XW-AIOT/wvp-platform
4
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

优化语音对讲

Browse Source
This commit is contained in:
648540858
2023-01-05 16:38:33 +08:00
parent 16f3b0553d
commit e14cef8062
6 changed files with 42 additions and 67 deletions
Download Patch File Download Diff File Expand all files Collapse all files

64
src/main/java/com/genersoft/iot/vmp/media/zlm/ZLMHttpHookListener.java
View File

@@ -344,70 +344,34 @@ public class ZLMHttpHookListener {
}
}else if ("broadcast".equals(param.getApp())){
// 语音对讲推流 stream需要满足格式deviceId_channelId
if (param.isRegist() && param.getStream().indexOf("_") > 0) {
if (param.getStream().indexOf("_") > 0) {
String[] streamArray = param.getStream().split("_");
if (streamArray.length == 2) {
String deviceId = streamArray[0];
String channelId = streamArray[1];
Device device = deviceService.getDevice(deviceId);
if (device != null) {
DeviceChannel deviceChannel = storager.queryChannel(deviceId, channelId);
if (deviceChannel != null) {
if (param.isRegist()) {
if (audioBroadcastManager.exit(deviceId, channelId)) {
// 直接推流
SendRtpItem sendRtpItem = redisCatchStorage.querySendRTPServer(null, null, param.getStream(), null);
if (sendRtpItem == null) {
// TODO 可能数据错误，重新开启语音通道
}else {
String is_Udp = sendRtpItem.isTcp() ? "0" : "1";
MediaServerItem mediaInfo = mediaServerService.getOne(sendRtpItem.getMediaServerId());
logger.info("rtp/{}开始向上级推流, 目标={}:{}，SSRC={}", sendRtpItem.getStreamId(), sendRtpItem.getIp(), sendRtpItem.getPort(), sendRtpItem.getSsrc());
Map<String, Object> sendParam = new HashMap<>(12);
sendParam.put("vhost","__defaultVhost__");
sendParam.put("app",sendRtpItem.getApp());
sendParam.put("stream",sendRtpItem.getStreamId());
sendParam.put("ssrc", sendRtpItem.getSsrc());
sendParam.put("src_port", sendRtpItem.getLocalPort());
sendParam.put("pt", sendRtpItem.getPt());
sendParam.put("use_ps", sendRtpItem.isUsePs() ? "1" : "0");
sendParam.put("only_audio", sendRtpItem.isOnlyAudio() ? "1" : "0");
JSONObject jsonObject;
if (sendRtpItem.isTcpActive()) {
jsonObject = zlmrtpServerFactory.startSendRtpPassive(mediaInfo, sendParam);
} else {
sendParam.put("is_udp", is_Udp);
sendParam.put("dst_url", sendRtpItem.getIp());
sendParam.put("dst_port", sendRtpItem.getPort());
jsonObject = zlmrtpServerFactory.startSendRtpStream(mediaInfo, sendParam);
}
if (jsonObject != null && jsonObject.getInteger("code") == 0) {
logger.info("[语音对讲] 自动推流成功, device: {}, channel: {}", deviceId, channelId);
}else {
logger.info("[语音对讲] 推流失败, 结果： {}", jsonObject);
}
}
}else {
// 开启语音对讲通道
try {
playService.audioBroadcastCmd(device, channelId, 60, (msg)->{
logger.info("[语音对讲] 通道建立成功, device: {}, channel: {}", deviceId, channelId);
});
} catch (InvalidArgumentException | ParseException | SipException e) {
logger.error("[命令发送失败] 语音对讲: {}", e.getMessage());
}
playService.stopAudioBroadcast(deviceId, channelId);
}
// 开启语音对讲通道
try {
playService.audioBroadcastCmd(device, channelId, 60, (msg)->{
logger.info("[语音对讲] 通道建立成功, device: {}, channel: {}", deviceId, channelId);
});
} catch (InvalidArgumentException | ParseException | SipException e) {
logger.error("[命令发送失败] 语音对讲: {}", e.getMessage());
}
}else {
logger.info("[语音对讲] 未找到通道：{}", channelId);
// 流注销
playService.stopAudioBroadcast(deviceId, channelId);
}
}else{
} else{
logger.info("[语音对讲] 未找到设备：{}", deviceId);
}
}
}
}else if ("talk".equals(param.getApp())){
// 语音对讲推流 stream需要满足格式deviceId_channelId
if (param.isRegist() && param.getStream().indexOf("_") > 0) {