6520 Commits (f3806126c4b11e698813170215a9b5fd5a6ed274)

 

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Author	SHA1	Message	Date
winlin	5ebf034aea	For #1694, Refine API for nb_bytes	4 years ago
winlin	c17474627b	Merge SRS3	4 years ago
winlin	25c76c1e8a	Fix #1694, Support DVR 2GB+ MP4 file. 3.0.155	4 years ago
winlin	576be75f00	Merge branch '4.0release' into develop	4 years ago
winlin	c97e943b07	Merge SRS3	4 years ago
winlin	ab5ddd24e2	Fix #1548, Add edts in MP4 for Windows10. 3.0.154	4 years ago
winlin	721173e6af	MP4: Fix warnings	4 years ago
winlin	1502560bcf	Update conf	4 years ago
winlin	9caeb606bf	ST: Support show coroutines.	4 years ago
winlin	c7c6d8778a	RTC: Fix warnings	4 years ago
jinxue.cgh	5309dbe18b	RTC: Refine RTCP process	4 years ago
winlin	18ae8d8571	RTC: Fix SDP bug for firefox	4 years ago
winlin	aad7c448bf	For #1998, Support Firefox.	4 years ago
winlin	0c113ff084	For #1998, support firefox	4 years ago
winlin	c5457e8241	RTC: Support unified-plan	4 years ago
winlin	27db60cc23	Fix #1996, Heap off-by-one in utest	4 years ago
winlin	c796c0d093	Fix #1689, fix typo	4 years ago
莫战	b38f30c3ee	support query parsing and escape	4 years ago
莫战	dc7124cd05	support base64 encode	4 years ago
jinxue.cgh	58b75c6f1b	tfsfu: add play red pt negotiate	4 years ago
winlin	140f8b0fce	For #1998, refine PT for firefox, support RED	4 years ago
winlin	f47329a94c	Update authors, for #2042, #2057	4 years ago
PieerePi	3d5c18c25a	GB28181 code crashed in ffmpeg after commit "RTC: Use FFmpeg to transcode aac to opus" <d5a0ad3dd8>. (#2057) ... Change the size from 64K to 256K.	4 years ago
ghostsf	d3e153e504	fix: update CMakeLists.txt for rtc (#2042)	4 years ago
winlin	d66082320f	For #2039, update authors	4 years ago
Jesse Xi	8515f5a91e	incomplete_len 在大华摄像头下，因为大华包头对音频的不标准处理，可能为负值，而sizeof(SrsPsPacketStartCode) 返回的是unsigned 类型， 因些增加判断 (#2039) ... Co-authored-by: jesse.xi <jj.xi@tianrang-inc.com>	4 years ago
winlin	c779d95246	GB28181: Remove chinese comments.	4 years ago
winlin	977e027d86	SIP: Fix build fail for Mac	4 years ago
winlin	fa3c491c0b	For #2014, Merged.	4 years ago
Pieere Pi	ffae1720ec	gb28181模块可用性增强 ... 主要改动， 1. 支持作为GB/T 28181上级平台 2. 新的目录接口sip_query_devicelist (/api/v1/gb28181?action=sip_query_devicelist) 3. 各种异常和问题修复 4. 其他一些小改动 以上改动基于feature/rtc分支，因为需要网页用WebRTC来拉GB28181的监控流，gb28181分支代码有点老了。 下面的序号n是指第n个差异块（"@@ -"之间的内容）。 srs_gb28181.html 1. 原页面上多加了一个端口号 2-4. 给摄像头加上名称显示 5. 查询目录去掉chid 6. 删除通道参数分解为id和chid 7. API端口固定为1985 srs_app_gb28181.cpp 1-4. 四处因为错误而退出GB28181媒体处理循环，修改为不退出 5. payload为空异常 6. 