diff --git a/trunk/research/arm/jmp_sp.cpp b/trunk/research/arm/jmp_sp.cpp index ddb3f56f2f..9d26c68a24 100644 --- a/trunk/research/arm/jmp_sp.cpp +++ b/trunk/research/arm/jmp_sp.cpp @@ -1,5 +1,4 @@ /* - # see: https://github.com/ossrs/srs/issues/190 # see: https://github.com/ossrs/srs/wiki/v1_CN_SrsLinuxArm g++ -g -O0 -o jmp_sp jmp_sp.cpp arm-linux-gnueabi-g++ -g -o jmp_sp jmp_sp.cpp -static diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp index 40fbc7959f..d1b65ace6e 100644 --- a/trunk/src/app/srs_app_config.hpp +++ b/trunk/src/app/srs_app_config.hpp @@ -746,7 +746,6 @@ class SrsConfig // all clients connected to edge must be tranverse to origin to verify. virtual bool get_vhost_edge_token_traverse(std::string vhost); // Get the transformed vhost for edge, - // @see https://github.com/ossrs/srs/issues/372 virtual std::string get_vhost_edge_transform_vhost(std::string vhost); // Whether enable the origin cluster. // @see https://github.com/ossrs/srs/wiki/v3_EN_OriginCluster @@ -921,7 +920,6 @@ class SrsConfig // Get the hls hls_on_error config. // The ignore will ignore error and disable hls. // The disconnect will disconnect publish connection. - // @see https://github.com/ossrs/srs/issues/264 virtual std::string get_hls_on_error(std::string vhost); // Get the HLS default audio codec. virtual std::string get_hls_acodec(std::string vhost); diff --git a/trunk/src/app/srs_app_edge.cpp b/trunk/src/app/srs_app_edge.cpp index 72c110d1a5..24ff586326 100644 --- a/trunk/src/app/srs_app_edge.cpp +++ b/trunk/src/app/srs_app_edge.cpp @@ -66,7 +66,6 @@ srs_error_t SrsEdgeRtmpUpstream::connect(SrsRequest* r, SrsLbRoundRobin* lb) if (true) { SrsConfDirective* conf = _srs_config->get_vhost_edge_origin(req->vhost); - // @see https://github.com/ossrs/srs/issues/79 // when origin is error, for instance, server is shutdown, // then user remove the vhost then reload, the conf is empty. if (!conf) { @@ -93,7 +92,6 @@ srs_error_t SrsEdgeRtmpUpstream::connect(SrsRequest* r, SrsLbRoundRobin* lb) selected_port = port; // support vhost tranform for edge, - // @see https://github.com/ossrs/srs/issues/372 std::string vhost = _srs_config->get_vhost_edge_transform_vhost(req->vhost); vhost = srs_string_replace(vhost, "[vhost]", req->vhost); @@ -478,7 +476,6 @@ srs_error_t SrsEdgeForwarder::start() srs_parse_hostport(server, server, port); // support vhost tranform for edge, - // @see https://github.com/ossrs/srs/issues/372 std::string vhost = _srs_config->get_vhost_edge_transform_vhost(req->vhost); vhost = srs_string_replace(vhost, "[vhost]", req->vhost); @@ -759,7 +756,6 @@ srs_error_t SrsPublishEdge::on_client_publish() return srs_error_new(ERROR_RTMP_EDGE_PUBLISH_STATE, "invalid state"); } - // @see https://github.com/ossrs/srs/issues/180 // to avoid multiple publish the same stream on the same edge, // directly enter the publish stage. if (true) { @@ -771,7 +767,6 @@ srs_error_t SrsPublishEdge::on_client_publish() // start to forward stream to origin. err = forwarder->start(); - // @see https://github.com/ossrs/srs/issues/180 // when failed, revert to init if (err != srs_success) { SrsEdgeState pstate = state; diff --git a/trunk/src/app/srs_app_forward.hpp b/trunk/src/app/srs_app_forward.hpp index 51a0195fb7..1c5e257138 100644 --- a/trunk/src/app/srs_app_forward.hpp +++ b/trunk/src/app/srs_app_forward.hpp @@ -40,7 +40,6 @@ class SrsForwarder : public ISrsCoroutineHandler SrsRtmpJitter* jitter; SrsMessageQueue* queue; // Cache the sequence header for retry when slave is failed. - // @see https://github.com/ossrs/srs/issues/150 SrsSharedPtrMessage* sh_audio; SrsSharedPtrMessage* sh_video; public: diff --git a/trunk/src/app/srs_app_hls.hpp b/trunk/src/app/srs_app_hls.hpp index 972771742e..e8afe4db4c 100644 --- a/trunk/src/app/srs_app_hls.hpp +++ b/trunk/src/app/srs_app_hls.hpp @@ -153,7 +153,6 @@ class SrsHlsMuxer // The current writing segment. SrsHlsSegment* current; // The ts context, to keep cc continous between ts. - // @see https://github.com/ossrs/srs/issues/375 SrsTsContext* context; public: SrsHlsMuxer(); @@ -187,7 +186,6 @@ class SrsHlsMuxer virtual bool wait_keyframe(); // Whether segment absolutely overflow, for pure audio to reap segment, // that is whether the current segment duration>=2*(the segment in config) - // @see https://github.com/ossrs/srs/issues/151#issuecomment-71155184 virtual bool is_segment_absolutely_overflow(); public: // Whether current hls muxer is pure audio mode. diff --git a/trunk/src/app/srs_app_http_hooks.cpp b/trunk/src/app/srs_app_http_hooks.cpp index aec4bd85aa..180bca26c2 100644 --- a/trunk/src/app/srs_app_http_hooks.cpp +++ b/trunk/src/app/srs_app_http_hooks.cpp @@ -509,7 +509,6 @@ srs_error_t SrsHttpHooks::do_post(SrsHttpClient* hc, std::string url, std::strin } // ensure the http status is ok. - // https://github.com/ossrs/srs/issues/158 if (code != SRS_CONSTS_HTTP_OK && code != SRS_CONSTS_HTTP_Created) { return srs_error_new(ERROR_HTTP_STATUS_INVALID, "http: status %d", code); } diff --git a/trunk/src/app/srs_app_http_stream.cpp b/trunk/src/app/srs_app_http_stream.cpp index a569d15dd1..becab70718 100755 --- a/trunk/src/app/srs_app_http_stream.cpp +++ b/trunk/src/app/srs_app_http_stream.cpp @@ -928,7 +928,6 @@ srs_error_t SrsHttpStreamServer::http_mount(SrsLiveSource* s, SrsRequest* r) // mount the http flv stream. // we must register the handler, then start the thread, // for the thread will cause thread switch context. - // @see https://github.com/ossrs/srs/issues/404 if ((err = mux.handle(mount, entry->stream)) != srs_success) { return srs_error_wrap(err, "http: mount flv stream for vhost=%s failed", sid.c_str()); } diff --git a/trunk/src/app/srs_app_mpegts_udp.cpp b/trunk/src/app/srs_app_mpegts_udp.cpp index 315d1ef33a..1d481ec690 100644 --- a/trunk/src/app/srs_app_mpegts_udp.cpp +++ b/trunk/src/app/srs_app_mpegts_udp.cpp @@ -449,7 +449,6 @@ srs_error_t SrsMpegtsOverUdp::write_h264_ipb_frame(char* frame, int frame_size, srs_error_t err = srs_success; // when sps or pps not sent, ignore the packet. - // @see https://github.com/ossrs/srs/issues/203 if (!h264_sps_pps_sent) { return srs_error_new(ERROR_H264_DROP_BEFORE_SPS_PPS, "drop sps/pps"); } diff --git a/trunk/src/app/srs_app_recv_thread.cpp b/trunk/src/app/srs_app_recv_thread.cpp index f3af47edc4..2ded41a828 100644 --- a/trunk/src/app/srs_app_recv_thread.cpp +++ b/trunk/src/app/srs_app_recv_thread.cpp @@ -93,8 +93,6 @@ srs_error_t SrsRecvThread::cycle() // the multiple messages writev improve performance large, // but the timeout recv will cause 33% sys call performance, // to use isolate thread to recv, can improve about 