feat(audio): end-to-end 5.1/7.1 surround across the native path + all clients
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Adds negotiated 5.1/7.1 surround to the punktfunk/1 protocol and every client
(previously stereo-only):

- core: new shared `audio` layout table (LAYOUT_51/71 + identity multistream
  mapping, canonical wire order FL FR FC LFE RL RR SL SR); Hello/Welcome
  `audio_channels` negotiation via the trailing-byte back-compat pattern (old
  peers fall back to stereo); C-ABI `punktfunk_connect_ex6`,
  `punktfunk_connection_audio_channels`, and in-core multistream decode
  `punktfunk_connection_next_audio_pcm` for embedders without a multistream
  Opus decoder. Real-libopus channel-identity round-trip test.
- host: native audio thread captures + Opus-(multi)stream-encodes at the
  negotiated count (with a cross-session cached-capturer channel-mismatch fix);
  GameStream surround unified onto the safe `opus::MSEncoder`, dropping
  `audiopus_sys` (~4 unsafe blocks) and un-gating Windows GameStream surround;
  WASAPI loopback capture relaxed to 2/6/8 with the correct dwChannelMask.
- clients: Linux (PipeWire), Windows (WASAPI), Android (AAudio) decode via
  `opus::MSDecoder` + render multichannel; Apple decodes in-core to PCM →
  AVAudioEngine with an explicit wire-order channel layout; each gains a
  Stereo/5.1/7.1 setting. `punktfunk-probe --audio-channels N` is the headless
  validator.

Verified on Linux: core/host/linux/probe test suites + the Android Rust
(cargo-ndk) build, clippy -D warnings, and rustfmt all green. Windows/Apple
builds, all on-glass checks, and the live native loopback are pending (CI / a
free box).

Also lands the concurrent in-tree HEVC 4:4:4 host work (PUNKTFUNK_444): it
shares the same touched files (quic.rs, punktfunk1.rs, encode/*, ...) and so
cannot be committed separately from the surround changes.

Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
This commit is contained in:
2026-06-28 21:11:05 +00:00
parent 6383e5f4fd
commit 75627c8afe
51 changed files with 2254 additions and 494 deletions
+71 -267
View File
@@ -41,8 +41,6 @@ type Aes128CbcEnc = cbc::Encryptor<aes::Aes128>;
/// `RTP_PAYLOAD_TYPE_FEC 127`).
const AUDIO_PACKET_TYPE: u8 = 97;
const AUDIO_FEC_PACKET_TYPE: u8 = 127;
/// Stereo Opus bitrate (unchanged from the live-validated stereo path).
const OPUS_BITRATE: i32 = 128_000;
/// Audio FEC geometry (moonlight-common-c `RtpAudioQueue.h`: `RTPA_DATA_SHARDS 4`,
/// `RTPA_FEC_SHARDS 2`). Blocks are aligned: the client synthesizes the block base as
@@ -82,67 +80,20 @@ impl Default for AudioParams {
}
}
/// One Opus (multi)stream layout. Channel order is the GameStream/Moonlight order
/// FL FR FC LFE RL RR [SL SR]; `mapping` is the libopus multistream mapping we *encode*
/// with — identical to Sunshine's `audio.cpp stream_configs` (verified verbatim 2026-06-10):
/// identity mapping, so normal quality couples (FL,FR) and (FC,LFE) [+ (RL,RR) on 7.1] with
/// the remaining channels as mono streams; high quality is one mono stream per channel.
