start transcoding audio

doc/DEV_LOG.md

@@ -28,6 +28,7 @@ go donutEngine.Stream(
 ```
 
 ref https://wiki.xiph.org/Opus_Recommended_Settings 48000 webrtc
+ref https://ffmpeg.org/ffmpeg-codecs.html#libopus-1 opus
 
 ## Date: 2/4/24
 ### Summary: Adding audio track

internal/controllers/engine/donut_engine_controller.go

@@ -12,7 +12,7 @@ import (
 type DonutEngine interface {
 	Prober() probers.DonutProber
 	Streamer() streamers.DonutStreamer
-	CompatibleStreamsFor(server, client *entities.StreamInfo) ([]entities.Stream, bool)
+	RecipeFor(server, client *entities.StreamInfo) *entities.DonutTransformRecipe
 }
 
 type DonutEngineParams struct {
@@ -79,7 +79,22 @@ func (d *donutEngine) Streamer() streamers.DonutStreamer {
 	return d.streamer
 }
 
-func (d *donutEngine) CompatibleStreamsFor(server, client *entities.StreamInfo) ([]entities.Stream, bool) {
+func (d *donutEngine) RecipeFor(server, client *entities.StreamInfo) *entities.DonutTransformRecipe {
 	// TODO: implement proper matching
-	return server.Streams, true
+	r := &entities.DonutTransformRecipe{
+		Video: entities.DonutMediaTask{
+			Action: entities.DonutBypass,
+		},
+		Audio: entities.DonutMediaTask{
+			Action: entities.DonutTranscode,
+			Codec:  entities.Opus,
+			// TODO: create method list options per Codec
+			CodecContextOptions: []entities.LibAVOptionsCodecContext{
+				// opus specifically works under 48000 Hz
+				entities.SetSampleRate(48000),
+			},
+		},
+	}
+
+	return r
 }

internal/controllers/streamers/libav_ffmpeg.go

@@ -51,13 +51,19 @@ func (c *LibAVFFmpegStreamer) Match(req *entities.RequestParams) bool {
 }
 
 type streamContext struct {
+	// IN
 	inputStream     *astiav.Stream
 	decCodec        *astiav.Codec
 	decCodecContext *astiav.CodecContext
 	decFrame        *astiav.Frame
+
+	// OUT
+	encCodec        *astiav.Codec
+	encCodecContext *astiav.CodecContext
+	encPkt          *astiav.Packet
 }
 
-type params struct {
+type libAVParams struct {
 	inputFormatContext *astiav.FormatContext
 	streams            map[int]*streamContext
 }
@@ -68,17 +74,36 @@ func (c *LibAVFFmpegStreamer) Stream(donut *entities.DonutParameters) {
 	closer := astikit.NewCloser()
 	defer closer.Close()
 
-	p := &params{
+	p := &libAVParams{
 		streams: make(map[int]*streamContext),
 	}
 
+	// it's really useful for debugging
+	astiav.SetLogLevel(astiav.LogLevelDebug)
+	astiav.SetLogCallback(func(l astiav.LogLevel, fmt, msg, parent string) {
+		c.l.Infof("ffmpeg %s: - %s", c.libAVLogToString(l), strings.TrimSpace(msg))
+	})
+
 	if err := c.prepareInput(p, closer, donut); err != nil {
 		c.onError(err, donut)
 		return
 	}
 
-	pkt := astiav.AllocPacket()
+	// the audio codec opus expects 48000 (for webrtc), therefore filters are needed
-	closer.Add(pkt.Free)
+	// so one can upscale 44100 to 48000 frames/samples through filters
+	// https://ffmpeg.org/ffmpeg-filters.html#aformat
+	// https://ffmpeg.org/ffmpeg-filters.html#aresample-1
+	// https://github.com/FFmpeg/FFmpeg/blob/8b6219a99d80cabf87c50170c009fe93092e32bd/doc/examples/resample_audio.c#L133
+	// https://github.com/FFmpeg/FFmpeg/blob/8b6219a99d80cabf87c50170c009fe93092e32bd/doc/examples/mux.c#L295
+	// ffmpeg error: more samples than frame size
+
+	if err := c.prepareOutput(p, closer, donut); err != nil {
+		c.onError(err, donut)
+		return
+	}
+
+	inPkt := astiav.AllocPacket()
