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https://github.com/jech/galene.git
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Simulcast.
This commit is contained in:
parent
f1a15f07db
commit
795a40ceaf
11 changed files with 218 additions and 112 deletions
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@ -135,8 +135,17 @@ A peer must explicitly request the streams that it wants to receive.
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```
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```
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The field `request` is a dictionary that maps the labels of requested
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The field `request` is a dictionary that maps the labels of requested
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streams to a list containing either 'audio', 'video' or both. An entry
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streams to a list containing either 'audio', or one of 'video' or
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with an empty key `''` serves as default.
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'video-low'. The empty key `''` serves as default. For example:
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```javascript
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{
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type: 'request',
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request: {
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camera: ['audio', 'video-low'],
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'': ['audio', 'video']
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}
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}
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## Pushing streams
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## Pushing streams
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@ -157,16 +166,22 @@ A stream is created by the sender with the `offer` message:
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If a stream with the same id exists, then this is a renegotation;
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If a stream with the same id exists, then this is a renegotation;
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otherwise this message creates a new stream. If the field `replace` is
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otherwise this message creates a new stream. If the field `replace` is
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not empty, then this request additionally requests that an existing stream
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not empty, then this request additionally requests that an existing stream
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with the given id should be closed, and the new stream should replace it.
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with the given id should be closed, and the new stream should replace it;
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this is used most notably when changing the simulcast envelope.
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The field `label` is one of `camera`, `screenshare` or `video`, as in the
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The field `label` is one of `camera`, `screenshare` or `video`, and will
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`request` message.
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be matched against the keys sent by the receiver in their `request` message.
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The field `sdp` contains the raw SDP string (i.e. the `sdp` field of
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The field `sdp` contains the raw SDP string (i.e. the `sdp` field of
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a JSEP session description). Galène will interpret the `nack`,
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a JSEP session description). Galène will interpret the `nack`,
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`nack pli`, `ccm fir` and `goog-remb` RTCP feedback types, and act
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`nack pli`, `ccm fir` and `goog-remb` RTCP feedback types, and act
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accordingly.
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accordingly.
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The sender may either send a single stream per media section in the SDP,
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or use rid-based simulcasting. In the latter case, it should send two
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video streams, one with rid 'h' and high throughput, and one with rid 'l'
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and throughput limited to roughly 100kbit/s.
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The receiver may either abort the stream immediately (see below), or send
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The receiver may either abort the stream immediately (see below), or send
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an answer.
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an answer.
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@ -25,6 +25,7 @@ type UpTrack interface {
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AddLocal(DownTrack) error
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AddLocal(DownTrack) error
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DelLocal(DownTrack) bool
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DelLocal(DownTrack) bool
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Kind() webrtc.RTPCodecType
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Kind() webrtc.RTPCodecType
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Label() string
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Codec() webrtc.RTPCodecCapability
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Codec() webrtc.RTPCodecCapability
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// get a recent packet. Returns 0 if the packet is not in cache.
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// get a recent packet. Returns 0 if the packet is not in cache.
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GetRTP(seqno uint16, result []byte) uint16
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GetRTP(seqno uint16, result []byte) uint16
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@ -33,7 +34,6 @@ type UpTrack interface {
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// Type Down represents a connection in the server to client direction.
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// Type Down represents a connection in the server to client direction.
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type Down interface {
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type Down interface {
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GetMaxBitrate(now uint64) uint64
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}
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}
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// Type DownTrack represents a track in the server to client direction.
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// Type DownTrack represents a track in the server to client direction.
