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。

Android GB28181设备接入端语音广播和语音对讲技术实现探究

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;

/*
* 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);
}

Android GB28181设备接入端语音广播和语音对讲技术实现探究

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;

/*
*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;

/*
*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);
}

Android GB28181设备接入端语音广播和语音对讲技术实现探究

  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

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
public native int SmartPlayerSetExternalAudioOutput(long;

/**
* 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
public native int SmartPlayerSetAudioDataCallback(long;


/**
* 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
public native int SmartPlayerSetBuffer(long handle, int;

/**
* 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
public native int SmartPlayerSetMute(long handle, int;

/**
* 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
public native int SmartPlayerSetAudioVolume(long handle, int;


/**
* 清除所有 rtp receivers
*
* @param handle: return value from SmartPlayerOpen()
*
* @return
public native int SmartPlayerClearRtpReceivers(long;


/**
* 增加 rtp receiver
*
* @param handle: return value from SmartPlayerOpen()
*
* @param rtp_receiver_handle: return value from CreateRTPReceiver()
*
* @return
public native int SmartPlayerAddRtpReceiver(long handle, long;


/**
* 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
public native int SmartPlayerSetUrl(long;


/**
* Start playback stream(开始播放)
*
* @param handle: return value from SmartPlayerOpen()
*
* @return
public native int SmartPlayerStartPlay(long;

/**
* Stop playback stream(停止播放)
*
* @param handle: return value from SmartPlayerOpen()
*
* @return
public native int SmartPlayerStopPlay(long;


/**
* 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
public native int SmartPlayerStartPullStream(long;

/**
* Stop pull stream(停止拉流)
*
* @param handle: return value from SmartPlayerOpen()
*
* @return
public native int SmartPlayerStopPullStream(long;

/**
* 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
public native int SmartPlayerClose(long;


/*++++++++++++++++++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;


/**
*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
public native int SetRTPReceiverTransportProtocol(long rtp_receiver_handle, int;


/**
*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
public native int SetRTPReceiverIPAddressType(long rtp_receiver_handle, int;


/**
*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
public native int SetRTPReceiverLocalPort(long rtp_receiver_handle, int;


/**
*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
public native int SetRTPReceiverSSRC(long;


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


/**
*获取 RTP Receiver RTP Socket本地端口
*
* @param rtp_receiver_handle, CreateRTPReceiver
*
* @return
public native int GetRTPReceiverLocalPort(long;


/**
*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
public native int SetRTPReceiverPayloadType(long rtp_receiver_handle, int payload_type, String encoding_name, int media_type, int;


/**
*设置 RTP Receiver 音频采样率
*
* @param rtp_receiver_handle, CreateRTPReceiver
* @param sampling_rate, 音频采样率
*
* @return
public native int SetRTPReceiverAudioSamplingRate(long rtp_receiver_handle, int;

/**
*设置 RTP Receiver 音频通道数
*
* @param rtp_receiver_handle, CreateRTPReceiver
* @param channels, 音频通道数
*
* @return
public native int SetRTPReceiverAudioChannels(long rtp_receiver_handle, int;


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

/**
*初始化 RTP Receiver
*
* @param rtp_receiver_handle, CreateRTPReceiver
*
* @return
public native int InitRTPReceiver(long;

/**
*UnInit RTP Receiver
*
* @param rtp_receiver_handle, CreateRTPReceiver
*
* @return
public native int UnInitRTPReceiver(long;


/**
*Destory RTP Receiver Session
*
* @param rtp_receiver_handle, CreateRTPReceiver
*
* @return
public native int DestoryRTPReceiverSession(long;


/**
*Destory RTP Receiver
*
* @param rtp_receiver_handle, CreateRTPReceiver
*
* @return
public native int DestoryRTPReceiver(long;


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

Android GB28181设备接入端语音广播和语音对讲技术实现探究

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);

@Override
public void ntsOnNotifyBroadcastCommand(String fromUserName, String fromUserNameAtDomain, String sn, String sourceID, String targetID) {
handler_.postDelayed(new Runnable() {
@Override
public 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);
}

@Override
public void ntsOnAudioBroadcast(String commandFromUserName, String commandFromUserNameAtDomain, String sourceID, String targetID) {
handler_.postDelayed(new Runnable() {
@Override
public void run() {
stopAudioPlayer();
destoryRTPReceiver();

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

boolean is_tcp = true; // 默认用TCP
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();
}

} 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);
}

@Override
public void ntsOnInviteAudioBroadcastException(String sourceID, String targetID, String errorInfo) {
handler_.postDelayed(new Runnable() {
@Override
public 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);
}

@Override
public void ntsOnInviteAudioBroadcastTimeout(String sourceID, String targetID) {
handler_.postDelayed(new Runnable() {
@Override
public 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;

@Override
public ByteBuffer getPcmByteBuffer(int {
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 {

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;

@Override
public ByteBuffer getAudioByteBuffer(int {
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_;
}

@Override
public ByteBuffer getAudioParameterInfo(int {
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 {
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;
}

@Override
public 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BYE
last_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;
}
}

@Override
public void ntsOnInviteAudioBroadcastResponse(String sourceID, String targetID, int {
handler_.postDelayed(new Runnable() {
@Override
public 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 {
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() {
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);
}

@Override
public void ntsOnTerminateAudioBroadcast(String sourceID, String targetID) {
handler_.postDelayed(new Runnable() {
@Override
public 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);
}
}

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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