Android GB28181设备接入端语音广播和语音对讲技术实现探究

上篇文章提到Android端GB28181接入端的语音广播和语音对讲的实现，从spec角度大概介绍了下流程和简单的接口设计，好多开发者私信我，希望展开说一下。其实这块难度不大，只是广播和对讲涉及到双向实现，如果之前没有相关的积累，从头实现麻烦一些而已。

语音广播的流程大家应该非常清楚了，简单来说，SIP服务器发送Broadcast语音广播命令到android接入端，接入端应答，在收到200 OK后，发送INVITE消息，Android接入端收到INVITE的200 OK响应后，回复ACK，开始读取并解析RTP包，然后对音频数据解码，输出到Android播放设备即可。

从DEMO来看，当有语音广播接入进来后，GB28181语音广播按钮会处于可用状态。

 

语音广播信令Listener如下：

package com.gb28181.ntsignalling;public interface GBSIPAgentListener
{/**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, PlaySessionDescription 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);
}

相关信令接口如下：

package com.gb28181.ntsignalling;public interface GBSIPAgent {/**语音广播应答*/void respondBroadcastCommand(String fromUserName, String fromUserNameAtDomain, String sn, String sourceID, String targetID, boolean result);/**语音广播接收者发送Invite消息, rtp ssrc暂时由sdk生成*@param addressType: ipv4:"IP4", ipv6:"IP6", 其他不支持, 填充SDP用*@param localAddress: 本地IP地址, 填充SDP用*@param localPort: 本地端口, 填充SDP用*@param mediaTransportProtocol: 媒体传输协议, rtp over udp:"RTP/AVP", rtp over tcp:"TCP/RTP/AVP". 其他不支持, 填充SDP用*/boolean inviteAudioBroadcast(String commandFromUserName, String commandFromUserNameAtDomain, String sourceID, String targetID,String addressType, String localAddress, int localPort, String mediaTransportProtocol);/**取消音频广播, 这个需要在invite收到临时响应之后，最终响应之前才能成功, 如果UAS已经发送过最终响应, UAS收到cancel不做处理, 具体参考RFC3261*/boolean cancelAudioBroadcast(String sourceID, String targetID);/**终止语音广播会话, 发送BYE消息*/boolean byeAudioBroadcast(String sourceID, String targetID);
}

  RTP音频包接收和解码输出接口，由于我们已经有非常成熟的RTMP和RTSP Player，我们是要在此基础上，扩展一些接口即可：

/** SmartPlayerJniV2.java* SmartPlayerJniV2** Github: https://github.com/daniulive/SmarterStreaming* */package com.daniulive.smartplayer;public class SmartPlayerJniV2 {
/*** Initialize Player(启动播放实例)** @param ctx: get by this.getApplicationContext()** <pre>This function must be called firstly.</pre>** @return player handle if successful, if return 0, which means init failed. */public native long SmartPlayerOpen(Object ctx);/*** Set External Audio Output(设置回调PCM数据)** @param handle: return value from SmartPlayerOpen()** @param external_audio_output:  External Audio Output** @return {0} if successful*/public native int SmartPlayerSetExternalAudioOutput(long handle, Object external_audio_output);/*** Set Audio Data Callback(设置回调编码后音频数据)** @param handle: return value from SmartPlayerOpen()** @param audio_data_callback: Audio Data Callback.** @return {0} if successful*/public native int SmartPlayerSetAudioDataCallback(long handle, Object audio_data_callback);/*** Set buffer(设置缓冲时间，单位:毫秒)** @param handle: return value from SmartPlayerOpen()** @param buffer:** <pre> NOTE: Unit is millisecond, range is 0-5000 ms </pre>** @return {0} if successful*/public native int SmartPlayerSetBuffer(long handle, int buffer);/*** Set mute or not(设置实时静音)** @param handle: return value from SmartPlayerOpen()** @param is_mute: if with 1:mute, if with 0: does not mute** @return {0} if successful*/public native int SmartPlayerSetMute(long handle, int is_mute);/*** 设置播放音量** @param handle: return value from SmartPlayerOpen()** @param volume: 范围是[0, 100], 0是静音，100是最大音量, 默认是100** @return {0} if successful*/public native int SmartPlayerSetAudioVolume(long handle, int volume);/*** 清除所有 rtp receivers** @param handle: return value from SmartPlayerOpen()** @return {0} if successful*/public native int SmartPlayerClearRtpReceivers(long handle);/*** 增加 rtp receiver** @param handle: return value from SmartPlayerOpen()** @param rtp_receiver_handle: return value from CreateRTPReceiver()** @return {0} if successful*/public native int SmartPlayerAddRtpReceiver(long handle, long rtp_receiver_handle);/*** 设置需要播放或录像的RTMP/RTSP url** @param handle: return value from SmartPlayerOpen()** @param uri: rtsp/rtmp playback/recorder uri** @return {0} if successful*/public native int SmartPlayerSetUrl(long handle, String uri);/*** Start playback stream(开始播放)** @param handle: return value from SmartPlayerOpen()** @return {0} if successful*/public native int SmartPlayerStartPlay(long handle);/*** Stop playback stream(停止播放)** @param handle: return value from SmartPlayerOpen()** @return {0} if successful*/public native int SmartPlayerStopPlay(long handle);/*** Start pull stream(开始拉流，用于数据转发，只拉流不播放)** @param handle: return value from SmartPlayerOpen()** @return {0} if successful*/public native int SmartPlayerStartPullStream(long handle);/*** Stop pull stream(停止拉流)** @param handle: return value from SmartPlayerOpen()** @return {0} if successful*/public native int SmartPlayerStopPullStream(long handle);/*** 关闭播放实例，结束时必须调用close接口释放资源** @param handle: return value from SmartPlayerOpen()** <pre> NOTE: it could not use player handle after call this function. </pre> ** @return {0} if successful*/public native int SmartPlayerClose(long handle);/*++++++++++++++++++RTP Receiver++++++++++++++++++++++*//** 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++++++++++++++++++++++*/}

