openRtpServer接口新增only_audio参数,优化语音对讲场景

This commit is contained in:
xiongziliang
2023-02-17 22:43:45 +08:00
parent 5cdaf982f3
commit 8f0ba6988b
10 changed files with 80 additions and 17 deletions

View File

@@ -391,7 +391,7 @@ Value makeMediaSourceJson(MediaSource &media){
}
#if defined(ENABLE_RTPPROXY)
uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc) {
uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio) {
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
if (s_rtpServerMap.find(stream_id) != s_rtpServerMap.end()) {
//为了防止RtpProcess所有权限混乱的问题不允许重复添加相同的stream_id
@@ -399,7 +399,7 @@ uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mod
}
RtpServer::Ptr server = std::make_shared<RtpServer>();
server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc);
server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_audio);
server->setOnDetach([stream_id]() {
//设置rtp超时移除事件
lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
@@ -1140,7 +1140,7 @@ void installWebApi() {
tcp_mode = 1;
}
auto port = openRtpServer(allArgs["port"], stream_id, tcp_mode, "::", allArgs["re_use_port"].as<bool>(),
allArgs["ssrc"].as<uint32_t>());
allArgs["ssrc"].as<uint32_t>(), allArgs["only_audio"].as<bool>());
if (port == 0) {
throw InvalidArgsException("该stream_id已存在");
}