/// Bitrates are Sunshine's per-config values (stereo keeps punktfunk's existing 128 kbps).
pub struct OpusLayout {
pub channels: u8,
pub streams: u8,
pub coupled: u8,
pub mapping: &'static [u8],
pub bitrate: i32,
}
pub const LAYOUT_STEREO: OpusLayout = OpusLayout {
channels: 2,
streams: 1,
coupled: 1,
mapping: &[0, 1],
bitrate: OPUS_BITRATE,
};
pub const LAYOUT_51: OpusLayout = OpusLayout {
channels: 6,
streams: 4,
coupled: 2,
mapping: &[0, 1, 2, 3, 4, 5],
bitrate: 256_000,
};
pub const LAYOUT_51_HQ: OpusLayout = OpusLayout {
channels: 6,
streams: 6,
coupled: 0,
mapping: &[0, 1, 2, 3, 4, 5],
bitrate: 1_536_000,
};
pub const LAYOUT_71: OpusLayout = OpusLayout {
channels: 8,
streams: 5,
coupled: 3,
mapping: &[0, 1, 2, 3, 4, 5, 6, 7],
bitrate: 450_000,
};
pub const LAYOUT_71_HQ: OpusLayout = OpusLayout {
channels: 8,
streams: 8,
coupled: 0,
mapping: &[0, 1, 2, 3, 4, 5, 6, 7],
bitrate: 2_048_000,
// The Opus surround layout table (channel order FL FR FC LFE RL RR [SL SR], identity mapping,
// Sunshine's per-config bitrates) now lives in `punktfunk_core::audio`, shared with the native
// `punktfunk/1` path and every client decoder. Re-export the pieces the GameStream module + its
// RTSP SDP (`rtsp.rs`) reference; the GFE-specific `surround_params` SDP rotation stays below.
pub use punktfunk_core::audio::{
OpusLayout, LAYOUT_51, LAYOUT_51_HQ, LAYOUT_71, LAYOUT_71_HQ, LAYOUT_STEREO,
};
/// Pick the encoder layout for the negotiated session parameters. Unknown channel counts
/// fall back to stereo (the client can only request 2/6/8 — `AUDIO_CONFIGURATION_*` in
/// Pick the encoder layout for the negotiated session parameters. Thin wrapper over the shared
/// [`punktfunk_core::audio::layout_for`] keyed on this module's [`AudioParams`] (unknown channel
/// counts fall back to stereo; the client can only request 2/6/8 — `AUDIO_CONFIGURATION_*` in
/// Limelight.h).
pub fn layout_for(params: &AudioParams) -> &'static OpusLayout {
match (params.channels, params.high_quality) {
(6, false) => &LAYOUT_51,
(6, true) => &LAYOUT_51_HQ,
(8, false) => &LAYOUT_71,
(8, true) => &LAYOUT_71_HQ,
_ => &LAYOUT_STEREO,
}
punktfunk_core::audio::layout_for(params.channels, params.high_quality)
}
/// The `a=fmtp:97 surround-params=` digit string for a layout: channelCount, streams,
@@ -345,21 +296,21 @@ fn run(
}
/// Opus encoder for one session: the plain stereo encoder (the live-validated path, byte
/// identical) or a libopus multistream encoder for 5.1/7.1.
/// identical) or the safe `opus::MSEncoder` multistream encoder for 5.1/7.1. Both are
/// cross-platform (Linux + Windows) — surround no longer needs `audiopus_sys`.
#[cfg(any(target_os = "linux", target_os = "windows"))]
enum SessionEncoder {
Stereo(opus::Encoder),
// Surround needs the libopus *multistream* encoder via `audiopus_sys` (Linux-only dep).
#[cfg(target_os = "linux")]
Surround(MsEncoder),
Surround(opus::MSEncoder),
}
#[cfg(any(target_os = "linux", target_os = "windows"))]
impl SessionEncoder {
fn new(layout: &'static OpusLayout) -> Result<SessionEncoder> {
// RESTRICTED_LOWDELAY (`opus::Application::LowDelay`) + hard CBR, matching Sunshine — CBR
// keeps the Opus packet size constant, which the GameStream audio FEC (equal-length shards)
// relies on, and the client asserts a constant per-stream TOC.
if layout.channels == 2 {
// RESTRICTED_LOWDELAY + CBR, matching Sunshine — CBR keeps the Opus TOC byte
// constant, which the client asserts per stream.
let mut enc = opus::Encoder::new(
SAMPLE_RATE,
opus::Channels::Stereo,
@@ -370,138 +321,32 @@ impl SessionEncoder {
enc.set_vbr(false).ok();
Ok(SessionEncoder::Stereo(enc))
} else {
#[cfg(target_os = "linux")]
{
Ok(SessionEncoder::Surround(MsEncoder::new(layout)?))