+	closer.Add(inPkt.Free)
 
 	for {
 		select {
@@ -91,42 +116,71 @@ func (c *LibAVFFmpegStreamer) Stream(donut *entities.DonutParameters) {
 			return
 		default:
 
-			if err := p.inputFormatContext.ReadFrame(pkt); err != nil {
+			if err := p.inputFormatContext.ReadFrame(inPkt); err != nil {
 				if errors.Is(err, astiav.ErrEof) {
 					break
 				}
 				c.onError(err, donut)
 			}
 
-			s, ok := p.streams[pkt.StreamIndex()]
+			s, ok := p.streams[inPkt.StreamIndex()]
 			if !ok {
 				continue
 			}
-			pkt.RescaleTs(s.inputStream.TimeBase(), s.decCodecContext.TimeBase())
+			// TODO: understand why it's necessary
+			inPkt.RescaleTs(s.inputStream.TimeBase(), s.decCodecContext.TimeBase())
 
-			audioDuration := c.defineAudioDuration(s, pkt)
+			isVideo := s.decCodecContext.MediaType() == astiav.MediaTypeVideo
-			videoDuration := c.defineVideoDuration(s, pkt)
+			isVideoBypass := donut.Recipe.Video.Action == entities.DonutBypass
-
+			if isVideo && isVideoBypass {
-			if s.inputStream.CodecParameters().MediaType() == astiav.MediaTypeVideo {
 				if donut.OnVideoFrame != nil {
-					if err := donut.OnVideoFrame(pkt.Data(), entities.MediaFrameContext{
+					if err := donut.OnVideoFrame(inPkt.Data(), entities.MediaFrameContext{
-						PTS:      int(pkt.Pts()),
+						PTS:      int(inPkt.Pts()),
-						DTS:      int(pkt.Dts()),
+						DTS:      int(inPkt.Dts()),
-						Duration: videoDuration,
+						Duration: c.defineVideoDuration(s, inPkt),
 					}); err != nil {
 						c.onError(err, donut)
 						return
 					}
 				}
+				continue
 			}
 
-			if s.inputStream.CodecParameters().MediaType() == astiav.MediaTypeAudio {
+			isAudio := s.decCodecContext.MediaType() == astiav.MediaTypeAudio
+			isAudioBypass := donut.Recipe.Audio.Action == entities.DonutBypass
+			if isAudio && isAudioBypass {
 				if donut.OnAudioFrame != nil {
-					donut.OnAudioFrame(pkt.Data(), entities.MediaFrameContext{
+					if err := donut.OnAudioFrame(inPkt.Data(), entities.MediaFrameContext{
-						PTS:      int(pkt.Pts()),
+						PTS:      int(inPkt.Pts()),
-						DTS:      int(pkt.Dts()),
+						DTS:      int(inPkt.Dts()),
-						Duration: audioDuration,
+						Duration: c.defineAudioDuration(s, inPkt),
-					})
+					}); err != nil {
+						c.onError(err, donut)
+						return
+					}
+				}
+				continue
+			}
+
+			// send the coded packet (compressed/encoded frame) to the decoder
+			if err := s.decCodecContext.SendPacket(inPkt); err != nil {
+				c.onError(err, donut)
+				return
+			}
+
+			for {
+				// receive the raw frame from the decoder
+				if err := s.decCodecContext.ReceiveFrame(s.decFrame); err != nil {
+					if errors.Is(err, astiav.ErrEof) || errors.Is(err, astiav.ErrEagain) {
+						break
+					}
+					c.onError(err, donut)
+					return
+				}
+				// send the raw frame to the encoder
+				if err := c.encodeFrame(s.decFrame, s, donut); err != nil {
+					c.onError(err, donut)
+					return
 				}
 			}
 		}
@@ -139,13 +193,7 @@ func (c *LibAVFFmpegStreamer) onError(err error, p *entities.DonutParameters) {
 	}
 }
 
-func (c *LibAVFFmpegStreamer) prepareInput(p *params, closer *astikit.Closer, donut *entities.DonutParameters) error {
+func (c *LibAVFFmpegStreamer) prepareInput(p *libAVParams, closer *astikit.Closer, donut *entities.DonutParameters) error {
-	// good for debugging
-	astiav.SetLogLevel(astiav.LogLevelDebug)