修正判断startcode越界一个字符导致内存写越界的问题 ps流有可能末尾是全零填充，而且越界的那个字符正好是0x01，这样会多出一个nalu（末尾的三个0x00和一个越界的0x01），后面写video_data内存越界（if (first_pos != pre_pos){块，此处size - pre_pos - 4为-1，uint32_t naluLen得到的值为0，video_data[pre_pos+3] = p[0];写越界）破坏了其他数据，后续video_stream析构出错程序异常退出。 7. 此处srs后来已修复 8. 更新ssrc为被叫返回的值 原代码只支持标准中的《点播域内设备媒体流SSRC处理方式》（设备注册上来），不支持《点播外域设备媒体流SSRC处理方式》（即作为上级平台）。 这是因为如果srs作为上级平台，ssrc不是自己生成的，而是下级平台生成的。 9. 删除通道参数分解为id和chid 10. notify_sip_unregister后delete_stream_channel无效 11. notify_sip_query_catalog清空内存中的设备列表 12. 新函数query_device_list srs_app_gb28181.hpp 1. update_rtmpmuxer_to_newssrc_by_id声明 2. 新函数get_gb28181_config_ptr和函数delete_stream_channel声明修改 3. 新函数query_device_list srs_app_gb28181_sip.cpp 1-4. 在调试界面给摄像头加上名称显示；新函数clear_device_list和新函数dumpItemList 5-6. 两处因为错误而退出GB28181信令处理循环，修改为不退出 7. 设备注册上来，不检查服务器ID匹不匹配（支持作为上级平台） 8. 收到一个目录上报消息，更新内存中的数据 9. 更新ssrc为被叫返回的值 10. 新函数query_device_list srs_app_gb28181_sip.hpp 1. 在调试界面给摄像头加上名称显示 2. 每个设备加上item_list，用于存储目录；新函数clear_device_list和新函数dumpItemList 3. 新函数clear_device_list srs_app_http_api.cpp 1. 删除通道参数分解为id和chid 2. 新的接口sip_query_devicelist，用于查询所有设备的目录 srs_sip_stack.cpp 1. GB2312转UTF-8类 2. 被叫返回的ssrc初始化 3. parse_xml声明修改 4. 对XML内容进行字符集检测和转换 5-7. parse_xml定义修改 8. SIP BODY里面也有可能有\r\n 9-10. 防止恶意SIP消息 by vicious sip prober 11-12. 新的XML解析目录代码 13. 获取被叫返回的ssrc srs_sip_stack.hpp 1. 依赖vector 2. 每个设备加上item_list，用于存储目录 3. 被叫返回的ssrc 4. parse_xml声明修改	4 years ago
yinjiaoyuan	fe65c7bf84	For 2034, GB28181: Support transport over TCP	4 years ago
winlin	751dab56d8	RTC: Refine player and publisher	4 years ago
winlin	529264f238	RTC: Refine player and publisher	4 years ago
winlin	3cf3047f97	Add conf/rtc_live.conf	4 years ago
winlin	7521bc86ad	For #1998, Update conf	4 years ago
winlin	7136af21de	For #1998, TODO: FIME: Should check packetization-mode=1 also.	4 years ago
winlin	5d27c62e95	For #1998, fix fetch remote payload bug. 4.0.56	4 years ago
winlin	4650d47082	For #1998, Support Firefox, use PT in offer. 4.0.55	4 years ago
winlin	57b5204a10	For #1998, Set default fmtp for H264 when transmux RTMP to RTC	4 years ago
winlin	9908433bc8	For #1508, Transform http header name to upper camel case. 4.0.54	4 years ago
winlin	07c04a042a	URI: Refine uri parser	4 years ago
winlin	efca38cd89	Player: Change default HTTP-API port to 1985 for WebRTC	4 years ago
winlin	cb4c668249	Merge SRS3	4 years ago
winlin	6e922b9589	Refine README for SRS3	4 years ago
winlin	e085250245	Update players	4 years ago
winlin	32c1832d64	For #1657, refine code	4 years ago
winlin	5709ee1b63	For #1657, add https configs	4 years ago
winlin	385e055c7b	For #1657, Fix read bug. 4.0.53	4 years ago
winlin	4618bfc137	For #1657, fix the http read bug	4 years ago
winlin	6dc9824495	For #1657, fix the http read bug	4 years ago