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android/kernel/bcm/23d376ef33aa4c500a5ea24a290f029d5f8e2de3/./sound/caph/dsp/shared_ap.h
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/************************************************************************************************/
/* */
/* Copyright 2011 Broadcom Corporation */
/* */
/* Unless you and Broadcom execute a separate written software license agreement governing */
/* use of this software, this software is licensed to you under the terms of the GNU */
/* General Public License version 2 (the GPL), available at */
/* */
/* http://www.broadcom.com/licenses/GPLv2.php */
/* */
/* with the following added to such license: */
/* */
/* As a special exception, the copyright holders of this software give you permission to */
/* link this software with independent modules, and to copy and distribute the resulting */
/* executable under terms of your choice, provided that you also meet, for each linked */
/* independent module, the terms and conditions of the license of that module. */
/* An independent module is a module which is not derived from this software. The special */
/* exception does not apply to any modifications of the software. */
/* */
/* Notwithstanding the above, under no circumstances may you combine this software in any */
/* way with any other Broadcom software provided under a license other than the GPL, */
/* without Broadcom's express prior written consent. */
/* */
/************************************************************************************************/
#ifndef _INC_SHARED_AP_H_
#define _INC_SHARED_AP_H_
#ifdef MSP
#include "mobcom_types.h"
#else
#include "types.h"
#endif
#include "consts.h"
/**
* \defgroup AP_Shared_Memory AP_Shared_Memory
* @{
*/
//******************************************************************************
// The RIP definition of the shared memory simply places each variable in a
// special data section.
//***********************************************************************************************************
// THE DEFINES ARE JUST FOR TAGS ONLY. THERE IS NO PAGING IN TLIII MEMORY ARCH.
// PLEASE DO NOT USE THEM AS PAGING CONCEPT
// DO NOT FORCE DEFINES TO HARD CODED ADDRESSES
// GENERAL
// SEC_CMD_STATUS
// AUDIO
// SEC_GEN_AUDIO
//
// DEBUG
// SEC_DIAGNOSIS
//
//
//***********************************************************************************************************
#ifdef RIP
// General
#define AP_SHARED_SEC_CMD_STATUS __attribute__( ( section( ".DSECT ap_shared_sec_cmd_status") ) )
// Audio
#define AP_SHARED_SEC_GEN_AUDIO __attribute__( ( section( ".DSECT ap_shared_sec_gen_audio") ) )
// Debug
#define AP_SHARED_SEC_DIAGNOS __attribute__( ( section( ".DSECT ap_shared_sec_diagnos") ) )
# undef EXTERN
# ifdef RIP_EXTERN_DECLARE
# define EXTERN
# else
# define EXTERN extern
# endif
#endif // RIP
#ifdef MSP
#define EXTERN
#define AP_SHARED_SEC_CMD_STATUS
#define AP_SHARED_SEC_GEN_AUDIO
#define AP_SHARED_SEC_DIAGNOS
#endif // MSP
#define VP_CMDQ_SIZE 16 // Number of command queue entries
#define VP_STATUSQ_SIZE 8 // Number of status queue entries
#define G711_FRAME_SIZE 40 // packed 80 samples (10ms) frame
#define NUM_OF_8K_SAMP_PER_INT0_INT 8 // Number of 8kHz samples per INT0 interrupt (for 16kHz * 2)
// Should be same as in hwregs.inc
#define NUM_OF_48K_SAMP_PER_INT0_INT (3*2*8) // Number of 48kHz samples per INT0 interrupt (for 16kHz * 2)
// Should be same as in hwregs.inc
#define NUM_OF_48K_SAMP_PER_LEG_INT0_INT (3) // Number of 48kHz samples per INT0 interrupt (for 16kHz * 2)
// Should be same as in hwregs.inc
/**
* \defgroup External_Modem_Interface External_Modem_Interface
* @{
*/
/**
* Define for number of samples to be transferred per AADMAC interrupt for
* external modem
*/
#define NUM_OF_EXT_MODEM_INTERF_SAMP_PER_INT 160
/**
* @}
*/
#define DJB_BUFFER_SIZE 600 // 600 words ~ 300ms
typedef struct
{
UInt32 array2[35];
} dummy2_config_t;
typedef struct
{
UInt32 array3[10];
} dummy3_queries_t;
typedef struct
{
UInt32 array4[2];
} dummy4_input_t;
typedef struct
{
UInt16 cmd;
UInt16 arg0;
UInt16 arg1;
UInt16 arg2;
} VPCmdQ_t;
typedef struct
{
UInt16 status;
UInt16 arg0;
UInt16 arg1;
UInt16 arg2;
UInt16 arg3;
} VPStatQ_t;
// VOIP data struct and defines
typedef enum
{
GOOD_FRAME,
BAD_FRAME
} G711_FrameType_t;
typedef struct // G.711 speech data for a 10ms frame
{
UInt16 frame_type; // 0 - good frame, 1 - bad frame
UInt16 payload[G711_FRAME_SIZE];
} VR_Frame_G711_t;
typedef struct // Data block of voice recording
{
UInt16 voip_vocoder;
union
{
UInt16 frame_pcm[320]; //voip mode 1/5: PCM, 8KHz use 160, 16KHz use 320
UInt16 frame_fr[79]; //voip mode 2: [VAD, SP, dtx_enable|BFI, SID, TAF], fr special format (76w)
UInt16 frame_amr[18]; //voip mode 3: frametype, mode, coded_bit(MSB first)
VR_Frame_G711_t frame_g711[2]; //voip mode 4: G.711 encoded samples
VR_Frame_AMR_WB_t frame_amr_wb; //voip mode 6: AMR-WB
} voip_frame;
} VOIP_Buffer_t;
//******************************************************************************
//
// Data structure for voice recording
//
//******************************************************************************
typedef struct // Data block of voice recording
{
UInt16 vp_speech_mode; ///< FR/EFR/AMR/Linear PCM speech mode
UInt16 nb_frame; ///< Number of vectors
union
{
VR_Frame_FR_t vectors_fr[RECORDING_FRAME_PER_BLOCK];
VR_Frame_EFR_t vectors_efr[RECORDING_FRAME_PER_BLOCK_EFR];
VR_Lin_PCM_t vectors_lin_PCM[RECORDING_FRAME_PER_BLOCK_LPCM];
VR_Frame_AMR_t vectors_amr[RECORDING_FRAME_PER_BLOCK_AMR];
}vr_frame;
} VRecording_Buffer_t;
//******************************************************************************
//
// Data structure for voice playback
//
//******************************************************************************
typedef struct // Data block of voice playback
{
UInt16 vp_speech_mode; // FR/EFR/AMR/Linear PCM speech mode
UInt16 nb_frame; // Number of vectors
union
{
VR_Frame_FR_t vectors_fr[PLAYBACK_FRAME_PER_BLOCK];
VR_Frame_EFR_t vectors_efr[PLAYBACK_FRAME_PER_BLOCK_EFR];
VR_Lin_PCM_t vectors_lin_PCM[PLAYBACK_FRAME_PER_BLOCK_LPCM];
VR_Frame_AMR_t vectors_amr[RECORDING_FRAME_PER_BLOCK_AMR];
}vp_frame;
} VPlayBack_Buffer_t;
//******************************************************************************
//
// Data structures for voice recording and playback
//
//******************************************************************************
typedef struct
{
VRecording_Buffer_t vr_buf[2]; // Voice recording (ping-pong) buffer
VPlayBack_Buffer_t vp_buf[2]; // Voice playback (ping-pong) buffer
} shared_voice_buf_t; // buffer for voice recording and playback
typedef struct
{
UInt16 *freeListp;
UInt16 *plStartp;
UInt16 *plEndp;
UInt16 freeListSize;
UInt16 numEntry;
UInt16 entrySizeInWords;
UInt16 numAlloc;
} DJB_PAYLOADQ;
/* Adaptive Jitter Control Module (decoder) Statistics */
typedef struct
{
/* Not in state structure because we want to clear it often with stats reset flag */
UInt16 peakHoldingTime; /* Peak Holding Time since last statistic */
/* Total number of superpackets arriving on the egress path */
UInt16 packetCount; /* # of packets received */
/* addTail and reorder are successful additions to the JB */
/* addTail is typically 1..N times the packetCount if superpackets are enabled */
UInt16 addTailCount; /* # of frames added to tail of JB - normal case */
UInt16 reorderCount; /* # of frames reordered */
/* overrun, duplicate, outOfRange, tooLate, jitterMax excess are packets that have been discarded */
UInt16 overrunCount; /* Decoder overrun count */
UInt16 duplicateCount; /* # of duplicate frames deleted */
UInt16 outOfRangeCount; /* # of frames with timestamps too far from current TS to be handled */
UInt16 tooLateCount; /* Packet arrived too late (it's redundant mate already played) */
/* cantDecode are packets that can't be played out due to algorithm not available */
UInt16 cantDecodeCount; /* Can't decode packet - decoder not in load or pkt hdr bad */
/* The following are internal to the AJC module - they do not represent physical packets */
UInt16 ajcUnderrunCount; /* Adaptive Jitter Control: jitter buffer underruns */
UInt16 ajcDeleteCount; /* Adaptive Jitter Control: # of packet deletes done to reduce holding time */
UInt16 ajcRepeatCount; /* Adaptive Jitter Control: # of packet repeats done to either increase holding time
or due to late frame or lost frames. */
UInt16 ajcResyncCount; /* Number of times ajb resynced (went through buildout state) */
UInt16 ajcPhaseJitterCount; /* Number of times ajb inserted a phase discontinuity
(possibly in silence or during CNG or due to a repeat/delete). */
UInt16 ajcHdrPayloadDiff; /* difference between header and payload buffer pool allocation (should be always 0) */
UInt16 ajcPayloadOverflowE; /* number of overruns in the payload buffer when the header pool is not full (should be always 0) */
UInt16 ajcStatsReserve1;
UInt16 ajcStatsReserve2;
UInt16 ajcStatsReserve3;
UInt16 ajcStatsReserve4;
} DJB_STATS;
//******************************************************************************
// Shared memory enumerations
//******************************************************************************
typedef enum
{
RECORD_NONE = 0,
MICROPHONE_ONLY = 1,
FROM_CHANNEL_ONLY = 2,
MICROPHONE_AND_CHANNEL = 3
} VP_Record_Mode_t;
typedef enum
{
PLAYBACK_NONE = 0,
SPEAKER_EAR_PIECE_ONLY = 1,
OVER_THE_CHANNEL_ONLY = 2,
SPEAKER_EAR_PIECE_AND_CHANNEL = 3,
AMR_TONE_DECODE = 4,
AMR_TONE_DECODE_2_MICROPHONE_ONLY = 5
} VP_Playback_Mode_t;
typedef enum
{
MIX_NONE = 0, // mix none
MIX_SPEAKER_EAR_PIECE_ONLY = 1, // mix vpu decodering with dnlink, send to speaker
MIX_OVER_THE_CHANNEL_ONLY = 2, // mix vpu decodering with microphone, send to uplink
MIX_SPEAKER_EAR_PIECE_AND_CHANNEL = 3 // mix vpu decodering with both, send respectively
} VP_PlaybackMix_Mode_t;
typedef enum
{
VP_MODE_IDLE,
VP_MODE_RECOGNITION, // **NOT USED ANYMORE**
VP_MODE_TRAINING, // **NOT USED ANYMORE**
VP_MODE_RECOG_PLAYBACK, // **NOT USED ANYMORE**
VP_MODE_MEMO_RECORDING,
VP_MODE_CALL_RECORDING,
VP_MODE_RECORDING_PLAYBACK,
VP_MODE_RECORD_AND_PLAYBACK_NOT_USED,
VP_MODE_NULL_FRAME_INT
}VPMode_t;
typedef enum
{
PR_MODE_IDLE,
PR_MODE_PLAY
}PRMode_t;
typedef enum
{
PR_OUTMODE_FIFO = 0,
PR_OUTMODE_8KHZ = 1
}PROUTMode_t;
typedef enum
{
PR_OUTDIR_STEREO_PR_HW = 0, //default, backward compatiable
PR_OUTDIR_SPEAKER_EAR_PIECE_ONLY = 1, //only work with PR_OUTMODE_8KHZ
PR_OUTDIR_OVER_THE_CHANNEL_ONLY = 2, //only work with PR_OUTMODE_8KHZ
PR_OUTDIR_SPEAKER_EAR_PIECE_AND_CHANNEL = 3 //only work with PR_OUTMODE_8KHZ
}PROUTDir_t;
typedef enum
{
SRC_0 = 0, //no SRC
SRC_48K8K = 1, //SRC 48KHz to 8KHz
SRC_48K16K = 2 //SRC 48KHz to 16KHz
}SampleRateC_t;
// ---- Typedef Declarations -----------------------------------------
/**
* @addtogroup AP_Queues
* @{
*/
/**
* @addtogroup VP_Queues
* @{
*/
/**
* @addtogroup VP_Command_Queue
* @{
*/
typedef enum
{// Command Code Arguments (** = unused command)
// ======================== ==== ====================================
/** \par Module
* Audio - VPU
* \par Command Code
* 0x00
* \par Description
* Not Used Anymore
*
* @param None
*/
VP_COMMAND_START_TRAINING, // 0x0 () **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x01
* \par Description
* Not Used Anymore
*
* @param None
*/
VP_COMMAND_CANCEL_TRAINING, // 0x1 () **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x02
* \par Description
* Not Used Anymore
*
* @param None
*
*/
VP_COMMAND_START_RECOGNITION, // 0x2 () **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x03
* \par Description
* Not Used Anymore
*
* @param None
*
*/
VP_COMMAND_CANCEL_RECOGNITION, // 0x3 () **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x04
* \par Description
* Not Used Anymore
*
* @param None
*
*/
VP_COMMAND_START_RECOGNITION_PLAYBACK, // 0x4 () **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x05
* \par Description
* Initiates Speech Call Recording (working in both idle and speech call mode).
* Call Recording is a legacy command name. It can be used in both idle mode
* (as Memo Recording) and speech call mode. In idle mode, the only source is
* the microphone. While during the speech call, it is flexible to record from
* a variety of sources:
*
* \htmlonly <ol>
* <li> Recording Near-End speech exclusively, i.e., from the microphone e.g. for voice recognition. </li>
* <li> Recording Far-End speech exclusively, i.e., from the channel and lastly </li>
* <li> Recording both Near and Far-End speech from the microphone and the channel default. </li>
* </ol>
* <br>
* \endhtmlonly
*
* For Call Recording, the recorded format can be either PCM or AMR format. \BR
*
* This is the main Speech recording command from the ARM. This command is to
* be sent at the start of the memo recording. Before getting this command
* ARM should have enabled the Audio Interrupts using COMMAND_AUDIO_ENABLE, and
* the audio inputs and outputs enabled by COMMAND_AUDIO_CONNECT. \BR
*
* After getting this
* command, DSP keeps getting 20ms worth of data from the microphone path and keeps
* encoding the data in the mode required by this command. After every speech buffer
* ((shared_voice_buf.vr_buf[rec_buf_index]).vr_frame)
* has been filled, DSP issues a VP_STATUS_RECORDING_DATA_READY reply to the ARM and
* generates an interrupt. This prompts the ARM to read out the speech data from the
* Shared memory buffer while the DSP continues to write new data into the other ping-pong
* buffer. This continues until the DSP is instructed to stop through receipt of either
* the VP_COMMAND_CANCEL_RECORDING or the VP_COMMAND_STOP_RECORDING command.
*
* @param VP_Record_Mode_t vp_record_mode; Indicates the source of recording
* @param UInt16 nr_frame; Number of 20ms frames to record per speech buffer.
* @param UInt16 {bit8: enables Noise Suppressor for Recording
* bit7: vp_dtx_enable; DTX enable for AMR encoder
* bit6-bit4: VP_Speech_Mode_t vp_speech_mode; Only support LINEAR_PCM and AMR_2.
* bit3: N/A
* bit2-bit0: AMR_Mode vp_amr_mode;}
* \see VP_Record_Mode_t, VP_Speech_Mode_t, AMR_Mode, COMMAND_AUDIO_ENABLE, COMMAND_AUDIO_CONNECT,
* VP_STATUS_RECORDING_DATA_READY, VP_COMMAND_CANCEL_RECORDING, VP_COMMAND_STOP_RECORDING,
* shared_voice_buf
*
*/
VP_COMMAND_START_CALL_RECORDING, // 0x5 ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x06
* \par Description
* Terminates recording modes at the nr_frame boundary of ping-pong buffer, i.e. after it completes
* emptying the current ping-pong buffer.
*
* @param None
*
*/
VP_COMMAND_STOP_RECORDING, // 0x6 ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x07
* \par Description
* Initiates Memo Recording. This is used for recording in idle mode. This is a
* legacy command and can be potentially be removed. The functionality of this
* command can be easily be provided by the VP_COMMAND_START_CALL_RECORDING command.
* This command is to be sent at the start of the memo recording. Before getting this
* command ARM should have enabled the Audio Interrupts using COMMAND_AUDIO_ENABLE,
* and the audio inputs and outputs enabled by COMMAND_AUDIO_CONNECT. After getting this
* command, DSP keeps getting 20ms worth of data from the microphone path and keeps
* collecting the data as required by this command. After every speech buffer
* ((shared_voice_buf.vr_buf[rec_buf_index]).vr_frame)
* has been filled, DSP issues a VP_STATUS_RECORDING_DATA_READY reply to the ARM and
* generates an interrupt. This prompts the ARM to read out the speech data from the
* Shared memory buffer while the DSP continues to write new data into the other ping-pong
* buffer. This continues until the DSP is instructed to stop through receipt of either
* the VP_COMMAND_CANCEL_RECORDING or the VP_COMMAND_STOP_RECORDING command.
