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update documentation (#1885)

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This commit is contained in:
Alessandro Ros
2023-05-31 18:16:39 +02:00
committed by GitHub
parent 0cccfa1928
commit 210603d7c4
4 changed files with 10 additions and 10 deletions
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README.md
+6 -6
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@@ -11,10 +11,10 @@ Live streams can be published to the server with:
|protocol|variants|video codecs|audio codecs|
|--------|--------|------------|------------|
|WebRTC|Browser-based, WHIP|AV1, VP9, VP8, H264|Opus, G722, G711|
|RTSP clients|UDP, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
|RTSP servers and cameras|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
|RTMP clients (OBS Studio)|RTMP, RTMPS, Enhanced RTMP|AV1, H265, H264|MPEG-4 Audio (AAC), MPEG-2 Audio (MP3)|
|RTMP servers and cameras|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-2 Audio (MP3)|
|RTSP clients|UDP, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
|RTSP servers and cameras|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
|RTMP clients (OBS Studio)|RTMP, RTMPS, Enhanced RTMP|AV1, H265, H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
|RTMP servers and cameras|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
|HLS servers and cameras|Low-Latency HLS, MP4-based HLS, legacy HLS|H265, H264|Opus, MPEG-4 Audio (AAC)|
|UDP/MPEG-TS streams|Unicast, broadcast, multicast|H265, H264|Opus, MPEG-4 Audio (AAC)|
|Raspberry Pi Cameras||H264||
@@ -24,8 +24,8 @@ And can be read from the server with:
|protocol|variants|video codecs|audio codecs|
|--------|--------|------------|------------|
|WebRTC|Browser-based, WHEP|AV1, VP9, VP8, H264|Opus, G722, G711|
|RTSP|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
|RTMP|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-2 Audio (MP3)|
|RTSP|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G722, G711, LPCM and any RTP-compatible codec|
|RTMP|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
|HLS|Low-Latency HLS, MP4-based HLS, legacy HLS|H265, H264|Opus, MPEG-4 Audio (AAC)|
Features:
internal/core/rtmp_conn.go
+2 -2
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@@ -407,7 +407,7 @@ func (c *rtmpConn) runRead(ctx context.Context, u *url.URL) error {
if videoFormat == nil && audioFormat == nil {
return fmt.Errorf(
"the stream doesn't contain any supported codec, which are currently H264, MPEG-2 Audio, MPEG-4 Audio")
"the stream doesn't contain any supported codec, which are currently H264, MPEG-1/2 Audio, MPEG-4 Audio")
}
defer res.stream.readerRemove(c)
@@ -655,7 +655,7 @@ func (c *rtmpConn) findAudioFormat(stream *stream, ringBuffer *ringbuffer.RingBu
}
if !(!h.MPEG2 && h.Layer == 3) {
return fmt.Errorf("RTMP only supports MPEG-1 audio layer 3")
return fmt.Errorf("RTMP only supports MPEG-1 layer 3 audio")
}
channels := uint8(flvio.SOUND_STEREO)
internal/formatprocessor/mpeg2audio.go
+1 -1
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@@ -11,7 +11,7 @@ import (
"github.com/bluenviron/mediamtx/internal/logger"
)
// UnitMPEG2Audio is a MPEG-2 Audio data unit.
// UnitMPEG2Audio is a MPEG-1/2 Audio data unit.
type UnitMPEG2Audio struct {
RTPPackets []*rtp.Packet
NTP time.Time
internal/formatprocessor/processor.go
+1 -1
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@@ -19,7 +19,7 @@ type Processor interface {
// cleans and normalizes a data unit.
Process(Unit, bool) error
// returns an unit for the given RTP packet.
// wraps a RTP packet into a Unit.
UnitForRTPPacket(pkt *rtp.Packet, ntp time.Time) Unit
}