33% performance. - // @see https://github.com/ossrs/srs/issues/194 - // @see: https://github.com/ossrs/srs/issues/217 rtmp->set_recv_timeout(SRS_UTIME_NO_TIMEOUT); pumper->on_start(); @@ -279,7 +277,6 @@ SrsPublishRecvThread::SrsPublishRecvThread(SrsRtmpServer* rtmp_sdk, SrsRequest* mr_fd = mr_sock_fd; // the mr settings, - // @see https://github.com/ossrs/srs/issues/241 mr = _srs_config->get_mr_enabled(req->vhost); mr_sleep = _srs_config->get_mr_sleep(req->vhost); @@ -405,7 +402,6 @@ void SrsPublishRecvThread::interrupt(srs_error_t err) recv_error = srs_error_copy(err); // when recv thread error, signal the conn thread to process it. - // @see https://github.com/ossrs/srs/issues/244 srs_cond_signal(error); } @@ -420,7 +416,6 @@ void SrsPublishRecvThread::on_start() set_socket_buffer(mr_sleep); // disable the merge read - // @see https://github.com/ossrs/srs/issues/241 rtmp->set_merge_read(true, this); } #endif @@ -432,13 +427,11 @@ void SrsPublishRecvThread::on_stop() // for we donot set to false yet. // when thread stop, signal the conn thread which wait. - // @see https://github.com/ossrs/srs/issues/244 srs_cond_signal(error); #ifdef SRS_PERF_MERGED_READ if (mr) { // disable the merge read - // @see https://github.com/ossrs/srs/issues/241 rtmp->set_merge_read(false, NULL); } #endif @@ -459,7 +452,6 @@ void SrsPublishRecvThread::on_read(ssize_t nread) * to improve read performance, merge some packets then read, * when it on and read small bytes, we sleep to wait more data., * that is, we merge some data to read together. - * @see https://github.com/ossrs/srs/issues/241 */ if (nread < SRS_MR_SMALL_BYTES) { srs_usleep(mr_sleep); @@ -476,7 +468,6 @@ srs_error_t SrsPublishRecvThread::on_reload_vhost_publish(string vhost) } // the mr settings, - // @see https://github.com/ossrs/srs/issues/241 bool mr_enabled = _srs_config->get_mr_enabled(req->vhost); srs_utime_t sleep_v = _srs_config->get_mr_sleep(req->vhost); @@ -489,13 +480,11 @@ srs_error_t SrsPublishRecvThread::on_reload_vhost_publish(string vhost) // mr enabled=>disabled if (mr && !mr_enabled) { // disable the merge read - // @see https://github.com/ossrs/srs/issues/241 rtmp->set_merge_read(false, NULL); } // mr disabled=>enabled if (!mr && mr_enabled) { // enable the merge read - // @see https://github.com/ossrs/srs/issues/241 rtmp->set_merge_read(true, this); } #endif diff --git a/trunk/src/app/srs_app_rtc_source.hpp b/trunk/src/app/srs_app_rtc_source.hpp index b8b8c1ebde..cfe694a41a 100644 --- a/trunk/src/app/srs_app_rtc_source.hpp +++ b/trunk/src/app/srs_app_rtc_source.hpp @@ -75,7 +75,6 @@ class SrsRtcConsumer // when source id changed, notice all consumers bool should_update_source_id; // The cond wait for mw. - // @see https://github.com/ossrs/srs/issues/251 srs_cond_t mw_wait; bool mw_waiting; int mw_min_msgs; diff --git a/trunk/src/app/srs_app_rtmp_conn.cpp b/trunk/src/app/srs_app_rtmp_conn.cpp index 246dcfd62f..2fc854c841 100644 --- a/trunk/src/app/srs_app_rtmp_conn.cpp +++ b/trunk/src/app/srs_app_rtmp_conn.cpp @@ -429,7 +429,6 @@ srs_error_t SrsRtmpConn::service_cycle() // logical accept and retry stream service. if (srs_error_code(err) == ERROR_CONTROL_RTMP_CLOSE) { // TODO: FIXME: use ping message to anti-death of socket. - // @see: https://github.com/ossrs/srs/issues/39 // set timeout to a larger