[干货分享]GB28181执法记录仪方案设计与技术探究

public class MainActivity extends Activity implements ViewTreeObserver.OnGlobalLayoutListener, Camera2Listener,
        GBSIPAgentListener, GBSIPAgentPlayListener, GBSIPAgentAudioBroadcastListener,
        GBSIPAgentDeviceControlListener, GBSIPAgentQueryCommandListener, 
        GBSIPAgentTalkListener, 
        GBSIPAgentQueryRecordInfoListener{
}

public interface GBSIPAgentListener
{
    /*注册成功
    * @param dateString: 服务器日期，用来校准设备端时间，用户自行决定是否校准设备时间
    */
    void ntsRegisterOK(String dateString);

    /*
    *注册超时
    */
    void ntsRegisterTimeout();

    /*
    *注册网络传输层异常
    */
    void ntsRegisterTransportError(String errorInfo);

    /*
    *心跳达到异常次数
    */
    void ntsOnHeartBeatException(int exceptionCount, String lastExceptionInfo);

    /*
     * 设备位置请求, 这个主要用在移动设备位置订阅上
     * @param interval 请求间隔, 单位是毫秒
     */
    void ntsOnDevicePositionRequest(String deviceId, int interval);
}

public interface GBSIPAgentPlayListener {

    /*
     *收到s=Play的实时视音频点播
     */
    void ntsOnInvitePlay(String deviceId, SessionDescription sessionDescription);

    /*
     *发送play invite response 异常
     */
    void ntsOnPlayInviteResponseException(String deviceId, int statusCode, String errorInfo);

    /*
     * 收到CANCEL play INVITE请求
     */
    void ntsOnCancelPlay(String deviceId);

    /*
     * 收到Ack
     */
    void ntsOnAckPlay(String deviceId);

    /*
     * 收到Bye
     */
    void ntsOnByePlay(String deviceId);

    /*
     * 不是在收到BYE Message情况下， 终止Play
     */
    void ntsOnTerminatePlay(String deviceId);

    /*
     * Play会话对应的对话终止, 一般不会出发这个回调,目前只有在响应了200K, 但在64*T1时间后还没收到ACK,才可能会出发
    收到这个, 请做相关清理处理
    */
    void ntsOnPlayDialogTerminated(String deviceId);
}

public interface GBSIPAgentAudioBroadcastListener {

    /*
     *收到语音广播通知
     */
    void ntsOnNotifyBroadcastCommand(String fromUserName, String fromUserNameAtDomain, String sn, String sourceID, String targetID);

    /*
     *需要准备接受语音广播的SDP内容
     */
    void ntsOnAudioBroadcast(String commandFromUserName, String commandFromUserNameAtDomain, String sourceID, String targetID);

    /*
     *音频广播, 发送Invite请求异常
     */
    void ntsOnInviteAudioBroadcastException(String sourceID, String targetID, String errorInfo);

    /*
     *音频广播, 等待Invite响应超时
     */
    void ntsOnInviteAudioBroadcastTimeout(String sourceID, String targetID);

    /*
     *音频广播, 收到Invite消息最终响应
     */
    void ntsOnInviteAudioBroadcastResponse(String sourceID, String targetID, int statusCode, SessionDescription sessionDescription);

    /*
     * 音频广播, 收到BYE Message
     */
    void ntsOnByeAudioBroadcast(String sourceID, String targetID);


    /*
     * 不是在收到BYE Message情况下， 终止音频广播
     */
    void ntsOnTerminateAudioBroadcast(String sourceID, String targetID);
}

public interface GBSIPAgentDeviceControlListener {

    /*
     * 收到远程启动控制命令
     */
    void ntsOnDeviceControlTeleBootCommand(String deviceId, String teleBootValue);

    /*
    * 云台控制
     */
    void ntsOnDeviceControlPTZCmd(String deviceId, String typeValue);
}

public interface GBSIPAgentQueryCommandListener {

    /*
     * 设备预置位查询
     */
    void ntsOnDevicePresetQueryCommand(String fromUserName, String fromUserNameAtDomain, String sn, String deviceId);
}

public interface GBSIPAgentTalkListener {
    /*
     *收到s=Talk 语音对讲
     */
    void ntsOnInviteTalk(String deviceId, SessionDescription sessionDescription);

    /*
     *发送talk invite response 异常
     */
    void ntsOnTalkInviteResponseException(String deviceId, int statusCode, String errorInfo);