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@ -42,4 +42,5 @@ type DownTrack interface {
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Accumulate(bytes uint32)
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Accumulate(bytes uint32)
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SetTimeOffset(ntp uint64, rtp uint32)
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SetTimeOffset(ntp uint64, rtp uint32)
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SetCname(string)
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SetCname(string)
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GetMaxBitrate() uint64
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}
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}
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@ -600,7 +600,7 @@ func (conn *diskConn) initWriter(width, height uint32) error {
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return nil
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return nil
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}
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}
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func (down *diskConn) GetMaxBitrate(now uint64) uint64 {
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func (t *diskTrack) GetMaxBitrate() uint64 {
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return ^uint64(0)
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return ^uint64(0)
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}
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}
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@ -13,6 +13,7 @@ import (
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"time"
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"time"
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"github.com/pion/ice/v2"
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"github.com/pion/ice/v2"
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"github.com/pion/sdp/v3"
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"github.com/pion/webrtc/v3"
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"github.com/pion/webrtc/v3"
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)
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)
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@ -60,7 +61,8 @@ type ChatHistoryEntry struct {
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}
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}
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const (
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const (
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MinBitrate = 200000
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LowBitrate = 100000
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MinBitrate = 2 * LowBitrate
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)
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)
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type Group struct {
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type Group struct {
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@ -252,6 +254,12 @@ func APIFromCodecs(codecs []webrtc.RTPCodecCapability) (*webrtc.API, error) {
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if UDPMin > 0 && UDPMax > 0 {
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if UDPMin > 0 && UDPMax > 0 {
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s.SetEphemeralUDPPortRange(UDPMin, UDPMax)
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s.SetEphemeralUDPPortRange(UDPMin, UDPMax)
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}
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}
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m.RegisterHeaderExtension(
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webrtc.RTPHeaderExtensionCapability{sdp.SDESMidURI},
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webrtc.RTPCodecTypeVideo)
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m.RegisterHeaderExtension(
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webrtc.RTPHeaderExtensionCapability{sdp.SDESRTPStreamIDURI},
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webrtc.RTPCodecTypeVideo)
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return webrtc.NewAPI(
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return webrtc.NewAPI(
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webrtc.WithSettingEngine(s),
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webrtc.WithSettingEngine(s),
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@ -77,15 +77,16 @@ type downTrackAtomics struct {
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}
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}
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type rtpDownTrack struct {
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type rtpDownTrack struct {
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track *webrtc.TrackLocalStaticRTP
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track *webrtc.TrackLocalStaticRTP
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sender *webrtc.RTPSender
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sender *webrtc.RTPSender
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remote conn.UpTrack
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remote conn.UpTrack
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ssrc webrtc.SSRC
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ssrc webrtc.SSRC
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maxBitrate *bitrate
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maxBitrate *bitrate
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rate *estimator.Estimator
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maxREMBBitrate *bitrate
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stats *receiverStats
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rate *estimator.Estimator
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atomics *downTrackAtomics
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stats *receiverStats
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cname atomic.Value
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atomics *downTrackAtomics
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cname atomic.Value
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}
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}
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func (down *rtpDownTrack) WriteRTP(packet *rtp.Packet) error {
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func (down *rtpDownTrack) WriteRTP(packet *rtp.Packet) error {
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@ -140,7 +141,6 @@ type rtpDownConnection struct {
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id string
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id string
