diff --git a/.circleci/config.yml b/.circleci/config.yml
index 59be9a326..518e1bb57 100644
--- a/.circleci/config.yml
+++ b/.circleci/config.yml
@@ -24,14 +24,6 @@ jobs:
- run: |
echo "Build SRS without NASM or SRTP-NASM" &&
cd trunk && ./configure --nasm=off --srtp-nasm=off && make
- build-c7-gb28181:
- docker:
- - image: ossrs/srs:dev
- steps:
- - checkout
- - run: |
- echo "Build SRS with GB28181" &&
- cd trunk && ./configure --gb28181=on && make
build-c7-srt:
docker:
- image: ossrs/srs:dev
@@ -111,7 +103,7 @@ jobs:
- checkout
- run: |
echo "Build and run utest for SRS" &&
- cd trunk && ./configure --gb28181=on --utest=on --gcov=on && make &&
+ cd trunk && ./configure --utest=on --gcov=on && make &&
./objs/srs_utest && bash auto/codecov.sh
run-regression-test:
docker:
@@ -140,7 +132,6 @@ workflows:
- run-regression-test
- build-c7-nortc
- build-c7-noasm
- - build-c7-gb28181
- build-c7-srt
- build-c8-baseline
- build-c8-srt
diff --git a/CHANGELOG.md b/CHANGELOG.md
index 34d6cc689..e2120c59a 100644
--- a/CHANGELOG.md
+++ b/CHANGELOG.md
@@ -6,16 +6,14 @@ The changelog for SRS.
## SRS 4.0 Changelog
+* v4.0, 2021-06-16, Change [GB28181](https://github.com/ossrs/srs/issues/1500) to [feature/gb28181](https://github.com/ossrs/srs/tree/feature/gb28181). 4.0.127
* v4.0, 2021-06-01, Support --shared-ffmpeg to link with *.so for LGPL license. 4.0.126
* v4.0, 2021-06-01, Support --shared-srt to link with *.so for MPL license. 4.0.125
* v4.0, 2021-05-31, Use [SPDX-License-Identifier: MIT](https://spdx.dev/ids/). 4.0.124
-* v4.0, 2021-05-28, Fix bugs for GB28181 and RTC. 4.0.123
* v4.0, 2021-05-21, Fix [#2370][bug #2370] bug for Firefox play stream(published by Chrome). 4.0.121
* v4.0, 2021-05-21, RTC: Refine sdk, migrate from onaddstream to ontrack. 4.0.120
* v4.0, 2021-05-21, Tools: Refine configure options. 4.0.119
* v4.0, 2021-05-20, Fix build fail when disable RTC by --rtc=off. 4.0.118
-* v4.0, 2021-05-19, Fix [#2362][bug #2362]: Allow WebRTC to play before publishing, for GB28181 as such. 4.0.117
-* v4.0, 2021-05-18, Fix [#2355][bug #2355]: GB28181: Fix play by RTC bug. 4.0.116
* v4.0, 2021-05-15, SRT: Build SRT from source by SRS. 4.0.115
* v4.0, 2021-05-15, Rename SrsConsumer* to SrsLiveConsumer*. 4.0.114
* v4.0, 2021-05-15, Rename SrsRtcStream* to SrsRtcSource*. 4.0.113
@@ -63,7 +61,6 @@ The changelog for SRS.
* v4.0, 2021-01-08, HTML5 video tag resolution adaptive. 4.0.59
* v4.0, 2021-01-08, Fix memory leak and bugs for RTC. 4.0.58
* v4.0, 2021-01-06, Merge #2109, Refine srs_string_split.
-* v4.0, 2021-01-06, Merge #2109, Fix bugs for GB28181.
* v4.0, 2020-12-24, Support disable CherryPy. 4.0.57
* v4.0, 2020-11-12, For [#1998][bug #1998], Support Firefox, use PT in offer. 4.0.55
* v4.0, 2020-11-11, For [#1508][bug #1508], Transform http header name to upper camel case. 4.0.54
@@ -81,10 +78,8 @@ The changelog for SRS.
* v4.0, 2020-07-25, RTC: Support multiple address for client. 4.0.36
* v4.0, 2020-07-11, Refine log context with random string. 4.0.35
* v4.0, 2020-07-04, Fix some bugs for RTC. 4.0.34
-* v4.0, 2020-07-03, Merge [#1830][bug #1830] to fix bugs in GB28181. 4.0.33
* v4.0, 2020-06-24, Support static link c++ libraries. 4.0.32
* v4.0, 2020-06-23, Change log cid from int to string. 4.0.31
-* v4.0, 2020-06-13, GB28181 with JitterBuffer support. 4.0.30
* v4.0, 2020-06-03, Support enable C++11. 4.0.29
* v4.0, 2020-05-31, Remove [srs-librtmp](https://github.com/ossrs/srs/issues/1535#issuecomment-633907655). 4.0.28
* v4.0, 2020-05-21, For [#307][bug #307], disable GSO and sendmmsg. 4.0.27
@@ -93,10 +88,8 @@ The changelog for SRS.
* v4.0, 2020-04-30, For [#307][bug #307], support publish RTC with passing opus. 4.0.24
* v4.0, 2020-04-14, For [#307][bug #307], support sendmmsg, GSO and reuseport. 4.0.23
* v4.0, 2020-04-05, For [#307][bug #307], SRTP ASM only works with openssl-1.0, auto detect it. 4.0.22
-* v4.0, 2020-04-04, Merge RTC and GB28181, with bugs fixed. 4.0.21
* v4.0, 2020-04-04, For [#307][bug #307], refine RTC latency from 600ms to 200ms. 4.0.20
* v4.0, 2020-04-03, For [#307][bug #307], build SRTP with openssl to improve performance. 4.0.19
-* v4.0, 2020-03-31, For [#1500][bug #1500], support push stream by GB28181. 4.0.18
* v4.0, 2020-03-31, Play stream by WebRTC on iOS/Android/PC browser. 4.0.17
* v4.0, 2020-03-28, Support multiple OS/Platform build cache. 4.0.16
* v4.0, 2020-03-28, For [#1250][bug #1250], support macOS, OSX, MacbookPro, Apple Darwin.
diff --git a/README.md b/README.md
index 3b1bc90a9..390cf8b64 100755
--- a/README.md
+++ b/README.md
@@ -5,9 +5,9 @@
[](https://codecov.io/gh/ossrs/srs/branch/4.0release)
[](https://github.com/ossrs/srs/wiki/v4_CN_Contact#wechat)
-SRS/4.0，[Leo][release4]，是一个简单高效的实时视频服务器，支持RTMP/WebRTC/HLS/HTTP-FLV/SRT/GB28181。
+SRS/4.0，[Leo][release4]，是一个简单高效的实时视频服务器，支持RTMP/WebRTC/HLS/HTTP-FLV/SRT。
-SRS is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT/GB28181.
+SRS is a simple, high efficiency and realtime video server, supports RTMP/WebRTC/HLS/HTTP-FLV/SRT.
SRS is licenced under [MIT][LICENSE], but some depended libraries are distributed using their [own licenses][LicenseMixing].
@@ -69,7 +69,6 @@ Fast index for Wikis:
Other important wiki:
-* Usage: How to publish GB28181 to SRS? [#1500](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679)
* Usage: How to delivery DASH(Experimental)?([CN][v4_CN_SampleDASH], [EN][v4_EN_SampleDASH])
* Usage: How to transode RTMP stream by FFMPEG?([CN][v4_CN_SampleFFMPEG], [EN][v4_EN_SampleFFMPEG])
* Usage: How to delivery HTTP FLV Live Streaming Cluster?([CN][v4_CN_SampleHttpFlvCluster], [EN][v4_EN_SampleHttpFlvCluster])
@@ -121,9 +120,6 @@ For optional stream caster services, to push streams to SRS:
* udp://8935, Stream Caster: [Push MPEGTS over UDP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-mpeg-ts-over-udp) server.
* tcp://554, Stream Caster: [Push RTSP](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-rtsp-to-srs) server.
* tcp://8936, Stream Caster: [Push HTTP-FLV](https://github.com/ossrs/srs/wiki/v4_CN_Streamer#push-http-flv-to-srs) server.
-* tcp://5060, Stream Caster: [Push GB28181 SIP](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679) server.
-* udp://9000, Stream Caster: [Push GB28181 Media(bundle)](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679) server.
-* udp://58200-58300, Stream Caster: [Push GB28181 Media(no-bundle)](https://github.com/ossrs/srs/issues/1500#issuecomment-606695679) server.
* udp://10080, Stream Caster: [Push SRT Media](https://github.com/ossrs/srs/issues/1147#issuecomment-577469119) server.
For external services to work with SRS:
@@ -176,7 +172,7 @@ For external services to work with SRS:
- [x] [Experimental] Support transcode RTMP/AAC to WebRTC/Opus, [#307][bug #307].
- [x] [Experimental] Support AV1 codec for WebRTC, [#2324][bug #2324].
- [x] [Experimental] Enhance HTTP Stream Server for HTTP-FLV, HTTPS, HLS etc. [#1657][bug #1657].
-- [x] [Experimental] Support push stream by GB28181, [#1500][bug #1500].
+- [ ] Support push stream by GB28181, [#1500][bug #1500].
- [x] [Experimental] Support DVR in MP4 format, read [#738][bug #738].
- [x] [Experimental] Support MPEG-DASH, the future live streaming protocol, read [#299][bug #299].
- [x] [Experimental] Support pushing MPEG-TS over UDP, please read [bug #250][bug #250].
@@ -306,7 +302,7 @@ The stream architecture of SRS.
| MediaSource(2) | | |
| (MPEGTSoverUDP | | |
| HTTP-FLV, --push-+->- StreamCaster(4) -(rtmp)-+-> SRS |
-| GB28181,SRT, | | |
+| SRT, | | |
| ......) | | |
+----------------------+ | |
| FFMPEG --push(srt)--+->- SRTModule(5) ---(rtmp)-+-> SRS |
diff --git a/trunk/auto/auto_headers.sh b/trunk/auto/auto_headers.sh
index fa6a55e58..589861869 100755
--- a/trunk/auto/auto_headers.sh
+++ b/trunk/auto/auto_headers.sh
@@ -85,12 +85,6 @@ else
srs_undefine_macro "SRS_SIMULATOR" $SRS_AUTO_HEADERS_H
fi
-if [ $SRS_GB28181 = YES ]; then
- srs_define_macro "SRS_GB28181" $SRS_AUTO_HEADERS_H
-else
- srs_undefine_macro "SRS_GB28181" $SRS_AUTO_HEADERS_H
-fi
-
if [ $SRS_HTTPS = YES ]; then
srs_define_macro "SRS_HTTPS" $SRS_AUTO_HEADERS_H
else
diff --git a/trunk/auto/depends.sh b/trunk/auto/depends.sh
index 36510a7a1..309f1c66c 100755
--- a/trunk/auto/depends.sh
+++ b/trunk/auto/depends.sh
@@ -299,15 +299,6 @@ function OSX_prepare()
echo "Please install pkg-config"; exit -1;
fi
- if [[ $SRS_GB28181 == YES ]]; then
- if [[ ! -f /usr/local/opt/libiconv/lib/libiconv.a ]]; then
- echo "install libiconv"
- echo "brew install libiconv"
- brew install libiconv; ret=$?; if [[ 0 -ne $ret ]]; then return $ret; fi
- echo "install libiconv success"
- fi
- fi
-
if [[ $SRS_SRT == YES ]]; then
echo "SRT enable, install depend tools"
tclsh <<< "exit" >/dev/null 2>&1; ret=$?; if [[ 0 -ne $ret ]]; then
diff --git a/trunk/auto/options.sh b/trunk/auto/options.sh
index 2b989c895..574730a0c 100755
--- a/trunk/auto/options.sh
+++ b/trunk/auto/options.sh
@@ -6,7 +6,6 @@ help=no
SRS_HDS=NO
SRS_SRT=NO
SRS_RTC=RESERVED
-SRS_GB28181=NO
SRS_CXX11=NO
SRS_CXX14=NO
SRS_NGINX=NO
@@ -115,7 +114,6 @@ Features:
--utest=on|off Whether build the utest. Default: $(value2switch $SRS_UTEST)
--srt=on|off Whether build the SRT. Default: $(value2switch $SRS_SRT)
--rtc=on|off Whether build the WebRTC. Default: $(value2switch $SRS_RTC)
- --gb28181=on|off Whether build the GB28181. Default: $(value2switch $SRS_GB28181)
--cxx11=on|off Whether enable the C++11. Default: $(value2switch $SRS_CXX11)
--cxx14=on|off Whether enable the C++14. Default: $(value2switch $SRS_CXX14)
--ffmpeg-fit=on|off Whether enable the FFmpeg fit(source code). Default: $(value2switch $SRS_FFMPEG_FIT)
@@ -265,10 +263,6 @@ function parse_user_option() {
--simulator) SRS_SIMULATOR=$(switch2value $value) ;;
--ffmpeg-fit) SRS_FFMPEG_FIT=$(switch2value $value) ;;
- --with-gb28181) SRS_GB28181=YES ;;
- --without-gb28181) SRS_GB28181=NO ;;
- --gb28181) SRS_GB28181=$(switch2value $value) ;;
-
--cxx11) SRS_CXX11=$(switch2value $value) ;;
--cxx14) SRS_CXX14=$(switch2value $value) ;;
@@ -473,7 +467,6 @@ function regenerate_options() {
SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --srt=$(value2switch $SRS_SRT)"
SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --rtc=$(value2switch $SRS_RTC)"
SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --simulator=$(value2switch $SRS_SIMULATOR)"
- SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --gb28181=$(value2switch $SRS_GB28181)"
SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --cxx11=$(value2switch $SRS_CXX11)"
SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --cxx14=$(value2switch $SRS_CXX14)"
SRS_AUTO_CONFIGURE="${SRS_AUTO_CONFIGURE} --ffmpeg-fit=$(value2switch $SRS_FFMPEG_FIT)"
diff --git a/trunk/conf/full.conf b/trunk/conf/full.conf
index 27416b7f2..2a1d76d59 100644
--- a/trunk/conf/full.conf
+++ b/trunk/conf/full.conf
@@ -366,89 +366,6 @@ stream_caster {
listen 8936;
}
-# GB28181
-stream_caster {
- # whether stream caster is enabled.
- # default: off
- enabled on;
- # the caster type of stream, the casters:
- # gb28181, Push GB28181 to SRS.
- caster gb28181;
- # the output rtmp url.
- # for gb28181 caster, the typically output url:
- # rtmp://127.0.0.1/live/[stream]
- # where the [stream] is the VideoChannelCodecID.
- output rtmp://127.0.0.1/live/[stream];
- # the listen port for stream caster.
- # for gb28181 caster, listen at udp port. for example, 9000.
- # @remark We can bundle all gb28181 to this port, to reuse this port.
- # User can choose to bundle port in API port_mode or SIP invite_port_fixed.
- listen 9000;
- # Listen as TCP if on; otherwise, listen as UDP.
- # default: off
- tcp_enable off;
- # If not bundle ports, use specified ports for each stream.
- rtp_port_min 58200;
- rtp_port_max 58300;
- # Whether wait for keyframe then forward to RTMP.
- # default: on
- wait_keyframe on;
- # Max timeout in seconds for RTP stream, if timeout, RTCP bye and close stream.
- # default: 30
- rtp_idle_timeout 30;
- # Whether has audio.
- # @remark Flash/RTMP only supports 11025 22050 44100 sample rate, if not the audio may corrupt.
- # default: off
- audio_enable off;
- # The exposed IP to receive media stream.
- # * Retrieve server IP automatically, from all network interfaces.
- # eth0 Retrieve server IP by specified network interface name. # TODO: Implements it.
- # $CANDIDATE Read the IP from ENV variable $EIP, use * if not set, see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
- # x.x.x.x A specified IP address or DNS name, which can be access by client such as Chrome.
- # You can specific more than one interface name:
- # eth0 eth1 Use network interface eth0 and eth1. # TODO: Implements it.
- # Also by IP or DNS names:
- # 192.168.1.3 10.1.2.3 rtc.me # TODO: Implements it.
- # And by multiple ENV variables:
- # $CANDIDATE $EIP # TODO: Implements it.
- # default: *
- host *;
- #The media channel is automatically created according to the received RTP packet,
- # and the channel ID is generated according to the RTP SSRC
- # channelid format: 'chid[ssrc]' [ssrc] is rtp's ssrc
- auto_create_channel off;
-
- sip {
- # Whether enable embeded SIP server.
