adm_ramp_gain {
+ struct apr_hdr hdr;
+ u16 session_id;
+ u16 copp_id;
+ u16 initial_gain;
+ u16 gain_increment;
+ u16 ramp_duration;
+ u16 reserved;
+} __attribute__ ((packed));
+
+struct adm_ramp_gains_command {
+ struct apr_hdr hdr;
+ u32 id;
+ u32 num_gains;
+ struct adm_ramp_gain gains[ADM_MAX_COPPS];
+} __attribute__ ((packed));
+
+
+#define ADM_CMD_COPP_OPEN 0x00010304
+struct adm_copp_open_command {
+ struct apr_hdr hdr;
+ u16 flags;
+ u16 mode; /* 1-RX, 2-Live TX, 3-Non Live TX */
+ u16 endpoint_id1;
+ u16 endpoint_id2;
+ u32 topology_id;
+ u16 channel_config;
+ u16 reserved;
+ u32 rate;
+} __attribute__ ((packed));
+
+#define ADM_CMD_COPP_CLOSE 0x00010305
+
+#define ADM_CMD_MULTI_CHANNEL_COPP_OPEN 0x00010310
+struct adm_multi_ch_copp_open_command {
+ struct apr_hdr hdr;
+ u16 flags;
+ u16 mode; /* 1-RX, 2-Live TX, 3-Non Live TX */
+ u16 endpoint_id1;
+ u16 endpoint_id2;
+ u32 topology_id;
+ u16 channel_config;
+ u16 reserved;
+ u32 rate;
+ u8 dev_channel_mapping[8];
+} __packed;
+
+#define ADM_CMD_MEMORY_MAP 0x00010C30
+struct adm_cmd_memory_map{
+ struct apr_hdr hdr;
+ u32 buf_add;
+ u32 buf_size;
+ u16 mempool_id;
+ u16 reserved;
+} __attribute__((packed));
+
+#define ADM_CMD_MEMORY_UNMAP 0x00010C31
+struct adm_cmd_memory_unmap{
+ struct apr_hdr hdr;
+ u32 buf_add;
+} __attribute__((packed));
+
+#define ADM_CMD_MEMORY_MAP_REGIONS 0x00010C47
+struct adm_memory_map_regions{
+ u32 phys;
+ u32 buf_size;
+} __attribute__((packed));
+
+struct adm_cmd_memory_map_regions{
+ struct apr_hdr hdr;
+ u16 mempool_id;
+ u16 nregions;
+} __attribute__((packed));
+
+#define ADM_CMD_MEMORY_UNMAP_REGIONS 0x00010C48
+struct adm_memory_unmap_regions{
+ u32 phys;
+} __attribute__((packed));
+
+struct adm_cmd_memory_unmap_regions{
+ struct apr_hdr hdr;
+ u16 nregions;
+ u16 reserved;
+} __attribute__((packed));
+
+#define DEFAULT_COPP_TOPOLOGY 0x00010be3
+#define DEFAULT_POPP_TOPOLOGY 0x00010be4
+#define VPM_TX_SM_ECNS_COPP_TOPOLOGY 0x00010F71
+#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY 0x00010F72
+#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY 0x00010F75
+
+/* SRS TRUMEDIA GUIDS */
+/* topology */
+#define SRS_TRUMEDIA_TOPOLOGY_ID 0x00010D90
+/* module */
+#define SRS_TRUMEDIA_MODULE_ID 0x10005010
+/* parameters */
+#define SRS_TRUMEDIA_PARAMS 0x10005011
+#define SRS_TRUMEDIA_PARAMS_WOWHD 0x10005012
+#define SRS_TRUMEDIA_PARAMS_CSHP 0x10005013
+#define SRS_TRUMEDIA_PARAMS_HPF 0x10005014
+#define SRS_TRUMEDIA_PARAMS_PEQ 0x10005015
+#define SRS_TRUMEDIA_PARAMS_HL 0x10005016
+
+#define ASM_MAX_EQ_BANDS 12
+
+struct asm_eq_band {
+ u32 band_idx; /* The band index, 0 .. 11 */
+ u32 filter_type; /* Filter band type */
+ u32 center_freq_hz; /* Filter band center frequency */
+ u32 filter_gain; /* Filter band initial gain (dB) */
+ /* Range is +12 dB to -12 dB with 1dB increments. */
+ u32 q_factor;
+} __attribute__ ((packed));
+
+struct asm_equalizer_params {
+ u32 enable;
+ u32 num_bands;
+ struct asm_eq_band eq_bands[ASM_MAX_EQ_BANDS];
+} __attribute__ ((packed));
+
+struct asm_master_gain_params {
+ u16 master_gain;
+ u16 padding;
+} __attribute__ ((packed));
+
+struct asm_lrchannel_gain_params {
+ u16 left_gain;
+ u16 right_gain;
+} __attribute__ ((packed));
+
+struct asm_mute_params {
+ u32 muteflag;
+} __attribute__ ((packed));
+
+struct asm_softvolume_params {
+ u32 period;
+ u32 step;
+ u32 rampingcurve;
+} __attribute__ ((packed));
+
+struct asm_softpause_params {
+ u32 enable;
+ u32 period;
+ u32 step;
+ u32 rampingcurve;
+} __packed;
+
+struct asm_pp_param_data_hdr {
+ u32 module_id;
+ u32 param_id;
+ u16 param_size;
+ u16 reserved;
+} __attribute__ ((packed));
+
+struct asm_pp_params_command {
+ struct apr_hdr hdr;
+ u32 *payload;
+ u32 payload_size;
+ struct asm_pp_param_data_hdr params;
+} __attribute__ ((packed));
+
+#define EQUALIZER_MODULE_ID 0x00010c27
+#define EQUALIZER_PARAM_ID 0x00010c28
+
+#define VOLUME_CONTROL_MODULE_ID 0x00010bfe
+#define MASTER_GAIN_PARAM_ID 0x00010bff
+#define L_R_CHANNEL_GAIN_PARAM_ID 0x00010c00
+#define MUTE_CONFIG_PARAM_ID 0x00010c01
+#define SOFT_PAUSE_PARAM_ID 0x00010D6A
+#define SOFT_VOLUME_PARAM_ID 0x00010C29
+
+#define IIR_FILTER_ENABLE_PARAM_ID 0x00010c03
+#define IIR_FILTER_PREGAIN_PARAM_ID 0x00010c04
+#define IIR_FILTER_CONFIG_PARAM_ID 0x00010c05
+
+#define MBADRC_MODULE_ID 0x00010c06
+#define MBADRC_ENABLE_PARAM_ID 0x00010c07
+#define MBADRC_CONFIG_PARAM_ID 0x00010c08
+
+
+#define ADM_CMD_SET_PARAMS 0x00010306
+#define ADM_CMD_GET_PARAMS 0x0001030B
+#define ADM_CMDRSP_GET_PARAMS 0x0001030C
+struct adm_set_params_command {
+ struct apr_hdr hdr;
+ u32 payload;
+ u32 payload_size;
+} __attribute__ ((packed));
+
+
+#define ADM_CMD_TAP_COPP_PCM 0x00010307
+struct adm_tap_copp_pcm_command {
+ struct apr_hdr hdr;
+} __attribute__ ((packed));
+
+
+/* QDSP6 to Client messages
+*/
+#define ADM_SERVICE_CMDRSP_GET_COPP_HANDLES 0x00010308
+struct adm_get_copp_handles_respond {
+ struct apr_hdr hdr;
+ u32 handles;
+ u32 copp_id;
+} __attribute__ ((packed));
+
+#define ADM_CMDRSP_COPP_OPEN 0x0001030A
+struct adm_copp_open_respond {
+ u32 status;
+ u16 copp_id;
+ u16 reserved;
+} __attribute__ ((packed));
+
+#define ADM_CMDRSP_MULTI_CHANNEL_COPP_OPEN 0x00010311
+
+
+#define ASM_STREAM_PRIORITY_NORMAL 0
+#define ASM_STREAM_PRIORITY_LOW 1
+#define ASM_STREAM_PRIORITY_HIGH 2
+#define ASM_STREAM_PRIORITY_RESERVED 3
+
+#define ASM_END_POINT_DEVICE_MATRIX 0
+#define ASM_END_POINT_STREAM 1
+
+#define AAC_ENC_MODE_AAC_LC 0x02
+#define AAC_ENC_MODE_AAC_P 0x05
+#define AAC_ENC_MODE_EAAC_P 0x1D
+
+#define ASM_STREAM_CMD_CLOSE 0x00010BCD
+#define ASM_STREAM_CMD_FLUSH 0x00010BCE
+#define ASM_STREAM_CMD_SET_PP_PARAMS 0x00010BCF
+#define ASM_STREAM_CMD_GET_PP_PARAMS 0x00010BD0
+#define ASM_STREAM_CMDRSP_GET_PP_PARAMS 0x00010BD1
+#define ASM_SESSION_CMD_PAUSE 0x00010BD3
+#define ASM_SESSION_CMD_GET_SESSION_TIME 0x00010BD4
+#define ASM_DATA_CMD_EOS 0x00010BDB
+#define ASM_DATA_EVENT_EOS 0x00010BDD
+
+#define ASM_SERVICE_CMD_GET_STREAM_HANDLES 0x00010C0B
+#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
+
+#define ASM_SESSION_EVENT_RX_UNDERFLOW 0x00010C17
+#define ASM_SESSION_EVENT_TX_OVERFLOW 0x00010C18
+#define ASM_SERVICE_CMD_GET_WALLCLOCK_TIME 0x00010C19
+#define ASM_DATA_CMDRSP_EOS 0x00010C1C
+
+/* ASM Data structures */
+
+/* common declarations */
+struct asm_pcm_cfg {
+ u16 ch_cfg;
+ u16 bits_per_sample;
+ u32 sample_rate;
+ u16 is_signed;
+ u16 interleaved;
+};
+
+#define PCM_CHANNEL_NULL 0
+
+/* Front left channel. */
+#define PCM_CHANNEL_FL 1
+
+/* Front right channel. */
+#define PCM_CHANNEL_FR 2
+
+/* Front center channel. */
+#define PCM_CHANNEL_FC 3
+
+/* Left surround channel.*/
+#define PCM_CHANNEL_LS 4
+
+/* Right surround channel.*/
+#define PCM_CHANNEL_RS 5
+
+/* Low frequency effect channel. */
+#define PCM_CHANNEL_LFE 6
+
+/* Center surround channel; Rear center channel. */
+#define PCM_CHANNEL_CS 7
+
+/* Left back channel; Rear left channel. */
+#define PCM_CHANNEL_LB 8
+
+/* Right back channel; Rear right channel. */
+#define PCM_CHANNEL_RB 9
+
+/* Top surround channel. */
+#define PCM_CHANNEL_TS 10
+
+/* Center vertical height channel.*/
+#define PCM_CHANNEL_CVH 11
+
+/* Mono surround channel.*/
+#define PCM_CHANNEL_MS 12
+
+/* Front left of center. */
+#define PCM_CHANNEL_FLC 13
+
+/* Front right of center. */
+#define PCM_CHANNEL_FRC 14
+
+/* Rear left of center. */
+#define PCM_CHANNEL_RLC 15
+
+/* Rear right of center. */
+#define PCM_CHANNEL_RRC 16
+
+#define PCM_FORMAT_MAX_NUM_CHANNEL 8
+
+/* Maximum number of channels supported
+ * in ASM_ENCDEC_DEC_CHAN_MAP command
+ */
+#define MAX_CHAN_MAP_CHANNELS 16
+/*
+ * Multiple-channel PCM decoder format block structure used in the
+ * #ASM_STREAM_CMD_OPEN_WRITE command.
+ * The data must be in little-endian format.
+ */
+struct asm_multi_channel_pcm_fmt_blk {
+
+ u16 num_channels; /*
+ * Number of channels.
+ * Supported values:1 to 8
+ */
+
+ u16 bits_per_sample; /*
+ * Number of bits per sample per channel.
+ * Supported values: 16, 24 When used for
+ * playback, the client must send 24-bit
+ * samples packed in 32-bit words. The
+ * 24-bit samples must be placed in the most
+ * significant 24 bits of the 32-bit word. When
+ * used for recording, the aDSP sends 24-bit
+ * samples packed in 32-bit words. The 24-bit
+ * samples are placed in the most significant
+ * 24 bits of the 32-bit word.
+ */
+
+ u32 sample_rate; /*
+ * Number of samples per second
+ * (in Hertz). Supported values:
+ * 2000 to 48000
+ */
+
+ u16 is_signed; /*
+ * Flag that indicates the samples
+ * are signed (1).
+ */
+
+ u16 is_interleaved; /*
+ * Flag that indicates whether the channels are
+ * de-interleaved (0) or interleaved (1).
+ * Interleaved format means corresponding
+ * samples from the left and right channels are
+ * interleaved within the buffer.
+ * De-interleaved format means samples from
+ * each channel are contiguous in the buffer.
+ * The samples from one channel immediately
+ * follow those of the previous channel.
