mirror of
https://github.com/aler9/rtsp-simple-server
synced 2025-10-31 19:13:22 +08:00
396 lines
8.7 KiB
Go
396 lines
8.7 KiB
Go
package webrtc
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import (
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"sync/atomic"
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"time"
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"github.com/bluenviron/gortsplib/v4/pkg/rtcpreceiver"
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"github.com/pion/rtcp"
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"github.com/pion/rtp"
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"github.com/pion/webrtc/v4"
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"github.com/bluenviron/mediamtx/internal/counterdumper"
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"github.com/bluenviron/mediamtx/internal/logger"
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)
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const (
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keyFrameInterval = 2 * time.Second
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mimeTypeMultiopus = "audio/multiopus"
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mimeTypeL16 = "audio/L16"
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)
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var incomingVideoCodecs = []webrtc.RTPCodecParameters{
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeAV1,
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ClockRate: 90000,
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SDPFmtpLine: "profile=1",
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},
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PayloadType: 96,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeAV1,
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ClockRate: 90000,
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},
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PayloadType: 97,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeVP9,
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ClockRate: 90000,
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SDPFmtpLine: "profile-id=3",
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},
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PayloadType: 98,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeVP9,
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ClockRate: 90000,
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SDPFmtpLine: "profile-id=2",
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},
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PayloadType: 99,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeVP9,
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ClockRate: 90000,
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SDPFmtpLine: "profile-id=1",
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},
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PayloadType: 100,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeVP9,
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ClockRate: 90000,
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SDPFmtpLine: "profile-id=0",
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},
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PayloadType: 101,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeVP8,
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ClockRate: 90000,
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},
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PayloadType: 102,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeH265,
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ClockRate: 90000,
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SDPFmtpLine: "level-id=93;profile-id=2;tier-flag=0;tx-mode=SRST",
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},
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PayloadType: 103,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeH265,
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ClockRate: 90000,
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SDPFmtpLine: "level-id=93;profile-id=1;tier-flag=0;tx-mode=SRST",
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},
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PayloadType: 104,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeH264,
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ClockRate: 90000,
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SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f",
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},
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PayloadType: 105,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeH264,
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ClockRate: 90000,
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SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f",
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},
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PayloadType: 106,
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},
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}
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var incomingAudioCodecs = []webrtc.RTPCodecParameters{
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: mimeTypeMultiopus,
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ClockRate: 48000,
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Channels: 3,
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SDPFmtpLine: "channel_mapping=0,2,1;num_streams=2;coupled_streams=1",
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},
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PayloadType: 112,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: mimeTypeMultiopus,
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ClockRate: 48000,
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Channels: 4,
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SDPFmtpLine: "channel_mapping=0,1,2,3;num_streams=2;coupled_streams=2",
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},
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PayloadType: 113,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: mimeTypeMultiopus,
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ClockRate: 48000,
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Channels: 5,
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SDPFmtpLine: "channel_mapping=0,4,1,2,3;num_streams=3;coupled_streams=2",
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},
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PayloadType: 114,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: mimeTypeMultiopus,
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ClockRate: 48000,
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Channels: 6,
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SDPFmtpLine: "channel_mapping=0,4,1,2,3,5;num_streams=4;coupled_streams=2",
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},
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PayloadType: 115,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: mimeTypeMultiopus,
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ClockRate: 48000,
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Channels: 7,
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SDPFmtpLine: "channel_mapping=0,4,1,2,3,5,6;num_streams=4;coupled_streams=4",
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},
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PayloadType: 116,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: mimeTypeMultiopus,
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ClockRate: 48000,
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Channels: 8,
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SDPFmtpLine: "channel_mapping=0,6,1,4,5,2,3,7;num_streams=5;coupled_streams=4",
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},
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PayloadType: 117,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeOpus,
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ClockRate: 48000,
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Channels: 2,
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SDPFmtpLine: "minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1",
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},
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PayloadType: 111,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypeG722,
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ClockRate: 8000,
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},
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PayloadType: 9,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypePCMU,
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ClockRate: 8000,
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Channels: 2,
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},
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PayloadType: 118,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypePCMA,
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ClockRate: 8000,
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Channels: 2,
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},
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PayloadType: 119,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypePCMU,
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ClockRate: 8000,
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},
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PayloadType: 0,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: webrtc.MimeTypePCMA,
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ClockRate: 8000,
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},
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PayloadType: 8,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: mimeTypeL16,
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ClockRate: 8000,
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Channels: 2,
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},
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PayloadType: 120,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: mimeTypeL16,
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ClockRate: 16000,
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Channels: 2,
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},
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PayloadType: 121,
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},
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{
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RTPCodecCapability: webrtc.RTPCodecCapability{
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MimeType: mimeTypeL16,
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ClockRate: 48000,
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Channels: 2,
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},
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PayloadType: 122,
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},
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}
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// IncomingTrack is an incoming track.
