mirror of
https://github.com/ossrs/srs.git
synced 2025-03-09 15:49:59 +00:00
For #2200, Enable RTC and FLV for GB28181
This commit is contained in:
xialixin@kanzhun.com
winlin
parent
adb6f723c7
commit
4df6fa540f
12 changed files with 1289 additions and 845 deletions
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@ -269,6 +269,15 @@ function OSX_prepare()
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echo "Please install pkg-config"; exit -1;
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fi
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if [[ $SRS_GB28181 == YES ]]; then
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if [[ ! -f /usr/local/opt/libiconv/lib/libiconv.a ]]; then
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echo "install libiconv"
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echo "brew install libiconv"
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brew install libiconv; ret=$?; if [[ 0 -ne $ret ]]; then return $ret; fi
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echo "install libiconv success"
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fi
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fi
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echo "OSX install tools success"
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return 0
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}
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@ -10,6 +10,11 @@ http_api {
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listen 1985;
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}
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http_server {
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enabled on;
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listen 8080;
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}
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stats {
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network 0;
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}
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@ -29,7 +34,7 @@ stream_caster {
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listen 9000;
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# 多路复用端口类型，on为tcp，off为udp
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# 默认：off
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tcp_enable on;
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tcp_enable off;
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# rtp接收监听端口范围，最小值
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rtp_port_min 58200;
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@ -111,5 +116,27 @@ stream_caster {
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query_catalog_interval 60;
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}
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}
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vhost __defaultVhost__ {
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rtc_server {
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enabled on;
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# Listen at udp://8000
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listen 8000;
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#
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# The $CANDIDATE means fetch from env, if not configed, use * as default.
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#
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# The * means retrieving server IP automatically, from all network interfaces,
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# @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
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candidate $CANDIDATE;
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}
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vhost __defaultVhost__ {
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rtc {
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enabled on;
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bframe discard;
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}
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http_remux {
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enabled on;
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mount [vhost]/[app]/[stream].flv;
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}
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}
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21
trunk/configure
vendored
21
trunk/configure
vendored
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@ -169,6 +169,11 @@ if [[ $SRS_SRT == YES ]]; then
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if [[ $SRS_SHARED_SRT == YES ]]; then LibSRTfile="-lsrt"; fi
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fi
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# For iconv on macOS only, CentOS seems ok.
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if [[ $SRS_GB28181 == YES && $SRS_OSX == YES ]]; then
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LibIconvRoot="/usr/local/opt/libiconv/include"; LibIconvfile="/usr/local/opt/libiconv/lib/libiconv.a"
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fi
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# the link options, always use static link
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SrsLinkOptions="-ldl -lpthread";
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if [[ $SRS_SSL == YES && $SRS_USE_SYS_SSL == YES ]]; then
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@ -231,9 +236,6 @@ if [[ $SRS_RTC == YES ]]; then
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MODULE_FILES+=("srs_rtc_stun_stack")
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ModuleLibIncs+=(${LibSrtpRoot})
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fi
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if [[ $SRS_GB28181 == YES ]]; then
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MODULE_FILES+=("srs_sip_stack")
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fi
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if [[ $SRS_FFMPEG_FIT == YES ]]; then
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ModuleLibIncs+=("${LibFfmpegRoot[*]}")
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fi
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@ -283,8 +285,16 @@ if [[ $SRS_FFMPEG_FIT == YES ]]; then
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MODULE_FILES+=("srs_app_rtc_codec")
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fi
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if [[ $SRS_GB28181 == YES ]]; then
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MODULE_FILES+=("srs_app_gb28181" "srs_app_gb28181_sip" "srs_app_gb28181_jitbuffer")
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MODULE_FILES+=("srs_app_gb28181" "srs_app_gb28181_sip")
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fi
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if [[ $SRS_GB28181 == YES || $SRS_RTC == YES ]]; then
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MODULE_FILES+=("srs_app_rtc_jitbuffer")
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fi
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if [[ $SRS_GB28181 == YES ]]; then
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MODULE_FILES+=("srs_app_sip")
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ModuleLibIncs+=(${LibIconvRoot})
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fi
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DEFINES=""
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# add each modules for app
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for SRS_MODULE in ${SRS_MODULES[*]}; do
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@ -358,6 +368,9 @@ fi
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if [[ $SRS_SRT == YES ]]; then
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ModuleLibFiles+=("${LibSRTfile[*]}")
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fi
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if [[ $SRS_GB28181 == YES ]]; then
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ModuleLibFiles+=("${LibIconvfile[*]}")
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fi
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# all depends objects
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MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${APP_OBJS[@]} ${SERVER_OBJS[@]}"
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ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibGperfRoot} ${LibSSLRoot})
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@ -122,7 +122,7 @@
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当前通道:<a id="gb28181ChannelId"></a>
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<div>
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<textarea class="span6" id="txt_rtmp_url" rows="2"></textarea>
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<button class="btn btn-primary" id="btn_play">RTMP播放</button>
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<button class="btn btn-primary" id="btn_play">FLV播放</button>
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</div>
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<div>
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@ -141,187 +141,17 @@
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</div>
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<div id="main_content" class="hide">
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<div id="link_modal" class="modal hide fade">
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<div class="modal-header">
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<button type="button" class="close" data-dismiss="modal" aria-hidden="true">×</button>
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<h3><a href="https://github.com/ossrs/srs">SRS Link Generator</a></h3>
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</div>
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<div class="modal-body">
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<div class="form-horizontal">
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<div class="control-group">
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<label class="control-label" for="link_server">服务器地址</label>
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<div class="controls">
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<span id="link_server" class="span4 uneditable-input"></span>
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</div>
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</div>
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<div class="control-group">
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<label class="control-label" for="link_port">服务器端口</label>
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<div class="controls">
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<span id="link_port" class="span2 uneditable-input"></span>
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</div>
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</div>
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<div class="control-group">
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<label class="control-label" for="link_vhost">RTMP Vhost</label>
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<div class="controls">
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<span id="link_vhost" class="span4 uneditable-input"></span>
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</div>
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</div>
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<div class="control-group">
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<label class="control-label" for="link_app">RTMP App</label>
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<div class="controls">
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<span id="link_app" class="span4 uneditable-input"></span>
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</div>
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</div>
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<div class="control-group">
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<label class="control-label" for="link_stream">RTMP Stream</label>
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<div class="controls">
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<span id="link_stream" class="span4 uneditable-input"></span>
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</div>
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</div>
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<div class="control-group">
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<label class="control-label" for="link_rtmp">RTMP地址</label>
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<div class="controls">
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<span id="link_rtmp" class="span4 uneditable-input"></span>
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</div>
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</div>
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<div class="control-group">
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<label class="control-label" for="link_url">播放链接地址</label>
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<div class="controls">
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<div style="margin-top:5px;"><a href="#" id="link_url" target="_blank">请右键拷贝此链接地址.</a></div>
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</div>
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</div>
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</div>
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</div>
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<div class="modal-footer"></div>
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</div>
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<div id="main_modal" class="modal hide fade">
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<div class="modal-header">
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<button type="button" class="close" data-dismiss="modal" aria-hidden="true">×</button>
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<h3><a href="https://github.com/ossrs/srs">SrsPlayer</a></h3>
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<h3><a href="https://github.com/ossrs/srs">SrsFlvPlayer</a></h3>
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</div>
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<div class="modal-body">
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<div id="player"></div>
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<div class="progress progress-striped active" id="pb_buffer_bg">
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<div class="bar" style="width: 0%;" id="pb_buffer"></div>
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<div>
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<video id="video_player" width="98%" autoplay controls></video>
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</div>
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</div>
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<div class="modal-footer" id="my_modal_footer">
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<div>
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<div class="btn-group dropup">
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<button class="btn dropdown-toggle" data-toggle="dropdown">
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全屏比例大小<span class="caret"></span>
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</button>
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<ul class="dropdown-menu">
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<li><a id="btn_fs_size_screen_100" href="#">屏幕大小(100%)</a></li>
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<li><a id="btn_fs_size_screen_75" href="#">屏幕大小(75%)</a></li>
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<li><a id="btn_fs_size_screen_50" href="#">屏幕大小(50%)</a></li>
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<li><a id="btn_fs_size_video_100" href="#">视频大小(100%)</a></li>
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<li><a id="btn_fs_size_video_75" href="#">视频大小(75%)</a></li>
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<li><a id="btn_fs_size_video_50" href="#">视频大小(50%)</a></li>
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</ul>
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</div>
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<div class="btn-group dropup">
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<button class="btn dropdown-toggle" data-toggle="dropdown">显示比例<span class="caret"></span></button>
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<ul class="dropdown-menu">
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<li><a id="btn_dar_original" href="#">视频原始比例</a></li>