+ */
+
+ u8 channel_mapping[8]; /*
+ * Supported values:
+ * PCM_CHANNEL_NULL, PCM_CHANNEL_FL,
+ * PCM_CHANNEL_FR, PCM_CHANNEL_FC,
+ * PCM_CHANNEL_LS, PCM_CHANNEL_RS,
+ * PCM_CHANNEL_LFE, PCM_CHANNEL_CS,
+ * PCM_CHANNEL_LB, PCM_CHANNEL_RB,
+ * PCM_CHANNEL_TS, PCM_CHANNEL_CVH,
+ * PCM_CHANNEL_MS, PCM_CHANNEL_FLC,
+ * PCM_CHANNEL_FRC, PCM_CHANNEL_RLC,
+ * PCM_CHANNEL_RRC.
+ * Channel[i] mapping describes channel I. Each
+ * element i of the array describes channel I
+ * inside the buffer where I < num_channels.
+ * An unused channel is set to zero.
+ */
+};
+
+struct asm_adpcm_cfg {
+ u16 ch_cfg;
+ u16 bits_per_sample;
+ u32 sample_rate;
+ u32 block_size;
+};
+
+struct asm_yadpcm_cfg {
+ u16 ch_cfg;
+ u16 bits_per_sample;
+ u32 sample_rate;
+};
+
+struct asm_midi_cfg {
+ u32 nMode;
+};
+
+struct asm_wma_cfg {
+ u16 format_tag;
+ u16 ch_cfg;
+ u32 sample_rate;
+ u32 avg_bytes_per_sec;
+ u16 block_align;
+ u16 valid_bits_per_sample;
+ u32 ch_mask;
+ u16 encode_opt;
+ u16 adv_encode_opt;
+ u32 adv_encode_opt2;
+ u32 drc_peak_ref;
+ u32 drc_peak_target;
+ u32 drc_ave_ref;
+ u32 drc_ave_target;
+};
+
+struct asm_wmapro_cfg {
+ u16 format_tag;
+ u16 ch_cfg;
+ u32 sample_rate;
+ u32 avg_bytes_per_sec;
+ u16 block_align;
+ u16 valid_bits_per_sample;
+ u32 ch_mask;
+ u16 encode_opt;
+ u16 adv_encode_opt;
+ u32 adv_encode_opt2;
+ u32 drc_peak_ref;
+ u32 drc_peak_target;
+ u32 drc_ave_ref;
+ u32 drc_ave_target;
+};
+
+struct asm_aac_cfg {
+ u16 format;
+ u16 aot;
+ u16 ep_config;
+ u16 section_data_resilience;
+ u16 scalefactor_data_resilience;
+ u16 spectral_data_resilience;
+ u16 ch_cfg;
+ u16 reserved;
+ u32 sample_rate;
+};
+
+struct asm_flac_cfg {
+ u16 stream_info_present;
+ u16 min_blk_size;
+ u16 max_blk_size;
+ u16 ch_cfg;
+ u16 sample_size;
+ u16 sample_rate;
+ u16 md5_sum;
+ u32 ext_sample_rate;
+ u32 min_frame_size;
+ u32 max_frame_size;
+};
+
+struct asm_vorbis_cfg {
+ u32 ch_cfg;
+ u32 bit_rate;
+ u32 min_bit_rate;
+ u32 max_bit_rate;
+ u16 bit_depth_pcm_sample;
+ u16 bit_stream_format;
+};
+
+struct asm_aac_read_cfg {
+ u32 bitrate;
+ u32 enc_mode;
+ u16 format;
+ u16 ch_cfg;
+ u32 sample_rate;
+};
+
+struct asm_amrnb_read_cfg {
+ u16 mode;
+ u16 dtx_mode;
+};
+
+struct asm_amrwb_read_cfg {
+ u16 mode;
+ u16 dtx_mode;
+};
+
+struct asm_evrc_read_cfg {
+ u16 max_rate;
+ u16 min_rate;
+ u16 rate_modulation_cmd;
+ u16 reserved;
+};
+
+struct asm_qcelp13_read_cfg {
+ u16 max_rate;
+ u16 min_rate;
+ u16 reduced_rate_level;
+ u16 rate_modulation_cmd;
+};
+
+struct asm_sbc_read_cfg {
+ u32 subband;
+ u32 block_len;
+ u32 ch_mode;
+ u32 alloc_method;
+ u32 bit_rate;
+ u32 sample_rate;
+};
+
+struct asm_sbc_bitrate {
+ u32 bitrate;
+};
+
+struct asm_immed_decode {
+ u32 mode;
+};
+
+struct asm_sbr_ps {
+ u32 enable;
+};
+
+struct asm_dual_mono {
+ u16 sce_left;
+ u16 sce_right;
+};
+
+struct asm_dec_chan_map {
+ u32 num_channels; /* Number of decoder output
+ * channels. 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];/* Channel array of size
+ * num_channels. It can grow
+ * till MAX_CHAN_MAP_CHANNELS.
+ * Channel[i] mapping
+ * describes channel I inside
+ * the decoder output buffer.
+ * Valid channel mapping
+ * values are to be present at
+ * the beginning of the array.
+ * All remaining elements of
+ * the array are to be filled
+ * with PCM_CHANNEL_NULL.
+ */
+};
+
+struct asm_encode_cfg_blk {
+ u32 frames_per_buf;
+ u32 format_id;
+ u32 cfg_size;
+ union {
+ struct asm_pcm_cfg pcm;
+ struct asm_aac_read_cfg aac;
+ struct asm_amrnb_read_cfg amrnb;
+ struct asm_evrc_read_cfg evrc;
+ struct asm_qcelp13_read_cfg qcelp13;
+ struct asm_sbc_read_cfg sbc;
+ struct asm_amrwb_read_cfg amrwb;
+ struct asm_multi_channel_pcm_fmt_blk mpcm;
+ } __attribute__((packed)) cfg;
+};
+
+struct asm_frame_meta_info {
+ u32 offset_to_frame;
+ u32 frame_size;
+ u32 encoded_pcm_samples;
+ u32 msw_ts;
+ u32 lsw_ts;
+ u32 nflags;
+};
+
+/* Stream level commands */
+#define ASM_STREAM_CMD_OPEN_READ 0x00010BCB
+struct asm_stream_cmd_open_read {
+ struct apr_hdr hdr;
+ u32 uMode;
+ u32 src_endpoint;
+ u32 pre_proc_top;
+ u32 format;
+} __attribute__((packed));
+
+/* Supported formats */
+#define LINEAR_PCM 0x00010BE5
+#define DTMF 0x00010BE6
+#define ADPCM 0x00010BE7
+#define YADPCM 0x00010BE8
+#define MP3 0x00010BE9
+#define MPEG4_AAC 0x00010BEA
+#define AMRNB_FS 0x00010BEB
+#define AMRWB_FS 0x00010BEC
+#define V13K_FS 0x00010BED
+#define EVRC_FS 0x00010BEE
+#define EVRCB_FS 0x00010BEF
+#define EVRCWB_FS 0x00010BF0
+#define MIDI 0x00010BF1
+#define SBC 0x00010BF2
+#define WMA_V10PRO 0x00010BF3
+#define WMA_V9 0x00010BF4
+#define AMR_WB_PLUS 0x00010BF5
+#define AC3_DECODER 0x00010BF6
+#define EAC3_DECODER 0x00010C3C
+#define DTS 0x00010D88
+#define ATRAC 0x00010D89
+#define MAT 0x00010D8A
+#define G711_ALAW_FS 0x00010BF7
+#define G711_MLAW_FS 0x00010BF8
+#define G711_PCM_FS 0x00010BF9
+#define MPEG4_MULTI_AAC 0x00010D86
+#define US_POINT_EPOS_FORMAT 0x00012310
+#define US_RAW_FORMAT 0x0001127C
+#define MULTI_CHANNEL_PCM 0x00010C66
+
+#define ASM_ENCDEC_SBCRATE 0x00010C13
+#define ASM_ENCDEC_IMMDIATE_DECODE 0x00010C14
+#define ASM_ENCDEC_CFG_BLK 0x00010C2C
+
+#define ASM_ENCDEC_SBCRATE 0x00010C13
+#define ASM_ENCDEC_IMMDIATE_DECODE 0x00010C14
+#define ASM_ENCDEC_CFG_BLK 0x00010C2C
+
+#define ASM_STREAM_CMD_OPEN_WRITE 0x00010BCA
+struct asm_stream_cmd_open_write {
+ struct apr_hdr hdr;
+ u32 uMode;
+ u16 sink_endpoint;
+ u16 stream_handle;
+ u32 post_proc_top;
+ u32 format;
+} __attribute__((packed));
+
+#define IEC_61937_MASK 0x00000001
+#define IEC_60958_MASK 0x00000002
+
+#define ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED 0x00010D84
+struct asm_stream_cmd_open_write_compressed {
+ struct apr_hdr hdr;
+ u32 flags;
+ u32 format;
+} __packed;
+
+#define ASM_STREAM_CMD_OPEN_READWRITE 0x00010BCC
+
+struct asm_stream_cmd_open_read_write {
+ struct apr_hdr hdr;
+ u32 uMode;
+ u32 post_proc_top;
+ u32 write_format;
+ u32 read_format;
+} __attribute__((packed));
+
+#define ADM_CMD_CONNECT_AFE_PORT 0x00010320
+
+struct adm_cmd_connect_afe_port {
+ struct apr_hdr hdr;
+ u8 mode; /*mode represent the interface is for RX or TX*/
+ u8 session_id; /*ASM session ID*/
+ u16 afe_port_id;
+} __packed;
+
+#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
+#define ASM_STREAM_CMD_GET_ENCDEC_PARAM 0x00010C11
+#define ASM_ENCDEC_CFG_BLK_ID 0x00010C2C
+#define ASM_ENABLE_SBR_PS 0x00010C63
+#define ASM_CONFIGURE_DUAL_MONO 0x00010C64
+struct asm_stream_cmd_encdec_cfg_blk{
+ struct apr_hdr hdr;
+ u32 param_id;
+ u32 param_size;
+ struct asm_encode_cfg_blk enc_blk;
+} __attribute__((packed));
+
+struct asm_stream_cmd_encdec_sbc_bitrate{
+ struct apr_hdr hdr;
+ u32 param_id;
+ struct asm_sbc_bitrate sbc_bitrate;
+} __attribute__((packed));
+
+struct asm_stream_cmd_encdec_immed_decode{
+ struct apr_hdr hdr;
+ u32 param_id;
+ u32 param_size;
+ struct asm_immed_decode dec;
+} __attribute__((packed));
+
+struct asm_stream_cmd_encdec_sbr{
+ struct apr_hdr hdr;
+ u32 param_id;
+ u32 param_size;
+ struct asm_sbr_ps sbr_ps;
+} __attribute__((packed));
+
+struct asm_stream_cmd_encdec_dualmono {
+ struct apr_hdr hdr;
+ u32 param_id;
+ u32 param_size;
+ struct asm_dual_mono channel_map;
+} __packed;
+
+#define ASM_ENCDEC_DEC_CHAN_MAP 0x00010D82
+struct asm_stream_cmd_encdec_channelmap {
+ struct apr_hdr hdr;
+ u32 param_id;
+ u32 param_size;
+ struct asm_dec_chan_map chan_map;
+} __packed;
+
+#define ASM_STREAM _CMD_ADJUST_SAMPLES 0x00010C0A
+struct asm_stream_cmd_adjust_samples{
+ struct apr_hdr hdr;
+ u16 nsamples;
+ u16 reserved;
+} __attribute__((packed));
+
+#define ASM_STREAM_CMD_TAP_POPP_PCM 0x00010BF9
+struct asm_stream_cmd_tap_popp_pcm{
+ struct apr_hdr hdr;
+ u16 enable;
+ u16 reserved;
+ u32 module_id;
+} __attribute__((packed));
+
+/* Session Level commands */
+#define ASM_SESSION_CMD_MEMORY_MAP 0x00010C32
+struct asm_stream_cmd_memory_map{
+ struct apr_hdr hdr;
+ u32 buf_add;
+ u32 buf_size;
+ u16 mempool_id;
+ u16 reserved;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_MEMORY_UNMAP 0x00010C33
+struct asm_stream_cmd_memory_unmap{
+ struct apr_hdr hdr;
+ u32 buf_add;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_MEMORY_MAP_REGIONS 0x00010C45
+struct asm_memory_map_regions{
+ u32 phys;
+ u32 buf_size;
+} __attribute__((packed));
+
+struct asm_stream_cmd_memory_map_regions{
+ struct apr_hdr hdr;
+ u16 mempool_id;
+ u16 nregions;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_MEMORY_UNMAP_REGIONS 0x00010C46
+struct asm_memory_unmap_regions{
+ u32 phys;
+} __attribute__((packed));
+
+struct asm_stream_cmd_memory_unmap_regions{
+ struct apr_hdr hdr;
+ u16 nregions;
+ u16 reserved;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_RUN 0x00010BD2
+struct asm_stream_cmd_run{
+ struct apr_hdr hdr;
+ u32 flags;
+ u32 msw_ts;
+ u32 lsw_ts;
+} __attribute__((packed));
+
+/* Session level events */
+#define ASM_SESSION_CMD_REGISTER_FOR_RX_UNDERFLOW_EVENTS 0x00010BD5
+struct asm_stream_cmd_reg_rx_underflow_event{
+ struct apr_hdr hdr;
+ u16 enable;
+ u16 reserved;
+} __attribute__((packed));
+
+#define ASM_SESSION_CMD_REGISTER_FOR_TX_OVERFLOW_EVENTS 0x00010BD6
+struct asm_stream_cmd_reg_tx_overflow_event{
+ struct apr_hdr hdr;
+ u16 enable;
+ u16 reserved;
+} __attribute__((packed));
+
+/* Data Path commands */
+#define ASM_DATA_CMD_WRITE 0x00010BD9
+struct asm_stream_cmd_write{
+ struct apr_hdr hdr;
+ u32 buf_add;
+ u32 avail_bytes;
+ u32 uid;
+ u32 msw_ts;
+ u32 lsw_ts;
+ u32 uflags;
+} __attribute__((packed));
+
+#define ASM_DATA_CMD_READ 0x00010BDA
+struct asm_stream_cmd_read{
+ struct apr_hdr hdr;
+ u32 buf_add;
+ u32 buf_size;
+ u32 uid;
+} __attribute__((packed));
+
+#define ASM_DATA_CMD_MEDIA_FORMAT_UPDATE 0x00010BDC
+#define ASM_DATA_EVENT_ENC_SR_CM_NOTIFY 0x00010BDE
+struct asm_stream_media_format_update{
+ struct apr_hdr hdr;
+ u32 format;
+ u32 cfg_size;
+ union {
+ struct asm_pcm_cfg pcm_cfg;
+ struct asm_adpcm_cfg adpcm_cfg;
+ struct asm_yadpcm_cfg yadpcm_cfg;
+ struct asm_midi_cfg midi_cfg;
+ struct asm_wma_cfg wma_cfg;
+ struct asm_wmapro_cfg wmapro_cfg;
+ struct asm_aac_cfg aac_cfg;
+ struct asm_flac_cfg flac_cfg;