    /*
     * 收到CANCEL Talk INVITE请求
     */
    void ntsOnCancelTalk(String deviceId);

    /*
     * 收到Ack
     */
    void ntsOnAckTalk(String deviceId);

    /*
     * 收到Bye
     */
    void ntsOnByeTalk(String deviceId);

    /*
     * 不是在收到BYE Message情况下， 终止Talk
     */
    void ntsOnTerminateTalk(String deviceId);

    /*
     * Talk会话对应的对话终止, 一般不会出发这个回调,目前只有在响应了200K, 但在64*T1时间后还没收到ACK,才可能会出发
    收到这个, 请做相关清理处理
    */
    void ntsOnTalkDialogTerminated(String deviceId);
}

public interface GBSIPAgentPlaybackListener {
    void ntsOnInvitePlayback(long var1, String var3, SessionDescription var4);

    void ntsOnPlaybackInviteResponseException(long var1, String var3, int var4, String var5);

    void ntsOnCancelPlayback(long var1, String var3);

    void ntsOnAckPlayback(long var1, String var3);

    void ntsOnPlaybackMANSRTSPPlayCommand(long var1, String var3);

    void ntsOnPlaybackMANSRTSPPauseCommand(long var1, String var3);

    void ntsOnPlaybackMANSRTSPScaleCommand(long var1, String var3, double var4);

    void ntsOnPlaybackMANSRTSPSeekCommand(long var1, String var3, double var4);

    void ntsOnPlaybackMANSRTSPTeardownCommand(long var1, String var3);

    void ntsOnByePlayback(long var1, String var3);

    void ntsOnTerminatePlayback(long var1, String var3);

    void ntsOnPlaybackDialogTerminated(long var1, String var3);
}

public interface GBSIPAgentDownloadListener {
    void ntsOnInviteDownload(long var1, String var3, SessionDescription var4);

    void ntsOnDownloadInviteResponseException(long var1, String var3, int var4, String var5);

    void ntsOnCancelDownload(long var1, String var3);

    void ntsOnAckDownload(long var1, String var3);

    void ntsOnDownloadMANSRTSPScaleCommand(long var1, String var3, double var4);

    void ntsOnByeDownload(long var1, String var3);

    void ntsOnTerminateDownload(long var1, String var3);

    void ntsOnDownloadDialogTerminated(long var1, String var3);
}

/*
 * SmartPublisherJniV2.java
 * Author: https://daniusdk.com
 * WeChat：xinsheng120
 */
/*
 * 创建RTP Sender实例
 *
 * @param reserve：保留参数传0
 *
 * @return RTP Sender 句柄，0表示失败
 */
public native long CreateRTPSender(int reserve);

/**
 *设置 RTP Sender传输协议
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param transport_protocol, 0:UDP, 1:TCP, 默认是UDP
 *
 * @return {0} if successful
 */
public native int SetRTPSenderTransportProtocol(long rtp_sender_handle, int transport_protocol);

/**
 *设置 RTP Sender IP地址类型
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param ip_address_type, 0:IPV4, 1:IPV6, 默认是IPV4, 当前仅支持IPV4
 *
 * @return {0} if successful
 */
public native int SetRTPSenderIPAddressType(long rtp_sender_handle, int ip_address_type);

/**
 *设置 RTP Sender RTP Socket本地端口
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param port, 必须是偶数，设置0的话SDK会自动分配, 默认值是0
 *
 * @return {0} if successful
 */
public native int SetRTPSenderLocalPort(long rtp_sender_handle, int port);

/**
 *设置 RTP Sender SSRC
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param ssrc, 如果设置的话，这个字符串要能转换成uint32类型, 否则设置失败
 *
 * @return {0} if successful
 */
public native int SetRTPSenderSSRC(long rtp_sender_handle, String ssrc);

/**
 *设置 RTP Sender RTP socket 发送Buffer大小
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param buffer_size, 必须大于0, 默认是512*1024, 当前仅对UDP socket有效, 根据视频码率考虑设置合适的值
 *
 * @return {0} if successful
 */
public native int SetRTPSenderSocketSendBuffer(long rtp_sender_handle, int buffer_size);

/**
 *设置 RTP Sender RTP时间戳时钟频率
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param clock_rate, 必须大于0, 对于GB28181 PS规定是90kHz, 也就是90000
 *
 * @return {0} if successful
 */
public native int SetRTPSenderClockRate(long rtp_sender_handle, int clock_rate);

