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feature/transcode2
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418 Commits

Author SHA1 Message Date
cqm
89e21038c2 Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit into feature/transcode2 
# Conflicts:
#	src/Common/MediaSource.h
#	src/Common/config.h
#	src/Rtsp/RtspMediaSourceImp.h
#	src/Rtsp/RtspMediaSourceMuxer.h
#	src/Rtsp/RtspMuxer.cpp
#	webrtc/WebRtcTransport.cpp
 2024-09-22 11:40:39 +08:00
alex
4152dcd409
AI automatically translates all comments in the code into English (#3917) 2024-09-19 14:53:50 +08:00
cqm
d329f76edf Merge branch 'master' of https://gitee.com/xia-chu/ZLMediaKit into transcode2 
# Conflicts:
#	src/Common/config.cpp
#	src/Common/config.h
 2024-08-19 18:03:03 +08:00
xia-chu
a14e40f8d2 修复webrtc rtp重发列队可能由于rtp时间戳异常导致内存溢出的bug 2024-07-21 19:16:34 +08:00
cqm
e6838f24af fixed build issue 2024-07-15 08:47:31 +08:00
cqm
2ffdfab71e Merge branch 'master' of github.com:ZLMediaKit/ZLMediaKit into transcode2 
# Conflicts:
#	CMakeLists.txt
#	conf/config.ini
#	src/Common/MediaSink.cpp
#	src/Common/MediaSink.h
#	src/Common/MediaSource.cpp
#	src/Common/MultiMediaSourceMuxer.h
#	src/Common/config.cpp
#	src/Common/config.h
#	src/Extension/AAC.cpp
#	src/Extension/AAC.h
#	src/Rtsp/RtpCodec.h
#	src/Rtsp/RtspMuxer.cpp
#	src/Rtsp/RtspMuxer.h
#	webrtc/Nack.cpp
#	webrtc/WebRtcTransport.cpp
 2024-07-15 08:47:30 +08:00
xiongziliang
2ead272187 修复webrtc播放时生成config rtp线程安全的bug 
rtsp直接代理时会从config frame生成rtp，在获取VideoTrack配置帧列表时存在线程安全风险；
同时简化getConfigFrames函数代码，去除缓存逻辑。
 2024-06-22 13:04:06 +08:00
xiongziliang
1ef3642ed4 优化webrtc jitter buffer相关参数 2024-06-22 13:04:06 +08:00
Xiaofeng Wang
84dbe4b076
Send config frames once before sending DirectProxy RTP packets (#3611) 2024-06-16 09:37:33 +08:00
Luke
fbac3f9ba0
RC4密码套件漏洞 (#3590) 
禁止使用RC4
 2024-06-09 10:45:40 +08:00
gongluck
5df2b80757
Nack相关参数可配置化 (#3510 #3507) 2024-05-12 18:15:11 +08:00
xia-chu
2d66dd7cb3 Fixing the problem of WebRTC handshake failure caused by unsupported RTP extensions (#3486) 2024-04-20 22:49:27 +08:00
Dw9
edca662208
fix webrtc echo error (#3442) 
遵循着pr #3360 的修改方式解决webrtc echo模式失败的问题
 2024-04-05 22:11:50 +08:00
xiongziliang
7aaafa18e7 Format code 2024-03-23 23:08:10 +08:00
xiongziliang
d8893877b2 Delete invalid code 2024-03-23 22:56:12 +08:00
johzzy
029813402d
feat: update negotiateSdp and WebRtcArgs (#3371) 
- update negotiateSdp
- update HttpAllArgs and alias
- update onRtcConfigure
- define setWebRtcArgs, handle set_webrtc_cands and setLocalIp

---------

Co-authored-by: xiongziliang <771730766@qq.com>
Co-authored-by: KkemChen <kkemchen@qq.com>
 2024-03-23 22:46:30 +08:00
xia-chu
db4c570d19 WebRTC audio preferred PCMA 2024-03-22 20:50:07 +08:00
小强先生
8e16a698b6
降低webrtc握手未结束，数据先到的日志等级 (#3368) 2024-03-13 10:57:43 +08:00
johzzy
1e39594335
fix for https://bugs.chromium.org/p/webrtc/issues/detail?id=15845 (#3360) 2024-03-10 21:34:03 +08:00
gongluck
5a6364bae2
Add datachannel c apis and callbacks(#3328) 
增加datachannel数据收发的回调通知 #3326，和控制datachannel回显的开关