+ struct asm_vorbis_cfg vorbis_cfg;
+ struct asm_multi_channel_pcm_fmt_blk multi_ch_pcm_cfg;
+ } __attribute__((packed)) write_cfg;
+} __attribute__((packed));
+
+
+/* Command Responses */
+#define ASM_STREAM_CMDRSP_GET_ENCDEC_PARAM 0x00010C12
+struct asm_stream_cmdrsp_get_readwrite_param{
+ struct apr_hdr hdr;
+ u32 status;
+ u32 param_id;
+ u16 param_size;
+ u16 padding;
+ union {
+ struct asm_sbc_bitrate sbc_bitrate;
+ struct asm_immed_decode aac_dec;
+ } __attribute__((packed)) read_write_cfg;
+} __attribute__((packed));
+
+
+#define ASM_SESSION_CMDRSP_GET_SESSION_TIME 0x00010BD8
+struct asm_stream_cmdrsp_get_session_time{
+ struct apr_hdr hdr;
+ u32 status;
+ u32 msw_ts;
+ u32 lsw_ts;
+} __attribute__((packed));
+
+#define ASM_DATA_EVENT_WRITE_DONE 0x00010BDF
+struct asm_data_event_write_done{
+ u32 buf_add;
+ u32 status;
+} __attribute__((packed));
+
+#define ASM_DATA_EVENT_READ_DONE 0x00010BE0
+struct asm_data_event_read_done{
+ u32 status;
+ u32 buffer_add;
+ u32 enc_frame_size;
+ u32 offset;
+ u32 msw_ts;
+ u32 lsw_ts;
+ u32 flags;
+ u32 num_frames;
+ u32 id;
+} __attribute__((packed));
+
+#define ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY 0x00010C65
+struct asm_data_event_sr_cm_change_notify {
+ u32 sample_rate;
+ u16 no_of_channels;
+ u16 reserved;
+ u8 channel_map[8];
+} __packed;
+
+/* service level events */
+
+#define ASM_SERVICE_CMDRSP_GET_STREAM_HANDLES 0x00010C1B
+struct asm_svc_cmdrsp_get_strm_handles{
+ struct apr_hdr hdr;
+ u32 num_handles;
+ u32 stream_handles;
+} __attribute__((packed));
+
+
+#define ASM_SERVICE_CMDRSP_GET_WALLCLOCK_TIME 0x00010C1A
+struct asm_svc_cmdrsp_get_wallclock_time{
+ struct apr_hdr hdr;
+ u32 status;
+ u32 msw_ts;
+ u32 lsw_ts;
+} __attribute__((packed));
+
+/*
+ * Error code
+*/
+#define ADSP_EOK 0x00000000 /* Success / completed / no errors. */
+#define ADSP_EFAILED 0x00000001 /* General failure. */
+#define ADSP_EBADPARAM 0x00000002 /* Bad operation parameter(s). */
+#define ADSP_EUNSUPPORTED 0x00000003 /* Unsupported routine/operation. */
+#define ADSP_EVERSION 0x00000004 /* Unsupported version. */
+#define ADSP_EUNEXPECTED 0x00000005 /* Unexpected problem encountered. */
+#define ADSP_EPANIC 0x00000006 /* Unhandled problem occurred. */
+#define ADSP_ENORESOURCE 0x00000007 /* Unable to allocate resource(s). */
+#define ADSP_EHANDLE 0x00000008 /* Invalid handle. */
+#define ADSP_EALREADY 0x00000009 /* Operation is already processed. */
+#define ADSP_ENOTREADY 0x0000000A /* Operation not ready to be processed*/
+#define ADSP_EPENDING 0x0000000B /* Operation is pending completion*/
+#define ADSP_EBUSY 0x0000000C /* Operation could not be accepted or
+ processed. */
+#define ADSP_EABORTED 0x0000000D /* Operation aborted due to an error. */
+#define ADSP_EPREEMPTED 0x0000000E /* Operation preempted by higher priority*/
+#define ADSP_ECONTINUE 0x0000000F /* Operation requests intervention
+ to complete. */
+#define ADSP_EIMMEDIATE 0x00000010 /* Operation requests immediate
+ intervention to complete. */
+#define ADSP_ENOTIMPL 0x00000011 /* Operation is not implemented. */
+#define ADSP_ENEEDMORE 0x00000012 /* Operation needs more data or resources*/
+
+/* SRS TRUMEDIA start */
+#define SRS_ID_GLOBAL 0x00000001
+#define SRS_ID_WOWHD 0x00000002
+#define SRS_ID_CSHP 0x00000003
+#define SRS_ID_HPF 0x00000004
+#define SRS_ID_PEQ 0x00000005
+#define SRS_ID_HL 0x00000006
+
+#define SRS_CMD_UPLOAD 0x7FFF0000
+#define SRS_PARAM_INDEX_MASK 0x80000000
+#define SRS_PARAM_OFFSET_MASK 0x3FFF0000
+#define SRS_PARAM_VALUE_MASK 0x0000FFFF
+
+struct srs_trumedia_params_GLOBAL {
+ uint8_t v1;
+ uint8_t v2;
+ uint8_t v3;
+ uint8_t v4;
+ uint8_t v5;
+ uint8_t v6;
+ uint8_t v7;
+ uint8_t v8;
+} __packed;
+
+struct srs_trumedia_params_WOWHD {
+ uint32_t v1;
+ uint16_t v2;
+ uint16_t v3;
+ uint16_t v4;
+ uint16_t v5;
+ uint16_t v6;
+ uint16_t v7;
+ uint16_t v8;
+ uint16_t v____A1;
+ uint32_t v9;
+ uint16_t v10;
+ uint16_t v11;
+ uint32_t v12[16];
+} __packed;
+
+struct srs_trumedia_params_CSHP {
+ uint32_t v1;
+ uint16_t v2;
+ uint16_t v3;
+ uint16_t v4;
+ uint16_t v5;
+ uint16_t v6;
+ uint16_t v____A1;
+ uint32_t v7;
+ uint16_t v8;
+ uint16_t v9;
+ uint32_t v10[16];
+} __packed;
+
+struct srs_trumedia_params_HPF {
+ uint32_t v1;
+ uint32_t v2[26];
+} __packed;
+
+struct srs_trumedia_params_PEQ {
+ uint32_t v1;
+ uint16_t v2;
+ uint16_t v3;
+ uint16_t v4;
+ uint16_t v____A1;
+ uint32_t v5[26];
+ uint32_t v6[26];
+} __packed;
+
+struct srs_trumedia_params_HL {
+ uint16_t v1;
+ uint16_t v2;
+ uint16_t v3;
+ uint16_t v____A1;
+ int32_t v4;
+ uint32_t v5;
+ uint16_t v6;
+ uint16_t v____A2;
+ uint32_t v7;
+} __packed;
+
+struct srs_trumedia_params {
+ struct srs_trumedia_params_GLOBAL global;
+ struct srs_trumedia_params_WOWHD wowhd;
+ struct srs_trumedia_params_CSHP cshp;
+ struct srs_trumedia_params_HPF hpf;
+ struct srs_trumedia_params_PEQ peq;
+ struct srs_trumedia_params_HL hl;
+} __packed;
+int srs_trumedia_open(int port_id, int srs_tech_id, void *srs_params);
+/* SRS TruMedia end */
+
+#endif /*_APR_AUDIO_H_*/
diff --git a/include/sound/compress_offload.h b/include/sound/compress_offload.h
index 05341a4..8d36c42 100644
--- a/include/sound/compress_offload.h
+++ b/include/sound/compress_offload.h
@@ -70,6 +70,7 @@
snd_pcm_uframes_t pcm_frames;
snd_pcm_uframes_t pcm_io_frames;
__u32 sampling_rate;
+ uint64_t timestamp;
};
/**
diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h
index da4a456..5aa7b09 100644
--- a/include/sound/compress_params.h
+++ b/include/sound/compress_params.h
@@ -51,8 +51,6 @@
#ifndef __SND_COMPRESS_PARAMS_H
#define __SND_COMPRESS_PARAMS_H
-#include <linux/types.h>
-
/* AUDIO CODECS SUPPORTED */
#define MAX_NUM_CODECS 32
#define MAX_NUM_CODEC_DESCRIPTORS 32
@@ -72,7 +70,10 @@
#define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B)
#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C)
#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D)
-
+#define SND_AUDIOCODEC_AC3 ((__u32) 0x0000000E)
+#define SND_AUDIOCODEC_DTS ((__u32) 0x0000000F)
+#define SND_AUDIOCODEC_AC3_PASS_THROUGH ((__u32) 0x00000010)
+#define SND_AUDIOCODEC_WMA_PRO ((__u32) 0x00000011)
/*
* Profile and modes are listed with bit masks. This allows for a
* more compact representation of fields that will not evolve
@@ -237,6 +238,9 @@
struct snd_enc_wma {
__u32 super_block_align; /* WMA Type-specific data */
+ __u32 bits_per_sample;
+ __u32 channelmask;
+ __u32 encodeopt;
};
diff --git a/include/sound/control.h b/include/sound/control.h
index 8332e86..1318164 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -40,7 +40,7 @@
snd_ctl_elem_iface_t iface; /* interface identifier */
unsigned int device; /* device/client number */
unsigned int subdevice; /* subdevice (substream) number */
- const unsigned char *name; /* ASCII name of item */
+ unsigned char *name; /* ASCII name of item */
unsigned int index; /* index of item */
unsigned int access; /* access rights */
unsigned int count; /* count of same elements */
diff --git a/include/sound/cs8427.h b/include/sound/cs8427.h
index f862cff..2004ec3 100644
--- a/include/sound/cs8427.h
+++ b/include/sound/cs8427.h
@@ -108,6 +108,7 @@
#define CS8427_SIDEL (1<<2) /* Delay of SDIN data relative to ILRCK for left-justified data formats, 0 = first ISCLK period, 1 = second ISCLK period */
#define CS8427_SISPOL (1<<1) /* ICLK clock polarity, 0 = rising edge of ISCLK, 1 = falling edge of ISCLK */
#define CS8427_SILRPOL (1<<0) /* ILRCK clock polarity, 0 = SDIN data left channel when ILRCK is high, 1 = SDIN right when ILRCK is high */
+#define CS8427_BITWIDTH_MASK 0xCF
/* CS8427_REG_SERIALOUTPUT */
#define CS8427_SOMS (1<<7) /* 0 = slave, 1 = master mode */
@@ -186,6 +187,31 @@
#define CS8427_VERSHIFT 0
#define CS8427_VER8427A 0x71
+/* possible address cs8427 can take
+ * based on the below combinations the upper four bits of 7bit
+ * address will be fixed for 0010b, abd lower 3 bits will decide
+ * the address combination based on the AD0 and AD1 and EMPH(AD2)
+ * Hardware pin configuration to cs8427 chip
+ */
+#define CS8427_ADDR0 0x10
+#define CS8427_ADDR1 0x11
+#define CS8427_ADDR2 0x12
+#define CS8427_ADDR3 0x13
+#define CS8427_ADDR4 0x14
+#define CS8427_ADDR5 0x15
+#define CS8427_ADDR6 0x16
+#define CS8427_ADDR7 0x17
+
+#define CHANNEL_STATUS_SIZE 24
+
+struct cs8427_platform_data {
+ int irq;
+ int irq_base;
+ int num_irqs;
+ int reset_gpio;
+ int (*enable) (int enable);
+};
+
struct snd_pcm_substream;
int snd_cs8427_create(struct snd_i2c_bus *bus, unsigned char addr,
@@ -197,5 +223,4 @@
struct snd_pcm_substream *capture_substream);
int snd_cs8427_iec958_active(struct snd_i2c_device *cs8427, int active);
int snd_cs8427_iec958_pcm(struct snd_i2c_device *cs8427, unsigned int rate);
-
#endif /* __SOUND_CS8427_H */
diff --git a/include/sound/dai.h b/include/sound/dai.h
new file mode 100644
index 0000000..4d3fb96
--- /dev/null
+++ b/include/sound/dai.h
@@ -0,0 +1,49 @@
+/* Copyright (c) 2010-2011, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ *
+ */
+#ifndef __DAI_H__
+#define __DAI_H__
+
+struct dai_dma_params {
+ u8 *buffer;
+ uint32_t src_start;
+ uint32_t bus_id;
+ int buffer_size;
+ int period_size;
+ int channels;
+};
+
+enum {
+ DAI_SPKR = 0,
+ DAI_MIC,
+ DAI_MI2S,
+ DAI_SEC_SPKR,
+ DAI_SEC_MIC,
+};
+
+/* Function Prototypes */
+int dai_open(uint32_t dma_ch);
+void dai_close(uint32_t dma_ch);
+int dai_start(uint32_t dma_ch);
+int dai_stop(uint32_t dma_ch);
+int dai_set_params(uint32_t dma_ch, struct dai_dma_params *params);
+uint32_t dai_get_dma_pos(uint32_t dma_ch);
+void register_dma_irq_handler(int dma_ch,
+ irqreturn_t (*callback) (int intrSrc, void *private_data),
+ void *private_data);
+void unregister_dma_irq_handler(int dma_ch);
+void dai_set_master_mode(uint32_t dma_ch, int mode);
+int dai_start_hdmi(uint32_t dma_ch);
+int wait_for_dma_cnt_stop(uint32_t dma_ch);
+void dai_stop_hdmi(uint32_t dma_ch);
+
+#endif
diff --git a/include/sound/jack.h b/include/sound/jack.h
index 5891657..1089ba4 100644
--- a/include/sound/jack.h
+++ b/include/sound/jack.h
@@ -35,27 +35,29 @@
* sound/core/jack.c.