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pc *webrtc.PeerConnection
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pc *webrtc.PeerConnection
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remote conn.Up
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remote conn.Up
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maxREMBBitrate *bitrate
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iceCandidates []*webrtc.ICECandidateInit
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iceCandidates []*webrtc.ICECandidateInit
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negotiationNeeded int
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negotiationNeeded int
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@ -174,31 +174,22 @@ func newDownConn(c group.Client, id string, remote conn.Up) (*rtpDownConnection,
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id: id,
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id: id,
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pc: pc,
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pc: pc,
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remote: remote,
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remote: remote,
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maxREMBBitrate: new(bitrate),
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}
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}
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return conn, nil
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return conn, nil
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}
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}
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func (down *rtpDownConnection) GetMaxBitrate(now uint64) uint64 {
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func (t *rtpDownTrack) GetMaxBitrate() uint64 {
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rate := down.maxREMBBitrate.Get(now)
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now := rtptime.Jiffies()
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var trackRate uint64
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r := t.maxBitrate.Get(now)
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tracks := down.getTracks()
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if r == ^uint64(0) {
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for _, t := range tracks {
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r = 512 * 1024
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r := t.maxBitrate.Get(now)
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if r == ^uint64(0) {
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if t.track.Kind() == webrtc.RTPCodecTypeAudio {
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r = 128 * 1024
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} else {
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r = 512 * 1024
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}
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}
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trackRate += r
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}
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}
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if trackRate < rate {
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rr := t.maxREMBBitrate.Get(now)
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return trackRate
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if rr == 0 || r < rr {
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return r
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}
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}
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return rate
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return rr
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}
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}
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func (down *rtpDownConnection) addICECandidate(candidate *webrtc.ICECandidateInit) error {
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func (down *rtpDownConnection) addICECandidate(candidate *webrtc.ICECandidateInit) error {
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@ -311,6 +302,10 @@ func (up *rtpUpTrack) GetRTP(seqno uint16, result []byte) uint16 {
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return up.cache.Get(seqno, result)
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return up.cache.Get(seqno, result)
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}
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}
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func (up *rtpUpTrack) Label() string {
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return up.track.RID()
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}
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func (up *rtpUpTrack) Kind() webrtc.RTPCodecType {
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func (up *rtpUpTrack) Kind() webrtc.RTPCodecType {
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return up.track.Kind()
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return up.track.Kind()
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}
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}
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@ -687,7 +682,7 @@ func rtcpUpListener(conn *rtpUpConnection, track *rtpUpTrack, r *webrtc.RTPRecei
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for {
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for {
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firstSR := false
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firstSR := false
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n, _, err := r.Read(buf)
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n, _, err := r.ReadSimulcast(buf, track.track.RID())
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if err != nil {
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if err != nil {
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if err != io.EOF && err != io.ErrClosedPipe {
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if err != io.EOF && err != io.ErrClosedPipe {
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log.Printf("Read RTCP: %v", err)
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log.Printf("Read RTCP: %v", err)
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@ -752,11 +747,11 @@ func rtcpUpListener(conn *rtpUpConnection, track *rtpUpTrack, r *webrtc.RTPRecei
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}
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}
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}
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}
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func sendUpRTCP(conn *rtpUpConnection) error {
