前往小程序,Get更优阅读体验!
立即前往
首页
学习
活动
专区
工具
TVP
发布
社区首页 >专栏 >如何在Android平台GB28181接入终端实现语音广播和语音对讲

如何在Android平台GB28181接入终端实现语音广播和语音对讲

原创
作者头像
音视频牛哥
发布2022-08-22 01:31:35
1.2K0
发布2022-08-22 01:31:35
举报
文章被收录于专栏:GB28181技术RTSP/RTMP直播相关
技术背景

在之前的blog,我们以Android平台国标接入终端为例,分别介绍了一些常规的功能,比如REGISTER、CATALOG、INVITE、Keepalive、SUBSCRIBE、NOTIFY等常规操作,今天主要介绍下语音广播和语音对讲这部分。

GB28181平台广播和对讲这块,重要性不言而喻,没有广播的接入终端,数据只是单向流入,加入后,指挥中心和终端之间的联系更紧密,实时双向沟通更方便,适用的行业范围也更广泛。

相关SPEC解读

关于语音广播和对讲,感兴趣的开发者可直接参阅GBT 28181-2016.pdf相关技术规范里面的9.12章节,以下是部分精选介绍:

命令交互流程

命令描述流程

a) 1:SIP服务器向语音流接收者发送语音广播通知消息,消息中通过 To头域标明作为目的地址 的语音流接收者ID,消息采用 Message方法携带。

举例说明:

代码语言:javascript
复制
MESSAGE sip:34020000001380000001@192.168.2.212:12070 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.154:15060;rport;branch=z9hG4bK311226558
From: <sip:34020000002000000001@3402000000>;tag=280226558
To: <sip:34020000001380000001@192.168.2.212:12070>
Call-ID: 172226558
CSeq: 207 MESSAGE
Content-Type: Application/MANSCDP+xml
Max-Forwards: 70
Content-Length: 206

<?xml version="1.0" encoding="GB2312"?>
<Notify>
  <CmdType>Broadcast</CmdType>
  <SN>461226558</SN>
  <SourceID>34020000002000000001</SourceID>
  <TargetID>34020000001380000001</TargetID>
</Notify>

b) 2:语音流接收者收到语音广播通知消息后,向SIP服务器发送200OK 响应。

代码语言:javascript
复制
SIP/2.0 200 OK
CSeq: 207 MESSAGE
Call-ID: 172226558
From: <sip:34020000002000000001@3402000000>;tag=280226558
To: <sip:34020000001380000001@192.168.2.212:12070>
Via: SIP/2.0/UDP 192.168.2.154:15060;rport=15060;branch=z9hG4bK311226558;received=192.168.2.154
Content-Length: 0

c) 3:语音流接收者向SIP服务器发送语音广播应答消息,消息中通过 To头域标明作为目的地 址的SIP服务器ID,消息采用 Message方法携带。

代码语言:javascript
复制
MESSAGE sip:34020000002000000001@3402000000 SIP/2.0
Call-ID: 0fc1f2c83c28898a29e146d7ef581908@192.168.2.212
CSeq: 337044229 MESSAGE
From: <sip:34020000001380000001@3402000000>;tag=93882333
To: <sip:34020000002000000001@3402000000>
Via: SIP/2.0/UDP 192.168.2.212:12070;rport;branch=z9hG4bK-363733-79fd88c45667975e5ebaf18f84b91a8e
Max-Forwards: 70
User-Agent: NT GB28181 User Agent V1.2(daniusdk.com)
Content-Type: Application/MANSCDP+xml
Content-Length: 180