- # default: on
- enabled on;
- # The SIP listen port.
- # default: 5060
- listen 5060;
- # The SIP server ID.
- # default: 34020000002000000001
- serial 34020000002000000001;
- # The SIP server domain.
- # default: 3402000000
- realm 3402000000;
- # The SIP ACK response timeout in seconds.
- # default: 30
- ack_timeout 30;
- # The keepalive timeout in seconds.
- # default: 120
- keepalive_timeout 120;
- # Whether play immediately after registered.
- # default: on
- auto_play on;
- # Whether bundle media stream port.
- # default: on
- invite_port_fixed on;
- # interval to query equipment list from equipment or subordinate domain, unit(s)
- # default: 60
- query_catalog_interval 60;
- }
-}
-
#############################################################################################
# SRT server section
#############################################################################################
diff --git a/trunk/conf/push.gb28181.conf b/trunk/conf/push.gb28181.conf
deleted file mode 100644
index 0890dd94f..000000000
--- a/trunk/conf/push.gb28181.conf
+++ /dev/null
@@ -1,143 +0,0 @@
-# push gb28181 stream to SRS.
-
-listen 1935;
-max_connections 1000;
-daemon off;
-srs_log_tank console;
-
-http_api {
- enabled on;
- listen 1985;
-}
-
-http_server {
- enabled on;
- listen 8080;
-}
-
-stats {
- network 0;
-}
-
-stream_caster {
- enabled on;
- caster gb28181;
-
- # 转发流到rtmp服务器地址与端口
- # TODO: https://github.com/ossrs/srs/pull/1679/files#r400875104
- # [stream] is VideoChannelCodecID(视频通道编码ID) for sip
- # 自动创建的道通[stream] 是‘chid[ssrc]’ [ssrc]是rtp的ssrc
- # [ssrc] rtp中的ssrc
- output rtmp://127.0.0.1:1935/live/[stream];
-
- # 接收设备端rtp流的多路复用端口
- listen 9000;
- # 多路复用端口类型，on为tcp，off为udp
- # 默认：on
- tcp_enable on;
-
- # rtp接收监听端口范围，最小值
- rtp_port_min 58200;
- # rtp接收监听端口范围，最大值
- rtp_port_max 58300;
-
- # 是否等待关键帧之后，再转发，
- # off:不需等待，直接转发
- # on:等第一个关键帧后，再转发
- wait_keyframe on;
-
- # rtp包空闲等待时间，如果指定时间没有收到任何包
- # rtp监听连接自动停止，发送BYE命令
- rtp_idle_timeout 30;
-
- # 是否转发音频流
- # 目前只支持aac格式，所以需要设备支持aac格式
- # on:转发音频
- # off:不转发音频，只有视频
- # *注意*!!!:flv 只支持11025 22050 44100 三种
- # 如果设备端没有三种中任何一个，转发时为自动选择一种格式
- # 同时也会将adts的头封装在flv aac raw数据中
- # 这样的话播放器为自动通过adts头自动选择采样频率
- # 像ffplay, vlc都可以，但是flash是没有声音，
- # 因为flash,只支持11025 22050 44100
- audio_enable off;
-
- # 是否开启rtp缓冲
- # 开启之后能有效解决rtp乱序等问题
- # tcp模式建议关闭
- jitterbuffer_enable off;
-
- # 服务器主机号，可以域名或ip地址
- # 也就是设备端将媒体发送的地址，如果是服务器是内外网
- # 需要写外网地址，
- # 调用api创建stream session时返回ip地址也是host
- # $CANDIDATE 是系统环境变量，从环境变量获取地址，如果没有配置，用*
- # *代表指定stats network 的网卡号地址，如果没有配置network，默认则是第0号网卡地址
- # TODO: https://github.com/ossrs/srs/pull/1679/files#r400917594
- host $CANDIDATE;
-
- #根据收到ps rtp包自带创建rtmp媒体通道，不需要api接口创建
- #rtmp地址参数[stream] 就是通道id 格式chid[ssrc]
- auto_create_channel off;
-
- sip {
- # 是否启用srs内部sip信令
- # 为on信令走srs, off 只转发ps流
- enabled on;
-
- # sip监听udp端口
- listen 5060;
-
- # SIP server ID(SIP服务器ID).
- # 设备端配置编号需要与该值一致，否则无法注册
- serial 34020000002000000001;
-
- # SIP server domain(SIP服务器域)
- realm 3402000000;
-
- # 服务端发送ack后，接收回应的超时时间，单位为秒
- # 如果指定时间没有回应，认为失败
- ack_timeout 30;
-
- # 设备心跳维持时间，如果指定时间内(秒）没有接收一个心跳
- # 认为设备离线
- keepalive_timeout 120;
-
- # 注册之后是否自动给设备端发送invite
- # on: 是 off 不是，需要通过api控制
- auto_play on;
- # 设备将流发送的端口，是否固定
- # on 发送流到多路复用端口 如9000
- # off 自动从rtp_mix_port - rtp_max_port 之间的值中
- # 选一个可以用的端口
- invite_port_fixed on;
-
- # 向设备或下级域查询设备列表的间隔，单位(秒)
- # 默认60秒
- query_catalog_interval 60;
- }
-}
-
-rtc_server {
- enabled on;
- # Listen at udp://8000
- listen 8000;
- #
- # The $CANDIDATE means fetch from env, if not configed, use * as default.
- #
- # The * means retrieving server IP automatically, from all network interfaces,
- # @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
- candidate $CANDIDATE;
-}
-
-vhost __defaultVhost__ {
- rtc {
- enabled on;
- bframe discard;
- }
-
- http_remux {
- enabled on;
- mount [vhost]/[app]/[stream].flv;
- }
-}
diff --git a/trunk/conf/push.gb28181.tcp.conf b/trunk/conf/push.gb28181.tcp.conf
deleted file mode 100644
index 257195afd..000000000
--- a/trunk/conf/push.gb28181.tcp.conf
+++ /dev/null
@@ -1,143 +0,0 @@
-# push gb28181 stream to SRS.
-
-listen 1935;
-max_connections 1000;
-daemon off;
-srs_log_tank console;
-
-http_api {
- enabled on;
- listen 1985;
-}
-
-http_server {
- enabled on;
- listen 8080;
-}
-
-stats {
- network 0;
-}
-
-stream_caster {
- enabled on;
- caster gb28181;
-
- # 转发流到rtmp服务器地址与端口
- # TODO: https://github.com/ossrs/srs/pull/1679/files#r400875104
- # [stream] is VideoChannelCodecID(视频通道编码ID) for sip
- # 自动创建的道通[stream] 是‘chid[ssrc]’ [ssrc]是rtp的ssrc
- # [ssrc] rtp中的ssrc
- output rtmp://127.0.0.1:1935/live/[stream];
-
- # 接收设备端rtp流的多路复用端口
- listen 9000;
- # 多路复用端口类型，on为tcp，off为udp
- # 默认：off
- tcp_enable on;
-
- # rtp接收监听端口范围，最小值
- rtp_port_min 58200;
- # rtp接收监听端口范围，最大值
- rtp_port_max 58300;
-
- # 是否等待关键帧之后，再转发，
- # off:不需等待，直接转发
- # on:等第一个关键帧后，再转发
- wait_keyframe on;
-
- # rtp包空闲等待时间，如果指定时间没有收到任何包
- # rtp监听连接自动停止，发送BYE命令
- rtp_idle_timeout 30;
-
- # 是否转发音频流
- # 目前只支持aac格式，所以需要设备支持aac格式
- # on:转发音频
- # off:不转发音频，只有视频
- # *注意*!!!:flv 只支持11025 22050 44100 三种
- # 如果设备端没有三种中任何一个，转发时为自动选择一种格式
- # 同时也会将adts的头封装在flv aac raw数据中
- # 这样的话播放器为自动通过adts头自动选择采样频率
- # 像ffplay, vlc都可以，但是flash是没有声音，
- # 因为flash,只支持11025 22050 44100
- audio_enable off;
-
- # 是否开启rtp缓冲
- # 开启之后能有效解决rtp乱序等问题
- # tcp模式建议关闭
- jitterbuffer_enable off;
-
- # 服务器主机号，可以域名或ip地址
- # 也就是设备端将媒体发送的地址，如果是服务器是内外网
- # 需要写外网地址，
- # 调用api创建stream session时返回ip地址也是host
- # $CANDIDATE 是系统环境变量，从环境变量获取地址，如果没有配置，用*
- # *代表指定stats network 的网卡号地址，如果没有配置network，默认则是第0号网卡地址
- # TODO: https://github.com/ossrs/srs/pull/1679/files#r400917594
- host $CANDIDATE;
-
- #根据收到ps rtp包自带创建rtmp媒体通道，不需要api接口创建
- #rtmp地址参数[stream] 就是通道id 格式chid[ssrc]
- auto_create_channel off;
-
- sip {
- # 是否启用srs内部sip信令
- # 为on信令走srs, off 只转发ps流
- enabled on;
-
- # sip监听udp端口
- listen 5060;
-
- # SIP server ID(SIP服务器ID).
- # 设备端配置编号需要与该值一致，否则无法注册
- serial 34020000002000000001;
-
- # SIP server domain(SIP服务器域)
- realm 3402000000;
-
- # 服务端发送ack后，接收回应的超时时间，单位为秒
- # 如果指定时间没有回应，认为失败
- ack_timeout 30;
-
- # 设备心跳维持时间，如果指定时间内(秒）没有接收一个心跳
- # 认为设备离线
- keepalive_timeout 120;
-
- # 注册之后是否自动给设备端发送invite
- # on: 是 off 不是，需要通过api控制
- auto_play on;
- # 设备将流发送的端口，是否固定
- # on 发送流到多路复用端口 如9000
- # off 自动从rtp_mix_port - rtp_max_port 之间的值中
- # 选一个可以用的端口
- invite_port_fixed on;
-
- # 向设备或下级域查询设备列表的间隔，单位(秒)
- # 默认60秒
- query_catalog_interval 60;
- }
-}
-
-rtc_server {
- enabled on;
- # Listen at udp://8000
- listen 8000;
- #
- # The $CANDIDATE means fetch from env, if not configed, use * as default.
- #
- # The * means retrieving server IP automatically, from all network interfaces,
- # @see https://github.com/ossrs/srs/issues/307#issuecomment-599028124
- candidate $CANDIDATE;
-}
-
-vhost __defaultVhost__ {
- rtc {
- enabled on;
- bframe discard;
- }
-
- http_remux {
- enabled on;
- mount [vhost]/[app]/[stream].flv;
- }
-}
diff --git a/trunk/configure b/trunk/configure
index 20cafeeab..92cf591bb 100755
--- a/trunk/configure
+++ b/trunk/configure
@@ -171,11 +171,6 @@ if [[ $SRS_SRT == YES ]]; then
if [[ $SRS_SHARED_SRT == YES ]]; then LibSRTfile="-L${SRS_OBJS_DIR}/srt/lib -lsrt"; fi
fi
-# For iconv on macOS only, CentOS seems ok.
-if [[ $SRS_GB28181 == YES && $SRS_OSX == YES ]]; then
- LibIconvRoot="/usr/local/opt/libiconv/include"; LibIconvfile="/usr/local/opt/libiconv/lib/libiconv.a"
-fi
-
# the link options, always use static link
SrsLinkOptions="-ldl -lpthread";
if [[ $SRS_SSL == YES && $SRS_USE_SYS_SSL == YES ]]; then
@@ -287,13 +282,6 @@ fi
if [[ $SRS_FFMPEG_FIT == YES ]]; then
MODULE_FILES+=("srs_app_rtc_codec")
fi
-if [[ $SRS_GB28181 == YES ]]; then
- MODULE_FILES+=("srs_app_gb28181" "srs_app_gb28181_sip" "srs_app_gb28181_jitter")
-fi
-if [[ $SRS_GB28181 == YES ]]; then
- MODULE_FILES+=("srs_app_gb28181_stack")
- ModuleLibIncs+=(${LibIconvRoot})
-fi
DEFINES=""
# add each modules for app
@@ -373,9 +361,6 @@ fi
if [[ $SRS_SRT == YES ]]; then
ModuleLibFiles+=("${LibSRTfile[*]}")
fi
-if [[ $SRS_GB28181 == YES ]]; then
- ModuleLibFiles+=("${LibIconvfile[*]}")
-fi
# all depends objects
MODULE_OBJS="${CORE_OBJS[@]} ${KERNEL_OBJS[@]} ${PROTOCOL_OBJS[@]} ${APP_OBJS[@]} ${SERVER_OBJS[@]}"
ModuleLibIncs=(${SRS_OBJS_DIR} ${LibSTRoot} ${LibGperfRoot} ${LibSSLRoot})
diff --git a/trunk/research/players/rtc_player.html b/trunk/research/players/rtc_player.html
index c1e574e23..ab0ad3ea4 100644
--- a/trunk/research/players/rtc_player.html
+++ b/trunk/research/players/rtc_player.html
@@ -31,7 +31,7 @@
- GB28181
+
diff --git a/trunk/research/players/rtc_publisher.html b/trunk/research/players/rtc_publisher.html
index 3d1681051..6b102a289 100644
--- a/trunk/research/players/rtc_publisher.html
+++ b/trunk/research/players/rtc_publisher.html
@@ -31,7 +31,7 @@
- GB28181
+
diff --git a/trunk/research/players/srs_bwt.html b/trunk/research/players/srs_bwt.html
index 961609685..a2858241b 100644
--- a/trunk/research/players/srs_bwt.html
+++ b/trunk/research/players/srs_bwt.html
@@ -29,7 +29,6 @@
SRS会议
SRS测网速
VLC播放器
- SRS-GB28181
diff --git a/trunk/research/players/srs_chat.html b/trunk/research/players/srs_chat.html
index 3ce343549..abed015fc 100644
--- a/trunk/research/players/srs_chat.html
+++ b/trunk/research/players/srs_chat.html
@@ -28,7 +28,7 @@
SRS会议
SRS测网速
VLC播放器
- SRS-GB28181
+
diff --git a/trunk/research/players/srs_player.html b/trunk/research/players/srs_player.html
index 3881f1e9c..62b89489c 100755
--- a/trunk/research/players/srs_player.html
+++ b/trunk/research/players/srs_player.html
@@ -26,7 +26,7 @@
- GB28181
+
diff --git a/trunk/research/players/srs_player_deprecated.html b/trunk/research/players/srs_player_deprecated.html
index e0e773884..26f21fcfc 100755
--- a/trunk/research/players/srs_player_deprecated.html
+++ b/trunk/research/players/srs_player_deprecated.html
@@ -40,7 +40,7 @@
- GB28181
+
diff --git a/trunk/research/players/vlc.html b/trunk/research/players/vlc.html
index 893b280a6..4422301ef 100644
--- a/trunk/research/players/vlc.html
+++ b/trunk/research/players/vlc.html
@@ -25,7 +25,7 @@
SRS会议
SRS测网速
VLC播放器
- SRS-GB28181
+
diff --git a/trunk/src/app/srs_app_config.cpp b/trunk/src/app/srs_app_config.cpp
index 2558a2796..d6da1a435 100644
--- a/trunk/src/app/srs_app_config.cpp
+++ b/trunk/src/app/srs_app_config.cpp
@@ -262,11 +262,6 @@ bool srs_stream_caster_is_flv(string caster)
return caster == "flv";
}
-bool srs_stream_caster_is_gb28181(string caster)
-{
- return caster == "gb28181";
-}
-
bool srs_config_apply_filter(SrsConfDirective* dvr_apply, SrsRequest* req)
{
static bool DEFAULT = true;
@@ -4590,302 +4585,6 @@ int SrsConfig::get_stream_caster_rtp_port_max(SrsConfDirective* conf)
return ::atoi(conf->arg0().c_str());
}
-srs_utime_t SrsConfig::get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf)
-{
- static srs_utime_t DEFAULT = 30 * SRS_UTIME_SECONDS;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("rtp_idle_timeout");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return (srs_utime_t)(::atoi(conf->arg0().c_str()) * SRS_UTIME_SECONDS);
-}
-
-srs_utime_t SrsConfig::get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf)
-{
- static srs_utime_t DEFAULT = 30 * SRS_UTIME_SECONDS;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("sip");
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("ack_timeout");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return (srs_utime_t)(::atoi(conf->arg0().c_str()) * SRS_UTIME_SECONDS);
-}
-
-srs_utime_t SrsConfig::get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* conf)
-{
- static srs_utime_t DEFAULT = 120 * SRS_UTIME_SECONDS;
-
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("sip");
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("keepalive_timeout");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return (srs_utime_t)(::atoi(conf->arg0().c_str()) * SRS_UTIME_SECONDS);
-}
-
-string SrsConfig::get_stream_caster_gb28181_host(SrsConfDirective* conf)
-{
- static string DEFAULT = "*";
-
- conf = conf->get("host");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- string eip = srs_getenv(conf->arg0());
- if (!eip.empty()) {
- return eip;
- }
-
- // If configed as ENV, but no ENV set, use default value.
