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| // called when a MULAW audio RTP packet arrives | ||
| if g.AudioLPCMMedia != nil && g.AudioLPCMForma != nil { | ||
| g.Client.OnPacketRTP(g.AudioLPCMMedia, g.AudioLPCMForma, func(rtppkt *rtp.Packet) { | ||
| pts, ok := g.Client.PacketPTS(g.AudioLPCMMedia, rtppkt) |
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Pull request overview
Adds support for recording non-AAC RTSP audio (PCMU and LPCM) by transcoding to AAC/ADTS, and improves MP4 audio timing/metadata so generated MP4s carry correct AAC sample-rate/channel info.
Changes:
- Introduce an ffmpeg-backed PCMU/LPCM → AAC (ADTS) transcoder and a recording audio writer that feeds MP4 muxing.
- Update capture pipeline to use the new audio writer and to roll over recordings on keyframes without dropping frames.
- Update MP4 muxer to derive AAC frame durations and audio timescale from ADTS headers, and set an AAC-LC descriptor with detected sample-rate/channels.
Reviewed changes
Copilot reviewed 5 out of 5 changed files in this pull request and generated 4 comments.
Show a summary per file
| File | Description |
|---|---|
| machinery/src/video/mp4.go | Switch audio duration math to AAC sample counts, derive audio timescale from ADTS, and set an AAC-LC descriptor with parsed parameters. |
| machinery/src/packets/stream.go | Add BitDepth to audio stream metadata for PCM/LPCM handling. |
| machinery/src/capture/pcmu_to_aac_transcoder.go | New ffmpeg-based transcoder + ADTS framing utilities + recordingAudioWriter integration with MP4. |
| machinery/src/capture/main.go | Route audio recording through recordingAudioWriter, pass bit depth for LPCM, and improve rollover logic. |
| machinery/src/capture/gortsplib.go | Add LPCM stream discovery/decoding and include default PCMU stream parameters (rate/channels) in stream metadata. |
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| pts, ok := g.Client.PacketPTS(g.AudioLPCMMedia, rtppkt) | ||
| pts2, ok := g.Client.PacketPTS2(g.AudioLPCMMedia, rtppkt) | ||
| if !ok { | ||
| log.Log.Debug("capture.golibrtsp.Start(): " + "unable to get PTS") | ||
| return | ||
| } |
| func (g *Golibrtsp) Start(ctx context.Context, streamType string, queue *packets.Queue, configuration *models.Configuration, communication *models.Communication) (err error) { | ||
| log.Log.Debug("capture.golibrtsp.Start(): started") | ||
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| // called when a MULAW audio RTP packet arrives |
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| // Set the audio descriptor to match the AAC-LC stream actually produced by ffmpeg/camera. | ||
| err := setAACLCDescriptor(init.Moov.Traks[1], audioSampleRate, audioChannels) | ||
| if err != nil { | ||
| } |
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| "-f", inputFormat, | ||
| "-ar", intToString(sampleRate), | ||
| "-ac", intToString(channels), | ||
| "-i", "pipe:0", | ||
| "-vn", | ||
| "-ac", "1", | ||
| "-ar", intToString(sampleRate), |
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Description
Enable PCMU and LPCM Audio Support via Transcoding to AAC
Motivation
IP cameras often stream audio in PCMU (G.711 mu-law) or LPCM formats over RTSP, but our recording pipeline previously skipped these, limiting recordings to video-only or direct AAC streams. This change introduces real-time transcoding of PCMU and LPCM to AAC using FFmpeg, ensuring audio is captured and muxed into MP4 files regardless of the source format. It also adds explicit LPCM detection in RTSP sessions and allows configuration overrides for sample rate/channels in LPCM streams to handle mismatches.
Key Changes
gortsplib.go): AddedFindLPCMto discover LPCM media/formats, setup decoding, and integrate into packet queues with proper metadata (sample rate, channels, bit depth). Extended PCMU streams with default metadata.main.go): IntroducedrecordingAudioWriterto abstract audio track creation and sample writing. For non-AAC inputs, it transcodes to AAC before muxing. Refactored continuous and motion-detection recording to use packet timestamps for precise rollovers, avoiding frame drops during max-duration limits or post-recording expiration. Added LPCM-specific config validation with warnings.pcmu_to_aac_transcoder.go): New FFmpeg-based transcoder for PCMU/LPCM → AAC (ADTS output). Supports input validation, non-blocking writes, timed flushes, and error handling. Checks FFmpeg availability at runtime; falls back to skipping audio if unavailable.Improvements
This enhances recording completeness and robustness, reducing lost audio in diverse camera setups while maintaining low overhead. Tested with sample RTSP streams; requires FFmpeg for full functionality.