24 Commits (704538355c24be072226d0c13b6f12d5dc70d3e9)

Author	SHA1	Message	Date
ZSC714725	3361edd01e	lal support opus	10 months ago
Jae-Sung Lee	6915ab3ddd	if audioPt knows returns true in isAnalyzeEnough	2 years ago
Jae-Sung Lee	1b80c869ae	parse audiocodecid from rtmp metadata	2 years ago
ZSC714725	f01c7b4010	rtmp推流支持enhanced RTMP	2 years ago
q191201771	2e3d90fab2	log	2 years ago
ZSC714725	411ba82181	rtmp2rtsp support g711a	2 years ago
q191201771	3029db08fd	patch	2 years ago
Jae-Sung Lee	373ef34c54	support sampling rate other than 8000Hz in G.711	2 years ago
Jae-Sung Lee	adc930fb0e	rtmp2rtsp support G711U	2 years ago
q191201771	b9bd066b84	new version gofmt to comment	2 years ago
q191201771	29e827a011	[patch] 修复前次提交导致转rtsp格式错误的bug #205	3 years ago
q191201771	e0cfceb4f0	[patch] 新增remux.RtspRemuxerAddSpsPps2KeyFrameFlag配置变量 #205	3 years ago
q191201771	8482b9e0c6	[refactor] base.AvPacket的时间戳类型修改为int64, avpacket_stream.go移入package base	3 years ago
q191201771	eb7bb115d8	[feat] lalserver支持集成自定义协议的输入流	3 years ago
q191201771	9969144c89	patch	3 years ago
q191201771	4e0db49b18	[refactor] 代码整理：检查rtmp msg长度有效性	3 years ago
joestarzxh	7d1fb50a75	[fix]rtmp2rtsp中数据包太小的过滤	3 years ago
q191201771	b4c85d98ec	[fix] rtsp: 增强容错性，修复rtmp输入流没有seq header时，rtmp转rtsp内崩溃的问题	3 years ago
q191201771	11c412cf84	- [refactor] 将rtmp转ts的代码从hls重构至remux中 - [fix] mpegts: 修复单音频场景，有一帧音频重复的问题	3 years ago
q191201771	abf50b107d	- [refactor] 规范所有导出符号的注释 - [refactor] base.ReadableNowTime()	3 years ago
q191201771	0f21334964	[opt] 支持pkg层面控制日志 #62	3 years ago
q191201771	20d5d56690	[fix] 增强兼容性，当收到的rtmp message中aac seq header payload长度为0时忽略，避免崩溃 #116	3 years ago
q191201771	d11074958d	(#97) [feat] lalserver支持用rtsp协议拉取rtmp的推流	4 years ago
q191201771	cfabc9043c	[feat] 新增demo pullrtmp2pushrtsp	4 years ago