- if (srs_string_starts_with(conf->arg0(), "$")) {
- return DEFAULT;
- }
-
- return conf->arg0();
-}
-
-string SrsConfig::get_stream_caster_gb28181_serial(SrsConfDirective* conf)
-{
- static string DEFAULT = "34020000002000000001";
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("sip");
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("serial");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return conf->arg0();
-}
-
-string SrsConfig::get_stream_caster_gb28181_realm(SrsConfDirective* conf)
-{
- static string DEFAULT = "3402000000";
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("sip");
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("realm");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return conf->arg0();
-}
-
-bool SrsConfig::get_stream_caster_gb28181_audio_enable(SrsConfDirective* conf)
-{
- static bool DEFAULT = false;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("audio_enable");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return SRS_CONF_PERFER_FALSE(conf->arg0());
-}
-
-bool SrsConfig::get_stream_caster_gb28181_jitterbuffer_enable(SrsConfDirective* conf)
-{
- static bool DEFAULT = true;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("jitterbuffer_enable");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return SRS_CONF_PERFER_FALSE(conf->arg0());
-}
-
-bool SrsConfig::get_stream_caster_gb28181_wait_keyframe(SrsConfDirective* conf)
-{
- static bool DEFAULT = true;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("wait_keyframe");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return SRS_CONF_PERFER_FALSE(conf->arg0());
-}
-
-bool SrsConfig::get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf)
-{
- static bool DEFAULT = true;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("sip");
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("enabled");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return SRS_CONF_PERFER_FALSE(conf->arg0());
-}
-
-bool SrsConfig::get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf)
-{
- static bool DEFAULT = true;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("sip");
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("auto_play");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return SRS_CONF_PERFER_FALSE(conf->arg0());
-
-}
-
-int SrsConfig::get_stream_caster_gb28181_sip_listen(SrsConfDirective* conf)
-{
- static int DEFAULT = 5060;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("sip");
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("listen");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return ::atoi(conf->arg0().c_str());
-
-}
-
-bool SrsConfig::get_stream_caster_gb28181_sip_invite_port_fixed(SrsConfDirective* conf)
-{
- static bool DEFAULT = true;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("sip");
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("invite_port_fixed");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return SRS_CONF_PERFER_FALSE(conf->arg0());
-
-}
-
-bool SrsConfig::get_stream_caster_gb28181_auto_create_channel(SrsConfDirective* conf)
-{
- static bool DEFAULT = false;
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("auto_create_channel");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return SRS_CONF_PERFER_FALSE(conf->arg0());
-}
-
-srs_utime_t SrsConfig::get_stream_caster_gb28181_sip_query_catalog_interval(SrsConfDirective* conf)
-{
- static srs_utime_t DEFAULT = 60 * SRS_UTIME_SECONDS;
-
-
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("sip");
- if (!conf) {
- return DEFAULT;
- }
-
- conf = conf->get("query_catalog_interval");
- if (!conf || conf->arg0().empty()) {
- return DEFAULT;
- }
-
- return (srs_utime_t)(::atoi(conf->arg0().c_str()) * SRS_UTIME_SECONDS);
-}
-
bool SrsConfig::get_rtc_server_enabled()
{
SrsConfDirective* conf = root->get("rtc_server");
diff --git a/trunk/src/app/srs_app_config.hpp b/trunk/src/app/srs_app_config.hpp
index 0a51183c3..502ce9f65 100644
--- a/trunk/src/app/srs_app_config.hpp
+++ b/trunk/src/app/srs_app_config.hpp
@@ -102,7 +102,6 @@ extern bool srs_config_dvr_is_plan_session(std::string plan);
extern bool srs_stream_caster_is_udp(std::string caster);
extern bool srs_stream_caster_is_rtsp(std::string caster);
extern bool srs_stream_caster_is_flv(std::string caster);
-extern bool srs_stream_caster_is_gb28181(std::string caster);
// Whether the dvr_apply active the stream specified by req.
extern bool srs_config_apply_filter(SrsConfDirective* dvr_apply, SrsRequest* req);
@@ -502,22 +501,6 @@ class SrsConfig
// Get the max udp port for rtp of stream caster rtsp.
virtual int get_stream_caster_rtp_port_max(SrsConfDirective* conf);
- virtual srs_utime_t get_stream_caster_gb28181_rtp_idle_timeout(SrsConfDirective* conf);
- virtual srs_utime_t get_stream_caster_gb28181_ack_timeout(SrsConfDirective* conf);
- virtual srs_utime_t get_stream_caster_gb28181_keepalive_timeout(SrsConfDirective* conf);
- virtual bool get_stream_caster_gb28181_audio_enable(SrsConfDirective* conf);
- virtual bool get_stream_caster_gb28181_jitterbuffer_enable(SrsConfDirective* conf);
- virtual std::string get_stream_caster_gb28181_host(SrsConfDirective* conf);
- virtual std::string get_stream_caster_gb28181_serial(SrsConfDirective* conf);
- virtual std::string get_stream_caster_gb28181_realm(SrsConfDirective* conf);
- virtual bool get_stream_caster_gb28181_wait_keyframe(SrsConfDirective* conf);
- virtual bool get_stream_caster_gb28181_sip_enable(SrsConfDirective* conf);
- virtual bool get_stream_caster_gb28181_sip_auto_play(SrsConfDirective* conf);
- virtual int get_stream_caster_gb28181_sip_listen(SrsConfDirective* conf);
- virtual bool get_stream_caster_gb28181_sip_invite_port_fixed(SrsConfDirective* conf);
- virtual bool get_stream_caster_gb28181_auto_create_channel(SrsConfDirective* conf);
- virtual srs_utime_t get_stream_caster_gb28181_sip_query_catalog_interval(SrsConfDirective* conf);
-
// rtc section
public:
virtual bool get_rtc_server_enabled();
diff --git a/trunk/src/app/srs_app_gb28181.cpp b/trunk/src/app/srs_app_gb28181.cpp
deleted file mode 100644
index 5329db0d9..000000000
--- a/trunk/src/app/srs_app_gb28181.cpp
+++ /dev/null
@@ -1,2685 +0,0 @@
-//
-// Copyright (c) 2013-2021 Lixin
-//
-// SPDX-License-Identifier: MIT
-//
-
-#include
-#include
-#include
-#include
-#include
-
-using namespace std;
-
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-#include
-
-//#define W_PS_FILE
-//#define W_VIDEO_FILE
-//#define W_AUDIO_FILE
-//#define W_UNKONW_FILE
-
-SrsPsRtpPacket::SrsPsRtpPacket()
-{
- isFirstPacket = false;
-}
-
-SrsPsRtpPacket::~SrsPsRtpPacket()
-{
-}
-
-srs_error_t SrsPsRtpPacket::decode(SrsBuffer* stream)
-{
- srs_error_t err = srs_success;
-
- // 12bytes header
- if (!stream->require(12)) {
- return srs_error_new(ERROR_RTP_HEADER_CORRUPT, "requires 12 only %d bytes", stream->left());
- }
-
- int8_t vv = stream->read_1bytes();
- version = (vv >> 6) & 0x03;
- padding = (vv >> 5) & 0x01;
- extension = (vv >> 4) & 0x01;
- csrc_count = vv & 0x0f;
-
- int8_t mv = stream->read_1bytes();
- marker = (mv >> 7) & 0x01;
- payload_type = mv & 0x7f;
-
- sequence_number = stream->read_2bytes();
- timestamp = stream->read_4bytes();
- ssrc = stream->read_4bytes();
-
- // TODO: FIXME: check sequence number.
-
- // video codec.
- if (payload_type == 96) {
- // ps stream playload atleast 1bytes content.
- if (!stream->require(1)) {
- return srs_error_new(ERROR_RTP_TYPE96_CORRUPT, "requires 1 only %d bytes", stream->left());
- }
- // append left bytes to payload.
- payload->append(stream->data() + stream->pos() , stream->size()-stream->pos());
- }
- return err;
-}
-
-//SrsPsRtpListener
-SrsPsRtpListener::SrsPsRtpListener(SrsGb28181Config* c, int p, std::string s)
-{
- rtp_processor = new SrsGb28181PsRtpProcessor(c, s);
- _port = p;
- // TODO: support listen at <[ip:]port>
- listener = new SrsUdpListener(this, srs_any_address_for_listener(), p);
-}
-
-SrsPsRtpListener::~SrsPsRtpListener()
-{
- srs_freep(listener);
- srs_freep(rtp_processor);
-}
-
-int SrsPsRtpListener::port()
-{
- return _port;
-}
-
-srs_error_t SrsPsRtpListener::listen()
-{
- return listener->listen();
-}
-
-srs_error_t SrsPsRtpListener::on_udp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf){
- srs_error_t err = srs_success;
- if (rtp_processor){
- err = rtp_processor->on_udp_packet(from, fromlen, buf, nb_buf);
- }
- return err;
-}
-
-//SrsGb28181RtpMuxService
-SrsGb28181RtpMuxService::SrsGb28181RtpMuxService(SrsConfDirective* c)
-{
- config = new SrsGb28181Config(c);
- rtp_processor = new SrsGb28181PsRtpProcessor(config,"");
-}
-
-SrsGb28181RtpMuxService::~SrsGb28181RtpMuxService()
-{
- srs_freep(config);
- srs_freep(rtp_processor);
-}
-
-srs_error_t SrsGb28181RtpMuxService::on_udp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf){
- srs_error_t err = srs_success;
- if (rtp_processor){
- err = rtp_processor->on_udp_packet(from, fromlen, buf, nb_buf);
- }
- return err;
-}
-
-//SrsGb28181PsRtpProcessor
-SrsGb28181PsRtpProcessor::SrsGb28181PsRtpProcessor(SrsGb28181Config* c, std::string id)
-{
- config = c;
- pprint = SrsPithyPrint::create_caster();
- channel_id = id;
-}
-
-SrsGb28181PsRtpProcessor::~SrsGb28181PsRtpProcessor()
-{
- dispose();
- srs_freep(pprint);
-}
-
-void SrsGb28181PsRtpProcessor::dispose()
-{
- map::iterator it2;
- for (it2 = cache_ps_rtp_packet.begin(); it2 != cache_ps_rtp_packet.end(); ++it2) {
- srs_freep(it2->second);
- }
- cache_ps_rtp_packet.clear();
-
- clear_pre_packet();
-
- return;
-}
-
-void SrsGb28181PsRtpProcessor::clear_pre_packet()
-{
- map::iterator it;
- for (it = pre_packet.begin(); it != pre_packet.end(); ++it) {
- srs_freep(it->second);
- }
- pre_packet.clear();
-}
-
-srs_error_t SrsGb28181PsRtpProcessor::on_udp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf)
-{
- if (config->jitterbuffer_enable){
- return on_rtp_packet_jitter(from, fromlen, buf, nb_buf);
- }else{
- return on_rtp_packet(from, fromlen, buf, nb_buf);
- }
-}
-srs_error_t SrsGb28181PsRtpProcessor::on_tcp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf)
-{
- on_udp_packet(from, fromlen, buf, nb_buf);
-}
-
-
-srs_error_t SrsGb28181PsRtpProcessor::on_rtp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf)
-{
- srs_error_t err = srs_success;
- bool completed = false;
-
- pprint->elapse();
-
- char address_string[64];
- char port_string[16];
- if (getnameinfo(from, fromlen,
- (char*)&address_string, sizeof(address_string),
- (char*)&port_string, sizeof(port_string),
- NI_NUMERICHOST|NI_NUMERICSERV)){
- // return srs_error_new(ERROR_SYSTEM_IP_INVALID, "bad address");
- srs_warn("gb28181 ps rtp: bad address");
- return srs_success;
- }
-
- int peer_port = atoi(port_string);
-
- if (true) {
- SrsBuffer stream(buf, nb_buf);
- SrsPsRtpPacket pkt;
-
- if ((err = pkt.decode(&stream)) != srs_success) {
- // return srs_error_wrap(err, "ps rtp decode error");
- srs_warn("gb28181 ps rtp: decode error");
- srs_freep(err);
- return srs_success;
- }
-
- //TODO: fixme: the same device uses the same SSRC to send with different local ports
- std::stringstream ss;
- ss << pkt.ssrc << ":" << pkt.timestamp << ":" << port_string;
- std::string pkt_key = ss.str();
-
- std::stringstream ss2;
- ss2 << pkt.ssrc << ":" << port_string;
- std::string pre_pkt_key = ss2.str();
-
- if (pre_packet.find(pre_pkt_key) == pre_packet.end()){
- pre_packet[pre_pkt_key] = new SrsPsRtpPacket();
- pre_packet[pre_pkt_key]->copy(&pkt);
- }
- //cache pkt by ssrc and timestamp
- if (cache_ps_rtp_packet.find(pkt_key) == cache_ps_rtp_packet.end()) {
- cache_ps_rtp_packet[pkt_key] = new SrsPsRtpPacket();
- }
-
- //get previous timestamp by ssrc
- uint32_t pre_timestamp = pre_packet[pre_pkt_key]->timestamp;
- uint32_t pre_sequence_number = pre_packet[pre_pkt_key]->sequence_number;
-
- //TODO: check sequence number out of order
- //it may be out of order, or multiple streaming ssrc are the same
- if (((pre_sequence_number + 1) % 65536) != pkt.sequence_number &&
- pre_sequence_number != pkt.sequence_number){
- srs_warn("gb28181: ps sequence_number out of order, ssrc=%#x, pre=%u, cur=%u, peer(%s, %s)",
- pkt.ssrc, pre_sequence_number, pkt.sequence_number, address_string, port_string);
- //return err;
- }
-
- //copy header to cache
- cache_ps_rtp_packet[pkt_key]->copy(&pkt);
- //accumulate one frame of data, to payload cache
- cache_ps_rtp_packet[pkt_key]->payload->append(pkt.payload);
-
- //detect whether it is a completed frame
- if (pkt.marker) {// rtp maker is true, is a completed frame
- completed = true;
- }else if (pre_timestamp != pkt.timestamp){
- //current timestamp is different from previous timestamp
- //previous timestamp, is a completed frame
- std::stringstream ss;
- ss << pkt.ssrc << ":" << pre_timestamp << ":" << port_string;
- pkt_key = ss.str();
- if (cache_ps_rtp_packet.find(pkt_key) != cache_ps_rtp_packet.end()) {
- completed = true;
- }
- }
-
- if (pprint->can_print()) {
- srs_trace("<- " SRS_CONSTS_LOG_GB28181_CASTER " gb28181: client_id %s, peer(%s, %d) ps rtp packet %dB, age=%d, vt=%d/%u, sts=%u/%u/%#x, paylod=%dB",
- channel_id.c_str(), address_string, peer_port, nb_buf, pprint->age(), pkt.version,
- pkt.payload_type, pkt.sequence_number, pkt.timestamp, pkt.ssrc,
- pkt.payload->length()
- );
- }
-
- //current packet becomes previous packet
- srs_freep(pre_packet[pre_pkt_key]);
- pre_packet[pre_pkt_key] = new SrsPsRtpPacket();
- pre_packet[pre_pkt_key]->copy(&pkt);;
-
- if (!completed){
- return err;
- }
- //process completed frame data
- //clear processed one ps frame
- //on completed frame data rtp packet in muxer enqueue
- map::iterator key = cache_ps_rtp_packet.find(pkt_key);
- if(key != cache_ps_rtp_packet.end())
- {
- SrsGb28181RtmpMuxer* muxer = NULL;
- muxer = fetch_rtmpmuxer(channel_id,pkt.ssrc);
-
- if (muxer){
- //TODO: fixme: the same device uses the same SSRC to send with different local ports
- //record the first peer port
- muxer->set_channel_peer_port(peer_port);
- muxer->set_channel_peer_ip(address_string);
- //not the first peer port's non processing
- if (muxer->channel_peer_port() != peer_port){
- srs_warn("<- " SRS_CONSTS_LOG_GB28181_CASTER " gb28181: client_id %s, ssrc=%#x, first peer_port=%d cur peer_port=%d",
- muxer->get_channel_id().c_str(), pkt.ssrc, muxer->channel_peer_port(), peer_port);
- srs_freep(key->second);
- }else {
- //put it in queue, wait for consumer to process, and then free
- muxer->ps_packet_enqueue(key->second);
- }
- }else{
- //no consumer process it, discarded
- srs_freep(key->second);
- }
- cache_ps_rtp_packet.erase(pkt_key);
- }
- }
- return err;
-}
-
-SrsGb28181RtmpMuxer* SrsGb28181PsRtpProcessor::fetch_rtmpmuxer(std::string channel_id, uint32_t ssrc)
-{
- if(true){
- SrsGb28181RtmpMuxer* muxer = NULL;
- //First, search according to the channel_id. Otherwise, search according to the SSRC.