-	astiav.SetLogCallback(func(l astiav.LogLevel, fmt, msg, parent string) {
-		c.l.Infof("ffmpeg log: %s (level: %d)", strings.TrimSpace(msg), l)
-	})
-
 	if p.inputFormatContext = astiav.AllocFormatContext(); p.inputFormatContext == nil {
 		return errors.New("ffmpeg/libav: input format context is nil")
 	}
@@ -159,7 +207,6 @@ func (c *LibAVFFmpegStreamer) prepareInput(p *params, closer *astikit.Closer, do
 	if err := p.inputFormatContext.OpenInput(donut.StreamURL, inputFormat, inputOptions); err != nil {
 		return fmt.Errorf("ffmpeg/libav: opening input failed %w", err)
 	}
-
 	closer.Add(p.inputFormatContext.CloseInput)
 
 	if err := p.inputFormatContext.FindStreamInfo(nil); err != nil {
@@ -209,6 +256,181 @@ func (c *LibAVFFmpegStreamer) prepareInput(p *params, closer *astikit.Closer, do
 	return nil
 }
 
+func (c *LibAVFFmpegStreamer) prepareOutput(p *libAVParams, closer *astikit.Closer, donut *entities.DonutParameters) error {
+	for _, is := range p.inputFormatContext.Streams() {
+		s, ok := p.streams[is.Index()]
+		if !ok {
+			c.l.Infof("skipping stream index = %d", is.Index())
+			continue
+		}
+
+		isVideo := s.decCodecContext.MediaType() == astiav.MediaTypeVideo
+		isVideoBypass := donut.Recipe.Video.Action == entities.DonutBypass
+		if isVideo && isVideoBypass {
+			c.l.Infof("skipping video transcoding for %+v", s.inputStream)
+			continue
+		}
+
+		isAudio := s.decCodecContext.MediaType() == astiav.MediaTypeAudio
+		isAudioBypass := donut.Recipe.Audio.Action == entities.DonutBypass
+		if isAudio && isAudioBypass {
+			c.l.Infof("skipping audio transcoding for %+v", s.inputStream)
+			continue
+		}
+
+		var codecID astiav.CodecID
+		if isAudio {
+			audioCodecID, err := c.m.FromStreamCodecToLibAVCodecID(donut.Recipe.Audio.Codec)
+			if err != nil {
+				return err
+			}
+			codecID = audioCodecID
+		}
+		if isVideo {
+			videoCodecID, err := c.m.FromStreamCodecToLibAVCodecID(donut.Recipe.Video.Codec)
+			if err != nil {
+				return err
+			}
+			codecID = videoCodecID
+		}
+
+		if s.encCodec = astiav.FindEncoder(codecID); s.encCodec == nil {
+			// TODO: migrate error to entity
+			return fmt.Errorf("cannot find a libav encoder for %+v", codecID)
+		}
+
+		if s.encCodecContext = astiav.AllocCodecContext(s.encCodec); s.encCodecContext == nil {
+			// TODO: migrate error to entity
+			return errors.New("ffmpeg/libav: codec context is nil")
+		}
+		closer.Add(s.encCodecContext.Free)
+
+		if isAudio {
+			if v := s.encCodec.ChannelLayouts(); len(v) > 0 {
+				s.encCodecContext.SetChannelLayout(v[0])
+			} else {
+				s.encCodecContext.SetChannelLayout(s.decCodecContext.ChannelLayout())
+			}
+			s.encCodecContext.SetChannels(s.decCodecContext.Channels())
+			s.encCodecContext.SetSampleRate(s.decCodecContext.SampleRate())
+			if v := s.encCodec.SampleFormats(); len(v) > 0 {
+				s.encCodecContext.SetSampleFormat(v[0])
+			} else {
+				s.encCodecContext.SetSampleFormat(s.decCodecContext.SampleFormat())
+			}
+			s.encCodecContext.SetTimeBase(s.decCodecContext.TimeBase())
+
+			// supplying custom config
+			if len(donut.Recipe.Audio.CodecContextOptions) > 0 {
+				for _, opt := range donut.Recipe.Audio.CodecContextOptions {
+					opt(s.encCodecContext)
+				}
+			}
+		}
+
+		if isVideo {
+			if v := s.encCodec.PixelFormats(); len(v) > 0 {
+				s.encCodecContext.SetPixelFormat(v[0])