* This command only supports recording of data in PCM format
*
* @param VP_Record_Mode_t vp_record_mode; Indicates the source of recording; Only VP_SPEECH_MODE_LINEAR_PCM_8K or
* VP_SPEECH_MODE_LINEAR_PCM_16K supported. Default (VP_SPEECH_MODE_LINEAR_PCM_8K)
* @param UInt16 nr_frame; Number of 20ms frames to record per speech buffer.
* @param {bit8: enables Noise Suppressor for Recording, bit7: DTX enable}
*
* \see VP_COMMAND_START_CALL_RECORDING, COMMAND_AUDIO_ENABLE, COMMAND_AUDIO_CONNECT,
* VP_STATUS_RECORDING_DATA_READY, VP_COMMAND_CANCEL_RECORDING, VP_COMMAND_STOP_RECORDING,
* shared_voice_buf
*
*/
VP_COMMAND_START_MEMO_RECORDING, // 0x7 ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x08
* \par Description
* This command Initiates Playback. The speech data can be in either PCM format or
* AMR format and is read by the DSP from Shared memory. The playback modes of
* operation are activated and de-activated via this same command.
*
* \htmlonly
* <pre>
* +-----------------------------+---------------------------------------------------------------------------------------------------------+
* |vp_playback_mode | vp_playbackmix_mode |
* | +-------------+----------------------------+---------------------------+----------------------------------+
* | | MIX_NONE | MIX_SPEAKER_EAR_PIECE_ONLY | MIX_OVER_THE_CHANNEL_ONLY | MIX_SPEAKER_EAR_PIECE_AND_CHANNEL|
* +-----------------------------+-------------+----------------------------+---------------------------+----------------------------------+
* |PLAYBACK_NONE | UL=UL_SP | UL=UL_SP | UL=UL_SP | UL=UL_SP |
* | | DL=DL_SP | DL=DL_SP | DL=DL_SP | DL=DL_SP |
* +-----------------------------+-------------+----------------------------+---------------------------+----------------------------------+
* |SPEAKER_EAR_PIECE_ONLY | UL=UL_SP | UL=UL_SP | UL=UL_SP | UL=UL_SP |
* | | DL=Playback | DL=DL_SP+Playback | DL=Playback | DL=DL_SP+Playback |
* +-----------------------------+-------------+----------------------------+---------------------------+----------------------------------+
* |OVER_THE_CHANNEL_ONLY | UL=Playback | UL=Playback | UL=UL_SP+Playback | UL=UL_SP+Playback |
* | | DL=DL_SP | DL=DL_SP | DL=DL_SP | DL=DL_SP |
* +-----------------------------+-------------+----------------------------+---------------------------+----------------------------------+
* |SPEAKER_EAR_PIECE_AND_CHANNEL| UL=Playback | UL=Playback | UL=UL_SP+Playback | UL=UL_SP+Playback |
* | | DL=Playback | DL=DL_SP+Playback | DL=Playback | DL=DL_SP+Playback |
* +-----------------------------+-------------+----------------------------+---------------------------+----------------------------------+
* </pre>
* <br>
* \endhtmlonly
*
* Upon activation, this operating mode continues under full control of the ARM.
* After every speech buffer has been read and processed, the DSP issues a
* VP_STATUS_PLAYBACK_DATA_EMPTY command to the ARM and generates an interrupt. This prompts the
* ARM to write new speech playback data into the Shared memory ping-pong buffer
* ((shared_voice_buf.vp_buf[play_buf_index]).vp_frame) while the DSP
* continues to read data from the other buffer. This sequence continues until the DSP is instructed
* to stop through receipt of either the VP_COMMAND_CANCEL_PLAYBACK or the VP_COMMAND_STOP_PLAYBACK command.
*
* \note To prevent the 20ms speech fragments left in the audio output buffer from repeatedly being played
* out to the speaker/ear-piece in IDLE mode, it is necessary for the ARM to instruct the DSP to disable the
* audio path via the vp_disable_audio_path flag sent in the VP_COMMAND_STOP_PLAYBACK command's argument.
*
* @param VP_Playback_Mode_t vp_playback_mode;
* @param VP_PlaybackMix_Mode_t vp_playbackmix_mode;
*
* \see VP_Playback_Mode_t, VP_PlaybackMix_Mode_t, VP_STATUS_PLAYBACK_DATA_EMPTY, VP_COMMAND_CANCEL_PLAYBACK,
* VP_COMMAND_STOP_PLAYBACK, shared_voice_buf
*/
VP_COMMAND_START_RECORDING_PLAYBACK, // 0x8 ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x09
* \par Description
* Terminates playback modes at the nb_frame boundary of ping-pong buffer.
*
* @param Boolean vp_disable_audio_path - if TRUE: DSP disables UL/DL audio connection.
*
* \note To prevent the 20ms speech fragments left in the audio output buffer from repeatedly being played
* out to the speaker/ear-piece in IDLE mode, it is necessary for the ARM to instruct the DSP to disable the
* audio path via the “vp_disable_audio_path” flag.
*/
VP_COMMAND_STOP_PLAYBACK, // 0x9 ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x0a
* \par Description
* Immediate interruption of recording modes.
*
* @param None
*
*/
VP_COMMAND_CANCEL_RECORDING, // 0xA ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x0b
* \par Description
* Immediate interruption of playback modes.
*
* @param None
*
*/
VP_COMMAND_CANCEL_PLAYBACK, // 0xB ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x0c
* \par Description
* This command initiates continuous Recording and Playback. This reuses the separate voice recording
* and playback modes/commands supports continuous call recording and playback. The only exception
* is that the recording and playback speech formats are constrained to be linear PCM. \BR
*
* After getting this
* command, DSP keeps getting 20ms worth of data from the microphone path and keeps
* encoding the data in the mode required by this command. After every speech buffer
* ((shared_voice_buf.vr_buf[rec_buf_index]).vr_frame)
* has been filled, DSP issues a VP_STATUS_RECORDING_DATA_READY reply to the ARM and
* generates an interrupt. This prompts the ARM to read out the speech data from the
* Shared memory buffer while the DSP continues to write new data into the other ping-pong
* buffer. This continues until the DSP is instructed to stop through receipt of either
* the VP_COMMAND_CANCEL_RECORDING or the VP_COMMAND_STOP_RECORDING command. \BR
*
* Upon activation, this operating mode continues under full control of the ARM.
* After every speech buffer has been read and processed, the DSP issues a
* VP_STATUS_PLAYBACK_DATA_EMPTY command to the ARM and generates an interrupt. This prompts the
* ARM to write new speech playback data into the Shared memory ping-pong buffer
* ((shared_voice_buf.vp_buf[play_buf_index]).vp_frame) while the DSP
* continues to read data from the other buffer. This sequence continues until the DSP is instructed
* to stop through receipt of either the VP_COMMAND_CANCEL_PLAYBACK or the VP_COMMAND_STOP_PLAYBACK command.
*
* \note PCM format only.
*
* @param VP_Playback_Mode_t vp_playback_mode;
* @param VP_Record_Mode_t vp_record_mode; Indicates the source of recording
* @param UInt16 {bit4: flag_memo_record_playback (TRUE: in idle mode),
* bit3-bit0: nr_frame Number of 20ms frames to record per speech buffer.}
*
* \see VP_Record_Mode_t, VP_Speech_Mode_t, AMR_Mode, COMMAND_AUDIO_ENABLE, COMMAND_AUDIO_CONNECT,
* VP_STATUS_RECORDING_DATA_READY, VP_COMMAND_CANCEL_RECORDING, VP_COMMAND_STOP_RECORDING,
* VP_Playback_Mode_t, VP_PlaybackMix_Mode_t, VP_STATUS_PLAYBACK_DATA_EMPTY, VP_COMMAND_CANCEL_PLAYBACK,
* VP_COMMAND_STOP_PLAYBACK, shared_voice_buf
*/
VP_COMMAND_START_RECORD_AND_PLAYBACK, // 0xC ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x0d
* \par Description
* This command Starts generation of 20ms frame interrupts if TRUE and stops it if FALSE.
*
* \note This command is used for debugging purpose only.
*
* @param Boolean Enable: If set to TRUE, enabled interrupts to ARM every 20ms with
* VP_STATUS_NULL_FRAME_INT reply
*
* \see VP_STATUS_NULL_FRAME_INT
*/
VP_COMMAND_NULL_FRAME_INT, // 0xD ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x0e
* \par Description
* Used to Pause Call/Memo Recording and/or Playback.
*
* @param UInt16 {bit1: PAUSE_REC, bit0: PAUSE_PLAY}
*
*/
VP_COMMAND_PAUSE, // 0xE () arg0.[b1=1 b0=1]=[PAUSE_REC PAUSE_PLAY], 0:no change
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x0f
* \par Description
* Used to Resume Call/Memo Recording and/or Playback.
*
* @param UInt16 {bit1: RESUME_REC, bit0: RESUME_PLAY}
*
*/
VP_COMMAND_RESUME, // 0xF () arg0.[b1=1 b0=1]=[RESUME_REC RESUME_PLAY], 0:no change
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x10
* \par Description
* This command enables/disables Digital Sound. \BR
*
* This command is used for connecting to mono-bluetooth chip using HCI interface.\BR
*
* The Bluetooth chip would be connected to the base-band using the PCM interface.
*
*
* @param UInt16 ds_enable_flag
* {
* bit0: = 1 Enable Digital Sound
* = 0 Disable Digital Sound
* bit1: = 1 Packed 16-bit AADMAC interface
* = 0 Unpacked 16-bit - one sample per 32-bit long-word in the lower 16-bits
* }
*/
VP_COMMAND_DIGITAL_SOUND, // 0x10 ( enable )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x11
* \par Description
* This command is used for configuring the ARM2SP interface for mixing/playing 8kHz or 16kHz PCM data to either
* Uplink or Downlink. \BR
*
* This command is sent by the ARM to the DSP for starting or stopping of data transfer from the
* ARM to the DSP. \BR
*
* \note Before getting this command ARM should have sent COMMAND_AUDIO_ENABLE, and
* the audio inputs and outputs enabled by COMMAND_AUDIO_CONNECT. \BR
*
* To start ARM2SP: ARM enables ARM2SP by VP_COMMAND_SET_ARM2SP.
* To resume ARM2SP (JunoC0): ARM enables ARM2SP by VP_COMMAND_SET_ARM2SP with arg1=1.
*
* To control shared_Arm2SP_InBuf[]: DSP interrupts ARM with
* VP_STATUS_ARM2SP_EMPTY( shared_Arm2SP_done_flag, shared_Arm2SP_InBuf_out, Arm2SP_flag )
* whenever finishing every half PCM data. shared_Arm2SP_InBuf_out indicates the next index DSP will read.
*
* To stop ARM2SP:\BR
* Quick stop: ARM disables ARM2SP using VP_COMMAND_SET_ARM2SP (arg0=0).\BR
* \note The ARM2SP will be stopped right away. There may be un-finished PCM data in shared_Arm2SP_InBuf[].\BR
*
* Finishing stop: After finishing filling the last 4-speech frames for 8kHz or last 2-speech frames for 16kHz PCM data, ARM set shared_Arm2SP_done_flag=1
* to indicate the end of PCM data. DSP ISR finishes the last 2 or 4-speech frames and disables ARM2SP. DSP sends
* VP_STATUS_ARM2SP_EMPTY and arg0=0. \BR
*
* To pause ARM2SP (JunoC0): ARM disables ARM2SP using VP_COMMAND_SET_ARM2SP (arg0=0, arg1=1).
*
*
* @param UInt16 Arm2SP_flag \BR
* \htmlonly
* <pre>
* {
* bit15 - Sampling Rate of ARM2SP stream 0 = 8kHz, 1 = 16kHz
* bits14:12 - Number of frames of data after which the DSP is supposed to send the VP_STATUS_ARM2SP_EMPTY reply
* (between 1 to 4 for 8kHz data and between 1 and 2 for 16kHz data - default is 4 for 8kHz and
* 2 for 16kHz)
* bit11 - reserved
* bit10 - DL_AFTER_PROC_bit 1= play PCM after DL audio processing (default=0) (JunoB0)
* bit9 - DL_OVERWRITE_bit 1= overwrite Arm2SP buffer to DL (only when DL_MIX_bit=0)
* bit8 - DL_MIX_bit 1= mixing Arm2SP buffer with DL (regardless DL_OVERWRITE_bit setting)
* bit7 - MONO_ST_bit 1= Stereo (not used if not 48k)
* 0= Mono
* bit6 - UL_BEFORE_PROC_bit 1= play PCM before UL audio processing (default=0) (JunoB0)
* bit5 - UL_OVERWRITE_bit 1= overwrite Arm2SP buffer to UL (only when UL_MIX_bit=0)
* bit4 - UL_MIX_bit 1= mixing Arm2SP buffer with UL (regardless UL_OVERWRITE_bit setting)
* bit3 - Record_bit 1= enable recording this PCM into memo/call recording
* bit2 - 48K input 1= Sampling rate of ARM2SP stream = 48k (irrespective of bit15 setting),
* 0= sampling rate determined by bit 15
* bit1 - UL_enable [1/0] = [enable/disable]
* bit0 - DL_enable [1/0] = [enable/disable]
* }
* </pre>
* \endhtmlonly
* @param UInt16 Reset_out_ptr_flag \BR
* =0, reset output pointer - shared_Arm2SP_InBuf_out - of buffer
* shared_Arm2SP_InBuf[] to 0. Used for new arm2sp session.\BR
* =1, keep output pointer - shared_Arm2SP_InBuf_out - unchange.
* Used for PAUSE/RESUME the same arm2sp session.
*
* \see ARM2SP_interface, shared_Arm2SP_InBuf, shared_Arm2SP_InBuf_out, shared_Arm2SP_done_flag,
* VP_STATUS_ARM2SP_EMPTY
*
*/
VP_COMMAND_SET_ARM2SP, // 0x11
VP_COMMAND_SET_BT_NB, // 0x12 ( not used for Rhea )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x13
* \par Description
* This command enables USB headset interface.
* Use USB headset sharedmem buffers as audio input and/or output ping-pong buffer.
*
* @param UInt16 pg_aud_usb_enable \BR
* \htmlonly
* <pre>
* {
* bit0 = [1/0] = [enable/disable] USB headset audio IN buffer.
* bit1 = [1/0] = [enable/disable] USB headset audio OUT buffer.
* bits7-4 = Input Sampling Rate Conversion index (SampleRateC_t).
* bits11-8 = Output Sampling Rate Conversion index (SampleRateC_t).
* }
* </pre>
* \endhtmlonly
*
* \note SRC should be done on arm side in general. Currently in this interface,
* only SRC_48K8K (0) and SRC_48K16K (1) with simple DOWN-SAMPLE are supported in DSP code.
*
* \see USB_Headset_Interface, VP_STATUS_USB_HEADSET_BUFFER, SampleRateC_t
*/
VP_COMMAND_USB_HEADSET, // 0x13 ( set pg_aud_usb_enable = arg0, [b15|b14|b13:b10|b9|b8|b7:b4|b3:b2|b1|b0]=[DL_GAIN|0|DL_SRC|0|UL_GAIN|UL_SRC|xx|DL|UL] )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x14
* \par Description
* This command starts the MultiMedia WB_AMR record OR playback. Prior to this command
* shared_WB_AMR_Ctrl_state must be configured.
*
* \note Prior to sending this command, APE should enable the Audio interrupts to
* come to the DSP at 16kHz using the COMMAND_AUDIO_ENABLE command, and the audio
* input or output should be enabled using the COMMAND_AUDIO_CONNECT command.
*
* @param UInt16 wb_amr_play_rec_mode = 1 for decoder (Play) \BR
* = 2 for encoder (Record) \BR
*
* \see shared_WB_AMR_Ctrl_state, MM_VPU_Interface, COMMAND_AUDIO_ENABLE,
* COMMAND_AUDIO_CONNECT, MM_VPU_RECORD, MM_VPU_PLAYBACK, VP_COMMAND_MM_VPU_DISABLE
*/
VP_COMMAND_MM_VPU_ENABLE, // 0x14
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x15
* \par Description
* This command stops the MultiMedia WB_AMR record OR playback.
*
* @param None
*
* \see shared_WB_AMR_Ctrl_state, MM_VPU_Interface, COMMAND_AUDIO_ENABLE,
* COMMAND_AUDIO_CONNECT, MM_VPU_RECORD, MM_VPU_PLAYBACK
*/
VP_COMMAND_MM_VPU_DISABLE, // 0x15
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x16
* \par Description
* Initializes the vocoder. This is not always needed as when the new vocoder is different than
* the old one an initialization of that vocoder is anyways done in GenDelayedNextEvents in
* int1_smc.c, by looking at the frame entries.
*
*
* @param SpeechMode_t Speech_mode - Speech Codec to initialize
*
* \see SpeechMode_t
*/
VP_COMMAND_VOCODER_INIT, // 0x16 ( efr_flag )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x17
* \par Description
* This command is used for configuring the ARM2SP2 interface for mixing/playing 8kHz or 16kHz PCM data to either
* Uplink or Downlink. \BR
*
* This command is sent by the ARM to the DSP for starting or stopping of data transfer from the
* ARM to the DSP. \BR
*
* \note Before getting this command ARM should have sent COMMAND_AUDIO_ENABLE, and
* the audio inputs and outputs enabled by COMMAND_AUDIO_CONNECT. \BR
*
* To start ARM2SP2: ARM enables ARM2SP2 by VP_COMMAND_SET_ARM2SP2.
* To resume ARM2SP2 (JunoC0): ARM enables ARM2SP2 by VP_COMMAND_SET_ARM2SP2 with arg1=1.
