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

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Jun 19, 2024
Merged

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

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4 changes: 2 additions & 2 deletions README.md
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Original file line number Diff line number Diff line change
Expand Up @@ -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|
Expand Down
41 changes: 31 additions & 10 deletions internal/protocols/webrtc/incoming_track.go
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Expand Up @@ -5,6 +5,7 @@
"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"
Expand Down Expand Up @@ -79,21 +80,28 @@
},
PayloadType: 102,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH265,
ClockRate: 90000,
},
PayloadType: 103,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH264,
ClockRate: 90000,
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
},
PayloadType: 103,
PayloadType: 104,
},
{
RTPCodecCapability: webrtc.RTPCodecCapability{
MimeType: webrtc.MimeTypeH264,
ClockRate: 90000,
SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
},
PayloadType: 104,
PayloadType: 105,
},
}

Expand Down Expand Up @@ -229,6 +237,7 @@
track *webrtc.TrackRemote
log logger.Writer

typ description.MediaType
format format.Format
reorderer *rtpreorderer.Reorderer
pkts []*rtp.Packet
Expand All @@ -246,41 +255,47 @@
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 {
Expand All @@ -292,9 +307,12 @@
}

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
Expand All @@ -313,6 +331,8 @@
}

case strings.ToLower(webrtc.MimeTypePCMA):
t.typ = description.MediaTypeAudio

channels := track.Codec().Channels
if channels == 0 {
channels = 1
Expand All @@ -331,6 +351,7 @@
}

case strings.ToLower(mimeTypeL16):
t.typ = description.MediaTypeAudio
t.format = &format.LPCM{
PayloadTyp: uint8(track.PayloadType()),
BitDepth: 16,
Expand All @@ -339,7 +360,7 @@
}

default:
return nil, fmt.Errorf("unsupported codec: %+v", track.Codec())
return nil, fmt.Errorf("unsupported codec: %+v", track.Codec().RTPCodecCapability)

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}

// read incoming RTCP packets to make interceptors work
Expand All @@ -354,7 +375,7 @@
}()

// send period key frame requests
if isVideo {
if t.typ == description.MediaTypeVideo {
go func() {
keyframeTicker := time.NewTicker(keyFrameInterval)
defer keyframeTicker.Stop()
Expand Down
10 changes: 10 additions & 0 deletions internal/protocols/webrtc/outgoing_track.go
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Expand Up @@ -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{
Expand Down Expand Up @@ -205,6 +214,7 @@ func (t *OutgoingTrack) isVideo() bool {
case *format.AV1,
*format.VP9,
*format.VP8,
*format.H265,
*format.H264:
return true
}
Expand Down
11 changes: 11 additions & 0 deletions internal/protocols/webrtc/peer_connection_test.go
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Expand Up @@ -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,
Expand Down
16 changes: 2 additions & 14 deletions internal/protocols/webrtc/tracks_to_medias.go
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Expand Up @@ -10,21 +10,9 @@
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},

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}
}

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