上层调用DEMO实例代码：

public class AndroidGB28181Demo implements GBSIPAgentListener {private String gb_source_id_ = null;private String gb_target_id_ = null;private long player_handle_ = 0;private long rtp_receiver_handle_ = 0;private AtomicLong last_receive_audio_data_time_ = new AtomicLong(0);@Overridepublic void ntsOnNotifyBroadcastCommand(String fromUserName, String fromUserNameAtDomain, String sn, String sourceID, String targetID) {handler_.postDelayed(new Runnable() {@Overridepublic void run() {if (gb28181_agent_ != null ) {gb28181_agent_.respondBroadcastCommand(from_user_name_, from_user_name_at_domain_,sn_,source_id_, target_id_, true);}}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);}@Overridepublic void ntsOnAudioBroadcast(String commandFromUserName, String commandFromUserNameAtDomain, String sourceID, String targetID) {handler_.postDelayed(new Runnable() {@Overridepublic void run() {stopAudioPlayer();destoryRTPReceiver();if (gb28181_agent_ != null ) {String local_ip_addr = IPAddrUtils.getIpAddress(context_);boolean is_tcp = true; // 默认用TCPrtp_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();}} else {destoryRTPReceiver();}}}}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);}@Overridepublic void ntsOnInviteAudioBroadcastException(String sourceID, String targetID, String errorInfo) {handler_.postDelayed(new Runnable() {@Overridepublic void run() {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);}@Overridepublic void ntsOnInviteAudioBroadcastTimeout(String sourceID, String targetID) {handler_.postDelayed(new Runnable() {@Overridepublic void run() {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);}class PlayerExternalPCMOutput implements NTExternalAudioOutput {private int buffer_size_ = 0;private ByteBuffer pcm_buffer_ = null;@Overridepublic ByteBuffer getPcmByteBuffer(int size)  {if(size < 1)return null;if(buffer_size_ != size) {buffer_size_ = size;pcm_buffer_ = ByteBuffer.allocateDirect(buffer_size_);}return pcm_buffer_;}public void onGetPcmFrame(int ret, int sampleRate, int channel, int sampleSize, int is_low_latency) {if (null == pcm_buffer_)return;pcm_buffer_.rewind();if (ret == 0 && isGB28181StreamRunning && publisherHandle != 0 )// 传给发送端做音频相关处理libPublisher.SmartPublisherOnFarEndPCMData(publisherHandle, pcm_buffer_, sampleRate, channel, sampleSize, is_low_latency);}}class PlayerAudioDataOutput implements NTAudioDataCallback {private int buffer_size_ = 0;private int param_info_size_ = 0;private ByteBuffer buffer_ = null;private ByteBuffer parameter_info_ = null;@Overridepublic ByteBuffer getAudioByteBuffer(int size) {if( size < 1 ) return null;if (size <= buffer_size_ && buffer_ != null )return buffer_;buffer_size_ = align(size + 256, 16);buffer_ = ByteBuffer.allocateDirect(buffer_size_);return buffer_;}@Overridepublic ByteBuffer getAudioParameterInfo(int size) {if(size < 1) return null;if ( size <= param_info_size_ &&  parameter_info_ != null )return  parameter_info_;param_info_size_ = align(size + 32, 16);parameter_info_ = ByteBuffer.allocateDirect(param_info_size_);return parameter_info_;}public void onAudioDataCallback(int ret, int audio_codec_id, int sample_size, int is_key_frame, long timestamp, int sample_rate, int channel, int parameter_info_size, long reserve)  {last_receive_audio_data_time_.set(SystemClock.elapsedRealtime());}}class AudioPlayerDataTimer implements Runnable {public static final int THRESHOLD_MS = 60*1000; public static final int INTERVAL_MS = 10*1000; public AudioPlayerDataTimer(long handle) {handle_ = handle;}@Overridepublic void run() {if (0 == handle_)return;if (handle_ != player_handle_)return;long last_update_time = last_receive_audio_data_time_.get();long cur_time = SystemClock.elapsedRealtime();if ( (last_update_time + this.THRESHOLD_MS) >  cur_time) {// 继续定时器handler_.postDelayed(new AudioPlayerDataTimer(this.handle_), this.INTERVAL_MS);}else {if (gb_source_id_!