<?xml version="1.0" encoding="GB2312"?>
<Response>
<CmdType>Broadcast</CmdType>
<SN>461226558</SN>
<DeviceID>34020000001380000001</DeviceID>
<Result>OK</Result>
</Response>

d) 4:SIP服务器收到语音广播应答消息后,向语音流接收者发送200OK 响应。

代码语言:javascript
复制
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.212:12070;rport=12070;received=192.168.2.212;branch=z9hG4bK-363733-79fd88c45667975e5ebaf18f84b91a8e
From: <sip:34020000001380000001@3402000000>;tag=93882333
To: <sip:34020000002000000001@3402000000>;tag=355226593
CSeq: 337044229 MESSAGE
Call-ID: 0fc1f2c83c28898a29e146d7ef581908@192.168.2.212
Content-Length: 0

e) 5:语音流接收者向SIP服务器发送Invite消息,消息中通过 To头域标明作为目的地址的语音 流发送者ID,消息头域中携带Subject字段,表明请求的语音流发送者ID、发送方媒体流序列 号、语音流接收者ID、接收方媒体流序列号等参数,SDP消息体中s字段为“Play”代表实时点 播,m 字段中媒体参数标识为“audio”表示请求语音媒体流。

代码语言:javascript
复制
INVITE sip:34020000002000000001@3402000000 SIP/2.0
Call-ID: 2b4f0f0512aa1a49ffc645618d0e8bae@192.168.2.212
CSeq: 44264 INVITE
From: <sip:34020000001380000001@3402000000>;tag=32ecf22a
To: <sip:34020000002000000001@3402000000>
Via: SIP/2.0/UDP 192.168.2.212:12070;rport;branch=z9hG4bK-363733-15283c8a0ea0a1e9dbf295ce2359dbe7
Max-Forwards: 70
Contact: <sip:34020000001380000001@192.168.2.212:12070>
Subject: 34020000002000000001:0200006727,34020000001380000001:0
User-Agent: NT GB28181 User Agent V1.2(daniusdk.com)
Content-Type: APPLICATION/SDP
Content-Length: 221

v=0
o=34020000002000000001 0 0 IN IP4 192.168.2.212
s=Play
c=IN IP4 192.168.2.212
t=0 0
m=audio 25002 TCP/RTP/AVP 8
a=setup:active
a=connection:new
a=recvonly
a=rtpmap:8 PCMA/8000
y=0200006727
f=v/////a/1/8/1

f) 6:SIP服务器收到Invite请求后,通过三方呼叫控制建立媒体服务器和语音流发送者之间的媒体连接。向媒体服务器发送Invite消息,此消息不携带SDP消息体。

g) 7:媒体服务器收到SIP服务器的Invite请求后,回复200OK 响应,携带SDP消息体,消息体 中描述了媒体服务器接收媒体流的IP、端口、媒体格式等内容。

h) 8:SIP服务器收到媒体服务器返回的200OK 响应后,向语音流发送者发送Invite请求,消息 中通过 To头域标明作为目的地址的语音流发送者ID,消息头域中携带 Subject字段,表明请 求的语音流发送者ID、发送方媒体流序列号、语音流接收者ID、接收方媒体流序列号等参数, 请求中携带消息7中媒体服务器回复的200OK 响应消息体,s字段为“Play”代表实时点播, m 字段中媒体参数标识为“audio”表示请求语音媒体流,增加y字段描述SSRC值,f字段描述 媒体参数。

代码语言:javascript
复制
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.212:12070;rport=12070;received=192.168.2.212;branch=z9hG4bK-363733-15283c8a0ea0a1e9dbf295ce2359dbe7
From: <sip:34020000001380000001@3402000000>;tag=32ecf22a
To: <sip:34020000002000000001@3402000000>;tag=954226632
CSeq: 44264 INVITE
Call-ID: 2b4f0f0512aa1a49ffc645618d0e8bae@192.168.2.212
Contact: <sip:34020000002000000001@192.168.2.154:15060>
Content-Length: 222
Content-Type: APPLICATION/SDP

v=0
o=34020000002000000001 0 0 IN IP4 192.168.2.154
s=Play
c=IN IP4 192.168.2.154
t=0 0
m=audio 30005 TCP/RTP/AVP 8
a=sendonly
a=rtpmap:8 PCMA/8000
a=setup:passive
a=connection:new
y=0200006727
f=v/////a/1/8/1

