Voice-to-Code Streaming Pipelines & Agent Orchestration

Streaming Audio to Tokens Over WebSockets

In PRAXIS, audio frames are streamed over WebSockets using binary framing, converted into mel-spectrogram features via SIMD AVX2 kernels, and dispatched to transcription and code-generation agent runtimes. Backpressure is managed through bounded MPSC channels to prevent runaway memory usage during network jitter:

use tokio::sync::mpsc;

pub struct AudioAgentPipeline {
    audio_tx: mpsc::Sender>,
}

impl AudioAgentPipeline {
    pub fn new(capacity: usize) -> (Self, mpsc::Receiver>) {
        let (audio_tx, rx) = mpsc::channel(capacity);
        (Self { audio_tx }, rx)
    }

    pub async fn ingest_chunk(&self, chunk: Vec) -> Result<(), &'static str> {
        match self.audio_tx.try_send(chunk) {
            Ok(()) => Ok(()),
            Err(mpsc::error::TrySendError::Full(_)) => {
                // Drop frame or apply graceful backpressure
                Err("Backpressure: Audio buffer full, dropping frame to maintain real-time sync")
            }
            Err(mpsc::error::TrySendError::Closed(_)) => Err("Pipeline closed"),
        }
    }
}