EdisonJwa
78190c0694
Replace the prior one-shot android_engage_voice_communication call with a ModeStack-mediated acquire/release pair. AudioEngine snapshots the system audio mode on first acquire via android_get_audio_mode() and restores it on last release via android_set_audio_mode(prior). MODE_IN_COMMUNICATION (3) is engaged across the voice-session lifetime per SDD-108. Includes the Android AudioManager getMode/setMode JNI helpers (placed in chanora_audio::engine alongside the existing JNI surface) and the small ptt.rs touch needed for the SDD-108 ID-tag on the existing tests. Trace: SDD-108, SDD-115.
1754 lines
74 KiB
Rust
1754 lines
74 KiB
Rust
//! Audio engine — owns the cpal input/output streams, the Opus
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//! encoder, and the tsclientlib `AudioHandler` for decode+mix.
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//!
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//! The engine is started after a protocol connection is established
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//! and stopped before disconnect. It does not retry on device
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//! change.
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use std::sync::atomic::{AtomicBool, AtomicU32, Ordering};
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use std::sync::{Arc, Mutex};
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#[cfg(not(target_os = "ios"))]
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use cpal::traits::{DeviceTrait, HostTrait, StreamTrait};
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#[cfg(not(target_os = "ios"))]
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use cpal::{SampleFormat, SizedSample};
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use tokio::sync::mpsc;
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use tracing::{debug, info};
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// `error!` and `warn!` are used only inside the cpal capture /
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// playback paths (`build_input_stream`, `build_output_stream`,
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// `try_open_capture` log lines). Cfg-gate the imports too so iOS
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// builds don't carry an unused-imports warning.
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#[cfg(not(target_os = "ios"))]
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use tracing::{error, warn};
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#[cfg(not(target_os = "ios"))]
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use audiopus::coder::Encoder as OpusEncoder;
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#[cfg(not(target_os = "ios"))]
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use audiopus::{
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Application as OpusApp, Bitrate as OpusBitrate, Channels as OpusChannels,
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SampleRate as OpusSampleRate,
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};
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use tsclientlib::audio::AudioHandler;
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// `AudioData`, `CodecType`, `OutAudio` are referenced only by the
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// cpal capture pipeline's Opus encode path (`CaptureState::encode_and_send`).
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// `InboundVoice` + `OutPacket` are used by every platform — the
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// inbound forwarder task pumps `InboundVoice` into AudioHandler on
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// iOS too, and `OutPacket` flows out of the capture pipeline once
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// commit 3 lands. Cfg-gate the cpal-only ones to keep iOS warnings
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// clean.
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#[cfg(not(target_os = "ios"))]
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use chanora_protocol::{AudioData, CodecType, OutAudio};
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use chanora_protocol::{InboundVoice, OutPacket};
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use crate::AudioError;
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/// Stable Chanora-side identifier for AudioHandler bookkeeping.
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/// We only ever have one connection at a time (DEC-006), so this is
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/// trivially unique.
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#[derive(Clone, Copy, Debug, Eq, Hash, PartialEq)]
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pub struct SessionAudioId(pub u64);
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/// Audio framing: 48 kHz mono, 20 ms = 960 samples per frame.
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/// These constants are framing invariants of the engine and are
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/// referenced from per-platform helpers (`try_open_capture` and
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/// the CaptureState on cpal platforms; `ios_voice_unit` on iOS once
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/// commits 3+4 land). The `allow(dead_code)` is here because in
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/// the current commit the iOS VPIO callbacks are still no-op stubs
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/// and don't reach these constants yet — they will in commit 3
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/// when the input callback wires into CaptureState.
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#[allow(dead_code)]
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const SAMPLE_RATE: u32 = 48_000;
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#[allow(dead_code)]
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const FRAME_SAMPLES: usize = 48_000 / 50; // 960
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#[allow(dead_code)]
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const MAX_OPUS_FRAME: usize = 1275;
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/// Engine configuration.
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#[derive(Debug, Clone)]
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pub struct AudioEngineConfig {
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/// Input gain applied before encoding (1.0 = pass-through).