- //Some channel_id are created by RTP pool, which are different ports.
- //No channel_id are created by multiplexing ports, which are the same port
- if (!channel_id.empty()){
- muxer = _srs_gb28181->fetch_rtmpmuxer(channel_id);
- }else {
- muxer = _srs_gb28181->fetch_rtmpmuxer_by_ssrc(ssrc);
- }
-
- //auto crate channel
- if (!muxer && config->auto_create_channel){
- //auto create channel generated id
- std::stringstream ss, ss1;
- ss << "chid" << ssrc;
- std::string tmp_id = ss.str();
-
- SrsGb28181StreamChannel channel;
- channel.set_channel_id(tmp_id);
- // channel.set_port_mode(RTP_PORT_MODE_FIXED);
- if (!config->sip_invite_port_fixed) {
- channel.set_port_mode(RTP_PORT_MODE_RANDOM);
- }else
- {
- channel.set_port_mode(RTP_PORT_MODE_FIXED);
- }
- channel.set_ssrc(ssrc);
-
- srs_error_t err2 = srs_success;
- if ((err2 = _srs_gb28181->create_stream_channel(&channel)) != srs_success){
- srs_warn("gb28181: RtpProcessor create stream channel error %s", srs_error_desc(err2).c_str());
- srs_error_reset(err2);
- };
-
- muxer = _srs_gb28181->fetch_rtmpmuxer(tmp_id);
- }
-
- return muxer;
- }//end if FoundFrame
-}
-
-srs_error_t SrsGb28181PsRtpProcessor::rtmpmuxer_enqueue_data(SrsGb28181RtmpMuxer *muxer, uint32_t ssrc,
- int peer_port, std::string address_string, SrsPsRtpPacket *pkt)
-{
- srs_error_t err = srs_success;
-
- if (!muxer)
- return err;
-
- if (muxer){
- //TODO: fixme: the same device uses the same SSRC to send with different local ports
- //record the first peer port
- muxer->set_channel_peer_port(peer_port);
- muxer->set_channel_peer_ip(address_string);
- //not the first peer port's non processing
- if (muxer->channel_peer_port() != peer_port){
- srs_warn("<- " SRS_CONSTS_LOG_GB28181_CASTER " gb28181: client_id %s, ssrc=%#x, first peer_port=%d cur peer_port=%d",
- muxer->get_channel_id().c_str(), ssrc, muxer->channel_peer_port(), peer_port);
- }else {
- //muxer->ps_packet_enqueue(pkt);
- muxer->insert_jitterbuffer(pkt);
- }//end if (muxer->channel_peer_port() != peer_port)
- }//end if (muxer)
-
- return err;
-}
-
-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;
-
- pprint->elapse();
-
- char address_string[64];
- char port_string[16];
- if (getnameinfo(from, fromlen,
- (char*)&address_string, sizeof(address_string),
- (char*)&port_string, sizeof(port_string),
- NI_NUMERICHOST|NI_NUMERICSERV)){
- // return srs_error_new(ERROR_SYSTEM_IP_INVALID, "bad address");
- srs_warn("gb28181 ps rtp: bad address");
- return srs_success;
- }
-
- int peer_port = atoi(port_string);
-
- if (true) {
- SrsBuffer stream(buf, nb_buf);
- SrsPsRtpPacket *pkt = new SrsPsRtpPacket();;
-
- if ((err = pkt->decode(&stream)) != srs_success) {
- srs_freep(pkt);
- // return srs_error_wrap(err, "ps rtp decode error");
- srs_warn("gb28181 ps rtp: decode error");
- srs_freep(err);
- return srs_success;
- }
-
- std::stringstream ss3;
- ss3 << pkt->ssrc << ":" << port_string;
- std::string jitter_key = ss3.str();
-
- pkt->completed = pkt->marker;
-
-
- if (pprint->can_print()) {
- srs_trace("<- " SRS_CONSTS_LOG_GB28181_CASTER " gb28181: client_id %s, peer(%s, %d) ps rtp packet %dB, age=%d, vt=%d/%u, sts=%u/%u/%#x, paylod=%dB",
- channel_id.c_str(), address_string, peer_port, nb_buf, pprint->age(), pkt->version,
- pkt->payload_type, pkt->sequence_number, pkt->timestamp, pkt->ssrc,
- pkt->payload->length()
- );
- }
-
- SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(channel_id, pkt->ssrc);
- if (muxer){
- rtmpmuxer_enqueue_data(muxer, pkt->ssrc, peer_port, address_string, pkt);
- }
-
- SrsAutoFree(SrsPsRtpPacket, pkt);
- }
-
- return err;
-}
-
-
-//ISrsPsStreamHander ps stream raw video/audio hander interface
-ISrsPsStreamHander::ISrsPsStreamHander()
-{
-}
-
-ISrsPsStreamHander::~ISrsPsStreamHander()
-{
-}
-
-//SrsPsStreamDemixer ps stream parse to h264/aac
-SrsPsStreamDemixer::SrsPsStreamDemixer(ISrsPsStreamHander *h, std::string id, bool a, bool k)
-{
- hander = h;
- audio_enable = a;
- wait_first_keyframe = k;
- channel_id = id;
- first_keyframe_flag = false;
-
- video_es_id = 0;
- video_es_type = 0;
- audio_es_id = 0;
- audio_es_type = 0;
- audio_check_aac_try_count = 0;
-
- aac = new SrsRawAacStream();
-}
-
-SrsPsStreamDemixer::~SrsPsStreamDemixer()
-{
- srs_freep(aac);
-}
-
-bool SrsPsStreamDemixer::can_send_ps_av_packet(){
- if (!wait_first_keyframe)
- return true;
-
- if (first_keyframe_flag)
- return true;
-
- return false;
-}
-
-std::string SrsPsStreamDemixer::get_ps_map_type_str(uint8_t type)
-{
- switch(type){
- case STREAM_TYPE_VIDEO_MPEG1: //0x01
- return "mpeg1";
- case STREAM_TYPE_VIDEO_MPEG2:// 0x02
- return "mpeg2";
- case STREAM_TYPE_AUDIO_MPEG1:// 0x03
- return "mpeg1";
- case STREAM_TYPE_AUDIO_MPEG2:// 0x04
- return "mpeg2";
- case STREAM_TYPE_PRIVATE_SECTION:// 0x05
- return "private_section";
- case STREAM_TYPE_PRIVATE_DATA:// 0x06
- return "private_data";
- case STREAM_TYPE_AUDIO_AAC:// 0x0f
- return "aac";
- case STREAM_TYPE_VIDEO_MPEG4:// 0x10
- return "mpeg4";
- case STREAM_TYPE_VIDEO_H264:// 0x1b
- return "h264";
- case STREAM_TYPE_VIDEO_HEVC:// 0x24
- return "hevc";
- case STREAM_TYPE_VIDEO_CAVS:// 0x42
- return "cavs";
- case STREAM_TYPE_VIDEO_SAVC:// 0x80
- return "savc";
-
- case STREAM_TYPE_AUDIO_AC3:// 0x81
- return "ac3";
-
- case STREAM_TYPE_AUDIO_G711:// 0x90
- return "g711";
- case STREAM_TYPE_AUDIO_G711ULAW:// 0x91
- return "g711ulaw";
- case STREAM_TYPE_AUDIO_G722_1:// 0x92
- return "g722_1";
- case STREAM_TYPE_AUDIO_G723_1:// 0x93
- return "g723_1";
- case STREAM_TYPE_AUDIO_G726:// 0x96
- return "g726";
- case STREAM_TYPE_AUDIO_G729_1:// 0x99
- return "g729_1";
- case STREAM_TYPE_AUDIO_SVAC:// 0x9b
- return "svac";
- case STREAM_TYPE_AUDIO_PCM:// 0x9c
- return "pcm";
- default:
- return "unknow";
- }
-}
-
-int64_t SrsPsStreamDemixer::parse_ps_timestamp(const uint8_t* p)
-{
- unsigned long b;
- //total 33 bits
- unsigned long val, val2, val3;
-
- //1st byte, 5、6、7 bit
- b = *p++;
- val = (b & 0x0e);
-
- //2 byte, all bit
- b = (*(p++)) << 8;
- //3 bytes 1--7 bit
- b += *(p++);
- val2 = (b & 0xfffe) >> 1;
-
- //4 byte, all bit
- b = (*(p++)) << 8;
- //5 byte 1--7 bit
- b += *(p++);
- val3 = (b & 0xfffe) >> 1;
-
- //<32--val--30> <29----val2----15> <14----val3----0>
- val = (val << 29) | (val2 << 15) | val3;
- return val;
-}
-
-srs_error_t SrsPsStreamDemixer::on_ps_stream(char* ps_data, int ps_size, uint32_t timestamp, uint32_t ssrc)
-{
- srs_error_t err = srs_success;
- int complete_len = 0;
- int incomplete_len = ps_size;
- char *next_ps_pack = ps_data;
-
- SrsSimpleStream video_stream;
- SrsSimpleStream audio_stream;
- uint64_t audio_pts = 0;
- uint64_t video_pts = 0;
- int pse_index = 0;
-
-#ifdef W_PS_FILE
- if (!ps_fw.is_open()) {
- std::string filename = "test_ps_" + channel_id + ".mpg";
- ps_fw.open(filename.c_str());
- }
- ps_fw.write(ps_data, ps_size, NULL);
-#endif
-
- while(incomplete_len > 0
- && incomplete_len >= sizeof(SrsPsPacketStartCode))
- {
- if (next_ps_pack
- && next_ps_pack[0] == (char)0x00
- && next_ps_pack[1] == (char)0x00
- && next_ps_pack[2] == (char)0x01
- && next_ps_pack[3] == (char)0xBA)
- {
- //ps header
- SrsPsPacketHeader *head = (SrsPsPacketHeader *)next_ps_pack;
- unsigned char pack_stuffing_length = head->stuffing_length & 0x07;
-
- next_ps_pack = next_ps_pack + sizeof(SrsPsPacketHeader) + pack_stuffing_length;
- complete_len = complete_len + sizeof(SrsPsPacketHeader) + pack_stuffing_length;
- incomplete_len = ps_size - complete_len;
- }
- else if(next_ps_pack
- && next_ps_pack[0] == (char)0x00
- && next_ps_pack[1] == (char)0x00
- && next_ps_pack[2] == (char)0x01
- && next_ps_pack[3] == (char)0xBB)
- {
- //ps system header
- SrsPsPacketBBHeader *bbhead=(SrsPsPacketBBHeader *)(next_ps_pack);
- int bbheaderlen = htons(bbhead->length);
- next_ps_pack = next_ps_pack + sizeof(SrsPsPacketBBHeader) + bbheaderlen;
- complete_len = complete_len + sizeof(SrsPsPacketBBHeader) + bbheaderlen;
- incomplete_len = ps_size - complete_len;
-
- first_keyframe_flag = true;
- }
- else if(next_ps_pack
- && next_ps_pack[0] == (char)0x00
- && next_ps_pack[1] == (char)0x00
- && next_ps_pack[2] == (char)0x01
- && next_ps_pack[3] == (char)0xBC)
- {
- //program stream map
-
- SrsPsMapPacket* psmap_pack = (SrsPsMapPacket*)next_ps_pack;
- psmap_pack->length = htons(psmap_pack->length);
-
- next_ps_pack = next_ps_pack + psmap_pack->length + sizeof(SrsPsMapPacket);
- complete_len = complete_len + psmap_pack->length + sizeof(SrsPsMapPacket);
- incomplete_len = ps_size - complete_len;
-
- //parse ps map
- uint16_t psm_length=0, ps_info_length=0, es_map_length=0;
- char *p = (char*)psmap_pack + sizeof(SrsPsMapPacket);
-
- SrsBuffer buf(p, (int)psmap_pack->length);
-
- psm_length =(int)psmap_pack->length;
- buf.read_1bytes();
- buf.read_1bytes();
-
- ps_info_length = buf.read_2bytes();
-
- /* skip program_stream_info */
- buf.skip(ps_info_length);
- /*es_map_length = */buf.read_2bytes();
- /* Ignore es_map_length, trust psm_length */
- es_map_length = psm_length - ps_info_length - 10;
-
- // /* at least one es available? */
- while (es_map_length >= 4) {
- uint8_t type = buf.read_1bytes();
- uint8_t es_id = buf.read_1bytes();
- uint16_t es_info_length = buf.read_2bytes();
- std::string s_type = get_ps_map_type_str(type);
-
- /* remember mapping from stream id to stream type */
- if (es_id >= PS_AUDIO_ID && es_id <= PS_AUDIO_ID_END){
- if (audio_es_type != type){
- srs_trace("gb28181: ps map audio es_type=%s(%x), es_id=%0x, es_info_length=%d",
- s_type.c_str(), type, es_id, es_info_length);
- }
-
- audio_es_id = es_id;
- audio_es_type = type;
- }else if (es_id >= PS_VIDEO_ID && es_id <= PS_VIDEO_ID_END){
-
- if (video_es_type != type){
- srs_trace("gb28181: ps map video es_type=%s(%x), es_id=%0x, es_info_length=%d",
- s_type.c_str(), type, es_id, es_info_length);
- }
-
- video_es_id = es_id;
- video_es_type = type;
- }
-
- /* skip program_stream_info */
- buf.skip(es_info_length);
- es_map_length -= 4 + es_info_length;
- }
-
- }
- else if(next_ps_pack
- && next_ps_pack[0] == (char)0x00
- && next_ps_pack[1] == (char)0x00
- && next_ps_pack[2] == (char)0x01
- && next_ps_pack[3] == (char)0xE0)
- {
- //pse video stream
- SrsPsePacket* pse_pack = (SrsPsePacket*)next_ps_pack;
-
- unsigned char pts_dts_flags = (pse_pack->info[0] & 0xF0) >> 6;
- //in a frame of data, pts is obtained from the first PSE packet
- if (pse_index == 0 && pts_dts_flags > 0) {
- video_pts = parse_ps_timestamp((unsigned char*)next_ps_pack + 9);
- srs_info("gb28181: ps stream video ts=%u pkt_ts=%u", video_pts, timestamp);
- }
- pse_index +=1;
-
- int packlength = htons(pse_pack->length);
- int payloadlen = packlength - 2 - 1 - pse_pack->stuffing_length;
-
- next_ps_pack = next_ps_pack + 9 + pse_pack->stuffing_length;
- complete_len = complete_len + 9 + pse_pack->stuffing_length;
-
- video_stream.append(next_ps_pack, payloadlen);
-
-#ifdef W_VIDEO_FILE
- if (!video_fw.is_open()) {
- std::string filename = "test_video_" + channel_id + ".h264";
- video_fw.open(filename.c_str());
- }
- video_fw.write(next_ps_pack, payloadlen, NULL);
-#endif
-
- next_ps_pack = next_ps_pack + payloadlen;
- complete_len = complete_len + payloadlen;
- incomplete_len = ps_size - complete_len;
- }
- else if (next_ps_pack
- && next_ps_pack[0] == (char)0x00
