Alessandro Ros
39c072edd6
change repository owner ( #1801 )
2023-05-16 16:14:20 +02:00
Alessandro Ros
e8124e2f56
support publishing H265 and AV1 tracks with Enhanced RTMP ( #1393 ) ( #1446 ) ( #1621 ) ( #1756 )
2023-05-04 20:37:25 +02:00
Alessandro Ros
22fe65509b
cleanup ( #1754 )
2023-05-02 13:05:19 +02:00
Alessandro Ros
2d17dff3b5
support publishing, reading and proxying MPEG-2 audio (MP3) tracks with RTMP ( #1102 ) ( #1736 )
2023-04-25 18:13:51 +02:00
Alessandro Ros
053f2ec282
rename repository and executable ( #1641 )
2023-04-01 19:52:06 +02:00
Alessandro Ros
2dffccf9c1
update gortsplib, gohlslib ( #1637 )
2023-04-01 18:39:12 +02:00
Alessandro Ros
c7938eb832
rtmp: fix panic when publishing audio-only streams ( #1459 ) ( #1502 )
2023-02-22 19:36:04 +01:00
Alessandro Ros
ef214b7649
rtmp server: fix compatibility with Neko ( #1405 )
2023-01-22 13:36:36 +01:00
aler9
b20abbed6c
webrtc muxer: keep the WebSocket connection
...
The WebSocket connection is kept open in order to use it to notify
shutdowns.
2023-01-08 15:37:47 +01:00
aler9
fbf8e82db5
update gortsplib
2022-12-28 20:32:03 +01:00
Alessandro Ros
ad52b3fab7
Support publishing with RTMP and H265 (for OBS Studio) ( #1333 )
...
* support publishing with RTMP and H265 (for OBS Studio)
* rtmp source: block H265 tracks
2022-12-27 13:55:30 +01:00
Alessandro Ros
c778c049ce
switch to gortsplib v2 ( #1301 )
...
Fixes #1103
gortsplib/v2 supports multiple formats inside a single track (media). This allows to apply the resizing algorithm to single formats inside medias.
For instance, if a media contains a a proprietary format and an H264 format, and the latter has oversized packets, they can now be resized.
2022-12-13 20:54:17 +01:00
aler9
282d155a4f
update gortsplib
2022-11-15 23:47:12 +01:00
Alessandro Ros
8bee4af86a
api, metrics: add number of bytes received and sent from/to all entities ( #1235 )
...
* API: number of bytes received/sent from/to RTSP connections
* API: number of bytes received/sent from/to RTSP sessions
* API: number of bytes received/sent from/to RTMP connections
* API: number of bytes sent to HLS connections
* API: number of bytes received from paths
* metrics of all the above
2022-11-11 11:59:52 +01:00
aler9
f1fb00b80f
update golangci-lint
2022-09-17 21:19:45 +02:00
aler9
27ae0b9812
rtmp client: validate command ID of results
2022-08-22 11:20:23 +02:00
aler9
59391a4366
rtmp client: fix play command id
2022-08-22 10:57:29 +02:00
aler9
d4945ab7bc
rtmp: cleanup
2022-08-22 10:55:06 +02:00
aler9
e255d004e3
rtmp server: change value of MessageStreamID of outgoing messages
2022-08-16 18:44:31 +02:00
aler9
4990e98993
rtmp: fix reading metadata from onMetadata
...
when there's no audio and Conn is a client, onMetadata was skipped and
tracks were read by using the fallback method. Fix this.
2022-08-16 18:44:31 +02:00
aler9
a19a20abfb
rtmp: set right command ID when replying to a play request
2022-08-16 18:44:31 +02:00
aler9
176f2f0729
rtmp: invert flag of InitializeServer() and InitializeClient()
2022-08-16 18:44:31 +02:00
aler9
0db2d3eb8c
rtmp: improve performance
...
reuse existing structs instead of allocating them during every read()
2022-08-15 16:11:23 +02:00
aler9
af7a815f83
update gortsplib
2022-08-05 23:50:45 +02:00
Alessandro Ros
9e6abc6e9f
rtmp: rewrite implementation of rtmp connection ( #1047 )
...
* 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
2022-07-17 15:17:18 +02:00
aler9
822a896a82
rtmp: fix rtmp -> rtsp audio conversion
2022-07-17 09:54:16 +02:00
aler9
67e8a01d56
rtmp: split net.Conn from rtmp.Conn
2022-07-09 17:25:33 +02:00
aler9
bf1f45df32
rtmp: add conn handshake tests
2022-07-09 16:19:49 +02:00
aler9
41b08c9f50
update gortsplib
2022-06-24 17:00:28 +02:00
aler9
ec4c40b222
update gortsplib
2022-06-23 13:54:48 +02:00
aler9
05bac43177
rtmp: fix compatibility with some dji drones ( #928 )
2022-06-11 00:19:06 +02:00
aler9
d3797d3139
rtmp: improve video / audio messages
2022-06-07 22:48:10 +02:00
aler9
db7ee22789
rtsp source: support AAC tracks with custom sizelength, indexlength and indexdeltalength
...
(https://github.com/aler9/gortsplib/pull/118 )
2022-04-15 13:17:00 +02:00
aler9
a34a01ebd9
RTMP client/source: support dynamic H264 SPS/PPS
2022-04-08 18:19:53 +02:00
aler9
983469a1f9
rtmp: support clients that publish with empty metadata or no metadata ( #386 ) ( #769 )
2022-02-12 17:48:55 +01:00
aler9
2bfdcc7d89
update gortsplib
2022-01-30 17:43:03 +01:00
aler9
811540b34b
tidy up rtmp
2021-12-22 17:37:15 +01:00
aler9
1dff3239d2
remove rtmp.Conn.NetConn()
2021-12-22 17:33:37 +01:00
aler9
99a07c0d33
rtmp client: speed up acceptance of clients by moving handshake inside client routine
2021-04-03 12:08:07 +02:00
aler9
897322e3a6
rename rtmputils into rtmp
2021-04-03 11:39:19 +02:00