---------

Co-authored-by: xiongziliang <771730766@qq.com>
 2024-03-02 16:52:51 +08:00
waken
fd1ebb1a51
Auto get icecand ip address from http request (#3251) 
Get ip address from http `Host` header, and set it to icecand ip for webrtc
 2024-01-26 10:26:30 +08:00
老衲不出家
62641f426b
Fix a bug in WebRTC where the DTLS certificate is too large(big than MTU) (#3250) 2024-01-24 15:14:23 +08:00
sandro-qiang
aeda47ec8c
修正SctpAssociation.cpp的访问越界 (#3182) 
ZLMediaKit/ZLMediaKit#3181
 2024-01-05 15:33:51 +08:00
Dw9
7c42ac4d79
Set the default port for WebRTC (#3175) 
Set the default port for WebRTC to 8000 to prevent the generated default configuration file from disabling WebRTC features.
 2023-12-29 10:53:47 +08:00
夏楚
bbe8f4a018
release 8.0 2023-12-09 16:23:51 +08:00
夏楚
76aebd74c6
整理MediaSource协议类型 2023-12-02 16:31:40 +08:00
LeiZhi.Mai
586bb73912
兼容webrtc offer sdp中重复行的情况 (#3038) 2023-12-01 10:44:59 +08:00
yujitai
92e7d8837e
RTC: 修复一组丢包分多次nack发送时blp-vector没有被重置的问题 (#3016) 
如果一组丢包，通过多个nack发送，那么多次执行blp_vec.resize(16,
false)并不会重置元素值，这会导致blp_vec的丢包状态错乱，从而产生一些不必要的重传请求。
 2023-11-23 21:35:23 +08:00
xiongguangjie
22f5f56130
rtc fix sdp ufag should not contain _ + = - by spec (#2993) 
rtc 中的sdp ufrag 只能有数字与字母组成，不能包含其他符号（根据RFC 5245），包含 _ + = - , 在chromium
项目中有人提出过，详情见( https://crbug.com/1053756 )。chrome 也能正常处理_ + = -
等符号，但是在日志中会有警告如下 `[WARNING:transport_description.cc(32)] '-', '=', '#'
and '-' are not valid ice-char and thus not permitted in ufrag or pwd.
This is a protocol violation that is permitted to allow upgrading but
will be rejected in the future. See https://crbug.com/1053756`
因此修改
 2023-11-19 14:36:03 +08:00
夏楚
ae662fa083
webrtc dtls默认采用https证书，如果https证书不存在则随机生成 (#2928) 
之前默认随机创建dtls证书，导致每次启动证书都不一致，而Firefox要求同主机的dtls证书必须一致，所以导致每次服务重启，Firefox可能拒绝dtls握手。
并且在集群模式下，如果Firefox接入多个不同集群实例的webrtc服务，也可能导致webrtc dtls握手失败。
 2023-10-25 17:50:29 +08:00
cqm
9bf48567b2 fixed issue #2874, thanks quxiaohui-debug 2023-10-16 09:51:20 +08:00
xia-chu
e3e7495c90 新增webrtc播放器datachannel消息广播机制和接口 2023-09-02 12:53:56 +08:00
xia-chu
39dd886ec1 优化获取播放器列表相关功能 2023-09-02 12:07:42 +08:00
xia-chu
d286ac1d73 NoticeCenter支持安全类型转换，解决监听者参数类型不一致导致的bug 2023-09-02 11:22:28 +08:00
Dw9
81966f2c48
修复未开启sctp时，webrtc datachannel sdp交换相关bug (#2791) 
主要优化如下：
1、通过设置port为0 声明不支持datachannel(而不是inactive)。
2、设置正确的ice用户名和密码，防止ice建联失败。
3、group attribute移除datachannel的mid。
3、datachannel 相关sdp移除rtcp相关描述。
 2023-08-29 11:22:30 +08:00
cqm
f4b2fd9c05 重构 audio_transcode 代码： 
- 独立出 RtcMediaSource，并只对rtc开放
- 增加Rtc g711转码开关
- 修改说明
 2023-08-14 13:04:18 +08:00
cqm
607d8ba0e7 实现音频转码功能: 
- 启用rtc:// schecma
- 增加音频转码码率配置
- aac转码使用原通道数
 2023-08-14 12:57:47 +08:00
xia-chu
a7d95461ee 过滤掉不支持的webrtc rtp ext，提高webrtc转发兼容性 
在测试obs simulcast推流时，发现chrome无法正常播放，
分析rtp ext扩展，发现是rtp mid ext未过滤导致，
zlmediakit在回复answer sdp时，已申明不支持mid扩展，
但是obs并未理会还是发送mid扩展。
根据answer sdp过滤rtp ext可兼容此问题。
 2023-08-12 15:25:16 +08:00
夏楚
73c8a5faf4
统一设置单字节对齐方式 (#2716) 
__attribute__((packed)) 替换为 #pragma pack(push, 1)
 2023-07-29 13:07:22 +08:00
夏楚
5464313b41
修复webrtc推流互斥锁重入死锁bug (#2713) 
simulcast推流时，在onRecvRtp函数中可能触发对totalReaderCount的调用，从而导致死锁。
 2023-07-29 13:04:24 +08:00
夏楚
09aa38334e
H264 rtsp sdp生成正确的packetization-mode (#2691) 2023-07-22 17:30:39 +08:00
Dw9
eb382c3fc9
修复whip delete_webrtc未注销流 (#2688) 
bug触发条件：
1、obs whip 推流到zlm
2、关闭推流
3、zlm 未注销流
 2023-07-22 10:46:17 +08:00
yogo-zhangyingzhe
9e8568e753
添加rtp ext: av1-rtp-spec (#2609) 
Co-authored-by: xia-chu <771730766@qq.com>
 2023-07-08 23:28:34 +08:00
夏楚
fad8dd74e7
修复webrtc开启simulcast推流时，统计观看人数线程安全相关bug (#2640) 2023-07-08 21:33:07 +08:00
johzzy
03770ff409
Replace the tuple (vhost/app/stream) with MediaTuple. (#2560) 2023-06-17 10:28:01 +08:00
xia-chu
af0ebf4633 优化Parser代码命名 2023-06-10 14:55:40 +08:00
Johnny
0232caf068 add MediaTuple 2023-05-30 11:01:27 +08:00
Johnny
f4ee607feb add RtspMediaSource::Clone 2023-05-30 11:01:27 +08:00
a-ucontrol
51e9313275
Make available the media_info variable for the class inherited from WebRtcSession and HttpSession. (#2505 #2461) 2023-05-29 14:48:13 +08:00
Johnny
bfec1b1e0e fix typo 2023-05-02 20:41:13 +08:00