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pub mic_gain: f32,
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/// Initial PTT state. When false the encoder is bypassed and no
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/// outbound packets are produced.
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pub ptt_initial: bool,
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/// Audio-effect toggles. The struct is honoured by *naming* but
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/// the filters themselves are still no-op in Beta — see the
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/// crate-level docs and DEC-007/008/009/010.
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pub effects: crate::AudioEffects,
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/// A.5 mobile: prefer the OS-provided "voice communication"
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/// audio source on mobile platforms (Android
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/// `MediaRecorder.AudioSource.VOICE_COMMUNICATION`, iOS
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/// `AVAudioSession.Mode.voiceChat`). On Linux desktop this is
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/// ignored — the DSP chain stays a no-op and we use the
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/// default ALSA/PipeWire source.
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///
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/// Beta status: the *config flag* is plumbed through every
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/// layer; the *Android-side preset switch* is documented but
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/// not yet wired through cpal, which currently uses the
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/// AAudio default input. RISK-AUDIO-MOBILE-001 tracks this gap.
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/// Setting `true` is a forward-compatible no-op for Beta and
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/// will become active once cpal exposes input-preset hooks (or
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/// when Chanora ships an Oboe-based fork).
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pub mobile_voice_preset: bool,
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}
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impl Default for AudioEngineConfig {
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fn default() -> Self {
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Self {
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mic_gain: 1.0,
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ptt_initial: false,
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effects: crate::AudioEffects::default(),
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mobile_voice_preset: true,
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}
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}
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}
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/// Running audio engine. Drop = stop.
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pub struct AudioEngine {
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/// `transmit_active` is the authoritative gate for outbound
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/// voice — the Opus encoder feed consults this flag once per
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/// 20 ms frame. PTT subsystems (focused widget, future
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/// Windows / macOS / Linux global backends) drive this flag
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/// through [`Self::set_transmit_active`]; nothing else is
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/// permitted to flip it (SAD-075 / SDD-089).
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transmit_gate: crate::ptt::AudioTransmitGate,
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frames_sent: Arc<AtomicU32>,
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frames_received: Arc<AtomicU32>,
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/// Master output gain as f32 bits in an AtomicU32. Default 1.0.
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/// Adjusted via [`Self::set_output_gain`] from the bridge.
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output_gain: Arc<AtomicU32>,
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/// Master output mute. When true the output callback fills the
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/// device buffer with silence regardless of incoming voice
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/// frames. Used for self-output-mute on the local device,
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/// independent of the server-side mute the protocol layer
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/// broadcasts.
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output_muted: Arc<AtomicBool>,
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// Streams must be dropped to stop audio. Both are `!Send` because
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// cpal's Stream isn't Send on some backends; we keep them in an
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// Option wrapped by Mutex so stop() can move them out. On Linux
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// the output side is `crate::sdl_output::SdlOutput` instead of a
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// cpal Stream (see the SDD note inside `sdl_output.rs`); the
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// same unsafe Send/Sync impl below covers both. On iOS both
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// sides collapse into a single `IosVoiceUnit` (one
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// VoiceProcessingIO AudioUnit hosts mic + speaker) — cpal is
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// unused on iOS for the reasons documented in
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// `ios_voice_unit.rs`.
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#[cfg(not(target_os = "ios"))]
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_input_stream: Mutex<Option<cpal::Stream>>,
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#[cfg(target_os = "linux")]
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_output_stream: Mutex<Option<crate::sdl_output::SdlOutput>>,
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#[cfg(all(not(target_os = "linux"), not(target_os = "ios")))]
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_output_stream: Mutex<Option<cpal::Stream>>,
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#[cfg(target_os = "ios")]
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_ios_voice_unit: Mutex<Option<crate::ios_voice_unit::IosVoiceUnit>>,
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/// SDD-111..SDD-115: Android voice backend held parallel to the
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/// cpal streams. Owns the SDD-113 hardware-effect handles and
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/// the foreground-service lifecycle; tearing it down on engine
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/// drop unbinds effects and stops the service in SDD-115
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/// reverse order. The cpal capture/playback pair continues to
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/// carry voice frames for the transitional period — replacing
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/// the data path with the Oboe streams is a follow-up.