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<li><a id="btn_dar_21_9" href="#">宽屏影院(21:9)</a></li>
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<li><a id="btn_dar_16_9" href="#">宽屏电视(16:9)</a></li>
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<li><a id="btn_dar_4_3" href="#">窄屏(4:3)</a></li>
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<li><a id="btn_dar_fill" href="#">填充(容器比例)</a></li>
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</ul>
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</div>
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<div class="btn-group dropup">
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<button class="btn dropdown-toggle" data-toggle="dropdown">缓冲区大小<span class="caret"></span></button>
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<ul class="dropdown-menu">
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<li><a id="btn_bt_0_1" href="#">0.1秒(实时)</a></li>
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<li><a id="btn_bt_0_2" href="#">0.2秒(实时)</a></li>
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<li><a id="btn_bt_0_3" href="#">0.3秒(实时)</a></li>
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<li><a id="btn_bt_0_5" href="#">0.5秒(实时)</a></li>
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<li><a id="btn_bt_0_8" href="#">0.8秒(会议)</a></li>
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<li><a id="btn_bt_1_0" href="#">1秒(低延迟)</a></li>
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<li><a id="btn_bt_2_0" href="#">2秒(较低延时)</a></li>
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<li><a id="btn_bt_3_0" href="#">3秒(流畅播放)</a></li>
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<li><a id="btn_bt_4_0" href="#">4秒(流畅播放)</a></li>
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<li><a id="btn_bt_5_0" href="#">5秒(网速较低)</a></li>
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<li><a id="btn_bt_6_0" href="#">6秒(网速较低)</a></li>
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<li><a id="btn_bt_8_0" href="#">8秒(网速较低)</a></li>
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<li><a id="btn_bt_10_0" href="#">10秒(无所谓延迟)</a></li>
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<li><a id="btn_bt_15_0" href="#">15秒(无所谓延迟)</a></li>
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<li><a id="btn_bt_20_0" href="#">20秒(无所谓延迟)</a></li>
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<li><a id="btn_bt_30_0" href="#">30秒(流畅第一)</a></li>
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</ul>
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</div>
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<div class="btn-group dropup">
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<button class="btn dropdown-toggle" data-toggle="dropdown">最大缓冲区<span class="caret"></span></button>
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<ul class="dropdown-menu">
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<li><a id="btn_mbt_0_6" href="#">0.6秒(实时)</a></li>
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<li><a id="btn_mbt_0_9" href="#">0.9秒(实时)</a></li>
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<li><a id="btn_mbt_1_2" href="#">1.2秒(实时)</a></li>
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<li><a id="btn_mbt_1_5" href="#">1.5秒(实时)</a></li>
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<li><a id="btn_mbt_2_4" href="#">2.4秒(会议)</a></li>
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<li><a id="btn_mbt_3_0" href="#">3秒(低延迟)</a></li>
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<li><a id="btn_mbt_6_0" href="#">6秒(较低延时)</a></li>
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<li><a id="btn_mbt_9_0" href="#">9秒(流畅播放)</a></li>
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<li><a id="btn_mbt_12_0" href="#">12秒(流畅播放)</a></li>
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<li><a id="btn_mbt_15_0" href="#">15秒(网速较低)</a></li>
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<li><a id="btn_mbt_18_0" href="#">18秒(网速较低)</a></li>
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<li><a id="btn_mbt_24_0" href="#">24秒(网速较低)</a></li>
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<li><a id="btn_mbt_30_0" href="#">30秒(无所谓延迟)</a></li>
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<li><a id="btn_mbt_45_0" href="#">45秒(无所谓延迟)</a></li>
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<li><a id="btn_mbt_60_0" href="#">60秒(无所谓延迟)</a></li>
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<li><a id="btn_mbt_90_0" href="#">90秒(流畅第一)</a></li>
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</ul>
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</div>
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<div class="btn-group dropup">
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<a id="btn_fullscreen" class="btn">全屏</a>
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</div>
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<div class="btn-group dropup">
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<button id="btn_pause" class="btn">暂停播放</button>
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<button id="btn_resume" class="btn hide">继续播放</button>
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</div>
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<div class="btn-group dropup">
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<button class="btn btn-primary" data-dismiss="modal" aria-hidden="true">关闭播放器</button>
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</div>
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</div>
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<div class="hide" id="fullscreen_tips">
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请<font color="red">点击视频</font>进入全屏模式~<br/>
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由于安全原因，Flash全屏无法使用JS触发
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</div>
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<div>
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<div class="input-prepend div_play_time" title="BufferLength/BufferTime/MaxBufferTime">
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<span class="add-on">@B</span>
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<input class="span2" style="width:80px" id="txt_buffer" type="text" placeholder="0/0/0s">
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</div>
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<div class="input-prepend div_play_time" title="视频的播放流畅度">
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<span class="add-on">@F</span>
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<input class="span2" style="width:57px" id="txt_fluency" type="text" placeholder="100%">
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</div>
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<div class="input-prepend div_play_time" title="视频总共卡顿次数">
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<span class="add-on">@E</span>
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<input class="span2" style="width:45px" id="txt_empty_count" type="text" placeholder="0">
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</div>
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<div class="input-prepend div_play_time" title="视频当前的帧率FPS">
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<span class="add-on">@F</span>
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<input class="span2" style="width:55px" id="txt_fps" type="text" placeholder="fps">
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</div>
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<div class="input-prepend div_play_time" title="视频当前的码率(视频+音频)，单位：Kbps">
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<span class="add-on">@B</span>
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<input class="span2" style="width:55px" id="txt_bitrate" type="text" placeholder="kbps">
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</div>
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<div class="input-prepend div_play_time" title="播放时长，格式：天 时:分:秒">
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<span class="add-on">@T</span>
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<input class="span2" style="width:85px" id="txt_time" type="text" placeholder="天 时:分:秒">
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</div>
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</div>
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<div style="margin-top:-12px;">
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<span id="debug_info"></span>
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URL: <a href="#" id="player_url"></a>
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<div class="input-prepend div_play_time" title="当前时间：年-月-日 时:分:秒">
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<span class="add-on">@N</span>
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<input class="span2" style="width:135px" id="player_clock" type="text" placeholder="年-月-日 时:分:秒">
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</div>
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</div>
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<div>
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<div class="btn-group dropup">
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<button class="btn btn-primary" id="btn_ptz_up"> 上↑ </button>
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@ -332,13 +162,8 @@
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<button class="btn btn-primary" id="btn_ptz_zoomout"> 缩小－ </button>
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</div>
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[注意] !!! 云台控制需要启用内部sip功能
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<div class="input-prepend" title="首播时间，点播放到开始播放的时间，秒">
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<span class="add-on">@PST</span>
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<input class="span1" style="width:60px" id="txt_pst" type="text" placeholder="N秒">
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</div>
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</div>
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</div>
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</div>
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<div id="rtc_player_modal" class="modal hide fade">
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@ -410,54 +235,17 @@
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</body>
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<script type="text/javascript" src="js/jquery-1.10.2.min.js"></script>
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<script type="text/javascript" src="js/bootstrap.min.js"></script>
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<script type="text/javascript" src="js/swfobject.js"></script>
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<script type="text/javascript" src="js/json2.js"></script>
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<script type="text/javascript" src="js/srs.page.js"></script>
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<script type="text/javascript" src="js/srs.log.js"></script>
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<script type="text/javascript" src="js/srs.player.js"></script>
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<script type="text/javascript" src="js/srs.publisher.js"></script>
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<script type="text/javascript" src="js/srs.utility.js"></script>
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<script type="text/javascript" src="js/winlin.utility.js"></script>
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<script type="text/javascript" src="js/flv-1.5.0.min.js"></script>
|
||||
<script type="text/javascript" src="js/hls-0.14.17.min.js"></script>
|
||||
<script type="text/javascript">
|
||||
var __on_flash_ready = null;
|
||||
|
||||
$(function(){
|
||||
// 探测Flash是否正常启用。
|
||||
$('#main_flash_hdr').html(
|
||||
'\
|
||||
<object classid="clsid:d27cdb6e-ae6d-11cf-96b8-444553540000" width="100%" height="100%"> \
|
||||
<param name="movie" value="srs_player/release/srs_player.swf"> \
|
||||
<param name="quality" value="autohigh"> \
|
||||
<param name="swliveconnect" value="true"> \
|
||||
<param name="allowScriptAccess" value="always"> \
|
||||
<param name="bgcolor" value="#0"> \
|
||||
<param name="allowFullScreen" value="true"> \
|
||||
<param name="wmode" value="opaque"> \
|
||||
<param name="FlashVars" value="log=1"> \
|
||||
<param name="flashvars" value="id=1&on_player_ready=__on_flash_ready"> \
|
||||
<embed src="srs_player/release/srs_player.swf" width="100%" height="100%" \
|
||||
quality="autohigh" bgcolor="#0" align="middle" allowfullscreen="true" allowscriptaccess="always" \
|
||||
type="application/x-shockwave-flash" swliveconnect="true" wmode="opaque" \
|
||||
flashvars="id=1&on_player_ready=__on_flash_ready" \
|
||||
pluginspage="http://www.macromedia.com/go/getflashplayer"> \
|
||||
</object> \
|
||||
'
|
||||
);
|
||||
$('#main_flash_hdr').show();
|
||||
|
||||
var showFlashHdr = setTimeout(function(){
|
||||
$('#main_flash_alert').show();
|
||||
}, 300);
|
||||
|
||||
__on_flash_ready = function (id) {
|
||||
clearTimeout(showFlashHdr);
|
||||
|
||||
$('#main_flash_alert').hide();
|
||||
$('#main_flash_hdr').hide();
|
||||
$('#main_content').show();
|
||||
|
||||
autoLoadPage();
|
||||
};
|
||||
});
|
||||
</script>
|
||||
<script type="text/javascript">
|
||||
|
|
@ -471,7 +259,6 @@
|
|||
} else {
|
||||
$("#txt_api_url").val("http://" + query.host + ":1985");
|
||||
}
|
||||
|
||||
var __active_dar = null;
|
||||
function select_dar(dar_id, num, den) {
|
||||
srs_player.set_dar(num, den);
|
||||
|
|
@ -631,13 +418,13 @@
|
|||
$('#gb28181ChannelMessage').html(syntaxHighlight(ret));
|
||||
|
||||
if (ret.code == 0 && ret.data != undefined ) {
|
||||
$("#txt_rtmp_url").val(ret.data.channels[0].rtmp_url);
|
||||
$("#txt_rtmp_url").val(ret.data.channels[0].flv_url);
|
||||
|
||||
ret.data.channels[0].rtmp_url.split("//")
|
||||
ret.data.channels[0].flv_url.split("//")
|
||||
|
||||
var urlObject = parse_rtmp_url(ret.data.channels[0].rtmp_url);
|
||||
var werbrtc_url = "webrtc://"+ urlObject.server + "/" + urlObject.app + "/" + urlObject.stream;
|
||||
$("#txt_rtc_url").val(werbrtc_url);
|
||||
//var urlObject = parse_rtmp_url(ret.data.channels[0].rtmp_url);
|
||||
//var werbrtc_url = "webrtc://"+ urlObject.server + "/" + urlObject.app + "/" + urlObject.stream;
|
||||
$("#txt_rtc_url").val(ret.data.channels[0].webrtc_url);
|
||||
}
|
||||
}
|
||||
|
||||
|
|
@ -659,6 +446,292 @@
|
|||
}
|
||||
|
||||
|
||||
// Async-await-promise based SRS RTC Player.
|
||||
function SrsRtcPlayerAsync() {
|
||||
var self = {};
|
||||
|
||||
// @see https://github.com/rtcdn/rtcdn-draft
|
||||
// @url The WebRTC url to play with, for example:
|
||||
// webrtc://r.ossrs.net/live/livestream
|
||||
// or specifies the API port:
|
||||
// webrtc://r.ossrs.net:11985/live/livestream
|
||||
// or autostart the play:
|
||||
// webrtc://r.ossrs.net/live/livestream?autostart=true
|
||||
// or change the app from live to myapp:
|
||||
// webrtc://r.ossrs.net:11985/myapp/livestream
|
||||
// or change the stream from livestream to mystream:
|
||||
// webrtc://r.ossrs.net:11985/live/mystream
|
||||
// or set the api server to myapi.domain.com:
|
||||
// webrtc://myapi.domain.com/live/livestream
|
||||
// or set the candidate(ip) of answer:
|
||||
// webrtc://r.ossrs.net/live/livestream?eip=39.107.238.185
|
||||
// or force to access https API:
|
||||
// webrtc://r.ossrs.net/live/livestream?schema=https
|
||||
// or use plaintext, without SRTP:
|
||||
// webrtc://r.ossrs.net/live/livestream?encrypt=false
|
||||
// or any other information, will pass-by in the query:
|
||||
// webrtc://r.ossrs.net/live/livestream?vhost=xxx
|
||||
// webrtc://r.ossrs.net/live/livestream?token=xxx
|
||||
self.play = async function(url) {
|
||||
var conf = self.__internal.prepareUrl(url);
|
||||
self.pc.addTransceiver("audio", {direction: "recvonly"});
|
||||
self.pc.addTransceiver("video", {direction: "recvonly"});
|
||||
|
||||
var offer = await self.pc.createOffer();
|
||||
await self.pc.setLocalDescription(offer);
|
||||
var session = await new Promise(function(resolve, reject) {
|
||||
// @see https://github.com/rtcdn/rtcdn-draft
|
||||
var data = {
|
||||
api: conf.apiUrl, streamurl: conf.streamUrl, clientip: null, sdp: offer.sdp
|
||||
};
|
||||
console.log("Generated offer: ", data);
|
||||
|
||||
$.ajax({
|
||||
type: "POST", url: conf.apiUrl, data: JSON.stringify(data),
|
||||
contentType:'application/json', dataType: 'json'
|
||||
}).done(function(data) {
|
||||
console.log("Got answer: ", data);
|
||||
if (data.code) {
|
||||
reject(data); return;
|
||||
}
|
||||
|
||||
resolve(data);
|
||||
}).fail(function(reason){
|
||||
reject(reason);
|
||||
});
|
||||
});
|
||||
await self.pc.setRemoteDescription(
|
||||
new RTCSessionDescription({type: 'answer', sdp: session.sdp})
|
||||
);
|
||||
return session;
|
||||
};
|
||||
|
||||
// Close the publisher.