i) 9:语音流发送者收到SIP服务器的Invite请求后,回复200OK 响应,携带SDP消息体,消息 体中描述了媒体流发送者发送媒体流的IP、端口、媒体格式、SSRC 字段等内容,s字段为 “Play”代表实时点播,m 字段中媒体参数标识为“audio”表示请求语音媒体流。

j) 10:SIP服务器收到语音流发送者返回的200OK 响应后,向媒体服务器发送 ACK 请求,请求 中携带消息9中语音流发送者回复的200OK 响应消息体,完成与媒体服务器的Invite会话 建立过程。

k) 11:SIP服务器收到语音流发送者返回的200OK 响应后,向语音流发送者发送 ACK 请求,请 求中不携带消息体,完成与语音流发送者的Invite会话建立过程。

l) 12:完成三方呼叫控制后,SIP服务器通过 B2BUA 代理方式建立语音流接收者和媒体服务器 之间的媒体连接。在消息5中增加SSRC值,转发给媒体服务器。

m)13:媒体服务器收到Invite请求,回复200OK 响应,携带SDP消息体,消息体中描述了媒体服 务器发送媒体流的IP、端口、媒体格式、SSRC值等内容,s字段为“Play”代表实时点播,m 字段 中媒体参数标识为“audio”表示请求语音媒体流。

n) 14:SIP服务器将消息13转发给语音流接收者。

o) 15:语音流接收者收到200OK 响应后,回复 ACK 消息,完成与SIP服务器的Invite会话建立 过程。

p) 16:SIP服务器将消息15转发给媒体服务器,完成与媒体服务器的Invite会话建立过程。

q) 17:SIP服务器向语音流接收者发送 BYE消息,断开消息5、14、15建立的Invite会话。

r) 18:语音流接收者收到 BYE消息后回复200OK 响应,会话断开。

s) 19:SIP服务器向媒体服务器发送 BYE 消息,断开消息 12、13、16 建立的同媒体服务器的 Invite会话。

t) 20:媒体服务器收到 BYE消息后回复200OK 响应,会话断开。

u) 21:SIP服务器向媒体服务器发送 BYE消息,断开消息6、7、10建立的同媒体服务器的Invite 会话。

v) 22:媒体服务器收到 BYE消息后回复200OK 响应,会话断开。

w)23:SIP服务器向语音流发送者发送 BYE 消息,断开消息8、9、11建立的同语音流发送者的 Invite会话。

x) 24:语音流发送者收到 BYE消息后回复200OK 响应,会话断开。

注:语音广播通知消息除上述流程中通过SIP服务器发出外,也可由语音流发送者发出,消息中通过 To头域标明 作为目的地址的语音流接收者ID,经SIP服务器中转后发往语音流接收者;语音流接收者处理后发送应答消 息,消息中通过 To头域标明作为目的地址的语音流发送者ID,经SIP服务器中转后回复给语音流发送者。后续呼叫流程与上述流程相同。

语音对讲

语音对讲功能实现中心用户与前端用户之间的一对一语音对讲功能。 语音对讲功能由下述两个独立的流程组合实现:

a) 通过9.2的实时视音频点播功能,中心用户获得前端设备的实时视音频媒体流;

b) 通过9.12的语音广播功能,中心用户向前端对讲设备发送实时音频媒体流,语音流的封装格 式见 C.2.4音频流的 RTP封装定义。

C.2.4 音频流的 RTP封装

语音比特流宜采用标准的 RTP协议进行打包,这里只摘录G.711A律的:

在一个 RTP包中,音频载荷数据应为整数个音频编码帧,且时间长度在20ms~180ms之间。

音频载荷数据的 RTP封装参数如下:

a) G.711的主要参数 G.711A律语音编码 RTP包的负载类型(PayloadType)的参数规定如下(见IETFRFC3551— 2003中的表4): 1)负载类型(PT):8; 2) 编码名称(encodingname):PCMA; 3) 时钟频率(clockrate):8kHz; 4) 通道数:1; 5) SDP描述中“m”字段的“media”项:audio。