*
* To control shared_Arm2SP2_InBuf[1280]: DSP interrupts ARM with
* VP_STATUS_ARM2SP2_EMPTY( shared_Arm2SP2_done_flag, shared_Arm2SP2_InBuf_out, Arm2SP2_flag )
* whenever finishing every 640 PCM data. shared_Arm2SP2_InBuf_out indicates the next index DSP will read.
*
* To stop ARM2SP2:\BR
* Quick stop: ARM disables ARM2SP2 using VP_COMMAND_SET_ARM2SP2 (arg0=0).\BR
* \note The ARM2SP2 will be stopped right away. There may be un-finished PCM data in shared_Arm2SP2_InBuf[1280].\BR
*
* Finishing stop: After finishing filling the last 2 or 4-speech frame PCM data, ARM set shared_Arm2SP2_done_flag=1
* to indicate the end of PCM data. DSP ISR finishes the last 2 or 4-speech frames and disables ARM2SP2. DSP sends
* VP_STATUS_ARM2SP2_EMPTY and arg0=0. \BR
*
* To pause ARM2SP2 (JunoC0): ARM disables ARM2SP2 using VP_COMMAND_SET_ARM2SP2 (arg0=0, arg1=1).
*
*
* @param UInt16 Arm2SP2_flag \BR
* \htmlonly
* <pre>
* {
* bit15 - Sampling Rate of ARM2SP2 stream 0 = 8kHz, 1 = 16kHz
* bits14:12 - Number of frames of data after which the DSP is supposed to send the VP_STATUS_ARM2SP2_EMPTY reply
* (between 1 to 4 for 8kHz data and between 1 and 2 for 16kHz data - default is 4 for 8kHz and
* 2 for 16kHz)
* bit11 - reserved
* bit10 - DL_AFTER_PROC_bit 1= play PCM after DL audio processing (default=0) (JunoB0)
* bit9 - DL_OVERWRITE_bit 1= overwrite Arm2SP2 buffer to DL (only when DL_MIX_bit=0)
* bit8 - DL_MIX_bit 1= mixing Arm2SP2 buffer with DL (regardless DL_OVERWRITE_bit setting)
* bit7 - MONO_ST_bit 1= Stereo (not used if not 48k)
* 0= Mono
* bit6 - UL_BEFORE_PROC_bit 1= play PCM before UL audio processing (default=0) (JunoB0)
* bit5 - UL_OVERWRITE_bit 1= overwrite Arm2SP2 buffer to UL (only when UL_MIX_bit=0)
* bit4 - UL_MIX_bit 1= mixing Arm2SP2 buffer with UL (regardless UL_OVERWRITE_bit setting)
* bit3 - Record_bit 1= enable recording this PCM into memo/call recording
* bit2 - reserved
* bit1 - UL_enable [1/0] = [enable/disable]
* bit0 - DL_enable [1/0] = [enable/disable]
* }
* </pre>
* \endhtmlonly
* @param UInt16 Reset_out_ptr_flag \BR
* =0, reset output pointer - shared_Arm2SP2_InBuf_out - of buffer
* shared_Arm2SP2_InBuf[] to 0. Used for new arm2sp2 session.\BR
* =1, keep output pointer - shared_Arm2SP2_InBuf_out - unchange.
* Used for PAUSE/RESUME the same arm2sp2 session.
*
* \see ARM2SP_interface, shared_Arm2SP2_InBuf, shared_Arm2SP2_InBuf_out, shared_Arm2SP2_done_flag,
* VP_STATUS_ARM2SP2_EMPTY
*
*/
VP_COMMAND_SET_ARM2SP2, // 0x17
VP_COMMAND_VOIF_CONTROL, // 0x18 (enable/disable)
VP_COMMAND_SP, // 0x19
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1A
* \par Description
* This command runs the main AMR vocoder during GSM idle to support WCDMA/VOIP voice call. This command has to be
* sent by the MCU every-time data is sent to the DSP to do a decode for WCDMA/VOIP call. The input to the speech
* decoder comes in DL_MainAMR_buf.param, and the output of the encoder is stored in UL_MainAMR_buf.param.
*
* \note Before getting this command ARM should have enabled the Audio Interrupts using COMMAND_AUDIO_ENABLE, and
* the audio inputs and outputs enabled by COMMAND_AUDIO_CONNECT.
*
* @param UInt16 For VOIP: {bit15-bit8: VOIP mode, bit1 - VoLTECallEnable, bit0 - Voip_DTX_Enable} \BR
* For Non-VOIP: {bit15-bit8:0, bit4: 1/0=WB_AMR/NB_AFS, bit3-0: AMR UL Mode set}
*
* @param UInt16 AMR_IF2_flag - AMR Iterface Format 2 flag
* @param UInt16 WCDMA_MST_flag - WCDMA MST SD decoding flag \BR
* = 2 - MainAMR_WCDMA_MST_flag = TRUE, MainAMR_WCDMA_MST_init = TRUE; \BR
* = 1 - MainAMR_WCDMA_MST_flag = TRUE, MainAMR_WCDMA_MST_init = FALSE; \BR
* = 0 - MainAMR_WCDMA_MST_flag = FALSE, MainAMR_WCDMA_MST_init = FALSE;
* \par Associated Replies
* For every VP_COMMAND_MAIN_AMR_RUN command, an associated VP_STATUS_MAIN_AMR_DONE reply would be sent in the status
* queue.
* \sa DL_MainAMR_buf, UL_MainAMR_buf, VP_STATUS_MAIN_AMR_DONE
*/
VP_COMMAND_MAIN_AMR_RUN, // 0x1A ( run main AMR vocoder during GSM idle to support WCDMA voice call, arg0=UL AMR codec mode request )
VP_COMMAND_CLEAR_VOIPMODE, // 0x1B (arg0 = 0 to clear VOIPmode)
VP_COMMAND_CNC_ENABLE, // 0x1C (arg0=1 to enable click removal)
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1D
* \par Description
* This command enables the output to the speaker path at 48kHz. This command has to be sent by the APE before
* sending COMMAND_SOFTWARE_ SPEAKER_PROTECTION_ENABLE. COMMAND_NEWAUDFIFO_START and before enabling EANC.
* APE should enable the Audio interrupts to
* come to the DSP at 16kHz using the COMMAND_AUDIO_ENABLE command, and the audio
* input or output should be enabled using the COMMAND_AUDIO_CONNECT command.
*
* @param UInt16 {
* Bit 0: Enable_48kHz_Speaker_Output: 0 = Disable, 1 = Enable
* Bit 1: Interrupt is enabled for this // Not supported currently
* Bit 2: = 0 Mono, = 1 Stereo // Currently only supports mono
* }
*
* \see COMMAND_SOFTWARE_SPEAKER_PROTECTION_ENABLE, NEWAUDFIFO Interface, COMMAND_NEWAUDFIFO_START
*/
VP_COMMAND_48KHZ_SPEAKER_OUTPUT_ENABLE, // 0x1D ( enable )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1E
* \par Description
* This command enables EANC
*
* \note Currently this command is only supporting audio coming in from the AADMAC channel 2 and looping it back to channel 1
* which is being used by IHF speaker.
*
* @param UInt16 {
* Bit 0: Enable_EANC: 0 = Disable, 1 = Enable
* }
*
* \see
*/
VP_COMMAND_ENABLE_EANC, // 0x1E ( enable )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1F
* \par Description
* This command enables VPU command queue processing, called from AP
*
*
* @param none
*
* \see
*/
VP_COMMAND_VPU_ENABLE, // 0x1F ( enable )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x20
* \par Description
* This command initializes Adaptive Jitter Buffer instance for VoLTE.
*/
VP_COMMAND_VOLTE_INIT, // 0x20
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x21
* \par Description
* This command flushes Adaptive Jitter Buffer for VoLTE.
*/
VP_COMMAND_VOLTE_START_STREAM, // 0x21
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x22
* \par Description
* This command delivers VoLTE DL speech frame to the DSP.
*
* \note Before getting this command ARM should have enabled AMR-NB or AMR-WB VoIP encoder with VP_COMMAND_MAIN_AMR_RUN
*
* @param UInt16 RTP timestamp (least 16-bit) \BR
*
* @param UInt16 {bit15-bit8: codec type, bit4: frame quality, bit3-0: frame type } \BR
* @param UInt16 {bit15-bit8: buffer index, bit7-0: frame index } \BR
*/
VP_COMMAND_VOLTE_PUT_FRAME, // 0x22
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x23
* \par Description
* This command enable/disable PTT session.
*
* @param UInt16 {bit0: 1: Enable; 0: disable;}
*/
VP_COMMAND_PTT_ENABLE, // 0x23
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x24
* \par Description
* This command enable/disable a call using an external modem.\BR
*
* This should be sent before the enabling of any audio at the start
* of a call and after the disabling of any audio at the end of a
* call.\BR
*
* For this command, the DSP enables the external modem interface, it
* configures the AADMAC channels for speaker and mic external modem
* paths and then sends a reply VP_STATUS_EXT_MODEM_CALL_DONE to the
* AP with the same argument as the one passed in the command.\BR
*
* The Audio clocks should not be shut-off before this command is
* sent, and the response to it has been received from the DSP by
* the AP.
*
* @param UInt16 arg0 = 1: Enable\BR
* = 0: Disable
*/
VP_COMMAND_EXT_MODEM_CALL, // 0x24
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x25
* \par Description
* This command is used for configuring the ARM2SP_HQ_DL interface
* for mixing/playing 48kHz PCM data to Downlink. \BR
*
* This command is sent by the ARM to the DSP for starting or
* stopping of 48kHz data transfer from the ARM to the DSP. \BR
*
* \note Before getting this command ARM should have sent
* COMMAND_AUDIO_ENABLE, and the audio inputs and outputs enabled
* by COMMAND_AUDIO_CONNECT, along with enabling of the 48kHz
* AADMAC interface in shared_aadmac_aud_enable. \BR
*
* To start ARM2SP_HQ_DL: ARM enables ARM2SP_HQ_DL by
* VP_COMMAND_SET_ARM2SP_HQ_DL.
* To resume ARM2SP_HQ_DL: ARM enables ARM2SP_HQ_DL by
* VP_COMMAND_SET_ARM2SP_HQ_DL with arg1=1.
*
* To control shared_Arm2SP_InBuf[]: DSP interrupts ARM
* with VP_STATUS_ARM2SP_HQ_DL_EMPTY( shared_ARM2SP_HQ_DL_done_flag,
* shared_ARM2SP_HQ_DL_InBuf_out, ARM2SP_HQ_DL_flag )
* whenever finishing every 2/4 20ms data.
* shared_ARM2SP_HQ_DL_InBuf_out indicates the next index DSP will
* read.\BR
*
* To stop ARM2SP_HQ_DL:\BR
* Quick stop: ARM disables ARM2SP_HQ_DL using
* VP_COMMAND_SET_ARM2SP_HQ_DL (arg0=0).\BR
* The ARM2SP_HQ_DL will be stopped right away. There may be
* un-finished PCM data in shared_Arm2SP_InBuf[].\BR
*
* Finishing stop: After finishing filling the last speech frames of
* PCM data, ARM set shared_ARM2SP_HQ_DL_done_flag=1
* to indicate the end of PCM data. DSP ISR finishes the last speech
* frames and disables ARM2SP_HQ_DL. DSP sends
* VP_STATUS_ARM2SP_HQ_DL_EMPTY and arg0=0. \BR
*
* To pause ARM2SP_HQ_DL : ARM disables ARM2SP_HQ_DL using
* VP_COMMAND_SET_ARM2SP_HQ_DL (arg0=0, arg1=1).
*
*
* @param UInt16 ARM2SP_HQ_DL_flag \BR
* \htmlonly
* <pre>
* {
* bits14:12 - Number of frames of data
* after which the DSP is
* supposed to send the
* VP_STATUS_ARM2SP_HQ_DL_EMPTY
* reply
* (between 1 and 4 or 1 and 2)
* bit11:10 - reserved
* bit9 - DL_OVERWRITE_bit
* 1= overwrite ARM2SP_HQ_DL
* buffer to DL
* (only when DL_MIX_bit=0)
* bit8 - DL_MIX_bit
* 1= mixing ARM2SP_HQ_DL
* buffer with DL
* (regardless
* DL_OVERWRITE_bit
* setting)
* bit7 - 48kHz Mono/Stereo
* [1/0] = [Mono/Stereo]
* bit6-1 - reserved
* bit0 - 48kHz DL PCM enable
* [1/0] = [enable/disable]
* }
* </pre>
* \endhtmlonly
* @param UInt16 Reset_out_ptr_flag \BR
* =0, reset output pointer -
* shared_ARM2SP_HQ_DL_InBuf_out
* - of buffer
* shared_Arm2SP_InBuf[]
* to 0. Used for new
* ARM2SP_HQ_DL session.\BR
* =1, keep output pointer -
* shared_ARM2SP_HQ_DL_InBuf_out
* - unchanged.
* Used for PAUSE/RESUME the
* same ARM2SP_HQ_DL session.
*
* \see ARM2SP_HQ_DL_interface, shared_Arm2SP_InBuf,
* shared_ARM2SP_HQ_DL_InBuf_out, shared_ARM2SP_HQ_DL_done_flag,
* VP_STATUS_ARM2SP_HQ_DL_EMPTY
*
*/
VP_COMMAND_SET_ARM2SP_HQ_DL, // 0x25
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x26
* \par Description
* This command is used for configuring the ARM2SP2_HQ_DL interface
* for mixing/playing 48kHz PCM data to Downlink. \BR
*
* This command is sent by the ARM to the DSP for starting or
* stopping of 48kHz data transfer from the ARM to the DSP. \BR
*
* \note Before getting this command ARM should have sent
* COMMAND_AUDIO_ENABLE, and the audio inputs and outputs enabled
* by COMMAND_AUDIO_CONNECT, along with enabling of the 48kHz
* AADMAC interface in shared_aadmac_aud_enable. \BR
*
* To start ARM2SP2_HQ_DL: ARM enables ARM2SP2_HQ_DL by
* VP_COMMAND_SET_ARM2SP2_HQ_DL.
* To resume ARM2SP2_HQ_DL: ARM enables ARM2SP2_HQ_DL by
* VP_COMMAND_SET_ARM2SP2_HQ_DL with arg1=1.
*
* To control shared_Arm2SP2_InBuf[]: DSP interrupts ARM
* with VP_STATUS_ARM2SP2_HQ_DL_EMPTY( shared_ARM2SP2_HQ_DL_done_flag,
* shared_ARM2SP2_HQ_DL_InBuf_out, ARM2SP2_HQ_DL_flag )
* whenever finishing every 2/4 20ms data.
* shared_ARM2SP2_HQ_DL_InBuf_out indicates the next index DSP will
* read.\BR
*
* To stop ARM2SP2_HQ_DL:\BR
* Quick stop: ARM disables ARM2SP2_HQ_DL using
* VP_COMMAND_SET_ARM2SP2_HQ_DL (arg0=0).\BR
* The ARM2SP2_HQ_DL will be stopped right away. There may be
* un-finished PCM data in shared_Arm2SP2_InBuf[].\BR
*
* Finishing stop: After finishing filling the last speech frames of
* PCM data, ARM set shared_ARM2SP2_HQ_DL_done_flag=1
* to indicate the end of PCM data. DSP ISR finishes the last speech
* frames and disables ARM2SP2_HQ_DL. DSP sends
* VP_STATUS_ARM2SP2_HQ_DL_EMPTY and arg0=0. \BR
*
* To pause ARM2SP2_HQ_DL : ARM disables ARM2SP2_HQ_DL using
* VP_COMMAND_SET_ARM2SP2_HQ_DL (arg0=0, arg1=1).
*
*
* @param UInt16 ARM2SP2_HQ_DL_flag \BR
* \htmlonly
* <pre>
* {
* bits14:12 - Number of frames of data
* after which the DSP is
* supposed to send the
* VP_STATUS_ARM2SP2_HQ_DL_EMPTY
* reply
* (between 1 and 4 or 1 and 2)
* bit11:10 - reserved
* bit9 - DL_OVERWRITE_bit
* 1= overwrite ARM2SP2_HQ_DL
* buffer to DL
* (only when DL_MIX_bit=0)
* bit8 - DL_MIX_bit
* 1= mixing ARM2SP2_HQ_DL
* buffer with DL
* (regardless
* DL_OVERWRITE_bit
* setting)
* bit7 - 48kHz Mono/Stereo
* [0/1] = [Mono/Stereo]
* bit6-1 - reserved
* bit0 - 48kHz DL PCM enable
* [1/0] = [enable/disable]
* }
* </pre>
* \endhtmlonly
* @param UInt16 Reset_out_ptr_flag \BR
* =0, reset output pointer -
* shared_ARM2SP2_HQ_DL_InBuf_out
* - of buffer
* shared_Arm2SP2_InBuf[]
* to 0. Used for new
* ARM2SP2_HQ_DL session.\BR
* =1, keep output pointer -
* shared_ARM2SP2_HQ_DL_InBuf_out
* - unchanged.
* Used for PAUSE/RESUME the
* same ARM2SP2_HQ_DL session.