/**
 *设置 RTP Sender 目的IP地址, 注意当前用在GB2818推送上,只设置一个地址,将来扩展如果用在其他地方，可能要设置多个目的地址，到时候接口可能会调整
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param address, IP地址
 * @param port, 端口
 *
 * @return {0} if successful
 */
public native int SetRTPSenderDestination(long rtp_sender_handle, String address, int port);

/**
 * 设置是否开启 RTP Receiver
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param is_enable, 0表示不收RTP包, 1表示收RTP包, SDK默认值为0.
 * @return
 */
public native int EnableRTPSenderReceive(long rtp_sender_handle, int is_enable);

/**
 *设置RTP Receiver SSRC
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param ssrc, 如果设置的话，这个字符串要能转换成uint32类型, 否则设置失败
 *
 * @return {0} if successful
 */
public native int SetRTPSenderReceiveSSRC(long rtp_sender_handle, String ssrc);

/**
 *设置RTP Receiver Payload 相关信息
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 *
 * @param payload_type, 请参考 RFC 3551
 *
 * @param encoding_name, 编码名, 请参考 RFC 3551, 如果payload_type不是动态的, 可能传null就好
 *
 * @param media_type, 媒体类型, 请参考 RFC 3551, 1 是视频, 2是音频
 *
 * @param clock_rate, 请参考 RFC 3551
 *
 * @return {0} if successful
 */
public native int SetRTPSenderReceivePayloadType(long rtp_sender_handle, int payload_type, String encoding_name, int media_type, int clock_rate);

/**
 *设置RTP Receiver PS的pts和dts clock frequency
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 *
 * @param ps_clock_frequency, 默认是90000, 一些特殊场景需要设置
 *
 * @return {0} if successful
 */
public native int SetRTPSenderReceivePSClockFrequency(long rtp_sender_handle, int ps_clock_frequency);

/**
 *设置 RTP Receiver 音频采样率
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param sampling_rate, 音频采样率
 *
 * @return {0} if successful
 */
public native int SetRTPSenderReceiveAudioSamplingRate(long rtp_sender_handle, int sampling_rate);

/**
 *设置 RTP Receiver 音频通道数
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param channels, 音频通道数
 *
 * @return {0} if successful
 */
public native int SetRTPSenderReceiveAudioChannels(long rtp_sender_handle, int channels);

/**
 *初始化RTP Sender, 初始化之前先调用上面的接口配置相关参数
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 *
 * @return {0} if successful
 */
public native int InitRTPSender(long rtp_sender_handle);

/**
 *获取RTP Sender RTP Socket本地端口
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 *
 * @return 失败返回0, 成功的话返回响应的端口, 请在InitRTPSender返回成功之后调用
 */
public native int GetRTPSenderLocalPort(long rtp_sender_handle);

/**
 * UnInit RTP Sender
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 *
 * @return {0} if successful
 */
public native int UnInitRTPSender(long rtp_sender_handle);

/**
 * 释放RTP Sender, 释放之后rtp_sender_handle就无效了，请不要再使用
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 *
 * @return {0} if successful
 */
public native int DestoryRTPSender(long rtp_sender_handle);

/*
 * SmartPlayerJniV2.java
 * Author: https://daniusdk.com
 * WeChat: xinsheng120
 */
/*
 * 创建RTP Receiver
 *
 * @param reserve：保留参数传0
 *
 * @return RTP Receiver 句柄，0表示失败
 */
public native long CreateRTPReceiver(int reserve);


/**
 *设置 RTP Receiver传输协议
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 * @param transport_protocol, 0:UDP, 1:TCP, 默认是UDP
 *
 * @return {0} if successful
 */
public native int SetRTPReceiverTransportProtocol(long rtp_receiver_handle, int transport_protocol);


/**
 *设置 RTP Receiver IP地址类型
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 * @param ip_address_type, 0:IPV4, 1:IPV6, 默认是IPV4
 *
 * @return {0} if successful
 */
public native int SetRTPReceiverIPAddressType(long rtp_receiver_handle, int ip_address_type);


/**
 *设置 RTP Receiver RTP Socket本地端口
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 * @param port, 必须是偶数，设置0的话SDK会自动分配, 默认值是0
 *
 * @return {0} if successful
 */
public native int SetRTPReceiverLocalPort(long rtp_receiver_handle, int port);


/**
 *设置 RTP Receiver SSRC
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 * @param ssrc, 如果设置的话，这个字符串要能转换成uint32类型, 否则设置失败
 *
 * @return {0} if successful
 */
public native int SetRTPReceiverSSRC(long rtp_receiver_handle, String ssrc);