技术实现

语音广播接收这块,由于有之前的RTMP和RTSP播放器积累,直接在player端做相应扩展即可,当收到广播后,GB28181语音广播按钮使能。

相关接口设计如下:

代码语言:javascript
复制
/*++++++++++++++++++RTP Receiver++++++++++++++++++++++*/
  //GitHub: https://github.com/daniulive/SmarterStreaming
  //WebSite: https://daniusdk.com
  /*
   * 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);


  /*++++++++++++++++++RTP Receiver++++++++++++++++++++++*/

相关调用代码:

代码语言:javascript
复制
class ButtonGB28181AudioBroadcastListener implements OnClickListener {
        public void onClick(View v) {
            if (gb_broadcast_source_id_ != null && gb_broadcast_target_id_ != null) {
                if (gb28181_agent_ != null ) {
                    if (gb28181_agent_.byeAudioBroadcast(gb_broadcast_source_id_, gb_broadcast_target_id_) ) {
                        gb_broadcast_source_id_ = null;
                        gb_broadcast_target_id_ = null;
                        btnGB28181AudioBroadcast.setText("GB28181语音广播");
                        btnGB28181AudioBroadcast.setEnabled(false);
                    }
                }
            }

            stopAudioPlayer();
            destoryRTPReceiver();
        }
    }
代码语言:javascript
复制
@Override
public void ntsOnNotifyBroadcastCommand(String fromUserName, String fromUserNameAtDomain, String sn, String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnNotifyBroadcastCommand, fromUserName:"+ from_user_name_ + ", fromUserNameAtDomain:"+ from_user_name_at_domain_
            + ", SN:" + sn_ + ", sourceID:" + source_id_ + ", targetID:" + target_id_);

      if (gb28181_agent_ != null ) {
        gb28181_agent_.respondBroadcastCommand(from_user_name_, from_user_name_at_domain_,sn_,source_id_, target_id_, true);
        btnGB28181AudioBroadcast.setText("收到GB28181语音广播通知");
      }
    }

    private String from_user_name_;
    private String from_user_name_at_domain_;
    private String sn_;
    private String source_id_;
    private String target_id_;

    public Runnable set(String from_user_name, String from_user_name_at_domain, String sn, String source_id, String target_id) {
      this.from_user_name_ = from_user_name;
      this.from_user_name_at_domain_ = from_user_name_at_domain;
      this.sn_ = sn;
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(fromUserName, fromUserNameAtDomain, sn, sourceID, targetID),0);
}

@Override
public void ntsOnAudioBroadcast(String commandFromUserName, String commandFromUserNameAtDomain, String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnAudioBroadcastPlay, fromFromUserName:" + command_from_user_name_
            + " FromUserNameAtDomain:" + command_from_user_name_at_domain_
            + " sourceID:" + source_id_ + ", targetID:" + target_id_);

      stopAudioPlayer();
      destoryRTPReceiver();

      if (gb28181_agent_ != null ) {
        String local_ip_addr = IPAddrUtils.getIpAddress(context_);

        boolean is_tcp = true; // 考虑到跨网段, 默认用TCP传输rtp包
        rtp_receiver_handle_ = lib_player_.CreateRTPReceiver(0);
        if (rtp_receiver_handle_ != 0 ) {
          lib_player_.SetRTPReceiverTransportProtocol(rtp_receiver_handle_, is_tcp?1:0);
          lib_player_.SetRTPReceiverIPAddressType(rtp_receiver_handle_, 0);