*/
enum snd_jack_types {
- SND_JACK_HEADPHONE = 0x0001,
- SND_JACK_MICROPHONE = 0x0002,
+ SND_JACK_HEADPHONE = 0x0000001,
+ SND_JACK_MICROPHONE = 0x0000002,
SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE,
- SND_JACK_LINEOUT = 0x0004,
- SND_JACK_MECHANICAL = 0x0008, /* If detected separately */
- SND_JACK_VIDEOOUT = 0x0010,
+ SND_JACK_LINEOUT = 0x0000004,
+ SND_JACK_MECHANICAL = 0x0000008, /* If detected separately */
+ SND_JACK_VIDEOOUT = 0x0000010,
SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_VIDEOOUT,
- SND_JACK_LINEIN = 0x0020,
-
+ /* */
+ SND_JACK_LINEIN = 0x0000020,
+ SND_JACK_OC_HPHL = 0x0000040,
+ SND_JACK_OC_HPHR = 0x0000080,
+ SND_JACK_UNSUPPORTED = 0x0000100,
/* Kept separate from switches to facilitate implementation */
- SND_JACK_BTN_0 = 0x4000,
- SND_JACK_BTN_1 = 0x2000,
- SND_JACK_BTN_2 = 0x1000,
- SND_JACK_BTN_3 = 0x0800,
- SND_JACK_BTN_4 = 0x0400,
- SND_JACK_BTN_5 = 0x0200,
+ SND_JACK_BTN_0 = 0x4000000,
+ SND_JACK_BTN_1 = 0x2000000,
+ SND_JACK_BTN_2 = 0x1000000,
+ SND_JACK_BTN_3 = 0x0800000,
+ SND_JACK_BTN_4 = 0x0400000,
+ SND_JACK_BTN_5 = 0x0200000,
+ SND_JACK_BTN_6 = 0x0100000,
+ SND_JACK_BTN_7 = 0x0080000,
};
-/* Keep in sync with definitions above */
-#define SND_JACK_SWITCH_TYPES 6
-
struct snd_jack {
struct input_dev *input_dev;
int registered;
diff --git a/include/sound/msm-dai-q6-v2.h b/include/sound/msm-dai-q6-v2.h
new file mode 100644
index 0000000..3d5ffdd
--- /dev/null
+++ b/include/sound/msm-dai-q6-v2.h
@@ -0,0 +1,42 @@
+/* Copyright (c) 2012, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+
+#ifndef __MSM_DAI_Q6_PDATA_H__
+
+#define __MSM_DAI_Q6_PDATA_H__
+
+#define MSM_MI2S_SD0 (1 << 0)
+#define MSM_MI2S_SD1 (1 << 1)
+#define MSM_MI2S_SD2 (1 << 2)
+#define MSM_MI2S_SD3 (1 << 3)
+#define MSM_MI2S_CAP_RX 0
+#define MSM_MI2S_CAP_TX 1
+
+struct msm_dai_auxpcm_pdata {
+ const char *clk;
+ u16 mode;
+ u16 sync;
+ u16 frame;
+ u16 quant;
+ /* modify slot to arr[4] to specify
+ * the slot number for each channel
+ * in multichannel scenario */
+ u16 slot;
+ u16 data;
+ int pcm_clk_rate;
+};
+
+struct msm_i2s_data {
+ u32 capability; /* RX or TX */
+ u16 sd_lines;
+};
+#endif
diff --git a/include/sound/msm-dai-q6.h b/include/sound/msm-dai-q6.h
new file mode 100644
index 0000000..042aa6f
--- /dev/null
+++ b/include/sound/msm-dai-q6.h
@@ -0,0 +1,45 @@
+/* Copyright (c) 2011, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+
+#ifndef __MSM_DAI_Q6_PDATA_H__
+
+#define __MSM_DAI_Q6_PDATA_H__
+
+#define MSM_MI2S_SD0 (1 << 0)
+#define MSM_MI2S_SD1 (1 << 1)
+#define MSM_MI2S_SD2 (1 << 2)
+#define MSM_MI2S_SD3 (1 << 3)
+#define MSM_MI2S_CAP_RX 0
+#define MSM_MI2S_CAP_TX 1
+
+struct msm_dai_auxpcm_config {
+ u16 mode;
+ u16 sync;
+ u16 frame;
+ u16 quant;
+ u16 slot;
+ u16 data;
+ int pcm_clk_rate;
+};
+
+struct msm_mi2s_pdata {
+ u16 rx_sd_lines;
+ u16 tx_sd_lines;
+};
+
+struct msm_dai_auxpcm_pdata {
+ const char *clk;
+ struct msm_dai_auxpcm_config mode_8k;
+ struct msm_dai_auxpcm_config mode_16k;
+};
+
+#endif
diff --git a/include/sound/omap-abe-dsp.h b/include/sound/omap-abe-dsp.h
new file mode 100644
index 0000000..60c405d
--- /dev/null
+++ b/include/sound/omap-abe-dsp.h
@@ -0,0 +1,19 @@
+/*
+ * omap-aess -- OMAP4 ABE DSP
+ *
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _OMAP4_ABE_DSP_H
+#define _OMAP4_ABE_DSP_H
+
+struct omap4_abe_dsp_pdata {
+ /* Return context loss count due to PM states changing */
+ int (*get_context_loss_count)(struct device *dev);
+};
+
+#endif
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 0d11128..6cb456e 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -264,7 +264,7 @@
struct snd_pcm_hw_constraint_list {
unsigned int count;
- const unsigned int *list;
+ unsigned int *list;
unsigned int mask;
};
@@ -413,6 +413,7 @@
#endif
/* misc flags */
unsigned int hw_opened: 1;
+ unsigned int hw_no_buffer: 1; /* substream may not have a buffer */
};
#define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0)
@@ -454,7 +455,6 @@
void *private_data;
void (*private_free) (struct snd_pcm *pcm);
struct device *dev; /* actual hw device this belongs to */
- bool internal; /* pcm is for internal use only */
#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
struct snd_pcm_oss oss;
#endif
@@ -476,9 +476,9 @@
int snd_pcm_new(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count,
struct snd_pcm **rpcm);
-int snd_pcm_new_internal(struct snd_card *card, const char *id, int device,
+int snd_pcm_new_soc_be(struct snd_card *card, const char *id, int device,
int playback_count, int capture_count,
- struct snd_pcm **rpcm);
+ struct snd_pcm ** rpcm);
int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count);
int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree);
@@ -785,8 +785,7 @@
unsigned int k, struct snd_interval *c);
void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k,
const struct snd_interval *b, struct snd_interval *c);
-int snd_interval_list(struct snd_interval *i, unsigned int count,
- const unsigned int *list, unsigned int mask);
+int snd_interval_list(struct snd_interval *i, unsigned int count, unsigned int *list, unsigned int mask);
int snd_interval_ratnum(struct snd_interval *i,
unsigned int rats_count, struct snd_ratnum *rats,
unsigned int *nump, unsigned int *denp);
diff --git a/include/sound/q6adm-v2.h b/include/sound/q6adm-v2.h
new file mode 100644
index 0000000..cb2f3d7
--- /dev/null
+++ b/include/sound/q6adm-v2.h
@@ -0,0 +1,50 @@
+/* Copyright (c) 2012, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+#ifndef __Q6_ADM_V2_H__
+#define __Q6_ADM_V2_H__
+
+
+#define ADM_PATH_PLAYBACK 0x1
+#define ADM_PATH_LIVE_REC 0x2
+#define ADM_PATH_NONLIVE_REC 0x3
+#include <sound/q6audio-v2.h>
+
+#define Q6_AFE_MAX_PORTS 32
+
+/* multiple copp per stream. */
+struct route_payload {
+ unsigned int copp_ids[Q6_AFE_MAX_PORTS];
+ unsigned short num_copps;
+ unsigned int session_id;
+};
+
+int adm_open(int port, int path, int rate, int mode, int topology);
+
+int adm_multi_ch_copp_open(int port, int path, int rate, int mode,
+ int topology);
+
+int adm_memory_map_regions(int port_id, uint32_t *buf_add, uint32_t mempool_id,
+ uint32_t *bufsz, uint32_t bufcnt);
+
+int adm_memory_unmap_regions(int port_id, uint32_t *buf_add, uint32_t *bufsz,
+ uint32_t bufcnt);
+
+int adm_close(int port);
+
+int adm_matrix_map(int session_id, int path, int num_copps,
+ unsigned int *port_id, int copp_id);
+
+int adm_connect_afe_port(int mode, int session_id, int port_id);
+
+int adm_get_copp_id(int port_id);
+
+#endif /* __Q6_ADM_V2_H__ */
diff --git a/include/sound/q6adm.h b/include/sound/q6adm.h
new file mode 100644
index 0000000..29fb606
--- /dev/null
+++ b/include/sound/q6adm.h
@@ -0,0 +1,49 @@
+/* Copyright (c) 2010-2012, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+#ifndef __Q6_ADM_H__
+#define __Q6_ADM_H__
+#include <sound/q6afe.h>
+
+#define ADM_PATH_PLAYBACK 0x1
+#define ADM_PATH_LIVE_REC 0x2
+#define ADM_PATH_NONLIVE_REC 0x3
+
+/* multiple copp per stream. */
+struct route_payload {
+ unsigned int copp_ids[AFE_MAX_PORTS];
+ unsigned short num_copps;
+ unsigned int session_id;
+};
+
+int adm_open(int port, int path, int rate, int mode, int topology);
+
+int adm_multi_ch_copp_open(int port, int path, int rate, int mode,
+ int topology);
+
+int adm_memory_map_regions(uint32_t *buf_add, uint32_t mempool_id,
+ uint32_t *bufsz, uint32_t bufcnt);
+
+int adm_memory_unmap_regions(uint32_t *buf_add, uint32_t *bufsz,
+ uint32_t bufcnt);
+
+int adm_close(int port);
+
+int adm_matrix_map(int session_id, int path, int num_copps,
+ unsigned int *port_id, int copp_id);
+
+int adm_connect_afe_port(int mode, int session_id, int port_id);
+
+#ifdef CONFIG_RTAC
+int adm_get_copp_id(int port_id);
+#endif
+
+#endif /* __Q6_ADM_H__ */
diff --git a/include/sound/q6afe-v2.h b/include/sound/q6afe-v2.h
new file mode 100644
index 0000000..1587d38
--- /dev/null
+++ b/include/sound/q6afe-v2.h
@@ -0,0 +1,107 @@
+/* Copyright (c) 2012, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+#ifndef __Q6AFE_V2_H__
+#define __Q6AFE_V2_H__
+#include <sound/apr_audio-v2.h>
+
+#define MSM_AFE_MONO 0
+#define MSM_AFE_MONO_RIGHT 1
+#define MSM_AFE_MONO_LEFT 2
+#define MSM_AFE_STEREO 3
+#define MSM_AFE_4CHANNELS 4
+#define MSM_AFE_6CHANNELS 6
+#define MSM_AFE_8CHANNELS 8
+
+#define MSM_AFE_I2S_FORMAT_LPCM 0
+#define MSM_AFE_I2S_FORMAT_COMPR 1
+#define MSM_AFE_I2S_FORMAT_IEC60958_LPCM 2
+#define MSM_AFE_I2S_FORMAT_IEC60958_COMPR 3
+
+#define MSM_AFE_PORT_TYPE_RX 0
+#define MSM_AFE_PORT_TYPE_TX 1
+
+#define RT_PROXY_DAI_001_RX 0xE0
+#define RT_PROXY_DAI_001_TX 0xF0
+#define RT_PROXY_DAI_002_RX 0xF1
+#define RT_PROXY_DAI_002_TX 0xE1
+#define VIRTUAL_ID_TO_PORTID(val) ((val & 0xF) | 0x2000)
+
+enum {
+ IDX_PRIMARY_I2S_RX = 0,
+ IDX_PRIMARY_I2S_TX = 1,
+ IDX_PCM_RX = 2,
+ IDX_PCM_TX = 3,
+ IDX_SECONDARY_I2S_RX = 4,
+ IDX_SECONDARY_I2S_TX = 5,
+ IDX_MI2S_RX = 6,
+ IDX_MI2S_TX = 7,
+ IDX_HDMI_RX = 8,
+ IDX_RSVD_2 = 9,
+ IDX_RSVD_3 = 10,
+ IDX_DIGI_MIC_TX = 11,
+ IDX_VOICE_RECORD_RX = 12,
+ IDX_VOICE_RECORD_TX = 13,
+ IDX_VOICE_PLAYBACK_TX = 14,
+ IDX_SLIMBUS_0_RX = 15,
+ IDX_SLIMBUS_0_TX = 16,
+ IDX_SLIMBUS_1_RX = 17,
+ IDX_SLIMBUS_1_TX = 18,
+ IDX_SLIMBUS_2_RX = 19,
+ IDX_SLIMBUS_2_TX = 20,
+ IDX_SLIMBUS_3_RX = 21,
+ IDX_SLIMBUS_3_TX = 22,
+ IDX_SLIMBUS_4_RX = 23,
+ IDX_SLIMBUS_4_TX = 24,
+ IDX_INT_BT_SCO_RX = 25,
+ IDX_INT_BT_SCO_TX = 26,
+ IDX_INT_BT_A2DP_RX = 27,
+ IDX_INT_FM_RX = 28,
+ IDX_INT_FM_TX = 29,
+ IDX_RT_PROXY_PORT_001_RX = 30,
+ IDX_RT_PROXY_PORT_001_TX = 31,
+ AFE_MAX_PORTS
+};
+
+int afe_open(u16 port_id, union afe_port_config *afe_config, int rate);
+int afe_close(int port_id);
+int afe_loopback(u16 enable, u16 rx_port, u16 tx_port);
+int afe_sidetone(u16 tx_port_id, u16 rx_port_id, u16 enable, uint16_t gain);
+int afe_loopback_gain(u16 port_id, u16 volume);
+int afe_validate_port(u16 port_id);
+int afe_start_pseudo_port(u16 port_id);
+int afe_stop_pseudo_port(u16 port_id);
+int afe_cmd_memory_map(u32 dma_addr_p, u32 dma_buf_sz);
+int afe_cmd_memory_map_nowait(int port_id, u32 dma_addr_p, u32 dma_buf_sz);
+int afe_cmd_memory_unmap(u32 dma_addr_p);
+int afe_cmd_memory_unmap_nowait(u32 dma_addr_p);
+
+int afe_register_get_events(u16 port_id,
+ void (*cb) (uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv),
+ void *private_data);
+int afe_unregister_get_events(u16 port_id);
+int afe_rt_proxy_port_write(u32 buf_addr_p, u32 mem_map_handle, int bytes);
+int afe_rt_proxy_port_read(u32 buf_addr_p, u32 mem_map_handle, int bytes);
+int afe_port_start_nowait(u16 port_id, union afe_port_config *afe_config,
+ u32 rate);
+int afe_port_stop_nowait(int port_id);
+int afe_apply_gain(u16 port_id, u16 gain);
+int afe_q6_interface_prepare(void);
+int afe_get_port_type(u16 port_id);
+/* if port_id is virtual, convert to physical..