+			} else {
+				s.encCodecContext.SetPixelFormat(s.decCodecContext.PixelFormat())
+			}
+			s.encCodecContext.SetSampleAspectRatio(s.decCodecContext.SampleAspectRatio())
+			s.encCodecContext.SetTimeBase(s.decCodecContext.TimeBase())
+			s.encCodecContext.SetHeight(s.decCodecContext.Height())
+			s.encCodecContext.SetWidth(s.decCodecContext.Width())
+			// s.encCodecContext.SetFramerate(p.inputFormatContext.GuessFrameRate(s.inputStream, nil))
+			s.encCodecContext.SetFramerate(s.inputStream.AvgFrameRate())
+
+			// supplying custom config
+			if len(donut.Recipe.Audio.CodecContextOptions) > 0 {
+				for _, opt := range donut.Recipe.Audio.CodecContextOptions {
+					opt(s.encCodecContext)
+				}
+			}
+		}
+
+		if s.decCodecContext.Flags().Has(astiav.CodecContextFlagGlobalHeader) {
+			s.encCodecContext.SetFlags(s.encCodecContext.Flags().Add(astiav.CodecContextFlagGlobalHeader))
+		}
+
+		if err := s.encCodecContext.Open(s.encCodec, nil); err != nil {
+			return fmt.Errorf("opening encoder context failed: %w", err)
+		}
+
+		s.encPkt = astiav.AllocPacket()
+		closer.Add(s.encPkt.Free)
+
+		// // Update codec parameters
+		// if err = s.outputStream.CodecParameters().FromCodecContext(s.encCodecContext); err != nil {
+		// 	err = fmt.Errorf("main: updating codec parameters failed: %w", err)
+		// 	return
+		// }
+
+		// // Update stream
+		// s.outputStream.SetTimeBase(s.encCodecContext.TimeBase())
+	}
+	return nil
+}
+
+func (c *LibAVFFmpegStreamer) encodeFrame(f *astiav.Frame, s *streamContext, donut *entities.DonutParameters) (err error) {
+	// Reset picture type
+	f.SetPictureType(astiav.PictureTypeNone)
+
+	s.encPkt.Unref()
+
+	// Send frame
+	if err = s.encCodecContext.SendFrame(f); err != nil {
+		err = fmt.Errorf("main: sending frame failed: %w", err)
+		return
+	}
+
+	// Loop
+	for {
+		// Receive packet
+		if err = s.encCodecContext.ReceivePacket(s.encPkt); err != nil {
+			if errors.Is(err, astiav.ErrEof) || errors.Is(err, astiav.ErrEagain) {
+				err = nil
+				break
+			}
+			err = fmt.Errorf("main: receiving packet failed: %w", err)
+			return
+		}
+
+		// Update pkt
+		// 		s.encPkt.RescaleTs(s.encCodecContext.TimeBase(), s.outputStream.TimeBase())
+		s.encPkt.RescaleTs(s.encCodecContext.TimeBase(), s.decCodecContext.TimeBase())
+
+		isVideo := s.decCodecContext.MediaType() == astiav.MediaTypeVideo
+		if isVideo {
+			if donut.OnVideoFrame != nil {
+				if err := donut.OnVideoFrame(s.encPkt.Data(), entities.MediaFrameContext{
+					PTS:      int(s.encPkt.Pts()),
+					DTS:      int(s.encPkt.Dts()),
+					Duration: c.defineVideoDuration(s, s.encPkt),
+				}); err != nil {
+					return err
+				}
+			}
+		}
+
+		isAudio := s.decCodecContext.MediaType() == astiav.MediaTypeAudio
+		if isAudio {
+			if donut.OnAudioFrame != nil {
+				if err := donut.OnAudioFrame(s.encPkt.Data(), entities.MediaFrameContext{
+					PTS:      int(s.encPkt.Pts()),
+					DTS:      int(s.encPkt.Dts()),
+					Duration: c.defineAudioDuration(s, s.encPkt),
+				}); err != nil {
+					return err
+				}
+			}
+		}
+
+	}
+
+	return nil
+}
+
 func (c *LibAVFFmpegStreamer) defineInputFormat(streamFormat string) (*astiav.InputFormat, error) {
 	if streamFormat != "" {
 		inputFormat := astiav.FindInputFormat(streamFormat)
@@ -246,14 +468,19 @@ func (c *LibAVFFmpegStreamer) defineAudioDuration(s *streamContext, pkt *astiav.
