Johnny
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82bc416546
|
add exchangeSdp
|
2023-04-21 20:40:37 +08:00 |
|
Johnny
|
5d33e4c9f9
|
refine: update static_cast in webrtc api
|
2023-04-21 20:24:23 +08:00 |
|
xia-chu
|
9443d68d6c
|
格式化代码
|
2023-04-18 10:33:22 +08:00 |
|
xiongguangjie
|
f949c6de2a
|
not retry when hook result code is int and !=0
|
2023-04-17 20:01:45 +08:00 |
|
xiongziliang
|
24eaaf68fb
|
初步支持webrtc whip/whep(推拉流)协议
whip推流地址: /index/api/whip?app=live&stream=test
whep拉流地址: /index/api/whep?app=live&stream=test
|
2023-04-08 21:44:08 +08:00 |
|
xiongziliang
|
23f9a42f72
|
格式化与精简代码
|
2023-04-01 23:59:13 +08:00 |
|
xiongguangjie
|
876aea33f5
|
avoid fps too big
|
2023-03-23 18:14:28 +08:00 |
|
Leonnash
|
cf342a6fdf
|
Update WebApi.cpp
|
2023-03-22 15:52:56 +08:00 |
|
Derek Liu
|
6008ae157a
|
修复addFFmepgSource接口参数非法时无法清除记录表的bug (#2305)
捕获addFFmepgSource接口参数dst_url解析错误的抛错,通过错误回调返回返回错误,清除s_ffmpegMap表中的无效KEY
|
2023-03-14 19:38:24 +08:00 |
|
ziyue
|
1f2ef82b46
|
新增支持获取gop大小与间隔信息: #1570
getMediaList/getMediaInfo接口、on_media_changed hook新增支持字段如下:
{
"codec_id" : 0,
"codec_id_name" : "H264",
"codec_type" : 0,
"fps" : 0.0,
"frames" : 1119, #累计接收帧数,不包含sei/aud/sps/pps等不能解码的帧
"gop_interval_ms" : 1993, #gop间隔时间,单位毫秒
"gop_size" : 60, #gop大小,单位帧数
"height" : 556,
"key_frames" : 21, #累计接收关键帧数
"ready" : true,
"width" : 990
}
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2023-02-20 16:11:10 +08:00 |
|
xiongziliang
|
5bcfba1da4
|
startSendRtpPassive接口新增连接超时参数:close_delay_ms
|
2023-02-17 23:02:09 +08:00 |
|
xiongziliang
|
8f0ba6988b
|
openRtpServer接口新增only_audio参数,优化语音对讲场景
|
2023-02-17 22:48:39 +08:00 |
|
xiongziliang
|
6eb36ec883
|
获取MediaSource信息相关接口返回帧数相关字段:#1570
|
2023-02-05 22:04:14 +08:00 |
|
xiongziliang
|
fcf70c6ef1
|
startSendRtpPassive接口支持同时接收流
|
2023-01-08 21:24:29 +08:00 |
|
xiongziliang
|
0374e7a660
|
startSendRtp接口支持同时接收流:#2109,#2149
|
2023-01-07 22:36:30 +08:00 |
|
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 |
|
xiongziliang
|
50fa671564
|
修复析构中调用getOwnerPoller抛异常导致崩溃的bug:#2117
|
2022-11-26 10:16:47 +08:00 |
|
ziyue
|
a9e53aae70
|
Merge branch 'master' of https://gitee.com/xia-chu/ZLMediaKit
|
2022-11-19 09:38:44 +08:00 |
|
ziyue
|
68948288e0
|
TcpSession/UdpSession统一为Session类
|
2022-11-19 09:33:10 +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 |
|
xiongziliang
|
3fdd5a86c9
|
MediaServer -v 打印代码日期
|
2022-11-13 00:13:02 +08:00 |
|
夏楚
|
a37268f003
|
格式化代码
|
2022-11-12 01:52:49 +00:00 |
|
xiongguangjie
|
bc63142712
|
add rtp server timeout hook
|
2022-11-10 16:58:02 +08:00 |
|
xiongziliang
|
44fd6b86bc
|
完善版本信息
|
2022-11-06 00:38:14 +08:00 |
|
monktan89
|
7e95bd2078
|
修复MSVC编译问题
|
2022-11-03 10:51:49 +08:00 |
|
xiongziliang
|
9498b96b95
|
确保rtp推流线程安全性
|
2022-10-30 21:36:35 +08:00 |
|
xiongziliang
|
c25e93fee3
|
解决启动ffmpeg进程导致shell终端假死的问题:#1662
|
2022-10-29 17:44:55 +08:00 |
|
ziyue
|
7d251e15b3
|
on_publish hook兼容非标准回复
|
2022-10-20 11:00:19 +08:00 |
|
ziyue
|
eac5a5b1dc
|
使用submodule方式添加jsoncpp源码
|
2022-10-16 21:10:18 +08:00 |
|
xiongziliang
|
a916760ac3
|
整理webrtc c接口
|
2022-10-06 12:35:14 +08:00 |
|
Dw9
|
43bf7c7918
|
c api support srt server
|
2022-09-22 21:18:34 +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 |
|
xiongziliang
|
12551be33c
|
提炼ProtocolOption赋值相关逻辑
|
2022-09-16 23:31:37 +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 |
|
custompal
|
d853075175
|
RtpServer新增tcp主动模式支持 (#1938)
|
2022-09-09 10:56:28 +08:00 |
|
huangxiuqi
|
258a4dd166
|
C API和WebHook未找到流回调添加直接关闭机制 (#1948)
|
2022-09-09 10:55:35 +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
|
00c9749b5d
|
防止多个track时获取rtp推流丢包率失败
|
2022-09-03 16:47:37 +08:00 |
|
PioLing
|
0948a3df31
|
支持在addStreamProxy和on_publish中控制单个流是否开启时间戳覆盖 (#1930)
|
2022-09-03 09:54:09 +08:00 |
|
custompal
|
dd6495cc07
|
补充getMediaPlayerList接口注释及postman示例
|
2022-09-02 17:46:09 +08:00 |
|
custompal
|
33e1e6b88d
|
getMediaPlayerList返回播放器id以及会话类型名称
|
2022-09-01 21:52:43 +08:00 |
|
custompal
|
d0214a13e1
|
防止getPlayerList返回的json数据为null
|
2022-09-01 17:45:06 +08:00 |
|
custompal
|
38170c702e
|
修正gcc4.8编译错误
|
2022-09-01 17:33:36 +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 |
|
xiongziliang
|
6a4297845f
|
新增发送rtp被动关闭hook
|
2022-08-27 10:53:47 +08:00 |
|
xiongziliang
|
9f0c15a4f0
|
startSendRtp接口支持rtcp接收超时主动停止
|
2022-08-20 12:48:27 +08:00 |
|
Dw9
|
30984d2076
|
mp4录制支持作为观看者参与播放人数统计 (#1880)
|
2022-08-16 11:47:24 +08:00 |
|
xiongguangjie
|
a1000da71f
|
add get version restful api
|
2022-08-12 18:09:44 +08:00 |
|