|
||||
self.close = function() {
|
||||
self.pc.close();
|
||||
};
|
||||
|
||||
// The callback when got remote stream.
|
||||
self.onaddstream = function (event) {};
|
||||
|
||||
// Internal APIs.
|
||||
self.__internal = {
|
||||
defaultPath: '/rtc/v1/play/',
|
||||
prepareUrl: function (webrtcUrl) {
|
||||
var urlObject = self.__internal.parse(webrtcUrl);
|
||||
|
||||
// If user specifies the schema, use it as API schema.
|
||||
var schema = urlObject.user_query.schema;
|
||||
schema = schema ? schema + ':' : window.location.protocol;
|
||||
|
||||
var port = urlObject.port || 1985;
|
||||
if (schema === 'https:') {
|
||||
port = urlObject.port || 443;
|
||||
}
|
||||
|
||||
// @see https://github.com/rtcdn/rtcdn-draft
|
||||
var api = urlObject.user_query.play || self.__internal.defaultPath;
|
||||
if (api.lastIndexOf('/') !== api.length - 1) {
|
||||
api += '/';
|
||||
}
|
||||
|
||||
apiUrl = schema + '//' + urlObject.server + ':' + port + api;
|
||||
for (var key in urlObject.user_query) {
|
||||
if (key !== 'api' && key !== 'play') {
|
||||
apiUrl += '&' + key + '=' + urlObject.user_query[key];
|
||||
}
|
||||
}
|
||||
// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
|
||||
var apiUrl = apiUrl.replace(api + '&', api + '?');
|
||||
|
||||
var streamUrl = urlObject.url;
|
||||
|
||||
return {apiUrl: apiUrl, streamUrl: streamUrl, schema: schema, urlObject: urlObject, port: port};
|
||||
},
|
||||
parse: function (url) {
|
||||
// @see: http://stackoverflow.com/questions/10469575/how-to-use-location-object-to-parse-url-without-redirecting-the-page-in-javascri
|
||||
var a = document.createElement("a");
|
||||
a.href = url.replace("rtmp://", "http://")
|
||||
.replace("webrtc://", "http://")
|
||||
.replace("rtc://", "http://");
|
||||
|
||||
var vhost = a.hostname;
|
||||
var app = a.pathname.substr(1, a.pathname.lastIndexOf("/") - 1);
|
||||
var stream = a.pathname.substr(a.pathname.lastIndexOf("/") + 1);
|
||||
|
||||
// parse the vhost in the params of app, that srs supports.
|
||||
app = app.replace("...vhost...", "?vhost=");
|
||||
if (app.indexOf("?") >= 0) {
|
||||
var params = app.substr(app.indexOf("?"));
|
||||
app = app.substr(0, app.indexOf("?"));
|
||||
|
||||
if (params.indexOf("vhost=") > 0) {
|
||||
vhost = params.substr(params.indexOf("vhost=") + "vhost=".length);
|
||||
if (vhost.indexOf("&") > 0) {
|
||||
vhost = vhost.substr(0, vhost.indexOf("&"));
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// when vhost equals to server, and server is ip,
|
||||
// the vhost is __defaultVhost__
|
||||
if (a.hostname === vhost) {
|
||||
var re = /^(\d+)\.(\d+)\.(\d+)\.(\d+)$/;
|
||||
if (re.test(a.hostname)) {
|
||||
vhost = "__defaultVhost__";
|
||||
}
|
||||
}
|
||||
|
||||
// parse the schema
|
||||
var schema = "rtmp";
|
||||
if (url.indexOf("://") > 0) {
|
||||
schema = url.substr(0, url.indexOf("://"));
|
||||
}
|
||||
|
||||
var port = a.port;
|
||||
if (!port) {
|
||||
if (schema === 'http') {
|
||||
port = 80;
|
||||
} else if (schema === 'https') {
|
||||
port = 443;
|
||||
} else if (schema === 'rtmp') {
|
||||
port = 1935;
|
||||
}
|
||||
}
|
||||
|
||||
var ret = {
|
||||
url: url,
|
||||
schema: schema,
|
||||
server: a.hostname, port: port,
|
||||
vhost: vhost, app: app, stream: stream
|
||||
};
|
||||
self.__internal.fill_query(a.search, ret);
|
||||
|
||||
// For webrtc API, we use 443 if page is https, or schema specified it.
|
||||
if (!ret.port) {
|
||||
if (schema === 'webrtc' || schema === 'rtc') {
|
||||
if (ret.user_query.schema === 'https') {
|
||||
ret.port = 443;
|
||||
} else if (window.location.href.indexOf('https://') === 0) {
|
||||
ret.port = 443;
|
||||
} else {
|
||||
// For WebRTC, SRS use 1985 as default API port.
|
||||
ret.port = 1985;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
return ret;
|
||||
},
|
||||
fill_query: function (query_string, obj) {
|
||||
// pure user query object.
|
||||
obj.user_query = {};
|
||||
|
||||
if (query_string.length === 0) {
|
||||
return;
|
||||
}
|
||||
|
||||
// split again for angularjs.
|
||||
if (query_string.indexOf("?") >= 0) {
|
||||
query_string = query_string.split("?")[1];
|
||||
}
|
||||
|
||||
var queries = query_string.split("&");
|
||||
for (var i = 0; i < queries.length; i++) {
|
||||
var elem = queries[i];
|
||||
|
||||
var query = elem.split("=");
|
||||
obj[query[0]] = query[1];
|
||||
obj.user_query[query[0]] = query[1];
|
||||
}
|
||||
|
||||
// alias domain for vhost.
|
||||
if (obj.domain) {
|
||||
obj.vhost = obj.domain;
|
||||
}
|
||||
}
|
||||
};
|
||||
|
||||
self.pc = new RTCPeerConnection(null);
|
||||
self.pc.onaddstream = function (event) {
|
||||
if (self.onaddstream) {
|
||||
self.onaddstream(event);
|
||||
}
|
||||
};
|
||||
|
||||
return self;
|
||||
}
|
||||
|
||||
var flvPlayer = null;
|
||||
var hlsPlayer = null;
|
||||
|
||||
var stopPlayers = function () {
|
||||
if (flvPlayer) {
|
||||
flvPlayer.destroy();
|
||||
flvPlayer = null;
|
||||
}
|
||||
if (hlsPlayer) {
|
||||
hlsPlayer.destroy();
|
||||
hlsPlayer = null;
|
||||
}
|
||||
};
|
||||
|
||||
var hide_for_error = function () {
|
||||
$('#main_flash_alert').show();
|
||||
$('#main_info').hide();
|
||||
$('#main_tips').hide();
|
||||
$('#video_player').hide();
|
||||
//$('#btn_play').hide();
|
||||
|
||||
stopPlayers();
|
||||
};
|
||||
|
||||
var show_for_ok = function () {
|
||||
$('#main_flash_alert').hide();
|
||||
$('#main_info').show();
|
||||
$('#main_tips').show();
|
||||
$('#video_player').show();
|
||||
//$('#btn_play').show();
|
||||
};
|
||||
|
||||
var start_play_live = function (r) {
|
||||
stopPlayers();
|
||||
if (!r) return;
|
||||
|
||||
// Start play HTTP-FLV.
|
||||
if (r.stream.indexOf('.flv') > 0) {
|
||||
if (!flvjs.isSupported()) {
|
||||
hide_for_error();
|
||||
return;
|
||||
}
|
||||
|
||||
show_for_ok();
|
||||
|
||||
flvPlayer = flvjs.createPlayer({type: 'flv', url: r.url});
|
||||
flvPlayer.attachMediaElement(document.getElementById('video_player'));
|
||||
flvPlayer.load();
|
||||
flvPlayer.play();
|
||||
return;
|
||||
}
|
||||
|
||||
// Start play HLS.
|
||||
if (r.stream.indexOf('.m3u8') > 0) {
|
||||
if (!Hls.isSupported()) {
|
||||
hide_for_error();
|
||||
return;
|
||||
}
|
||||
|
||||
show_for_ok();
|
||||
|
||||
hlsPlayer = new Hls();
|
||||
hlsPlayer.loadSource(r.url);
|
||||
hlsPlayer.attachMedia(document.getElementById('video_player'));
|
||||
return;
|
||||
}
|
||||
|
||||
console.error('不支持的URL', r.url, r);
|
||||
$('#video_player').hide();
|
||||
};
|
||||
|
||||
|
||||
/****
|
||||
|
|
@ -693,154 +766,11 @@
|
|||
// the play startup time.
|
||||
var pst = new Date();
|
||||
|
||||
var pc = null; // Global handler to do cleanup when replaying.
|
||||
|
||||
$("#main_modal").on("show", function(){
|
||||
if (srs_player) {
|
||||
return;
|
||||
}
|
||||
|
||||
$("#div_container").remove();
|
||||
$("#debug_info").text("");
|
||||
|
||||
var div_container = $("<div/>");
|
||||
$(div_container).attr("id", "div_container");
|
||||
$("#player").append(div_container);
|
||||
|
||||
var player = $("<div/>");
|
||||
$(player).attr("id", "player_id");
|
||||
$(div_container).append(player);
|
||||
|
||||
srs_player = new SrsPlayer("player_id", srs_get_player_width(), srs_get_player_height());
|
||||
srs_player.on_player_ready = function() {
|
||||
var buffer_time = 0.5;
|
||||
if (url.indexOf('.m3u8') > 0) {
|
||||
buffer_time = 2;
|
||||
}
|
||||
|
||||
if (query.buffer) {
|
||||
for (var i = 0; i < bts.length - 1; i++) {
|
||||
var cur = bts[i];
|
||||
var next = bts[i+1];
|
||||
if (Number(query.buffer) >= cur && Number(query.buffer) < next) {
|
||||
buffer_time = cur;
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
select_buffer(buffer_time);
|
||||
this.play(url);
|
||||
|
||||
pst = new Date();
|
||||
};
|
||||
srs_player.on_player_status = function(code, desc) {
|
||||
//console.log("[播放器状态] code=" + code + ", desc=" + desc);
|
||||
};
|
||||
srs_player.on_player_metadata = function(metadata) {
|
||||
$("#btn_dar_original").text("视频原始比例" + "(" + metadata.width + ":" + metadata.height + ")");
|
||||
if (metadata.ip && metadata.pid && metadata.cid) {
|
||||
$("#debug_info").text("ID:" + metadata.ip + '/' + metadata.pid + '/' + metadata.cid + '');
|
||||
}
|
||||
select_dar("#btn_dar_original", 0, 0);
|
||||
select_fs_size("#btn_fs_size_screen_100", "screen", 100);
|
||||
};
|
||||
srs_player.on_player_timer = function(time, buffer_length, kbps, fps, rtime) {
|
||||
if (time > 0 && pst) {
|
||||
var diff = (new Date().getTime() - pst.getTime()) / 1000.0;
|
||||
$("#txt_pst").val(Number(diff).toFixed(2) + "秒");
|
||||
pst = null;
|
||||
}
|
||||
|
||||
var buffer = buffer_length / this.max_buffer_time * 100;
|
||||
$("#pb_buffer").width(Number(buffer).toFixed(1) + "%");
|
||||
|
||||
$("#pb_buffer_bg").attr("title",
|
||||
"缓冲区:" + buffer_length.toFixed(1) + "秒, 最大缓冲区:"
|
||||
+ this.max_buffer_time.toFixed(1) + "秒, 当前:"
|
||||
+ buffer.toFixed(1) + "%");
|
||||
|
||||
var bts = this.buffer_time >= 1? this.buffer_time.toFixed(0) : this.buffer_time.toFixed(1);
|
||||
var mbts = this.buffer_time >= 1? this.max_buffer_time.toFixed(0) : this.max_buffer_time.toFixed(1);
|
||||
$("#txt_buffer").val(buffer_length.toFixed(1) + "/" + bts + "/" + mbts + "s");
|
||||
|
||||
$("#txt_bitrate").val(kbps.toFixed(0) + "kbps");
|
||||
$("#txt_fps").val(fps.toFixed(1) + "fps");
|
||||
$("#txt_empty_count").val(srs_player.empty_count() + "次");
|
||||
$("#txt_fluency").val(srs_player.fluency().toFixed(2) + "%");
|
||||
|
||||
var time_str = "";
|
||||
// day
|
||||
time_str = padding(parseInt(time / 24 / 3600), 2, '0') + " ";
|
||||
// hour
|
||||
time = time % (24 * 3600);
|
||||
time_str += padding(parseInt(time / 3600), 2, '0') + ":";
|
||||
// minute
|
||||
time = time % (3600);
|
||||
time_str += padding(parseInt(time / 60), 2, '0') + ":";
|
||||
// seconds
|
||||
time = time % (60);
|
||||
time_str += padding(parseInt(time), 2, '0');
|
||||
// show
|
||||
$("#txt_time").val(time_str);
|
||||
|
||||
var clock = new Date().getTime() / 1000;
|
||||
$("#player_clock").val(absolute_seconds_to_YYYYmmdd(clock) + " " + absolute_seconds_to_HHMMSS(clock));
|
||||
};
|
||||
srs_player.start();
|
||||
});
|
||||
|
||||
$("#main_modal").on("hide", function(){
|
||||
if (srs_player) {
|
||||
// report the log to backend.