value, for user paused. rtmp->set_recv_timeout(SRS_PAUSED_RECV_TIMEOUT); rtmp->set_send_timeout(SRS_PAUSED_SEND_TIMEOUT); @@ -670,7 +669,6 @@ srs_error_t SrsRtmpConn::playing(SrsLiveSource* source) } // Use receiving thread to receive packets from peer. - // @see: https://github.com/ossrs/srs/issues/217 SrsQueueRecvThread trd(consumer, rtmp, SRS_PERF_MW_SLEEP, _srs_context->get_id()); if ((err = trd.start()) != srs_success) { @@ -731,8 +729,6 @@ srs_error_t SrsRtmpConn::do_playing(SrsLiveSource* source, SrsLiveConsumer* cons pprint->elapse(); // to use isolate thread to recv, can improve about 33% performance. - // @see: https://github.com/ossrs/srs/issues/196 - // @see: https://github.com/ossrs/srs/issues/217 while (!rtrd->empty()) { SrsCommonMessage* msg = rtrd->pump(); if ((err = process_play_control_msg(consumer, msg)) != srs_success) { @@ -1114,7 +1110,6 @@ srs_error_t SrsRtmpConn::process_play_control_msg(SrsLiveConsumer* consumer, Srs SrsAutoFree(SrsPacket, pkt); // for jwplayer/flowplayer, which send close as pause message. - // @see https://github.com/ossrs/srs/issues/6 SrsCloseStreamPacket* close = dynamic_cast(pkt); if (close) { return srs_error_new(ERROR_CONTROL_RTMP_CLOSE, "rtmp: close stream"); @@ -1122,7 +1117,6 @@ srs_error_t SrsRtmpConn::process_play_control_msg(SrsLiveConsumer* consumer, Srs // call msg, // support response null first, - // @see https://github.com/ossrs/srs/issues/106 // TODO: FIXME: response in right way, or forward in edge mode. SrsCallPacket* call = dynamic_cast(pkt); if (call) { diff --git a/trunk/src/app/srs_app_rtmp_conn.hpp b/trunk/src/app/srs_app_rtmp_conn.hpp index f88e6f2fd2..bc4fada942 100644 --- a/trunk/src/app/srs_app_rtmp_conn.hpp +++ b/trunk/src/app/srs_app_rtmp_conn.hpp @@ -82,7 +82,6 @@ class SrsRtmpConn : public ISrsStartableConneciton, public ISrsReloadHandler ISrsWakable* wakable; // The elapsed duration in srs_utime_t // For live play duration, for instance, rtmpdump to record. - // @see https://github.com/ossrs/srs/issues/47 srs_utime_t duration; // The MR(merged-write) sleep time in srs_utime_t. srs_utime_t mw_sleep; diff --git a/trunk/src/app/srs_app_security.hpp b/trunk/src/app/srs_app_security.hpp index 5d004506eb..389368a7ce 100644 --- a/trunk/src/app/srs_app_security.hpp +++ b/trunk/src/app/srs_app_security.hpp @@ -16,7 +16,6 @@ class SrsConfDirective; // The security apply on vhost. -// @see https://github.com/ossrs/srs/issues/211 class SrsSecurity { public: diff --git a/trunk/src/kernel/srs_kernel_mp3.cpp b/trunk/src/kernel/srs_kernel_mp3.cpp index 5b3909b54d..6e92a49a06 100644 --- a/trunk/src/kernel/srs_kernel_mp3.cpp +++ b/trunk/src/kernel/srs_kernel_mp3.cpp @@ -6,7 +6,6 @@ #include -// for srs-librtmp, @see https://github.com/ossrs/srs/issues/213 #ifndef _WIN32 #include #endif diff --git a/trunk/src/kernel/srs_kernel_utility.cpp b/trunk/src/kernel/srs_kernel_utility.cpp index 645c333456..7e783382b9 100644 --- a/trunk/src/kernel/srs_kernel_utility.cpp +++ b/trunk/src/kernel/srs_kernel_utility.cpp @@ -6,7 +6,6 @@ #include -// for srs-librtmp, @see https://github.com/ossrs/srs/issues/213 #ifndef _WIN32 #include #include @@ -120,7 +119,6 @@ srs_utime_t srs_update_system_time() return -1; } - // @see: https://github.com/ossrs/srs/issues/35 // we must convert the tv_sec/tv_usec to int64_t. int64_t