+ * if port_id is already physical, return physical
+ */
+int afe_convert_virtual_to_portid(u16 port_id);
+
+int afe_pseudo_port_start_nowait(u16 port_id);
+int afe_pseudo_port_stop_nowait(u16 port_id);
+#endif /* __Q6AFE_V2_H__ */
diff --git a/include/sound/q6afe.h b/include/sound/q6afe.h
new file mode 100644
index 0000000..8cdcc18
--- /dev/null
+++ b/include/sound/q6afe.h
@@ -0,0 +1,109 @@
+/* Copyright (c) 2010-2012, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+#ifndef __Q6AFE_H__
+#define __Q6AFE_H__
+#include <sound/apr_audio.h>
+
+#define MSM_AFE_MONO 0
+#define MSM_AFE_MONO_RIGHT 1
+#define MSM_AFE_MONO_LEFT 2
+#define MSM_AFE_STEREO 3
+#define MSM_AFE_4CHANNELS 4
+#define MSM_AFE_6CHANNELS 6
+#define MSM_AFE_8CHANNELS 8
+
+#define MSM_AFE_I2S_FORMAT_LPCM 0
+#define MSM_AFE_I2S_FORMAT_COMPR 1
+#define MSM_AFE_I2S_FORMAT_IEC60958_LPCM 2
+#define MSM_AFE_I2S_FORMAT_IEC60958_COMPR 3
+
+#define MSM_AFE_PORT_TYPE_RX 0
+#define MSM_AFE_PORT_TYPE_TX 1
+
+#define RT_PROXY_DAI_001_RX 0xE0
+#define RT_PROXY_DAI_001_TX 0xF0
+#define RT_PROXY_DAI_002_RX 0xF1
+#define RT_PROXY_DAI_002_TX 0xE1
+#define VIRTUAL_ID_TO_PORTID(val) ((val & 0xF) | 0x2000)
+
+enum {
+ IDX_PRIMARY_I2S_RX = 0,
+ IDX_PRIMARY_I2S_TX = 1,
+ IDX_PCM_RX = 2,
+ IDX_PCM_TX = 3,
+ IDX_SECONDARY_I2S_RX = 4,
+ IDX_SECONDARY_I2S_TX = 5,
+ IDX_MI2S_RX = 6,
+ IDX_MI2S_TX = 7,
+ IDX_HDMI_RX = 8,
+ IDX_RSVD_2 = 9,
+ IDX_RSVD_3 = 10,
+ IDX_DIGI_MIC_TX = 11,
+ IDX_VOICE_RECORD_RX = 12,
+ IDX_VOICE_RECORD_TX = 13,
+ IDX_VOICE_PLAYBACK_TX = 14,
+ IDX_SLIMBUS_0_RX = 15,
+ IDX_SLIMBUS_0_TX = 16,
+ IDX_SLIMBUS_1_RX = 17,
+ IDX_SLIMBUS_1_TX = 18,
+ IDX_SLIMBUS_2_RX = 19,
+ IDX_SLIMBUS_2_TX = 20,
+ IDX_SLIMBUS_3_RX = 21,
+ IDX_SLIMBUS_3_TX = 22,
+ IDX_SLIMBUS_4_RX = 23,
+ IDX_SLIMBUS_4_TX = 24,
+ IDX_INT_BT_SCO_RX = 25,
+ IDX_INT_BT_SCO_TX = 26,
+ IDX_INT_BT_A2DP_RX = 27,
+ IDX_INT_FM_RX = 28,
+ IDX_INT_FM_TX = 29,
+ IDX_RT_PROXY_PORT_001_RX = 30,
+ IDX_RT_PROXY_PORT_001_TX = 31,
+ AFE_MAX_PORTS
+};
+
+int afe_open(u16 port_id, union afe_port_config *afe_config, int rate);
+int afe_close(int port_id);
+int afe_loopback(u16 enable, u16 rx_port, u16 tx_port);
+int afe_loopback_cfg(u16 enable, u16 dst_port, u16 src_port, u16 mode);
+int afe_sidetone(u16 tx_port_id, u16 rx_port_id, u16 enable, uint16_t gain);
+int afe_loopback_gain(u16 port_id, u16 volume);
+int afe_validate_port(u16 port_id);
+int afe_get_port_index(u16 port_id);
+int afe_start_pseudo_port(u16 port_id);
+int afe_stop_pseudo_port(u16 port_id);
+int afe_cmd_memory_map(u32 dma_addr_p, u32 dma_buf_sz);
+int afe_cmd_memory_map_nowait(u32 dma_addr_p, u32 dma_buf_sz);
+int afe_cmd_memory_unmap(u32 dma_addr_p);
+int afe_cmd_memory_unmap_nowait(u32 dma_addr_p);
+
+int afe_register_get_events(u16 port_id,
+ void (*cb) (uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv),
+ void *private_data);
+int afe_unregister_get_events(u16 port_id);
+int afe_rt_proxy_port_write(u32 buf_addr_p, int bytes);
+int afe_rt_proxy_port_read(u32 buf_addr_p, int bytes);
+int afe_port_start_nowait(u16 port_id, union afe_port_config *afe_config,
+ u32 rate);
+int afe_port_stop_nowait(int port_id);
+int afe_apply_gain(u16 port_id, u16 gain);
+int afe_q6_interface_prepare(void);
+int afe_get_port_type(u16 port_id);
+/* if port_id is virtual, convert to physical..
+ * if port_id is already physical, return physical
+ */
+int afe_convert_virtual_to_portid(u16 port_id);
+
+int afe_pseudo_port_start_nowait(u16 port_id);
+int afe_pseudo_port_stop_nowait(u16 port_id);
+#endif /* __Q6AFE_H__ */
diff --git a/include/sound/q6asm-v2.h b/include/sound/q6asm-v2.h
new file mode 100644
index 0000000..7ef15ac
--- /dev/null
+++ b/include/sound/q6asm-v2.h
@@ -0,0 +1,303 @@
+/* Copyright (c) 2012, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+#ifndef __Q6_ASM_V2_H__
+#define __Q6_ASM_V2_H__
+
+#include <mach/qdsp6v2/apr.h>
+#include <mach/msm_subsystem_map.h>
+#include <sound/apr_audio-v2.h>
+#include <linux/list.h>
+#include <linux/ion.h>
+
+#define IN 0x000
+#define OUT 0x001
+#define CH_MODE_MONO 0x001
+#define CH_MODE_STEREO 0x002
+
+#define FORMAT_LINEAR_PCM 0x0000
+#define FORMAT_DTMF 0x0001
+#define FORMAT_ADPCM 0x0002
+#define FORMAT_YADPCM 0x0003
+#define FORMAT_MP3 0x0004
+#define FORMAT_MPEG4_AAC 0x0005
+#define FORMAT_AMRNB 0x0006
+#define FORMAT_AMRWB 0x0007
+#define FORMAT_V13K 0x0008
+#define FORMAT_EVRC 0x0009
+#define FORMAT_EVRCB 0x000a
+#define FORMAT_EVRCWB 0x000b
+#define FORMAT_MIDI 0x000c
+#define FORMAT_SBC 0x000d
+#define FORMAT_WMA_V10PRO 0x000e
+#define FORMAT_WMA_V9 0x000f
+#define FORMAT_AMR_WB_PLUS 0x0010
+#define FORMAT_MPEG4_MULTI_AAC 0x0011
+#define FORMAT_MULTI_CHANNEL_LINEAR_PCM 0x0012
+
+#define ENCDEC_SBCBITRATE 0x0001
+#define ENCDEC_IMMEDIATE_DECODE 0x0002
+#define ENCDEC_CFG_BLK 0x0003
+
+#define CMD_PAUSE 0x0001
+#define CMD_FLUSH 0x0002
+#define CMD_EOS 0x0003
+#define CMD_CLOSE 0x0004
+#define CMD_OUT_FLUSH 0x0005
+
+/* bit 0:1 represents priority of stream */
+#define STREAM_PRIORITY_NORMAL 0x0000
+#define STREAM_PRIORITY_LOW 0x0001
+#define STREAM_PRIORITY_HIGH 0x0002
+
+/* bit 4 represents META enable of encoded data buffer */
+#define BUFFER_META_ENABLE 0x0010
+
+/* Enable Sample_Rate/Channel_Mode notification event from Decoder */
+#define SR_CM_NOTIFY_ENABLE 0x0004
+
+#define ASYNC_IO_MODE 0x0002
+#define SYNC_IO_MODE 0x0001
+#define NO_TIMESTAMP 0xFF00
+#define SET_TIMESTAMP 0x0000
+
+#define SOFT_PAUSE_ENABLE 1
+#define SOFT_PAUSE_DISABLE 0
+
+#define SESSION_MAX 0x08
+
+#define SOFT_PAUSE_PERIOD 30 /* ramp up/down for 30ms */
+#define SOFT_PAUSE_STEP 2000 /* Step value 2ms or 2000us */
+enum {
+ SOFT_PAUSE_CURVE_LINEAR = 0,
+ SOFT_PAUSE_CURVE_EXP,
+ SOFT_PAUSE_CURVE_LOG,
+};
+
+#define SOFT_VOLUME_PERIOD 30 /* ramp up/down for 30ms */
+#define SOFT_VOLUME_STEP 2000 /* Step value 2ms or 2000us */
+enum {
+ SOFT_VOLUME_CURVE_LINEAR = 0,
+ SOFT_VOLUME_CURVE_EXP,
+ SOFT_VOLUME_CURVE_LOG,
+};
+
+typedef void (*app_cb)(uint32_t opcode, uint32_t token,
+ uint32_t *payload, void *priv);
+
+struct audio_buffer {
+ dma_addr_t phys;
+ void *data;
+ uint32_t used;
+ uint32_t size;/* size of buffer */
+ uint32_t actual_size; /* actual number of bytes read by DSP */
+ struct ion_handle *handle;
+ struct ion_client *client;
+};
+
+struct audio_aio_write_param {
+ unsigned long paddr;
+ uint32_t len;
+ uint32_t uid;
+ uint32_t lsw_ts;
+ uint32_t msw_ts;
+ uint32_t flags;
+};
+
+struct audio_aio_read_param {
+ unsigned long paddr;
+ uint32_t len;
+ uint32_t uid;
+};
+
+struct audio_port_data {
+ struct audio_buffer *buf;
+ uint32_t max_buf_cnt;
+ uint32_t dsp_buf;
+ uint32_t cpu_buf;
+ struct list_head mem_map_handle;
+ uint32_t tmp_hdl;
+ /* read or write locks */
+ struct mutex lock;
+ spinlock_t dsp_lock;
+};
+
+struct audio_client {
+ int session;
+ app_cb cb;
+ atomic_t cmd_state;
+ /* Relative or absolute TS */
+ uint32_t time_flag;
+ void *priv;
+ uint32_t io_mode;
+ uint64_t time_stamp;
+ struct apr_svc *apr;
+ struct apr_svc *mmap_apr;
+ struct mutex cmd_lock;
+ /* idx:1 out port, 0: in port*/
+ struct audio_port_data port[2];
+ wait_queue_head_t cmd_wait;
+};
+
+void q6asm_audio_client_free(struct audio_client *ac);
+
+struct audio_client *q6asm_audio_client_alloc(app_cb cb, void *priv);
+
+struct audio_client *q6asm_get_audio_client(int session_id);
+
+int q6asm_audio_client_buf_alloc(unsigned int dir/* 1:Out,0:In */,
+ struct audio_client *ac,
+ unsigned int bufsz,
+ unsigned int bufcnt);
+int q6asm_audio_client_buf_alloc_contiguous(unsigned int dir
+ /* 1:Out,0:In */,
+ struct audio_client *ac,
+ unsigned int bufsz,
+ unsigned int bufcnt);
+
+int q6asm_audio_client_buf_free_contiguous(unsigned int dir,
+ struct audio_client *ac);
+
+int q6asm_open_read(struct audio_client *ac, uint32_t format
+ /*, uint16_t bits_per_sample*/);
+
+int q6asm_open_write(struct audio_client *ac, uint32_t format
+ /*, uint16_t bits_per_sample*/);
+
+int q6asm_open_read_write(struct audio_client *ac,
+ uint32_t rd_format,
+ uint32_t wr_format);
+
+int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags);
+int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags);
+
+int q6asm_async_write(struct audio_client *ac,
+ struct audio_aio_write_param *param);
+
+int q6asm_async_read(struct audio_client *ac,
+ struct audio_aio_read_param *param);
+
+int q6asm_read(struct audio_client *ac);
+int q6asm_read_nolock(struct audio_client *ac);
+
+int q6asm_memory_map(struct audio_client *ac, uint32_t buf_add,
+ int dir, uint32_t bufsz, uint32_t bufcnt);
+
+int q6asm_memory_unmap(struct audio_client *ac, uint32_t buf_add,
+ int dir);
+
+int q6asm_run(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts);
+
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts);
+
+int q6asm_reg_tx_overflow(struct audio_client *ac, uint16_t enable);
+
+int q6asm_cmd(struct audio_client *ac, int cmd);
+
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
+
+void *q6asm_is_cpu_buf_avail(int dir, struct audio_client *ac,
+ uint32_t *size, uint32_t *idx);
+
+void *q6asm_is_cpu_buf_avail_nolock(int dir, struct audio_client *ac,
+ uint32_t *size, uint32_t *idx);
+
+int q6asm_is_dsp_buf_avail(int dir, struct audio_client *ac);
+
+/* File format specific configurations to be added below */
+
+int q6asm_enc_cfg_blk_aac(struct audio_client *ac,
+ uint32_t frames_per_buf,
+ uint32_t sample_rate, uint32_t channels,
+ uint32_t bit_rate,
+ uint32_t mode, uint32_t format);
+
+int q6asm_enc_cfg_blk_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels);
+
+int q6asm_set_encdec_chan_map(struct audio_client *ac,
+ uint32_t num_channels);
+
+int q6asm_enable_sbrps(struct audio_client *ac,
+ uint32_t sbr_ps);
+
+int q6asm_cfg_dual_mono_aac(struct audio_client *ac,
+ uint16_t sce_left, uint16_t sce_right);
+
+int q6asm_enc_cfg_blk_qcelp(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t min_rate, uint16_t max_rate,
+ uint16_t reduced_rate_level, uint16_t rate_modulation_cmd);
+
+int q6asm_enc_cfg_blk_evrc(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t min_rate, uint16_t max_rate,
+ uint16_t rate_modulation_cmd);
+
+int q6asm_enc_cfg_blk_amrnb(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t band_mode, uint16_t dtx_enable);
+
+int q6asm_enc_cfg_blk_amrwb(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t band_mode, uint16_t dtx_enable);
+
