3606472e82
generate RTP packets after H264 remuxing
...
Previously, RTP packets coming from sources other than RTSP (that
actually are RTMP and HLS) were generated before the H264 remuxing, and
that leaded to invalid streams, expecially when sourceOnDemand is true
and the stream has invalid or dynamic SPS/PPS.
3 years ago
5ba46e6c8f
update gortsplib
3 years ago
a8822b9f15
hls muxer / source: route AAC units singularly
...
This aligns the HLS implementation with the rest of the server. In case
of HLS/MPEGTS, the server now generates an ADTS packet for each AU,
without grouping multiple AUs into a single ADTS packet.
3 years ago
3f0771bb31
fix variable name
3 years ago
af7a815f83
update gortsplib
3 years ago
092a2be3a8
api: add "created" field to RTSP sessions, RTMP connections, HLS muxers
3 years ago
055e08ac6c
remove "on" prefix from most communication functions between components
3 years ago
ea528f47e1
speed up tests
3 years ago
7067c02030
api: return static sources in paths/list even if sourceOnDemand is true
3 years ago
6afbdf1015
add intermediate layer between path and static sources
3 years ago
34eb1d1e7a
normalize sources
3 years ago
8a4743fe9a
hls muxer: when hlsAlwaysRemux is on, automatically recreate muxers in
...
case of errors
3 years ago
9e6abc6e9f
rtmp: rewrite implementation of rtmp connection ( #1047 )
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* rtmp: improve MsgCommandAMF0
* rtmp: fix MsgSetPeerBandwidth
* rtmp: add message tests
* rtmp: replace implementation with new one
* rtmp: rename handshake functions
* rtmp: avoid calling useless function
* rtmp: use time.Duration for PTSDelta
* rtmp: fix decoding chunks with relevant size
* rtmp: rewrite implementation of rtmp connection
* rtmp: fix tests
* rtmp: improve error message
* rtmp: replace h264 config implementation
* link against github.com/notedit/rtmp
* normalize MessageStreamID
* rtmp: make acknowledge optional
* rtmp: fix decoding of chunk2 + chunk3
* avoid using encoding/binary
3 years ago
67e8a01d56
rtmp: split net.Conn from rtmp.Conn
3 years ago
bf1f45df32
rtmp: add conn handshake tests
3 years ago
f28540b9f4
update gortsplib
3 years ago
08ddc9f83d
update gortsplib
3 years ago
fb5aa7bbf2
fix bug that caused on-demand commands to be closed after 10secs ( #1000 )
3 years ago
19e38930bd
hls muxer: compensate latency by using maxLiveSyncPlaybackRate ( #972 )
3 years ago
41b08c9f50
update gortsplib
3 years ago
ec4c40b222
update gortsplib
3 years ago
e45820b2c0
hls server: show real client IPs when behind a proxy ( #955 )
3 years ago
3e5f62156d
fix DTS error in case of H264 NALUs without POC
3 years ago
05bac43177
rtmp: fix compatibility with some dji drones ( #928 )
3 years ago
cb610a707c
hls muxer: fix code 500 when a stream is not found
3 years ago
9d3fd3bc37
update gortsplib
3 years ago
9bd8b2cfb6
rtmp server, hls muxer: fix DTS in case of B-frames and remove PTS-DTS offset
3 years ago
4073013f68
hls muxer: stop normalizing PTS
3 years ago
acd788d632
update gortsplib
3 years ago
2ed1aa3d11
hls muxer, rtmp server: extract DTS from samples
3 years ago
e115983296
Implement Low-Latency HLS ( #938 )
...
* add hlsVariant parameter
* hls: split muxer into variants
* hls: implement fmp4 segments
* hls muxer: implement low latency mode
* hls muxer: support audio with fmp4 mode
* hls muxer: rewrite file router
* hls muxer: implement preload hint
* hls muxer: add various error codes
* hls muxer: use explicit flags
* hls muxer: fix error in aac pts
* hls muxer: fix sudden freezes with video+audio
* hls muxer: skip empty parts
* hls muxer: fix video FPS
* hls muxer: add parameter hlsPartDuration
* hls muxer: refactor fmp4 muxer
* hls muxer: fix CAN-SKIP-UNTIL
* hls muxer: refactor code
* hls muxer: show only parts of last 2 segments
* hls muxer: implementa playlist delta updates
* hls muxer: change playlist content type
* hls muxer: improve video dts precision
* hls muxer: fix video sample flags
* hls muxer: improve iphone audio support
* hls muxer: improve mp4 timestamp precision
* hls muxer: add offset between pts and dts
* hls muxer: close muxer in case of error
* hls muxer: stop logging requests with the info level
* hls muxer: rename entry into sample
* hls muxer: compensate video dts error over time
* hls muxer: change default segment count
* hls muxer: add starting gap
* hls muxer: set default part duration to 200ms
* hls muxer: fix audio-only streams on ios
* hls muxer: add playsinline attribute to video tag of default web page
* hls muxer: keep mpegts as the default hls variant
* hls muxer: implement encryption
* hls muxer: rewrite dts estimation
* hls muxer: improve DTS precision
* hls muxer: use right SPS/PPS for each sample
* hls muxer: adjust part duration dynamically
* add comments
* update readme
* hls muxer: fix memory leak
* hls muxer: decrease ram consumption
3 years ago
c5afa69174
fix regression that caused a source to be closed after 10secs when sourceOnDemand is yes ( #949 )
3 years ago
23ac079646
rtsp source: fix regression ( #949 )
...
This caused a periodic disconnection when sourceOnDemand is yes
3 years ago
13fb24da39
fix rtmp -> rtsp audio conversion ( #932 )
3 years ago
d6a804f592
rtsp server: improve performance when reading with TCP
3 years ago
0c4f6e2d43
rtmp server: fix bias error in AAC DTS
3 years ago
b6b99b142a
hls muxer: prefer hls.js over native HLS
3 years ago
901eae2f6b
fix bias error in AAC timestamp
3 years ago
58e3fa358e
split handling of on-demand sources and on-demand publishers
3 years ago
98b3538289
fix panic that happens when publishing to a path with source = redirect ( #933 )
3 years ago
35b3541e4f
hls client: add limit on AU size
3 years ago
709d727eab
hls muxer: tune hls.js parameters
3 years ago
dedca93eca
hls muxer: update hls.js
3 years ago
ae7e68c914
hls muxer: remove progressive flag; add liveSyncDurationCount, liveMaxLatencyDurationCount to hls.js
3 years ago
1e07636f86
change default RTSPS port ( #867 )
3 years ago
6b86607092
rtsp source: improve support for AAC tracks with custom parameters
3 years ago
f71b7d8967
fix tests
3 years ago
ce42c53a03
hls, rtmp: fix video/audio sync
3 years ago
f620484757
rtmp: always send decoder config before IDR frames
3 years ago
98c6cd4650
RTSP: automatically remux oversized RTP/H264 packets; drop parameter ReadBufferSize
3 years ago
aler9
Alessandro Ros