          if (0 == lib_player_.CreateRTPReceiverSession(rtp_receiver_handle_, 0) ) {
            int local_port = lib_player_.GetRTPReceiverLocalPort(rtp_receiver_handle_);
            boolean ret = gb28181_agent_.inviteAudioBroadcast(command_from_user_name_,command_from_user_name_at_domain_,
                                                              source_id_, target_id_, "IP4", local_ip_addr, local_port, is_tcp?"TCP/RTP/AVP":"RTP/AVP");

            if (!ret ) {
              destoryRTPReceiver();
              btnGB28181AudioBroadcast.setText("GB28181语音广播");
            }
            else {
              btnGB28181AudioBroadcast.setText("GB28181语音广播呼叫中");
            }
          } else {
            destoryRTPReceiver();
            btnGB28181AudioBroadcast.setText("GB28181语音广播");
          }
        }
      }
    }

    private String command_from_user_name_;
    private String command_from_user_name_at_domain_;
    private String source_id_;
    private String target_id_;

    public Runnable set(String command_from_user_name, String command_from_user_name_at_domain, String source_id, String target_id) {
      this.command_from_user_name_ = command_from_user_name;
      this.command_from_user_name_at_domain_ = command_from_user_name_at_domain;
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(commandFromUserName, commandFromUserNameAtDomain, sourceID, targetID),0);
}

@Override
public void ntsOnInviteAudioBroadcastException(String sourceID, String targetID, String errorInfo) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnInviteAudioBroadcastException, sourceID:" + source_id_ + ", targetID:" + target_id_);

      destoryRTPReceiver();
      btnGB28181AudioBroadcast.setText("GB28181语音广播");
    }

    private String source_id_;
    private String target_id_;

    public Runnable set(String source_id, String target_id) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(sourceID, targetID),0);
}

@Override
public void ntsOnInviteAudioBroadcastTimeout(String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnInviteAudioBroadcastTimeout, sourceID:" + source_id_ + ", targetID:" + target_id_);

      destoryRTPReceiver();
      btnGB28181AudioBroadcast.setText("GB28181语音广播");
    }

    private String source_id_;
    private String target_id_;

    public Runnable set(String source_id, String target_id) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(sourceID, targetID),0);
}
代码语言:javascript
复制
@Override
public void ntsOnInviteAudioBroadcastResponse(String sourceID, String targetID, int statusCode, PlaySessionDescription sessionDescription) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnInviteAudioBroadcastResponse, statusCode:" + status_code_ +" sourceID:" + source_id_ + ", targetID:" + target_id_);

      boolean is_need_destory_rtp = true;

      if (gb28181_agent_ != null ) {
        boolean is_need_bye = 200==status_code_;

        if (200 == status_code_ && session_description_ != null && rtp_receiver_handle_ != 0 ) {
          MediaSessionDescription audio_des = session_description_.getAudioDescription();

          SDPRtpMapAttribute audio_attr = null;
          if (audio_des != null && audio_des.getRtpMapAttributes() != null && !audio_des.getRtpMapAttributes().isEmpty() )
            audio_attr = audio_des.getRtpMapAttributes().get(0);

          if ( audio_des != null && audio_attr != null ) {
            lib_player_.SetRTPReceiverSSRC(rtp_receiver_handle_, audio_des.getSSRC());

            lib_player_.SetRTPReceiverPayloadType(rtp_receiver_handle_, audio_attr.getPayloadType(),
                                                  audio_attr.getEncodingName(), 2, audio_attr.getClockRate());

            lib_player_.SetRTPReceiverRemoteAddress(rtp_receiver_handle_, audio_des.getAddress(), audio_des.getPort());
            lib_player_.InitRTPReceiver(rtp_receiver_handle_);

            if (startAudioPlay()) {
              is_need_bye = false;
              is_need_destory_rtp = false;

              gb_broadcast_source_id_ = source_id_;
              gb_broadcast_target_id_ = target_id_;
              btnGB28181AudioBroadcast.setText("终止GB28181语音广播");
              btnGB28181AudioBroadcast.setEnabled(true);
            }
          }

