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388 Commits

Author SHA1 Message Date
mtdxc
754073918a
Header refactor (#2115) 
* 优化MultiMediaSourceMuxer头文件包含

* 将MediaSinkDelegate和Demux移到MediaSink中

* MediaSource头文件重构, 独立出PacketCache.h
精简Frame和Track的头文件

* Rtmp头文件重构

* Rtsp头文件重构

* webrtc头文件重构

* 规范.h头文件包含，并将其移到.cpp中:
- 尽量不包含Common\config.h
- Util\File.h
- Rtsp/RtspPlayer.h
- Rtmp/RtmpPlayer.h

* 删除多余的Stamp.h和Base64包含
 2022-11-29 11:07:13 +08:00
Johnny
59ecdd13e5 fix: WebRtcTransport 切换 poller 线程时，创建 WebRtcSession 不需要加锁 2022-11-28 17:54:16 +08:00
johzzy
bc4fa2bc53
fix(rtcp/bye): 收到rtcp bye时不断开webrtc连接 (#2123) 
bye(203) 会在 sender audio track mute 时出现, 不能作为 shutdown 的依据
 2022-11-28 17:52:37 +08:00
Dw9
fef9d31631
webrtc支持通过http参数指定是否优先tcp模式 (#2105) 
* webrtc push/play支持通过http参数指定tcp

* force_tcp改成perferred_tcp

Co-authored-by: xiongziliang <771730766@qq.com>
 2022-11-27 12:43:16 +08:00
ziyue
9fd5152aa1 确保webrtc关闭udp模式时可用 2022-11-19 09:52:10 +08:00
ziyue
d910c80a00 优化webrtc iceandidate端口相关代码 2022-11-19 09:46:04 +08:00
xiongguangjie
0d6fa1281a add rtc tcp port config and ignore candidate when port is 0 2022-11-19 01:51:53 +08:00
Dw9
47530ce830
新增支持webrtc over tcp模式 (#2092) 
* webrtc server/session/cadidate 改为tcp

* 先屏蔽检查isCurrentThread

* 接受和发送的数据处理tcp 2字节头

* 处理rtc tcp 分片

* 完善webrtc over tcp

* 精简rtp服务器相关代码

* 适配webrtc AV1编码: #2091

* webrtc tcp模式支持Firefox

* webrtc tcp模式支持线程安全

* c sdk支持webrtc tcp

Co-authored-by: ziyue <1213642868@qq.com>
 2022-11-18 22:52:57 +08:00
ziyue
fc433de9ac 适配webrtc AV1编码: #2091 2022-11-18 16:47:02 +08:00
ziyue
520945c2e9 提高线程安全性 2022-11-01 10:53:38 +08:00
baiyfcu
4a51d6b488 修复第三方引用ZLM编译webrtc避免生成dll文件 2022-10-17 10:53:39 +08:00
xiongziliang
a916760ac3 整理webrtc c接口 2022-10-06 12:35:14 +08:00
ziyue
0b355759de 整理webrtc相关代码命名空间 2022-09-18 21:03:05 +08:00
ziyue
15affeff1d 优化关闭媒体源相关逻辑: #1963 2022-09-18 20:36:47 +08:00
xiongguangjie
d04a6d4355 rtc receive sr reset alive ticker 2022-09-09 23:04:28 +08:00
夏楚
4a35ddbddb
Merge pull request #1942 from mtdxc/reduce_code 
简化代码
 2022-09-09 11:10:55 +08:00
xiongguangjie
7f86299c80 rtc stop push unregister mediasource 2022-09-07 19:10:06 +08:00
xiongguangjie
96c19ac85f 修复webrtcsession getIdentifier 为空的bug,导致GetAllSession webrtcsession的id 为空 2022-09-07 13:58:26 +08:00
cqm
999e0b274e 简化代码: 
- MediaSource引入shortUrl和getUrl来简化日志输出
- WebApi引入fillSockInfo
 2022-09-07 11:47:15 +08:00
xiongziliang
20f1275c58 rtsp拉流、rtp单端口推流新增支持获取丢包率: #1877 2022-09-03 16:32:09 +08:00
custompal
0d6509ce03 统一部分变量名和代码样式风格 2022-08-30 21:24:25 +08:00
custompal
04aa3ef41f 增加获取媒体流播放器列表功能 2022-08-30 21:05:19 +08:00
xiongzilaing
679c79802f 整理媒体事件相关代码，删除无须重载时的多余代码逻辑 2022-08-28 17:25:56 +08:00
Dw9
b7ef766701
支持单独的datachannel通道 (#1894) 
* 支持单独的datachannel 通道