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func sendUpRTCP(up *rtpUpConnection) error {
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tracks := conn.getTracks()
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tracks := up.getTracks()
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if len(conn.tracks) == 0 {
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if len(up.tracks) == 0 {
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state := conn.pc.ConnectionState()
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state := up.pc.ConnectionState()
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if state == webrtc.PeerConnectionStateClosed {
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if state == webrtc.PeerConnectionStateClosed {
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return io.ErrClosedPipe
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return io.ErrClosedPipe
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}
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}
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@ -765,7 +760,7 @@ func sendUpRTCP(conn *rtpUpConnection) error {
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now := rtptime.Jiffies()
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now := rtptime.Jiffies()
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reports := make([]rtcp.ReceptionReport, 0, len(conn.tracks))
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reports := make([]rtcp.ReceptionReport, 0, len(up.tracks))
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for _, t := range tracks {
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for _, t := range tracks {
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updateUpTrack(t)
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updateUpTrack(t)
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stats := t.cache.GetStats(true)
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stats := t.cache.GetStats(true)
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@ -810,29 +805,38 @@ func sendUpRTCP(conn *rtpUpConnection) error {
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},
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},
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}
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}
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rate := ^uint64(0)
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local := conn.getLocal()
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for _, l := range local {
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r := l.GetMaxBitrate(now)
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if r < rate {
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rate = r
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}
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}
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if rate < group.MinBitrate {
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rate = group.MinBitrate
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}
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var ssrcs []uint32
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var ssrcs []uint32
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var rate uint64
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for _, t := range tracks {
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for _, t := range tracks {
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if !t.hasRtcpFb("goog-remb", "") {
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if !t.hasRtcpFb("goog-remb", "") {
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continue
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continue
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}
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}
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ssrcs = append(ssrcs, uint32(t.track.SSRC()))
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ssrcs = append(ssrcs, uint32(t.track.SSRC()))
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var r uint64
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if t.Kind() == webrtc.RTPCodecTypeAudio {
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r = 100 * 1024
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} else if t.Label() == "l" {
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r = group.LowBitrate
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} else {
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local := t.getLocal()
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r = ^uint64(0)
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for _, down := range local {
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rr := down.GetMaxBitrate()
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if rr < group.MinBitrate {
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rr = group.MinBitrate
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}
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if r > rr {
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r = rr
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}
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}
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if r == ^uint64(0) {
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r = 512 * 1024
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}
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}
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rate += r
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}
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}
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if len(ssrcs) > 0 {
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if rate < ^uint64(0) && len(ssrcs) > 0 {
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packets = append(packets,
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packets = append(packets,
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&rtcp.ReceiverEstimatedMaximumBitrate{
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&rtcp.ReceiverEstimatedMaximumBitrate{
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Bitrate: rate,
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Bitrate: rate,
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@ -840,7 +844,7 @@ func sendUpRTCP(conn *rtpUpConnection) error {
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},
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},
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)
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)
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}
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}
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return conn.pc.WriteRTCP(packets)