- && next_ps_pack[1] == (char)0x00
- && next_ps_pack[2] == (char)0x01
- && next_ps_pack[3] == (char)0xBD)
- {
- //private stream
-
- SrsPsePacket* pse_pack = (SrsPsePacket*)next_ps_pack;
-
- int packlength = htons(pse_pack->length);
- int payload_len = packlength - 2 - 1 - pse_pack->stuffing_length;
-
- next_ps_pack = next_ps_pack + payload_len + 9 + pse_pack->stuffing_length;
- complete_len = complete_len + (payload_len + 9 + pse_pack->stuffing_length);
- incomplete_len = ps_size - complete_len;
- }
- else if (next_ps_pack
- && next_ps_pack[0] == (char)0x00
- && next_ps_pack[1] == (char)0x00
- && next_ps_pack[2] == (char)0x01
- && next_ps_pack[3] == (char)0xC0)
- {
- //audio stream
-
- SrsPsePacket* pse_pack = (SrsPsePacket*)next_ps_pack;
-
- unsigned char pts_dts_flags = (pse_pack->info[0] & 0xF0) >> 6;
- if (pts_dts_flags > 0 ) {
- audio_pts = parse_ps_timestamp((unsigned char*)next_ps_pack + 9);
- srs_info("gb28181: ps stream video ts=%u pkt_ts=%u", audio_pts, timestamp);
- }
-
- int packlength = htons(pse_pack->length);
- int payload_len = packlength - 2 - 1 - pse_pack->stuffing_length;
- next_ps_pack = next_ps_pack + 9 + pse_pack->stuffing_length;
-
- //if ps map is not aac, but stream many be aac adts , try update type,
- //TODO: dahua audio ps map type always is 0x90(g711)
-
- 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]);
-
- if (audio_enable && audio_es_type != STREAM_TYPE_AUDIO_AAC &&
- (p1 & 0xFF) == 0xFF && (p2 & 0xF0) == 0xF0) {
-
- //try update aac type
- SrsBuffer avs(next_ps_pack, payload_len);
- char* frame = NULL;
- int frame_size = 0;
- SrsRawAacStreamCodec codec;
-
- srs_error_t err2 = srs_success;
- if ((err2 = aac->adts_demux(&avs, &frame, &frame_size, codec)) != srs_success) {
- srs_info("gb28181: client_id %s, audio data not aac adts (%#x/%u) %02x %02x %02x %02x\n",
- channel_id.c_str(), ssrc, timestamp, p1, p2, p3, p4);
- srs_error_reset(err);
- }else{
- srs_warn("gb28181: client_id %s, ps map is not aac (%s) type, but stream many be aac adts, try update type",
- channel_id.c_str(), get_ps_map_type_str(audio_es_type).c_str());
- audio_es_type = STREAM_TYPE_AUDIO_AAC;
- }
- }
-
- audio_stream.append(next_ps_pack, payload_len);
-
-#ifdef W_AUDIO_FILE
- if (!audio_fw.is_open()) {
- std::string filename = "test_audio_" + channel_id + ".aac";
- audio_fw.open(filename.c_str());
- }
- audio_fw.write(next_ps_pack, payload_len, NULL);
-#endif
-
- next_ps_pack = next_ps_pack + payload_len;
- complete_len = complete_len + (payload_len + 9 + pse_pack->stuffing_length);
- incomplete_len = ps_size - complete_len;
-
- if (hander && audio_enable && audio_stream.length() && can_send_ps_av_packet()) {
- if ((err = hander->on_rtp_audio(&audio_stream, audio_pts, audio_es_type)) != srs_success) {
- return srs_error_wrap(err, "process ps audio packet");
- }
- }
- }
- else
- {
-
-#ifdef W_UNKONW_FILE
- if (!unknow_fw.is_open()) {
- std::string filename = "test_unknow_" + channel_id + ".mpg";
- unknow_fw.open(filename.c_str());
- }
- unknow_fw.write(next_ps_pack, incomplete_len, NULL);
-#endif
- //TODO: fixme unkonw ps data parse
- if (next_ps_pack
- && next_ps_pack[0] == (char)0x00
- && next_ps_pack[1] == (char)0x00
- && next_ps_pack[2] == (char)0x00
- && next_ps_pack[3] == (char)0x01){
- //dahua's PS header may lose packets. It is sent by an RTP packet of Dahua's PS header
- //dahua rtp send format:
- //ts=1000 seq=1 mark=false payload= ps header
- //ts=1000 seq=2 mark=false payload= video
- //ts=1000 seq=3 mark=true payload= video
- //ts=1000 seq=4 mark=true payload= audio
- incomplete_len = ps_size - complete_len;
- complete_len = complete_len + incomplete_len;
- }
-
- first_keyframe_flag = false;
- srs_trace("gb28181: client_id %s, unkonw ps data (%#x/%u) %02x %02x %02x %02x\n",
- channel_id.c_str(), ssrc, timestamp,
- next_ps_pack[0]&0xFF, next_ps_pack[1]&0xFF, next_ps_pack[2]&0xFF, next_ps_pack[3]&0xFF);
- break;
- }
- }
-
- if (complete_len != ps_size){
- srs_trace("gb28181: client_id %s decode ps packet error (%#x/%u)! ps_size=%d complete=%d \n",
- channel_id.c_str(), ssrc, timestamp, ps_size, complete_len);
- }else if (hander && video_stream.length() && can_send_ps_av_packet() && video_es_type == STREAM_TYPE_VIDEO_H264) {
- if ((err = hander->on_rtp_video(&video_stream, video_pts)) != srs_success) {
- video_es_type = 0;
- return srs_error_wrap(err, "process ps video packet");
- }
- }
-
- return err;
-}
-
-static std::string get_host_candidate_ips(SrsConfDirective* c)
-{
- string candidate = _srs_config->get_stream_caster_gb28181_host(c);
- if (candidate == "*" || candidate == "0.0.0.0") {
- std::vector& ips = srs_get_local_ips();
- int index = _srs_config->get_stats_network();
- SrsIPAddress* ip = ips.at(index);
- return ip->ip;
- } else {
- return candidate;
- }
-}
-
-//Gb28181 Config
-SrsGb28181Config::SrsGb28181Config(SrsConfDirective* c)
-{
- // TODO: FIXME: support reload.
- host = get_host_candidate_ips(c);
- output = _srs_config->get_stream_caster_output(c);
- rtp_mux_port = _srs_config->get_stream_caster_listen(c);
- rtp_mux_tcp_enable = _srs_config->get_stream_caster_tcp_enable(c);
- rtp_port_min = _srs_config->get_stream_caster_rtp_port_min(c);
- rtp_port_max = _srs_config->get_stream_caster_rtp_port_max(c);
- rtp_idle_timeout = _srs_config->get_stream_caster_gb28181_rtp_idle_timeout(c);
-
- wait_keyframe = _srs_config->get_stream_caster_gb28181_wait_keyframe(c);
- audio_enable = _srs_config->get_stream_caster_gb28181_audio_enable(c);
- auto_create_channel = _srs_config->get_stream_caster_gb28181_auto_create_channel(c);
- jitterbuffer_enable = _srs_config->get_stream_caster_gb28181_jitterbuffer_enable(c);
-
- //sip config
- sip_enable = _srs_config->get_stream_caster_gb28181_sip_enable(c);
- sip_port = _srs_config->get_stream_caster_gb28181_sip_listen(c);
- sip_realm = _srs_config->get_stream_caster_gb28181_realm(c);
- sip_serial = _srs_config->get_stream_caster_gb28181_serial(c);
- sip_auto_play = _srs_config->get_stream_caster_gb28181_sip_auto_play(c);
- sip_ack_timeout = _srs_config->get_stream_caster_gb28181_ack_timeout(c);
- sip_keepalive_timeout = _srs_config->get_stream_caster_gb28181_keepalive_timeout(c);
- sip_invite_port_fixed = _srs_config->get_stream_caster_gb28181_sip_invite_port_fixed(c);
- sip_query_catalog_interval = _srs_config->get_stream_caster_gb28181_sip_query_catalog_interval(c);
-}
-
-SrsGb28181Config::~SrsGb28181Config()
-{
-
-}
-
-//SrsGb28181RtmpMuxer gb28181 rtmp muxer, process ps stream to rtmp
-SrsGb28181RtmpMuxer::SrsGb28181RtmpMuxer(SrsGb28181Manger* c, std::string id, bool a, bool k)
-{
- channel_id = id;
- gb28181_manger = c;
- channel = new SrsGb28181StreamChannel();
-
- pprint = SrsPithyPrint::create_caster();
- trd = new SrsSTCoroutine("gb28181rtmpmuxer", this);
- //change stack size to 256K, fix crash when call FFMpeg
- ((SrsSTCoroutine*)trd)->set_stack_size(1 << 18);
-
- sdk = NULL;
- vjitter = new SrsRtpTimeJitter();
- ajitter = new SrsRtpTimeJitter();
-
- avc = new SrsRawH264Stream();
- aac = new SrsRawAacStream();
-
- ps_demixer = new SrsPsStreamDemixer(this, id, a, k);
- wait_ps_queue = srs_cond_new();
-
- stream_idle_timeout = -1;
- recv_rtp_stream_time = 0;
- send_rtmp_stream_time = 0;
-
- _rtmp_url = "";
-
- h264_sps = "";
- h264_pps = "";
- aac_specific_config = "";
-
- req = NULL;
- server = NULL;
- source = NULL;
- source_publish = true;
-
- jitter_buffer = new SrsRtpJitterBuffer(id);
- jitter_buffer_audio = new SrsRtpJitterBuffer(id);
-
- ps_buflen = 0;
- ps_buffer = NULL;
-
- ps_buflen_auido = 0;
- ps_buffer_audio = NULL;
-}
-
-SrsGb28181RtmpMuxer::~SrsGb28181RtmpMuxer()
-{
-
- close();
-
- srs_cond_destroy(wait_ps_queue);
-
- srs_freep(jitter_buffer);
- srs_freep(jitter_buffer_audio);
- srs_freepa(ps_buffer);
- srs_freepa(ps_buffer_audio);
-
- srs_freep(channel);
- srs_freep(ps_demixer);
- srs_freep(trd);
- srs_freep(sdk);
- srs_freep(vjitter);
- srs_freep(ajitter);
- srs_freep(pprint);
-
- destroy();
-}
-
-srs_error_t SrsGb28181RtmpMuxer::serve()
-{
- srs_error_t err = srs_success;
-
- if ((err = trd->start()) != srs_success) {
- return srs_error_wrap(err, "gb28181rtmpmuxer");
- }
-
- return err;
-}
-
-std::string SrsGb28181RtmpMuxer::remote_ip()
-{
- return "";
-}
-
-const SrsContextId& SrsGb28181RtmpMuxer::get_id()
-{
- return _srs_context->get_id();
-}
-
-std::string SrsGb28181RtmpMuxer::desc()
-{
- return "GBConn";
-}
-
-std::string SrsGb28181RtmpMuxer::get_channel_id()
-{
- return channel_id;
-}
-
-void SrsGb28181RtmpMuxer::copy_channel(SrsGb28181StreamChannel *s)
-{
- channel->copy(s);
-}
-
-SrsGb28181StreamChannel SrsGb28181RtmpMuxer::get_channel()
-{
- return *channel;
-}
-
-void SrsGb28181RtmpMuxer::set_channel_peer_ip(std::string ip)
-{
- if (channel->get_rtp_peer_ip().empty()){
- channel->set_rtp_peer_ip(ip);
- }
-}
-
-void SrsGb28181RtmpMuxer::set_channel_peer_port(int port)
-{
- if (channel->get_rtp_peer_port() == 0){
- channel->set_recv_time_str(srs_sip_get_utc_date());
- channel->set_recv_time(srs_get_system_time());
- channel->set_rtp_peer_port(port);
- }
-}
-
-int SrsGb28181RtmpMuxer::channel_peer_port()
-{
- return channel->get_rtp_peer_port();
-}
-
-std::string SrsGb28181RtmpMuxer::channel_peer_ip()
-{
- return channel->get_rtp_peer_ip();
-}
-
-void SrsGb28181RtmpMuxer::set_rtmp_url(std::string url)
-{
- _rtmp_url = url;
-}
-std::string SrsGb28181RtmpMuxer::rtmp_url()
-{
- return _rtmp_url;
-}
-
-srs_utime_t SrsGb28181RtmpMuxer::get_recv_stream_time()
-{
- return recv_rtp_stream_time;
-}
-
-
-void SrsGb28181RtmpMuxer::destroy()
-{
- while(!ps_queue.empty()){
- SrsPsRtpPacket* pkt = ps_queue.front();
- ps_queue.pop();
- //must be free pkt
- srs_freep(pkt);
- }
-}
-
-srs_error_t SrsGb28181RtmpMuxer::initialize(SrsServer *s, SrsRequest* r)
-{
- srs_error_t err = srs_success;
-
- if (!jitter_buffer) {
- jitter_buffer = new SrsRtpJitterBuffer(channel_id);
- }
-
- jitter_buffer->SetDecodeErrorMode(kSelectiveErrors);
- jitter_buffer->SetNackMode(kNack, -1, -1);
- jitter_buffer->SetNackSettings(250, 450, 0);
-
- if (!jitter_buffer_audio) {
- jitter_buffer_audio = new SrsRtpJitterBuffer(channel_id);
- }
-
- jitter_buffer_audio->SetDecodeErrorMode(kSelectiveErrors);
- jitter_buffer_audio->SetNackMode(kNack, -1, -1);
- jitter_buffer_audio->SetNackSettings(250, 450, 0);
-
- if (!source_publish) return err;
-
- req = r;
- server = s;
-
- if ((err = _srs_sources->fetch_or_create(req, (ISrsLiveSourceHandler*)server, &source)) != srs_success) {
- return srs_error_wrap(err, "create source");
- }
-
- #ifdef SRS_RTC
- SrsRtcSource *rtc = NULL;
- bool rtc_server_enabled = _srs_config->get_rtc_server_enabled();
- bool rtc_enabled = _srs_config->get_rtc_enabled(req->vhost);
- if (rtc_server_enabled && rtc_enabled) {
- if ((err = _srs_rtc_sources->fetch_or_create(req, &rtc)) != srs_success) {
- return srs_error_wrap(err, "create source");
- }
-
- if (!rtc->can_publish()) {
- return srs_error_new(ERROR_RTC_SOURCE_BUSY, "gb28181 rtc stream %s busy", req->get_stream_url().c_str());
- }
- }
-#endif
-
- // Check whether RTMP stream is busy.
- if (!source->can_publish(false)) {
- return srs_error_new(ERROR_SYSTEM_STREAM_BUSY, "gb28181 rtmp: stream %s is busy", req->get_stream_url().c_str());
- }
-
- // Bridge to RTC streaming.