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type IncomingTrack struct {
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OnPacketRTP func(*rtp.Packet, time.Time)
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useAbsoluteTimestamp bool
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track *webrtc.TrackRemote
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receiver *webrtc.RTPReceiver
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writeRTCP func([]rtcp.Packet) error
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log logger.Writer
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rtpPacketsReceived *uint64
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rtpPacketsLost *uint64
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packetsLost *counterdumper.CounterDumper
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rtcpReceiver *rtcpreceiver.RTCPReceiver
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}
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func (t *IncomingTrack) initialize() {
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t.OnPacketRTP = func(*rtp.Packet, time.Time) {}
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}
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// Codec returns the track codec.
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func (t *IncomingTrack) Codec() webrtc.RTPCodecParameters {
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return t.track.Codec()
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}
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// ClockRate returns the clock rate. Needed by rtptime.GlobalDecoder
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func (t *IncomingTrack) ClockRate() int {
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return int(t.track.Codec().ClockRate)
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}
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// PTSEqualsDTS returns whether PTS equals DTS. Needed by rtptime.GlobalDecoder
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func (*IncomingTrack) PTSEqualsDTS(*rtp.Packet) bool {
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return true
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}
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func (t *IncomingTrack) start() {
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t.packetsLost = &counterdumper.CounterDumper{
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OnReport: func(val uint64) {
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t.log.Log(logger.Warn, "%d RTP %s lost",
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val,
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func() string {
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if val == 1 {
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return "packet"
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}
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return "packets"
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}())
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},
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}
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t.packetsLost.Start()
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t.rtcpReceiver = &rtcpreceiver.RTCPReceiver{
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ClockRate: int(t.track.Codec().ClockRate),
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UnrealiableTransport: true,
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Period: 1 * time.Second,
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WritePacketRTCP: func(p rtcp.Packet) {
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t.writeRTCP([]rtcp.Packet{p}) //nolint:errcheck
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},
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}
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err := t.rtcpReceiver.Initialize()
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if err != nil {
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panic(err)
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}
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// read incoming RTCP packets.
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// incoming RTCP packets must always be read to make interceptors work.
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go func() {
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buf := make([]byte, 1500)
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for {
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n, _, err2 := t.receiver.Read(buf)
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if err2 != nil {
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return
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}
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pkts, err2 := rtcp.Unmarshal(buf[:n])
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if err2 != nil {
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panic(err2)
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}
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for _, pkt := range pkts {
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if sr, ok := pkt.(*rtcp.SenderReport); ok {
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t.rtcpReceiver.ProcessSenderReport(sr, time.Now())
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}
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}
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}
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}()
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// send period key frame requests
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if t.track.Kind() == webrtc.RTPCodecTypeVideo {
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go func() {
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keyframeTicker := time.NewTicker(keyFrameInterval)
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defer keyframeTicker.Stop()
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for range keyframeTicker.C {
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err2 := t.writeRTCP([]rtcp.Packet{
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&rtcp.PictureLossIndication{
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MediaSSRC: uint32(t.track.SSRC()),
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},
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})
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if err2 != nil {
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return
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}
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}
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}()
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}
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// read incoming RTP packets.
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go func() {
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for {
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pkt, _, err2 := t.track.ReadRTP()
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if err2 != nil {
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return
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}
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packets, lost, err2 := t.rtcpReceiver.ProcessPacket2(pkt, time.Now(), true)
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if err2 != nil {
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t.log.Log(logger.Warn, err2.Error())
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continue
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}
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if lost != 0 {
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atomic.AddUint64(t.rtpPacketsLost, lost)
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t.packetsLost.Add(lost)
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// do not return
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}
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atomic.AddUint64(t.rtpPacketsReceived, uint64(len(packets)))
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var ntp time.Time
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if t.useAbsoluteTimestamp {
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var avail bool
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ntp, avail = t.rtcpReceiver.PacketNTP(pkt.Timestamp)
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if !avail {
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t.log.Log(logger.Warn, "received RTP packet without absolute time, skipping it")
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continue
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}
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} else {
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ntp = time.Now()
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}
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for _, pkt := range packets {
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// sometimes Chrome sends empty RTP packets. ignore them.
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if len(pkt.Payload) == 0 {
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continue
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}
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t.OnPacketRTP(pkt, ntp)
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}
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}
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}()
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}
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func (t *IncomingTrack) close() {
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if t.packetsLost != nil {
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t.packetsLost.Stop()
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}
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if t.rtcpReceiver != nil {
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t.rtcpReceiver.Close()
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}
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}
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