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return up.pc.WriteRTCP(packets)
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}
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}
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func rtcpUpSender(conn *rtpUpConnection) {
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func rtcpUpSender(conn *rtpUpConnection) {
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@ -1049,7 +1053,7 @@ func rtcpDownListener(conn *rtpDownConnection, track *rtpDownTrack, s *webrtc.RT
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log.Printf("sendFIR: %v", err)
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log.Printf("sendFIR: %v", err)
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}
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}
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case *rtcp.ReceiverEstimatedMaximumBitrate:
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case *rtcp.ReceiverEstimatedMaximumBitrate:
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conn.maxREMBBitrate.Set(p.Bitrate, jiffies)
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track.maxREMBBitrate.Set(p.Bitrate, jiffies)
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case *rtcp.ReceiverReport:
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case *rtcp.ReceiverReport:
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for _, r := range p.Reports {
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for _, r := range p.Reports {
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if r.SSRC == uint32(track.ssrc) {
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if r.SSRC == uint32(track.ssrc) {
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@ -149,6 +149,16 @@ func readLoop(conn *rtpUpConnection, track *rtpUpTrack) {
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kf, _ := isKeyframe(codec.MimeType, &packet)
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kf, _ := isKeyframe(codec.MimeType, &packet)
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if packet.Extension {
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packet.Extension = false
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packet.Extensions = nil
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bytes, err = packet.MarshalTo(buf)
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if err != nil {
|
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log.Printf("%v", err)
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continue
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}
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}
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|
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first, index := track.cache.Store(
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first, index := track.cache.Store(
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packet.SequenceNumber, packet.Timestamp,
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packet.SequenceNumber, packet.Timestamp,
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kf, packet.Marker, buf[:bytes],
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kf, packet.Marker, buf[:bytes],
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@ -47,7 +47,6 @@ func (c *webClient) GetStats() *stats.Client {
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for _, down := range c.down {
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for _, down := range c.down {
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conns := stats.Conn{
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conns := stats.Conn{
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Id: down.id,
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Id: down.id,
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MaxBitrate: down.GetMaxBitrate(jiffies),
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}
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}
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for _, t := range down.tracks {
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for _, t := range down.tracks {
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rate, _ := t.rate.Estimate()
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rate, _ := t.rate.Estimate()
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@ -380,14 +380,15 @@ func addDownTrackUnlocked(conn *rtpDownConnection, remoteTrack *rtpUpTrack, remo
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}
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}
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track := &rtpDownTrack{
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track := &rtpDownTrack{
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track: local,
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track: local,
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sender: sender,
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sender: sender,
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ssrc: parms.Encodings[0].SSRC,
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ssrc: parms.Encodings[0].SSRC,
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remote: remoteTrack,
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remote: remoteTrack,
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maxBitrate: new(bitrate),
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maxBitrate: new(bitrate),
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stats: new(receiverStats),
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maxREMBBitrate: new(bitrate),
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rate: estimator.New(time.Second),
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stats: new(receiverStats),
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atomics: &downTrackAtomics{},
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rate: estimator.New(time.Second),
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atomics: &downTrackAtomics{},
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}
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}
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|
||||||
conn.tracks = append(conn.tracks, track)
|
conn.tracks = append(conn.tracks, track)
|
||||||
|
@ -646,33 +647,60 @@ func requestedTracks(c *webClient, up conn.Up, tracks []conn.UpTrack) []conn.UpT
|
||||||
return nil
|
return nil
|
||||||
}
|
}
|
||||||
|
|
||||||
var audio, video bool
|
var audio, video, videoLow bool
|
||||||
for _, s := range r {
|
for _, s := range r {
|
||||||
switch s {
|
switch s {
|
||||||
case "audio":
|
case "audio":
|
||||||
audio = true
|
audio = true
|
||||||
case "video":
|
case "video":
|
||||||
video = true
|
video = true
|
||||||
|
case "video-low":
|
||||||
|
videoLow = true
|
||||||
default:
|
default:
|
||||||
