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webrtc: support publishing H265 tracks (#3435) (#3492)

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IMPORTANT NOTE: this doesn't allow to read H265 tracks with WebRTC,
just to publish them. The inability to read H265 tracks with WebRTC is
not in any way related to the server but depends on browsers and on the
fact that they are not legally entitled to embed a H265 decoder inside
them.
This commit is contained in:
Alessandro Ros 2024-06-19 21:02:08 +02:00 • committed by GitHub
parent 65d90f7cc6
commit a1dc9f45f5
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GPG key ID: B5690EEEBB952194
5 changed files with 58 additions and 28 deletions
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4
README.md
View file

@ -22,8 +22,8 @@ Live streams can be published to the server with:
|--------|--------|------------|------------|
|[SRT clients](#srt-clients)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[SRT cameras and servers](#srt-cameras-and-servers)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[WebRTC clients](#webrtc-clients)|Browser-based, WHIP|AV1, VP9, VP8, H264|Opus, G722, G711 (PCMA, PCMU)|
|[WebRTC servers](#webrtc-servers)|WHEP|AV1, VP9, VP8, H264|Opus, G722, G711 (PCMA, PCMU)|
|[WebRTC clients](#webrtc-clients)|Browser-based, WHIP|AV1, VP9, VP8, H265, H264|Opus, G722, G711 (PCMA, PCMU)|
|[WebRTC servers](#webrtc-servers)|WHEP|AV1, VP9, VP8, H265, H264|Opus, G722, G711 (PCMA, PCMU)|
|[RTSP clients](#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), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
|[RTSP cameras and servers](#rtsp-cameras-and-servers)|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), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
|[RTMP clients](#rtmp-clients)|RTMP, RTMPS, Enhanced RTMP|AV1, VP9, H265, H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G711 (PCMA, PCMU), LPCM|