/**
 *创建 RTP Receiver 会话
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 * @param reserve, 保留值，目前传0
 *
 * @return {0} if successful
 */
public native int CreateRTPReceiverSession(long rtp_receiver_handle, int reserve);


/**
 *获取 RTP Receiver RTP Socket本地端口
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 *
 * @return 失败返回0, 成功的话返回响应的端口, 请在CreateRTPReceiverSession返回成功之后调用
 */
public native int GetRTPReceiverLocalPort(long rtp_receiver_handle);


/**
 *设置 RTP Receiver Payload 相关信息
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 *
 * @param payload_type, 请参考 RFC 3551
 *
 * @param encoding_name, 编码名, 请参考 RFC 3551, 如果payload_type不是动态的, 可能传null就好
 *
 * @param media_type, 媒体类型, 请参考 RFC 3551, 1 是视频, 2是音频
 *
 * @param clock_rate, 请参考 RFC 3551
 *
 * @return {0} if successful
 */
public native int SetRTPReceiverPayloadType(long rtp_receiver_handle, int payload_type, String encoding_name, int media_type, int clock_rate);


/**
 *设置 RTP Receiver 音频采样率
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 * @param sampling_rate, 音频采样率
 *
 * @return {0} if successful
 */
public native int SetRTPReceiverAudioSamplingRate(long rtp_receiver_handle, int sampling_rate);

/**
 *设置 RTP Receiver 音频通道数
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 * @param channels, 音频通道数
 *
 * @return {0} if successful
 */
public native int SetRTPReceiverAudioChannels(long rtp_receiver_handle, int channels);


/**
 *设置 RTP Receiver 远端地址
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 * @param address, IP地址
 * @param port, 端口
 *
 * @return {0} if successful
 */
public native int SetRTPReceiverRemoteAddress(long rtp_receiver_handle, String address, int port);

/**
 *初始化 RTP Receiver
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 *
 * @return {0} if successful
 */
public native int InitRTPReceiver(long rtp_receiver_handle);

/**
 *UnInit RTP Receiver
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 *
 * @return {0} if successful
 */
public native int UnInitRTPReceiver(long rtp_receiver_handle);


/**
 *Destory RTP Receiver Session
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 *
 * @return {0} if successful
 */
public native int DestoryRTPReceiverSession(long rtp_receiver_handle);


/**
 *Destory RTP Receiver
 *
 * @param rtp_receiver_handle, CreateRTPReceiver
 *
 * @return {0} if successful
 */
public native int DestoryRTPReceiver(long rtp_receiver_handle);

/*
 * SmartPlayerJniV2.java
 * Author: https://daniusdk.com
 */
/**
 * 投递音频包给外部Live source, 注意ByteBuffer对象必须是DirectBuffer
 *
 * @param handle: return value from SmartPlayerOpen()
 *
 * @return {0} if successful
 */
public native int PostAudioPacket(long handle, int codec_id,
                          java.nio.ByteBuffer packet, int offset, int size, long pts, boolean is_pts_discontinuity,
                          java.nio.ByteBuffer extra_data, int extra_data_offset, int extra_data_size, int sample_rate, int channels);

/*
 * SmartPublisherJniV2.java
 * Author: https://daniusdk.com
 * WeChat: xinsheng120
 */
/**
 * 设置GB28181 RTP Sender
 *
 * @param rtp_sender_handle, CreateRTPSender返回值
 * @param rtp_payload_type, 对于GB28181 PS, 协议定义是96, 具体以SDP为准,  RFC 3551有定义
 * @param encoding_name, 编码名, 请参考 RFC 3551, 当前仅支持: "PS", 其他值返回失败
 * @return {0} if successful
 */
public native int SetGB28181RTPSender(long handle, long rtp_sender_handle, int rtp_payload_type, String encoding_name);

/**
 * 设置GB28181 RTP 收到的音频包回调
 * @param handle
 * @param audio_packet_callback
 * @return
 */
public native int SetGB28181ReceiveAudioPacketCallback(long handle, NTAudioPacketCallback audio_packet_callback);

/**
 * 启动 GB28181 媒体流
 *
 * @return {0} if successful
 */
public native int StartGB28181MediaStream(long handle);

/**
 * 停止 GB28181 媒体流
 *
 * @return {0} if successful
 */
public native int StopGB28181MediaStream(long handle);