log.Printf("client requested unknown value %v", s)
|
log.Printf("client requested unknown value %v", s)
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
|
find := func(kind webrtc.RTPCodecType, labels ...string) conn.UpTrack {
|
||||||
|
for _, l := range labels {
|
||||||
|
for _, t := range tracks {
|
||||||
|
if t.Kind() != kind {
|
||||||
|
continue
|
||||||
|
}
|
||||||
|
if t.Label() == l {
|
||||||
|
return t
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
for _, t := range tracks {
|
||||||
|
if t.Kind() != kind {
|
||||||
|
continue
|
||||||
|
}
|
||||||
|
return t
|
||||||
|
}
|
||||||
|
return nil
|
||||||
|
}
|
||||||
|
|
||||||
var ts []conn.UpTrack
|
var ts []conn.UpTrack
|
||||||
if audio {
|
if audio {
|
||||||
for _, t := range tracks {
|
t := find(webrtc.RTPCodecTypeAudio)
|
||||||
if t.Kind() == webrtc.RTPCodecTypeAudio {
|
if t != nil {
|
||||||
ts = append(ts, t)
|
ts = append(ts, t)
|
||||||
break
|
|
||||||
}
|
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
if video {
|
if video {
|
||||||
for _, t := range tracks {
|
t := find(
|
||||||
if t.Kind() == webrtc.RTPCodecTypeVideo {
|
webrtc.RTPCodecTypeVideo, "h", "m", "video",
|
||||||
ts = append(ts, t)
|
)
|
||||||
break
|
if t != nil {
|
||||||
}
|
ts = append(ts, t)
|
||||||
|
}
|
||||||
|
} else if videoLow {
|
||||||
|
t := find(
|
||||||
|
webrtc.RTPCodecTypeVideo, "l", "m", "video",
|
||||||
|
)
|
||||||
|
if t != nil {
|
||||||
|
ts = append(ts, t)
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
|
|
|
@ -213,7 +213,9 @@
|
||||||
<select id="requestselect" class="select select-inline">
|
<select id="requestselect" class="select select-inline">
|
||||||
<option value="">nothing</option>
|
<option value="">nothing</option>
|
||||||
<option value="audio">audio only</option>
|
<option value="audio">audio only</option>
|
||||||
|
<option value="screenshare-low">screen share (low)</option>
|
||||||
<option value="screenshare">screen share</option>
|
<option value="screenshare">screen share</option>
|
||||||
|
<option value="everything-low">everything (low)</option>
|
||||||
<option value="everything" selected>everything</option>
|
<option value="everything" selected>everything</option>
|
||||||
</select>
|
</select>
|
||||||
</form>
|
</form>
|
||||||
|
|
|
@ -78,6 +78,7 @@ function getUserPass() {
|
||||||
* @property {boolean} [localMute]
|
* @property {boolean} [localMute]
|
||||||
* @property {string} [video]
|
* @property {string} [video]
|
||||||
* @property {string} [audio]
|
* @property {string} [audio]
|
||||||
|
* @property {boolean} [simulcast]
|
||||||
* @property {string} [send]
|
* @property {string} [send]
|
||||||
* @property {string} [request]
|
* @property {string} [request]
|
||||||
* @property {boolean} [activityDetection]
|
* @property {boolean} [activityDetection]
|
||||||
|
@ -550,9 +551,15 @@ function mapRequest(what) {
|
||||||
case 'audio':
|
case 'audio':
|
||||||
return {'': ['audio']};
|
return {'': ['audio']};
|
||||||
break;
|
break;
|
||||||
|
case 'screenshare-low':
|
||||||
|
return {screenshare: ['audio','video-low'], '': ['audio']};
|
||||||
|
break;
|
||||||
case 'screenshare':
|
case 'screenshare':
|
||||||
return {screenshare: ['audio','video'], '': ['audio']};
|
return {screenshare: ['audio','video'], '': ['audio']};
|
||||||
break;
|
break;
|
||||||
|
case 'everything-low':
|
||||||
|
return {'': ['audio','video-low']};
|
||||||
|
break;
|
||||||
case 'everything':
|
case 'everything':
|
||||||
return {'': ['audio','video']}
|
return {'': ['audio','video']}
|
||||||
break;
|
break;
|
||||||
|
@ -611,20 +618,25 @@ getInputElement('fileinput').onchange = function(e) {
|
||||||
function gotUpStats(stats) {
|
function gotUpStats(stats) {
|
||||||
let c = this;
|
let c = this;
|
||||||
|
|
||||||
let text = '';
|
let values = [];
|
||||||
|
|
||||||
c.pc.getSenders().forEach(s => {
|
for(let id in stats) {
|
||||||
let tid = s.track && s.track.id;
|
if(stats[id] && stats[id]['outbound-rtp']) {
|
||||||
let stats = tid && c.stats[tid];
|
let rate = stats[id]['outbound-rtp'].rate;
|
||||||
let rate = stats && stats['outbound-rtp'] && stats['outbound-rtp'].rate;
|
if(typeof rate === 'number') {
|
||||||
if(typeof rate === 'number') {
|
values.push(rate);
|
||||||
if(text)
|
}
|
||||||
text = text + ' + ';
|
|
||||||
text = text + Math.round(rate / 1000) + 'kbps';
|
|
||||||
}
|
}
|
||||||
});
|
}
|
||||||
|
|
||||||
setLabel(c, text);
|
if(values.length === 0) {
|
||||||
|
setLabel(c, '');
|
||||||
|
} else {
|
||||||
|
values.sort((x,y) => x - y);
|
||||||
|
setLabel(c, values
|
||||||
|
.map(x => Math.round(x / 1000).toString())
|
||||||
|
.reduce((x, y) => x + '+' + y));
|
||||||
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
/**
|
/**
|
||||||
|
@ -800,6 +812,7 @@ function newUpStream(localId) {
|
||||||
* @param {number} [bps]
|
* @param {number} [bps]
|
||||||
*/
|
*/
|
||||||
async function setMaxVideoThroughput(c, bps) {
|
async function setMaxVideoThroughput(c, bps) {
|
||||||
|
let simulcast = doSimulcast();
|
||||||
let senders = c.pc.getSenders();
|
let senders = c.pc.getSenders();
|
||||||
for(let i = 0; i < senders.length; i++) {
|
for(let i = 0; i < senders.length; i++) {
|
||||||
let s = senders[i];
|
let s = senders[i];
|
||||||
|
@ -808,17 +821,17 @@ async function setMaxVideoThroughput(c, bps) {
|
||||||
let p = s.getParameters();
|
let p = s.getParameters();
|
||||||
if(!p.encodings)
|
if(!p.encodings)
|
||||||
p.encodings = [{}];
|
p.encodings = [{}];
|
||||||
p.encodings.forEach(e => {
|
if((!simulcast && p.encodings.length != 1) ||
|
||||||
if(bps > 0)
|
(simulcast && p.encodings.length != 2)) {
|
||||||
e.maxBitrate = bps;
|
// change the simulcast envelope
|
||||||
else
|
await replaceUpStream(c);
|
||||||
delete e.maxBitrate;
|
return;
|
||||||
});
|
|
||||||
try {
|
|
||||||
await s.setParameters(p);
|
|
||||||
} catch(e) {
|
|
||||||
console.error(e);
|
|
||||||
}
|
}
|
||||||
|
p.encodings.forEach(e => {
|
||||||
|
if(!e.rid || e.rid === 'h')
|
||||||
|
e.maxBitrate = bps || unlimitedRate;
|
||||||
|
});
|
||||||
|
await s.setParameters(p);
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
|
@ -1022,6 +1035,19 @@ function isSafari() {
|
||||||
return ua.indexOf('safari') >= 0 && ua.indexOf('chrome') < 0;
|
return ua.indexOf('safari') >= 0 && ua.indexOf('chrome') < 0;
|
||||||
}
|
}
|
||||||
|
|
||||||
|
const unlimitedRate = 1000000000;
|
||||||
|
const simulcastRate = 100000;
|
||||||
|
|
||||||
|
/**
|
||||||
|
* @returns {boolean}
|
||||||
|
*/
|
||||||
|
function doSimulcast() {
|
||||||
|
if(!getSettings().simulcast)
|
||||||
|
return false;
|
||||||
|
let bps = getMaxVideoThroughput();
|
||||||
|
return bps <= 0 || bps >= 2 * simulcastRate;
|
||||||
|
}
|
||||||
|
|
||||||
/**
|
/**
|
||||||
* Sets up c to send the given stream. Some extra parameters are stored
|
* Sets up c to send the given stream. Some extra parameters are stored
|
||||||
* in c.userdata.
