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Fixed issue that set use_ps in startSendRtp api does not take effect (#3353)

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修复使用startSendRtp接口转发ps流，设置use_ps为1后，还发送es流的问题

---------

Co-authored-by: huangcaichun <cchuang@secusoft.cc>
This commit is contained in:
huangcaichun 2024-03-10 16:17:29 +08:00 committed by GitHub
parent fe61b572e0
commit 78a6f041a8
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GPG Key ID: B5690EEEBB952194
1 changed files with 8 additions and 7 deletions
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15
server/WebApi.cpp
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@ -1299,9 +1299,10 @@ void installWebApi() {
if (!src) {
throw ApiRetException("can not find the source stream", API::NotFound);
}
auto type = allArgs["type"].as<int>();
if (!allArgs["use_ps"].empty()) {
// 兼容之前的use_ps参数
allArgs["type"] = allArgs["use_ps"].as<int>();
type = allArgs["use_ps"].as<int>();
}
MediaSourceEvent::SendRtpArgs args;
args.passive = false;
@ -1312,11 +1313,11 @@ void installWebApi() {
args.is_udp = allArgs["is_udp"];
args.src_port = allArgs["src_port"];
args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
args.type = (MediaSourceEvent::SendRtpArgs::Type)(allArgs["type"].as<int>());
args.type = (MediaSourceEvent::SendRtpArgs::Type)type;
args.only_audio = allArgs["only_audio"].as<bool>();
args.udp_rtcp_timeout = allArgs["udp_rtcp_timeout"];
args.recv_stream_id = allArgs["recv_stream_id"];
TraceL << "startSendRtp, pt " << int(args.pt) << " rtp type " << args.type << " audio " << args.only_audio;
TraceL << "startSendRtp, pt " << int(args.pt) << " rtp type " << type << " audio " << args.only_audio;
src->getOwnerPoller()->async([=]() mutable {
src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {
@ -1338,10 +1339,10 @@ void installWebApi() {
if (!src) {
throw ApiRetException("can not find the source stream", API::NotFound);
}
auto type = allArgs["type"].as<int>();
if (!allArgs["use_ps"].empty()) {
// 兼容之前的use_ps参数
allArgs["type"] = allArgs["use_ps"].as<int>();
type = allArgs["use_ps"].as<int>();
}
MediaSourceEvent::SendRtpArgs args;
@ -1350,12 +1351,12 @@ void installWebApi() {
args.is_udp = false;
args.src_port = allArgs["src_port"];
args.pt = allArgs["pt"].empty() ? 96 : allArgs["pt"].as<int>();
args.type = (MediaSourceEvent::SendRtpArgs::Type)(allArgs["type"].as<int>());
args.type = (MediaSourceEvent::SendRtpArgs::Type)type;
args.only_audio = allArgs["only_audio"].as<bool>();
args.recv_stream_id = allArgs["recv_stream_id"];
//tcp被动服务器等待链接超时时间
args.tcp_passive_close_delay_ms = allArgs["close_delay_ms"];
TraceL << "startSendRtpPassive, pt " << int(args.pt) << " rtp type " << args.type << " audio " << args.only_audio;
TraceL << "startSendRtpPassive, pt " << int(args.pt) << " rtp type " << type << " audio " << args.only_audio;
src->getOwnerPoller()->async([=]() mutable {
src->startSendRtp(args, [val, headerOut, invoker](uint16_t local_port, const SockException &ex) mutable {