*
* \see ARM2SP2_HQ_DL_interface, shared_Arm2SP2_InBuf,
* shared_ARM2SP2_HQ_DL_InBuf_out, shared_ARM2SP2_HQ_DL_done_flag,
* VP_STATUS_ARM2SP2_HQ_DL_EMPTY
*
*/
VP_COMMAND_SET_ARM2SP2_HQ_DL // 0x26
} VPCommand_t;
/**
* @}
*/
/**
* @addtogroup VP_Status_Queue
* @{
*/
typedef enum
{// Status Code Arguments (** = unused status)
// ======================== ==== ====================================
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x00
* \par Description
* Not Used Anymore
*
* @param None
*
*/
VP_STATUS_TRAINING_DONE, // 0x0 () **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x01
* \par Description
* Not Used Anymore
*
* @param None
*
*/
VP_STATUS_NEW_VECT_READY, // 0x1 () **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x02
* \par Description
* Not Used Anymore
*
* @param None
*
*/
VP_STATUS_NAME_ID, // 0x2 (} **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x03
* \par Description
* Not Used Anymore
*
* @param None
*
*/
VP_STATUS_RECOGNITION_DONE, // 0x3 () **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x04
* \par Description
* Not Used Anymore
*
* @param None
*
*/
VP_STATUS_RECOGNITION_PLAYBACK_DONE , // 0x4 () **NOT USED ANYMORE**
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x05
* \par Description
* This reply is sent by the DSP to the ARM during both voice recording (working
* in both idle and speech call mode), after every speech buffer has been filled
* ((shared_voice_buf.vr_buf[rec_buf_index]).vr_frame). The voice recording can
* be started by either VP_COMMAND_START_CALL_RECORDING or
* VP_COMMAND_START_MEMO_RECORDING.
* This prompts the ARM to read out the speech data from the
* Shared memory buffer while the DSP continues to write new data into the other ping-pong
* buffer. This continues until the DSP is instructed to stop through receipt of either
* the VP_COMMAND_CANCEL_RECORDING or the VP_COMMAND_STOP_RECORDING command.
*
* @param rec_buf_index {0 or 1}: Index of the recording buffer in Shared memory.
* @param stop_recording_flag {0 or 1}: Indicates if this is the last buffer of recording data.
* @param AMR_Mode vp_amr_mode;
*
* \see VP_Record_Mode_t, VP_Speech_Mode_t, AMR_Mode, COMMAND_AUDIO_ENABLE, COMMAND_AUDIO_CONNECT,
* VP_COMMAND_START_CALL_RECORDING, VP_COMMAND_START_MEMO_RECORDING, VP_COMMAND_CANCEL_RECORDING,
* VP_COMMAND_STOP_RECORDING, shared_voice_buf
*
*/
VP_STATUS_RECORDING_DATA_READY, // 0x5 ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x06
* \par Description
* This reply is sent by the DSP to the ARM during both voice playback (working
* in both idle and speech call mode), after every speech buffer has been emptied
* ((shared_voice_buf.vp_buf[play_buf_index]).vp_frame) by the DSP. The voice playback can
* be started by VP_COMMAND_START_RECORDING_PLAYBACK. \BR
*
* This reply prompts the
* ARM to write new speech playback data into the Shared memory ping-pong buffer
* ((shared_voice_buf.vp_buf[play_buf_index]).vp_frame) while the DSP
* continues to read data from the other buffer. This sequence continues until the DSP is instructed
* to stop through receipt of either the VP_COMMAND_CANCEL_PLAYBACK or the VP_COMMAND_STOP_PLAYBACK command.
*
* @param play_buf_index {0 or 1]: Index of the playback buffer in Shared memory.
* @param stop_playback_flag {0 or 1}: Indicates if this is the last buffer of playback data.
*
* \see VP_COMMAND_START_RECORDING_PLAYBACK, VP_COMMAND_CANCEL_PLAYBACK,
* VP_COMMAND_STOP_PLAYBACK, shared_voice_buf
*
*/
VP_STATUS_PLAYBACK_DATA_EMPTY, // 0x6 ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x07
* \par Description
* This status reply is sent every 20ms by the DSP to the ARM after enabling of it by
* VP_COMMAND_NULL_FRAME_INT command
*
* \note This reply is only used for debugging
*
* @param None
*
* \see VP_COMMAND_NULL_FRAME_INT
*
*/
VP_STATUS_NULL_FRAME_INT, // 0x7 ()
/** \HR */
/** \par Module
* Audio - VPU
* \par Command Code
* 0x08
* \par Description
* Not Used Anymore
*
* @param None
*
*/
VP_STATUS_AMR2_ARORT, // 0x8 () AMR_2 aborted by DSP because playing polyringer
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x09
* \par Description
* This reply is part of the ARM2SP interface for mixing/playing 8kHz/16kHz PCM data to either
* Uplink or Downlink. \BR
* This reply is sent by the DSP whenever finishing every 640 PCM data.
* shared_Arm2SP_InBuf_out indicates the next index DSP will read.
*
* @param UInt16 shared_Arm2SP_done_flag - flag indicating the end of transfer of data
* @param UInt16 shared_Arm2SP_InBuf_out - index indicating next index DSP will read from the
* shared_Arm2SP_InBuf[] buffer.
* @param UInt16 Arm2SP_flag - if shared_Arm2SP_done_flag is set then Arm2SP_flag = 0 \BR
* else send back Arm2SP_flag received in the VP_COMMAND_SET_ARM2SP command \BR
* \htmlonly
* <pre>
* {
* bit15 - Sampling Rate of ARM2SP2 stream 0 = 8kHz, 1 = 16kHz
* bits14:12 - Number of frames of data after which the DSP is supposed to send the VP_STATUS_ARM2SP_EMPTY reply
* (between 1 to 4 for 8kHz data and between 1 and 2 for 16kHz data - default is 4 for 8kHz and
* 2 for 16kHz)
* bit11 - reserved
* bit10 - DL_AFTER_PROC_bit 1= play PCM after DL audio processing (default=0) (JunoB0)
* bit9 - DL_OVERWRITE_bit 1= overwrite Arm2SP buffer to DL (only when DL_MIX_bit=0)
* bit8 - DL_MIX_bit 1= mixing Arm2SP buffer with DL (regardless DL_OVERWRITE_bit setting)
* bit7 - MONO_ST_bit 1= Stereo (not used if not 48k)
* 0= Mono
* bit6 - UL_BEFORE_PROC_bit 1= play PCM before UL audio processing (default=0) (JunoB0)
* bit5 - UL_OVERWRITE_bit 1= overwrite Arm2SP buffer to UL (only when UL_MIX_bit=0)
* bit4 - UL_MIX_bit 1= mixing Arm2SP buffer with UL (regardless UL_OVERWRITE_bit setting)
* bit3 - Record_bit 1= enable recording this PCM into memo/call recording
* bit2 - reserved
* bit1 - UL_enable [1/0] = [enable/disable]
* bit0 - DL_enable [1/0] = [enable/disable]
* }
* </pre>
* \endhtmlonly
*
*
* \see ARM2SP_interface, shared_Arm2SP_InBuf, shared_Arm2SP_InBuf_out, shared_Arm2SP_done_flag,
* VP_COMMAND_SET_ARM2SP
*
*/
VP_STATUS_ARM2SP_EMPTY, // 0x09
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x0A
* \par Description
* This reply is part of the ARM2SP2 interface for mixing/playing 8kHz/16kHz PCM data to either
* Uplink or Downlink. \BR
* This reply is sent by the DSP whenever finishing every 640 PCM data.
* shared_Arm2SP2_InBuf_out indicates the next index DSP will read.
*
* @param UInt16 shared_Arm2SP2_done_flag - flag indicating the end of transfer of data
* @param UInt16 shared_Arm2SP2_InBuf_out - index indicating next index DSP will read from the
* shared_Arm2SP2_InBuf[] buffer.
* @param UInt16 Arm2SP2_flag - if shared_Arm2SP2_done_flag is set then Arm2SP2_flag = 0 \BR
* else send back Arm2SP2_flag received in the VP_COMMAND_SET_ARM2SP2 command \BR
* \htmlonly
* <pre>
* {
* bit15 - Sampling Rate of ARM2SP2 stream 0 = 8kHz, 1 = 16kHz
* bits14:12 - Number of frames of data after which the DSP is supposed to send the VP_STATUS_ARM2SP2_EMPTY reply
* (between 1 to 4 for 8kHz data and between 1 and 2 for 16kHz data - default is 4 for 8kHz and
* 2 for 16kHz)
* bit11 - reserved
* bit10 - DL_AFTER_PROC_bit 1= play PCM after DL audio processing (default=0) (JunoB0)
* bit9 - DL_OVERWRITE_bit 1= overwrite Arm2SP2 buffer to DL (only when DL_MIX_bit=0)
* bit8 - DL_MIX_bit 1= mixing Arm2SP2 buffer with DL (regardless DL_OVERWRITE_bit setting)
* bit7 - MONO_ST_bit 1= Stereo (not used if not 48k)
* 0= Mono
* bit6 - UL_BEFORE_PROC_bit 1= play PCM before UL audio processing (default=0) (JunoB0)
* bit5 - UL_OVERWRITE_bit 1= overwrite Arm2SP2 buffer to UL (only when UL_MIX_bit=0)
* bit4 - UL_MIX_bit 1= mixing Arm2SP2 buffer with UL (regardless UL_OVERWRITE_bit setting)
* bit3 - Record_bit 1= enable recording this PCM into memo/call recording
* bit2 - reserved
* bit1 - UL_enable [1/0] = [enable/disable]
* bit0 - DL_enable [1/0] = [enable/disable]
* }
* </pre>
* \endhtmlonly
*
*
* \see ARM2SP_interface, shared_Arm2SP2_InBuf, shared_Arm2SP2_InBuf_out, shared_Arm2SP2_done_flag,
* VP_COMMAND_SET_ARM2SP2
*
*/
VP_STATUS_ARM2SP2_EMPTY, // 0x0A
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x0B
* \par Description
* This reply is part of the USB_Headset_Interface, which is used for connecting a USB microphone
* and/or speaker to the baseband chip.
*
* This interface is enabled using VP_COMMAND_USB_HEADSET command.
*
* This reply is sent by the DSP after every 20ms speech frame. It indicates
* to the ARM to go ahead and write the microphone data to the consumed input buffer and
* read the speaker data from filled up output buffer. The pointers to the input buffer and
* the output buffers are contained in this reply.
*
* @param UInt16 pg_aud_usb_enable (This is the same first parameter received in VP_COMMAND_USB_HEADSET)
* @param UInt16 Input buffer pointer already read/consumed by the DSP.
* @param UInt16 Output buffer pointer already filled by the DSP.
*
*
* \see USB_Headset_Interface, VP_COMMAND_USB_HEADSET
*
*/
VP_STATUS_USB_HEADSET_BUFFER, // 0x0B ( arg0=pg_aud_usb_enable, arg1=pg_aud_in_ptr_usb_arm, arg2=pg_aud_out_ptr_usb_arm )
VP_STATUS_BT_NB_BUFFER_DONE, // 0x0C ( arg0=isr ptr, arg1=bk ptr(used by arm or dsp bk) )
VP_STATUS_VOIF_BUFFER_READY, // 0x0D
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x0E
* \par Description
* This reply is sent by the DSP during two cases
*
*
* \section WCDMA_encoded_data_interface GSM idle to support WCDMA/VOIP voice call.
* This reply is sent by the DSP
* every-time it completes processing the COMMAND_MAIN_AMR_RUN command - i.e. it has encoded the speech data and
* stored it in UL_MainAMR_buf buffer for ARM to pick up and when it has completed speech decoding the data
* sent by the ARM in DL_MainAMR_buf
*
* @param UInt16 For Non-VOIP: amr_tx_type
* @param UInt16 {bit4: =1/0 - WBAMR/NBAMR, bit3-bit0: active_ulink_mode_set}
* @param Boolean dtx_enable
*
* \note For every COMMAND_MAIN_AMR_RUN command, an associated STATUS_MAIN_AMR_DONE reply would be sent
* in the status queue.
* \see DL_MainAMR_buf, UL_MainAMR_buf, COMMAND_MAIN_AMR_RUN, WCDMA_encoded_data_interface
*
* \section MM_VPU_PLAYBACK
* If the amount of input encoded samples from the ARM in shared_decoder_InputBuffer becomes lower
* than one frame's worth of samples, DSP sends an interrupt to the ARM with a status reply
* VP_STATUS_PRAM_CODEC_INPUT_EMPTY, requesting the ARM to send more downlink data immediately,
* and informing that under-flow is happening.
*
* \see MM_VPU_PLAYBACK
*
* @param UInt16 WB_AMR_InBuf_data_available (Amount of input data available)
* @param UInt16 WB_AMR_InBuf_in (Read Index)
* @param UInt16 WB_AMR_InBuf_out (Write Index)
*
*/
VP_STATUS_PRAM_CODEC_INPUT_EMPTY, // 0x0E
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x0F
* \par Description
* This reply is sent by the DSP during MM_VPU_PLAYBACK.
* If the amount of input encoded samples from the ARM in shared_decoder_InputBuffer,
* becomes lower than the threshold shared_Inbuf_LOW_Sts_TH, the DSP sends a interrupt to
* the ARM with a status reply VP_STATUS_PRAM_CODEC_INPUT_LOW, requesting the ARM to send
* more encoded downlink data.
*
* @param UInt16 Threshold
* @param UInt16 Read Index
* @param UInt16 Write Index
*
* \see MM_VPU_PLAYBACK, shared_Inbuf_LOW_Sts_TH
*/
VP_STATUS_PRAM_CODEC_INPUT_LOW, // 0x0F
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x10
* \par Description
* This reply is sent by the DSP during MM_VPU_PLAYBACK.
* If the amount of remaining space in the output PCM buffer
* (as selected by shared_WB_AMR_Ctrl_state variable) becomes
* lower than one frame's worth of samples, DSP sends an interrupt
* to the ARM with a status reply VP_STATUS_PRAM_CODEC_OUTPUT_FULL,
* requesting the ARM to stop more downlink data immediately, and
* informing that over-flow is happening.
*
* @param UInt16 Threshold = 0
* @param UInt16 Read Index
* @param UInt16 Write Index
*
* \see MM_VPU_PLAYBACK, shared_WB_AMR_Ctrl_state
*/
VP_STATUS_PRAM_CODEC_OUTPUT_FULL, // 0x10
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x11
* \par Description
* This reply is sent by the DSP during MM_VPU_PLAYBACK.
* If the amount of remaining space in the output PCM buffer (as selected by
* shared_WB_AMR_Ctrl_state variable) becomes lower than the threshold
* shared_Outbuf_LOW_Sts_TH, the DSP sends a interrupt to the ARM with a
* status reply STATUS_PRAM_CODEC_OUTPUT_LOW, requesting the ARM to slow down
* in sending more downlink data.
*
* @param UInt16 Threshold
* @param UInt16 Read Index
* @param UInt16 Write Index
*
* \see MM_VPU_PLAYBACK, shared_Inbuf_LOW_Sts_TH, shared_WB_AMR_Ctrl_state
*/
VP_STATUS_PRAM_CODEC_OUTPUT_LOW, // 0x11
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x12
* \par Description
* This reply is sent by the DSP during MM_VPU_PLAYBACK.
* When ARM was to complete the decoding process, it would set the
* appropriate Input Data Done Flag (as selected by the
* shared_WB_AMR_Ctrl_state variable). The DSP stops any further decoding
* after it completes decoding the last frame of data and sends back a
* STATUS_PRAM_CODEC_DONEPLAY reply to the ARM.
*
* @param UInt16 Input done flag as selected by shared_WB_AMR_Ctrl_state
* @param UInt16 Number of Wide-band AMR frames (higher word)
* @param UInt16 Number of Wide-band AMR frames (higher word)
*
* \see MM_VPU_PLAYBACK, shared_WB_AMR_Ctrl_state
*/
VP_STATUS_PRAM_CODEC_DONEPLAY, // 0x12
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x13
* \par Description
* This reply is part of the MM VPU Record interface, which records 16kHz microphone samples to
* WB-AMR encoded data.
*
* This recording is enabled using VP_COMMAND_MM_VPU_ENABLE command.
*
* This reply is sent by the DSP after completing encoding of 4 20ms frames of data. It indicates
* to the ARM to go ahead and read this data from
* shared_encoder_OutputBuffer[shared_encodedSamples_buffer_out[0]]
*
* @param UInt16 wb_amr_encoded_total_size - Size of the encoded data size in 8-bit bytes
* @param UInt16 WB_AMR_OutBuf_in - Write Index - copy of shared_encodedSamples_buffer_in[0]
* @param UInt16 WB_AMR_OutBuf_out - Read Index - copy of shared_encodedSamples_buffer_out[0]
*
*
* \see MM_VPU_RECORD, MM_VPU_Interface, VP_COMMAND_MM_VPU_ENABLE, shared_encoder_OutputBuffer,
* shared_encodedSamples_buffer_in, shared_encodedSamples_buffer_out, VP_COMMAND_MM_VPU_DISABLE
*
*/
VP_STATUS_PRAM_CODEC_DONE_RECORD, // 0x13
VP_STATUS_SP, // 0x14
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x15
* \par Description
* This reply is sent by the DSP during GSM idle to support WCDMA/VOIP voice call. This reply is sent by the DSP
* every-time it completes processing the VP_COMMAND_MAIN_AMR_RUN command - i.e. it has encoded the speech data and
* stored it in UL_MainAMR_buf buffer for ARM to pick up and when it has completed speech decoding the data
* sent by the ARM in DL_MainAMR_buf
*
* @param UInt16 For Non-VOIP: amr_tx_type
* @param UInt16 {bit4: =1/0 - WBAMR/NBAMR, bit3-bit0: active_ulink_mode_set}
* @param Boolean dtx_enable
*
* \par Associated Command
* For every VP_COMMAND_MAIN_AMR_RUN command, an associated VP_STATUS_MAIN_AMR_DONE reply would be sent in the status
* queue.