}
#[cfg(not(target_os = "linux"))]
{
anyhow::bail!(
"surround audio ({} ch) needs the libopus multistream encoder (Linux only) — \
use a stereo session",
layout.channels
)
}
let mut enc = opus::MSEncoder::new(
SAMPLE_RATE,
layout.streams,
layout.coupled,
layout.mapping,
opus::Application::LowDelay,
)
.map_err(|e| anyhow::anyhow!("create Opus multistream encoder: {e}"))?;
enc.set_bitrate(opus::Bitrate::Bits(layout.bitrate)).ok();
enc.set_vbr(false).ok();
Ok(SessionEncoder::Surround(enc))
}
}
/// Encode one interleaved frame (`samples_per_channel * channels` f32s) into `out`,
/// returning the packet length.
fn encode_float(
&mut self,
frame: &[f32],
samples_per_channel: usize,
out: &mut [u8],
) -> Result<usize> {
// `samples_per_channel` only feeds the multistream (surround) encoder; stereo infers it.
#[cfg(not(target_os = "linux"))]
let _ = samples_per_channel;
/// Encode one interleaved frame into `out`, returning the packet length. Both encoders infer
/// the per-channel sample count from `frame.len()` and their channel count.
fn encode_float(&mut self, frame: &[f32], out: &mut [u8]) -> Result<usize> {
match self {
SessionEncoder::Stereo(enc) => enc.encode_float(frame, out).context("opus encode"),
#[cfg(target_os = "linux")]
SessionEncoder::Surround(enc) => enc.encode_float(frame, samples_per_channel, out),
SessionEncoder::Surround(enc) => enc
.encode_float(frame, out)
.context("opus multistream encode"),
}
}
}
/// RAII wrapper for `OpusMSEncoder` (the safe `opus` crate is stereo-only; the multistream
/// API comes from `audiopus_sys`, the same libopus the crate already links). Configured like
/// the stereo path: RESTRICTED_LOWDELAY, hard CBR, per-layout bitrate.
#[cfg(target_os = "linux")]
struct MsEncoder {
st: std::ptr::NonNull<audiopus_sys::OpusMSEncoder>,
}
// SAFETY: `MsEncoder` owns a unique `OpusMSEncoder` via `NonNull` (it is neither `Clone` nor
// `Sync`, so the pointer is never aliased). libopus's multistream encoder state is a self-contained
// heap allocation with no thread-local or thread-affine state, so moving ownership to another thread
// is sound; every method takes `&mut self`, keeping access single-threaded at any instant.
#[cfg(target_os = "linux")]
unsafe impl Send for MsEncoder {}
#[cfg(target_os = "linux")]
impl MsEncoder {
fn new(layout: &OpusLayout) -> Result<MsEncoder> {
use std::os::raw::c_int;
let mut err: c_int = 0;
// SAFETY: every scalar arg is a valid libopus input (sample rate, channel/stream/coupled
// counts, the RESTRICTED_LOWDELAY application constant). `layout.mapping.as_ptr()` addresses
// a 'static slice of exactly `layout.channels` bytes (every `OpusLayout` constant upholds
// that), which is the element count `opus_multistream_encoder_create` reads through it, and
// `&mut err` is a live local the call writes its status into. libopus copies the mapping into
// its own allocation, so the pointer need only be valid for the call; the returned pointer is
// null/`OPUS_OK`-checked below before any use.