|
* in c.userdata.
|
||||||
|
@ -1029,6 +1055,7 @@ function isSafari() {
|
||||||
* @param {Stream} c
|
* @param {Stream} c
|
||||||
* @param {MediaStream} stream
|
* @param {MediaStream} stream
|
||||||
*/
|
*/
|
||||||
|
|
||||||
function setUpStream(c, stream) {
|
function setUpStream(c, stream) {
|
||||||
if(c.stream != null)
|
if(c.stream != null)
|
||||||
throw new Error("Setting nonempty stream");
|
throw new Error("Setting nonempty stream");
|
||||||
|
@ -1073,11 +1100,20 @@ function setUpStream(c, stream) {
|
||||||
c.close();
|
c.close();
|
||||||
};
|
};
|
||||||
|
|
||||||
let encodings = [{}];
|
let encodings = [];
|
||||||
if(t.kind === 'video') {
|
if(t.kind === 'video') {
|
||||||
|
let simulcast = doSimulcast();
|
||||||
let bps = getMaxVideoThroughput();
|
let bps = getMaxVideoThroughput();
|
||||||
if(bps > 0)
|
encodings.push({
|
||||||
encodings[0].maxBitrate = bps;
|
rid: 'h',
|
||||||
|
maxBitrate: bps || unlimitedRate,
|
||||||
|
});
|
||||||
|
if(simulcast)
|
||||||
|
encodings.push({
|
||||||
|
rid: 'l',
|
||||||
|
scaleResolutionDownBy: 2,
|
||||||
|
maxBitrate: simulcastRate,
|
||||||
|
});
|
||||||
}
|
}
|
||||||
c.pc.addTransceiver(t, {
|
c.pc.addTransceiver(t, {
|
||||||
direction: 'sendonly',
|
direction: 'sendonly',
|
||||||
|
|
|
@ -1246,17 +1246,20 @@ Stream.prototype.updateStats = async function() {
|
||||||
if(report) {
|
if(report) {
|
||||||
for(let r of report.values()) {
|
for(let r of report.values()) {
|
||||||
if(stid && r.type === 'outbound-rtp') {
|
if(stid && r.type === 'outbound-rtp') {
|
||||||
|
let id = stid;
|
||||||
|
if(r.rid)
|
||||||
|
id = id + '-' + r.rid
|
||||||
if(!('bytesSent' in r))
|
if(!('bytesSent' in r))
|
||||||
continue;
|
continue;
|
||||||
if(!stats[stid])
|
if(!stats[id])
|
||||||
stats[stid] = {};
|
stats[id] = {};
|
||||||
stats[stid][r.type] = {};
|
stats[id][r.type] = {};
|
||||||
stats[stid][r.type].timestamp = r.timestamp;
|
stats[id][r.type].timestamp = r.timestamp;
|
||||||
stats[stid][r.type].bytesSent = r.bytesSent;
|
stats[id][r.type].bytesSent = r.bytesSent;
|
||||||
if(old[stid] && old[stid][r.type])
|
if(old[id] && old[id][r.type])
|
||||||
stats[stid][r.type].rate =
|
stats[id][r.type].rate =
|
||||||
((r.bytesSent - old[stid][r.type].bytesSent) * 1000 /
|
((r.bytesSent - old[id][r.type].bytesSent) * 1000 /
|
||||||
(r.timestamp - old[stid][r.type].timestamp)) * 8;
|
(r.timestamp - old[id][r.type].timestamp)) * 8;
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
Loading…
Reference in a new issue