        } else {
          btnGB28181AudioBroadcast.setText("GB28181语音广播");
        }

        if (is_need_bye)
          gb28181_agent_.byeAudioBroadcast(source_id_, target_id_);
      }

      if (is_need_destory_rtp)
        destoryRTPReceiver();
    }

    private String source_id_;
    private String target_id_;
    private int status_code_;
    private PlaySessionDescription session_description_;

    public Runnable set(String source_id, String target_id, int status_code, PlaySessionDescription session_description) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      this.status_code_ = status_code;
      this.session_description_ = session_description;
      return this;
    }

  }.set(sourceID, targetID, statusCode, sessionDescription),0);
}

@Override
public void ntsOnByeAudioBroadcast(String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnByeAudioBroadcast sourceID:" + source_id_ + " targetID:" + target_id_);

      gb_broadcast_source_id_ = null;
      gb_broadcast_target_id_ = null;
      btnGB28181AudioBroadcast.setText("GB28181语音广播");
      btnGB28181AudioBroadcast.setEnabled(false);

      stopAudioPlayer();
      destoryRTPReceiver();
    }

    private String source_id_;
    private String target_id_;

    public Runnable set(String source_id, String target_id) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(sourceID, targetID),0);
}

@Override
public void ntsOnTerminateAudioBroadcast(String sourceID, String targetID) {
  handler_.postDelayed(new Runnable() {
    @Override
    public void run() {
      Log.i(TAG, "ntsOnTerminateAudioBroadcast sourceID:" + source_id_ + " targetID:" + target_id_);

      gb_broadcast_source_id_ = null;
      gb_broadcast_target_id_ = null;
      btnGB28181AudioBroadcast.setText("GB28181语音广播");
      btnGB28181AudioBroadcast.setEnabled(false);

      stopAudioPlayer();
      destoryRTPReceiver();
    }

    private String source_id_;
    private String target_id_;

    public Runnable set(String source_id, String target_id) {
      this.source_id_ = source_id;
      this.target_id_ = target_id;
      return this;
    }

  }.set(sourceID, targetID),0);
}
总结

至此、Android平台GB28181接入终端,如位置订阅、语音广播和语音对讲这块已经全面覆盖,加上之前的技术积累,看了下,已覆盖了以下部分:

  • ​[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;
  • 支持国标GB/T28181—2016平台接入;
  • 支持语音广播及语音对讲;
  • [实时水印]支持动态文字水印、png水印;
  • [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。​

特别是语音广播和语音对讲这块,是GB28181终端接入模块的一个核心扩展功能,在智能门禁、工业与物联网、监控等行业,用途非常广泛,技术实现这块,不要忽略的技术点还有降噪和回音消除这块,由于之前我们有技术积累,可以直接复用,对新入手的开发者来说,也提出了新的挑战,感兴趣的开发者,可以酌情参考。

原创声明:本文系作者授权腾讯云开发者社区发表,未经许可,不得转载。

如有侵权,请联系 cloudcommunity@tencent.com 删除。

原创声明:本文系作者授权腾讯云开发者社区发表,未经许可,不得转载。

如有侵权,请联系 cloudcommunity@tencent.com 删除。

评论
登录后参与评论
0 条评论
热度
最新
推荐阅读
目录
  • 技术背景
  • 相关SPEC解读
    • 命令交互流程
      • 命令描述流程
        • 语音对讲
          • C.2.4 音频流的 RTP封装
          • 技术实现
          • 总结
          相关产品与服务
          云直播
          云直播(Cloud Streaming Services,CSS)为您提供极速、稳定、专业的云端直播处理服务,根据业务的不同直播场景需求,云直播提供了标准直播、快直播、云导播台三种服务,分别针对大规模实时观看、超低延时直播、便捷云端导播的场景,配合腾讯云视立方·直播 SDK,为您提供一站式的音视频直播解决方案。
          领券
          问题归档专栏文章快讯文章归档关键词归档开发者手册归档开发者手册 Section 归档