1
use crate::{Error, Result};
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use std::time::Instant;
3
use str0m::{
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    Rtc,
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    format::Codec,
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    media::{Frequency, Mid},
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    rtp::{RtpWrite, SeqNo},
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};
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10
/// One Opus packet per RTP packet, with sequence state tied to the SSRC lifetime.
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#[derive(Debug)]
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pub(crate) struct Audio {
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    mid: Mid,
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    next_seq: SeqNo,
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}
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17
impl Audio {
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    pub(crate) fn new(mid: Mid) -> Self {
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        Self {
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            mid,
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            // Match str0m's random initial sequence and start with a zero ROC.
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            next_seq: SeqNo::default(),
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        }
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    }
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26
    pub(crate) fn write(&mut self, rtc: &mut Rtc, pts_ms: u32, bytes: &[u8]) -> Result<()> {
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        if bytes.is_empty() || bytes.len() > radio_core::music::MAX_OPUS_BYTES {
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            return Err(Error::new("invalid Opus packet size"));
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        }
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        // Direct writes retain their payload, so never queue audio without a
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        // live ICE/DTLS/SRTP transport to consume it.
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        if !rtc.is_alive() || !rtc.is_connected() {
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            return Err(Error::new("audio transport unavailable"));
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        }
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        let media = rtc
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            .media(self.mid)
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            .filter(|media| !media.disabled() && media.direction().is_sending())
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            .ok_or(Error::new("audio media unavailable"))?;
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        let pt = rtc
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            .codec_config()
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            .params()
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            .iter()
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            .find(|params| {
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                let spec = params.spec();
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                media.remote_pts().contains(&params.pt())
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                    && spec.codec == Codec::Opus
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                    && spec.clock_rate == Frequency::FORTY_EIGHT_KHZ
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                    && spec.channels == Some(2)
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            })
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            .ok_or(Error::new("Opus was not negotiated"))?
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            .pt();
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        let mut direct = rtc.direct_api();
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        let stream = direct
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            .stream_tx_by_mid(self.mid, None)
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            .ok_or(Error::new("audio stream unavailable"))?;
56
 
57
        // Pauses send encoded silence, without DTX/talkspurt markers. str0m
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        // supplies negotiated header extensions, pacing and SRTP encryption.
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        // The owned Arc payload outlives the caller's reusable music buffer.
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        stream.write_rtp(RtpWrite::new(
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            pt,
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            self.next_seq.inc(),
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            pts_ms.wrapping_mul(48),
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            Instant::now(),
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            bytes,
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        ));
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        Ok(())
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    }
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}
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71
#[cfg(test)]
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mod tests;