feat(audio): an encode backend seam and an AAC output codec

doc/lib/c/index.md

@@ -35,7 +35,7 @@ and `target/include/moq.h`.
 - **Threading.** Any function from any thread. Raw publish calls block until the codec takes the frame, which paces a publisher.
 - **Connection health.** `moq_session_stats()` reports available metrics with per-field validity flags. `moq_session_snapshot()` samples those metrics and the negotiated draft name together from the same connection. Its protocol string is backed by static storage. Both return an offline error between reconnects and leave the destination untouched. `moq_session_bandwidth()` mints an allocator over the send estimate; `moq_bandwidth_reserve` claims a share for an app-owned track, and `moq_encode_video` / `moq_encode_audio` take the same handle so the built-in video encoder follows the grant.
 - **Raw playback.** Raw audio and video consumers start at the newest cached group when opened, so rebuilding a live decoder skips the retained backlog.
-- **Raw audio encode.** `moq_audio_encoder_output.codec` names the codec (only `"opus"` today), and `frame_duration_us` sets the Opus frame length: 2500, 5000, 10000, 20000, 40000, or 60000, with 0 meaning the 20 ms default.
+- **Raw audio encode.** `moq_audio_encoder_output.codec` names the codec: `"opus"`, `"pcm"`, or `"aac"`. `frame_duration_us` sets the Opus frame length: 2500, 5000, 10000, 20000, 40000, or 60000, with 0 meaning the codec's default (20 ms for Opus, 1024 samples for AAC). AAC-LC encodes through the platform's encoder, so a host without one refuses it.
 - **Audio channel layouts.** A `channels` count also names the speaker layout, by the WAVE convention: 1 is mono, 2 stereo, 3 2.1, 4 quad, 5 5.0, 6 5.1, 7 6.1, and 8 7.1, interleaved front left, front right, center, LFE, back, then side. `moq_decode_audio` remixes to the count you ask for; past 8 channels the samples pass through but can't be remixed.
 - **Raw decode output.** `moq_video_decoder_output` selects the decoded CPU pixel format (`MOQ_VIDEO_PIXEL_FORMAT_I420` or `_RGBA`) and target size (`width`/`height`, both zero for native; otherwise even and non-zero). Unknown formats and invalid sizes fail `moq_decode_video` before subscribing; accepted requests deliver exactly that layout or fail on the terminal callback.
 - **Encoded video metadata.** `moq_video_init.hint` is a zero-initialized `moq_video_hint` with `has_*` flags for coded dimensions, bitrate (bits per second), frame rate, and latency preference. Hints seed a video codec track's catalog; detected dimensions take precedence.

doc/lib/go/index.md

@@ -115,6 +115,12 @@ one: `FetchGroup`/`FetchMediaGroup`, `Dynamic()` with `Requests(ctx)`,
 `AppendDatagram`/`Datagrams(ctx)`, `SetCatalogSection`, `Demand()` for `Used`/`Unused`,
 `Session().Stats()`. `moq.IsAuthError` and `moq.IsShutdown` classify errors. `moq.ProtocolError(err)` is the structured protocol failure (scope, verbatim code, kind) when the peer sent one.
 
+`EncodeAudio` encodes raw PCM inside the binding. Its codec is `OpusAudioCodec()`
+or `AacAudioCodec()`, and `AudioEncoderOutput.FrameDurationUs` sets the Opus
+frame length: 2500, 5000, 10000, 20000 (the default), 40000, or 60000. 0 takes
+the codec's own frame, which AAC needs. AAC-LC encodes through the platform's
+encoder, so a host without one refuses it.
+
 Audio `Channels` also names the speaker layout, by the WAVE convention: 1 is
 mono, 2 stereo, 3 2.1, 4 quad, 5 5.0, 6 5.1, 7 6.1, and 8 7.1, interleaved
 front left, front right, center, LFE, back, then side. Decoding remixes to the

doc/lib/kt/index.md

@@ -91,8 +91,11 @@ connection's send estimate; pass it to `encodeVideo` / `encodeAudio` or
 native side.
 