 		// ref https://developer.apple.com/documentation/coreaudiotypes/audiostreambasicdescription/1423257-mframesperpacket
 
 		// TODO: properly handle wraparound / roll over
-		c.currentAudioFrameSize = float64(pkt.Dts()) - c.lastAudioFrameDTS
+		// or explore av frame_size https://ffmpeg.org/doxygen/trunk/structAVCodecContext.html#aec57f0d859a6df8b479cd93ca3a44a33
-		if c.currentAudioFrameSize < 0 {
+		// and libAV pts roll over
-			c.currentAudioFrameSize = c.lastAudioFrameDTS*2 - c.lastAudioFrameDTS
+		if float64(pkt.Dts())-c.lastAudioFrameDTS > 0 {
+			c.currentAudioFrameSize = float64(pkt.Dts()) - c.lastAudioFrameDTS
 		}
 
 		c.lastAudioFrameDTS = float64(pkt.Dts())
 		sampleRate := float64(s.inputStream.CodecParameters().SampleRate())
 		audioDuration = time.Duration((c.currentAudioFrameSize / sampleRate) * float64(time.Second))
+		c.l.Infow("audio duration",
+			"framesize", s.inputStream.CodecParameters().FrameSize(),
+			"audioDuration", audioDuration,
+		)
 	}
 	return audioDuration
 }
@@ -270,6 +497,42 @@ func (c *LibAVFFmpegStreamer) defineVideoDuration(s *streamContext, pkt *astiav.
 
 		// we're assuming fixed video frame rate
 		videoDuration = time.Duration((float64(1) / float64(s.inputStream.AvgFrameRate().Num())) * float64(time.Second))
+		c.l.Infow("video duration",
+			"framesize", s.inputStream.CodecParameters().FrameSize(),
+			"videoDuration", videoDuration,
+		)
 	}
 	return videoDuration
 }
+
+// TODO: move this either to a mapper or make a PR for astiav
+func (*LibAVFFmpegStreamer) libAVLogToString(l astiav.LogLevel) string {
+	const _Ciconst_AV_LOG_DEBUG = 0x30
+	const _Ciconst_AV_LOG_ERROR = 0x10
+	const _Ciconst_AV_LOG_FATAL = 0x8
+	const _Ciconst_AV_LOG_INFO = 0x20
+	const _Ciconst_AV_LOG_PANIC = 0x0
+	const _Ciconst_AV_LOG_QUIET = -0x8
+	const _Ciconst_AV_LOG_VERBOSE = 0x28
+	const _Ciconst_AV_LOG_WARNING = 0x18
+	switch l {
+	case _Ciconst_AV_LOG_WARNING:
+		return "WARN"
+	case _Ciconst_AV_LOG_VERBOSE:
+		return "VERBOSE"
+	case _Ciconst_AV_LOG_QUIET:
+		return "QUIET"
+	case _Ciconst_AV_LOG_PANIC:
+		return "PANIC"
+	case _Ciconst_AV_LOG_INFO:
+		return "INFO"
+	case _Ciconst_AV_LOG_FATAL:
+		return "FATAL"
+	case _Ciconst_AV_LOG_DEBUG:
+		return "DEBUG"
+	case _Ciconst_AV_LOG_ERROR:
+		return "ERROR"
+	default:
+		return "UNKNOWN LEVEL"
+	}
+}

internal/controllers/webrtc_controller.go

@@ -74,8 +74,8 @@ func (c *WebRTCController) CreatePeerConnection(cancel context.CancelFunc) (*web
 	return peerConnection, nil
 }
 
-func (c *WebRTCController) CreateTrack(peer *webrtc.PeerConnection, track entities.Stream, id string, streamId string) (*webrtc.TrackLocalStaticSample, error) {
+func (c *WebRTCController) CreateTrack(peer *webrtc.PeerConnection, codec entities.Codec, id string, streamId string) (*webrtc.TrackLocalStaticSample, error) {
-	codecCapability := c.m.FromTrackToRTPCodecCapability(track)