43
internal/protocols/webrtc/incoming_track.go
View file

@ -5,6 +5,7 @@ import (
"strings"
"time"
"github.com/bluenviron/gortsplib/v4/pkg/description"
"github.com/bluenviron/gortsplib/v4/pkg/format"
"github.com/bluenviron/gortsplib/v4/pkg/liberrors"
"github.com/bluenviron/gortsplib/v4/pkg/rtpreorderer"
@ -81,9 +82,8 @@ var incomingVideoCodecs = []webrtc.RTPCodecParameters{
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH264,
MimeType: webrtc.MimeTypeH265,
ClockRate: 90000,
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
},
PayloadType: 103,
},
@ -91,10 +91,18 @@ var incomingVideoCodecs = []webrtc.RTPCodecParameters{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH264,
ClockRate: 90000,
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
},
PayloadType: 104,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH264,
ClockRate: 90000,
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
},
PayloadType: 105,
},
}
var incomingAudioCodecs = []webrtc.RTPCodecParameters{
@ -229,6 +237,7 @@ type IncomingTrack struct {
track *webrtc.TrackRemote
log logger.Writer
typ description.MediaType
format format.Format
reorderer *rtpreorderer.Reorderer
pkts []*rtp.Packet
@ -246,41 +255,47 @@ func newIncomingTrack(
reorderer: rtpreorderer.New(),
}
isVideo := false
switch strings.ToLower(track.Codec().MimeType) {
case strings.ToLower(webrtc.MimeTypeAV1):
isVideo = true
t.typ = description.MediaTypeVideo
t.format = &format.AV1{
PayloadTyp: uint8(track.PayloadType()),
}
case strings.ToLower(webrtc.MimeTypeVP9):
isVideo = true
t.typ = description.MediaTypeVideo
t.format = &format.VP9{
PayloadTyp: uint8(track.PayloadType()),
}
case strings.ToLower(webrtc.MimeTypeVP8):
isVideo = true
t.typ = description.MediaTypeVideo
t.format = &format.VP8{
PayloadTyp: uint8(track.PayloadType()),
}
case strings.ToLower(webrtc.MimeTypeH265):
t.typ = description.MediaTypeVideo
t.format = &format.H265{
PayloadTyp: uint8(track.PayloadType()),
}
case strings.ToLower(webrtc.MimeTypeH264):
isVideo = true
t.typ = description.MediaTypeVideo
t.format = &format.H264{
PayloadTyp: uint8(track.PayloadType()),
PacketizationMode: 1,
}
case strings.ToLower(mimeTypeMultiopus):
t.typ = description.MediaTypeAudio
t.format = &format.Opus{
PayloadTyp: uint8(track.PayloadType()),
ChannelCount: int(track.Codec().Channels),
}
case strings.ToLower(webrtc.MimeTypeOpus):
t.typ = description.MediaTypeAudio
t.format = &format.Opus{
PayloadTyp: uint8(track.PayloadType()),
ChannelCount: func() int {
@ -292,9 +307,12 @@ func newIncomingTrack(
}
case strings.ToLower(webrtc.MimeTypeG722):
t.typ = description.MediaTypeAudio
t.format = &format.G722{}
case strings.ToLower(webrtc.MimeTypePCMU):
t.typ = description.MediaTypeAudio
channels := track.Codec().Channels
if channels == 0 {
channels = 1
@ -313,6 +331,8 @@ func newIncomingTrack(
}
case strings.ToLower(webrtc.MimeTypePCMA):
t.typ = description.MediaTypeAudio
channels := track.Codec().Channels
if channels == 0 {
channels = 1
@ -331,6 +351,7 @@ func newIncomingTrack(
}
case strings.ToLower(mimeTypeL16):
t.typ = description.MediaTypeAudio
t.format = &format.LPCM{
PayloadTyp: uint8(track.PayloadType()),
BitDepth: 16,
@ -339,7 +360,7 @@ func newIncomingTrack(
}
default:
return nil, fmt.Errorf("unsupported codec: %+v", track.Codec())
return nil, fmt.Errorf("unsupported codec: %+v", track.Codec().RTPCodecCapability)
}
// read incoming RTCP packets to make interceptors work
@ -354,7 +375,7 @@ func newIncomingTrack(
}()
// send period key frame requests
if isVideo {
if t.typ == description.MediaTypeVideo {
go func() {
keyframeTicker := time.NewTicker(keyFrameInterval)
defer keyframeTicker.Stop()

10
internal/protocols/webrtc/outgoing_track.go
View file

@ -54,6 +54,15 @@ func (t *OutgoingTrack) codecParameters() (webrtc.RTPCodecParameters, error) {
PayloadType: 96,
}, nil
case *format.H265:
return webrtc.RTPCodecParameters{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH265,
ClockRate: 90000,
},
PayloadType: 96,
}, nil
case *format.H264:
return webrtc.RTPCodecParameters{
RTPCodecCapability: webrtc.RTPCodecCapability{
@ -205,6 +214,7 @@ func (t *OutgoingTrack) isVideo() bool {
case *format.AV1,
*format.VP9,
*format.VP8,
*format.H265,
*format.H264:
return true
}

11
internal/protocols/webrtc/peer_connection_test.go
View file

@ -83,6 +83,17 @@ func TestPeerConnectionPublishRead(t *testing.T) {
PayloadTyp: 96,
},
},
{
"h265",
test.FormatH265,
webrtc.RTPCodecCapability{
MimeType: "video/H265",
ClockRate: 90000,
},
&format.H265{
PayloadTyp: 96,
},
},
{
"h264",
test.FormatH264,

16
internal/protocols/webrtc/tracks_to_medias.go
View file

@ -10,21 +10,9 @@ func TracksToMedias(tracks []*IncomingTrack) []*description.Media {
ret := make([]*description.Media, len(tracks))
for i, track := range tracks {
forma := track.Format()
var mediaType description.MediaType
switch forma.(type) {
case *format.AV1, *format.VP9, *format.VP8, *format.H264:
mediaType = description.MediaTypeVideo
default:
mediaType = description.MediaTypeAudio
}
ret[i] = &description.Media{
Type: mediaType,
Formats: []format.Format{forma},
Type: track.typ,
Formats: []format.Format{track.format},
}
}