mbtk/include/gsw/gsw_voice_interface.h

/**
*  @file  : gsw_voice_interface.h
*  @brief : voice mic and rtp
*  @date : 2022-06-29
*  @author : 
*  @version : v1.0
*  @copyright copyright Copyright (c) 2022 www.gosuncn.com
*/
#ifndef GSW_VOICE_INTERFACE_H
#define GSW_VOICE_INTERFACE_H

#include <stdint.h>
#include <stdbool.h>
#include "gsw_hal_errcode.h"

#ifdef __cplusplus
extern "C" {
#endif

#define INVALID_CALL_HANDLE (-1)
#define GSW_VOICE_SUCCESS      GSW_HAL_SUCCESS
#define GSW_VOICE_ERROR        GSW_HAL_NORMAL_FAIL
#define ERR_FAILED         GSW_HAL_NORMAL_FAIL
#define ERR_SUCCESS        GSW_HAL_SUCCESS

#define GSW_VOICE_MAX_SPEAKER_VOLUME (7)
#define GSW_VOICE_MIN_SPEAKER_VOLUME (1)
#define GSW_VOICE_MAX_MIC_VOLUME     (7)
#define GSW_VOICE_MIN_MIC_VOLUME     (1)

typedef int CallHandle;

typedef enum {
    GSW_VOICE_CALL_HOLDING = 0, /**< holding */
    GSW_VOICE_CALL_DIALING,     /**< dialing */
    GSW_VOICE_CALL_ALERTING,    /**< alerting */
    GSW_VOICE_CALL_CONNECTED,   /**< connected */
    GSW_VOICE_CALL_INCOMING,    /**< incoming */
    GSW_VOICE_CALL_WAITING,     /**< waiting */
    GSW_VOICE_CALL_END,         /**< call end */
} VoiceCallState;

typedef enum {
    GSW_AUDIO_MODE_CODEC = 0,   /**< Codec */
    GSW_AUDIO_MODE_RTP = 1,     /**< RTP */
} AudioMode;

typedef enum {
    GSW_RTP_CLIENT = 0, /**< rtp client */
    GSW_RTP_SERVER,     /**< rtp server */
} RTPMode;

typedef void (*CallStateInd)(CallHandle, VoiceCallState);

/**
 * @brief voice sdk init function
 * @param  [in] ind callback function to handle new voice call state
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_sdk_init(CallStateInd ind);

/**
 * @brief voice sdk deinit function
 * @retval 0: success
 * @retval other: fail
 */
void gsw_voice_sdk_deinit(void);

/**
 * @brief normal voice start
 * @param  [in] handle
 * @param  [in] callNumber
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_normal_voice_start(CallHandle *handle, const char *callNumber);

/**
 * @brief hangup voice
 * @param  [in] handle
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_hangup(CallHandle handle);

/**
 * @brief answer voice
 * @param  [in] handle
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_answer(CallHandle handle);

/**
 * @brief auto answer mode
 * @param  [in] mode: ture/false
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_set_auto_answer_mode(int32_t mode);

/**
 * @brief get current call state
 * @param  [in] handle
 * @param  [in] state
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_get_current_call_state(CallHandle handle, VoiceCallState *state);

/**
 * @brief get speaker volume
 * @param  [out] volume: 0 ~ 7
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_get_speaker_volume(int32_t *volume);

/**
 * @brief set speaker volume
 * @param  [in] volume: 0 ~ 7
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_set_speaker_volume(int32_t volume);

/**
 * @brief set mic volume
 * @param  [out] volume: 0 ~ 7
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_get_mic_volume(int32_t *volume);

/**
 * @brief set mic volume
 * @param  [in] volume: 0 ~ 7
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_set_mic_volume(int32_t volume);

/**
 * @brief set audio mode. 
 * @details mode change from GSW_AUDIO_MODE_RTP to GSW_AUDIO_MODE_CODEC, should close RTP immediately and change to codec mode.  
 * change from GSW_AUDIO_MODE_CODEC to GSW_AUDIO_MODE_RTP, should close codec immediately and change to RTP mode.
 * 
 * @param  [in] audioMode
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_set_audio_mode(AudioMode audioMode);

/**
 * @brief get audio mode
 * @param  [out] audioMode
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_get_audio_mode(AudioMode *audioMode);

/**
 * @brief set rtp ip
 * @param  [in] ip rtp ip
 * @param  [in] len rtp ip length
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_set_remote_rtp_ip(const char *ip, int32_t len);

/**
 * @brief get rtp ip
 * @param  [out] ip rtp ip
 * @param  [out] len rtp ip length
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_get_remote_rtp_ip(char *ip, int32_t len);

/**
 * @brief set rtp port
 * @param  [in] RTPMode rtp mode
 * @param  [in] port rtp port
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_set_rtp_port(RTPMode rtpMode, int32_t port);

/**
 * @brief get rtp port
 * @param  [out] RTPMode rtp mode
 * @param  [out] port rtp port
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_get_rtp_port(RTPMode rtpMode, int32_t *port);

/**
 * @brief set rtp port
 * @param  [in] clockRate rtp clock rate
 * @param  [in] channel rtp channel
 * @param  [in] latency rtp latency
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_set_rtp_param(int32_t clockRate, int32_t channel, int32_t latency);

/**
 * @brief get rtp param
 * @param  [out] clockRate rtp clock rate
 * @param  [out] channel rtp channel
 * @param  [out] latency rtp latency
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_get_rtp_param(int32_t *clockRate, int32_t *channel, int32_t *latency);

/**
 * @brief set rtp vlan
 * @param  [in] interfaceName network interface name
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_set_rtp_vlan_info(const char *interfaceName);

/**
 * @brief get rtp vlan
 * @param  [out] interfaceName network interface name
 * @param  [out] len network interface name length
 * @retval 0: success
 * @retval other: fail
 */
int32_t gsw_voice_get_rtp_vlan_info(char *interfaceName, uint32_t len);

/**
 * @brief get current call end reason
 * @param  [in] handle
 * @retval call_end_reason
 */
int32_t gsw_voice_get_current_call_end_reason(CallHandle handle);

#ifdef __cplusplus
}
#endif
#endif /* GSW_VOICE_INTERFACE_H */