now_us = ((int64_t)now.tv_sec) * 1000 * 1000 + (int64_t)now.tv_usec; @@ -140,7 +138,6 @@ srs_utime_t srs_update_system_time() diff = srs_max(0, diff); if (diff < 0 || diff > 1000 * SYS_TIME_RESOLUTION_US) { srs_warn("clock jump, history=%" PRId64 "us, now=%" PRId64 "us, diff=%" PRId64 "us", _srs_system_time_us_cache, now_us, diff); - // @see: https://github.com/ossrs/srs/issues/109 _srs_system_time_startup_time += diff; } @@ -580,7 +577,6 @@ int srs_do_create_dir_recursively(string dir) } // create curren dir. - // for srs-librtmp, @see https://github.com/ossrs/srs/issues/213 #ifdef _WIN32 if (::_mkdir(dir.c_str()) < 0) { #else diff --git a/trunk/src/protocol/srs_protocol_stream.cpp b/trunk/src/protocol/srs_protocol_stream.cpp index 59dd03ce0a..453c586924 100755 --- a/trunk/src/protocol/srs_protocol_stream.cpp +++ b/trunk/src/protocol/srs_protocol_stream.cpp @@ -167,7 +167,6 @@ srs_error_t SrsFastStream::grow(ISrsReader* reader, int required_size) * to improve read performance, merge some packets then read, * when it on and read small bytes, we sleep to wait more data., * that is, we merge some data to read together. - * @see https://github.com/ossrs/srs/issues/241 */ if (merged_read && _handler) { _handler->on_read(nread); diff --git a/trunk/src/protocol/srs_protocol_utility.cpp b/trunk/src/protocol/srs_protocol_utility.cpp index e0b999a735..690e2414e4 100644 --- a/trunk/src/protocol/srs_protocol_utility.cpp +++ b/trunk/src/protocol/srs_protocol_utility.cpp @@ -6,7 +6,6 @@ #include -// for srs-librtmp, @see https://github.com/ossrs/srs/issues/213 #ifndef _WIN32 #include #endif @@ -345,7 +344,6 @@ srs_error_t srs_write_large_iovs(ISrsProtocolReadWriter* skt, iovec* iovs, int s srs_error_t err = srs_success; // the limits of writev iovs. - // for srs-librtmp, @see https://github.com/ossrs/srs/issues/213 #ifndef _WIN32 // for linux, generally it's 1024. static int limits = (int)sysconf(_SC_IOV_MAX); diff --git a/trunk/src/protocol/srs_protocol_utility.hpp b/trunk/src/protocol/srs_protocol_utility.hpp index d3a4f9df9d..7b8c34bd94 100644 --- a/trunk/src/protocol/srs_protocol_utility.hpp +++ b/trunk/src/protocol/srs_protocol_utility.hpp @@ -9,7 +9,6 @@ #include -// for srs-librtmp, @see https://github.com/ossrs/srs/issues/213 #ifndef _WIN32 #include #endif diff --git a/trunk/src/protocol/srs_raw_avc.cpp b/trunk/src/protocol/srs_raw_avc.cpp index 86ad861f07..2e1c513793 100644 --- a/trunk/src/protocol/srs_raw_avc.cpp +++ b/trunk/src/protocol/srs_raw_avc.cpp @@ -281,7 +281,6 @@ srs_error_t SrsRawAacStream::adts_demux(SrsBuffer* stream, char** pframe, int* p // decode the ADTS. // @see ISO_IEC_13818-7-AAC-2004.pdf, page 26 // 6.2 Audio Data Transport Stream, ADTS - // @see https://github.com/ossrs/srs/issues/212#issuecomment-64145885 // byte_alignment() // adts_fixed_header: @@ -342,7 +341,7 @@ srs_error_t SrsRawAacStream::adts_demux(SrsBuffer* stream, char** pframe, int* p int8_t channel_configuration = (sfiv >> 6) & 0x07; /*int8_t original = (sfiv >> 5) & 0x01;*/ /*int8_t home = (sfiv >> 4) & 0x01;*/ - //int8_t Emphasis; @remark, Emphasis is removed, @see https://github.com/ossrs/srs/issues/212#issuecomment-64154736 + //int8_t Emphasis; @remark, Emphasis is removed // 4bits left. // adts_variable_header(), 1.A.2.2.2 Variable Header of ADTS // copyright_identification_bit 