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Avoid overflow in bitrate computation.
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parent
0d8fdc5c20
commit
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3 changed files with 36 additions and 8 deletions
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@ -64,6 +64,7 @@ type ChatHistoryEntry struct {
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const (
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LowBitrate = 100 * 1024
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MinBitrate = LowBitrate * 2
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MaxBitrate = 1024 * 1024 * 1024
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)
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type Group struct {
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@ -881,6 +881,15 @@ func rtcpUpListener(track *rtpUpTrack) {
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}
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}
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// saturating addition
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func sadd(x, y uint64) uint64 {
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s, c := bits.Add64(x, y, 0)
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if c != 0 {
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return ^uint64(0)
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}
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return s
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}
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func maxUpBitrate(t *rtpUpTrack) uint64 {
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minrate := ^uint64(0)
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maxrate := uint64(group.MinBitrate)
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@ -908,15 +917,13 @@ func maxUpBitrate(t *rtpUpTrack) uint64 {
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// assume that lower spatial layers take up 1/5 of
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// the throughput
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if maxsid > 0 {
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maxrate = maxrate * 5 / 4
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maxrate = sadd(maxrate, maxrate / 4)
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}
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// assume that each layer takes two times less
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// throughput than the higher one. Then we've
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// got enough slack for a factor of 2^(layers-1).
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for i := 0; i < maxtid; i++ {
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if minrate < ^uint64(0)/2 {
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minrate *= 2
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}
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minrate = sadd(minrate, minrate)
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}
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if minrate < maxrate {
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return minrate
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@ -990,15 +997,18 @@ func sendUpRTCP(up *rtpUpConnection) error {
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}
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ssrcs = append(ssrcs, uint32(t.track.SSRC()))
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if t.Kind() == webrtc.RTPCodecTypeAudio {
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rate += 100 * 1024
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rate = sadd(rate, 100 * 1024)
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} else if t.Label() == "l" {
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rate += group.LowBitrate
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rate = sadd(rate, group.LowBitrate)
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} else {
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rate += maxUpBitrate(t)
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rate = sadd(rate, maxUpBitrate(t))
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}
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}
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if rate < ^uint64(0) && len(ssrcs) > 0 {
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if rate > group.MaxBitrate {
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rate = group.MaxBitrate
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}
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if len(ssrcs) > 0 {
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packets = append(packets,
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&rtcp.ReceiverEstimatedMaximumBitrate{
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Bitrate: float32(rate),
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@ -37,3 +37,20 @@ func TestDownTrackAtomics(t *testing.T) {
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t.Errorf("Expected %v, got %v", info, info2)
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}
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}
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func TestSadd(t *testing.T) {
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ts := []struct{ x, y, z uint64 }{
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{0, 0, 0},
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{1, 2, 3},
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{^uint64(0) - 10, 5, ^uint64(0) - 5},
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{^uint64(0) - 10, 15, ^uint64(0)},
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}
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for _, tt := range ts {
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z := sadd(tt.x, tt.y)
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if z != tt.z {
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t.Errorf("%v + %v: expected %v, got %v",
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tt.x, tt.y, tt.z, z,
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)
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}
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}
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}
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