+	codecCapability := c.m.FromTrackToRTPCodecCapability(codec)
 	webRTCtrack, err := webrtc.NewTrackLocalStaticSample(codecCapability, id, streamId)
 	if err != nil {
 		return nil, err

internal/entities/entities.go

@@ -5,6 +5,7 @@ import (
 	"fmt"
 	"time"
 
+	"github.com/asticode/go-astiav"
 	"github.com/pion/webrtc/v3"
 )
 
@@ -131,8 +132,7 @@ type DonutParameters struct {
 	StreamFormat string // ie: flv, mpegts
 	StreamURL    string // ie: srt://host:9080, rtmp://host:4991
 
-	TranscodeVideoCodec Codec // ie: vp8
+	Recipe *DonutTransformRecipe
-	TranscodeAudioCodec Codec // ie: opus
 
 	OnClose      func()
 	OnError      func(err error)
@@ -141,6 +141,45 @@ type DonutParameters struct {
 	OnAudioFrame func(data []byte, c MediaFrameContext) error
 }
 
+type DonutMediaTaskAction string
+
+var DonutTranscode DonutMediaTaskAction = "transcode"
+var DonutBypass DonutMediaTaskAction = "bypass"
+
+// TODO: split entities per domain or files avoiding cluttered names.
+
+// DonutMediaTask is a transformation template to apply over a media.
+type DonutMediaTask struct {
+	// Action the action that needs to be performed
+	Action DonutMediaTaskAction
+	// Codec is the main codec, it might be used depending on the action.
+	Codec Codec
+	// CodecContextOptions is a list of options to be applied on codec context.
+	// If no value is provided ffmpeg will use defaults.
+	// For instance, if one does not provide bit rate, it'll fallback to 64000 bps (opus)
+	CodecContextOptions []LibAVOptionsCodecContext
+}
+
+// DonutTransformRecipe is a recipe to run on medias
+type DonutTransformRecipe struct {
+	Video DonutMediaTask
+	Audio DonutMediaTask
+}
+
+// LibAVOptionsCodecContext is option pattern to change codec context
+type LibAVOptionsCodecContext func(c *astiav.CodecContext)
+
+func SetSampleRate(sampleRate int) LibAVOptionsCodecContext {
+	return func(c *astiav.CodecContext) {
+		c.SetSampleRate(sampleRate)
+	}
+}
+
+// TODO: implement proper matching
+// DonutTransformRecipe
+//  AudioTask: {Action: Transcode, From: AAC, To: Opus}
+//  VideoTask: {Action: Bypass, From: H264, To: H264}
+
 type Config struct {
 	HTTPPort       int32  `required:"true" default:"8080"`
 	HTTPHost       string `required:"true" default:"0.0.0.0"`

internal/mapper/mapper.go

@@ -1,6 +1,7 @@
 package mapper
 
 import (
+	"fmt"
 	"strings"
 
 	"github.com/asticode/go-astiav"
@@ -19,15 +20,18 @@ func NewMapper(l *zap.SugaredLogger) *Mapper {
 	return &Mapper{l: l}
 }
 
-func (m *Mapper) FromTrackToRTPCodecCapability(track entities.Stream) webrtc.RTPCodecCapability {
+func (m *Mapper) FromTrackToRTPCodecCapability(codec entities.Codec) webrtc.RTPCodecCapability {
+	// TODO: enrich codec capability, check if it's necessary
 	response := webrtc.RTPCodecCapability{}
 
-	if track.Codec == entities.H264 {
+	if codec == entities.H264 {
 		response.MimeType = webrtc.MimeTypeH264
-	} else if track.Codec == entities.H265 {