* \sa DL_MainAMR_buf, UL_MainAMR_buf, VP_COMMAND_MAIN_AMR_RUN
*/
VP_STATUS_MAIN_AMR_DONE, // 0x15 ( report main AMR encoder done, TX frame type, AMR mode, dtx_enable )
VP_STATUS_CNC_EMERGENCY_CALL, // 0x16 ( )
VP_STATUS_CNC_DETECTED, // 0x17 ( )
VP_STATUS_VOIP_DL_DONE, // 0x18 ( )
VP_STATUS_AUDIO_STREAM_DATA_READY, // 0x19 ( arg0 = shared_audio_stream_0_crtl, arg1 = shared_audio_stream_1_crtl, arg2 = audio_stream_buf_idx )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1A
* \par Description
* This reply is sent by the DSP after the initializations to be done by COMMAND_AUDIO_ENABLE are complete.
*
* @param UInt16 Value of shared_aadmac_aud_enable used in the initialization.
* @param None
* @param None
* @param None
*
* \par Associated Command
* For every COMMAND_AUDIO_ENABLE command in the CP's command queue, an associated VP_STATUS_AUDIO_ENABLE_DONE reply
* would be sent in the vp_status queue.
* \sa shared_aadmac_aud_enable, UCOMMAND_AUDIO_ENABLE
*/
VP_STATUS_AUDIO_ENABLE_DONE, // 0x1A ( )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1B
* \par Description
* This reply is sent by the DSP to echo VP_COMMAND_PTT_ENABLE.
*
* @param UInt16 PTT_flag
* @param UInt16 curr_int_rate_flag
* @param UInt16 shared_voif_enable_flag (debugging)
* @param None
*
*/
VP_STATUS_PTT_STATUS, // 0x1B ( )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1C
* \par Description
* This reply is sent by the DSP to indicate PTT UL frame is read and ask next frame.
*
* @param UInt16 PTT_UL_buffer_index
* @param UInt16 PTT_flag
* @param UInt16 curr_int_rate_flag
* @param None
*
*/
VP_STATUS_PTT_UL_FRAME_DONE, // 0x1C ( )
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1D
* \par Description
* This reply from the DSP lets the AP know that the DSP has
* enabled the external modem interface, and configured the
* AADMAC channels for UL and DL external modem paths The
* argument is the same as the one passed in the command.
*
* @param UInt16 arg0 = 1: Enable\BR
* = 0: Disable
*
*/
VP_STATUS_EXT_MODEM_CALL_DONE, // 0x1D (0=disable, 1=Enable)
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1E
* \par Description
* This reply is part of the ARM2SP_HQ_DL interface for
* mixing/playing 48kHz to Downlink. \BR
* This reply is sent by the DSP whenever finishing every 2/4 20ms
* PCM frame.
* shared_ARM2SP_HQ_DL_InBuf_out indicates the next index DSP will
* read.
*
* @param UInt16 shared_ARM2SP_HQ_DL_done_flag - flag
* indicating the end of transfer of data
* @param UInt16 shared_ARM2SP_HQ_DL_InBuf_out - index
* indicating next index DSP will read
* from the shared_Arm2SP_InBuf[]
* buffer.
* @param UInt16 ARM2SP_HQ_DL_flag - if
* shared_ARM2SP_HQ_DL_done_flag is set
* then ARM2SP_HQ_DL_flag = 0 \BR
* else send back ARM2SP_HQ_DL_flag
* received in the
* VP_COMMAND_SET_ARM2SP_HQ_DL command \BR
* \htmlonly
* <pre>
* {
* bits14:12 - Number of frames of data
* after which the DSP is
* supposed to send the
* VP_STATUS_ARM2SP_HQ_DL_EMPTY
* reply
* (between 1 and 4 or 1 and 2)
* bit11:10 - reserved
* bit9 - DL_OVERWRITE_bit
* 1= overwrite ARM2SP_HQ_DL
* buffer to DL
* (only when DL_MIX_bit=0)
* bit8 - DL_MIX_bit
* 1= mixing ARM2SP_HQ_DL
* buffer with DL
* (regardless
* DL_OVERWRITE_bit
* setting)
* bit7 - 48kHz Mono/Stereo
* [1/0] = [Mono/Stereo]
* bit6-1 - reserved
* bit0 - 48kHz DL PCM enable
* [1/0] = [enable/disable]
* }
* </pre>
* \endhtmlonly
*
*
* \see ARM2SP_HQ_DL_interface, shared_Arm2SP_InBuf,
* shared_ARM2SP_HQ_DL_InBuf_out, shared_ARM2SP_HQ_DL_done_flag,
* VP_COMMAND_SET_ARM2SP_HQ_DL
*
*/
VP_STATUS_ARM2SP_HQ_DL_EMPTY, // 0x1E
/** \HR */
/** \par Module
* Audio
* \par Command Code
* 0x1F
* \par Description
* This reply is part of the ARM2SP2_HQ_DL interface for
* mixing/playing 48kHz to Downlink. \BR
* This reply is sent by the DSP whenever finishing every 2/4 20ms
* PCM frame.
* shared_ARM2SP2_HQ_DL_InBuf_out indicates the next index DSP will
* read.
*
* @param UInt16 shared_ARM2SP2_HQ_DL_done_flag - flag
* indicating the end of transfer of data
* @param UInt16 shared_ARM2SP2_HQ_DL_InBuf_out - index
* indicating next index DSP will read
* from the shared_Arm2SP2_InBuf[]
* buffer.
* @param UInt16 ARM2SP2_HQ_DL_flag - if
* shared_ARM2SP2_HQ_DL_done_flag is set
* then ARM2SP2_HQ_DL_flag = 0 \BR
* else send back ARM2SP2_HQ_DL_flag
* received in the
* VP_COMMAND_SET_ARM2SP2_HQ_DL command \BR
* \htmlonly
* <pre>
* {
* bits14:12 - Number of frames of data
* after which the DSP is
* supposed to send the
* VP_STATUS_ARM2SP2_HQ_DL_EMPTY
* reply
* (between 1 and 4 or 1 and 2)
* bit11:10 - reserved
* bit9 - DL_OVERWRITE_bit
* 1= overwrite ARM2SP2_HQ_DL
* buffer to DL
* (only when DL_MIX_bit=0)
* bit8 - DL_MIX_bit
* 1= mixing ARM2SP2_HQ_DL
* buffer with DL
* (regardless
* DL_OVERWRITE_bit
* setting)
* bit7 - 48kHz Mono/Stereo
* [1/0] = [Mono/Stereo]
* bit6-1 - reserved
* bit0 - 48kHz DL PCM enable
* [1/0] = [enable/disable]
* }
* </pre>
* \endhtmlonly
*
*
* \see ARM2SP2_HQ_DL_interface, shared_Arm2SP2_InBuf,
* shared_ARM2SP2_HQ_DL_InBuf_out, shared_ARM2SP2_HQ_DL_done_flag,
* VP_COMMAND_SET_ARM2SP2_HQ_DL
*
*/
VP_STATUS_ARM2SP2_HQ_DL_EMPTY // 0x1F
} VPStatus_t;
/**
* @}
*/
/**
* @}
*/
/**
* @}
*/
//******************************************************************************
// Shared Memory Definition
//******************************************************************************
// SINCE TLIII CORE/MSS, SHARED MEMORY PAGING IS REMOVED.
// THREE BIG CATERGORIES ARE DEFINED:
//
//
// 1. MODEM
// SEC_CMD_STATUS
// SEC_MST_SAIC_GEN_MODEM
// SEC_MODEM_MET
// SEC_RESERVED_PART1
// SEC_SMC_AFC_AGC
// SEC_RFIC
// SEC_RESERVED_PART2
//
// 2. AUDIO
// SEC_GEN_AUDIO
// SEC_NS
// SEC_NLP
// SEC_ECHO
// SEC_VPU
// SEC_AUX_AUDIO
//
// 3. DEBUG
// SEC_DIAGNOSIS
// SEC_DSP_GEN_DEBUG
//
//******************************************************************************
#ifdef MSP
typedef struct
{
#endif
EXTERN UInt32 NOT_USE_shared_memory_align AP_SHARED_SEC_CMD_STATUS;
/*-----------------------------------------------------------------------------------------------*/
/**
* @addtogroup AP_Queues
* @{
*/
/**
* @addtogroup VP_Queues
* @{
*/
/**
* @addtogroup VP_Command_Queue
* @{
*/
EXTERN UInt16 vp_shared_cmdq_in AP_SHARED_SEC_CMD_STATUS;
EXTERN UInt16 vp_shared_cmdq_out AP_SHARED_SEC_CMD_STATUS;
/**
* @}
*/
/**
* @addtogroup VP_Status_Queue
* @{
*/
EXTERN UInt16 vp_shared_statusq_in AP_SHARED_SEC_CMD_STATUS;
EXTERN UInt16 vp_shared_statusq_out AP_SHARED_SEC_CMD_STATUS;
/**
* @}
*/
/**
* @addtogroup VP_Command_Queue
* @{
*/
EXTERN VPCmdQ_t vp_shared_cmdq[ VP_CMDQ_SIZE ] AP_SHARED_SEC_CMD_STATUS;
/**
* @}
*/
/**
* @addtogroup VP_Status_Queue
* @{
*/
EXTERN VPStatQ_t vp_shared_statusq[ VP_STATUSQ_SIZE ] AP_SHARED_SEC_CMD_STATUS;
/**
* @}
*/
/**
* @}
*/
/**
* @}
*/
/**
* @addtogroup Shared_Audio_Buffers
* @{ */
/**
* @addtogroup MM_VPU_Interface
*
* The Multi-Media VPU Interface is used to do 16kHz record and playback in WB-AMR
* format. This interface cannot do simulteneous record and playback.
* It is meant to run only in idle mode. There is no uplink and/or downlink mixing
* with a call data.
*
* \note As it stands the MM_VPU_RECORD interface can only be used for Recording and
* NOT for encoding of WB-AMR. \BR
*
* \note The MM_VPU_PLAYBACK interface can be used BOTH for WB-AMR decoding AND
* for Playback
*
* @{ */
/**
* @addtogroup MM_VPU_RECORD
*
* The Voice codec is programmed to generate an Voice ISR
* at 16Khz providing 320 PCM samples to DSP every 20 ms speech frame, using
* the COMMAND_AUDIO_ENABLE command. \BR
*
* Recording of WB-AMR encoded data is started
* by setting the appropriate mode in the shared_WB_AMR_Ctrl_state variable and
* by sending the COMMAND_MM_VPU_ENABLE command. \BR
*
* Raw PCM samples from the microphone are directly copied to the a DSP internal
* memory buffer and then read back by
* the encoder to process for the compression. \BR
*
* Compressed output bits are written back to the shared
* memory buffer shared_encoder_OutputBuffer for Flash or buffer storage. \BR
*
* When the encoder has completed encoding 4 20ms frames, the DSP sends an interrupt to
* the ARM with the reply STATUS_PRAM_CODEC_DONE_RECORD, to go ahead and read the encoded
* data.
*
* \see COMMAND_MM_VPU_ENABLE, COMMAND_AUDIO_ENABLE, COMMAND_AUDIO_CONNECT,
* STATUS_PRAM_CODEC_DONE_RECORD, shared_WB_AMR_Ctrl_state, shared_encoder_OutputBuffer,
* shared_encodedSamples_buffer_in, shared_encodedSamples_buffer_out, COMMAND_MM_VPU_DISABLE
*
* @{
*/
/**
* @}
*/
/**
* @addtogroup MM_VPU_PLAYBACK
*
* \section WB-AMR Encoding
* This interface can be used to decode WB-AMR and playback (if needed). It can be also used to
* just do WB-AMR decoding with the output present
* in any of the three output shared memory buffers as selected by shared_WB_AMR_Ctrl_state. \BR
*
* Decoding of WB-AMR encoded data from an appropriate input shared memory buffer is started
* by setting the appropriate mode in the shared_WB_AMR_Ctrl_state variable and
* by sending the COMMAND_MM_VPU_ENABLE command. Also, prior to sending the
* COMMAND_MM_VPU_ENABLE command, APE should set the shared_Inbuf_LOW_Sts_TH and
* shared_Outbuf_LOW_Sts_TH variables. \BR
*
* \note These thresholds are currently not used in Athena. Currently these thresholds are
* hard-coded in the code to 0x400 \BR
*
* DSP reads the input buffer (shared_decoder_InputBuffer) and runs the decoder to generate
* the synthesized speech to the output buffer, as selected by the shared_WB_AMR_Ctrl_state
* variable. \BR
*
* If the amount of input encoded samples from the ARM in shared_decoder_InputBuffer,
* becomes lower than the threshold shared_Inbuf_LOW_Sts_TH, the DSP sends a interrupt
* to the ARM with a status reply STATUS_PRAM_CODEC_INPUT_LOW, requesting the ARM to send
* more downlink data. \BR
*
* If the amount of input encoded samples from the ARM in shared_decoder_InputBuffer
* becomes lower than one frame's worth of samples, DSP sends an interrupt to the ARM
* with a status reply STATUS_PRAM_CODEC_INPUT_EMPTY, requesting the ARM to send
* more downlink data immediately, and informing that under-flow is happening. \BR
*
* If the amount of remaining space in the output PCM buffer (as selected by
* shared_WB_AMR_Ctrl_state variable) becomes lower than the threshold
* shared_Outbuf_LOW_Sts_TH, the DSP sends a interrupt
* to the ARM with a status reply STATUS_PRAM_CODEC_OUTPUT_LOW, requesting the ARM to slow
* down in sending more downlink data. \BR
*
* If the amount of remaining space in the output PCM buffer (as selected by
* shared_WB_AMR_Ctrl_state variable) becomes lower than one frame's worth of samples,
* DSP sends an interrupt to the ARM
* with a status reply STATUS_PRAM_CODEC_OUTPUT_FULL, requesting the ARM to stop
* more downlink data immediately, and informing that over-flow is happening. \BR
*
* When ARM was to complete the decoding process, it would set the appropriate Input Data Done Flag
* (as selected by the shared_WB_AMR_Ctrl_state variable). The DSP stops any further decoding
* after it completes decoding the last frame of data and sends back a STATUS_PRAM_CODEC_DONEPLAY
* reply to the ARM.
*
* \note Depending on the setting of shared_WB_AMR_Ctrl_state, the output buffers
* of MM_VPU_PLAYBACK overlap with the ones used by the NEWAUDFIFO interface.
*
* \see COMMAND_MM_VPU_ENABLE, shared_WB_AMR_Ctrl_state, COMMAND_MM_VPU_DISABLE,
* shared_decoder_InputBuffer, shared_NEWAUD_InBuf_in, shared_NEWAUD_InBuf_out,
* shared_NEWAUD_InBuf_done_flag, shared_pram_codec_out0, shared_NEWAUD_OutBuf_in,
* shared_NEWAUD_OutBuf_out, shared_newpr_codec_out0, shared_decoder_OutputBuffer,
* shared_encodedSamples_buffer_in, shared_encodedSamples_buffer_out,
* shared_encodedSamples_done_flag, shared_decodedSamples_buffer_in,
* shared_decodedSamples_buffer_out, shared_Inbuf_LOW_Sts_TH,
* shared_Outbuf_LOW_Sts_TH, STATUS_PRAM_CODEC_INPUT_LOW,
* STATUS_PRAM_CODEC_INPUT_EMPTY, STATUS_PRAM_CODEC_OUTPUT_LOW, STATUS_PRAM_CODEC_OUTPUT_FULL,
* STATUS_PRAM_CODEC_DONEPLAY
*
* \section Play-back
* If in addition to decoding of WB-AMR, playback to the speaker is needed,
* in addition to sending the COMMAND_MM_VPU_ENABLE command, AP enable the Voice
* codec to generate a Voice audio at 16Khz using the COMMAND_AUDIO_ENABLE command. \BR
*
* \note Unlike MM_VPU_RECORD, the playback Audio interrupts DO NOT need to come
* at 16kHz frequency (even though the DAC is running at 16kHz).
*
* \note From Athena Onwards, the MM_VPU_PLAYBACK no longer uses the NEWAUDFIFO
* interface for playback.
*
* @{
*/
/**
* This buffer is part of the MM_VPU_PLAYBACK interface. It is filled by ARM with
* WB-AMR encoded speech for DSP to decode.
*
* \see shared_WB_AMR_Ctrl_state, shared_encodedSamples_buffer_in,
* shared_encodedSamples_buffer_out, shared_NEWAUD_InBuf_in,
* shared_NEWAUD_InBuf_out
*/
EXTERN UInt16 shared_decoder_InputBuffer[AUDIO_SIZE_PER_PAGE] AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is part of the MM_VPU_PLAYBACK interface. It is filled by the DSP with
* decoded PCM speech for ARM to take and play. \BR
*
* ARM can select this buffer to be filled by the DSP, using shared_WB_AMR_Ctrl_state.
*
* \see shared_WB_AMR_Ctrl_state, shared_decodedSamples_buffer_in,
* shared_decodedSamples_buffer_out
*
*/
EXTERN UInt16 shared_decoder_OutputBuffer[1] AP_SHARED_SEC_GEN_AUDIO; // This buffer is not used and hence change to only 1 word - should be removed later on
/**
* @}
*/
/**
* \ingroup MM_VPU_RECORD
*/
/**
* This buffer is used to send back MM_VPU recoder's (encoder's) encoded data from
* the DSP to the AP. The DSP writes the encoded data in shared_encoder_OutputBuffer
* with an index from shared_encodedSamples_buffer_in[0], and the ARM reads the encoded
* data from shared_encoder_OutputBuffer with an index of shared_encodedSamples_buffer_out[0]
*
* \see shared_encodedSamples_buffer_in, shared_encodedSamples_buffer_out
*
*/
#ifdef FPGA_AUDIO_HUB_VERIFICATION
EXTERN UInt32 shared_encoder_OutputBuffer[AUDIO_SIZE_PER_PAGE/2] AP_SHARED_SEC_GEN_AUDIO;
#else
EXTERN UInt16 shared_encoder_OutputBuffer[AUDIO_SIZE_PER_PAGE] AP_SHARED_SEC_GEN_AUDIO;
#endif
/** @} */
/**
* @addtogroup USB_Headset_Interface
*
* This interface is used for connecting a USB microphone and/or speaker to the
* baseband chip. Instead of routing the data to the regular microphone and the speaker
* voice paths, the DSP routes the data from and/or to the USB shared memory buffers.
