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Fix UDP stereo streaming buffer size
Add SRT streaming output
…yback Fix SRT streaming by chunking
…ction-available--messages Improve SRT logging and add usage notes
…mats-and-logs Add WAV support for SRT output
…and-address-ffplay-warnings Improve SRT documentation
…tput Add min_rx_seconds and post_write_script options
Author
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Been running this server for ~2 weeks now, appears stable. I'm using it here: https://github.com/yegors/co-atc |
- Change audio format from 32-bit float to 16-bit signed PCM across all formats (pcm, mp3, wav) - Add configurable `mode` setting with two options: - `live` (default): Standard SRT with TSBPD for broad client compatibility (~120ms latency) - `raw`: Minimal latency mode with TSBPD disabled (ffplay/ffmpeg only) - Improve error handling: drop data on buffer full (SRT_EASYNCSND) instead of disconnecting client - Update documentation with separate ffplay commands for different formats and new mode configuration This provides users flexibility to choose between compatibility (live mode) and minimal latency (raw mode) based on their specific use case and client applications.
…in srt_stream_init
Collaborator
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@yegors thanks for the PR and sorry for the delay. It looks like there are a few things in here that could be pulled out into independent smaller PRs and get in sooner (addition of |
Author
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Yeah, I mostly just kept committing to my branch for changes I needed along the way for my project to make it output clean streams I can feed directly into TTS with no encoding. Feel free to copy whatever you want. |
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This update introduces a new SRT output feature that allows RTLSDR‑Airband to stream audio using the Secure Reliable Transport protocol.
ffplay -fflags nobuffer -flags low_delay srt://<host>:<port>Configuration example
I'm not a C++ dev, and this code has been entirely AI written so definitely needs some human eyes on it. But it does work.
Resolves #392