+int q6asm_media_format_block_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels);
+
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels);
+
+int q6asm_media_format_block_aac(struct audio_client *ac,
+ struct asm_aac_cfg *cfg);
+
+int q6asm_media_format_block_multi_aac(struct audio_client *ac,
+ struct asm_aac_cfg *cfg);
+
+int q6asm_media_format_block_wma(struct audio_client *ac,
+ void *cfg);
+
+int q6asm_media_format_block_wmapro(struct audio_client *ac,
+ void *cfg);
+
+/* PP specific */
+int q6asm_equalizer(struct audio_client *ac, void *eq);
+
+/* Send Volume Command */
+int q6asm_set_volume(struct audio_client *ac, int volume);
+
+/* Set SoftPause Params */
+int q6asm_set_softpause(struct audio_client *ac,
+ struct asm_softpause_params *param);
+
+/* Set Softvolume Params */
+int q6asm_set_softvolume(struct audio_client *ac,
+ struct asm_softvolume_params *param);
+
+/* Send left-right channel gain */
+int q6asm_set_lrgain(struct audio_client *ac, int left_gain, int right_gain);
+
+/* Enable Mute/unmute flag */
+int q6asm_set_mute(struct audio_client *ac, int muteflag);
+
+uint64_t q6asm_get_session_time(struct audio_client *ac);
+
+/* Client can set the IO mode to either AIO/SIO mode */
+int q6asm_set_io_mode(struct audio_client *ac, uint32_t mode);
+
+/* Get Service ID for APR communication */
+int q6asm_get_apr_service_id(int session_id);
+
+/* Common format block without any payload
+*/
+int q6asm_media_format_block(struct audio_client *ac, uint32_t format);
+
+#endif /* __Q6_ASM_H__ */
diff --git a/include/sound/q6asm.h b/include/sound/q6asm.h
new file mode 100644
index 0000000..54a9187
--- /dev/null
+++ b/include/sound/q6asm.h
@@ -0,0 +1,318 @@
+/* Copyright (c) 2010-2012, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+#ifndef __Q6_ASM_H__
+#define __Q6_ASM_H__
+
+#include <mach/qdsp6v2/apr.h>
+#include <sound/apr_audio.h>
+#ifdef CONFIG_MSM_MULTIMEDIA_USE_ION
+#include <linux/ion.h>
+#endif
+
+#define IN 0x000
+#define OUT 0x001
+#define CH_MODE_MONO 0x001
+#define CH_MODE_STEREO 0x002
+
+#define FORMAT_LINEAR_PCM 0x0000
+#define FORMAT_DTMF 0x0001
+#define FORMAT_ADPCM 0x0002
+#define FORMAT_YADPCM 0x0003
+#define FORMAT_MP3 0x0004
+#define FORMAT_MPEG4_AAC 0x0005
+#define FORMAT_AMRNB 0x0006
+#define FORMAT_AMRWB 0x0007
+#define FORMAT_V13K 0x0008
+#define FORMAT_EVRC 0x0009
+#define FORMAT_EVRCB 0x000a
+#define FORMAT_EVRCWB 0x000b
+#define FORMAT_MIDI 0x000c
+#define FORMAT_SBC 0x000d
+#define FORMAT_WMA_V10PRO 0x000e
+#define FORMAT_WMA_V9 0x000f
+#define FORMAT_AMR_WB_PLUS 0x0010
+#define FORMAT_MPEG4_MULTI_AAC 0x0011
+#define FORMAT_MULTI_CHANNEL_LINEAR_PCM 0x0012
+#define FORMAT_AC3 0x0013
+#define FORMAT_DTS 0x0014
+#define FORMAT_EAC3 0x0015
+#define FORMAT_ATRAC 0x0016
+#define FORMAT_MAT 0x0017
+#define FORMAT_AAC 0x0018
+
+#define ENCDEC_SBCBITRATE 0x0001
+#define ENCDEC_IMMEDIATE_DECODE 0x0002
+#define ENCDEC_CFG_BLK 0x0003
+
+#define CMD_PAUSE 0x0001
+#define CMD_FLUSH 0x0002
+#define CMD_EOS 0x0003
+#define CMD_CLOSE 0x0004
+#define CMD_OUT_FLUSH 0x0005
+
+/* bit 0:1 represents priority of stream */
+#define STREAM_PRIORITY_NORMAL 0x0000
+#define STREAM_PRIORITY_LOW 0x0001
+#define STREAM_PRIORITY_HIGH 0x0002
+
+/* bit 4 represents META enable of encoded data buffer */
+#define BUFFER_META_ENABLE 0x0010
+
+/* Enable Sample_Rate/Channel_Mode notification event from Decoder */
+#define SR_CM_NOTIFY_ENABLE 0x0004
+
+#define ASYNC_IO_MODE 0x0002
+#define SYNC_IO_MODE 0x0001
+#define NO_TIMESTAMP 0xFF00
+#define SET_TIMESTAMP 0x0000
+
+#define SOFT_PAUSE_ENABLE 1
+#define SOFT_PAUSE_DISABLE 0
+
+#define SESSION_MAX 0x08
+
+#define SOFT_PAUSE_PERIOD 30 /* ramp up/down for 30ms */
+#define SOFT_PAUSE_STEP 2000 /* Step value 2ms or 2000us */
+enum {
+ SOFT_PAUSE_CURVE_LINEAR = 0,
+ SOFT_PAUSE_CURVE_EXP,
+ SOFT_PAUSE_CURVE_LOG,
+};
+
+#define SOFT_VOLUME_PERIOD 30 /* ramp up/down for 30ms */
+#define SOFT_VOLUME_STEP 2000 /* Step value 2ms or 2000us */
+enum {
+ SOFT_VOLUME_CURVE_LINEAR = 0,
+ SOFT_VOLUME_CURVE_EXP,
+ SOFT_VOLUME_CURVE_LOG,
+};
+
+typedef void (*app_cb)(uint32_t opcode, uint32_t token,
+ uint32_t *payload, void *priv);
+
+struct audio_buffer {
+ dma_addr_t phys;
+ void *data;
+ uint32_t used;
+ uint32_t size;/* size of buffer */
+ uint32_t actual_size; /* actual number of bytes read by DSP */
+#ifdef CONFIG_MSM_MULTIMEDIA_USE_ION
+ struct ion_handle *handle;
+ struct ion_client *client;
+#else
+ void *mem_buffer;
+#endif
+};
+
+struct audio_aio_write_param {
+ unsigned long paddr;
+ uint32_t uid;
+ uint32_t len;
+ uint32_t msw_ts;
+ uint32_t lsw_ts;
+ uint32_t flags;
+};
+
+struct audio_aio_read_param {
+ unsigned long paddr;
+ uint32_t len;
+ uint32_t uid;
+};
+
+struct audio_port_data {
+ struct audio_buffer *buf;
+ uint32_t max_buf_cnt;
+ uint32_t dsp_buf;
+ uint32_t cpu_buf;
+ /* read or write locks */
+ struct mutex lock;
+ spinlock_t dsp_lock;
+};
+
+struct audio_client {
+ int session;
+ /* idx:1 out port, 0: in port*/
+ struct audio_port_data port[2];
+
+ struct apr_svc *apr;
+ struct mutex cmd_lock;
+
+ atomic_t cmd_state;
+ atomic_t time_flag;
+ wait_queue_head_t cmd_wait;
+ wait_queue_head_t time_wait;
+
+ app_cb cb;
+ void *priv;
+ uint32_t io_mode;
+ uint64_t time_stamp;
+};
+
+void q6asm_audio_client_free(struct audio_client *ac);
+
+struct audio_client *q6asm_audio_client_alloc(app_cb cb, void *priv);
+
+struct audio_client *q6asm_get_audio_client(int session_id);
+
+int q6asm_audio_client_buf_alloc(unsigned int dir/* 1:Out,0:In */,
+ struct audio_client *ac,
+ unsigned int bufsz,
+ unsigned int bufcnt);
+int q6asm_audio_client_buf_alloc_contiguous(unsigned int dir
+ /* 1:Out,0:In */,
+ struct audio_client *ac,
+ unsigned int bufsz,
+ unsigned int bufcnt);
+
+int q6asm_audio_client_buf_free_contiguous(unsigned int dir,
+ struct audio_client *ac);
+
+int q6asm_open_read(struct audio_client *ac, uint32_t format);
+
+int q6asm_open_write(struct audio_client *ac, uint32_t format);
+
+int q6asm_open_write_compressed(struct audio_client *ac, uint32_t format);
+
+int q6asm_open_read_write(struct audio_client *ac,
+ uint32_t rd_format,
+ uint32_t wr_format);
+
+int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags);
+int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
+ uint32_t lsw_ts, uint32_t flags);
+
+int q6asm_async_write(struct audio_client *ac,
+ struct audio_aio_write_param *param);
+
+int q6asm_async_read(struct audio_client *ac,
+ struct audio_aio_read_param *param);
+
+int q6asm_read(struct audio_client *ac);
+int q6asm_read_nolock(struct audio_client *ac);
+
+int q6asm_memory_map(struct audio_client *ac, uint32_t buf_add,
+ int dir, uint32_t bufsz, uint32_t bufcnt);
+
+int q6asm_memory_unmap(struct audio_client *ac, uint32_t buf_add,
+ int dir);
+
+int q6asm_run(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts);
+
+int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
+ uint32_t msw_ts, uint32_t lsw_ts);
+
+int q6asm_reg_tx_overflow(struct audio_client *ac, uint16_t enable);
+
+int q6asm_cmd(struct audio_client *ac, int cmd);
+
+int q6asm_cmd_nowait(struct audio_client *ac, int cmd);
+
+void *q6asm_is_cpu_buf_avail(int dir, struct audio_client *ac,
+ uint32_t *size, uint32_t *idx);
+
+void *q6asm_is_cpu_buf_avail_nolock(int dir, struct audio_client *ac,
+ uint32_t *size, uint32_t *idx);
+
+int q6asm_is_dsp_buf_avail(int dir, struct audio_client *ac);
+
+/* File format specific configurations to be added below */
+
+int q6asm_enc_cfg_blk_aac(struct audio_client *ac,
+ uint32_t frames_per_buf,
+ uint32_t sample_rate, uint32_t channels,
+ uint32_t bit_rate,
+ uint32_t mode, uint32_t format);
+
+int q6asm_enc_cfg_blk_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels);
+
+int q6asm_enc_cfg_blk_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels);
+
+int q6asm_enable_sbrps(struct audio_client *ac,
+ uint32_t sbr_ps);
+
+int q6asm_cfg_dual_mono_aac(struct audio_client *ac,
+ uint16_t sce_left, uint16_t sce_right);
+
+int q6asm_set_encdec_chan_map(struct audio_client *ac,
+ uint32_t num_channels);
+
+int q6asm_enc_cfg_blk_qcelp(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t min_rate, uint16_t max_rate,
+ uint16_t reduced_rate_level, uint16_t rate_modulation_cmd);
+
+int q6asm_enc_cfg_blk_evrc(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t min_rate, uint16_t max_rate,
+ uint16_t rate_modulation_cmd);
+
+int q6asm_enc_cfg_blk_amrnb(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t band_mode, uint16_t dtx_enable);
+
+int q6asm_enc_cfg_blk_amrwb(struct audio_client *ac, uint32_t frames_per_buf,
+ uint16_t band_mode, uint16_t dtx_enable);
+
+int q6asm_media_format_block_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels);
+
+int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t rate, uint32_t channels);
+
+int q6asm_media_format_block_aac(struct audio_client *ac,
+ struct asm_aac_cfg *cfg);
+
+int q6asm_media_format_block_multi_aac(struct audio_client *ac,
+ struct asm_aac_cfg *cfg);
+
+int q6asm_media_format_block_wma(struct audio_client *ac,
+ void *cfg);
+
+int q6asm_media_format_block_wmapro(struct audio_client *ac,
+ void *cfg);
+
+/* PP specific */
+int q6asm_equalizer(struct audio_client *ac, void *eq);
+
+/* Send Volume Command */
+int q6asm_set_volume(struct audio_client *ac, int volume);
+
+/* Set SoftPause Params */
+int q6asm_set_softpause(struct audio_client *ac,
+ struct asm_softpause_params *param);
+
+/* Set Softvolume Params */
+int q6asm_set_softvolume(struct audio_client *ac,
+ struct asm_softvolume_params *param);
+
+/* Send left-right channel gain */
+int q6asm_set_lrgain(struct audio_client *ac, int left_gain, int right_gain);
+
+/* Enable Mute/unmute flag */
+int q6asm_set_mute(struct audio_client *ac, int muteflag);
+
+uint64_t q6asm_get_session_time(struct audio_client *ac);
+
+/* Client can set the IO mode to either AIO/SIO mode */
+int q6asm_set_io_mode(struct audio_client *ac, uint32_t mode);
+
+#ifdef CONFIG_RTAC
+/* Get Service ID for APR communication */
+int q6asm_get_apr_service_id(int session_id);
+#endif
+
+/* Common format block without any payload
+*/
+int q6asm_media_format_block(struct audio_client *ac, uint32_t format);
+
+#endif /* __Q6_ASM_H__ */
diff --git a/include/sound/q6audio-v2.h b/include/sound/q6audio-v2.h
new file mode 100644
index 0000000..1a5dce1
--- /dev/null