* 当仅有datachannel时 ，忽略rtp和rtcp超时

* 单独开启datachannel时，通过dtls包维持心跳

Co-authored-by: xiongziliang <771730766@qq.com>
 2022-08-20 10:28:19 +08:00
gongluck
0764e48f6c
兼容安卓平台，to_string模板处理pt值 2022-08-18 19:13:22 +08:00
ziyue
ba947cb6b5 调整switch case语句缩进 2022-08-05 17:42:17 +08:00
Xiaofeng Wang
c14a777598 修正启用 SCTP 后编译问题 2022-07-28 23:15:22 +08:00
Xiaofeng Wang
0fc38bbadb 避免对外传递 srtp 依赖 2022-07-26 00:04:03 +08:00
Xiaofeng Wang
9d3ead61de 整理 CMakeLists.txt 2022-07-25 00:22:30 +08:00
xiongguangjie
cffc0743d7 format code use clang-format 2022-07-14 22:49:31 +08:00
xiongguangjie
602b475a3c for rtc send can receive rtcp xr, player can calculate rtt , for metaRTC player send rtcp xr 2022-07-12 19:26:17 +08:00
Haibo Chen
576056c465
get ip from env, for externIP in file 'config.ini' (#1786) 2022-07-09 21:44:57 +08:00
bin-work
2e1a45895a
修改webrtc码率带宽约束字段 (#1771, #1768) 2022-07-07 16:44:42 +08:00
xiongziliang
2490fd2115 MS_ASSERT改成抛异常 2022-06-18 21:09:57 +08:00
xiongziliang
94839fd93c 添加bom头 2022-06-18 14:34:58 +08:00
xiongziliang
df7a3242bb 修复获取webrtc推流丢包率除0的bug 2022-06-11 14:23:43 +08:00
xiongziliang
b6d3ec5251 完善获取丢包率pull request; 新增MediaSource::getOwnerPoller接口 2022-06-11 12:56:03 +08:00
Leon
be995f9cd2 !17 【功能请求】 /index/api/getMediaList接口 增加LOSS字段用于统计每个轨道丢包率反馈 
* update webrtc/WebRtcPusher.h.
* update webrtc/WebRtcPusher.cpp.
* update webrtc/WebRtcTransport.h.
* update webrtc/WebRtcTransport.cpp.
* update src/Common/MediaSource.h.
* update src/Common/MediaSource.cpp.
* update server/WebApi.cpp.
 2022-06-11 04:31:06 +00:00
sha-long-ba-si
3a9408bcec
webrtc answer m-lines次序错误 (#1695) 
Co-authored-by: james <mrjamesli@126.com>
 2022-06-11 12:26:31 +08:00
xiongziliang
b23cbaa0f8 on_publish hook新增continue_push_ms参数，用于断连续推延时控制 2022-06-04 11:06:35 +08:00
xiongziliang
b0f0bdb6ae 调整优化webrtc sdp处理相关代码 2022-05-20 22:14:21 +08:00
cqm
2b29e58f71 sdp代码重构, 增加addItem和addAttr 2022-05-18 18:56:09 +08:00
cqm
9d9315d8c6 支持SDP munging方式simulicast 2022-05-18 09:21:21 +08:00
xiongziliang
4f768cacf1 修复编译警告 2022-05-14 18:46:50 +08:00
xiongziliang
b0f15c9647 sctp关闭编译时,关闭datachannel相关功能 2022-05-13 20:45:27 +08:00
xiongziliang
7e58d91a57 修复rtp two byte ext解析相关bug: #1631 2022-05-13 20:01:57 +08:00
mtdxc
301cbf0a83
支持多个rtc候选地址 (#1622) 
* 支持多个rtc候选地址