-#if defined(SRS_RTC) && defined(SRS_FFMPEG_FIT)
- if (rtc) {
- SrsRtcFromRtmpBridger *bridger = new SrsRtcFromRtmpBridger(rtc);
- if ((err = bridger->initialize(req)) != srs_success) {
- srs_freep(bridger);
- return srs_error_wrap(err, "bridger init");
- }
-
- source->set_bridger(bridger);
- }
-#endif
-
- if ((err = source->on_publish()) != srs_success) {
- return srs_error_wrap(err, "on publish");
- }
-
- return err;
-}
-
-
-srs_error_t SrsGb28181RtmpMuxer::do_cycle()
-{
- srs_error_t err = srs_success;
- recv_rtp_stream_time = srs_get_system_time();
- 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) {
-
- pprint->elapse();
-
- if ((err = trd->pull()) != srs_success) {
- return srs_error_wrap(err, "gb28181 rtmp muxer cycle");
- }
-
- SrsGb28181Config config = gb28181_manger->get_gb28181_config();
-
- if (config.jitterbuffer_enable){
- if(jitter_buffer->FoundFrame(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){
- srs_warn("gb28181: demix ps stream error:%s", srs_error_desc(err).c_str());
- srs_freep(err);
- };
- }
- }
-
- if(jitter_buffer_audio->FoundFrame(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){
- srs_warn("gb28181: demix ps stream error:%s", srs_error_desc(err).c_str());
- srs_freep(err);
- };
- }
- }
- }else {
- //demix ps to h264/aac, to rtmp
- while(!ps_queue.empty()){
- SrsPsRtpPacket* pkt = ps_queue.front();
- if (pkt){
- if ((err = ps_demixer->on_ps_stream(pkt->payload->bytes(),
- pkt->payload->length(), pkt->timestamp, pkt->ssrc)) != srs_success){
- srs_warn("gb28181: demix ps stream error:%s", srs_error_desc(err).c_str());
- srs_freep(err);
- };
- }
- ps_queue.pop();
- //must be free pkt
- srs_freep(pkt);
- }
- }
-
- if (pprint->can_print()) {
- srs_trace("gb28181: client id=%s, ssrc=%#x, peer(%s, %d), rtmp muxer is alive",
- channel_id.c_str(), channel->get_ssrc(),
- channel->get_rtp_peer_ip().c_str(),
- channel->get_rtp_peer_port());
- }
-
- srs_utime_t now = srs_get_system_time();
- srs_utime_t duration = now - recv_rtp_stream_time;
-
- //if no RTP data is received within 2 seconds,
- //the peer-port and peer-ip will be cleared and
- //other port data will be received again
- if (duration > (2 * SRS_UTIME_SECONDS) && channel->get_rtp_peer_port() != 0){
- srs_warn("gb28181: client id=%s ssrc=%#x, peer(%s, %d), no rtp data %d in seconds, clean it, wait other port!",
- channel_id.c_str(), channel->get_ssrc(), channel->get_rtp_peer_ip().c_str(),
- channel->get_rtp_peer_port(), duration/SRS_UTIME_SECONDS);
- channel->set_rtp_peer_port(0);
- channel->set_rtp_peer_ip("");
- }
-
-
- if (duration > config.rtp_idle_timeout){
- srs_trace("gb28181: client id=%s, stream idle timeout, stop!!!", channel_id.c_str());
- break;
- }
-
- //RTMP connection is about to timeout without receiving any data.,
- //waiting for the next time there is data automatically connected
- //it is related to the following two parameter settings of the rtmp server
- //the publish 1st packet timeout in srs_utime_t
- //publish_1stpkt_timeout default 20000ms
- //the publish normal packet timeout in srs_utime_t
- //publish_normal_timeout default 5000ms
- duration = now - send_rtmp_stream_time;
- bool will_timeout = duration > (5 * SRS_UTIME_SECONDS);
- if (will_timeout && sdk){
- srs_warn("gb28181: client id=%s RTMP connection is about to time out without receiving any data",
- channel_id.c_str());
- rtmp_close();
- }
-
- srs_usleep(30 * SRS_UTIME_MILLISECONDS);
- }
-
- return err;
-}
-
-
-void SrsGb28181RtmpMuxer::stop()
-{
- if (trd){
- trd->interrupt();
- }
- //stop rtmp publish
- close();
-}
-
-
-void SrsGb28181RtmpMuxer::insert_jitterbuffer(SrsPsRtpPacket *pkt)
-{
- if (!pkt){
- return;
- }
-
- recv_rtp_stream_time = srs_get_system_time();
-
- char *payload = pkt->payload->bytes();
- if (!payload) {
- return;
- }
-
- uint8_t p1 = (uint8_t)(payload[0]);
- uint8_t p2 = (uint8_t)(payload[1]);
- uint8_t p3 = (uint8_t)(payload[2]);
- uint8_t p4 = (uint8_t)(payload[3]);
-
-
- //check for rtp ps audio streaming
- bool av_same_ts = true;
-
- if (p1 == 0x00 && p2 == 0x00 && p3 == 0x01 && p4 == 0xC0 &&
- ps_rtp_video_ts != pkt->timestamp) {
- av_same_ts = false;
- }
-
- //if audio and video are the same clock,
- //if both audio and video use jitter_buffer,
- //otherwise audio uses jitter_buffer_audio, and video uses jitter_buffer
- if (av_same_ts){
- pkt->marker = false;
- 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->sequence_number, pkt->timestamp, pkt->marker,
- pkt->payload->bytes(), pkt->payload->length(), NULL);
- }
-
- //srs_cond_signal(wait_ps_queue);
-}
-
-void SrsGb28181RtmpMuxer::ps_packet_enqueue(SrsPsRtpPacket *pkt)
-{
- srs_assert(pkt);
-
- recv_rtp_stream_time = srs_get_system_time();
-
- //prevent consumers from being unable to process data
- //and accumulating in the queue
- uint32_t size = ps_queue.size();
- if (size > 100){
- srs_warn("gb28181: rtmpmuxer too much queue data, need to clear!!!");
- while(!ps_queue.empty()) {
- SrsPsRtpPacket* pkt = ps_queue.front();
- ps_queue.pop();
- srs_freep(pkt);
- }
- }
-
- ps_queue.push(pkt);
- //srs_cond_signal(wait_ps_queue);
-}
-
-srs_error_t SrsGb28181RtmpMuxer::cycle()
-{
- // serve the rtmp muxer.
- srs_error_t err = do_cycle();
-
- gb28181_manger->stop_rtp_listen(channel_id);
-
- gb28181_manger->remove(this);
- srs_trace("gb28181: client id=%s rtmp muxer is remove", channel_id.c_str());
-
- if (err == srs_success) {
- srs_trace("client finished.");
- } else if (srs_is_client_gracefully_close(err)) {
- srs_warn("client disconnect peer. code=%d", srs_error_code(err));
- srs_freep(err);
- }
-
- return err;
-}
-
-srs_error_t SrsGb28181RtmpMuxer::on_rtp_video(SrsSimpleStream *stream, int64_t fpts)
-{
- srs_error_t err = srs_success;
-
- if (!source_publish){
- // ensure rtmp connected.
- if ((err = connect()) != srs_success) {
- //after the connection fails, need to clear flag
- //and send the av header again next time
- h264_sps = "";
- h264_pps = "";
- aac_specific_config = "";
- return srs_error_wrap(err, "connect");
- }
- }
-
- if ((err = vjitter->correct(fpts)) != srs_success) {
- return srs_error_wrap(err, "jitter");
- }
-
- // ts tbn to flv tbn.
- uint32_t dts = (uint32_t)(fpts / 90);
- uint32_t pts = (uint32_t)(fpts / 90);
- srs_info("gb28181rtmpmuxer: on_rtp_video dts=%u", dts);
-
- if (true) {
- char *data = stream->bytes();
- int length = stream->length();
-
- err = replace_startcode_with_nalulen(data, length, dts, pts);
- }
-
- return err;
-}
-
-srs_error_t SrsGb28181RtmpMuxer::write_h264_ipb_frame2(char *frame, int frame_size, uint32_t pts, uint32_t dts)
-{
- srs_error_t err = srs_success;
-
- if (!frame){
- return srs_error_new(ERROR_GB28181_H264_FRAME_FULL, "h264 frame null");
- }
-
- if (frame_size <= 0){
- return srs_error_new(ERROR_GB28181_H264_FRAMESIZE, "h264 frame size");
- }
-
- SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(frame[0] & 0x1f);
- // ignore the nalu type sei(6) aud(9)
- if (nal_unit_type == SrsAvcNaluTypeAccessUnitDelimiter ||
- nal_unit_type == SrsAvcNaluTypeSEI) {
- return err;
- }
-
- // for sps
- if (avc->is_sps(frame, frame_size)) {
- std::string sps;
- if ((err = avc->sps_demux(frame, frame_size, sps)) != srs_success) {
- return srs_error_wrap(err, "demux sps");
- }
-
- if (h264_sps == sps) {
- return err;
- }
- h264_sps = sps;
-
- if ((err = write_h264_sps_pps(dts, pts)) != srs_success) {
- return srs_error_wrap(err, "write sps/pps");
- }
- return err;
- }
-
- // for pps
- if (avc->is_pps(frame, frame_size)) {
- std::string pps;
- if ((err = avc->pps_demux(frame, frame_size, pps)) != srs_success) {
- return srs_error_wrap(err, "demux pps");
- }
-
- if (h264_pps == pps) {
- return err;
- }
- h264_pps = pps;
-
- if ((err = write_h264_sps_pps(dts, pts)) != srs_success) {
- return srs_error_wrap(err, "write sps/pps");
- }
- return err;
- }
-
- srs_info("gb28181: demux avc ibp frame size=%d, dts=%d", frame_size, dts);
- if ((err = write_h264_ipb_frame(frame, frame_size, dts, pts)) != srs_success) {
- return srs_error_wrap(err, "write frame");
- }
-
- return err;
-}
-
- srs_error_t SrsGb28181RtmpMuxer::replace_startcode_with_nalulen(char *video_data, int &size, uint32_t pts, uint32_t dts)
- {
- srs_error_t err = srs_success;
-
- int index = 0;
- std::list list_index;
-
- for(; index < size; index++){
- if (index > (size-4))
- break;
- if (video_data[index] == 0x00 && video_data[index+1] == 0x00 &&
- video_data[index+2] == 0x00 && video_data[index+3] == 0x01){
- list_index.push_back(index);
- }
- }
-
- if (list_index.size() == 1){
- int cur_pos = list_index.front();
- list_index.pop_front();
-
- //0001xxxxxxxxxx
- //xxxx0001xxxxxxx
- uint32_t naluLen = size - cur_pos - 4;
-
- char *frame = video_data + cur_pos + 4;
- int frame_size = naluLen;
-
- err = write_h264_ipb_frame2(frame, frame_size, dts, pts);
-
- }else if (list_index.size() > 1){
- int pre_pos = list_index.front();
- list_index.pop_front();
- int first_pos = pre_pos;
-
- while(list_index.size() > 0){
- int cur_pos = list_index.front();
- list_index.pop_front();
-
- //pre=========cur======================
- //0001xxxxxxxx0001xxxxxxxx0001xxxxxxxxx
- //xxxxxxxxxxxx0001xxxxxxxx0001xxxxxxxxx
- uint32_t naluLen = cur_pos - pre_pos - 4;
-
- char *frame = video_data + pre_pos + 4;
- int frame_size = naluLen;
-
- pre_pos = cur_pos;
- err = write_h264_ipb_frame2(frame, frame_size, dts, pts);
- }
-
- //========================pre==========
- //0001xxxxxxxx0001xxxxxxxx0001xxxxxxxxx
- if (first_pos != pre_pos){
-
- uint32_t naluLen = size - pre_pos - 4;
-
- char *frame = video_data + pre_pos + 4;
- int frame_size = naluLen;
-
- err = write_h264_ipb_frame2(frame, frame_size, dts, pts);
- }
- }else{
- //xxxxxxxxxxxxxxxxxxx
- char *frame = video_data;
- int frame_size = size;
- err = write_h264_ipb_frame2(frame, frame_size, dts, pts);
- }
-
- return err;
-}
-
-srs_error_t SrsGb28181RtmpMuxer::on_rtp_audio(SrsSimpleStream* stream, int64_t fdts, int type)
-{
- srs_error_t err = srs_success;
-
- if (!source_publish){
- // ensure rtmp connected.
- if ((err = connect()) != srs_success) {
- //after the connection fails, need to clear flag
- //and send the av header again next time
- h264_sps = "";
- h264_pps = "";
- aac_specific_config = "";
- return srs_error_wrap(err, "connect");
- }
- }
-
- if ((err = ajitter->correct(fdts)) != srs_success) {
- return srs_error_wrap(err, "jitter");
- }
-
- uint32_t dts = (uint32_t)(fdts / 90);
-
- // send each frame.
- SrsBuffer *avs = new SrsBuffer(stream->bytes(), stream->length());
- SrsAutoFree(SrsBuffer, avs);
- if (!avs->empty()) {
- if (type == STREAM_TYPE_AUDIO_AAC) {
- char* frame = NULL;
- int frame_size = 0;
- SrsRawAacStreamCodec codec;
- if ((err = aac->adts_demux(avs, &frame, &frame_size, codec)) != srs_success) {
- return srs_error_wrap(err, "demux adts");
- }
-
- if (frame_size <= 0) {
- return err;
- }
-
- bool send_adts = false;
- static int srs_aac_srates[] = {
- 96000, 88200, 64000, 48000,
- 44100, 32000, 24000, 22050,
- 16000, 12000, 11025, 8000,
- 7350, 0, 0, 0
- };
- switch (srs_aac_srates[codec.sampling_frequency_index]) {
- case 11025:
- codec.sound_rate = SrsAudioSampleRate11025;
- break;
- case 22050:
- codec.sound_rate = SrsAudioSampleRate22050;
- break;
- case 44100:
- codec.sound_rate = SrsAudioSampleRate44100;
- break;
- default:
- send_adts = true; //raw with adts
- break;
- };
-
- std::string sh;
- if ((err = aac->mux_sequence_header(&codec, sh)) != srs_success) {
- return srs_error_wrap(err, "mux sequence header");
- }
-
- if (aac_specific_config != sh){
- std::string sh;
- if ((err = aac->mux_sequence_header(&codec, sh)) != srs_success) {
- return srs_error_wrap(err, "mux sequence header");
- }
- aac_specific_config = sh;
- codec.aac_packet_type = 0;
- if ((err = write_audio_raw_frame((char*)sh.data(), (int)sh.length(), &codec, dts)) != srs_success) {
- return srs_error_wrap(err, "write raw audio frame");
- }
- }
-
- codec.aac_packet_type = 1;
- if (send_adts) { // audio raw data. with adts header
- if ((err = write_audio_raw_frame(stream->bytes(), stream->length(), &codec, dts)) != srs_success) {
- return srs_error_wrap(err, "write audio raw frame");
- }
- }else { // audio raw data. without adts header
- if ((err = write_audio_raw_frame(frame, frame_size, &codec, dts)) != srs_success) {
- return srs_error_wrap(err, "write audio raw frame");
- }
- }
- }else if (type != 0) {
- SrsRawAacStreamCodec codec;
- codec.aac_packet_type = 0;
-
- if (type == STREAM_TYPE_AUDIO_G711){
- codec.sound_format = SrsAudioCodecIdReservedG711AlawLogarithmicPCM;
- codec.sound_rate = SrsAudioSampleRate5512;
- codec.sound_type = 0; //MONO = 0, STEREO = 1
- codec.sound_size = 0; //0=8K, 1=16K
- }else if(type == STREAM_TYPE_AUDIO_G711ULAW){
- codec.sound_format = SrsAudioCodecIdReservedG711MuLawLogarithmicPCM;
- codec.sound_rate = SrsAudioSampleRate5512;
- codec.sound_type = 0;
- codec.sound_size = 0;
- }else {
- return srs_error_wrap(err, "write audio raw frame, type=%d not suppered", type);
- }
-
- char* frame = stream->bytes();
- int frame_size = stream->length();
- if ((err = write_audio_raw_frame(frame, frame_size, &codec, dts)) != srs_success) {
- return srs_error_wrap(err, "write audio raw frame");
- }
- }
- }//end if (!avs->empty())
-
- return err;
-}
-
-srs_error_t SrsGb28181RtmpMuxer::write_h264_sps_pps(uint32_t dts, uint32_t pts)
-{
- srs_error_t err = srs_success;
-
- if (h264_sps == "" || h264_pps == ""){
- return err;
- }
-
- // h264 raw to h264 packet.
- std::string sh;
- if ((err = avc->mux_sequence_header(h264_sps, h264_pps, dts, pts, sh)) != srs_success) {
- return srs_error_wrap(err, "mux sequence header");
- }
-
- // h264 packet to flv packet.
- int8_t frame_type = SrsVideoAvcFrameTypeKeyFrame;
- int8_t avc_packet_type = SrsVideoAvcFrameTraitSequenceHeader;
- char* flv = NULL;
- int nb_flv = 0;
- if ((err = avc->mux_avc2flv(sh, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != srs_success) {
- return srs_error_wrap(err, "mux avc to flv");
- }
-
- // the timestamp in rtmp message header is dts.
- uint32_t timestamp = dts;
- if ((err = rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv)) != srs_success) {
- return srs_error_wrap(err, "write packet");
- }
-
- return err;
-}
-
-srs_error_t SrsGb28181RtmpMuxer::write_h264_ipb_frame(char* frame, int frame_size, uint32_t dts, uint32_t pts, bool writelen)
-{
- srs_error_t err = srs_success;
-
- // 5bits, 7.3.1 NAL unit syntax,
- // ISO_IEC_14496-10-AVC-2003.pdf, page 44.
- // 7: SPS, 8: PPS, 5: I Frame, 1: P Frame
- SrsAvcNaluType nal_unit_type = (SrsAvcNaluType)(frame[0] & 0x1f);
-
- // for IDR frame, the frame is keyframe.
- SrsVideoAvcFrameType frame_type = SrsVideoAvcFrameTypeInterFrame;
- if (nal_unit_type == SrsAvcNaluTypeIDR) {
- frame_type = SrsVideoAvcFrameTypeKeyFrame;
- }
-
- std::string ibp;
-
- if (writelen){
- if ((err = avc->mux_ipb_frame(frame, frame_size, ibp)) != srs_success) {
- return srs_error_wrap(err, "mux ibp frame");
- }
- }else{
- ibp = string(frame, frame_size);
- }
-
- int8_t avc_packet_type = SrsVideoAvcFrameTraitNALU;
- char* flv = NULL;
- int nb_flv = 0;
- if ((err = avc->mux_avc2flv(ibp, frame_type, avc_packet_type, dts, pts, &flv, &nb_flv)) != srs_success) {
- return srs_error_wrap(err, "mux avc to flv");
- }
-
- // the timestamp in rtmp message header is dts.