+++ b/include/sound/q6audio-v2.h
@@ -0,0 +1,26 @@
+/* Copyright (c) 2012, Code Aurora Forum. 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
+ * only version 2 as published by the Free Software Foundation.
+ *
+ * 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.
+ */
+
+#ifndef _Q6_AUDIO_H_
+#define _Q6_AUDIO_H_
+
+#include <mach/qdsp6v2/apr.h>
+
+int q6audio_get_port_index(u16 port_id);
+
+int q6audio_convert_virtual_to_portid(u16 port_id);
+
+int q6audio_validate_port(u16 port_id);
+
+int q6audio_get_port_id(u16 port_id);
+
+#endif
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index c429f24..4676a02 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -2,6 +2,7 @@
* linux/sound/soc-dai.h -- ALSA SoC Layer
*
* Copyright: 2005-2008 Wolfson Microelectronics. PLC.
+ * Copyright (c) 2012, Code Aurora Forum. 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 as
@@ -122,6 +123,10 @@
unsigned int tx_num, unsigned int *tx_slot,
unsigned int rx_num, unsigned int *rx_slot);
+int snd_soc_dai_get_channel_map(struct snd_soc_dai *dai,
+ unsigned int *tx_num, unsigned int *tx_slot,
+ unsigned int *rx_num, unsigned int *rx_slot);
+
int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
/* Digital Audio Interface mute */
@@ -151,6 +156,9 @@
unsigned int rx_num, unsigned int *rx_slot);
int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
+ int (*get_channel_map)(struct snd_soc_dai *dai,
+ unsigned int *tx_num, unsigned int *tx_slot,
+ unsigned int *rx_num, unsigned int *rx_slot);
/*
* DAI digital mute - optional.
* Called by soc-core to minimise any pops.
@@ -173,6 +181,8 @@
struct snd_soc_dai *);
int (*trigger)(struct snd_pcm_substream *, int,
struct snd_soc_dai *);
+ int (*bespoke_trigger)(struct snd_pcm_substream *, int,
+ struct snd_soc_dai *);
/*
* For hardware based FIFO caused delay reporting.
* Optional.
@@ -257,6 +267,13 @@
struct list_head list;
struct list_head card_list;
+
+ /* runtime AIF widget and channel mmap updates */
+ u64 playback_channel_map;
+ u64 capture_channel_map;
+ struct snd_soc_dapm_widget *playback_aif;
+ struct snd_soc_dapm_widget *capture_aif;
+ bool channel_map_instanciated;
};
static inline void *snd_soc_dai_get_dma_data(const struct snd_soc_dai *dai,
@@ -287,4 +304,98 @@
return dev_get_drvdata(dai->dev);
}
+/* Backend DAI PCM ops */
+static inline int snd_soc_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int ret = 0;
+
+ mutex_lock(&rtd->pcm_mutex);
+
+ if (dai->driver->ops->startup)
+ ret = dai->driver->ops->startup(substream, dai);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dai->playback_active++;
+ else
+ dai->capture_active++;
+
+ dai->active++;
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+static inline void snd_soc_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ mutex_lock(&rtd->pcm_mutex);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dai->playback_active--;
+ else
+ dai->capture_active--;
+
+ dai->active--;
+
+ if (dai->driver->ops->shutdown)
+ dai->driver->ops->shutdown(substream, dai);
+ mutex_unlock(&rtd->pcm_mutex);
+}
+
+static inline int snd_soc_dai_hw_params(struct snd_pcm_substream * substream,
+ struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int ret = 0;
+
+ mutex_lock(&rtd->pcm_mutex);
+
+ if (dai->driver->ops->hw_params)
+ ret = dai->driver->ops->hw_params(substream, hw_params, dai);
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+static inline int snd_soc_dai_hw_free(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int ret = 0;
+
+ mutex_lock(&rtd->pcm_mutex);
+
+ if (dai->driver->ops->hw_free)
+ ret = dai->driver->ops->hw_free(substream, dai);
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+static inline int snd_soc_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ int ret = 0;
+
+ mutex_lock(&rtd->pcm_mutex);
+
+ if (dai->driver->ops->prepare)
+ ret = dai->driver->ops->prepare(substream, dai);
+
+ mutex_unlock(&rtd->pcm_mutex);
+ return ret;
+}
+
+static inline int snd_soc_dai_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ if (dai->driver->ops->trigger)
+ return dai->driver->ops->trigger(substream, cmd, dai);
+ return 0;
+}
#endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 8da3c24..fed2e0a 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -13,11 +13,10 @@
#ifndef __LINUX_SND_SOC_DAPM_H
#define __LINUX_SND_SOC_DAPM_H
+#include <linux/device.h>
#include <linux/types.h>
#include <sound/control.h>
-struct device;
-
/* widget has no PM register bit */
#define SND_SOC_NOPM -1
@@ -244,10 +243,6 @@
{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .event = wevent, \
.event_flags = wflags}
-#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay) \
-{ .id = snd_soc_dapm_regulator_supply, .name = wname, \
- .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
- .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
/* dapm kcontrol types */
#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
@@ -280,6 +275,12 @@
.get = xget, \
.put = xput, \
.private_value = (unsigned long)&xenum }
+#define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_enum_double, \
+ .get = xget, \
+ .put = xput, \
+ .private_value = (unsigned long)&xenum }
#define SOC_DAPM_VALUE_ENUM(xname, xenum) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_enum_double, \
@@ -324,11 +325,10 @@
struct snd_soc_dapm_pin;
struct snd_soc_dapm_route;
struct snd_soc_dapm_context;
+struct snd_soc_dapm_widget_list;
int dapm_reg_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event);
-int dapm_regulator_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event);
/* dapm controls */
int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
@@ -353,12 +353,11 @@
struct snd_ctl_elem_value *uncontrol);
int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *uncontrol);
+int snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm,
+ const struct snd_soc_dapm_widget *widget);
int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
const struct snd_soc_dapm_widget *widget,
int num);
-int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
- struct snd_soc_dai *dai);
-int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
/* dapm path setup */
int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
@@ -369,15 +368,19 @@
const struct snd_soc_dapm_route *route, int num);
/* dapm events */
-int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
- struct snd_soc_dai *dai, int event);
+void snd_soc_dapm_codec_stream_event(struct snd_soc_codec *codec,
+ const char *stream, int event);
+int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd,
+ const char *stream, int event);
+void snd_soc_dapm_rtd_stream_event(struct snd_soc_pcm_runtime *rtd,
+ int stream, int event);
void snd_soc_dapm_shutdown(struct snd_soc_card *card);
-
/* external DAPM widget events */
int snd_soc_dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kcontrol, int connect);
int snd_soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
- struct snd_kcontrol *kcontrol, int mux, struct soc_enum *e);
+ struct snd_kcontrol *kcontrol, int change,
+ int mux, struct soc_enum *e);
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
@@ -402,6 +405,15 @@
/* Mostly internal - should not normally be used */
void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason);
+struct snd_soc_dapm_widget *snd_soc_get_codec_widget(struct snd_soc_card *card,
+ struct snd_soc_codec *codec, const char *name);
+struct snd_soc_dapm_widget *snd_soc_get_platform_widget(struct snd_soc_card *card,
+ struct snd_soc_platform *platform, const char *name);
+
+/* dapm path query */
+int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
+ struct snd_soc_dapm_widget_list **list);
+
/* dapm widget types */
enum snd_soc_dapm_type {
snd_soc_dapm_input = 0, /* input pin */
@@ -425,11 +437,9 @@
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
snd_soc_dapm_supply, /* power/clock supply */
- snd_soc_dapm_regulator_supply, /* external regulator */
snd_soc_dapm_aif_in, /* audio interface input */
snd_soc_dapm_aif_out, /* audio interface output */
snd_soc_dapm_siggen, /* signal generator */
- snd_soc_dapm_dai, /* link to DAI structure */
};
/*
@@ -450,8 +460,8 @@
/* dapm audio path between two widgets */
struct snd_soc_dapm_path {
- const char *name;
- const char *long_name;
+ char *name;
+ char *long_name;
/* source (input) and sink (output) widgets */
struct snd_soc_dapm_widget *source;
@@ -474,15 +484,14 @@
/* dapm widget */
struct snd_soc_dapm_widget {
enum snd_soc_dapm_type id;
- const char *name; /* widget name */
+ char *name; /* widget name */
const char *sname; /* stream name */
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
+ struct snd_soc_dai *dai;
struct list_head list;
struct snd_soc_dapm_context *dapm;
- void *priv; /* widget specific data */
-
/* dapm control */
short reg; /* negative reg = no direct dapm */
unsigned char shift; /* bits to shift */
@@ -502,6 +511,7 @@
unsigned char new_power:1; /* power from this run */
unsigned char power_checked:1; /* power checked this run */
int subseq; /* sort within widget type */
+ void *private_data; /* for widget specific data */
int (*power_check)(struct snd_soc_dapm_widget *w);
@@ -549,6 +559,7 @@
struct device *dev; /* from parent - for debug */
struct snd_soc_codec *codec; /* parent codec */
struct snd_soc_platform *platform; /* parent platform */
+ struct snd_soc_dai *dai; /* parent DAI */
struct snd_soc_card *card; /* parent card */
/* used during DAPM updates */
@@ -574,4 +585,16 @@
int neighbour_checks;
};
+/* Accessors for snd_soc_dapm_widget->private_data */
+static inline void *snd_soc_dapm_widget_get_pdata(struct snd_soc_dapm_widget *w)
+{
+ return w->private_data;
+}
+
+static inline void snd_soc_dapm_widget_set_pdata(struct snd_soc_dapm_widget *w,
+ void *data)
+{
+ w->private_data = data;
+}
+
#endif
diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
new file mode 100644
index 0000000..1f99cba
--- /dev/null
+++ b/include/sound/soc-dpcm.h
@@ -0,0 +1,109 @@
+/*
+ * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support
+ *
+ * Author: Liam Girdwood <lrg@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_SOC_DPCM_H
+#define __LINUX_SND_SOC_DPCM_H
+
+#include <sound/pcm.h>
+
+/*
+ * Types of runtime_update to perform (e.g. originated from FE PCM ops
+ * or audio route changes triggered by muxes/mixers.