|
||||
//console.log(srs_player.dump_log());
|
||||
|
||||
srs_player.stop();
|
||||
srs_player = null;
|
||||
}
|
||||
});
|
||||
|
||||
var apply_url_change = function() {
|
||||
var rtmp = parse_rtmp_url($("#txt_url").val());
|
||||
var url = "http://" + query.host + query.pathname + "?"
|
||||
+ "app=" + rtmp.app + "&stream=" + rtmp.stream
|
||||
+ "&server=" + rtmp.server + "&port=" + rtmp.port
|
||||
+ "&autostart=true";
|
||||
if (query.shp_identify) {
|
||||
url += "&shp_identify=" + query.shp_identify;
|
||||
}
|
||||
if (rtmp.vhost == "__defaultVhost__") {
|
||||
url += "&vhost=" + rtmp.server;
|
||||
} else {
|
||||
url += "&vhost=" + rtmp.vhost;
|
||||
}
|
||||
if (rtmp.schema == "http") {
|
||||
url += "&schema=http";
|
||||
}
|
||||
if (query.buffer) {
|
||||
url += "&buffer=" + query.buffer;
|
||||
}
|
||||
if (query.api_port) {
|
||||
url += "&api_port=" + query.api_port;
|
||||
}
|
||||
|
||||
var queries = user_extra_params(query);
|
||||
if (queries && queries.length) {
|
||||
url += '&' + queries.join('&');
|
||||
}
|
||||
$("#player_url").text($("#txt_url").val()).attr("href", url);
|
||||
|
||||
$("#link_server").text(rtmp.server);
|
||||
$("#link_port").text(rtmp.port);
|
||||
$("#link_vhost").text(rtmp.vhost);
|
||||
$("#link_app").text(rtmp.app);
|
||||
$("#link_stream").text(rtmp.stream);
|
||||
$("#link_rtmp").text($("#txt_url").val());
|
||||
$("#link_url").attr("href", url);
|
||||
};
|
||||
|
||||
$("#txt_url").change(function(){
|
||||
apply_url_change();
|
||||
});
|
||||
|
||||
$("#btn_generate_link").click(function(){
|
||||
|
|
@ -848,91 +778,11 @@
|
|||
});
|
||||
|
||||
$("#btn_play").click(function(){
|
||||
url = $("#txt_rtmp_url").val();
|
||||
$("#main_modal").modal({show:true, keyboard:true});
|
||||
var r = parse_rtmp_url($("#txt_rtmp_url").val());
|
||||
start_play_live(r);
|
||||
});
|
||||
|
||||
$("#btn_fullscreen").click(function(){
|
||||
$("#fullscreen_tips").toggle();
|
||||
});
|
||||
|
||||
$("#btn_pause").click(function() {
|
||||
$("#btn_resume").toggle();
|
||||
$("#btn_pause").toggle();
|
||||
srs_player.pause();
|
||||
});
|
||||
$("#btn_resume").click(function(){
|
||||
$("#btn_resume").toggle();
|
||||
$("#btn_pause").toggle();
|
||||
srs_player.resume();
|
||||
});
|
||||
|
||||
if (true) {
|
||||
$("#srs_publish").click(function () {
|
||||
url = $("#srs_publish").text();
|
||||
$("#main_modal").modal({show: true, keyboard: false});
|
||||
});
|
||||
$("#srs_publish_ld").click(function () {
|
||||
url = $("#srs_publish_ld").text();
|
||||
$("#main_modal").modal({show: true, keyboard: false});
|
||||
});
|
||||
$("#srs_publish_sd").click(function () {
|
||||
url = $("#srs_publish_sd").text();
|
||||
$("#main_modal").modal({show: true, keyboard: false});
|
||||
});
|
||||
$("#srs_publish_fw").click(function () {
|
||||
url = $("#srs_publish_fw").text();
|
||||
$("#main_modal").modal({show: true, keyboard: false});
|
||||
});
|
||||
$("#srs_publish_fw_ld").click(function () {
|
||||
url = $("#srs_publish_fw_ld").text();
|
||||
$("#main_modal").modal({show: true, keyboard: false});
|
||||
});
|
||||
$("#srs_publish_fw_sd").click(function () {
|
||||
url = $("#srs_publish_fw_sd").text();
|
||||
$("#main_modal").modal({show: true, keyboard: false});
|
||||
});
|
||||
}
|
||||
|
||||
if (true) {
|
||||
$("#btn_dar_original").click(function(){
|
||||
select_dar("#btn_dar_original", 0, 0);
|
||||
});
|
||||
$("#btn_dar_21_9").click(function(){
|
||||
select_dar("#btn_dar_21_9", 21, 9);
|
||||
});
|
||||
$("#btn_dar_16_9").click(function(){
|
||||
select_dar("#btn_dar_16_9", 16, 9);
|
||||
});
|
||||
$("#btn_dar_4_3").click(function(){
|
||||
select_dar("#btn_dar_4_3", 4, 3);
|
||||
});
|
||||
$("#btn_dar_fill").click(function(){
|
||||
select_dar("#btn_dar_fill", -1, -1);
|
||||
});
|
||||
}
|
||||
|
||||
if (true) {
|
||||
$("#btn_fs_size_video_100").click(function(){
|
||||
select_fs_size("#btn_fs_size_video_100", "video", 100);
|
||||
});
|
||||
$("#btn_fs_size_video_75").click(function(){
|
||||
select_fs_size("#btn_fs_size_video_75", "video", 75);
|
||||
});
|
||||
$("#btn_fs_size_video_50").click(function(){
|
||||
select_fs_size("#btn_fs_size_video_50", "video", 50);
|
||||
});
|
||||
$("#btn_fs_size_screen_100").click(function(){
|
||||
select_fs_size("#btn_fs_size_screen_100", "screen", 100);
|
||||
});
|
||||
$("#btn_fs_size_screen_75").click(function(){
|
||||
select_fs_size("#btn_fs_size_screen_75", "screen", 75);
|
||||
});
|
||||
$("#btn_fs_size_screen_50").click(function(){
|
||||
select_fs_size("#btn_fs_size_screen_50", "screen", 50);
|
||||
});
|
||||
}
|
||||
|
||||
if (true) {
|
||||
for (var i = 0; i < bts.length; i++) {
|
||||
var bt = bts[i];
|
||||
|
|
@ -1160,92 +1010,50 @@
|
|||
$("#create_channel_modal").modal('hide');
|
||||
});
|
||||
|
||||
var sdk = null; // Global handler to do cleanup when replaying.
|
||||
var startPlay = function() {
|
||||
$('#rtc_media_player').show();
|
||||
|
||||
// Close PC when user replay.
|
||||
if (sdk) {
|
||||
sdk.close();
|
||||
}
|
||||
|
||||
sdk = new SrsRtcPlayerAsync();
|
||||
sdk.onaddstream = function (event) {
|
||||
console.log('Start play, event: ', event);
|
||||
$('#rtc_media_player').prop('srcObject', event.stream);
|
||||
};
|
||||
|
||||
// For example:
|
||||
// webrtc://r.ossrs.net/live/livestream
|
||||
var url = $("#txt_rtc_url").val();
|
||||
sdk.play(url).then(function(session){
|
||||
$('#sessionid').html(session.sessionid);
|
||||
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
|
||||
}).catch(function (reason) {
|
||||
sdk.close();
|
||||
$('#rtc_media_player').hide();
|
||||
console.error(reason);
|
||||
});
|
||||
};
|
||||
|
||||
$("#btn_rtc_play").click(function(){
|
||||
$('#rtc_media_player').width(srs_get_player_width);
|
||||
$('#rtc_media_player').height(srs_get_player_height);
|
||||
$("#rtc_player_modal").modal({show: true, keyboard: false});
|
||||
|
||||
startPlay($("#txt_rtc_url").val());
|
||||
startPlay();
|
||||
});
|
||||
|
||||
|
||||
$("#rtc_player_modal").on("hide", function(){
|
||||
if (pc) {
|
||||
pc.close();
|
||||
if (sdk) {
|
||||
sdk.close();
|
||||
}
|
||||
});
|
||||
|
||||
var startPlay = function(url) {
|
||||
$('#rtc_media_player').show();
|
||||
var urlObject = parse_rtmp_url(url);
|
||||
var schema = window.location.protocol;
|
||||
|
||||
// Close PC when user replay.