+	} else if codec == entities.H265 {
 		response.MimeType = webrtc.MimeTypeH265
+	} else if codec == entities.Opus {
+		response.MimeType = webrtc.MimeTypeOpus
 	} else {
-		m.l.Info("[[[[TODO: mapper not implemented]]]] for ", track)
+		m.l.Info("[[[[TODO: mapper not implemented]]]] for ", codec)
 	}
 
 	return response
@@ -203,3 +207,16 @@ func (m *Mapper) FromLibAVStreamToEntityStream(libavStream *astiav.Stream) entit
 
 	return st
 }
+
+func (m *Mapper) FromStreamCodecToLibAVCodecID(codec entities.Codec) (astiav.CodecID, error) {
+	if codec == entities.H264 {
+		return astiav.CodecIDH264, nil
+	} else if codec == entities.H265 {
+		return astiav.CodecIDHevc, nil
+	} else if codec == entities.Opus {
+		return astiav.CodecIDOpus, nil
+	}
+
+	// TODO: port error to entities
+	return astiav.CodecIDH264, fmt.Errorf("cannot find a libav codec id for donut codec id %+v", codec)
+}

internal/web/handlers/signaling.go

@@ -70,41 +70,19 @@ func (h *SignalingHandler) ServeHTTP(w http.ResponseWriter, r *http.Request) err
 		return err
 	}
 
-	// TODO: introduce a mode to deal with transcoding recipes
+	donutRecipe := donutEngine.RecipeFor(serverStreamInfo, clientStreamInfo)
-	// selects proper media that client and server has adverted.
+	if donutRecipe == nil {
-	// donutEngine preferable vp8, ogg???
-	// From: [] To: [] or Transcode:[], Bypass: []
-	// libav_streamer.go, libav_streamer_format.go, libav_streamer_codec.go...
-	// reads from Server (input) and generates h264 raw, and ogg and send it with timing attributes
-	compatibleStreams, ok := donutEngine.CompatibleStreamsFor(serverStreamInfo, clientStreamInfo)
-	if !ok {
-		h.l.Info("we must transcode")
-	}
-
-	if len(compatibleStreams) == 0 {
 		return entities.ErrMissingCompatibleStreams
 	}
 
 	var videoTrack *webrtc.TrackLocalStaticSample
-	// var audioTrack *webrtc.TrackLocalStaticSample
+	videoTrack, err = h.webRTCController.CreateTrack(peer, donutRecipe.Video.Codec, string(entities.VideoType), params.SRTStreamID)
-
+	if err != nil {
-	for _, st := range compatibleStreams {
+		return err
-		// TODO: make the mapping less dependent on type
-		if st.Type == entities.VideoType {
-			videoTrack, err = h.webRTCController.CreateTrack(
-				peer,
-				st,
-				string(st.Type), // "video" or "audio"
-				params.SRTStreamID,
-			)
-			if err != nil {
-				return err
-			}
-
-		}
-		// if st.Type == entities.AudioType {
 	}
 
+	// var audioTrack *webrtc.TrackLocalStaticSample
+
 	metadataSender, err := h.webRTCController.CreateDataChannel(peer, entities.MetadataChannelID)
 	if err != nil {
 		return err
@@ -123,6 +101,8 @@ func (h *SignalingHandler) ServeHTTP(w http.ResponseWriter, r *http.Request) err
 		Cancel: cancel,
 		Ctx:    ctx,
 
+		Recipe: donutRecipe,
+
 		// TODO: add an UI element for the sub-type (format) when input is srt://
 		// We're assuming that SRT is carrying mpegts.
 		StreamFormat: "mpegts",