* The ARM fills in the microphone data from the USB stack on the ARM side and passes
* this data to the DSP using the shared memory buffers. The DSP fills in the
* speaker data to the shared memory buffers from which the ARM reads out to the
* USB stack on the ARM side.
*
* To start DSP interface for USB headset:
* - ARM enables the voice interrupts to the DSP using the COMMAND_AUDIO_ENABLE command,
* and enables the input and/or the output to the USB using the COMMAND_AUDIO_CONNECT
* command.
* - ARM enables USB headset shared memory interface by COMMAND_USB_HEADSET.
* - At COMMAND_USB_HEADSET enable command, DSP starts using shared_usb_headset_audio_in_buf_8or16k0[ ]
* and shared_usb_headset_audio_out_buf_8or16k0[ ] as audio in sources and audio out sink.
* - When DSP audio ISR changes 20ms boundary, DSP sends STATUS_USB_HEADSET_BUFFER with the
* in/out buffer pointers for ARM use. ARM then should write audio in buffer and
* read audio out buffer accordingly.
*
* To stop USB headset:
* - ARM disables the voice interrupts using the COMMAND_AUDIO_ENABLE command,
* and disables the input and/or the output to the USB using the COMMAND_AUDIO_CONNECT
* command.
* - ARM disables USB headset interface using COMMAND_USB_HEADSET (arg0=0).
*
* \note
* - USB headset Uplink (Microphone) and Downlink (Speaker) can be enabled/disabled separately.
* - USB headset UL and DL can have different Sampling Rates.
* SRC should be done on arm side in general. Currently in this interface,
* only SRC_48K8K (0) and SRC_48K16K (1) with simple DOWN-SAMPLE are supported in DSP code.
*
* \see COMMAND_USB_HEADSET, COMMAND_AUDIO_ENABLE, COMMAND_AUDIO_CONNECT, STATUS_USB_HEADSET_BUFFER
* @{
*/
/**
* This buffer is part of the USB_Headset_Interface. It contains 20ms worth of input
* 8kHz/16kHz microphone samples to be sent on the uplink path. It is part of the
* 2 input ping pong buffers. The other one is shared_usb_headset_audio_in_buf_8or16k1. \BR
*
* The USB interface is started using the COMMAND_USB_HEADSET command. \BR
*
* This buffer contains the first frame of input samples. Once the DSP is done
* playing the samples from this buffer, it would go ahead and read the samples
* from shared_usb_headset_audio_in_buf_8or16k1. Once it has completed reading
* samples from shared_usb_headset_audio_in_buf_8or16k1, it would go ahead and
* read samples back from this buffer. The change of buffers occurs every
* 20ms speech frames.
*
* \see shared_usb_headset_audio_in_buf_8or16k1, COMMAND_USB_HEADSET
*/
EXTERN UInt16 shared_usb_headset_audio_in_buf_8or16k0[320] AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is part of the USB_Headset_Interface. It contains 20ms worth of input
* 48kHz microphone samples to be sent on the uplink path. It is part of the
* 2 input ping pong buffers. The other one is shared_usb_headset_audio_in_buf_48k1. \BR
*
* The USB interface is started using the COMMAND_USB_HEADSET command. \BR
*
* This buffer contains the first frame of input samples. Once the DSP is done
* playing the samples from this buffer, it would go ahead and read the samples
* from shared_usb_headset_audio_in_buf_48k1. Once it has completed reading
* samples from shared_usb_headset_audio_in_buf_48k1, it would go ahead and
* read samples back from this buffer. The change of buffers occurs every
* 20ms speech frames.
*
* \note In Athena currently this buffer is being used, even though this label
* will not be seen anywhere. It has to be after shared_usb_headset_audio_in_buf_8or16k0 !!!!!
*
* \see shared_usb_headset_audio_in_buf_48k1, COMMAND_USB_HEADSET,
* shared_usb_headset_audio_in_buf_8or16k0
*/
EXTERN UInt16 shared_usb_headset_audio_in_buf_48k0[960] AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is part of the USB_Headset_Interface. It contains 20ms worth of input
* 8kHz/16kHz microphone samples to be sent on the uplink path. It is part of the
* 2 input ping pong buffers. The other one is shared_usb_headset_audio_in_buf_8or16k0. \BR
*
* The USB interface is started using the COMMAND_USB_HEADSET command. \BR
*
* This buffer contains the second frame of input samples. First the DSP
* plays the samples from shared_usb_headset_audio_in_buf_8or160, then
* it would go ahead and read the samples
* from this buffer. Once it has completed reading
* samples from this buffer , it would go ahead and
* read samples back from shared_usb_headset_audio_in_buf_8or160.
* The change of buffers occurs every
* 20ms speech frames.
*
* \see shared_usb_headset_audio_in_buf_8or16k0, COMMAND_USB_HEADSET
*/
EXTERN UInt16 shared_usb_headset_audio_in_buf_8or16k1[320] AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is part of the USB_Headset_Interface. It contains 20ms worth of input
* 48kHz microphone samples to be sent on the uplink path. It is part of the
* 2 input ping pong buffers. The other one is shared_usb_headset_audio_in_buf_48k0. \BR
*
* The USB interface is started using the COMMAND_USB_HEADSET command. \BR
*
* This buffer contains the second frame of input samples. First the DSP
* plays the samples from shared_usb_headset_audio_in_buf_48k0, then
* it would go ahead and read the samples
* from this buffer. Once it has completed reading
* samples from this buffer , it would go ahead and
* read samples back from shared_usb_headset_audio_in_buf_48k0.
* The change of buffers occurs every
* 20ms speech frames.
*
* \note In Athena currently this buffer is being used, even though this label
* will not be seen anywhere. It has to be after shared_usb_headset_audio_in_buf_8or16k1 !!!!!
*
* \see shared_usb_headset_audio_in_buf_48k0, COMMAND_USB_HEADSET,
* shared_usb_headset_audio_in_buf_8or16k1
*/
EXTERN UInt16 shared_usb_headset_audio_in_buf_48k1[960] AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is part of the USB_Headset_Interface. It is used by the DSP to send
* 20ms worth of output 8kHz/16kHz speaker samples to the ARM to be sent to the USB speaker.
* It is part of the 2 input ping pong buffers. The other one is
* shared_usb_headset_audio_out_buf_8or16k1. \BR
*
* The USB interface is started using the COMMAND_USB_HEADSET command. \BR
*
* This buffer contains the first frame of output samples. Once the DSP is done
* decoding the speech samples from the modem, it would go ahead and write the samples
* to this buffer. The second speech frame would then go to
* shared_usb_headset_audio_out_buf_8or16k1 after 20ms. Further 20ms later,
* it will again fill this buffer. The change of buffers occurs every
* 20ms speech frames.
*
* \see shared_usb_headset_audio_out_buf_8or16k1, COMMAND_USB_HEADSET
*/
EXTERN UInt16 shared_usb_headset_audio_out_buf_8or16k0[320] AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is part of the USB_Headset_Interface. It is used by the DSP to send
* 20ms worth of output 48kHz speaker samples to the ARM to be sent to the USB speaker.
* It is part of the 2 input ping pong buffers. The other one is
* shared_usb_headset_audio_out_buf_48k1. \BR
*
* The USB interface is started using the COMMAND_USB_HEADSET command. \BR
*
* This buffer contains the first frame of output samples. Once the DSP is done
* decoding the speech samples from the modem, it would go ahead and write the samples
* to this buffer. The second speech frame would then go to
* shared_usb_headset_audio_out_buf_48k1 after 20ms. Further 20ms later,
* it will again fill this buffer. The change of buffers occurs every
* 20ms speech frames.
*
* \note In Athena currently this buffer is being used, even though this label
* will not be seen anywhere. It has to be after shared_usb_headset_audio_out_buf_8or16k0 !!!!!
*
* \see shared_usb_headset_audio_out_buf_48k1, COMMAND_USB_HEADSET,
* shared_usb_headset_audio_out_buf_8or16k0
*/
EXTERN UInt16 shared_usb_headset_audio_out_buf_48k0[960] AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is part of the USB_Headset_Interface. It is used by the DSP to send
* 20ms worth of output 8kHz/16kHz speaker samples to the ARM to be sent to the USB speaker.
* It is part of the 2 input ping pong buffers. The other one is
* shared_usb_headset_audio_out_buf_8or16k0. \BR
*
* The USB interface is started using the COMMAND_USB_HEADSET command. \BR
*
* This buffer contains the second frame of output samples. Once the DSP is done
* decoding the speech samples from the modem, it would go ahead and write the samples
* to shared_usb_headset_audio_out_buf_8or16k0. The second speech frame would then go to
* this buffer after 20ms. Further 20ms later, it will again fill
* shared_usb_headset_audio_out_buf_8or16k0. The change of buffers occurs every
* 20ms speech frames.
*
* \see shared_usb_headset_audio_out_buf_8or16k0, COMMAND_USB_HEADSET
*/
EXTERN UInt16 shared_usb_headset_audio_out_buf_8or16k1[320] AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is part of the USB_Headset_Interface. It is used by the DSP to send
* 20ms worth of output 48kHz speaker samples to the ARM to be sent to the USB speaker.
* It is part of the 2 input ping pong buffers. The other one is
* shared_usb_headset_audio_out_buf_48k0. \BR
*
* The USB interface is started using the COMMAND_USB_HEADSET command. \BR
*
* This buffer contains the second frame of output samples. Once the DSP is done
* decoding the speech samples from the modem, it would go ahead and write the samples
* to shared_usb_headset_audio_out_buf_48k0. The second speech frame would then go to
* this buffer after 20ms. Further 20ms later, it will again fill
* shared_usb_headset_audio_out_buf_48k0. The change of buffers occurs every
* 20ms speech frames.
*
* \note In Athena currently this buffer is being used, even though this label
* will not be seen anywhere. It has to be after shared_usb_headset_audio_out_buf_8or16k1 !!!!!
*
* \see shared_usb_headset_audio_out_buf_48k0, COMMAND_USB_HEADSET,
* shared_usb_headset_audio_out_buf_8or16k1
*/
EXTERN UInt16 shared_usb_headset_audio_out_buf_48k1[960] AP_SHARED_SEC_GEN_AUDIO;
/**
* @}
*/
EXTERN UInt16 shared_BTnbdinLR[2][640] AP_SHARED_SEC_GEN_AUDIO;
/**
* @addtogroup VPU_Shared_Memory_Interface NOT_USED
*
* This frame-aware interface is provided for transfer of both data and control
* information between the ARM and DSP during NB-AMR or 8k PCM record and
* playback. \BR
*
* This interface supports individual or simultaneous record and playback either
* stand-alone (memo mode) or along with a phone-call, of either uplink or downlink
* or both. \BR
*
* Dedicated ping-pong buffers will
* be used for transferring speech data between the two processors while the
* VP_Command_Queue and VP_Status_Queue queues, also residing in Shared memory,
* will be used for exchanging control information.
*
* \BR
*
* This interface supports simultaneous record and playback. It has two dedicated
* buffers each for record (shared_voice_buf.vr_buf[2]) and for playback
* (shared_voice_buf.vp_buf[2]). These buffers are identical in size and structure.
* The data is exchanged in a ping-pong fashion - i.e. when DSP is reading data from
* one buffer, ARM is writing into another buffer and vice-versa.
*
* \note Two words are placed before each buffer for indicating the format of the
* speech data, i.e., AMR/Linear PCM and the number of 20 ms speech frames contained.
*
* \BR
*
* Speech data in an uncompressed form will be exchanged between ARM and DSP in a
* maximum of 640 word allocation buffer sizes (representing 80 msec worth of speech
* data) or in smaller sizes for compressed speech (size is in multiples of 20ms frames).
* Since double buffering is required this results in a requirement of 1280 words of
* Shared memory per buffer (2560 words total), which will be operated in a cyclic mode.
*
* \note The transfer of each data buffer from the DSP to the ARM will always be
* accompanied by an associated command. However, the converse is not always true
* for data sent from the ARM to the DSP.
*
* \BR
*
* This interface needs to be enabled using COMMAND_VPU_ENABLE command in the main
* command queue.
*
* \note This interface provides for recording/playback in either \Bold{AMR-NB OR
* PCM - Not Both}
*
*
* \see VP_Command_Queue, VP_Status_Queue, shared_voice_buf, COMMAND_VPU_ENABLE
* @{ */
EXTERN shared_voice_buf_t shared_voice_buf AP_SHARED_SEC_GEN_AUDIO; //!< This buffer is used
//!< to transfer data
//!< between the DSP and the
//!< ARM for VPU - recording
//!< and playback
//!< \see VP_Queues
/** @} */
/**
* @addtogroup WCDMA_encoded_data_interface
*
* This interface is provided to transfer data back and forth simultaneously between the ARM and the DSP
* for WCDMA or VoIP call. The data being transfered is in \Bold{AMR-NB\, AMR-WB\, FR or PCM formats only}. \BR
*
* The COMMAND_MAIN_AMR_RUN
* has to be sent by the MCU every-time data is sent to the DSP to do a decode for WCDMA/VOIP call.
* The input to the speech decoder comes in DL_MainAMR_buf.param, and the output of the encoder is
* stored in UL_MainAMR_buf.param \BR
*
* For every COMMAND_MAIN_AMR_RUN command, an associated STATUS_MAIN_AMR_DONE reply would be sent in the
* status queue.
*
* \note This interface can be only used for simultaneous capture and playback. It cannot be used for only
* capture or only playback of AMR/VoIP data.
*
* \BR
*
* \note This interface has to be able to run independently of the record and playback interface, as there
* can be use cases of record and playback during a WCDMA or VoIP call.
*
* \see COMMAND_MAIN_AMR_RUN, DL_MainAMR_buf, UL_MainAMR_buf, AP_DL_MainAMR_buf, AP_UL_MainAMR_buf, STATUS_MAIN_AMR_DONE.
* @{ */
EXTERN VR_Frame_AMR_WB_t AP_DL_MainAMR_buf AP_SHARED_SEC_GEN_AUDIO; //!< This Buffer is used
//!< to transfer the input to
//!< the speech decoder in the
//!< DSP from
//!< WCDMA channel decoder
//!< This buffer is filled by
//!< the ARM (as well as emtying
//!< the UL_MainAMR_buf buffer)
//!< and then it sends
//!< the COMMAND_MAIN_AMR_RUN
//!< command to the DSP to start
//!< decoding the encoded
//!< AMR data
//!< \sa COMMAND_MAIN_AMR_RUN,
//!< STATUS_MAIN_AMR_DONE,
//!< UL_MainAMR_buf
EXTERN VR_Frame_AMR_WB_t AP_UL_MainAMR_buf AP_SHARED_SEC_GEN_AUDIO; //!< This Buffer is used
//!< to transfer the output of
//!< the speech encoder in
//!< the DSP to
//!< WCDMA channel encoder
//!< This buffer is filled by
//!< the DSP (as well as emtying
//!< the DL_MainAMR_buf buffer)
//!< and then it sends
//!< the STATUS_MAIN_AMR_DONE
//!< reply to the ARM to start
//!< channel encoding the
//!< AMR data
//!< \sa COMMAND_MAIN_AMR_RUN,
//!< STATUS_MAIN_AMR_DONE,
//!< DL_MainAMR_buf
/** @} */
/**
* @addtogroup ARM2SP_interface
*
* This interface is used for mixing/playing \Bold{8kHz/16kHz PCM data to either Uplink or
* Downlink or Both}.\BR
*
* There are 2 instances of this interface.\BR
*
* This PCM data can be mixed before or after Uplink or Downlink speech processing.
*
* \note This interface works in only one way - from ARM to DSP. \BR
*
* One use-case of this interface is mixing of call-record warning tones on UL and DL.
* The other is mixing of un-equalized music playback (music sharing) to the Uplink. \BR
*
* In this interface 4-speech frames worth of data is transferred at a time from
* ARM to DSP in ping-pong buffers - i.e. ARM fills empty half of shared_Arm2SP_InBuf[].
* When shared_Arm2SP_InBuf_out=0, ARM fills next half number of words from shared_Arm2SP_InBuf[half].
* When shared_Arm2SP_InBuf_out=half, ARM fills half number of words from shared_Arm2SP_InBuf[0]. \BR
*
* To start ARM2SP: ARM enables ARM2SP by COMMAND_SET_ARM2SP.\BR
*
* To resume ARM2SP (JunoC0): ARM enables ARM2SP by COMMAND_SET_ARM2SP with arg1=1.
*
* To control shared_Arm2SP_InBuf[]: DSP interrupts ARM with
* STATUS_ARM2SP_EMPTY( shared_Arm2SP_done_flag, shared_Arm2SP_InBuf_out, Arm2SP_flag )
* whenever finishing every half PCM data. shared_Arm2SP_InBuf_out indicates the next index DSP will read.
*
* To stop ARM2SP:\BR
* Quick stop: ARM disables ARM2SP using COMMAND_SET_ARM2SP (arg0=0).\BR
* \note The ARM2SP will be stopped right away. There may be un-finished PCM data in shared_Arm2SP_InBuf[].\BR
*
* Finishing stop: After finishing filling the last 2/4-speech frame PCM data, ARM set shared_Arm2SP_done_flag=1
* to indicate the end of PCM data. DSP ISR finishes the last 2/4-speech frames and disables ARM2SP. DSP sends
* STATUS_ARM2SP_EMPTY and arg0=0. \BR
*
* To pause ARM2SP (JunoC0): ARM disables ARM2SP using COMMAND_SET_ARM2SP (arg0=0, arg1=1).
