Remove unreachable issues in code (#3793)

remove unreachable links by python scripts:
```
def is_delete_issue(link):
    try:
        response = requests.get(link)
    except RequestException as e:
        print(f"An error occurred while trying to get the link: {e}")
        return False

    return "This issue has been deleted." in response.text


def remove_unreachable_links(dir):
    string_to_search = re.compile(r'// @see https://github\.com/ossrs/srs/issues/.*')
    file_list = util.find_files_with_extension(dir, ".cpp", True)
    for file in file_list:
        lines = []
        with open(file, "r", encoding="utf-8") as f:
            lines = f.readlines()
        with open(file, "w", encoding="utf-8", newline="\n") as f:    
            for line in lines:
                if string_to_search.search(line):
                    result = re.search(r'https://github\.com/ossrs/srs/issues/\d+', line)
                    if result:
                        link = result.group()
                        if is_delete_issue(link):
                            print("is_delete_issue link: file: %s, line: %s" % (file, line))
                            continue
                    
                f.write(line)

if __name__ == "__main__":
    remove_unreachable_links("srs/trunk/src/")
```

trunk/src/app/srs_app_dvr.cpp

@@ -868,7 +868,6 @@ srs_error_t SrsDvrSegmentPlan::update_duration(SrsSharedPtrMessage* msg)
     }
     
     // when wait keyframe, ignore if no frame arrived.
-    // @see https://github.com/ossrs/srs/issues/177
     if (wait_keyframe) {
         if (!msg->is_video()) {
             return err;

trunk/src/app/srs_app_edge.cpp

@@ -201,7 +201,6 @@ srs_error_t SrsEdgeFlvUpstream::do_connect(SrsRequest* r, SrsLbRoundRobin* lb, i
     if (redirect_depth == 0) {
         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) {

trunk/src/app/srs_app_ffmpeg.cpp

@@ -173,7 +173,6 @@ srs_error_t SrsFFMPEG::initialize_transcode(SrsConfDirective* engine)
     
     // for not rtmp input, donot append the iformat,
     // for example, "-f flv" before "-i udp://192.168.1.252:2222"
-    // @see https://github.com/ossrs/srs/issues/290
     if (!srs_string_starts_with(input, "rtmp://")) {
         iformat = "";
     }

trunk/src/app/srs_app_hls.cpp

@@ -554,7 +554,6 @@ bool SrsHlsMuxer::wait_keyframe()
 
 bool SrsHlsMuxer::is_segment_absolutely_overflow()
 {
-    // @see https://github.com/ossrs/srs/issues/151#issuecomment-83553950
     srs_assert(current);
     
     // to prevent very small segment.
@@ -812,7 +811,6 @@ srs_error_t SrsHlsMuxer::_refresh_m3u8(string m3u8_file)
      * rounded to the nearest integer. Its value MUST NOT change. A
      * typical target duration is 10 seconds.
      */
-    // @see https://github.com/ossrs/srs/issues/304#issuecomment-74000081
     srs_utime_t max_duration = segments->max_duration();
     int target_duration = (int)ceil(srsu2msi(srs_max(max_duration, max_td)) / 1000.0);
     
@@ -1025,9 +1023,7 @@ srs_error_t SrsHlsController::write_audio(SrsAudioFrame* frame, int64_t pts)
     // for example, pure audio when start, audio/video when publishing,
     // pure audio again for audio disabled.
     // so we reap event when the audio incoming when segment overflow.
-    // @see https://github.com/ossrs/srs/issues/151
     // we use absolutely overflow of segment to make jwplayer/ffplay happy
-    // @see https://github.com/ossrs/srs/issues/151#issuecomment-71155184
     if (tsmc->audio && muxer->is_segment_absolutely_overflow()) {
         if ((err = reap_segment()) != srs_success) {
             return srs_error_wrap(err, "hls: reap segment");

trunk/src/app/srs_app_recv_thread.hpp

@@ -136,7 +136,6 @@ private:
     // The video frames we got.
     uint64_t video_frames;
     // For mr(merged read),
-    // @see https://github.com/ossrs/srs/issues/241
     bool mr;
     int mr_fd;
     srs_utime_t mr_sleep;
@@ -149,7 +148,6 @@ private:
     // The params for conn callback.
     SrsLiveSource* _source;
     // The error timeout cond
-    // @see https://github.com/ossrs/srs/issues/244
     srs_cond_t error;
     // The merged context id.
     SrsContextId cid;

trunk/src/app/srs_app_rtmp_conn.cpp

@@ -1020,7 +1020,6 @@ srs_error_t SrsRtmpConn::do_publishing(SrsLiveSource* source, SrsPublishRecvThre
         // cond wait for timeout.
         if (nb_msgs == 0) {
             // when not got msgs, wait for a larger timeout.
-            // @see https://github.com/ossrs/srs/issues/441
             rtrd->wait(publish_1stpkt_timeout);
         } else {
             rtrd->wait(publish_normal_timeout);

trunk/src/app/srs_app_source.cpp

@@ -967,7 +967,6 @@ srs_error_t SrsOriginHub::on_audio(SrsSharedPtrMessage* shared_audio)
     