* fixed missing extern_ips check
 2022-05-12 15:21:09 +08:00
xiongziliang
d66845425c 完善ipv6网络适配 2022-05-08 00:26:01 +08:00
xiongziliang
fb003714e9 新增webrtc datachannel功能: #1216 2022-05-04 22:15:21 +08:00
xiongziliang
1ed793fe97 初步实现webrtc datachannel sdp握手 2022-04-03 17:12:23 +08:00
xiongziliang
03dfcbad36 优化webrtc sdp处理相关代码 2022-04-03 17:10:39 +08:00
xiongziliang
b0af3334c1 修复firefox上webrtc echo测试无法通过的问题: #1493 2022-03-26 22:11:50 +08:00
xiongziliang
4314006551 调整nack相关参数 2022-03-13 19:22:19 +08:00
xiongziliang
aef2ae65af 优化代码 2022-03-12 19:29:58 +08:00
xiongziliang
b8277974aa webrtc sdp msid不存在时，不添加 2022-03-12 19:27:32 +08:00
xiongziliang
4dc621e1bb 转协议选项抽象为ProtocolOption对象 2022-03-12 13:24:23 +08:00
zx
cd4e772985
remove dumplicate line rtc_media.type 
remove  dumplicate line 

`rtc_media.type = mline.type;`
 2022-03-04 16:18:26 +08:00
ziyue
d88fe077ed on_publish hook新增originType字段 2022-03-02 18:03:44 +08:00
xiongziliang
7f66e6b4bf 消除编译警告 2022-02-27 21:52:12 +08:00
xiongziliang
497bf289b6 完善rtc网络迁移日志打印 2022-02-27 21:40:32 +08:00
xiongziliang
f8ef5579ee 整理webrtc相关日志 2022-02-27 21:26:49 +08:00
xiongziliang
a0b464958c 优化rtp重发列队性能 2022-02-27 12:26:11 +08:00
xiongziliang
6707b13418 rtp nack重传列队长度限制为5秒 2022-02-27 02:08:32 +08:00
xiongziliang
ba5ca8f5c7 修复rtp seq重复导致rtp nack重传列队异常的bug 2022-02-27 02:07:25 +08:00
xiongziliang
7a5cf925b0 Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit 2022-02-27 01:03:49 +08:00
xiongziliang
b82cd5a500 确保ntp时间戳获取失败时，webrtc rtp重传缓存列队长度正常 2022-02-27 01:02:54 +08:00
夏楚
3a2a60c40a
防止开启模拟丢包代码时，生成twcc rtcp紊乱的bug 2022-02-27 00:33:39 +08:00
夏楚
db3df7b92e
修复ssrc大小端转换bug 2022-02-26 22:33:00 +08:00
ziyue
72caa43c97 主动关闭的流不支持断流续推: #1401 2022-02-09 14:31:40 +08:00
夏楚
c72cf4cbcc
整理命名空间 (#1409) 
* feat: remove using namespace mediakit in header files.

(cherry picked from commit d44aeb339a8a0e1f0455be82b21fe4b1b536299f)

* feat: remove using namespace mediakit in FFmpegSource.h

* feat: remove using namespace mediakit in RtpExt.h

* feat: remove using namespace mediakit in header files.

* feat: remove using namespace std in header files.