 `encodeAudio` encodes raw PCM inside the binding. Its codec is an object,
-`AudioCodec.opus()`, and `AudioEncoderOutput.frameDurationUs` sets the Opus
-frame length: 2500, 5000, 10000, 20000 (the default), 40000, or 60000.
+`AudioCodec.opus()` or `AudioCodec.aac()`, and
+`AudioEncoderOutput.frameDurationUs` sets the Opus frame length: 2500, 5000,
+10000, 20000 (the default), 40000, or 60000. 0 takes the codec's own frame,
+which AAC needs. AAC-LC encodes through the platform's encoder, so a host
+without one refuses it.
 
 Audio `channels` also names the speaker layout, by the WAVE convention: 1 is
 mono, 2 stereo, 3 2.1, 4 quad, 5 5.0, 6 5.1, 7 6.1, and 8 7.1, interleaved

doc/lib/py/index.md

@@ -107,8 +107,11 @@ Each server request reports a `moq.Transport` enum, including QUIC, Iroh,
 WebSocket, TCP, and Unix sockets.
 
 `encode_audio` encodes raw PCM inside the binding. Its codec is an object,
-`moq.AudioCodec.opus()`, and `AudioEncoderOutput.frame_duration_us` sets the
-Opus frame length: 2500, 5000, 10000, 20000 (the default), 40000, or 60000.
+`moq.AudioCodec.opus()` or `moq.AudioCodec.aac()`, and
+`AudioEncoderOutput.frame_duration_us` sets the Opus frame length: 2500, 5000,
+10000, 20000 (the default), 40000, or 60000. 0 takes the codec's own frame,
+which AAC needs. AAC-LC encodes through the platform's encoder, so a host
+without one refuses it.
 
 Audio `channels` also names the speaker layout, by the WAVE convention: 1 is
 mono, 2 stereo, 3 2.1, 4 quad, 5 5.0, 6 5.1, 7 6.1, and 8 7.1, interleaved

doc/lib/rs/moq-audio.md

@@ -28,7 +28,7 @@ policy. Decoding likewise separates low-level `decode::Config`, PCM
 | Module | Does |
 | --- | --- |
 | `capture` | Microphones via CoreAudio, WASAPI, ALSA (and PipeWire/PulseAudio hosts), plus macOS system audio |
-| `encode` | PCM to Opus (with DTX and voice-activity signaling) or raw PCM for the lowest latency |
+| `encode` | PCM to Opus (with DTX and voice-activity signaling), raw PCM for the lowest latency, or AAC-LC through a platform encoder |
 | `decode` | Opus, PCM, and AAC-LC back to PCM, resampled to the rate you want |
 | `playback` | One output device mixing every track in a call, with click-free volume ramps |
 | `aec` | Acoustic echo cancellation (a port of WebRTC's), so a laptop with no headset doesn't feed itself back |
@@ -48,6 +48,25 @@ its config are refused at construction on every host. HE-AAC signaled only in
 band plays as its half-rate LC core. Linux has no OS audio decoder, so it will
 stay that way there.
 
+`encode` selects the same way, through `encode::Settings::kind`, and
+`Encoder::name()` reports what opened.
+
+| Backend | Encodes | Hosts |
+| --- | --- | --- |
+| `libopus` | Opus, mono or stereo | all |
+| `pcm` | PCM | all |
+
+`encode::Codec::Aac` is AAC-LC (`mp4a.40.2`) at the input's rate and layout:
+mono, stereo, 3.0, 4.0, 5.0, 5.1, or 7.1, the layouts with an AAC
+channelConfiguration. Frames are 1024 samples, so `Settings::from_input` sets
+`frame_duration` to match. The catalog's AudioSpecificConfig is built from the
+settings when the track is registered, and since it has no field for the
+encoder's delay, packets are stamped that much earlier so the first input
+sample still lands at the first timestamp. There is no software AAC encoder,
+and no platform encoder is wired in yet, so `Codec::Aac` is refused at
+construction on every host for now. Linux has no OS encoder, so it will stay
+that way there.
+
 Highlights:
 
 - **`encode::Publication`** advertises the track and opens the microphone only while someone listens. Stop, swap devices, and restart without changing the track subscribers know; read a level meter for the UI.

doc/lib/swift/index.md

@@ -95,8 +95,11 @@ divides the connection's send estimate; pass it to `encodeVideo` /
 (scope, verbatim code, kind) when the peer sent one.
 