     if ((err = hls->on_audio(msg, format)) != srs_success) {
         // apply the error strategy for hls.
-        // @see https://github.com/ossrs/srs/issues/264
         std::string hls_error_strategy = _srs_config->get_hls_on_error(req_->vhost);
         if (srs_config_hls_is_on_error_ignore(hls_error_strategy)) {
             srs_warn("hls: ignore audio error %s", srs_error_desc(err).c_str());
@@ -1061,7 +1060,6 @@ srs_error_t SrsOriginHub::on_video(SrsSharedPtrMessage* shared_video, bool is_se
     if ((err = hls->on_video(msg, format)) != srs_success) {
         // TODO: We should support more strategies.
         // apply the error strategy for hls.
-        // @see https://github.com/ossrs/srs/issues/264
         std::string hls_error_strategy = _srs_config->get_hls_on_error(req_->vhost);
         if (srs_config_hls_is_on_error_ignore(hls_error_strategy)) {
             srs_warn("hls: ignore video error %s", srs_error_desc(err).c_str());
@@ -1629,7 +1627,6 @@ srs_error_t SrsMetaCache::dumps(SrsLiveConsumer* consumer, bool atc, SrsRtmpJitt
     
     // copy sequence header
     // copy audio sequence first, for hls to fast parse the "right" audio codec.
-    // @see https://github.com/ossrs/srs/issues/301
     if (aformat && aformat->acodec && aformat->acodec->id != SrsAudioCodecIdMP3) {
         if (ds && audio && (err = consumer->enqueue(audio, atc, ag)) != srs_success) {
             return srs_error_wrap(err, "enqueue audio sh");
@@ -1858,7 +1855,6 @@ srs_error_t SrsLiveSourceManager::notify(int event, srs_utime_t interval, srs_ut
 
         // TODO: FIXME: support source cleanup.
         // @see https://github.com/ossrs/srs/issues/713
-        // @see https://github.com/ossrs/srs/issues/714
 #if 0
         // When source expired, remove it.
         if (source->stream_is_dead()) {

trunk/src/kernel/srs_kernel_aac.cpp

@@ -115,7 +115,6 @@ srs_error_t SrsAacTransmuxer::write_audio(int64_t timestamp, char* data, int siz
     // write the ADTS header.
     // @see ISO_IEC_14496-3-AAC-2001.pdf, page 75,
     //      1.A.2.2 Audio_Data_Transport_Stream frame, ADTS
-    // @see https://github.com/ossrs/srs/issues/212#issuecomment-64145885
     // byte_alignment()
     
     // adts_fixed_header:

trunk/src/kernel/srs_kernel_ts.hpp

@@ -1258,7 +1258,6 @@ class SrsTsContextWriter
 {
 private:
     // User must config the codec in right way.
-    // @see https://github.com/ossrs/srs/issues/301
     SrsVideoCodecId vcodec_;
     SrsAudioCodecId acodec_;
 private:

trunk/src/main/srs_main_ingest_hls.cpp

@@ -1118,7 +1118,6 @@ int SrsIngestHlsOutput::write_h264_ipb_frame(string ibps, SrsVideoAvcFrameType f
     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 ERROR_H264_DROP_BEFORE_SPS_PPS;
     }

trunk/src/protocol/srs_protocol_http_stack.hpp

@@ -184,7 +184,6 @@ public:
     // @param data, the data to send. NULL to flush header only.
     virtual srs_error_t write(char* data, int size) = 0;
     // for the HTTP FLV, to writev to improve performance.
-    // @see https://github.com/ossrs/srs/issues/405
     virtual srs_error_t writev(const iovec* iov, int iovcnt, ssize_t* pnwrite) = 0;
     
     // WriteHeader sends an HTTP response header with status code.
@@ -261,7 +260,6 @@ public:
     // @param data, the data to send. NULL to flush header only.
     virtual srs_error_t write(char* data, int size) = 0;
     // for the HTTP FLV, to writev to improve performance.
-    // @see https://github.com/ossrs/srs/issues/405
     virtual srs_error_t writev(const iovec* iov, int iovcnt, ssize_t* pnwrite) = 0;
 
     // WriteHeader sends an HTTP request header with status code.

trunk/src/protocol/srs_protocol_rtmp_stack.cpp

@@ -799,7 +799,6 @@ srs_error_t SrsProtocol::recv_interlaced_message(SrsCommonMessage** pmsg)
     SrsChunkStream* chunk = NULL;
     
     // use chunk stream cache to get the chunk info.
-    // @see https://github.com/ossrs/srs/issues/249
     if (cid < SRS_PERF_CHUNK_STREAM_CACHE) {
         // already init, use it direclty
         chunk = cs_cache[cid];
@@ -2469,7 +2468,6 @@ srs_error_t SrsRtmpServer::identify_client(int stream_id, SrsRtmpConnType& type,
 
         // 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<SrsCallPacket*>(pkt);
         if (call) {

trunk/src/protocol/srs_protocol_stream.hpp

@@ -56,7 +56,6 @@ private:
 #endif
     // the user-space buffer to fill by reader,
     // which use fast index and reset when chunk body read ok.
-    // @see https://github.com/ossrs/srs/issues/248
     // ptr to the current read position.
     char* p;
     // ptr to the content end.