- uint32_t timestamp = dts;
- return rtmp_write_packet(SrsFrameTypeVideo, timestamp, flv, nb_flv);
-}
-
-srs_error_t SrsGb28181RtmpMuxer::write_audio_raw_frame(char* frame, int frame_size, SrsRawAacStreamCodec* codec, uint32_t dts)
-{
- srs_error_t err = srs_success;
-
- char* data = NULL;
- int size = 0;
- if ((err = aac->mux_aac2flv(frame, frame_size, codec, dts, &data, &size)) != srs_success) {
- return srs_error_wrap(err, "mux aac to flv");
- }
-
- return rtmp_write_packet(SrsFrameTypeAudio, dts, data, size);
-}
-
-srs_error_t SrsGb28181RtmpMuxer::rtmp_write_packet(char type, uint32_t timestamp, char* data, int size)
-{
- srs_error_t err = srs_success;
-
- if (source_publish){
- return rtmp_write_packet_by_source(type, timestamp, data, size);
- }
-
- if ((err = connect()) != srs_success) {
- return srs_error_wrap(err, "connect");
- }
-
- SrsSharedPtrMessage* msg = NULL;
-
- send_rtmp_stream_time = srs_get_system_time();
-
- if ((err = srs_rtmp_create_msg(type, timestamp, data, size, sdk->sid(), &msg)) != srs_success) {
- return srs_error_wrap(err, "create message");
- }
- srs_assert(msg);
-
- // send out encoded msg.
- if ((err = sdk->send_and_free_message(msg)) != srs_success) {
- close();
- return srs_error_wrap(err, "write message");
- }
- return err;
-}
-
-srs_error_t SrsGb28181RtmpMuxer::rtmp_write_packet_by_source(char type, uint32_t timestamp, char* data, int size)
-{
- srs_error_t err = srs_success;
-
- send_rtmp_stream_time = srs_get_system_time();
-
- //create a source that will process stream without the need for internal rtmpclient
- if (type == SrsFrameTypeAudio) {
- SrsMessageHeader header;
- header.message_type = RTMP_MSG_AudioMessage;
- // TODO: FIXME: Maybe the tbn is not 90k.
- header.timestamp = timestamp & 0x3fffffff;
-
- SrsCommonMessage* shared_video = new SrsCommonMessage();
- SrsAutoFree(SrsCommonMessage, shared_video);
-
- // TODO: FIXME: Check error.
- shared_video->create(&header, data, size);
- source->on_audio(shared_video);
- }else if(type == SrsFrameTypeVideo) {
- SrsMessageHeader header;
- header.message_type = RTMP_MSG_VideoMessage;
- // TODO: FIXME: Maybe the tbn is not 90k.
- header.timestamp = timestamp & 0x3fffffff;
-
- SrsCommonMessage* shared_video = new SrsCommonMessage();
- SrsAutoFree(SrsCommonMessage, shared_video);
-
- // TODO: FIXME: Check error.
- shared_video->create(&header, data, size);
- source->on_video(shared_video);
- }
-
- return err;
-}
-
-srs_error_t SrsGb28181RtmpMuxer::connect()
-{
- srs_error_t err = srs_success;
-
- // Ignore when connected.
- if (sdk) {
- return err;
- }
-
- // generate rtmp url to connect to.
- std::string url = _rtmp_url;
-
- // connect host.
- srs_utime_t cto = SRS_CONSTS_RTMP_TIMEOUT;
- srs_utime_t sto = SRS_CONSTS_RTMP_PULSE;
- sdk = new SrsSimpleRtmpClient(url, cto, sto);
-
- srs_trace("gb28181: rtmp connect url=%s", url.c_str());
-
- if ((err = sdk->connect()) != srs_success) {
- close();
- return srs_error_wrap(err, "connect %s failed, cto=%dms, sto=%dms.", url.c_str(), srsu2msi(cto), srsu2msi(sto));
- }
-
- // publish.
- if ((err = sdk->publish(SRS_CONSTS_RTMP_PROTOCOL_CHUNK_SIZE)) != srs_success) {
- close();
- return srs_error_wrap(err, "publish %s failed", url.c_str());
- }
-
- return err;
-}
-
-void SrsGb28181RtmpMuxer::close()
-{
- srs_freep(sdk);
-
- // cleared and sequence header will be sent again next time.
- // RTMP close may stop through API(rtmp_close)
- h264_sps = "";
- h264_pps = "";
- aac_specific_config = "";
-
- // BUGFIX: if don't unpublish, it will always be in the /api/v1/streams list
- //if (source_publish && !source){
- if (source_publish && source){
- source->on_unpublish();
- }
-}
-
-void SrsGb28181RtmpMuxer::rtmp_close(){
- close();
-}
-
-SrsGb28181StreamChannel::SrsGb28181StreamChannel(){
- channel_id = "";
- port_mode = "";
- app = "";
- stream = "";
- ip = "";
- rtp_port = 0;
- rtmp_port = 0;
- ssrc = 0;
- rtp_peer_port = 0;
- rtp_peer_ip = "";
- rtmp_url = "";
- flv_url = "";
- hls_url = "";
- webrtc_url = "";
- recv_time = 0;
- recv_time_str = "";
-}
-
-SrsGb28181StreamChannel::~SrsGb28181StreamChannel()
-{
-
-}
-
-void SrsGb28181StreamChannel::copy(const SrsGb28181StreamChannel *s){
- channel_id = s->get_channel_id();
- port_mode = s->get_port_mode();
- app = s->get_app();
- stream = s->get_stream();
-
- ip = s->get_ip();
- rtp_port = s->get_rtp_port();
- rtmp_port = s->get_rtmp_port();
- ssrc = s->get_ssrc();
-
- rtp_peer_ip = s->get_rtp_peer_ip();
- 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();
-
-}
-
-void SrsGb28181StreamChannel::dumps(SrsJsonObject* obj)
-{
- obj->set("id", SrsJsonAny::str(channel_id.c_str()));
- obj->set("ip", SrsJsonAny::str(ip.c_str()));
- obj->set("rtmp_port", SrsJsonAny::integer(rtmp_port));
- 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));
- obj->set("port_mode", SrsJsonAny::str(port_mode.c_str()));
- obj->set("rtp_peer_port", SrsJsonAny::integer(rtp_peer_port));
- obj->set("rtp_peer_ip", SrsJsonAny::str(rtp_peer_ip.c_str()));
- obj->set("recv_time", SrsJsonAny::integer(recv_time/SRS_UTIME_SECONDS));
- obj->set("recv_time_str", SrsJsonAny::str(recv_time_str.c_str()));
-
-}
-
-
-//Global Singleton instance, init in SrsServer
-SrsGb28181Manger* _srs_gb28181 = NULL;
-
-//SrsGb28181Manger
-SrsGb28181Manger::SrsGb28181Manger(SrsServer *s, SrsConfDirective* c)
-{
- // TODO: FIXME: support reload.
- server = s;
- config = new SrsGb28181Config(c);
- manager = new SrsResourceManager("GB28181");
-}
-
-SrsGb28181Manger::~SrsGb28181Manger()
-{
- used_ports.clear();
- destroy();
-
- srs_freep(manager);
- srs_freep(config);
-}
-
-srs_error_t SrsGb28181Manger::initialize()
-{
- srs_error_t err = srs_success;
- if ((err = manager->start()) != srs_success) {
- return srs_error_wrap(err, "start manager");
- }
-
- return err;
-}
-
-SrsGb28181Config SrsGb28181Manger::get_gb28181_config()
-{
- return *config;
-}
-
-void SrsGb28181Manger::alloc_port(int* pport)
-{
- // use a pair of port.
- for (int i = config->rtp_port_min; i < config->rtp_port_max - 1; i += 2) {
- if (!used_ports[i]) {
- used_ports[i] = true;
- used_ports[i + 1] = true;
- *pport = i;
- break;
- }
- }
- srs_info("gb28181: alloc port=%d-%d", *pport, *pport + 1);
-}
-
-void SrsGb28181Manger::free_port(int lpmin, int lpmax)
-{
- for (int i = lpmin; i < lpmax; i++) {
- used_ports[i] = false;
- }
- srs_trace("gb28181: free rtp port=%d-%d", lpmin, lpmax);
-}
-
-uint32_t SrsGb28181Manger::hash_code(std::string str)
-{
- uint32_t h = 0;
- int len = str.length();
-
- if (h == 0) {
- int off = 0;
- const char *val = str.c_str();
-
- for (int i = 0; i < len; i++) {
- h = 31 * h + val[off++];
- }
- }
- return h;
-}
-
-uint32_t SrsGb28181Manger::generate_ssrc(std::string id)
-{
- srand(uint(time(0)));
- // TODO: SSRC rules can be customized,
- //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 = 0x00FFFFFF & (hash_code(id));
- srs_trace("gb28181: generate ssrc id=%s, ssrc=%u", id.c_str(), ssrc);
- return ssrc;
-}
-
-srs_error_t SrsGb28181Manger::fetch_or_create_rtmpmuxer(std::string id, SrsRequest *req, SrsGb28181RtmpMuxer** gb28181)
-{
- srs_error_t err = srs_success;
-
- SrsGb28181RtmpMuxer* muxer = NULL;
- if ((muxer = fetch_rtmpmuxer(id)) != NULL) {
- *gb28181 = muxer;
- return err;
- }
-
- muxer = new SrsGb28181RtmpMuxer(this, id, config->audio_enable, config->wait_keyframe);
- if ((err = muxer->initialize(server, req)) != srs_success) {
- return srs_error_wrap(err, "gb28181: rtmp muxer initialize %s", id.c_str());
- }
-
- if ((err = muxer->serve()) != srs_success) {
- return srs_error_wrap(err, "gb28181: rtmp muxer serve %s", id.c_str());
- }
- rtmpmuxers[id] = muxer;
- *gb28181 = muxer;
-
- return err;
-}
-
-SrsGb28181RtmpMuxer* SrsGb28181Manger::fetch_rtmpmuxer(std::string id)
-{
- SrsGb28181RtmpMuxer* muxer = NULL;
-
- if (rtmpmuxers.find(id) == rtmpmuxers.end()) {
- return NULL;
- }
-
- muxer = rtmpmuxers[id];
- return muxer;
-}
-
-void SrsGb28181Manger::update_rtmpmuxer_to_newssrc_by_id(std::string id, uint32_t ssrc)
-{
- SrsGb28181RtmpMuxer* muxer = NULL;
-
- if (rtmpmuxers.find(id) == rtmpmuxers.end()) {
- srs_warn("gb28181: at update_rtmpmuxer_to_newssrc_by_id() client_id not found. client_id=%s",id.c_str());
- return;
- }
-
- muxer = rtmpmuxers[id];
-
- SrsGb28181StreamChannel mc = muxer->get_channel();
- uint32_t old_ssrc = mc.get_ssrc();
- if (old_ssrc == ssrc) {
- return;
- } else {
- srs_trace("gb28181: update ssrc of muxer %s from %x to %x", id.c_str(), old_ssrc, ssrc);
- }
- rtmpmuxer_unmap_by_ssrc(old_ssrc);
- mc.set_ssrc(ssrc);
- muxer->copy_channel(&mc);
- rtmpmuxer_map_by_ssrc(muxer, ssrc);
-}
-
-SrsGb28181RtmpMuxer* SrsGb28181Manger::fetch_rtmpmuxer_by_ssrc(uint32_t ssrc)
-{
- SrsGb28181RtmpMuxer* muxer = NULL;
- if (rtmpmuxers_ssrc.find(ssrc) == rtmpmuxers_ssrc.end()) {
- return NULL;
- }
-
- muxer = rtmpmuxers_ssrc[ssrc];
- return muxer;
-}
-
-void SrsGb28181Manger::rtmpmuxer_map_by_ssrc(SrsGb28181RtmpMuxer*muxer, uint32_t ssrc)
-{
- if (rtmpmuxers_ssrc.find(ssrc) == rtmpmuxers_ssrc.end()) {
- rtmpmuxers_ssrc[ssrc] = muxer;
- }
-}
-
-void SrsGb28181Manger::rtmpmuxer_unmap_by_ssrc(uint32_t ssrc)
-{
- std::map::iterator it = rtmpmuxers_ssrc.find(ssrc);
- if (it != rtmpmuxers_ssrc.end()) {
- rtmpmuxers_ssrc.erase(it);
- }
-}
-
-void SrsGb28181Manger::destroy()
-{
- if (!config->rtp_mux_tcp_enable) {
- //destory ps rtp listen
- std::map::iterator it;
- for (it = rtp_pool.begin(); it != rtp_pool.end(); ++it) {
- SrsPsRtpListener* listener = it->second;
- srs_freep(listener);
- }
- rtp_pool.clear();
- }
-
- //destory gb28181 muxer
- std::map::iterator it;
- for (it = rtmpmuxers.begin(); it != rtmpmuxers.end(); ++it) {
- SrsGb28181RtmpMuxer* muxer = it->second;
- SrsGb28181StreamChannel sess = muxer->get_channel();
- rtmpmuxer_unmap_by_ssrc(sess.get_ssrc());
- manager->remove(muxer);
- }
- rtmpmuxers.clear();
-}
-
-void SrsGb28181Manger::remove(SrsGb28181RtmpMuxer* muxer)
-{
- std::string id = muxer->get_channel_id();
-
- map::iterator it = rtmpmuxers.find(id);
- if (it != rtmpmuxers.end()) {
- SrsGb28181RtmpMuxer* muxer = it->second;
- SrsGb28181StreamChannel sess = muxer->get_channel();
- rtmpmuxer_unmap_by_ssrc(sess.get_ssrc());
- rtmpmuxers.erase(it);
- }
- manager->remove(muxer);
-}
-
-void SrsGb28181Manger::remove_sip_session(SrsGb28181SipSession* sess)
-{
- manager->remove(sess);
-}
-
-srs_error_t SrsGb28181Manger::start_ps_rtp_listen(std::string id, int port)
-{
- srs_error_t err = srs_success;
- if (port == config->rtp_mux_port) {
- return srs_error_wrap(err, "start rtp listen port is mux port");
- }
-
- /* delete by xbpeng 20201222 should not check rtmpmuxers, becasue it always not find*/
- // map::iterator key = rtmpmuxers.find(id);
- // if (key == rtmpmuxers.end()){
- // srs_warn("start rtp listen port rtmp muxer is null. id=%s,port=%d", id.c_str(),port);
- // return srs_error_wrap(err, "start rtp listen port rtmp muxer is null");
- // }
-
- if (!config->rtp_mux_tcp_enable) {
- if (rtp_pool.find(port) == rtp_pool.end())
- {
- SrsPsRtpListener* rtp = new SrsPsRtpListener(this->config, port, id);
- rtp_pool[port] = rtp;
- if ((err = rtp_pool[port]->listen()) != srs_success) {
- stop_rtp_listen(id);
- return srs_error_wrap(err, "rtp listen");
- }
-
- srs_trace("gb28181: start rtp ps stream over server-port=%d", port);
- }
- }
-
- return err;
-}
-
-void SrsGb28181Manger::stop_rtp_listen(std::string id)
-{
- map::iterator it = rtmpmuxers.find(id);
- if (it == rtmpmuxers.end()){
- return;
- }
-
- SrsGb28181RtmpMuxer* muxer = it->second;
- SrsGb28181StreamChannel sess = muxer->get_channel();
-
- int port = sess.get_rtp_port();
- if (port == config->rtp_mux_port) {
- return;
- }
-
- if (!config->rtp_mux_tcp_enable) {
- map::iterator it2 = rtp_pool.find(port);
- if (it2 != rtp_pool.end()) {
- srs_freep(it2->second);
- rtp_pool.erase(it2);
- }
- }
- free_port(port, port+1);
-}
-
-//api
-srs_error_t SrsGb28181Manger::create_stream_channel(SrsGb28181StreamChannel *channel)
-{
- srs_error_t err = srs_success;
- srs_assert(channel);
-
- std::string id = channel->get_channel_id();
- SrsGb28181RtmpMuxer *muxer = NULL;
-
- muxer = fetch_rtmpmuxer(id);
- if (muxer){
- SrsGb28181StreamChannel s = muxer->get_channel();
- channel->copy(&s);
- //return ERROR_GB28181_SESSION_IS_EXIST;
- return err;
- }
-
- //Start RTP listening port, receive gb28181 stream,
- //fixed is mux port,
- //random is random allocation port
- int rtp_port = 0;
- std::string port_mode = channel->get_port_mode();
-
- if (port_mode.empty()){
- port_mode = RTP_PORT_MODE_FIXED;
- channel->set_port_mode(port_mode);
- }
-
- if (port_mode == RTP_PORT_MODE_RANDOM){
- alloc_port(&rtp_port);
- if (rtp_port <= 0){