+ */
+#define SND_SOC_DPCM_UPDATE_NO 0
+#define SND_SOC_DPCM_UPDATE_BE 1
+#define SND_SOC_DPCM_UPDATE_FE 2
+
+/*
+ * Dynamic PCM Frontend -> Backend link state.
+ */
+enum snd_soc_dpcm_link_state {
+ SND_SOC_DPCM_LINK_STATE_NEW = 0, /* newly created path */
+ SND_SOC_DPCM_LINK_STATE_FREE, /* path to be dismantled */
+};
+
+/*
+ * Dynamic PCM params link
+ * This links together a FE and BE DAI at runtime and stores the link
+ * state information and the hw_params configuration.
+ */
+struct snd_soc_dpcm_params {
+ /* FE and BE DAIs*/
+ struct snd_soc_pcm_runtime *be;
+ struct snd_soc_pcm_runtime *fe;
+
+ /* link state */
+ enum snd_soc_dpcm_link_state state;
+
+ struct list_head list_be;
+ struct list_head list_fe;
+
+ /* hw params for this link - may be different for each link */
+ struct snd_pcm_hw_params hw_params;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_state;
+#endif
+};
+
+/*
+ * Bespoke Trigger() Helper API
+ */
+
+/* is the PCM operation for this FE ? */
+static inline int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe,
+ int stream)
+{
+ return (fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE);
+}
+
+/* is the PCM operation for this BE ? */
+static inline int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream)
+{
+ if ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) ||
+ ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_BE) &&
+ be->dpcm[stream].runtime_update))
+ return 1;
+ else
+ return 0;
+}
+
+/* trigger platform driver only */
+static inline int
+ snd_soc_dpcm_platform_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_platform *platform)
+{
+ if (platform->driver->ops->trigger)
+ return platform->driver->ops->trigger(substream, cmd);
+ return 0;
+}
+
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+ struct snd_soc_pcm_runtime *be, int stream);
+
+static inline struct snd_pcm_substream *
+ snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream)
+{
+ return be->pcm->streams[stream].substream;
+}
+
+static inline enum snd_soc_dpcm_state
+ snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream)
+{
+ return be->dpcm[stream].state;
+}
+
+static inline void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be,
+ int stream, enum snd_soc_dpcm_state state)
+{
+ be->dpcm[stream].state = state;
+}
+#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 2ebf787..7886e84 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -55,6 +55,16 @@
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
.put = snd_soc_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+#define SOC_SINGLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+ SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+ .tlv.p = (tlv_array), \
+ .info = snd_soc_info_volsw_s8, .get = snd_soc_get_volsw_s8, \
+ .put = snd_soc_put_volsw_s8, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .min = xmin, .max = xmax, \
+ .platform_max = xmax} }
#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
@@ -131,6 +141,14 @@
.get = xhandler_get, .put = xhandler_put, \
.private_value = \
SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert) }
+ #define SOC_SINGLE_MULTI_EXT(xname, xreg, xshift, xmax, xinvert, xcount,\
+ xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+ .info = snd_soc_info_multi_ext, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_multi_mixer_control) \
+ {.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \
+ .count = xcount, .platform_max = xmax, .invert = xinvert} }
#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
xhandler_get, xhandler_put, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -185,20 +203,6 @@
.rreg = xreg_right, .shift = xshift, \
.min = xmin, .max = xmax} }
-#define SND_SOC_BYTES(xname, xbase, xregs) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
- .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
- .put = snd_soc_bytes_put, .private_value = \
- ((unsigned long)&(struct soc_bytes) \
- {.base = xbase, .num_regs = xregs }) }
-
-#define SND_SOC_BYTES_MASK(xname, xbase, xregs, xmask) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
- .info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
- .put = snd_soc_bytes_put, .private_value = \
- ((unsigned long)&(struct soc_bytes) \
- {.base = xbase, .num_regs = xregs, \
- .mask = xmask }) }
/*
* Simplified versions of above macros, declaring a struct and calculating
@@ -217,6 +221,15 @@
#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
+
+/* DAI Link Host Mode Support */
+#define SND_SOC_DAI_LINK_NO_HOST 0x1
+#define SND_SOC_DAI_LINK_OPT_HOST 0x2
+
+#define snd_soc_get_enum_text(soc_enum, idx) \
+ (soc_enum->texts ? soc_enum->texts[idx] : soc_enum->dtexts[idx])
+
+
/*
* Component probe and remove ordering levels for components with runtime
* dependencies.
@@ -263,6 +276,7 @@
struct snd_soc_jack_zone;
struct snd_soc_jack_pin;
struct snd_soc_cache_ops;
+struct snd_soc_dpcm_link;
#include <sound/soc-dapm.h>
#ifdef CONFIG_GPIOLIB
@@ -288,6 +302,35 @@
SND_SOC_PCM_CLASS_BE = 1,
};
+/*
+ * Dynamic PCM DAI link states.
+ */
+enum snd_soc_dpcm_state {
+ SND_SOC_DPCM_STATE_NEW = 0,
+ SND_SOC_DPCM_STATE_OPEN,
+ SND_SOC_DPCM_STATE_HW_PARAMS,
+ SND_SOC_DPCM_STATE_PREPARE,
+ SND_SOC_DPCM_STATE_START,
+ SND_SOC_DPCM_STATE_STOP,
+ SND_SOC_DPCM_STATE_PAUSED,
+ SND_SOC_DPCM_STATE_SUSPEND,
+ SND_SOC_DPCM_STATE_HW_FREE,
+ SND_SOC_DPCM_STATE_CLOSE,
+};
+
+/*
+ * Dynamic PCM trigger ordering. Triggering flexibility is required as some
+ * DSPs require triggering before/after their clients/hosts.
+ *
+ * i.e. some clients may want to manually order this call in their PCM
+ * trigger() whilst others will just use the regular core ordering.
+ */
+enum snd_soc_dpcm_trigger {
+ SND_SOC_DPCM_TRIGGER_PRE = 0,
+ SND_SOC_DPCM_TRIGGER_POST,
+ SND_SOC_DPCM_TRIGGER_BESPOKE,
+};
+
int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
int source, unsigned int freq, int dir);
int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@@ -333,6 +376,11 @@
unsigned int reg, unsigned int val);
int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
+struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
+ const char *dai_link, int stream);
+struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
+ const char *dai_link);
+
/* Utility functions to get clock rates from various things */
int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params);
@@ -347,6 +395,8 @@
int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
struct snd_soc_jack *jack);
void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask);
+void snd_soc_jack_report_no_dapm(struct snd_soc_jack *jack, int status,
+ int mask);
int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
struct snd_soc_jack_pin *pins);
void snd_soc_jack_notifier_register(struct snd_soc_jack *jack,
@@ -380,7 +430,7 @@
*Controls
*/
struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
- void *data, const char *long_name,
+ void *data, char *long_name,
const char *prefix);
int snd_soc_add_codec_controls(struct snd_soc_codec *codec,
const struct snd_kcontrol_new *controls, int num_controls);
@@ -404,6 +454,8 @@
struct snd_ctl_elem_value *ucontrol);
int snd_soc_info_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
+int snd_soc_info_multi_ext(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo);
int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo);
#define snd_soc_info_bool_ext snd_ctl_boolean_mono_info
@@ -427,13 +479,6 @@
struct snd_ctl_elem_value *ucontrol);
int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol);
-int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo);
-int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol);
-
/**
* struct snd_soc_reg_access - Describes whether a given register is
@@ -524,6 +569,7 @@
/* SoC PCM stream information */
struct snd_soc_pcm_stream {
const char *stream_name;
+ const char *aif_name; /* DAPM AIF widget name */
u64 formats; /* SNDRV_PCM_FMTBIT_* */
unsigned int rates; /* SNDRV_PCM_RATE_* */
unsigned int rate_min; /* min rate */
@@ -664,8 +710,6 @@
/* codec stream completion event */
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
- bool ignore_pmdown_time; /* Doesn't benefit from pmdown delay */
-
/* probe ordering - for components with runtime dependencies */
int probe_order;
int remove_order;
@@ -711,6 +755,8 @@
/* platform IO - used for platform DAPM */
unsigned int (*read)(struct snd_soc_platform *, unsigned int);
int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
+
+ int (*bespoke_trigger)(struct snd_pcm_substream *, int);
};
struct snd_soc_platform {
@@ -718,7 +764,6 @@
int id;
struct device *dev;
struct snd_soc_platform_driver *driver;
- struct mutex mutex;
unsigned int suspended:1; /* platform is suspended */
unsigned int probed:1;
@@ -749,11 +794,21 @@
unsigned int dai_fmt; /* format to set on init */
+ enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */
+
/* Keep DAI active over suspend */
unsigned int ignore_suspend:1;
/* Symmetry requirements */
unsigned int symmetric_rates:1;
+ /* No PCM created for this DAI link */
+ unsigned int no_pcm:1;
+ /* This DAI link can change CODEC and platform at runtime*/
+ unsigned int dynamic:1;
+ /* This DAI has a Backend ID */
+ unsigned int be_id;
+ /* This DAI can support no host IO (no pcm data is copied to from host) */
+ unsigned int no_host_mode:2;
/* pmdown_time is ignored at stop */
unsigned int ignore_pmdown_time:1;
@@ -761,6 +816,10 @@
/* codec/machine specific init - e.g. add machine controls */
int (*init)(struct snd_soc_pcm_runtime *rtd);
+ /* hw_params re-writing for BE and FE sync */
+ int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params);
+
/* machine stream operations */
struct snd_soc_ops *ops;
};
@@ -800,6 +859,12 @@
struct list_head list;
struct mutex mutex;
+ struct mutex dpcm_mutex;
+
+ struct mutex dapm_mutex;
+ struct mutex dapm_power_mutex;
+ struct mutex dsp_mutex;
+ spinlock_t dsp_spinlock;
bool instantiated;
@@ -829,6 +894,8 @@
int num_links;
struct snd_soc_pcm_runtime *rtd;
int num_rtd;
+ int num_playback_channels;
+ int num_capture_channels;
/* optional codec specific configuration */
struct snd_soc_codec_conf *codec_conf;
@@ -880,6 +947,17 @@
void *drvdata;
};
+/* DSP runtime data */
+struct snd_soc_dpcm_runtime {
+ struct list_head be_clients;
+ struct list_head fe_clients;
+ int users;
+ struct snd_pcm_runtime *runtime;
+ struct snd_pcm_hw_params hw_params;
+ int runtime_update;
+ enum snd_soc_dpcm_state state;
+};
+
/* SoC machine DAI configuration, glues a codec and cpu DAI together */
struct snd_soc_pcm_runtime {
struct device *dev;
@@ -892,6 +970,9 @@
unsigned int complete:1;
unsigned int dev_registered:1;
+ /* Dynamic PCM BE runtime data */
+ struct snd_soc_dpcm_runtime dpcm[2];
+
long pmdown_time;
/* runtime devices */
@@ -902,6 +983,11 @@
struct snd_soc_dai *cpu_dai;
struct delayed_work delayed_work;
+
+#ifdef CONFIG_DEBUG_FS
+ struct dentry *debugfs_dpcm_root;
+ struct dentry *debugfs_dpcm_state;
+#endif
};
/* mixer control */
@@ -909,13 +995,12 @@
int min, max, platform_max;
unsigned int reg, rreg, shift, rshift, invert;
};
-
-struct soc_bytes {
- int base;
- int num_regs;
- u32 mask;
+struct soc_multi_mixer_control {
+ int min, max, platform_max, count;
+ unsigned int reg, rreg, shift, rshift, invert;
};
+
/* enumerated kcontrol */
struct soc_enum {
unsigned short reg;
@@ -925,6 +1010,7 @@
unsigned int max;
unsigned int mask;
const char * const *texts;
+ char **dtexts;
const unsigned int *values;
void *dapm;
};