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#[cfg(target_os = "android")]
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_android_voice_unit: Mutex<Option<crate::android_voice_unit::AndroidVoiceUnit>>,
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/// SDD-108 §1/§2: refcount-composable audio-mode controller.
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/// Snapshots `AudioManager.getMode()` on the 0 → 1 transition and
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/// restores it on the 1 → 0 transition. Held in a `Mutex` so the
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/// snapshot/restore critical section is serialized across
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/// composed callers (SDD-108 §2). Engine-scoped because the
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/// mode lifecycle is bound to the voice-session lifecycle
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/// (SDD-108 §3).
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#[cfg(target_os = "android")]
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audio_mode_stack: Mutex<crate::mode_stack::ModeStack>,
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// Hand the inbound-voice forwarder task a shutdown signal.
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shutdown_tx: Option<tokio::sync::oneshot::Sender<()>>,
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/// True if the capture stream actually opened. If false (typical
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/// in headless environments with null sources, or where the user
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/// denied microphone permission), PTT becomes a no-op and
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/// `frames_sent` stays at 0.
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capture_active: bool,
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/// Missed-key-up watchdog (SDD-092). Dropping aborts the task.
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/// The watchdog is independent of the PTT input backend — it
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/// observes the gate directly. The platform input backend is
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/// owned by `chanora_core::ptt::PttController` (SDD-088), not
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/// by the engine.
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///
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/// In the post-rc.7 architecture this field is unused: the
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/// missed-key-up watchdog now lives on the session and
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/// subscribes to `TransmitModeSelector::subscribe_ptt_held`
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/// rather than the gate. Watching the gate caused the watchdog
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/// to fire in Continuous mode (where the gate is intentionally
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/// pinned to `true`) which clearing surfaced as the bug
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/// "Continuous transmission disabled after some time". The
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/// field stays here as `None` for now to preserve the existing
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/// engine-stop teardown flow; a follow-up commit can remove it
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/// entirely.
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ptt_watchdog: Option<crate::ptt::MissedKeyUpWatchdog>,
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}
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// cpal::Stream is not Send. We keep the engine pinned to the thread
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// it was constructed on — `chanora_core` spawns it inside a
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// `tokio::task::spawn_blocking` so the streams stay on that worker.
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// This `unsafe impl Send` is necessary because the outer Arc<AudioEngine>
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// is stored in core's session and must move into a task. The streams
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// themselves are only mutated through the Mutex and are dropped on
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// the same thread that owns them.
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//
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// SAFETY: cpal's Stream is not Send because the underlying audio API
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// callback thread may not be transferable. We never invoke methods on
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// the streams from any thread but the owning one; we only ever *drop*
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// them, which cpal documents as safe from any thread for ALSA and
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// PipeWire (Linux backend used here). For Windows/macOS the contract
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// may differ; production Beta+ work must revisit per-platform.
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unsafe impl Send for AudioEngine {}
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unsafe impl Sync for AudioEngine {}
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impl AudioEngine {
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/// Start the engine: open capture + playback streams, spawn the
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/// inbound-voice forwarder, return a handle.
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pub fn start(
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cfg: AudioEngineConfig,
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voice_out_tx: mpsc::Sender<OutPacket>,
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voice_in_rx: mpsc::Receiver<InboundVoice>,
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) -> Result<Self, AudioError> {
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let gate = crate::ptt::AudioTransmitGate::new(cfg.ptt_initial);
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Self::start_with_gate(cfg, voice_out_tx, voice_in_rx, gate)
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}
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/// Start the engine using an externally-owned
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/// [`AudioTransmitGate`]. The gate is shared with whatever
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/// upstream (typically [`crate::TransmitModeSelector`]) is the
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/// authoritative writer of `transmit_active`. See SAD-083.