* feat: remove using namespace std in header files when zltoolkit remove std in header

* 补充命名空间

* 整理命名空间

* 整理命名空间2

* 修复macos ci

* 修复编译问题

* 修复编译问题2

* 修复编译问题3

Co-authored-by: Johnny <hellojinqiang@gmail.com>
Co-authored-by: Xiaofeng Wang <wasphin@gmail.com>
 2022-02-02 20:34:50 +08:00
xgj
341459fbf5 fix non rtc push rtc play g711 audio loss 2022-01-24 11:57:24 +08:00
ziyue
09ac333e35 webrtc支持断连续推 2022-01-10 17:43:28 +08:00
ziyue
5a592e6db6 优化循环池性能 2022-01-06 14:30:44 +08:00
ziyue
80d24d2884 修复Windows下编译问题 2021-12-31 14:24:15 +08:00
ziyue
7f6be9e118 Refine: 消除编译警告 2021-12-30 15:28:27 +08:00
ziyue
254bc6889e RTC: 链接迁移后确保老链接及时销毁 2021-12-15 15:55:43 +08:00
xgj
78d014cf51 fix chrome ufrag too short to play webrtc 2021-11-29 16:54:40 +08:00
xiongziliang
f45830cbb8 Refine: 优化webrtc对象追踪机制 2021-11-28 21:43:21 +08:00
xiongziliang
61c55f4030 Refine: 使用socket fd作为日志追踪id 2021-11-28 21:42:54 +08:00
xiongziliang
bc4453015f RTC: 降低rtx重传日志优先级 2021-11-27 22:37:15 +08:00
xiongziliang
8a1f979375 RTC: 修复twcc rtp ext seq回退处理逻辑bug 2021-11-27 22:35:51 +08:00
xiongziliang
8b3a5dda1c Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit 2021-11-27 21:34:51 +08:00
xiongziliang
90315ebce5 RTC: srtp发送减少一次内存拷贝，提高webrtc发送性能 2021-11-27 21:30:46 +08:00
xiongguangjie
c1b78629e1
fix rtc play signal stream only audio or video failed (#1239) 2021-11-25 20:42:51 +08:00
xiongziliang
dd30ab22cf Performance: 使用循环池优化webrtc分发性能 2021-11-21 11:36:07 +08:00
wxf
1cba3f4d58 修正 gcc 4.8 下编译问题 2021-11-19 17:04:24 +08:00
ziyue
afdc5e4267 Refine: 优化tcp/udp服务器异常管理断开机制 2021-11-19 15:26:33 +08:00
Johnny
7e6467615c refine: rename TwccContext::checkIfNeedSendTwcc to TwccContext::needSendTwcc. 2021-11-19 11:46:10 +08:00
Johnny
da0a7219e9 refactor: 梳理 rtx 处理逻辑 2021-11-19 11:46:10 +08:00
ziyue
fb97c2139a Refine: 优化webrtc sdp处理代码 2021-11-18 21:36:10 +08:00
ziyue
daa06ffd4f Refine: 精简webrtc sdp生成ssrc line相关代码 2021-11-18 21:25:14 +08:00
johzzy
f7963a9032
调整 addSSRCItem 函数 (#1224) 
* update readme.

* refactor(sdp): update addSSRCItem.

* refactor(sdp): remove addSSRCItem.

* refactor(sdp): 调整逻辑，生成a=ssrc-group:FID字段
 2021-11-18 20:55:18 +08:00
ziyue
950d957c86 RTC: 删除过多stun包日志 2021-11-15 10:40:31 +08:00
ziyue
a7d6e2ba38 RTC: webrtc偏好音视频codec可配置(#1214) 2021-11-15 10:33:16 +08:00
ziyue
6d384c4a1e 修复webrtc推流播放鉴权广播参数类型不匹配导致的bug 2021-10-21 11:05:15 +08:00
ziyue
7c016ad14b Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit 2021-10-20 17:02:26 +08:00
ziyue
758f1b414e webrtc是否允许发送rtp逻辑移至基类 2021-10-20 16:58:59 +08:00
夏楚
73aed907e7
Update readme.md 2021-10-20 14:49:56 +08:00
ziyue
5ee9b69568 webrtc新增自定义插件模式 2021-10-19 15:23:12 +08:00
ziyue
a3177de658 修复webrtc日志相关编译问题 2021-10-18 17:47:16 +08:00