 `encodeAudio` encodes raw PCM inside the binding. Its codec is an object,
-`AudioCodec.opus()`, and `AudioEncoderOutput.frameDurationUs` sets the Opus
-frame length: 2500, 5000, 10000, 20000 (the default), 40000, or 60000.
+`AudioCodec.opus()` or `AudioCodec.aac()`, and
+`AudioEncoderOutput.frameDurationUs` sets the Opus frame length: 2500, 5000,
+10000, 20000 (the default), 40000, or 60000. 0 takes the codec's own frame,
+which AAC needs. AAC-LC encodes through the platform's encoder, so a host
+without one refuses it.
 
 Audio `channels` also names the speaker layout, by the WAVE convention: 1 is
 mono, 2 stereo, 3 2.1, 4 quad, 5 5.0, 6 5.1, 7 6.1, and 8 7.1, interleaved

go/wrapper/publish.go

@@ -193,7 +193,7 @@ func (b *BroadcastProducer) PublishContainerStream(format ContainerFormat) (*Con
 
 // EncodeAudio publishes a raw-audio track with an in-process encoder.
 //
-// Select the codec with OpusAudioCodec (currently the only constructor).
+// Select the codec with OpusAudioCodec or AacAudioCodec.
 // Pass bandwidth to reserve this track's bitrate against the session's
 // allocator so a co-resident video encoder sizes itself against what is left.
 func (b *BroadcastProducer) EncodeAudio(name string, input AudioEncoderInput, output AudioEncoderOutput, bandwidth *Bandwidth) (*AudioProducer, error) {

go/wrapper/types.go

@@ -8,14 +8,14 @@ import ffi "moq.dev/moq-ffi/moq"
 type (
 	// Audio describes one audio rendition in a broadcast catalog: codec, sample rate, channel count, and container.
 	Audio = ffi.MoqAudio
-	// AudioCodec selects the audio encoder codec. Build one with OpusAudioCodec;
+	// AudioCodec selects the audio encoder codec. Build one with OpusAudioCodec or AacAudioCodec;
 	// adding a codec later adds a constructor, not a breaking enum change.
 	AudioCodec = ffi.MoqAudioCodec
 	// AudioDecoderOutput configures the PCM format, sample rate, and channels DecodeAudio delivers.
 	AudioDecoderOutput = ffi.MoqAudioDecoderOutput
 	// AudioEncoderInput declares the PCM sample format, sample rate, and channel count of frames written to an audio producer.
 	AudioEncoderInput = ffi.MoqAudioEncoderInput
-	// AudioEncoderOutput configures the Opus encoder: codec, optional sample rate, channels, bitrate, and frame duration.
+	// AudioEncoderOutput configures the encoder: codec, optional sample rate, channels, bitrate, and frame duration.
 	AudioEncoderOutput = ffi.MoqAudioEncoderOutput
 	// AudioSampleFormat is a raw PCM sample layout, mirroring WebCodecs AudioData.format.
 	AudioSampleFormat = ffi.MoqAudioSampleFormat
@@ -184,6 +184,12 @@ func OpusAudioCodec() *AudioCodec {
 	return ffi.MoqAudioCodecOpus()
 }
 
+// AacAudioCodec selects AAC-LC through the platform's encoder for EncodeAudio.
+// A host without one refuses it. Leave FrameDurationUs at 0 for AAC's own frame.
+func AacAudioCodec() *AudioCodec {
+	return ffi.MoqAudioCodecAac()
+}
+
 // VideoPixelFormat values: the raw pixel layout fed to the in-process encoder,
 // and the one the in-process decoder delivers.
 const (