*
* \note When used for mixing warning tone during speech recording, ARM2SP should be enabled periodic
* (every a few seconds) instead of being enabled all the time with tone plus lots 0's PCM.
* For example, ARM can fill 2/4-speech frames (640 PCM samples) of tone plus 0's and set
* shared_Arm2SP_done_flag=1. DSP will mix or overwrite these 2/4-speech frames and disable ARM2SP, then
* send STATUS_ARM2SP_EMPTY( shared_Arm2SP_done_flag,
* shared_Arm2SP_InBuf_out, Arm2SP_flag )
*
* \note The speech processing occurs at the rate specified by AP. If there is a mis-match
* between the rate set by the AP and ARM2SP\Â’s rate, the Sample Rate Conversion occurs right at the start.
*
* \note For the second instance of the ARM2SP, please read ARM2SP2 everywhere ARM2SP is written above.
*
* \see shared_Arm2SP_InBuf, shared_Arm2SP_InBuf_out, shared_Arm2SP_done_flag,
* COMMAND_SET_ARM2SP, STATUS_ARM2SP_EMPTY,
* shared_Arm2SP2_InBuf, shared_Arm2SP2_InBuf_out, shared_Arm2SP2_done_flag,
* COMMAND_SET_ARM2SP2, STATUS_ARM2SP2_EMPTY
*
* @{ */
/**
* This flag indicates the end of the transfer of data from the ARM to the DSP.
* The transfer of data by ARM2SP interface can be disabled by setting
* shared_Arm2SP_done_flag=1 after filling the shared_Arm2SP_InBuf[] buffer.
* DSP will disable the feature when it reaches the next 2/4-speech frame boundary. */
EXTERN UInt16 shared_Arm2SP_done_flag AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is used for mixing PCM data in sharedmem buffer to DL and/or
* UL during speech call. One usage is to send warning tone when doing call
* recording. The buffer is used as ping-pong buffer, each with 2/4-speech frame
* (4*160) or (2*320).
*/
EXTERN UInt32 shared_Arm2SP_InBuf[ARM2SP_INPUT_SIZE_48K/2] AP_SHARED_SEC_GEN_AUDIO;
/**
* This index indicates the next index DSP will read from the shared_Arm2SP_InBuf[]
* buffer.
*
* \note This index is changed only by the DSP.
*
* */
EXTERN UInt16 shared_Arm2SP_InBuf_out AP_SHARED_SEC_GEN_AUDIO;
/** @} */
/**
* @addtogroup Shared_Audio_Buffers
*
* @{ */
/**
* @addtogroup ARM2SP2_interface
* @{ */
/**
* This flag indicates the end of the transfer of data from the ARM to the DSP.
* The transfer of data by ARM2SP2 interface can be disabled by setting
* shared_Arm2SP2_done_flag=1 after filling the shared_Arm2SP2_InBuf[] buffer.
* DSP will disable the feature when it reaches the next 2/4-speech frame boundary. */
EXTERN UInt16 shared_Arm2SP2_done_flag AP_SHARED_SEC_GEN_AUDIO;
/**
* This buffer is used for mixing PCM data in sharedmem buffer to DL and/or
* UL during speech call. One usage is to send warning tone when doing call
* recording. The buffer is used as ping-pong buffer, each with 2/4-speech frame
* (4*160) or (2*320).
*/
EXTERN UInt32 shared_Arm2SP2_InBuf[ARM2SP_INPUT_SIZE_48K/2] AP_SHARED_SEC_GEN_AUDIO;
/**
* This index indicates the next index DSP will read from the shared_Arm2SP2_InBuf[]
* buffer.
*
* \note This index is changed only by the DSP.
*
* */
EXTERN UInt16 shared_Arm2SP2_InBuf_out AP_SHARED_SEC_GEN_AUDIO;
/** @} */
/** @} */
/**
* @addtogroup MM_VPU_Interface
* @{
*/
/**
* \ingroup MM_VPU_RECORD
* \par For MM_VPU Encoder
* The first index is used to send back MM_VPU recoder's (encoder's) encoded data from
* the DSP to the AP. The DSP writes the encoded data in shared_encoder_OutputBuffer
* with an index from shared_encodedSamples_buffer_in[0]. DSP updates the first index.
*
* \see shared_encoder_OutputBuffer
*
*/
EXTERN UInt16 shared_encodedSamples_buffer_in[2] AP_SHARED_SEC_GEN_AUDIO;
/**
* \ingroup MM_VPU_RECORD
* \par For MM_VPU Encoder
* The first index is used to send back MM_VPU recoder's (encoder's) encoded data from
* the DSP to the AP. The ARM reads the encoded
* data from shared_encoder_OutputBuffer with an index of shared_encodedSamples_buffer_out[0].
* ARM updates the first index.
*
* \see shared_encoder_OutputBuffer
*
*/
EXTERN UInt16 shared_encodedSamples_buffer_out[2] AP_SHARED_SEC_GEN_AUDIO;
/**
* This variable should be configured by the AP before sending the COMMAND_MM_VPU_ENABLE command to the DSP.
* It controls the encoder and decoder of the COMMAND_MM_VPU_ENABLE.
*
* The bit map of this variable is as follows: \BR
*
* \htmlonly
* <pre>
* { bits 3-0: Bit-rate mode -
* 0: 6.60 kbit/s
* 1: 8.85 kbit/s
* 2: 12.65 kbit/s
* 3: 14.25 kbit/s
* 4: 15.85 kbit/s
* 5: 18.25 kbit/s
* 6: 19.85 kbit/s
* 7: 23.05 kbit/s
* 8: 23.85 kbit/s
* bit 4 : DTX mode - 0=disabled, 1=enabled
* bits 6-5: Bitstream mode - 0=ITU Format 1, 2=MIME
* bit 10 : Download mode - 0=DMA, 1=Software (Not used currently)
* bit 12-11: Output Select for the decoder (does not affect the encoder.
* Encoder's output is always sent in shared_encoder_OutputBuffer)
* }
*
* MM VPU Decode:
* =============
* +--------------------------------------+--------------------------------+--------------------------------+-----------------------------------+
* | Output Select Value | 00 | 01 | 10 (used only for WB-AMR Decode) |
* +--------------------------------------+--------------------------------+--------------------------------+-----------------------------------+
* | Encoded Speech Inp Buf |shared_decoder_InputBuffer |shared_decoder_InputBuffer |shared_decoder_InputBuffer |
* | ARM's Encoded Speech Write Indx |shared_NEWAUD_InBuf_in[0] |shared_NEWAUD_InBuf_in[1] |shared_encodedSamples_buffer_in[0] |
* | DSP's Encoded Speech Read Indx |shared_NEWAUD_InBuf_out[0] |shared_NEWAUD_InBuf_out[1] |shared_encodedSamples_buffer_out[0]|
* | Input Data Done Flag |shared_NEWAUD_InBuf_done_flag[0]|shared_NEWAUD_InBuf_done_flag[1]|shared_encodedSamples_done_flag[0] |
* | PCM Output Data Buffer |shared_pram_codec_out0 |shared_newpr_codec_out0 |shared_decoder_OutputBuffer |
* | DSP's Decoded Speech (PCM) Write Indx|shared_NEWAUD_OutBuf_in[0] |shared_NEWAUD_OutBuf_in[1] |shared_decodedSamples_buffer_in[0] |
* | Decoded (PCM) Speech Read Indx |shared_NEWAUD_OutBuf_out[0] |shared_NEWAUD_OutBuf_out[1] |shared_decodedSamples_buffer_out[0]|
* +--------------------------------------+--------------------------------+--------------------------------+-----------------------------------+
*
* MM VPU Encode:
* =============
* +--------------------------------------+-----------------------------------+
* | PCM Speech (from MIC) Inp Buf |Internal to the DSP |
* | DSP's PCM Speech Write Indx |N/A |
* | ARM's PCM Speech Read Indx |N/A |
* | Speech Capture Done Flag(Not useful) |shared_decodedSamples_done_flag[0] |
* | Encoded Speech Out Buf |shared_encoder_OutputBuffer |
* | DSP's Encoded Speech Write Indx |shared_encodedSamples_buffer_in[0] |
* | ARM's Encoded Speech Read Indx |shared_encodedSamples_buffer_out[0]|
* +--------------------------------------+-----------------------------------+
*
* </pre>
* \endhtmlonly
* \note Case of Output Select Value = %10 - does not seem to support support playback to the speaker in the code !!! \BR
*
* \note In the case of Output Select Value = %10 - the indexes of the encoded speech are the same as the indexes
* used by the MM_VPU_RECORD for encoded speech, except the encoded speech buffer is different
*
* \see shared_decoder_InputBuffer, shared_NEWAUD_InBuf_in, shared_NEWAUD_InBuf_out, shared_NEWAUD_InBuf_done_flag
* shared_pram_codec_out0, shared_NEWAUD_OutBuf_in, shared_NEWAUD_OutBuf_out, shared_newpr_codec_out0, shared_decoder_OutputBuffer,
* shared_encodedSamples_buffer_in, shared_encodedSamples_buffer_out, shared_encodedSamples_done_flag, shared_decodedSamples_buffer_in,
* shared_decodedSamples_buffer_out, shared_encoder_OutputBuffer,
* MM_VPU_RECORD, MM_VPU_PLAYBACK
*/
EXTERN UInt16 shared_WB_AMR_Ctrl_state AP_SHARED_SEC_GEN_AUDIO; // prgrammable states
/**
* @addtogroup MM_VPU_PLAYBACK
* @{
*/
/**
* These Read indices are ONLY used for MM_VPU_PLAYBACK interface (even though the name
* suggests otherwise). \BR
*
* They are used to the point till which compressed WB-AMR data is written.\BR
*
* They both point inside shared_decoder_InputBuffer.\BR
*
* ARM updates these indices.
*
* \see MM_VPU_PLAYBACK, shared_WB_AMR_Ctrl_state, shared_decoder_InputBuffer
*/
EXTERN UInt16 shared_NEWAUD_InBuf_in[2] AP_SHARED_SEC_GEN_AUDIO; //arm updates after adding the input data
/**
* These Write indices are ONLY used for MM_VPU_PLAYBACK interface (even though the name
* suggests otherwise). \BR
*
* They are used to the point from which compressed WB-AMR data is to be read.\BR
*
* They both point inside shared_decoder_InputBuffer.\BR
*
* DSP updates these indices.
*
* \see MM_VPU_PLAYBACK, shared_WB_AMR_Ctrl_state, shared_decoder_InputBuffer
*/
EXTERN UInt16 shared_NEWAUD_InBuf_out[2] AP_SHARED_SEC_GEN_AUDIO; //dsp updates after using the input data
/**
* @}
*/
/**
* @}
*/
/**
* @addtogroup NEWAUDFIFO_Interface
* @{
*/
/**
* These Done Flags are used for BOTH NEWAUDFIFO_Interface and
* MM_VPU_PLAYBACK interface (depending on the settings in shared_WB_AMR_Ctrl_state).\BR
*
* In both the cases they are used to indicate that ARM has no more data to send to the DSP.
* In case of NEWAUDFIFO_Interface interface, it implies end of un-compressed PCM
* data, whereas in case of MM_VPU_PLAYBACK, it implies end of compressed WB-AMR encoded
* data. \BR
*
* Their associated buffers are shared_pram_codec_out0 and shared_newpr_codec_out0 respectively.\BR
*
* \see NEWAUDFIFO_Interface, MM_VPU_PLAYBACK, shared_WB_AMR_Ctrl_state, shared_pram_codec_out0,
* shared_newpr_codec_out0
*/
EXTERN Int16 shared_NEWAUD_InBuf_done_flag[2] AP_SHARED_SEC_GEN_AUDIO; //arm set it to 1 if InBuf is all loaded (end of file)
/** @} */
/** @addtogroup MM_VPU_Interface
* @{ */
/**
* \ingroup MM_VPU_PLAYBACK
*
* This variable specifies the low threshold for the input buffer of MM_VPU_PLAYBACK.
*
* If the amount of input encoded samples from the ARM in shared_decoder_InputBuffer,
* becomes lower than the threshold shared_Inbuf_LOW_Sts_TH, the DSP sends a interrupt to
* the ARM with a status reply STATUS_PRAM_CODEC_INPUT_LOW, requesting the ARM to send
* more downlink data.
*
* \note This threshold is currently not used in Athena. Currently these thresholds
* are hard-coded in the code to 0x400
*
*/
EXTERN Int16 shared_Inbuf_LOW_Sts_TH AP_SHARED_SEC_GEN_AUDIO;
/**
* \ingroup MM_VPU_PLAYBACK
*
* This variable specifies the low threshold for the output buffer of MM_VPU_PLAYBACK.
*
* If the amount of remaining space in the output PCM buffer (as selected by
* shared_WB_AMR_Ctrl_state variable) becomes lower than the threshold shared_Outbuf_LOW_Sts_TH,
* the DSP sends a interrupt to the ARM with a status reply STATUS_PRAM_CODEC_OUTPUT_LOW,
* requesting the ARM to slow down in sending more downlink data.
*
* \note This threshold is currently not used in Athena. Currently these thresholds are
* hard-coded in the code to 0x400
*
*/
EXTERN Int16 shared_Outbuf_LOW_Sts_TH AP_SHARED_SEC_GEN_AUDIO;
/**
* @}
*/
/**
* @}
*/
EXTERN UInt16 shared_voif_enable_flag AP_SHARED_SEC_GEN_AUDIO;
EXTERN UInt16 shared_voif_DL_buffer_index AP_SHARED_SEC_GEN_AUDIO;
EXTERN Int16 shared_voif_DL_buffer[640] AP_SHARED_SEC_GEN_AUDIO;
EXTERN Int16 shared_voif_UL_buffer[640] AP_SHARED_SEC_GEN_AUDIO;
EXTERN dummy4_input_t shared_SP_input AP_SHARED_SEC_GEN_AUDIO;
EXTERN dummy2_config_t shared_SP_left_config AP_SHARED_SEC_GEN_AUDIO;
EXTERN dummy2_config_t shared_SP_right_config AP_SHARED_SEC_GEN_AUDIO;
EXTERN dummy3_queries_t shared_SP_params_left AP_SHARED_SEC_GEN_AUDIO;
EXTERN dummy3_queries_t shared_SP_params_right AP_SHARED_SEC_GEN_AUDIO;
/**
* This variable tells about the audio sampling rate to the DSP, while using AADMAC during a regular
* phone call.
*/
EXTERN UInt16 shared_aadmac_audio_samp_rate AP_SHARED_SEC_GEN_AUDIO;
/**
* This variable tells what audio paths are enabled.
*/
EXTERN UInt16 shared_aadmac_aud_enable AP_SHARED_SEC_GEN_AUDIO;
/** @addtogroup Shared_Audio_Buffers
* @{ */
/**
* @addtogroup IHF_48K_OUTPUT
*
* This interface is used for sending 48KHz output data to the AADMAC which then passes the data
* out to the IHF device. This interface shoule be able to support both mono and stereo data - however
* currently it supports only mono.
*
* \see COMMAND_48KHZ_SPEAKER_OUTPUT_ENABLE, COMMAND_AUDIO_ENABLE, COMMAND_AUDIO_CONNECT
* @{
*/
/**
* These buffers are used for transfer of mono data between DSP and AADMAC.
*
* \see COMMAND_48KHZ_SPEAKER_OUTPUT_ENABLE
*/
EXTERN UInt32 shared_aud_out_buf_48k[2][IHF_48K_BUF_SIZE] AP_SHARED_SEC_GEN_AUDIO;
/**
* @}
*/
/**
* @addtogroup HANDSET_AADMAC_BUFFERS
*
* This interface is used for receiving 8/16 KHz input data from the AADMAC which then passed over to the regular hand-set.
*
* \see
* @{
*/
/**
* These buffers are used for getting mono data for primary mic from AADMAC.
*
* \see
*
* \note The order of the buffers below is important. First should be the low buffer followed by
* the high buffer.
*/
EXTERN UInt32 shared_aadmac_pri_mic_low[NUM_OF_8K_SAMP_PER_INT0_INT*2] AP_SHARED_SEC_GEN_AUDIO;
EXTERN UInt32 shared_aadmac_pri_mic_high[NUM_OF_8K_SAMP_PER_INT0_INT*2] AP_SHARED_SEC_GEN_AUDIO;
/**
* These buffers are used for getting mono data for primary mic from AADMAC.
*
* \see
*
* \note The order of the buffers below is important. First should be the low buffer followed by
* the high buffer.
*/
EXTERN UInt32 shared_aadmac_sec_mic_low[NUM_OF_8K_SAMP_PER_INT0_INT*2] AP_SHARED_SEC_GEN_AUDIO;
EXTERN UInt32 shared_aadmac_sec_mic_high[NUM_OF_8K_SAMP_PER_INT0_INT*2] AP_SHARED_SEC_GEN_AUDIO;
/**
* These buffers are used for sending mono data for speaker from AADMAC (all paths other than IHF).
*
* \see
*
* \note The order of the buffers below is important. First should be the low buffer followed by
* the high buffer.