1 bslbf diff --git a/trunk/src/protocol/srs_rtmp_handshake.cpp b/trunk/src/protocol/srs_rtmp_handshake.cpp index bbbce8eae5..1bdebcaabd 100644 --- a/trunk/src/protocol/srs_rtmp_handshake.cpp +++ b/trunk/src/protocol/srs_rtmp_handshake.cpp @@ -571,7 +571,6 @@ namespace srs_internal } // directly generate the public key. - // @see: https://github.com/ossrs/srs/issues/148 int pkey_size = 128; if ((err = dh.copy_shared_key(c1->get_key(), 128, key.key, pkey_size)) != srs_success) { return srs_error_wrap(err, "copy shared key"); diff --git a/trunk/src/protocol/srs_rtmp_stack.cpp b/trunk/src/protocol/srs_rtmp_stack.cpp index 45a467233f..f4b44a0721 100644 --- a/trunk/src/protocol/srs_rtmp_stack.cpp +++ b/trunk/src/protocol/srs_rtmp_stack.cpp @@ -1036,7 +1036,6 @@ srs_error_t SrsProtocol::read_message_header(SrsChunkStream* chunk, char fmt) // 0x04 where: message_type=4(protocol control user-control message) // 0x00 0x06 where: event Ping(0x06) // 0x00 0x00 0x0d 0x0f where: event data 4bytes ping timestamp. - // @see: https://github.com/ossrs/srs/issues/98 if (fmt == RTMP_FMT_TYPE1) { srs_warn("fresh chunk starts with fmt=1"); } else { @@ -1175,7 +1174,6 @@ srs_error_t SrsProtocol::read_message_header(SrsChunkStream* chunk, char fmt) pp[0] = *p++; // always use 31bits timestamp, for some server may use 32bits extended timestamp. - // @see https://github.com/ossrs/srs/issues/111 timestamp &= 0x7fffffff; /** diff --git a/trunk/src/protocol/srs_rtmp_stack.hpp b/trunk/src/protocol/srs_rtmp_stack.hpp index 17eb0ef6e0..83b623e197 100644 --- a/trunk/src/protocol/srs_rtmp_stack.hpp +++ b/trunk/src/protocol/srs_rtmp_stack.hpp @@ -13,7 +13,6 @@ #include #include -// For srs-librtmp, @see https://github.com/ossrs/srs/issues/213 #ifndef _WIN32 #include #endif @@ -161,7 +160,6 @@ class SrsProtocol std::map chunk_streams; // Cache some frequently used chunk header. // cs_cache, the chunk stream cache. - // @see https://github.com/ossrs/srs/issues/249 SrsChunkStream** cs_cache; // The bytes buffer cache, recv from skt, provide services for stream. SrsFastStream* in_buffer; @@ -179,7 +177,6 @@ class SrsProtocol bool show_debug_info; // Whether auto response when recv messages. // default to true for it's very easy to use the protocol stack. - // @see: https://github.com/ossrs/srs/issues/217 bool auto_response_when_recv; // When not auto response message, manual flush the messages in queue. std::vector manual_response_queue; @@ -208,7 +205,6 @@ class SrsProtocol public: // Set the auto response message when recv for protocol stack. // @param v, whether auto response message when recv message. - // @see: https://github.com/ossrs/srs/issues/217 virtual void set_auto_response(bool v); // Flush for manual response when the auto response is disabled // by set_auto_response(false), we default use auto response, so donot @@ -222,13 +218,11 @@ class SrsProtocol // that is, we merge some data to read together. // @param v true to ename merged read. // @param handler the handler when merge read is enabled. - // @see https://github.com/ossrs/srs/issues/241 virtual void set_merge_read(bool v, IMergeReadHandler* handler); // Create buffer with specifeid size. // @param buffer the size of buffer. // @remark when MR(SRS_PERF_MERGED_READ) disabled, always set to 8K. // @remark when