kt/moq/src/jvmAndAndroidMain/kotlin/dev/moq/Aliases.kt

@@ -134,7 +134,7 @@ typealias FetchGroupOptions = uniffi.moq.MoqFetchGroupOptions
 typealias TrackInfo = uniffi.moq.MoqTrackInfo
 /** One audio frame: PCM payload bytes plus a presentation timestamp. */
 typealias AudioFrame = uniffi.moq.MoqAudioFrame
-/** Selects the audio encoder codec. Build one with `AudioCodec.opus()`. */
+/** Selects the audio encoder codec. Build one with `AudioCodec.opus()` or `AudioCodec.aac()`. */
 typealias AudioCodec = uniffi.moq.MoqAudioCodec
 /** A raw PCM sample format, mirroring WebCodecs `AudioData.format`. */
 typealias AudioSampleFormat = uniffi.moq.MoqAudioSampleFormat

py/moq-rs/README.md

@@ -154,7 +154,7 @@ client = moq.Client(
   - `.publish_audio(format, init, *, label=None, track=None) → MediaProducer`. `init` is required: an OpusHead or AudioSpecificConfig resolves the whole rendition. `track` names the track; otherwise a unique name is derived from the format.
   - `.publish_video(format, init=b"", *, label=None, hint=None, track=None) → MediaProducer`. `init` may be empty for a format that resolves in band; a `VideoHint` pins catalog fields the stream can't reveal (bitrate) or publishes the catalog before the first keyframe. `track` names the track as in `publish_audio`.
   - `.encode_video(input, output, *, bandwidth=None) → VideoProducer`. Encode raw `VideoFrame`s inside the binding; `.write(frame)` each one.
-  - `.encode_audio(name, input, output, *, bandwidth=None) → AudioProducer`. Encode raw PCM `AudioFrame`s; the codec is `output.codec`, e.g. `AudioCodec.opus()`, with `output.frame_duration_us` setting the Opus frame length.
+  - `.encode_audio(name, input, output, *, bandwidth=None) → AudioProducer`. Encode raw PCM `AudioFrame`s; the codec is `output.codec`, e.g. `AudioCodec.opus()` or `AudioCodec.aac()`, with `output.frame_duration_us` setting the Opus frame length (0 takes the codec's own frame, which AAC needs).
   - `.finish()`
 - **`BroadcastDynamic`**. Async source of tracks requested by subscribers.
   - `await .requested_track() → TrackRequest`. Call `.accept()` on it for a `TrackProducer`, or `.abort(code)` to reject.

py/moq-rs/moq/publish.py

@@ -749,8 +749,8 @@ def encode_audio(
     ) -> AudioProducer:
         """Publish a raw-audio track with an in-process encoder.
 
-        Select the codec with ``moq.AudioCodec.opus()`` (currently the only
-        constructor), placed in ``output``.
+        Select the codec with ``moq.AudioCodec.opus()`` or
+        ``moq.AudioCodec.aac()``, placed in ``output``.
 
         Pass ``bandwidth`` to reserve this track's bitrate against the session's
         allocator so a co-resident video encoder sizes itself against what is left.

quest/m1/audio-codecs/README.md

@@ -45,7 +45,6 @@ its own decode and encode quest so verification stays per host.
 
 - [AudioToolbox decode](/quest/m1/audio-codecs/decode-audiotoolbox.md) - macOS and iOS decode HE-AAC, multichannel AAC, and what else the framework offers
 - [Opus surround](/quest/m1/audio-codecs/opus-surround.md) - mapping family 1 decodes on every host through the multistream decoder
-- [Encode seam](/quest/m1/audio-codecs/encode-backend.md) - `encode::backend` and `Codec::Aac`, so a native publisher can produce AAC-LC
 - [AudioToolbox encode](/quest/m1/audio-codecs/encode-audiotoolbox.md) - macOS and iOS encode AAC-LC
 
 ## Related

quest/m1/audio-codecs/encode-audiotoolbox.md

@@ -15,11 +15,15 @@ the encode seam as the platform candidate on macOS and iOS.
   `kAudioConverterPrimeInfo` gives the delay the timestamps fold in.
 - Bitrate through `kAudioConverterEncodeBitRate`, updated live where the
   converter allows.
+- The seam assumes one packet per frame. If the converter holds output back,
+  the backend needs a `flush` and a zero-or-more return, which changes
+  `Encoder::encode` and so targets `dev`.
+- Gate the seam's "AAC refused without a platform encoder" test to hosts
+  without one.
 - Regression: a stereo and a 5.1 encode round-trip through the AudioToolbox
   decoder and through symphonia (stereo only), with timestamps continuous
   across the priming.
 
