Adds support WebRTC + H265 to Safari

This commit is contained in:
Alexey Khit
2022-09-21 22:28:59 +03:00
parent 62c13f016b
commit 04a6e64650
4 changed files with 79 additions and 6 deletions
+1
View File
@@ -36,3 +36,4 @@ H.264/high | avc1.6400xx | FFmpeg superfast
- [AVC levels](https://en.wikipedia.org/wiki/Advanced_Video_Coding#Levels)
- [AVC profiles table](https://developer.mozilla.org/ru/docs/Web/Media/Formats/codecs_parameter)
- [Supported Media for Google Cast](https://developers.google.com/cast/docs/media)
- [Two stream formats, Annex-B, AVCC (H.264) and HVCC (H.265)](https://www.programmersought.com/article/3901815022/)
+66 -4
View File
@@ -21,10 +21,6 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
//)
switch naluType {
case h265parser.NAL_UNIT_CODED_SLICE_TRAIL_R:
case h265parser.NAL_UNIT_VPS:
case h265parser.NAL_UNIT_SPS:
case h265parser.NAL_UNIT_PPS:
case h265parser.NAL_UNIT_UNSPECIFIED_49:
data := packet.Payload
switch data[2] >> 6 {
@@ -55,3 +51,69 @@ func RTPDepay(track *streamer.Track) streamer.WrapperFunc {
}
}
}
// SafariPay - generate Safari friendly payload for H265
func SafariPay(mtu uint16) streamer.WrapperFunc {
sequencer := rtp.NewRandomSequencer()
size := int(mtu - 12) // rtp.Header size
var buffer []byte
return func(push streamer.WriterFunc) streamer.WriterFunc {
return func(packet *rtp.Packet) error {
if packet.Version != h264.RTPPacketVersionAVC {
return push(packet)
}
data := packet.Payload
data[0] = 0
data[1] = 0
data[2] = 0
data[3] = 1
var start byte
nut := (data[4] >> 1) & 0b111111
switch nut {
case h265parser.NAL_UNIT_VPS, h265parser.NAL_UNIT_SPS, h265parser.NAL_UNIT_PPS:
buffer = append(buffer, data...)
return nil
case h265parser.NAL_UNIT_CODED_SLICE_IDR_W_RADL:
buffer = append([]byte{3}, buffer...)
data = append(buffer, data...)
start = 1
default:
data = append([]byte{2}, data...)
start = 0
}
for len(data) > size {
clone := rtp.Packet{
Header: rtp.Header{
Version: 2,
Marker: false,
SequenceNumber: sequencer.NextSequenceNumber(),
Timestamp: packet.Timestamp,
},
Payload: data[:size],
}
if err := push(&clone); err != nil {
return err
}
data = append([]byte{start}, data[size:]...)
}
clone := rtp.Packet{
Header: rtp.Header{
Version: 2,
Marker: true,
SequenceNumber: sequencer.NextSequenceNumber(),
Timestamp: packet.Timestamp,
},
Payload: data,
}
return push(&clone)
}
}
}
-1
View File
@@ -59,7 +59,6 @@ func (c *Conn) Init() {
}
fmt.Printf("TODO: webrtc ontrack %+v\n", remote)
fmt.Printf("TODO: webrtc ontrack %#v\n", remote)
})
// OK connection:
@@ -3,6 +3,7 @@ package webrtc
import (
"encoding/json"
"github.com/AlexxIT/go2rtc/pkg/h264"
"github.com/AlexxIT/go2rtc/pkg/h265"
"github.com/AlexxIT/go2rtc/pkg/streamer"
"github.com/pion/rtp"
"github.com/pion/webrtc/v3"
@@ -51,7 +52,8 @@ func (c *Conn) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.
return trackLocal.WriteRTP(packet)
}
if codec.Name == streamer.CodecH264 {
switch codec.Name {
case streamer.CodecH264:
wrapper := h264.RTPPay(1200)
push = wrapper(push)
@@ -61,6 +63,15 @@ func (c *Conn) AddTrack(media *streamer.Media, track *streamer.Track) *streamer.
wrapper = h264.RTPDepay(track)
}
push = wrapper(push)
case streamer.CodecH265:
// SafariPay because it is the only browser in the world
// that supports WebRTC + H265
wrapper := h265.SafariPay(1200)
push = wrapper(push)
wrapper = h265.RTPDepay(track)
push = wrapper(push)
}
track = track.Bind(push)