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rename org simple-rtmp-server to ossrs
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241 changed files with 957 additions and 957 deletions
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@ -1,7 +1,7 @@
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/*
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The MIT License (MIT)
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Copyright (c) 2013-2015 SRS(simple-rtmp-server)
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Copyright (c) 2013-2015 SRS(ossrs)
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Permission is hereby granted, free of charge, to any person obtaining a copy of
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this software and associated documentation files (the "Software"), to deal in
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@ -373,7 +373,7 @@ int SrsRtmpConn::service_cycle()
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}
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// do token traverse before serve it.
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// @see https://github.com/simple-rtmp-server/srs/pull/239
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// @see https://github.com/ossrs/srs/pull/239
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if (true) {
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bool vhost_is_edge = _srs_config->get_vhost_is_edge(req->vhost);
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bool edge_traverse = _srs_config->get_vhost_edge_token_traverse(req->vhost);
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@ -387,7 +387,7 @@ int SrsRtmpConn::service_cycle()
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// set chunk size to larger.
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// set the chunk size before any larger response greater than 128,
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// to make OBS happy, @see https://github.com/simple-rtmp-server/srs/issues/454
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// to make OBS happy, @see https://github.com/ossrs/srs/issues/454
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int chunk_size = _srs_config->get_chunk_size(req->vhost);
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if ((ret = rtmp->set_chunk_size(chunk_size)) != ERROR_SUCCESS) {
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srs_error("set chunk_size=%d failed. ret=%d", chunk_size, ret);
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@ -439,7 +439,7 @@ int SrsRtmpConn::service_cycle()
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// logical accept and retry stream service.
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if (ret == ERROR_CONTROL_RTMP_CLOSE) {
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// TODO: FIXME: use ping message to anti-death of socket.
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// @see: https://github.com/simple-rtmp-server/srs/issues/39
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// @see: https://github.com/ossrs/srs/issues/39
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// set timeout to a larger value, for user paused.
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rtmp->set_recv_timeout(SRS_PAUSED_RECV_TIMEOUT_US);
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rtmp->set_send_timeout(SRS_PAUSED_SEND_TIMEOUT_US);
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@ -606,7 +606,7 @@ int SrsRtmpConn::playing(SrsSource* source)
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srs_verbose("consumer created success.");
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// use isolate thread to recv,
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// @see: https://github.com/simple-rtmp-server/srs/issues/217
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// @see: https://github.com/ossrs/srs/issues/217
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SrsQueueRecvThread trd(consumer, rtmp, SRS_PERF_MW_SLEEP);
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// start isolate recv thread.
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@ -678,8 +678,8 @@ int SrsRtmpConn::do_playing(SrsSource* source, SrsConsumer* consumer, SrsQueueRe
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}
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// to use isolate thread to recv, can improve about 33% performance.
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// @see: https://github.com/simple-rtmp-server/srs/issues/196
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// @see: https://github.com/simple-rtmp-server/srs/issues/217
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// @see: https://github.com/ossrs/srs/issues/196
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// @see: https://github.com/ossrs/srs/issues/217
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while (!trd->empty()) {
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SrsCommonMessage* msg = trd->pump();
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srs_verbose("pump client message to process.");
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@ -705,8 +705,8 @@ int SrsRtmpConn::do_playing(SrsSource* source, SrsConsumer* consumer, SrsQueueRe
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srs_verbose("send thread now=%"PRId64"us, wait %dms", srs_update_system_time_ms(), mw_sleep);
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// wait for message to incoming.
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// @see https://github.com/simple-rtmp-server/srs/issues/251
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// @see https://github.com/simple-rtmp-server/srs/issues/257
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// @see https://github.com/ossrs/srs/issues/251
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// @see https://github.com/ossrs/srs/issues/257
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if (realtime) {
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// for realtime, min required msgs is 0, send when got one+ msgs.
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consumer->wait(0, mw_sleep);
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@ -789,7 +789,7 @@ int SrsRtmpConn::do_playing(SrsSource* source, SrsConsumer* consumer, SrsQueueRe
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}
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// if duration specified, and exceed it, stop play live.
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// @see: https://github.com/simple-rtmp-server/srs/issues/45
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// @see: https://github.com/ossrs/srs/issues/45
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if (user_specified_duration_to_stop) {
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if (duration >= (int64_t)req->duration) {
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ret = ERROR_RTMP_DURATION_EXCEED;
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@ -825,7 +825,7 @@ int SrsRtmpConn::publishing(SrsSource* source)
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bool vhost_is_edge = _srs_config->get_vhost_is_edge(req->vhost);
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if ((ret = acquire_publish(source, vhost_is_edge)) == ERROR_SUCCESS) {
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// use isolate thread to recv,
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// @see: https://github.com/simple-rtmp-server/srs/issues/237
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// @see: https://github.com/ossrs/srs/issues/237
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SrsPublishRecvThread trd(rtmp, req,
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st_netfd_fileno(stfd), 0, this, source, true, vhost_is_edge);
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@ -839,7 +839,7 @@ int SrsRtmpConn::publishing(SrsSource* source)
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// whatever the acquire publish, always release publish.