= null && gb_target_id_ != null) {if (gb28181_agent_ != null)gb28181_agent_.byeAudioBroadcast(gb_source_id_, gb_target_id_);}gb_source_id_= null;gb_target_id_ = null;stopAudioPlayer();destoryRTPReceiver();}}private long handle_;}private boolean startAudioPlay() {if (player_handle_ != 0 )return false;player_handle_ = lib_player_.SmartPlayerOpen(context_);if (player_handle_ == 0)return false;// lib_player_.SetSmartPlayerEventCallbackV2(player_handle_,new EventHandePlayerV2());lib_player_.SmartPlayerSetBuffer(player_handle_, 0);lib_player_.SmartPlayerSetReportDownloadSpeed(player_handle_, 1, 10);lib_player_.SmartPlayerClearRtpReceivers(player_handle_);lib_player_.SmartPlayerAddRtpReceiver(player_handle_, rtp_receiver_handle_);lib_player_.SmartPlayerSetSurface(player_handle_, null);// lib_player_.SmartPlayerSetRenderScaleMode(player_handle_, 1);lib_player_.SmartPlayerSetAudioOutputType(player_handle_, 1);lib_player_.SmartPlayerSetMute(player_handle_, 0);lib_player_.SmartPlayerSetAudioVolume(player_handle_, 100);lib_player_.SmartPlayerSetExternalAudioOutput(player_handle_, new PlayerExternalPCMOutput());lib_player_.SmartPlayerSetUrl(player_handle_, "rtp://xxxxxxxxxxxxxxxxxxx");if (0 != lib_player_.SmartPlayerStartPlay(player_handle_)) {lib_player_.SmartPlayerClose(player_handle_);player_handle_ = 0;Log.e(TAG,  "start audio paly failed");return false;}lib_player_.SmartPlayerSetAudioDataCallback(player_handle_, new PlayerAudioDataOutput());if (0 ==lib_player_.SmartPlayerStartPullStream(player_handle_) ) {// 启动定时器，长时间收不到音频数据，则停止播放，发送BYElast_receive_audio_data_time_.set(SystemClock.elapsedRealtime());handler_.postDelayed(new AudioPlayerDataTimer(player_handle_), AudioPlayerDataTimer.INTERVAL_MS);}return true;}private void stopAudioPlayer() {if (player_handle_ != 0 ) {lib_player_.SmartPlayerStopPullStream(player_handle_);lib_player_.SmartPlayerStopPlay(player_handle_);lib_player_.SmartPlayerClose(player_handle_);player_handle_ = 0;}}private void destoryRTPReceiver() {if (rtp_receiver_handle_ != 0) {lib_player_.UnInitRTPReceiver(rtp_receiver_handle_);lib_player_.DestoryRTPReceiverSession(rtp_receiver_handle_);lib_player_.DestoryRTPReceiver(rtp_receiver_handle_);rtp_receiver_handle_ = 0;}}@Overridepublic void ntsOnInviteAudioBroadcastResponse(String sourceID, String targetID, int statusCode, PlaySessionDescription sessionDescription) {handler_.postDelayed(new Runnable() {@Overridepublic void run() {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());int clock_rate = audio_attr.getClockRate();lib_player_.SetRTPReceiverPayloadType(rtp_receiver_handle_, audio_attr.getPayloadType(),  audio_attr.getEncodingName(), 2, clock_rate);// 如果是PCMA, 会默认填采样率8000, 通道1, 其他音频编码需要手动填入// lib_player_.SetRTPReceiverAudioSamplingRate(rtp_receiver_handle_, 8000);// lib_player_.SetRTPReceiverAudioChannels(rtp_receiver_handle_, 1);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_source_id_ = source_id_;gb_target_id_ = target_id_;}}} 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);}@Overridepublic void ntsOnByeAudioBroadcast(String sourceID, String targetID) {handler_.postDelayed(new Runnable() {@Overridepublic void run() {gb_source_id_ = null;gb_target_id_ = null;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);}@Overridepublic void ntsOnTerminateAudioBroadcast(String sourceID, String targetID) {handler_.postDelayed(new Runnable() {@Overridepublic void run() {gb_source_id_ = null;gb_target_id_ = null;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);}
}

以上是大概的流程，感兴趣的开发者，可Q我89030985，通过自测和现场的反馈，由于我们有回音消除机制，整体的体验还是非常不错的。

有开发者私信我们，如果从头开发Android平台的GB28181接入端，需要多久？我想说的是，如果是按照SPEC实现个DEMO，验证技术可行性的话不难，但是如果是产品级，确保功能完备性能优异长时间运行稳定的话，从头开发，难度还是挺大的。

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