let st = unsafe {
audiopus_sys::opus_multistream_encoder_create(
SAMPLE_RATE as i32,
layout.channels as c_int,
layout.streams as c_int,
layout.coupled as c_int,
layout.mapping.as_ptr(),
audiopus_sys::OPUS_APPLICATION_RESTRICTED_LOWDELAY,
&mut err,
)
};
let st = std::ptr::NonNull::new(st)
.filter(|_| err == audiopus_sys::OPUS_OK)
.ok_or_else(|| anyhow::anyhow!("opus_multistream_encoder_create failed ({err})"))?;
// SAFETY: `st` is the non-null encoder `opus_multistream_encoder_create` just returned, owned
// exclusively here. Each `opus_multistream_encoder_ctl` call passes a valid request constant
// with the single by-value `c_int` argument that request's variadic ABI expects
// (`OPUS_SET_BITRATE_REQUEST` → bitrate, `OPUS_SET_VBR_REQUEST` → 0). No pointer escapes the
// call and the encoder outlives it.
unsafe {
audiopus_sys::opus_multistream_encoder_ctl(
st.as_ptr(),
audiopus_sys::OPUS_SET_BITRATE_REQUEST,
layout.bitrate as c_int,
);
audiopus_sys::opus_multistream_encoder_ctl(
st.as_ptr(),
audiopus_sys::OPUS_SET_VBR_REQUEST,
0 as c_int, // hard CBR (constant packet size — also what audio FEC relies on)
);
}
Ok(MsEncoder { st })
}
fn encode_float(
&mut self,
frame: &[f32],
samples_per_channel: usize,
out: &mut [u8],
) -> Result<usize> {
// SAFETY: `self.st` is the live encoder from `new`. libopus reads `samples_per_channel *
// channels` f32s through `frame.as_ptr()`; every caller passes a `frame` of exactly that
// length together with the matching `samples_per_channel` (`audio_body`'s `frame_len =
// samples_per_channel * layout.channels`; the round-trip tests size identically), so the read
// stays in bounds. `out.as_mut_ptr()` is written for at most `out.len()` bytes, which is
// passed as the capacity bound. Both buffers are live locals outliving this synchronous call;
// the return value is range-checked before being used as a length.
let n = unsafe {
audiopus_sys::opus_multistream_encode_float(
self.st.as_ptr(),
frame.as_ptr(),
samples_per_channel as std::os::raw::c_int,
out.as_mut_ptr(),
out.len() as i32,
)
};
anyhow::ensure!(n > 0, "opus_multistream_encode_float failed ({n})");
Ok(n as usize)
}
}
#[cfg(target_os = "linux")]
impl Drop for MsEncoder {
fn drop(&mut self) {
// SAFETY: `self.st` is the encoder `opus_multistream_encoder_create` returned; this
// `MsEncoder` owns it uniquely and `drop` runs exactly once, so the destroy frees it once
// with no subsequent use.
unsafe { audiopus_sys::opus_multistream_encoder_destroy(self.st.as_ptr()) }
}
}
#[cfg(any(target_os = "linux", target_os = "windows"))]
fn audio_body(
cap: &mut dyn AudioCapturer,
@@ -565,7 +410,7 @@ fn audio_body(
*s = (*s * gain).clamp(-1.0, 1.0);
}
}
let n = enc.encode_float(&frame, samples_per_channel, &mut out)?;
let n = enc.encode_float(&frame, &mut out)?;
// AES-128-CBC the Opus payload (RTP header stays plaintext). Per-packet IV =
// BE32(rikeyid + seq) in [0..4], zero elsewhere; PKCS7 padding.
let iv_seq = (rikeyid as u32).wrapping_add(seq as u32);
@@ -775,41 +620,33 @@ mod tests {
/// Real-codec proof of the 5.1 mapping math: encode with our encoder layout, decode with
/// the mapping a stock Moonlight client derives from our advertised surround-params
/// (parse → GFE swap), and verify a tone fed into each input channel comes out on the
/// same output channel.
#[cfg(target_os = "linux")]
/// same output channel. Cross-platform via the safe `opus` crate — this also guards the
/// (now un-gated) Windows GameStream surround build.
