技术背景
今天，我们主要讲讲Android平台GB28181接入模块的技术对接，Android平台GB28181接入模块设计的目的，可实现不具备国标音视频能力的 Android终端，通过平台注册接入到现有的GB/T28181—服务，可用于如智能监控、智慧零售、智慧教育、远程办公、生产运输、智慧交通、车载或执法记录仪等场景。
Android终端除支持常规的音视频数据接入外，还可以支持移动设备位置(MobilePosition)订阅和通知、语音广播和语音对讲、云台控制回调和预置位查询，支持对接数据类型如下：
编码前数据（目前支持的有YV12/NV21/NV12/I420/RGB24/RGBA32/RGB565等数据类型）；编码后数据（如无人机等264/HEVC数据，或者本地解析的MP4音视频数据）；拉取RTSP或RTMP流并接入至GB28181平台（比如其他IPC的RTSP流，可通过Android平台GB28181接入到国标平台）。
功能支持
[视频格式]H.264/H.265(Android H.265硬编码)；[音频格式]G.711 A律、AAC；[音量调节]Android平台采集端支持实时音量调节；[H.264硬编码]支持H.264特定机型硬编码；[H.265硬编码]支持H.265特定机型硬编码；[软硬编码参数配置]支持gop间隔、帧率、bit-rate设置；[软编码参数配置]支持软编码profile、软编码速度、可变码率设置；支持纯视频、音视频PS打包传输；支持RTP OVER UDP和RTP OVER TCP被动模式；支持信令通道网络传输协议TCP/UDP设置；支持注册、注销，支持注册刷新及注册有效期设置；支持设备目录查询应答；支持心跳机制，支持心跳间隔、心跳检测次数设置；支持移动设备位置(MobilePosition)订阅和通知；支持语音广播；支持语音对讲；支持云台控制和预置位查询；[实时水印]支持动态文字水印、png水印；[镜像]Android平台支持前置摄像头实时镜像功能；[实时静音]支持实时静音/取消静音；[实时快照]支持实时快照；[降噪]支持环境音、手机干扰等引起的噪音降噪处理、自动增益、VAD检测；[外部编码前视频数据对接]支持YUV数据对接；[外部编码前音频数据对接]支持PCM对接；[外部编码后视频数据对接]支持外部H.264数据对接；[外部编码后音频数据对接]外部AAC数据对接；[扩展录像功能]支持和录像SDK组合使用，录像相关功能。系统要求
SDK支持Android 4.4及以上版本；支持的CPU架构：armv7, arm64, x86, x86_64。准备工作
确保SmartPublisherJniV2.java放到com.daniulive.smartpublisher包名下(可在其他包名下调用)；如需集成语音广播、语音对讲功能，确保SmartPlayerJniV2.java放到com.daniulive.smartplayer包名下(可在其他包名下调用)；smartavengine.jar和smartgbsipagent.jar加入到工程；拷贝libSmartPublisher.so和libSmartPlayer.so（如需语音广播或语音对讲）到工程；AndroidManifast.xml添加相关权限：<uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE" ></uses-permission><uses-permission android:name="android.permission.INTERNET" ></uses-permission><uses-permission android:name="android.permission.MOUNT_UNMOUNT_FILESYSTEMS" /><uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" /><uses-permission android:name="android.permission.ACCESS_COARSE_LOCATION"></uses-permission><uses-permission android:name="android.permission.ACCESS_FINE_LOCATION"></uses-permission>
Load相关so：
static { System.loadLibrary("SmartPublisher");System.loadLibrary("SmartPlayer");}
build.gradle配置32/64位库：
splits {abi {enable truereset()// Specifies a list of ABIs that Gradle should create APKs forinclude 'armeabi-v7a', 'arm64-v8a', 'x86', 'x86_64' //select ABIs to build APKs for// Specify that we do not want to also generate a universal APK that includes all ABIsuniversalApk true}}
如需集成到自己系统测试，请用大牛直播SDK的app name，授权版按照授权app name正常使用即可；如何改app-name，strings.xml做以下修改：
<string name="app_name">SmartPublisherSDKDemo</string>
接口详解
以Android平台Camera2对接为例，信令部分需要实现如下标红接口：
public class MainActivity extends Activity implements ViewTreeObserver.OnGlobalLayoutListener, Camera2Listener,GBSIPAgentListener, GBSIPAgentPlayListener, GBSIPAgentAudioBroadcastListener,GBSIPAgentDeviceControlListener, GBSIPAgentQueryCommandListener, GBSIPAgentTalkListener{}
媒体数据处理接口，可参照SmartPublisherJniV2.java，如需语音广播或语音对讲，可参照SmartPlayerJniV2.java。
信令处理
GBSIPAgentListener主要系GB28181注册、心跳、DevicePosition等，如注册成功、注册超时、注册网络传输层错误、心跳异常、设备位置请求处理：
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);}
GBSIPAgentPlayListener主要系GB28181的Invite、Ack、Bye等处理：
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);}
GBSIPAgentAudioBroadcastListener主要系GB28181语音广播处理相关，如有语音广播相关需求，可参照demo实例实现：
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);}
GBSIPAgentDeviceControlListener主要系GB28181设备控制相关，比如远程启动、云台控制：
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);}
GBSIPAgentQueryCommandListener主要系GB28181查询命令，如预置位查询：
public interface GBSIPAgentQueryCommandListener {/** 设备预置位查询*/void ntsOnDevicePresetQueryCommand(String fromUserName, String fromUserNameAtDomain, String sn, String deviceId);}
GBSIPAgentTalkListener主要系GB28181语音对讲相关处理：
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);}
媒体数据处理
RTP数据发送
RTP Sender（SmartPublisherJniV2.java）相关接口设计：
/** SmartPublisherJniV2.java* Author: *//** 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);
RTP数据接收
对应RTP Receiver（SmartPlayerJniV2.java）相关接口设计，如无语音广播或语音对讲相关技术需求，这部分可忽略：
/** SmartPlayerJniV2.java* Author: *//** 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);
PostAudioPacket（SmartPlayerJniV2.java）,投递音频包给外部Live source，目前仅于语音对讲使用：
/** SmartPlayerJniV2.java* Author: *//*** 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);
GB28181接口调用
对应GB28181相关接口调用相关设计如下：
/** SmartPublisherJniV2.java* Author: *//*** 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);
总结
以上是大牛直播SDK发布的Android平台GB28181设备接入模块的相关说明，除了上述接口设计外，模块还可以扩展实现实时静音、实时快照、按需录像、实时音量调节等，可扩展性非常好。