- return srs_error_new(ERROR_GB28181_RTP_PORT_FULL, "gb28181: rtp port full");
- }
-
- if ((err = start_ps_rtp_listen(id, rtp_port)) != srs_success){
- free_port(rtp_port, rtp_port + 1);
- return err;
- }
- }
- else if(port_mode == RTP_PORT_MODE_FIXED) {
- rtp_port = config->rtp_mux_port;
- }
- else{
- return srs_error_new(ERROR_GB28181_PORT_MODE_INVALID, "gb28181: port mode invalid");
- }
-
- uint32_t ssrc = channel->get_ssrc();
- if (ssrc == 0){
- //auto generate SSRC according to the hash code,
- //of the string value of the id
- ssrc = generate_ssrc(id);
- }
-
- //generate RTMP push stream address,
- //if the app and stream in the API are empty,
- //RTMP URL is generated using the output template parameter
- std::string url = "";
- int rtmp_port;
- string app = channel->get_app();
- string stream = channel->get_stream();
-
- SrsRequest request;
-
- if (true) {
- string tcUrl, stream_name;
-
- //get template rtmp url configuration
- std::string output = config->output;
- srs_parse_rtmp_url(output, tcUrl, stream_name);
-
- string _schema, _host, _vhost, _param, _app, _stream;
- srs_discovery_tc_url(tcUrl, _schema, _host, _vhost, _app, _stream, rtmp_port, _param);
-
- //if the stream name is not parameterized,
- //it needs to be parameterized to ensure that the stream name is different
- if (!srs_string_contains(stream_name, "[stream]") &&
- !srs_string_contains(stream_name, "[timestamp]") &&
- !srs_string_contains(stream_name, "[ssrc]")){
- stream_name = stream_name + "_[stream]";
- }
-
- if (app.empty()){
- app = _app;
- }
-
- if (stream.empty())
- {
- stream = stream_name;
- }
-
- url = srs_generate_rtmp_url(_host, rtmp_port, "", "", app, stream, "");
- url = srs_string_replace(url, "[app]", "live");
- url = srs_string_replace(url, "[stream]", id);
- std::stringstream ss;
- ss << ssrc;
- url = srs_string_replace(url, "[ssrc]", ss.str());
- url = srs_path_build_timestamp(url);
-
- //update channel app stream value
- srs_parse_rtmp_url(url, tcUrl, stream_name);
- srs_discovery_tc_url(tcUrl, _schema, _host, _vhost, _app, _stream, rtmp_port, _param);
-
- //generate the value returned to the api response
- channel->set_rtp_port(rtp_port);
- channel->set_ssrc(ssrc);
-
- channel->set_app(_app);
- channel->set_stream(stream_name);
- 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;
- //request.vhost = config->host;
- }
-
- //create on rtmp muxer, gb28181 stream to rtmp
- if ((err = fetch_or_create_rtmpmuxer(id, &request, &muxer)) != srs_success){
- srs_warn("gb28181: create rtmp muxer error, %s", srs_error_desc(err).c_str());
- return err;
- }
-
- rtmpmuxer_map_by_ssrc(muxer, ssrc);
- muxer->set_rtmp_url(url);
- srs_trace("gb28181: create new stream channel id:%s rtmp url=%s", id.c_str(), url.c_str());
-
- muxer->copy_channel(channel);
-
- return err;
-}
-
-srs_error_t SrsGb28181Manger::delete_stream_channel(std::string id, std::string chid)
-{
- srs_error_t err = srs_success;
-
- //notify the device to stop streaming
- //if an internal sip service controlled channel
- notify_sip_bye(id, chid);
-
- string channel_id = id + "@" + chid;
-
- SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(channel_id);
- if (muxer){
- stop_rtp_listen(channel_id);
- muxer->stop();
- return err;
- }else {
- return srs_error_new(ERROR_GB28181_SESSION_IS_NOTEXIST, "stream channel is not exists");
- }
-}
-
-
-srs_error_t SrsGb28181Manger::query_stream_channel(std::string id, SrsJsonArray* arr)
-{
- srs_error_t err = srs_success;
-
- if (!id.empty()){
- SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(id);
- if (!muxer){
- return srs_error_new(ERROR_GB28181_SESSION_IS_NOTEXIST, "stream channel not exists");
- }
- SrsJsonObject* obj = SrsJsonAny::object();
- arr->append(obj);
- muxer->get_channel().dumps(obj);
- }else {
- std::map::iterator it2;
- for (it2 = rtmpmuxers.begin(); it2 != rtmpmuxers.end(); ++it2) {
- SrsGb28181RtmpMuxer* muxer = it2->second;
- SrsJsonObject* obj = SrsJsonAny::object();
- arr->append(obj);
- muxer->get_channel().dumps(obj);
- }
- }
-
- return err;
-}
-
-srs_error_t SrsGb28181Manger::notify_sip_invite(std::string id, std::string ip, int port, uint32_t ssrc, std::string chid)
-{
- srs_error_t err = srs_success;
-
- if (!sip_service){
- return srs_error_new(ERROR_GB28181_SIP_NOT_RUN, "sip not run");
- }
-
- //if RTMP Muxer does not exist, you need to create
- std::string key = id+"@"+chid;
- SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(key);
-
- if (!muxer){
- //if there is an invalid parameter, the channel will be created automatically
- if (ip.empty() || port == 0 || ssrc == 0){
- //channel not exist
- SrsGb28181StreamChannel channel;
- channel.set_channel_id(key);
- if (!this->config->sip_invite_port_fixed) {
- channel.set_port_mode(RTP_PORT_MODE_RANDOM);
- }else
- {
- channel.set_port_mode(RTP_PORT_MODE_FIXED);
- }
- err = create_stream_channel(&channel);
- if (err != srs_success){
- return err;
- }
-
- ip = channel.get_ip();
- port = channel.get_rtp_port();
- ssrc = channel.get_ssrc();
- }
- }else {
- //channel exit, use channel config
- SrsGb28181StreamChannel channel = muxer->get_channel();
- ip = channel.get_ip();
- port = channel.get_rtp_port();
- ssrc = channel.get_ssrc();
- }
-
- SrsSipRequest req;
- req.sip_auth_id = id;
- return sip_service->send_invite(&req, ip, port, ssrc, chid);
-}
-
-srs_error_t SrsGb28181Manger::notify_sip_bye(std::string id, std::string chid)
-{
- if (!sip_service){
- return srs_error_new(ERROR_GB28181_SIP_NOT_RUN, "sip not run");
- }
-
- SrsSipRequest req;
- req.sip_auth_id = id;
- return sip_service->send_bye(&req, chid);
-}
-
-srs_error_t SrsGb28181Manger::notify_sip_ptz(std::string id, std::string chid, std::string cmd,
- uint8_t speed, int priority)
-{
- if (!sip_service){
- return srs_error_new(ERROR_GB28181_SIP_NOT_RUN, "sip not run");
- }
-
- SrsSipRequest req;
- req.sip_auth_id = id;
- return sip_service->send_ptz(&req, chid, cmd, speed, priority);
-}
-
-srs_error_t SrsGb28181Manger::notify_sip_raw_data(std::string id, std::string data)
-{
- if (!sip_service){
- return srs_error_new(ERROR_GB28181_SIP_NOT_RUN, "sip not run");
- }
-
- SrsSipRequest req;
- req.sip_auth_id = id;
- return sip_service->send_sip_raw_data(&req, data);
-
-}
-
-srs_error_t SrsGb28181Manger::notify_sip_unregister(std::string id)
-{
- if (!sip_service){
- return srs_error_new(ERROR_GB28181_SIP_NOT_RUN, "sip not run");
- }
- sip_service->remove_session(id);
- return srs_success;
- // useless, because
- // sip session has been removed
- // id is not channel id like id@chid
- //return delete_stream_channel(id);
-}
-
-srs_error_t SrsGb28181Manger::notify_sip_query_catalog(std::string id)
-{
- if (!sip_service){
- return srs_error_new(ERROR_GB28181_SIP_NOT_RUN, "sip not run");
- }
-
- SrsSipRequest req;
- req.sip_auth_id = id;
- SrsGb28181SipSession *sip_session = sip_service->fetch(req.sip_auth_id);
- if (sip_session) {
- sip_session->item_list.clear();
- sip_session->clear_device_list();
- srs_trace("notify_sip_query_catalog, clear sip session item and device list");
- }
- return sip_service->send_query_catalog(&req);
-}
-
-srs_error_t SrsGb28181Manger::query_sip_session(std::string id, SrsJsonArray* arr)
-{
- if (!sip_service){
- return srs_error_new(ERROR_GB28181_SIP_NOT_RUN, "sip not run");
- }
-
- return sip_service->query_sip_session(id, arr);
-}
-
-srs_error_t SrsGb28181Manger::query_device_list(std::string id, SrsJsonArray* arr)
-{
- if (!sip_service){
- return srs_error_new(ERROR_GB28181_SIP_NOT_RUN, "sip not run");
- }
-
- return sip_service->query_device_list(id, arr);
-}
-#define SRS_RTSP_BUFFER 8192
-#define RTP_TCP_HEADER 2
-#define MAX_PACKAGE_SIZE 1024 * 10
-SrsGb28181Conn::SrsGb28181Conn(SrsGb28181Caster* c, srs_netfd_t fd, SrsGb28181PsRtpProcessor *rtp_processor)
-{
- caster = c;
- stfd = fd;
- skt = new SrsStSocket();
- rtsp = new SrsRtspStack(skt);
- trd = new SrsSTCoroutine("rtsp", this);
- mbuffer = (char*)malloc(SRS_RTSP_BUFFER);
- processor = rtp_processor;
-}
-
-SrsGb28181Conn::~SrsGb28181Conn()
-{
- free(mbuffer);
- srs_close_stfd(stfd);
-
- srs_freep(trd);
- srs_freep(skt);
- srs_freep(rtsp);
-}
-
-srs_error_t SrsGb28181Conn::serve()
-{
- srs_error_t err = srs_success;
-
- if ((err = skt->initialize(stfd)) != srs_success) {
- return srs_error_wrap(err, "socket initialize");
- }
-
- if ((err = trd->start()) != srs_success) {
- return srs_error_wrap(err, "rtsp connection");
- }
- return err;
-}
-
-std::string SrsGb28181Conn::remote_ip()
-{
- // TODO: FIXME: Implement it.
- return "";
-}
-
-srs_error_t SrsGb28181Conn::do_cycle()
-{
- srs_error_t err = srs_success;
-
- // retrieve ip of client.
- int fd = srs_netfd_fileno(stfd);
- std::string ip = srs_get_peer_ip(fd);
- int port = srs_get_peer_port(fd);
- int addr_len = sizeof(sockaddr_in);
- sockaddr_in *peer_sockaddr = (sockaddr_in*)malloc(addr_len);
- peer_sockaddr->sin_family = AF_INET; //设置地址家族
- peer_sockaddr->sin_port = htons(port); //设置端口
- peer_sockaddr->sin_addr.s_addr = inet_addr(ip.c_str());
-
-
- if (ip.empty() && !_srs_config->empty_ip_ok()) {
- srs_warn("empty ip for fd=%d", srs_netfd_fileno(stfd));
- }
- srs_trace("gb28181 new connect by rtp-tcp from: %s:%d", ip.c_str(), port);
-
- uint32_t left_data_len = 0; //缓存剩余数据
- ssize_t nb_read = 0;
- uint16_t packet_len = 0; //rtp包长度
-
- // consume all rtp data.
- while (true) {
- if ((err = trd->pull()) != srs_success) {
- return srs_error_wrap(err, "rtsp cycle");
- }
- nb_read = 0;
- if ((err = skt->read(mbuffer + left_data_len, SRS_RTSP_BUFFER - left_data_len, &nb_read)) != srs_success) {
- return srs_error_wrap(err, "recv data");
- }
-
- left_data_len = nb_read + left_data_len;
- char * buf = mbuffer;
-
- uint32_t index = 0;
- for( ; index < left_data_len; ){
- if (index + RTP_TCP_HEADER >= left_data_len){ //less rtp package
- break;
- }
- packet_len = (((uint8_t *) buf)[index] << 8) | ((uint8_t *) buf)[index + 1];
- if (packet_len > MAX_PACKAGE_SIZE){
- //FIXME 自动重新invite?
- srs_error("abnormal RTP packet length:%d, close the tcp conn:%s", packet_len, remote_ip().c_str());
- return err;
- }
- if (index + RTP_TCP_HEADER + packet_len >= left_data_len){
- break;
- }
- processor->on_tcp_packet((sockaddr*)peer_sockaddr, addr_len, buf + index + RTP_TCP_HEADER, packet_len);
- index = index + RTP_TCP_HEADER + packet_len;
- }
- if (index != 0) { //update left data
- left_data_len = left_data_len - index;
- memmove(mbuffer, buf + index, left_data_len);
- }
-
- }
- free(peer_sockaddr);
- return err;
-}
-
-srs_error_t SrsGb28181Conn::cycle()
-{
- // serve the rtsp client.
- srs_error_t err = do_cycle();
-
- caster->remove(this);
-
- if (err == srs_success) {
- srs_trace("client finished.");
- }
- else if (srs_is_client_gracefully_close(err)) {
- srs_warn("client disconnect peer. code=%d", srs_error_code(err));
- srs_freep(err);
- }
-
- return err;
-}
-
-std::string SrsGb28181Conn::desc()
-{
- return "GB28181TcpConn";
-}
-
-const SrsContextId& SrsGb28181Conn::get_id()
-{
- return trd->cid();
-}
-
-SrsGb28181Caster::SrsGb28181Caster(SrsConfDirective* c)
-{
- // TODO: FIXME: support reload.
- output = _srs_config->get_stream_caster_output(c);
- config = new SrsGb28181Config(c);
- rtp_processor = new SrsGb28181PsRtpProcessor(config, "");
- manager = new SrsResourceManager("GB28181TCP", true);
-}
-
-SrsGb28181Caster::~SrsGb28181Caster()
-{
- std::vector::iterator it;
- for (it = clients.begin(); it != clients.end(); ++it) {
- SrsGb28181Conn* conn = *it;
- manager->remove(conn);
- }
- clients.clear();
-
- srs_freep(manager);
-}
-
-srs_error_t SrsGb28181Caster::initialize()
-{
- srs_error_t err = srs_success;
- if ((err = manager->start()) != srs_success) {
- return srs_error_wrap(err, "start manager");
- }
- return err;
-}
-
-srs_error_t SrsGb28181Caster::on_tcp_client(srs_netfd_t stfd)
-{
- srs_error_t err = srs_success;
-
- SrsGb28181Conn* conn = new SrsGb28181Conn(this, stfd, rtp_processor);
-
- if ((err = conn->serve()) != srs_success) {
- srs_freep(conn);
- return srs_error_wrap(err, "serve conn");
- }
-
- clients.push_back(conn);
-
- return err;
-}
-
-void SrsGb28181Caster::remove(SrsGb28181Conn* conn)
-{
- std::vector::iterator it = find(clients.begin(), clients.end(), conn);
- if (it != clients.end()) {
- clients.erase(it);
- }
- srs_info("rtsp: remove connection from caster.");
-
- manager->remove(conn);
-}
-
diff --git a/trunk/src/app/srs_app_gb28181.hpp b/trunk/src/app/srs_app_gb28181.hpp
deleted file mode 100644
index ba28056a1..000000000
--- a/trunk/src/app/srs_app_gb28181.hpp
+++ /dev/null
@@ -1,581 +0,0 @@
-//
-// Copyright (c) 2013-2021 Lixin
-//
-// SPDX-License-Identifier: MIT
-//
-
-#ifndef SRS_APP_GB28181_HPP
-#define SRS_APP_GB28181_HPP
-
-#include
-
-#include
-#include
-#include
-#include
-#include