#[test]
fn multistream_51_roundtrip_channel_identity() {
let layout = &LAYOUT_51;
let samples = 240; // 5 ms
let ch = layout.channels as usize;
// Client-side decoder mapping derived exactly as moonlight-common-c does.
// Client-side decoder mapping derived exactly as moonlight-common-c does (GFE swap).
let s = surround_params(layout, false);
let digits: Vec<u8> = s.bytes().map(|b| b - b'0').collect();
let client_mapping = client_swap(&digits[3..]);
let mut err = 0i32;
// SAFETY: scalar args are valid libopus inputs. `client_mapping.as_ptr()` addresses a
// `Vec<u8>` of exactly `ch` entries (derived from the advertised surround-params), which is
// the element count the decoder reads through it, and `&mut err` is a live local the call
// writes. The returned pointer is `OPUS_OK`/non-null-checked immediately below before use.
let dec = unsafe {
audiopus_sys::opus_multistream_decoder_create(
SAMPLE_RATE as i32,
ch as i32,
layout.streams as i32,
layout.coupled as i32,
client_mapping.as_ptr(),
&mut err,
)
};
assert_eq!(err, audiopus_sys::OPUS_OK);
assert!(!dec.is_null());
let mut dec =
opus::MSDecoder::new(SAMPLE_RATE, layout.streams, layout.coupled, &client_mapping)
.expect("multistream decoder");
for tone_ch in 0..ch {
let mut enc = MsEncoder::new(layout).unwrap();
let mut enc = opus::MSEncoder::new(
SAMPLE_RATE,
layout.streams,
layout.coupled,
layout.mapping,
opus::Application::LowDelay,
)
.expect("multistream encoder");
let mut out = vec![0u8; 1400];
let mut decoded = vec![0f32; samples * ch];
let mut energy = vec![0f64; ch];
// A few frames so the codec converges past its startup transient.
for f in 0..8 {
@@ -819,28 +656,15 @@ mod tests {
/ SAMPLE_RATE as f32;
frame[t * ch + tone_ch] = 0.5 * phase.sin();
}
let n = enc.encode_float(&frame, samples, &mut out).unwrap();
let n = enc.encode_float(&frame, &mut out).unwrap();
assert!(n > 0);
// SAFETY: `dec` is the non-null decoder asserted above. `out.as_ptr()` is read for
// the `n` encoded bytes just produced by `encode_float`; `decoded.as_mut_ptr()` is
// written for up to `samples * ch` f32s and `decoded` is exactly that long; `samples`
// is the per-channel frame size. All buffers are live locals outliving the call; the
// return is checked to equal `samples`.
let got = unsafe {
audiopus_sys::opus_multistream_decode_float(
dec,
out.as_ptr(),
n as i32,
decoded.as_mut_ptr(),
samples as i32,
0,
)
};
assert_eq!(got as usize, samples);
let mut decoded = vec![0f32; samples * ch];
let got = dec.decode_float(&out[..n], &mut decoded, false).unwrap();
assert_eq!(got, samples);
if f >= 4 {
for t in 0..samples {
for c in 0..ch {
energy[c] += (decoded[t * ch + c] as f64).powi(2);
for (c, e) in energy.iter_mut().enumerate() {
*e += (decoded[t * ch + c] as f64).powi(2);
}
}
}
@@ -854,9 +678,6 @@ mod tests {
(energies: {energy:?})"
);
}
// SAFETY: `dec` is the decoder `opus_multistream_decoder_create` returned; the test owns it
// and destroys it exactly once here, after the final decode — no later use, no double free.
unsafe { audiopus_sys::opus_multistream_decoder_destroy(dec) };
}
/// Live 5.1 capture → multistream encode → decode, against a real PipeWire session.