 ## Required
 
-- [Encode seam](/quest/m1/audio-codecs/encode-backend.md) - the candidate order this backend joins
 - [AudioToolbox decode](/quest/m1/audio-codecs/decode-audiotoolbox.md) - the round-trip regression decodes through it

quest/m1/obs-moq-video/audio-publish.md

@@ -8,15 +8,11 @@ MoQ publishing can encode OBS's mixed audio with moq-audio Opus while preserving
 
 - Expose `moq_audio::encode::Encoder` through a codec-only moq-ffi type with owned encoder/packet handles, so every binding gets it. The existing raw audio producer combines encoding and publication and must not create a second publication alongside the OBS encoded output. Reuse frame sizing, bitrate updates, catalog configuration, and finish/padding behavior.
 - Implement the OBS audio encoder interface, including fixed input frame size, mono/stereo PCM conversion, timestamps, codec headers, final padding, and packet release. Ask OBS for the input layout the encoder supports; moq-audio does not implement arbitrary channel remapping. Keep capture/mixing/device ownership in OBS.
-- Publish Opus initially. Leave AAC with the existing OBS mode and defer PCM publishing UI, since the output currently declares AAC/Opus. Correct stale binding documentation that describes the raw codec parser as Opus-only if that API is touched.
+- Publish Opus initially. `encode::Codec::Aac` lets the adapter offer AAC where a platform encoder exists; until then leave AAC with the existing OBS mode and defer PCM publishing UI, since the output currently declares AAC/Opus. Correct stale binding documentation that describes the raw codec parser as Opus-only if that API is touched.
 - Apply the shared presets and independent audio bitrate. Test 10/20 ms packetization, frame-size changes at stream boundaries, partial final frames, silence, reconnect, saturation, and stop with pending output. Verify decoded audio and A/V timestamps with a real subscriber.
 - Land the internal adapter independently; the combined Use MoQ encoders UI becomes available when the video adapter also lands. Do not expose a temporary video/audio mix-and-match product UI.
 
 ## Required
 
 - [OBS migration](/quest/m1/cpp/obs.md) - the plugin is on the generated C++ first
 - [Encoder presets](/quest/m1/obs-moq-video/presets.md) - shared policy and truthful reporting
-
-## Related
-
-- [Encode seam](/quest/m1/audio-codecs/encode-backend.md) - `Codec::Aac` lets the adapter offer AAC instead of leaving it to the OBS encoder mode

quest/m2/audio-encode-mediacodec.md

@@ -13,12 +13,14 @@ behind the `mediacodec` feature and the encode seam.
 - `audio/mp4a-latm` with `KEY_AAC_PROFILE` = LC. The catalog ASC is
   synthesized at construction per the encode seam, since `csd-0` only arrives
   with the first output buffer; assert the two match.
+- MediaCodec pipelines output, which the one-packet-per-frame seam does not
+  allow yet: add a `flush` and a zero-or-more return, a change to
+  `Encoder::encode` that targets `dev`, unless the AudioToolbox quest already did.
 - Multichannel is device-dependent; probe the encoder's capabilities at open
   and refuse a layout it does not list.
 - Round-trip regression through the MediaCodec decoder; runtime proof on a
   device or emulator.
 
 ## Required
 
-- [Encode seam](/quest/m1/audio-codecs/encode-backend.md) - the candidate order this backend joins
 - [MediaCodec decode](/quest/m2/audio-decode-mediacodec.md) - the round-trip regression decodes through it

quest/m2/audio-encode-mediafoundation.md

@@ -20,7 +20,6 @@ behind the encode seam on Windows.
 
 ## Required
 
-- [Encode seam](/quest/m1/audio-codecs/encode-backend.md) - the candidate order this backend joins
 - [Media Foundation decode](/quest/m2/audio-decode-mediafoundation.md) - the round-trip regression decodes through it
 
 ## Related