e8124e2f56
support publishing H265 and AV1 tracks with Enhanced RTMP ( #1393 ) ( #1446 ) ( #1621 ) ( #1756 )
2023-05-04 20:37:25 +02:00
22fe65509b
cleanup ( #1754 )
2023-05-02 13:05:19 +02:00
2d17dff3b5
support publishing, reading and proxying MPEG-2 audio (MP3) tracks with RTMP ( #1102 ) ( #1736 )
2023-04-25 18:13:51 +02:00
053f2ec282
rename repository and executable ( #1641 )
2023-04-01 19:52:06 +02:00
2dffccf9c1
update gortsplib, gohlslib ( #1637 )
2023-04-01 18:39:12 +02:00
c7938eb832
rtmp: fix panic when publishing audio-only streams ( #1459 ) ( #1502 )
2023-02-22 19:36:04 +01:00
ef214b7649
rtmp server: fix compatibility with Neko ( #1405 )
2023-01-22 13:36:36 +01:00
97c1e68c0b
improve tests
2023-01-19 19:30:19 +01:00
e3d00878b3
rtmp server: fix handshake and compatibility with streamlabs ( #1244 ) ( #1398 )
2023-01-19 16:03:48 +01:00
c79c3c83cb
rtmp server: improve efficiency of outgoing packets ( #1395 )
...
group together messages by using a buffered writer between the network
connection and the WriteMessage() function
2023-01-19 13:27:54 +01:00
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
fbf8e82db5
update gortsplib
2022-12-28 20:32:03 +01:00
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
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
282d155a4f
update gortsplib
2022-11-15 23:47:12 +01:00
6471800b52
rtmp server: fix responses to ping requests ( #1245 )
2022-11-14 11:39:33 +01:00
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
f1fb00b80f
update golangci-lint
2022-09-17 21:19:45 +02:00
27ae0b9812
rtmp client: validate command ID of results
2022-08-22 11:20:23 +02:00
59391a4366
rtmp client: fix play command id
2022-08-22 10:57:29 +02:00
d4945ab7bc
rtmp: cleanup
2022-08-22 10:55:06 +02:00
b06498d24b
rtmp: remove useless comments from tests
2022-08-22 09:49:35 +02:00
ca46d56184
fix linting
2022-08-16 18:44:31 +02:00
e255d004e3
rtmp server: change value of MessageStreamID of outgoing messages
2022-08-16 18:44:31 +02:00
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
a19a20abfb
rtmp: set right command ID when replying to a play request
2022-08-16 18:44:31 +02:00
176f2f0729
rtmp: invert flag of InitializeServer() and InitializeClient()
2022-08-16 18:44:31 +02:00
0db2d3eb8c
rtmp: improve performance
...
reuse existing structs instead of allocating them during every read()
2022-08-15 16:11:23 +02:00
af7a815f83
update gortsplib
2022-08-05 23:50:45 +02:00
f7c08f577a
rtmp: fix decoding of chunk3 + chunk3
2022-07-19 16:42:11 +02:00
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
50d205274f
fix tests
2022-07-17 12:02:17 +02:00
822a896a82
rtmp: fix rtmp -> rtsp audio conversion
2022-07-17 09:54:16 +02:00
ef3e18a9e9
rtmp: add handshake functions
2022-07-16 15:18:04 +02:00
772c5b2363
rtmp: improve tests
2022-07-16 12:42:48 +02:00
2a0567ab4a
rtmp: improve tests
2022-07-14 21:49:18 +02:00
67e8a01d56
rtmp: split net.Conn from rtmp.Conn
2022-07-09 17:25:33 +02:00
bf1f45df32
rtmp: add conn handshake tests
2022-07-09 16:19:49 +02:00
a19f45f736
rtmp: fix s1 generation
2022-07-09 15:00:47 +02:00
8e14dff6d0
rtmp: fix tests
2022-07-09 14:04:51 +02:00
e771f9eba0
rename file
2022-07-02 22:11:26 +02:00
45c1370145
rtmp: normalize API
2022-06-27 17:52:20 +02:00
41b08c9f50
update gortsplib
2022-06-24 17:00:28 +02:00
ec4c40b222
update gortsplib
2022-06-23 13:54:48 +02:00
06b09efa91
rtmp: add tests
2022-06-11 00:31:43 +02:00
58cbd072c8
rtmp: support ping requests
2022-06-11 00:31:05 +02:00
05bac43177
rtmp: fix compatibility with some dji drones ( #928 )
2022-06-11 00:19:06 +02:00
2601ca5661
rtmp: implement acknowledge mechanism
2022-06-08 20:47:36 +02:00
ee2908081e
rtmp: make chunk writes atomic
2022-06-08 14:07:30 +02:00
d0781e5e4a
rtmp: add DTS field to video / audio messages
2022-06-07 23:13:31 +02:00
Alessandro Ros
aler9