buffer changed, the previous ptr maybe invalid. - // @see https://github.com/ossrs/srs/issues/241 virtual void set_recv_buffer(int buffer_size); #endif public: @@ -429,7 +423,6 @@ class SrsRequest std::string stream; // For play live stream, // used to specified the stop when exceed the duration. - // @see https://github.com/ossrs/srs/issues/45 // in srs_utime_t. srs_utime_t duration; // The token in the connect request, @@ -614,7 +607,6 @@ class SrsRtmpServer public: // Set the auto response message when recv for protocol stack. // @param v, whether auto response message when recv message. - // @see: https://github.com/ossrs/srs/issues/217 virtual void set_auto_response(bool v); #ifdef SRS_PERF_MERGED_READ // To improve read performance, merge some packets then read, @@ -622,13 +614,11 @@ class SrsRtmpServer // that is, we merge some data to read together. // @param v true to ename merged read. // @param handler the handler when merge read is enabled. - // @see https://github.com/ossrs/srs/issues/241 virtual void set_merge_read(bool v, IMergeReadHandler* handler); // Create buffer with specifeid size. // @param buffer the size of buffer. // @remark when MR(SRS_PERF_MERGED_READ) disabled, always set to 8K. // @remark when buffer changed, the previous ptr maybe invalid. - // @see https://github.com/ossrs/srs/issues/241 virtual void set_recv_buffer(int buffer_size); #endif // To set/get the recv timeout in srs_utime_t. @@ -669,7 +659,6 @@ class SrsRtmpServer // @param stream_id, the stream id of packet to send over, 0 for control message. // // @remark performance issue, to support 6k+ 250kbps client, - // @see https://github.com/ossrs/srs/issues/194 virtual srs_error_t send_and_free_messages(SrsSharedPtrMessage** msgs, int nb_msgs, int stream_id); // Send the RTMP packet and always free it. // user must never free or use the packet after this method, @@ -1330,11 +1319,9 @@ class SrsSampleAccessPacket : public SrsPacket // Name of command. Set to "|RtmpSampleAccess". std::string command_name; // Whether allow access the sample of video. - // @see: https://github.com/ossrs/srs/issues/49 // @see: http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/net/NetStream.html#videoSampleAccess bool video_sample_access; // Whether allow access the sample of audio. - // @see: https://github.com/ossrs/srs/issues/49 // @see: http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/3/flash/net/NetStream.html#audioSampleAccess bool audio_sample_access; public: diff --git a/trunk/src/protocol/srs_service_rtmp_conn.cpp b/trunk/src/protocol/srs_service_rtmp_conn.cpp index 4ea32dd2de..1bda9e6609 100644 --- a/trunk/src/protocol/srs_service_rtmp_conn.cpp +++ b/trunk/src/protocol/srs_service_rtmp_conn.cpp @@ -97,7 +97,6 @@ srs_error_t SrsBasicRtmpClient::do_connect_app(string local_ip, bool debug) } // notify server the edge identity, - // @see https://github.com/ossrs/srs/issues/147 SrsAmf0Object* data = req->args; data->set("srs_sig", SrsAmf0Any::str(RTMP_SIG_SRS_KEY)); data->set("srs_server", SrsAmf0Any::str(RTMP_SIG_SRS_SERVER)); @@ -121,7 +120,6 @@ srs_error_t SrsBasicRtmpClient::do_connect_app(string local_ip, bool debug) req->tcUrl = tc_url; // upnode server identity will show in the connect_app of client. - // @see https://github.com/ossrs/srs/issues/160 // the debug_srs_upnode