|
||||
if (pc) {
|
||||
pc.close();
|
||||
}
|
||||
|
||||
pc = new RTCPeerConnection(null);
|
||||
pc.onaddstream = function (event) {
|
||||
console.debug(event.stream);
|
||||
$('#rtc_media_player').prop('srcObject', event.stream);
|
||||
};
|
||||
new Promise(function(resolve, reject) {
|
||||
pc.addTransceiver("audio", {direction: "recvonly"});
|
||||
pc.addTransceiver("video", {direction: "recvonly"});
|
||||
|
||||
pc.createOffer(function(offer){
|
||||
resolve(offer);
|
||||
},function(reason){
|
||||
reject(reason);
|
||||
});
|
||||
}).then(function(offer) {
|
||||
return pc.setLocalDescription(offer).then(function(){ return offer; });
|
||||
}).then(function(offer) {
|
||||
return new Promise(function(resolve, reject) {
|
||||
// var port = urlObject.port || 1985;
|
||||
var port = 1985;
|
||||
|
||||
// @see https://github.com/rtcdn/rtcdn-draft
|
||||
var api = urlObject.user_query.play || '/rtc/v1/play/';
|
||||
if (api.lastIndexOf('/') != api.length - 1) {
|
||||
api += '/';
|
||||
}
|
||||
|
||||
var url = schema + '//' + urlObject.server + ':' + port + api;
|
||||
for (var key in urlObject.user_query) {
|
||||
if (key != 'api' && key != 'play') {
|
||||
url += '&' + key + '=' + urlObject.user_query[key];
|
||||
}
|
||||
}
|
||||
// Replace /rtc/v1/play/&k=v to /rtc/v1/play/?k=v
|
||||
url = url.replace(api + '&', api + '?');
|
||||
|
||||
// @see https://github.com/rtcdn/rtcdn-draft
|
||||
var data = {
|
||||
api: url, streamurl: urlObject.url, clientip: null, sdp: offer.sdp
|
||||
};
|
||||
console.log("offer: " + JSON.stringify(data));
|
||||
|
||||
$.ajax({
|
||||
type: "POST", url: url, data: JSON.stringify(data),
|
||||
contentType:'application/json', dataType: 'json'
|
||||
}).done(function(data) {
|
||||
console.log("answer: " + JSON.stringify(data));
|
||||
resolve(data.sdp);
|
||||
}).fail(function(reason){
|
||||
reject(reason);
|
||||
});
|
||||
});
|
||||
}).then(function(answer) {
|
||||
return pc.setRemoteDescription(new RTCSessionDescription({type: 'answer', sdp: answer}));
|
||||
}).catch(function(reason) {
|
||||
throw reason;
|
||||
});
|
||||
};
|
||||
}
|
||||
|
||||
apply_url_change();
|
||||
};
|
||||
</script>
|
||||
</html>
|
||||
|
|
|
|||
|
|
@ -50,7 +50,7 @@ using namespace std;
|
|||
#include <srs_app_rtmp_conn.hpp>
|
||||
#include <srs_protocol_utility.hpp>
|
||||
#include <srs_protocol_format.hpp>
|
||||
#include <srs_sip_stack.hpp>
|
||||
#include <srs_app_sip.hpp>
|
||||
|
||||
//#define W_PS_FILE
|
||||
//#define W_VIDEO_FILE
|
||||
|
|
@ -405,7 +405,6 @@ srs_error_t SrsGb28181PsRtpProcessor::rtmpmuxer_enqueue_data(SrsGb28181RtmpMuxer
|
|||
srs_error_t SrsGb28181PsRtpProcessor::on_rtp_packet_jitter(const sockaddr* from, const int fromlen, char* buf, int nb_buf)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
bool completed = false;
|
||||
|
||||
pprint->elapse();
|
||||
|
||||
|
|
@ -739,13 +738,12 @@ srs_error_t SrsGb28181TcpPsRtpProcessor::rtmpmuxer_enqueue_data(SrsGb28181RtmpMu
|
|||
srs_error_t SrsGb28181TcpPsRtpProcessor::on_rtp_packet_jitter(char* buf, int nb_buf, std::string ip, int port)
|
||||
{
|
||||
srs_error_t err = srs_success;
|
||||
bool completed = false;
|
||||
|
||||
pprint->elapse();
|
||||
|
||||
char address_string[64] = {0};
|
||||
char port_string[16] = {0};
|
||||
/*if (getnameinfo(from, fromlen,
|
||||
/*char port_string[16] = {0};
|
||||
if (getnameinfo(from, fromlen,
|
||||
(char*)&address_string, sizeof(address_string),
|
||||
(char*)&port_string, sizeof(port_string),
|
||||
NI_NUMERICHOST | NI_NUMERICSERV)) {
|
||||
|
|
@ -1109,8 +1107,8 @@ srs_error_t SrsPsStreamDemixer::on_ps_stream(char* ps_data, int ps_size, uint32_
|
|||
|
||||
uint8_t p1 = (uint8_t)(next_ps_pack[0]);
|
||||
uint8_t p2 = (uint8_t)(next_ps_pack[1]);
|
||||
uint8_t p3 = (uint8_t)(next_ps_pack[2]);
|
||||
uint8_t p4 = (uint8_t)(next_ps_pack[3]);
|
||||
//uint8_t p3 = (uint8_t)(next_ps_pack[2]);
|
||||
//uint8_t p4 = (uint8_t)(next_ps_pack[3]);
|
||||
|
||||
if (audio_enable && audio_es_type != STREAM_TYPE_AUDIO_AAC &&
|
||||
(p1 & 0xFF) == 0xFF && (p2 & 0xF0) == 0xF0) {
|
||||
|
|
@ -1260,8 +1258,8 @@ SrsGb28181RtmpMuxer::SrsGb28181RtmpMuxer(SrsGb28181Manger* c, std::string id, bo
|
|||
((SrsSTCoroutine*)trd)->set_stack_size(1 << 18);
|
||||
|
||||
sdk = NULL;
|
||||
vjitter = new SrsRtspJitter();
|
||||
ajitter = new SrsRtspJitter();
|
||||
vjitter = new SrsRtpTimeJitter();
|
||||
ajitter = new SrsRtpTimeJitter();
|
||||
|
||||
avc = new SrsRawH264Stream();
|
||||
aac = new SrsRawAacStream();
|
||||
|
|
@ -1284,8 +1282,8 @@ SrsGb28181RtmpMuxer::SrsGb28181RtmpMuxer(SrsGb28181Manger* c, std::string id, bo
|
|||
source = NULL;
|
||||
source_publish = true;
|
||||
|
||||
jitter_buffer = new SrsPsJitterBuffer(id);
|
||||
jitter_buffer_audio = new SrsPsJitterBuffer(id);
|
||||
jitter_buffer = new SrsRtpJitterBuffer(id);
|
||||
jitter_buffer_audio = new SrsRtpJitterBuffer(id);
|
||||
|
||||
ps_buflen = 0;
|
||||
ps_buffer = NULL;
|
||||
|
|
@ -1414,7 +1412,7 @@ srs_error_t SrsGb28181RtmpMuxer::initialize(SrsServer *s, SrsRequest* r)
|
|||
srs_error_t err = srs_success;
|
||||
|
||||
if (!jitter_buffer) {
|
||||
jitter_buffer = new SrsPsJitterBuffer(channel_id);
|
||||
jitter_buffer = new SrsRtpJitterBuffer(channel_id);
|
||||
}
|
||||
|
||||
jitter_buffer->SetDecodeErrorMode(kSelectiveErrors);
|
||||
|
|
@ -1422,7 +1420,7 @@ srs_error_t SrsGb28181RtmpMuxer::initialize(SrsServer *s, SrsRequest* r)
|
|||
jitter_buffer->SetNackSettings(250, 450, 0);
|
||||
|
||||
if (!jitter_buffer_audio) {
|
||||
jitter_buffer_audio = new SrsPsJitterBuffer(channel_id);
|
||||
jitter_buffer_audio = new SrsRtpJitterBuffer(channel_id);
|
||||
}
|
||||
|
||||
jitter_buffer_audio->SetDecodeErrorMode(kSelectiveErrors);
|
||||
|
|
@ -1458,6 +1456,7 @@ srs_error_t SrsGb28181RtmpMuxer::do_cycle()
|
|||
send_rtmp_stream_time = srs_get_system_time();
|
||||
uint32_t cur_timestamp = 0;
|
||||
int buffer_size = 0;
|
||||
bool keyframe = false;
|
||||
|
||||
//consume ps stream, and check status
|
||||
while (true) {
|
||||
|
|
@ -1472,7 +1471,7 @@ srs_error_t SrsGb28181RtmpMuxer::do_cycle()
|
|||
|
||||
if (config.jitterbuffer_enable){
|
||||
if(jitter_buffer->FoundFrame(cur_timestamp)){
|
||||
jitter_buffer->GetPsFrame(&ps_buffer, ps_buflen, buffer_size, cur_timestamp);
|
||||
jitter_buffer->GetFrame(&ps_buffer, ps_buflen, buffer_size, keyframe, cur_timestamp);
|
||||
|
||||
if (buffer_size > 0){
|
||||
if ((err = ps_demixer->on_ps_stream(ps_buffer, buffer_size, cur_timestamp, 0)) != srs_success){
|
||||
|
|
@ -1483,7 +1482,7 @@ srs_error_t SrsGb28181RtmpMuxer::do_cycle()
|
|||
}
|
||||
|
||||
if(jitter_buffer_audio->FoundFrame(cur_timestamp)){
|
||||
jitter_buffer_audio->GetPsFrame(&ps_buffer_audio, ps_buflen_auido, buffer_size, cur_timestamp);
|
||||
jitter_buffer_audio->GetFrame(&ps_buffer_audio, ps_buflen_auido, buffer_size, keyframe, cur_timestamp);
|
||||
|
||||
if (buffer_size > 0){
|
||||
if ((err = ps_demixer->on_ps_stream(ps_buffer_audio, buffer_size, cur_timestamp, 0)) != srs_success){
|
||||
|
|
@ -1600,10 +1599,12 @@ void SrsGb28181RtmpMuxer::insert_jitterbuffer(SrsPsRtpPacket *pkt)
|
|||
//otherwise audio uses jitter_buffer_audio, and video uses jitter_buffer
|
||||
if (av_same_ts){
|
||||
pkt->marker = false;
|
||||
jitter_buffer->InsertPacket(*pkt, pkt->payload->bytes(), pkt->payload->length(), NULL);
|
||||
jitter_buffer->InsertPacket(pkt->sequence_number, pkt->timestamp, pkt->marker,
|
||||
pkt->payload->bytes(), pkt->payload->length(), NULL);
|
||||
ps_rtp_video_ts = pkt->timestamp;
|
||||
}else {
|
||||
jitter_buffer_audio->InsertPacket(*pkt, pkt->payload->bytes(), pkt->payload->length(), NULL);
|
||||
jitter_buffer_audio->InsertPacket(pkt->sequence_number, pkt->timestamp, pkt->marker,
|
||||
pkt->payload->bytes(), pkt->payload->length(), NULL);
|
||||
}
|
||||
|
||||
//srs_cond_signal(wait_ps_queue);
|
||||
|
|
@ -1771,13 +1772,7 @@ srs_error_t SrsGb28181RtmpMuxer::write_h264_ipb_frame2(char *frame, int frame_si
|
|||
|
||||
//0001xxxxxxxxxx
|
||||
//xxxx0001xxxxxxx
|
||||
uint32_t naluLen = size - cur_pos;
|
||||
char *p = (char*)&naluLen;
|
||||
|
||||
video_data[cur_pos] = p[3];
|
||||
video_data[cur_pos+1] = p[2];
|
||||
video_data[cur_pos+2] = p[1];
|
||||
video_data[cur_pos+3] = p[0];
|
||||
uint32_t naluLen = size - cur_pos - 4;
|
||||
|
||||
char *frame = video_data + cur_pos + 4;
|
||||
int frame_size = naluLen;
|
||||
|
|
@ -1797,12 +1792,6 @@ srs_error_t SrsGb28181RtmpMuxer::write_h264_ipb_frame2(char *frame, int frame_si
|
|||
//0001xxxxxxxx0001xxxxxxxx0001xxxxxxxxx
|
||||
//xxxxxxxxxxxx0001xxxxxxxx0001xxxxxxxxx
|
||||
uint32_t naluLen = cur_pos - pre_pos - 4;
|
||||
char *p = (char*)&naluLen;
|
||||
|
||||
video_data[pre_pos] = p[3];
|
||||