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// when the acquire error in the midlle-way, the publish state changed,
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// but failed, so we must cleanup it.
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// @see https://github.com/simple-rtmp-server/srs/issues/474
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// @see https://github.com/ossrs/srs/issues/474
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// @remark when stream is busy, should never release it.
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if (ret != ERROR_SYSTEM_STREAM_BUSY) {
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release_publish(source, vhost_is_edge);
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@ -896,7 +896,7 @@ int SrsRtmpConn::do_publishing(SrsSource* source, SrsPublishRecvThread* trd)
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// cond wait for timeout.
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if (nb_msgs == 0) {
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// when not got msgs, wait for a larger timeout.
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// @see https://github.com/simple-rtmp-server/srs/issues/441
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// @see https://github.com/ossrs/srs/issues/441
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trd->wait(publish_1stpkt_timeout);
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} else {
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trd->wait(publish_normal_timeout);
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@ -1107,7 +1107,7 @@ int SrsRtmpConn::process_play_control_msg(SrsConsumer* consumer, SrsCommonMessag
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SrsAutoFree(SrsPacket, pkt);
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// for jwplayer/flowplayer, which send close as pause message.
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// @see https://github.com/simple-rtmp-server/srs/issues/6
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// @see https://github.com/ossrs/srs/issues/6
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SrsCloseStreamPacket* close = dynamic_cast<SrsCloseStreamPacket*>(pkt);
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if (close) {
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ret = ERROR_CONTROL_RTMP_CLOSE;
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@ -1117,7 +1117,7 @@ int SrsRtmpConn::process_play_control_msg(SrsConsumer* consumer, SrsCommonMessag
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// call msg,
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// support response null first,
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// @see https://github.com/simple-rtmp-server/srs/issues/106
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// @see https://github.com/ossrs/srs/issues/106
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// TODO: FIXME: response in right way, or forward in edge mode.
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SrsCallPacket* call = dynamic_cast<SrsCallPacket*>(pkt);
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if (call) {
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@ -1337,7 +1337,7 @@ int SrsRtmpConn::http_hooks_on_connect()
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// the http hooks will cause context switch,
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// so we must copy all hooks for the on_connect may freed.
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// @see https://github.com/simple-rtmp-server/srs/issues/475
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// @see https://github.com/ossrs/srs/issues/475
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vector<string> hooks;
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if (true) {
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@ -1372,7 +1372,7 @@ void SrsRtmpConn::http_hooks_on_close()
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// the http hooks will cause context switch,
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// so we must copy all hooks for the on_connect may freed.
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// @see https://github.com/simple-rtmp-server/srs/issues/475
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// @see https://github.com/ossrs/srs/issues/475
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vector<string> hooks;
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if (true) {
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@ -1404,7 +1404,7 @@ int SrsRtmpConn::http_hooks_on_publish()
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// the http hooks will cause context switch,
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// so we must copy all hooks for the on_connect may freed.
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// @see https://github.com/simple-rtmp-server/srs/issues/475
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// @see https://github.com/ossrs/srs/issues/475
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vector<string> hooks;
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if (true) {
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@ -1439,7 +1439,7 @@ void SrsRtmpConn::http_hooks_on_unpublish()
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// the http hooks will cause context switch,
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// so we must copy all hooks for the on_connect may freed.
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// @see https://github.com/simple-rtmp-server/srs/issues/475
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// @see https://github.com/ossrs/srs/issues/475
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vector<string> hooks;
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if (true) {
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@ -1471,7 +1471,7 @@ int SrsRtmpConn::http_hooks_on_play()
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// the http hooks will cause context switch,
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// so we must copy all hooks for the on_connect may freed.
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// @see https://github.com/simple-rtmp-server/srs/issues/475
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// @see https://github.com/ossrs/srs/issues/475
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vector<string> hooks;
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if (true) {
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@ -1506,7 +1506,7 @@ void SrsRtmpConn::http_hooks_on_stop()
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// the http hooks will cause context switch,
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// so we must copy all hooks for the on_connect may freed.
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// @see https://github.com/simple-rtmp-server/srs/issues/475
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// @see https://github.com/ossrs/srs/issues/475
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vector<string> hooks;
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if (true) {
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