is config in vhost and default to true. SrsServerInfo si; if ((err = client->connect_app(req->app, tc_url, req, debug, &si)) != srs_success) { diff --git a/trunk/src/protocol/srs_service_st.cpp b/trunk/src/protocol/srs_service_st.cpp index ee9774119f..577c1204b5 100644 --- a/trunk/src/protocol/srs_service_st.cpp +++ b/trunk/src/protocol/srs_service_st.cpp @@ -41,7 +41,6 @@ srs_error_t srs_st_init() { #ifdef __linux__ // check epoll, some old linux donot support epoll. - // @see https://github.com/ossrs/srs/issues/162 if (!srs_st_epoll_is_supported()) { return srs_error_new(ERROR_ST_SET_EPOLL, "linux epoll disabled"); } @@ -495,7 +494,6 @@ srs_error_t SrsStSocket::read(void* buf, size_t size, ssize_t* nread) // (a value of 0 means the network connection is closed or end of file is reached). // Otherwise, a value of -1 is returned and errno is set to indicate the error. if (nb_read <= 0) { - // @see https://github.com/ossrs/srs/issues/200 if (nb_read < 0 && errno == ETIME) { return srs_error_new(ERROR_SOCKET_TIMEOUT, "timeout %d ms", srsu2msi(rtm)); } @@ -531,7 +529,6 @@ srs_error_t SrsStSocket::read_fully(void* buf, size_t size, ssize_t* nread) // (a value less than nbyte means the network connection is closed or end of file is reached) // Otherwise, a value of -1 is returned and errno is set to indicate the error. if (nb_read != (ssize_t)size) { - // @see https://github.com/ossrs/srs/issues/200 if (nb_read < 0 && errno == ETIME) { return srs_error_new(ERROR_SOCKET_TIMEOUT, "timeout %d ms", srsu2msi(rtm)); } @@ -566,7 +563,6 @@ srs_error_t SrsStSocket::write(void* buf, size_t size, ssize_t* nwrite) // On success a non-negative integer equal to nbyte is returned. // Otherwise, a value of -1 is returned and errno is set to indicate the error. if (nb_write <= 0) { - // @see https://github.com/ossrs/srs/issues/200 if (nb_write < 0 && errno == ETIME) { return srs_error_new(ERROR_SOCKET_TIMEOUT, "write timeout %d ms", srsu2msi(stm)); } @@ -597,7 +593,6 @@ srs_error_t SrsStSocket::writev(const iovec *iov, int iov_size, ssize_t* nwrite) // On success a non-negative integer equal to nbyte is returned. // Otherwise, a value of -1 is returned and errno is set to indicate the error. if (nb_write <= 0) { - // @see https://github.com/ossrs/srs/issues/200 if (nb_write < 0 && errno == ETIME) { return srs_error_new(ERROR_SOCKET_TIMEOUT, "writev timeout %d ms", srsu2msi(stm)); } diff --git a/trunk/src/utest/srs_utest_protocol.cpp b/trunk/src/utest/srs_utest_protocol.cpp index 80d14fbd57..cbf4873d2e 100644 --- a/trunk/src/utest/srs_utest_protocol.cpp +++ b/trunk/src/utest/srs_utest_protocol.cpp @@ -727,7 +727,6 @@ VOID TEST(ProtocolStackTest, ProtocolRecvMessage) // 0x04 where: message_type=4(protocol control user-control message) // 0x00 0x06 where: event Ping(0x06) // 0x00 0x00 0x0d 0x0f where: event data 4bytes ping timestamp. -// @see: https://github.com/ossrs/srs/issues/98 VOID TEST(ProtocolStackTest, ProtocolRecvMessageBug98) { MockBufferIO bio; @@ -3824,7 +3823,6 @@ VOID TEST(ProtocolStackTest, ProtocolRecvExtTimeMessage2) * always use 31bits timestamp. */ // always use 31bits timestamp, for some server may use 32bits extended timestamp. -// @see https://github.com/ossrs/srs/issues/111 VOID TEST(ProtocolStackTest, ProtocolRecvExtTimeMessage3) { MockBufferIO bio;