video_data[pre_pos+1] = p[2];
|
||||
video_data[pre_pos+2] = p[1];
|
||||
video_data[pre_pos+3] = p[0];
|
||||
|
||||
char *frame = video_data + pre_pos + 4;
|
||||
int frame_size = naluLen;
|
||||
|
|
@ -1816,12 +1805,6 @@ srs_error_t SrsGb28181RtmpMuxer::write_h264_ipb_frame2(char *frame, int frame_si
|
|||
if (first_pos != pre_pos){
|
||||
|
||||
uint32_t naluLen = size - pre_pos - 4;
|
||||
char *p = (char*)&naluLen;
|
||||
|
||||
video_data[pre_pos] = p[3];
|
||||
video_data[pre_pos+1] = p[2];
|
||||
video_data[pre_pos+2] = p[1];
|
||||
video_data[pre_pos+3] = p[0];
|
||||
|
||||
char *frame = video_data + pre_pos + 4;
|
||||
int frame_size = naluLen;
|
||||
|
|
@ -2167,6 +2150,9 @@ SrsGb28181StreamChannel::SrsGb28181StreamChannel(){
|
|||
rtp_peer_port = 0;
|
||||
rtp_peer_ip = "";
|
||||
rtmp_url = "";
|
||||
flv_url = "";
|
||||
hls_url = "";
|
||||
webrtc_url = "";
|
||||
recv_time = 0;
|
||||
recv_time_str = "";
|
||||
}
|
||||
|
|
@ -2191,6 +2177,9 @@ void SrsGb28181StreamChannel::copy(const SrsGb28181StreamChannel *s){
|
|||
rtp_peer_port = s->get_rtp_peer_port();
|
||||
|
||||
rtmp_url = s->get_rtmp_url();
|
||||
flv_url = s->get_flv_url();
|
||||
hls_url = s->get_hls_url();
|
||||
webrtc_url = s->get_webrtc_url();
|
||||
|
||||
recv_time_str = s->get_recv_time_str();
|
||||
recv_time = s->get_recv_time();
|
||||
|
|
@ -2205,6 +2194,9 @@ void SrsGb28181StreamChannel::dumps(SrsJsonObject* obj)
|
|||
obj->set("app", SrsJsonAny::str(app.c_str()));
|
||||
obj->set("stream", SrsJsonAny::str(stream.c_str()));
|
||||
obj->set("rtmp_url", SrsJsonAny::str(rtmp_url.c_str()));
|
||||
obj->set("flv_url", SrsJsonAny::str(flv_url.c_str()));
|
||||
obj->set("hls_url", SrsJsonAny::str(hls_url.c_str()));
|
||||
obj->set("webrtc_url", SrsJsonAny::str(webrtc_url.c_str()));
|
||||
|
||||
obj->set("ssrc", SrsJsonAny::integer(ssrc));
|
||||
obj->set("rtp_port", SrsJsonAny::integer(rtp_port));
|
||||
|
|
@ -2298,7 +2290,7 @@ uint32_t SrsGb28181Manger::generate_ssrc(std::string id)
|
|||
//gb28181 live ssrc max value 0999999999(3B9AC9FF)
|
||||
//gb28181 vod ssrc max value 1999999999(773593FF)
|
||||
uint8_t index = uint8_t(rand() % (0x0F - 0x01 + 1) + 0x01);
|
||||
uint32_t ssrc = 0x2FFFF00 & (hash_code(id) << 8) | index;
|
||||
uint32_t ssrc = ((0x2FFFF00) & (hash_code(id) << 8)) | index;
|
||||
//uint32_t ssrc = 0x00FFFFFF & (hash_code(id));
|
||||
srs_trace("gb28181: generate ssrc id=%s, ssrc=%u", id.c_str(), ssrc);
|
||||
return ssrc;
|
||||
|
|
@ -2599,7 +2591,21 @@ srs_error_t SrsGb28181Manger::create_stream_channel(SrsGb28181StreamChannel *cha
|
|||
channel->set_rtmp_port(rtmp_port);
|
||||
channel->set_ip(config->host);
|
||||
std::string play_url = srs_generate_rtmp_url(config->host, rtmp_port, "", "", app, stream_name, "");
|
||||
|
||||
std::string flv_url = srs_string_replace(play_url, "rtmp://", "http://");
|
||||
std::stringstream port;
|
||||
port << ":" << rtmp_port;
|
||||
flv_url = srs_string_replace(flv_url, port.str(), ":"+_srs_config->get_http_stream_listen());
|
||||
std::string hls_url = flv_url + ".m3u8";
|
||||
flv_url = flv_url + ".flv";
|
||||
|
||||
std::string webrtc_url = srs_string_replace(play_url, "rtmp://", "webrtc://");
|
||||
webrtc_url = srs_string_replace(webrtc_url, port.str(), ":"+_srs_config->get_http_api_listen());
|
||||
|
||||
channel->set_rtmp_url(play_url);
|
||||
channel->set_flv_url(flv_url);
|
||||
channel->set_hls_url(hls_url);
|
||||
channel->set_webrtc_url(webrtc_url);
|
||||
|
||||
request.app = app;
|
||||
request.stream = stream_name;
|
||||
|
|
@ -2876,7 +2882,6 @@ srs_error_t SrsGb28181Conn::do_cycle()
|
|||
|
||||
nb_read = nb_read + leftDataLength;
|
||||
|
||||
length;
|
||||
pp = (char*)&length;
|
||||
p = &(mbuffer[0]);
|
||||
pp[1] = *p++;
|
||||
|
|
|
|||
|
|
@ -34,13 +34,12 @@
|
|||
|
||||
#include <srs_app_st.hpp>
|
||||
#include <srs_app_listener.hpp>
|
||||
#include <srs_rtsp_stack.hpp>
|
||||
#include <srs_kernel_stream.hpp>
|
||||
#include <srs_app_log.hpp>
|
||||
#include <srs_kernel_file.hpp>
|
||||
#include <srs_protocol_json.hpp>
|
||||
#include <srs_app_gb28181_sip.hpp>
|
||||
#include <srs_app_gb28181_jitbuffer.hpp>
|
||||
#include <srs_app_rtc_jitbuffer.hpp>
|
||||
#include <srs_rtmp_stack.hpp>
|
||||
#include <srs_app_source.hpp>
|
||||
#include <srs_service_conn.hpp>
|
||||
|
|
@ -90,7 +89,7 @@ class SrsPithyPrint;
|
|||
class SrsSimpleRtmpClient;
|
||||
class SrsSipStack;
|
||||
class SrsGb28181Manger;
|
||||
class SrsRtspJitter;
|
||||
class SrsRtpTimeJitter;
|
||||
class SrsSipRequest;
|
||||
class SrsGb28181RtmpMuxer;
|
||||
class SrsGb28181Config;
|
||||
|
|
@ -99,7 +98,7 @@ class SrsGb28181TcpPsRtpProcessor;
|
|||
class SrsGb28181SipService;
|
||||
class SrsGb28181StreamChannel;
|
||||
class SrsGb28181SipSession;
|
||||
class SrsPsJitterBuffer;
|
||||
class SrsRtpJitterBuffer;
|
||||
class SrsServer;
|
||||
class SrsSource;
|
||||
class SrsRequest;
|
||||
|
|
@ -316,8 +315,8 @@ private:
|
|||
srs_cond_t wait_ps_queue;
|
||||
|
||||
SrsSimpleRtmpClient* sdk;
|
||||
SrsRtspJitter* vjitter;
|
||||
SrsRtspJitter* ajitter;
|
||||
SrsRtpTimeJitter* vjitter;
|
||||
SrsRtpTimeJitter* ajitter;
|
||||
|
||||
SrsRawH264Stream* avc;
|
||||
std::string h264_sps;
|
||||
|
|
@ -330,8 +329,8 @@ private:
|
|||
SrsSource* source;
|
||||
SrsServer* server;
|
||||
|
||||
SrsPsJitterBuffer *jitter_buffer;
|
||||
SrsPsJitterBuffer *jitter_buffer_audio;
|
||||
SrsRtpJitterBuffer *jitter_buffer;
|
||||
SrsRtpJitterBuffer *jitter_buffer_audio;
|
||||
|
||||
char *ps_buffer;
|
||||
char *ps_buffer_audio;
|
||||
|
|
@ -340,7 +339,6 @@ private:
|
|||
int ps_buflen_auido;
|
||||
|
||||
uint32_t ps_rtp_video_ts;
|
||||
uint32_t ps_rtp_audio_ts;
|
||||
|
||||
bool source_publish;
|
||||
|
||||
|
|
@ -444,6 +442,9 @@ private:
|
|||
std::string app;
|
||||
std::string stream;
|
||||
std::string rtmp_url;
|
||||
std::string flv_url;
|
||||
std::string hls_url;
|
||||
std::string webrtc_url;
|
||||
|
||||
std::string ip;
|
||||
int rtp_port;
|
||||
|
|
@ -472,6 +473,9 @@ public:
|
|||
uint32_t get_rtp_peer_port() const { return rtp_peer_port; }
|
||||
std::string get_rtp_peer_ip() const { return rtp_peer_ip; }
|
||||
std::string get_rtmp_url() const { return rtmp_url; }
|
||||
std::string get_flv_url() const { return flv_url; }
|
||||
std::string get_hls_url() const { return hls_url; }
|
||||
std::string get_webrtc_url() const { return webrtc_url; }
|
||||
srs_utime_t get_recv_time() const { return recv_time; }
|
||||
std::string get_recv_time_str() const { return recv_time_str; }
|
||||
|
||||
|
|
@ -486,6 +490,9 @@ public:
|
|||
void set_rtp_peer_ip( const std::string &p) { rtp_peer_ip = p; }
|
||||
void set_rtp_peer_port( const int &s) { rtp_peer_port = s;}
|
||||
void set_rtmp_url( const std::string &u) { rtmp_url = u; }
|
||||
void set_flv_url( const std::string &u) { flv_url = u; }
|
||||
void set_hls_url( const std::string &u) { hls_url = u; }
|
||||
void set_webrtc_url( const std::string &u) { webrtc_url = u; }
|
||||
void set_recv_time( const srs_utime_t &u) { recv_time = u; }
|
||||
void set_recv_time_str( const std::string &u) { recv_time_str = u; }
|
||||
|
||||
|
|
|
|||
|
|
@ -43,7 +43,7 @@ using namespace std;
|
|||
#include <srs_kernel_utility.hpp>
|
||||
#include <srs_kernel_codec.hpp>
|
||||
#include <srs_app_pithy_print.hpp>
|
||||
#include <srs_sip_stack.hpp>
|
||||
#include <srs_app_sip.hpp>
|
||||
#include <srs_app_gb28181.hpp>
|
||||
|
||||
|
||||
|
|
|
|||
|
|
@ -31,7 +31,7 @@
|
|||
#include <map>
|
||||
|
||||
#include <srs_app_log.hpp>
|
||||
#include <srs_sip_stack.hpp>
|
||||
#include <srs_app_sip.hpp>
|
||||
#include <srs_app_gb28181.hpp>
|
||||
#include <srs_app_pithy_print.hpp>
|
||||
#include <srs_service_conn.hpp>
|
||||
|
|
|
|||
File diff suppressed because it is too large
Load diff
|
|
@ -21,8 +21,8 @@
|
|||
* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
||||
*/
|
||||
|
||||
#ifndef SRS_APP_GB28181_JITBUFFER_HPP
|
||||
#define SRS_APP_GB28181_JITBUFFER_HPP
|
||||
#ifndef SRS_APP_RTC_RTP_JITBUFFER_HPP
|
||||
#define SRS_APP_RTC_RTP_JITBUFFER_HPP
|
||||
|
||||
#include <srs_core.hpp>
|
||||
|
||||
|
|
@ -36,13 +36,15 @@
|
|||
|
||||
#include <srs_app_log.hpp>
|
||||
#include <srs_kernel_utility.hpp>
|
||||
#include <srs_app_gb28181.hpp>
|
||||
#include <srs_kernel_rtc_rtp.hpp>
|
||||
#include <srs_kernel_flv.hpp>
|
||||
|
||||
class SrsPsRtpPacket;
|
||||
class SrsPsFrameBuffer;
|
||||
class PsDecodingState;
|
||||
class SrsRtpFrameBuffer;
|
||||
class SrsRtpDecodingState;
|
||||
class SrsGb28181RtmpMuxer;
|
||||
class VCMPacket;
|
||||
class SrsRtpPacket2;
|
||||
|
||||
///jittbuffer
|
||||
|
||||
|
|
@ -57,7 +59,7 @@ enum FrameType {
|
|||
};
|
||||
|
||||
// Used to indicate which decode with errors mode should be used.