*/
EXTERN UInt32 shared_aadmac_spkr_low[NUM_OF_48K_SAMP_PER_INT0_INT] AP_SHARED_SEC_GEN_AUDIO;
EXTERN UInt32 shared_aadmac_spkr_high[NUM_OF_48K_SAMP_PER_INT0_INT] AP_SHARED_SEC_GEN_AUDIO;
/**
* @}
*/
/**
* @}
*/
EXTERN UInt32 shared_UNUSED[95-10] AP_SHARED_SEC_GEN_AUDIO; // 10 - sizeof(DJB_STATS) in UInt32
/**
* @addtogroup Adaptive Jitter Buffer
* @{
*/
/**
* Adaptive Jitter Buffer statistics for reporting to ARM .\BR
*
*/
EXTERN DJB_STATS shared_AjcStatistics AP_SHARED_SEC_GEN_AUDIO;
/**
* @}
*/
EXTERN UInt16 shared_unused_extra AP_SHARED_SEC_GEN_AUDIO;
EXTERN UInt16 shared_unused_voif AP_SHARED_SEC_GEN_AUDIO;
#ifdef FPGA_AUDIO_HUB_VERIFICATION
EXTERN UInt16 shared_ap_test_flag_for_sspi4 AP_SHARED_SEC_GEN_AUDIO;
EXTERN UInt16 shared_dsp_test_flag_for_sspi4 AP_SHARED_SEC_GEN_AUDIO;
EXTERN UInt16 *shared_ap_test_inptr_for_sspi4 AP_SHARED_SEC_GEN_AUDIO;
EXTERN UInt16 *shared_dsp_test_outptr_for_sspi4 AP_SHARED_SEC_GEN_AUDIO;
#endif
EXTERN UInt16 shared_cnc_emergency_flag AP_SHARED_SEC_GEN_AUDIO; // 1: emergency call flag
EXTERN UInt16 shared_cnc_debug AP_SHARED_SEC_GEN_AUDIO; //
EXTERN UInt16 shared_cnc_init_flag AP_SHARED_SEC_GEN_AUDIO; // 1: init cnc then clear this variable
EXTERN VOIP_Buffer_t VOIP_DL_buf AP_SHARED_SEC_GEN_AUDIO;
EXTERN VOIP_Buffer_t VOIP_UL_buf AP_SHARED_SEC_GEN_AUDIO;
/**
* @addtogroup Audio_Gains
* @{
*/
/**
* The gain factor, shared_inp_sp_gain_to_arm2sp_mixer_dl is applied on the downlink speech right before
* either ARM2SP or ARM2SP2 or BTNB signals are "added" to the downlink (does not matter whether the
* ARM2SP, ARM2SP2, BTRNB data is added before or after audio processing).\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 1.14 format).\BR
*
* @note This gain is only applied when ARM2SP, ARM2SP2 or BTNB are mixed in the downlink
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_inp_sp_gain_to_arm2sp_mixer_dl[5] AP_SHARED_SEC_GEN_AUDIO;
/**
* The gain factor, shared_inp_sp_gain_to_arm2sp_mixer_ul is applied on the uplink speech right before
* either ARM2SP or ARM2SP2 or BTNB or TONEUL signals are "added" to the uplink (does not matter whether the
* ARM2SP, ARM2SP2, BTNB TONEUL data is added before or after audio processing).\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 1.14 format).\BR
*
* @note This gain is only applied when ARM2SP, ARM2SP2 or BTNB are mixed in the uplink
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_inp_sp_gain_to_arm2sp_mixer_ul[5] AP_SHARED_SEC_GEN_AUDIO;
/**
* The gain factor, shared_arm2speech2_call_gain_dl is applied on the shared_Arm2SP2_InBuf[1280] PCM data
* on the downlink path (does not matter whether the ARM2SP2 data is added before or after audio processing.\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 2.14 format).\BR
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_arm2speech2_call_gain_dl[5] AP_SHARED_SEC_GEN_AUDIO;
/**
* The gain factor, shared_arm2speech2_call_gain_ul is applied on the shared_Arm2SP2_InBuf[1280] PCM data
* on the uplink path (does not matter whether the ARM2SP2 data is added before or after audio processing.\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 2.14 format).\BR
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_arm2speech2_call_gain_ul[5] AP_SHARED_SEC_GEN_AUDIO;
/**
* The gain factor, shared_arm2speech2_call_gain_rec is applied on the ARM2SP2 PCM data
* getting recorded.\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 2.14 format).\BR
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_arm2speech2_call_gain_rec[5] AP_SHARED_SEC_GEN_AUDIO;
/**
* The gain factor, shared_arm2speech_call_gain_dl is applied on the shared_Arm2SP_InBuf[] PCM data
* on the downlink path (does not matter whether the ARM2SP data is added before or after audio processing.\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 2.14 format).\BR
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_arm2speech_call_gain_dl[5] AP_SHARED_SEC_GEN_AUDIO;
/**
* The gain factor, shared_arm2speech_call_gain_ul is applied on the shared_Arm2SP_InBuf[] PCM data
* on the uplink path (does not matter whether the ARM2SP data is added before or after audio processing.\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 2.14 format).\BR
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_arm2speech_call_gain_ul[5] AP_SHARED_SEC_GEN_AUDIO;
/**
* The gain factor, shared_arm2speech_call_gain_rec is applied on the ARM2SP PCM data
* getting recorded.\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 2.14 format).\BR
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_arm2speech_call_gain_rec[5] AP_SHARED_SEC_GEN_AUDIO;
/**
* This gain, is applied on the VPU playback path on the downlink path before going into record.\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 2.14 format).\BR
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_speech_rec_gain_dl[5] AP_SHARED_SEC_GEN_AUDIO;
/**
* @}
*/
/**
* @addtogroup Adaptive Jitter Buffer
* @{
*/
/**
* Adaptive Jitter Buffer queue and payload.\BR
*
*/
EXTERN DJB_PAYLOADQ ajcPayloadQueue AP_SHARED_SEC_GEN_AUDIO;
EXTERN UInt16 ajcPayloadBuffer[DJB_BUFFER_SIZE] AP_SHARED_SEC_GEN_AUDIO;
/**
* @}
*/
EXTERN UInt16 shared_audio_stream_0_crtl AP_SHARED_SEC_DIAGNOS; // Ctrl info specifying 1 out of N capture points for audio stream_0
EXTERN UInt16 shared_audio_stream_1_crtl AP_SHARED_SEC_DIAGNOS; // Ctrl info specifying 1 out of N capture points for audio stream_1
EXTERN UInt16 shared_audio_stream_2_crtl AP_SHARED_SEC_DIAGNOS; // Ctrl info specifying 1 out of N capture points for audio stream_0
EXTERN UInt16 shared_audio_stream_3_crtl AP_SHARED_SEC_DIAGNOS; // Ctrl info specifying 1 out of N capture points for audio stream_1
EXTERN Audio_Logging_Buf_t shared_audio_stream_0[2] AP_SHARED_SEC_DIAGNOS; // 8KHz 20ms double buffer of Audio
EXTERN Audio_Logging_Buf_t shared_audio_stream_1[2] AP_SHARED_SEC_DIAGNOS; // 8KHz 20ms double buffer of Audio
EXTERN Audio_Logging_Buf_t shared_audio_stream_2[2] AP_SHARED_SEC_DIAGNOS; // 8KHz 20ms double buffer of Audio
EXTERN Audio_Logging_Buf_t shared_audio_stream_3[2] AP_SHARED_SEC_DIAGNOS; // 8KHz 20ms double buffer of Audio
EXTERN UInt16 shared_usb_status_option AP_SHARED_SEC_DIAGNOS; // STATUS_USB_HEADSET_BUFFER ptr option 0 or 1
EXTERN UInt16 shared_rhea_audio_test_select AP_SHARED_SEC_DIAGNOS; // STATUS_USB_HEADSET_BUFFER ptr option 0 or 1
//#ifdef PTT_SUPPORT
EXTERN Int16 shared_PTT_UL_buffer[2][320] AP_SHARED_SEC_DIAGNOS;
//#endif
/** @addtogroup Shared_Audio_Buffers
* @{ */
#if defined(EXT_MODEM_INTERFACE)
/**
* @addtogroup External_Modem_Interface
* @{
*/
/**
* DDR ping-pong buffer for UL data transfer between Rhea and External modem.
* This are the ping-pong buffers from which the AADMAC UL channel would take
* the data from to be transmitted to the External modem.
*/
EXTERN UInt32 shared_ext_ul_buf[2][ NUM_OF_EXT_MODEM_INTERF_SAMP_PER_INT/2] AP_SHARED_SEC_DIAGNOS;
/**
* DDR ping-pong buffer for DL data transfer between Rhea and External modem.
* This are the ping-pong buffers to which the AADMAC DL channel would put
* the data received from the External modem.
*/
EXTERN UInt32 shared_ext_dl_buf[2][ NUM_OF_EXT_MODEM_INTERF_SAMP_PER_INT/2] AP_SHARED_SEC_DIAGNOS;
/**
* @}
*/
#endif
/**
* @addtogroup ARM2SP_HQ_DL_interface
*
* This interface is used for mixing/playing \Bold{48kHz Mono/Stereo PCM data
* to Downlink}.\BR
*
* This PCM data will be mixed after Downlink speech processing in the 48kHz
* mode.
*
* \note This interface works in only one way - from ARM to DSP. \BR
*
* One use-case of this interface is to support S-barge-in feature where a
* music might be playing in the downlink using this interface and an
* uplink record/speech recognition happens at the same time, then this
* UL would get the benefit of echo cancellation of the music.
*
* In this interface 2 or 4-speech frames worth of data is transferred at a
* time from ARM to DSP in ping-pong buffers - i.e. ARM fills empty half of
* shared_Arm2SP_InBuf[].
*
* When shared_ARM2SP_HQ_DL_InBuf_out=0, ARM fills next half number of words from
* shared_Arm2SP_InBuf[half].
* When shared_ARM2SP_HQ_DL_InBuf_out=half, ARM fills half number of words from
* shared_Arm2SP_InBuf[0]. \BR
*
* To start ARM2SP_HQ_DL: ARM enables ARM2SP_HQ_DL by
* VP_COMMAND_SET_ARM2SP_HQ_DL.\BR
*
* To resume ARM2SP_HQ_DL : ARM enables ARM2SP_HQ_DL by
* VP_COMMAND_SET_ARM2SP_HQ_DL with arg1=1.
*
* To control shared_Arm2SP_InBuf[]: DSP interrupts ARM with \BR
* VP_STATUS_ARM2SP_HQ_DL_EMPTY( shared_ARM2SP_HQ_DL_done_flag, \BR
* shared_ARM2SP_HQ_DL_InBuf_out, \BR
* ARM2SP_HQ_DL_flag )\BR
* whenever finishing every half PCM data. shared_ARM2SP_HQ_DL_InBuf_out
* indicates the next index DSP will read.
*
* To stop ARM2SP_HQ_DL:\BR
* Quick stop: ARM disables ARM2SP_HQ_DL using
* VP_COMMAND_SET_ARM2SP_HQ_DL (arg0=0).\BR
* \note The ARM2SP_HQ_DL will be stopped right away.
* There may be un-finished PCM data in shared_Arm2SP_InBuf[].\BR
*
* Finishing stop: After finishing filling the last 2/4-speech frame PCM data,
* ARM set shared_ARM2SP_HQ_DL_done_flag=1 to indicate the end of PCM data.
* DSP ISR finishes the last 2/4-speech frames and disables ARM2SP_HQ_DL.
* DSP sends VP_STATUS_ARM2SP_HQ_DL_EMPTY and arg0=0. \BR
*
* To pause ARM2SP_HQ_DL: ARM disables ARM2SP_HQ_DL using
* VP_COMMAND_SET_ARM2SP_HQ_DL (arg0=0, arg1=1).
*
* \see shared_Arm2SP_InBuf, shared_ARM2SP_HQ_DL_InBuf_out,
* shared_ARM2SP_HQ_DL_done_flag, VP_COMMAND_SET_ARM2SP_HQ_DL,
* VP_STATUS_ARM2SP_HQ_DL_EMPTY,
*
* @{ */
/**
* This flag indicates the end of the transfer of data from the ARM to the
* DSP. The transfer of data by ARM2SP_HQ_DL interface can be disabled by
* setting shared_ARM2SP_HQ_DL_done_flag=1 after filling the
* shared_Arm2SP_InBuf[] buffer.
* DSP will disable the feature when it reaches the next 1/2-speech frame
* boundary. */
EXTERN UInt16 shared_ARM2SP_HQ_DL_done_flag AP_SHARED_SEC_DIAGNOS;
/**
* This index indicates the next index DSP will read from the shared_Arm2SP_InBuf[]
* buffer.
*
* \note This index is changed only by the DSP.
*
* */
EXTERN UInt16 shared_ARM2SP_HQ_DL_InBuf_out AP_SHARED_SEC_DIAGNOS;
/**
* The gain factor, shared_arm2speech_hq_call_gain_dl is applied on the 48kHz PCM data
* on the downlink path.\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 2.14 format).\BR
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_arm2speech_hq_call_gain_dl[5] AP_SHARED_SEC_DIAGNOS;
/** @} */
/**
* @addtogroup ARM2SP2_HQ_DL_interface
*
* This interface is used for mixing/playing \Bold{48kHz Mono/Stereo PCM data
* to Downlink}.\BR
*
* This PCM data will be mixed after Downlink speech processing in the 48kHz
* mode.
*
* \note This interface works in only one way - from ARM to DSP. \BR
*
* One use-case of this interface is to support S-barge-in feature where a
* music might be playing in the downlink using this interface and an
* uplink record/speech recognition happens at the same time, then this
* UL would get the benefit of echo cancellation of the music.
*
* In this interface 2 or 4-speech frames worth of data is transferred at a
* time from ARM to DSP in ping-pong buffers - i.e. ARM fills empty half of
* shared_Arm2SP2_InBuf[].
*
* When shared_ARM2SP2_HQ_DL_InBuf_out=0, ARM fills next half number of words from
* shared_Arm2SP2_InBuf[half].
* When shared_ARM2SP2_HQ_DL_InBuf_out=half, ARM fills half number of words from
* shared_Arm2SP2_InBuf[0]. \BR
*
* To start ARM2SP2_HQ_DL: ARM enables ARM2SP2_HQ_DL by
* VP_COMMAND_SET_ARM2SP2_HQ_DL.\BR
*
* To resume ARM2SP2_HQ_DL : ARM enables ARM2SP2_HQ_DL by
* VP_COMMAND_SET_ARM2SP2_HQ_DL with arg1=1.
*
* To control shared_Arm2SP2_InBuf[]: DSP interrupts ARM with \BR
* VP_STATUS_ARM2SP2_HQ_DL_EMPTY( shared_ARM2SP2_HQ_DL_done_flag, \BR
* shared_ARM2SP2_HQ_DL_InBuf_out, \BR
* ARM2SP2_HQ_DL_flag )\BR
* whenever finishing every half PCM data. shared_ARM2SP2_HQ_DL_InBuf_out
* indicates the next index DSP will read.
*
* To stop ARM2SP2_HQ_DL:\BR
* Quick stop: ARM disables ARM2SP2_HQ_DL using
* VP_COMMAND_SET_ARM2SP2_HQ_DL (arg0=0).\BR
* \note The ARM2SP2_HQ_DL will be stopped right away.
* There may be un-finished PCM data in shared_Arm2SP2_InBuf[].\BR
*
* Finishing stop: After finishing filling the last 2/4-speech frame PCM data,
* ARM set shared_ARM2SP2_HQ_DL_done_flag=1 to indicate the end of PCM data.
* DSP ISR finishes the last 2/4-speech frames and disables ARM2SP2_HQ_DL.
* DSP sends VP_STATUS_ARM2SP2_HQ_DL_EMPTY and arg0=0. \BR
*
* To pause ARM2SP2_HQ_DL: ARM disables ARM2SP2_HQ_DL using
* VP_COMMAND_SET_ARM2SP2_HQ_DL (arg0=0, arg1=1).
*
* \see shared_Arm2SP2_InBuf, shared_ARM2SP2_HQ_DL_InBuf_out,
* shared_ARM2SP2_HQ_DL_done_flag, VP_COMMAND_SET_ARM2SP2_HQ_DL,
* VP_STATUS_ARM2SP2_HQ_DL_EMPTY,
*
* @{ */
/**
* This flag indicates the end of the transfer of data from the ARM to the
* DSP. The transfer of data by ARM2SP2_HQ_DL interface can be disabled by
* setting shared_ARM2SP2_HQ_DL_done_flag=1 after filling the
* shared_Arm2SP2_InBuf[] buffer.
* DSP will disable the feature when it reaches the next 1/2-speech frame
* boundary. */
EXTERN UInt16 shared_ARM2SP2_HQ_DL_done_flag AP_SHARED_SEC_DIAGNOS;
/**
* This index indicates the next index DSP will read from the shared_Arm2SP2_InBuf[]
* buffer.
*
* \note This index is changed only by the DSP.
*
* */
EXTERN UInt16 shared_ARM2SP2_HQ_DL_InBuf_out AP_SHARED_SEC_DIAGNOS;
/**
* The gain factor, shared_arm2speech2_hq_call_gain_dl is applied on the 48kHz PCM data
* on the downlink path.\BR
*
* This gain is a ramped gain in Q14 format as shown below (Q14 means 2.14 format).\BR
*
* {
* - Int16 Start_Gain_in_Q14,
* - Int16 Target_Gain_in_Q14,
* - Int16 Step_size_to_increment_or_decrement_the_Gain_in_Q14, (<= 0 make Start_Gain_in_Q14 = Target_Gain_in_Q14)
* - Int16 Reserved,
* - Int16 Reserved \BR
* }
*
*/
EXTERN Int16 shared_arm2speech2_hq_call_gain_dl[5] AP_SHARED_SEC_DIAGNOS;
/** @} */
/**
* @}
*/
EXTERN UInt32 NOT_USE_shared_memory_end AP_SHARED_SEC_DIAGNOS;
#ifdef MSP
} AP_SharedMem_t;
#endif
//******************************************************************************
// Function Prototypes
//******************************************************************************
#ifdef RIP
void VPSHARED_Init( void ); // Initialize the shared memory
#endif // RIP
/**
* @}
*/
#endif // _INC_SHARED_AP_H_