@@ -869,7 +690,15 @@ mod tests {
fn surround_capture_live() {
let mut cap = crate::audio::open_audio_capture(6).expect("open 6ch capture");
let layout = &LAYOUT_51;
let mut enc = MsEncoder::new(layout).unwrap();
let mut enc = opus::MSEncoder::new(
SAMPLE_RATE,
layout.streams,
layout.coupled,
layout.mapping,
opus::Application::LowDelay,
)
.unwrap();
enc.set_vbr(false).ok(); // hard CBR so packet sizes are constant (audio FEC relies on it)
let mut out = vec![0u8; 1400];
let mut acc: Vec<f32> = Vec::new();
let frame_len = 240 * 6;
@@ -880,49 +709,24 @@ mod tests {
acc.extend_from_slice(&chunk);
while acc.len() >= frame_len && packets < 100 {
let frame: Vec<f32> = acc.drain(..frame_len).collect();
let n = enc.encode_float(&frame, 240, &mut out).unwrap();
let n = enc.encode_float(&frame, &mut out).unwrap();
sizes.insert(n);
packets += 1;
}
}
// Hard CBR: every multistream packet must be the same size (audio FEC relies on it).
assert_eq!(sizes.len(), 1, "CBR sizes: {sizes:?}");
// And a stock client's decoder must accept them.
// And a stock client's GFE-derived decoder must accept them.
let s = surround_params(layout, false);
let digits: Vec<u8> = s.bytes().map(|b| b - b'0').collect();
let client_mapping = client_swap(&digits[3..]);
let mut err = 0i32;
// SAFETY: scalar args are valid; `client_mapping.as_ptr()` addresses a 6-entry `Vec<u8>`
// (matches the 6-channel layout the decoder reads through it), alive past the call, and
// `&mut err` is a live local. The pointer is `OPUS_OK`-checked before use.
let dec = unsafe {
audiopus_sys::opus_multistream_decoder_create(
48000,
6,
layout.streams as i32,
layout.coupled as i32,
client_mapping.as_ptr(),
&mut err,
)
};
assert_eq!(err, audiopus_sys::OPUS_OK);
let mut dec =
opus::MSDecoder::new(SAMPLE_RATE, layout.streams, layout.coupled, &client_mapping)
.unwrap();
let mut pcm = vec![0f32; 240 * 6];
// SAFETY: `dec` is the non-null decoder from create. `out.as_ptr()` is read for the CBR
// packet length passed in (`*sizes.first()`, a real encoded packet size in `out`);
// `pcm.as_mut_ptr()` is written for up to `240 * 6` f32s and `pcm` is exactly that long;
// `240` is the per-channel frame size. All buffers are live locals outliving the call.
let got = unsafe {
audiopus_sys::opus_multistream_decode_float(
dec,
out.as_ptr(),
*sizes.first().unwrap() as i32,
pcm.as_mut_ptr(),
240,
0,
)
};
// SAFETY: `dec` is owned by the test; destroyed exactly once here after the final decode.
unsafe { audiopus_sys::opus_multistream_decoder_destroy(dec) };
let got = dec
.decode_float(&out[..*sizes.first().unwrap()], &mut pcm, false)
.unwrap();
assert_eq!(got, 240);
}
}
@@ -431,6 +431,9 @@ fn stream_body(
cfg.bitrate_kbps as u64 * 1000,
frame.is_cuda(),
8, // GameStream/Moonlight path: 8-bit (its own codec negotiation)
// GameStream/Moonlight stays 4:2:0 — stock Moonlight clients can't decode 4:4:4, and the
// protocol has no chroma negotiation. 4:4:4 is punktfunk/1-native only.
encode::ChromaFormat::Yuv420,
)
.context("open video encoder for stream")?;
// FEC overhead percent (Sunshine default 20). Override with PUNKTFUNK_FEC_PCT (0 = data-only).
@@ -560,6 +563,7 @@ fn stream_body(
cfg.bitrate_kbps as u64 * 1000,
frame.is_cuda(),
8,
encode::ChromaFormat::Yuv420, // GameStream stays 4:2:0
)
.context("reopen encoder after rebuild")?;
supports_rfi = enc.caps().supports_rfi;