|
||||
enum PsDecodeErrorMode {
|
||||
enum SrsRtpDecodeErrorMode {
|
||||
kNoErrors, // Never decode with errors. Video will freeze
|
||||
// if nack is disabled.
|
||||
kSelectiveErrors, // Frames that are determined decodable in
|
||||
|
|
@ -79,7 +81,7 @@ enum { kMaxVideoDelayMs = 10000 };
|
|||
enum { kPacketsPerFrameMultiplier = 5 };
|
||||
enum { kFastConvergeThreshold = 5};
|
||||
|
||||
enum PsJitterBufferEnum {
|
||||
enum SrsRtpJitterBufferEnum {
|
||||
kMaxConsecutiveOldFrames = 60,
|
||||
kMaxConsecutiveOldPackets = 300,
|
||||
kMaxPacketsInSession = 800,
|
||||
|
|
@ -87,7 +89,7 @@ enum PsJitterBufferEnum {
|
|||
kMaxJBFrameSizeBytes = 4000000 // sanity don't go above 4Mbyte.
|
||||
};
|
||||
|
||||
enum PsFrameBufferEnum {
|
||||
enum SrsRtpFrameBufferEnum {
|
||||
kOutOfBoundsPacket = -7,
|
||||
kNotInitialized = -6,
|
||||
kOldPacket = -5,
|
||||
|
|
@ -102,18 +104,65 @@ enum PsFrameBufferEnum {
|
|||
kDuplicatePacket = 5 // We're receiving a duplicate packet.
|
||||
};
|
||||
|
||||
enum PsFrameBufferStateEnum {
|
||||
enum SrsRtpFrameBufferStateEnum {
|
||||
kStateEmpty, // frame popped by the RTP receiver
|
||||
kStateIncomplete, // frame that have one or more packet(s) stored
|
||||
kStateComplete, // frame that have all packets
|
||||
kStateDecodable // Hybrid mode - frame can be decoded
|
||||
};
|
||||
|
||||
enum PsNackMode {
|
||||
enum SrsRtpNackMode {
|
||||
kNack,
|
||||
kNoNack
|
||||
};
|
||||
|
||||
// Used to indicate if a received packet contain a complete NALU (or equivalent)
|
||||
enum VCMNaluCompleteness {
|
||||
kNaluUnset = 0, // Packet has not been filled.
|
||||
kNaluComplete = 1, // Packet can be decoded as is.
|
||||
kNaluStart, // Packet contain beginning of NALU
|
||||
kNaluIncomplete, // Packet is not beginning or end of NALU
|
||||
kNaluEnd, // Packet is the end of a NALU
|
||||
};
|
||||
|
||||
enum RtpVideoCodecTypes {
|
||||
kRtpVideoNone,
|
||||
kRtpVideoGeneric,
|
||||
kRtpVideoVp8,
|
||||
kRtpVideoVp9,
|
||||
kRtpVideoH264,
|
||||
kRtpVideoPS
|
||||
};
|
||||
|
||||
|
||||
// Video codec types
|
||||
enum VideoCodecType {
|
||||
kVideoCodecVP8,
|
||||
kVideoCodecVP9,
|
||||
kVideoCodecH264,
|
||||
kVideoCodecH264SVC,
|
||||
kVideoCodecI420,
|
||||
kVideoCodecRED,
|
||||
kVideoCodecULPFEC,
|
||||
kVideoCodecGeneric,
|
||||
kVideoCodecH264PS,
|
||||
kVideoCodecUnknown
|
||||
};
|
||||
|
||||
// The packetization types that we support: single, aggregated, and fragmented.
|
||||
enum H264PacketizationTypes {
|
||||
kH264SingleNalu, // This packet contains a single NAL unit.
|
||||
kH264StapA, // This packet contains STAP-A (single time
|
||||
// aggregation) packets. If this packet has an
|
||||
// associated NAL unit type, it'll be for the
|
||||
// first such aggregated packet.
|
||||
kH264FuA, // This packet contains a FU-A (fragmentation
|
||||
// unit) packet, meaning it is a part of a frame
|
||||
// that was too large to fit into a single packet.
|
||||
};
|
||||
|
||||
enum { kH264StartCodeLengthBytes = 4};
|
||||
|
||||
// Used to pass data from jitter buffer to session info.
|
||||
// This data is then used in determining whether a frame is decodable.
|
||||
struct FrameData {
|
||||
|
|
@ -147,7 +196,7 @@ inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2)
|
|||
return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2;
|
||||
}
|
||||
|
||||
typedef std::list<SrsPsFrameBuffer*> UnorderedFrameList;
|
||||
typedef std::list<SrsRtpFrameBuffer*> UnorderedFrameList;
|
||||
|
||||
class TimestampLessThan {
|
||||
public:
|
||||
|
|
@ -159,16 +208,16 @@ public:
|
|||
};
|
||||
|
||||
class FrameList
|
||||
: public std::map<uint32_t, SrsPsFrameBuffer*, TimestampLessThan> {
|
||||
: public std::map<uint32_t, SrsRtpFrameBuffer*, TimestampLessThan> {
|
||||
public:
|
||||
void InsertFrame(SrsPsFrameBuffer* frame);
|
||||
SrsPsFrameBuffer* PopFrame(uint32_t timestamp);
|
||||
SrsPsFrameBuffer* Front() const;
|
||||
SrsPsFrameBuffer* FrontNext() const;
|
||||
SrsPsFrameBuffer* Back() const;
|
||||
void InsertFrame(SrsRtpFrameBuffer* frame);
|
||||
SrsRtpFrameBuffer* PopFrame(uint32_t timestamp);
|
||||
SrsRtpFrameBuffer* Front() const;
|
||||
SrsRtpFrameBuffer* FrontNext() const;
|
||||
SrsRtpFrameBuffer* Back() const;
|
||||
int RecycleFramesUntilKeyFrame(FrameList::iterator* key_frame_it,
|
||||
UnorderedFrameList* free_frames);
|
||||
void CleanUpOldOrEmptyFrames(PsDecodingState* decoding_state, UnorderedFrameList* free_frames);
|
||||
void CleanUpOldOrEmptyFrames(SrsRtpDecodingState* decoding_state, UnorderedFrameList* free_frames);
|
||||
void Reset(UnorderedFrameList* free_frames);
|
||||
};
|
||||
|
||||
|
|
@ -180,7 +229,13 @@ public:
|
|||
size_t size,
|
||||
uint16_t seqNum,
|
||||
uint32_t timestamp,
|
||||
bool markerBit);
|
||||
bool markerBit,
|
||||
H264PacketizationTypes type,
|
||||
RtpVideoCodecTypes rtpType,
|
||||
bool singlenual,
|
||||
bool isfirst,
|
||||
FrameType ftype
|
||||
);
|
||||
|
||||
void Reset();
|
||||
|
||||
|
|
@ -194,24 +249,32 @@ public:
|
|||
bool markerBit;
|
||||
|
||||
FrameType frameType;
|
||||
//cloopenwebrtc::VideoCodecType codec;
|
||||
VideoCodecType codec;
|
||||
|
||||
bool isFirstPacket; // Is this first packet in a frame.
|
||||
//VCMNaluCompleteness completeNALU; // Default is kNaluIncomplete.
|
||||
VCMNaluCompleteness completeNALU; // Default is kNaluIncomplete.
|
||||
bool insertStartCode; // True if a start code should be inserted before this
|
||||
// packet.
|
||||
int width;
|
||||
int height;
|
||||
//RTPVideoHeader codecSpecificHeader;
|
||||
|
||||
|
||||
//H264 header
|
||||
H264PacketizationTypes h264packetizationType;
|
||||
bool h264singleNalu;
|
||||
|
||||
public:
|
||||
void CopyCodecSpecifics(RtpVideoCodecTypes codecType, bool H264single_nalu, bool firstPacket);
|
||||
};
|
||||
|
||||
class SrsPsFrameBuffer {
|
||||
class SrsRtpFrameBuffer {
|
||||
public:
|
||||
SrsPsFrameBuffer();
|
||||
virtual ~SrsPsFrameBuffer();
|
||||
SrsRtpFrameBuffer();
|
||||
virtual ~SrsRtpFrameBuffer();
|
||||
|
||||
public:
|
||||
PsFrameBufferEnum InsertPacket(const VCMPacket& packet, const FrameData& frame_data);
|
||||
SrsRtpFrameBufferEnum InsertPacket(const VCMPacket& packet, const FrameData& frame_data);
|
||||
void UpdateCompleteSession();
|
||||
void UpdateDecodableSession(const FrameData& frame_data);
|
||||
bool HaveFirstPacket() const;
|
||||
|
|
@ -220,7 +283,7 @@ public:
|
|||
|
||||
uint32_t GetTimeStamp() const;
|
||||
FrameType GetFrameType() const;
|
||||
PsFrameBufferStateEnum GetState() const;
|
||||
SrsRtpFrameBufferStateEnum GetState() const;
|
||||
|
||||
int32_t GetHighSeqNum() const;
|
||||
int32_t GetLowSeqNum() const;
|
||||
|
|
@ -233,7 +296,6 @@ public:
|
|||
bool complete() const;
|
||||
bool decodable() const;
|
||||
|
||||
bool GetPsPlayload(SrsSimpleStream **ps_data, int &count);
|
||||
bool DeletePacket(int &count);
|
||||
void PrepareForDecode(bool continuous);
|
||||
|
||||
|
|
@ -248,7 +310,7 @@ private:
|
|||
const PacketIterator& prev_packet_it);
|
||||
|
||||
size_t InsertBuffer(uint8_t* frame_buffer, PacketIterator packet_it);
|
||||
size_t Insert(const uint8_t* buffer, size_t length, uint8_t* frame_buffer);
|
||||
size_t Insert(const uint8_t* buffer, size_t length, bool insert_start_code, uint8_t* frame_buffer);
|
||||
void ShiftSubsequentPackets(PacketIterator it, int steps_to_shift);
|
||||
void VerifyAndAllocate(const uint32_t minimumSize);
|
||||
void UpdateDataPointers(const uint8_t* old_base_ptr, const uint8_t* new_base_ptr);
|
||||
|
|
@ -269,11 +331,11 @@ private:
|
|||
uint32_t timeStamp_;
|
||||
FrameType frame_type_;
|
||||
|
||||
PsDecodeErrorMode decode_error_mode_;
|
||||
PsFrameBufferStateEnum state_;
|
||||
SrsRtpDecodeErrorMode decode_error_mode_;
|
||||
SrsRtpFrameBufferStateEnum state_;
|
||||
|
||||
uint16_t nackCount_;
|
||||
int64_t latestPacketTimeMs_;
|
||||
//uint16_t nackCount_;
|
||||
//int64_t latestPacketTimeMs_;
|
||||
|
||||
// The payload.
|
||||
uint8_t* _buffer;
|
||||
|
|
@ -281,20 +343,20 @@ private:
|
|||
size_t _length;
|
||||
};
|
||||
|
||||
class PsDecodingState {
|
||||
class SrsRtpDecodingState {
|
||||
public:
|
||||
PsDecodingState();
|
||||
~PsDecodingState();
|
||||
SrsRtpDecodingState();
|
||||
~SrsRtpDecodingState();
|
||||
// Check for old frame
|
||||
bool IsOldFrame(const SrsPsFrameBuffer* frame) const;
|
||||
bool IsOldFrame(const SrsRtpFrameBuffer* frame) const;
|
||||
// Check for old packet
|
||||
bool IsOldPacket(const VCMPacket* packet);
|
||||
// Check for frame continuity based on current decoded state. Use best method
|
||||
// possible, i.e. temporal info, picture ID or sequence number.
|
||||
bool ContinuousFrame(const SrsPsFrameBuffer* frame) const;
|
||||
void SetState(const SrsPsFrameBuffer* frame);
|
||||
void CopyFrom(const PsDecodingState& state);
|
||||
bool UpdateEmptyFrame(const SrsPsFrameBuffer* frame);
|
||||
bool ContinuousFrame(const SrsRtpFrameBuffer* frame) const;
|
||||
void SetState(const SrsRtpFrameBuffer* frame);
|
||||
void CopyFrom(const SrsRtpDecodingState& state);
|
||||
bool UpdateEmptyFrame(const SrsRtpFrameBuffer* frame);
|
||||
// Update the sequence number if the timestamp matches current state and the
|
||||
// sequence number is higher than the current one. This accounts for packets
|
||||
// arriving late.
|
||||
|
|
@ -309,64 +371,81 @@ public:
|
|||
bool full_sync() const;
|
||||
|
||||
private:
|
||||
void UpdateSyncState(const SrsPsFrameBuffer* frame);
|
||||
void UpdateSyncState(const SrsRtpFrameBuffer* frame);
|
||||
// Designated continuity functions
|
||||
//bool ContinuousPictureId(int picture_id) const;
|
||||
bool ContinuousSeqNum(uint16_t seq_num) const;
|
||||
//bool ContinuousLayer(int temporal_id, int tl0_pic_id) const;
|
||||
//bool UsingPictureId(const SrsPsFrameBuffer* frame) const;
|
||||
//bool UsingPictureId(const SrsRtpFrameBuffer* frame) const;
|
||||
|
||||
// Keep state of last decoded frame.
|
||||
// TODO(mikhal/stefan): create designated classes to handle these types.
|
||||
uint16_t sequence_num_;
|
||||
uint32_t time_stamp_;
|
||||
int picture_id_;
|
||||
int temporal_id_;
|
||||
int tl0_pic_id_;
|
||||
bool full_sync_; // Sync flag when temporal layers are used.
|
||||
bool in_initial_state_;
|
||||
|
||||
bool m_firstPacket;
|
||||
};
|
||||
|
||||
class SrsPsJitterBuffer
|
||||
// The time jitter correct for rtp.
|
||||
class SrsRtpTimeJitter
|
||||
{
|
||||
private:
|
||||
int64_t previous_timestamp;
|
||||
int64_t pts;
|
||||
int delta;
|
||||
public:
|
||||
SrsRtpTimeJitter();
|
||||
virtual ~SrsRtpTimeJitter();
|
||||
public:
|
||||
int64_t timestamp();
|
||||
srs_error_t correct(int64_t& ts);
|
||||
void reset();
|
||||
};
|
||||
|
||||
class SrsRtpJitterBuffer
|
||||
{
|
||||
public:
|
||||
SrsPsJitterBuffer(std::string key);
|
||||
virtual ~SrsPsJitterBuffer();
|
||||
SrsRtpJitterBuffer(std::string key);
|
||||
virtual ~SrsRtpJitterBuffer();
|
||||
|
||||
public:
|
||||
srs_error_t start();
|
||||
void Reset();
|
||||
PsFrameBufferEnum InsertPacket(const SrsPsRtpPacket &packet, char *buf, int size, bool* retransmitted);
|
||||
void ReleaseFrame(SrsPsFrameBuffer* frame);
|
||||
SrsRtpFrameBufferEnum InsertPacket2(const SrsRtpPacket2 &pkt, bool* retransmitted);
|
||||
SrsRtpFrameBufferEnum InsertPacket(uint16_t seq, uint32_t ts, bool maker, char *buf, int size,
|
||||
bool* retransmitted);
|
||||
void ReleaseFrame(SrsRtpFrameBuffer* frame);
|
||||
bool FoundFrame(uint32_t& time_stamp);
|
||||
bool GetPsFrame(char **buffer, int &buf_len, int &size, const uint32_t time_stamp);
|
||||
void SetDecodeErrorMode(PsDecodeErrorMode error_mode);
|
||||
void SetNackMode(PsNackMode mode,int64_t low_rtt_nack_threshold_ms,
|
||||
bool GetFrame(char **buffer, int &buf_len, int &size, bool &keyframe, const uint32_t time_stamp);
|
||||
void SetDecodeErrorMode(SrsRtpDecodeErrorMode error_mode);
|
||||
void SetNackMode(SrsRtpNackMode mode,int64_t low_rtt_nack_threshold_ms,
|
||||
int64_t high_rtt_nack_threshold_ms);
|
||||
void SetNackSettings(size_t max_nack_list_size,int max_packet_age_to_nack,
|
||||
int max_incomplete_time_ms);
|
||||
uint16_t* GetNackList(uint16_t* nack_list_size, bool* request_key_frame);
|
||||
void Flush();
|
||||
void ResetJittter();
|
||||
|
||||
bool isFirstKeyFrame;
|
||||
private:
|
||||
|
||||
PsFrameBufferEnum GetFrame(const VCMPacket& packet, SrsPsFrameBuffer** frame,
|
||||
SrsRtpFrameBufferEnum GetFrameByRtpPacket(const VCMPacket& packet, SrsRtpFrameBuffer** frame,
|
||||
FrameList** frame_list);
|
||||
SrsPsFrameBuffer* GetEmptyFrame();
|
||||
SrsRtpFrameBuffer* GetEmptyFrame();
|
||||
bool NextCompleteTimestamp(uint32_t max_wait_time_ms, uint32_t* timestamp);
|
||||
bool NextMaybeIncompleteTimestamp(uint32_t* timestamp);
|
||||
SrsPsFrameBuffer* ExtractAndSetDecode(uint32_t timestamp);
|
||||
SrsPsFrameBuffer* NextFrame() const;
|
||||
SrsRtpFrameBuffer* ExtractAndSetDecode(uint32_t timestamp);
|
||||
SrsRtpFrameBuffer* NextFrame() const;
|
||||
|
||||
|
||||
bool TryToIncreaseJitterBufferSize();
|
||||
bool RecycleFramesUntilKeyFrame();
|
||||
bool IsContinuous(const SrsPsFrameBuffer& frame) const;
|
||||
bool IsContinuousInState(const SrsPsFrameBuffer& frame,
|
||||
const PsDecodingState& decoding_state) const;
|
||||
void FindAndInsertContinuousFrames(const SrsPsFrameBuffer& new_frame);
|
||||
bool IsContinuous(const SrsRtpFrameBuffer& frame) const;
|
||||
bool IsContinuousInState(const SrsRtpFrameBuffer& frame,
|
||||
const SrsRtpDecodingState& decoding_state) const;
|
||||
void FindAndInsertContinuousFrames(const SrsRtpFrameBuffer& new_frame);
|
||||
void CleanUpOldOrEmptyFrames();
|
||||
|
||||
//nack
|
||||
|
|
@ -376,11 +455,14 @@ private:
|
|||
bool MissingTooOldPacket(uint16_t latest_sequence_number) const;
|
||||
bool HandleTooOldPackets(uint16_t latest_sequence_number);
|
||||
void DropPacketsFromNackList(uint16_t last_decoded_sequence_number);
|
||||
PsNackMode nack_mode() const;
|
||||
SrsRtpNackMode nack_mode() const;
|
||||
int NonContinuousOrIncompleteDuration();
|
||||
uint16_t EstimatedLowSequenceNumber(const SrsPsFrameBuffer& frame) const;
|
||||
uint16_t EstimatedLowSequenceNumber(const SrsRtpFrameBuffer& frame) const;
|
||||
bool WaitForRetransmissions();
|
||||
|
||||
bool IsPacketInOrder(uint16_t sequence_number);
|
||||
bool IsFirstPacketInFrame(uint32_t ts, uint16_t seq);
|
||||
|
||||
private:
|
||||
class SequenceNumberLessThan {
|
||||
public:
|
||||
|
|
@ -402,7 +484,7 @@ private:
|
|||
UnorderedFrameList free_frames_;
|
||||
FrameList decodable_frames_;
|
||||
FrameList incomplete_frames_;
|
||||
PsDecodingState last_decoded_state_;
|
||||
SrsRtpDecodingState last_decoded_state_;
|
||||
bool first_packet_since_reset_;
|
||||
|
||||
// Statistics.
|
||||
|
|
@ -438,7 +520,7 @@ private:
|
|||
int64_t rtt_ms_;
|
||||
|
||||
// NACK and retransmissions.
|
||||
PsNackMode nack_mode_;
|
||||
SrsRtpNackMode nack_mode_;
|
||||
int64_t low_rtt_nack_threshold_ms_;
|
||||
int64_t high_rtt_nack_threshold_ms_;
|
||||
// Holds the internal NACK list (the missing sequence numbers).
|
||||
|
|
@ -449,12 +531,17 @@ private:
|
|||
int max_packet_age_to_nack_; // Measured in sequence numbers.
|
||||
int max_incomplete_time_ms_;
|
||||
|
||||
PsDecodeErrorMode decode_error_mode_;
|
||||
SrsRtpDecodeErrorMode decode_error_mode_;
|
||||
// Estimated rolling average of packets per frame
|
||||
float average_packets_per_frame_;
|
||||
// average_packets_per_frame converges fast if we have fewer than this many
|
||||
// frames.
|
||||
int frame_counter_;
|
||||
|
||||
uint32_t last_received_timestamp_;
|
||||
uint16_t last_received_sequence_number_;
|
||||
bool first_packet_;
|
||||
|
||||
};
|
||||
|
||||
#endif
|
||||
|
|
@ -21,7 +21,7 @@
|
|||
* CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
|
||||
*/
|
||||
|
||||
#include <srs_sip_stack.hpp>
|
||||
#include <srs_app_sip.hpp>
|
||||
|
||||
#if !defined(SRS_EXPORT_LIBRTMP)
|
||||
|
||||
Loading…
Add table
Add a link
Reference in a new issue