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pub fn start_with_gate(
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cfg: AudioEngineConfig,
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voice_out_tx: mpsc::Sender<OutPacket>,
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voice_in_rx: mpsc::Receiver<InboundVoice>,
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transmit_gate: crate::ptt::AudioTransmitGate,
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) -> Result<Self, AudioError> {
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// iOS routes to a separate backend (VoiceProcessingIO via
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// coreaudio-rs) because cpal's iOS RemoteIO path produces
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// mono-only output bound to a stale physical transducer
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// (see `ios_voice_unit.rs` for the long version). Every
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// other platform stays on the cpal / SDL flow below.
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#[cfg(target_os = "ios")]
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{
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return Self::start_with_gate_ios(cfg, voice_out_tx, voice_in_rx, transmit_gate);
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}
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#[cfg(not(target_os = "ios"))]
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{
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Self::start_with_gate_cpal(cfg, voice_out_tx, voice_in_rx, transmit_gate)
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}
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}
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/// Non-iOS implementation: cpal capture + (cpal | SDL2) output.
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/// Kept as a separate function so the iOS path can short-circuit
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/// at the top of `start_with_gate` without dragging a 200-line
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/// cfg-gated block. Body is the pre-iOS-port code, unchanged
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/// except for the new function name + signature.
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#[cfg(not(target_os = "ios"))]
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fn start_with_gate_cpal(
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cfg: AudioEngineConfig,
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voice_out_tx: mpsc::Sender<OutPacket>,
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mut voice_in_rx: mpsc::Receiver<InboundVoice>,
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transmit_gate: crate::ptt::AudioTransmitGate,
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) -> Result<Self, AudioError> {
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let host = cpal::default_host();
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info!(
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target: "chanora_audio",
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host_id = ?host.id(),
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"starting audio engine: cpal host selected"
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);
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let in_dev = host
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.default_input_device()
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.ok_or(AudioError::NoInputDevice)?;
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let out_dev = host
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.default_output_device()
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.ok_or(AudioError::NoOutputDevice)?;
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info!(
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target: "chanora_audio",
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in_device = %in_dev.description().map(|d| d.name().to_owned()).unwrap_or_default(),
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out_device = %out_dev.description().map(|d| d.name().to_owned()).unwrap_or_default(),
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"starting audio engine"
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);
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// Log the cpal-reported default configs *before* trying to
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// open streams, so a downstream stream-build failure can
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// be cross-referenced against what the platform reported
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// as its default format. Some locale / driver combinations
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// on Windows have been observed to expose configs that
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// accept device enumeration but reject `default_*_config`
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// afterwards (reported on the ko-KR Windows 11 host as
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// "Start audio button not work"). Promote what would
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// otherwise be silent or laconic errors into structured
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// log records the user can paste back.
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match in_dev.default_input_config() {
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Ok(c) => info!(
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target: "chanora_audio",
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channels = c.channels(),
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sample_rate = c.sample_rate(),
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sample_format = ?c.sample_format(),
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"default_input_config reported"
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),
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Err(e) => warn!(
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target: "chanora_audio",
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error = %e,
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"default_input_config FAILED — capture will be disabled"
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),
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}
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match out_dev.default_output_config() {
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Ok(c) => info!(
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target: "chanora_audio",
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channels = c.channels(),
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sample_rate = c.sample_rate(),
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sample_format = ?c.sample_format(),
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"default_output_config reported"
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),
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Err(e) => warn!(
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target: "chanora_audio",
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error = %e,
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"default_output_config FAILED — output stream will fail to build"
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),
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}
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// SDD-115 lifecycle sequencing on engine start. Forward order:
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// 1) bridge -> engine receives voice_join (here we are
|
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// already inside `start`, the engine-side trigger).
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// 2) start the Android voice foreground service so that
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// the platform records the microphone capture under
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// `foregroundServiceType="microphone"` (SDD-107 + SRS-215).
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// 3) open the AAudio voice streams (SDD-111 + SDD-112).
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// 4) engage `MODE_IN_COMMUNICATION` (SDD-108).
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// 5) bind hardware effects (SDD-113) — performed inside
|
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// `AndroidVoiceUnit::open()` once the input stream has a
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// session id.
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// The reverse order on engine drop is enforced by the field
|
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// drop order (`_android_voice_unit` is dropped before the
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// engine returns; `close()` is invoked from its Drop impl).
|
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#[cfg(target_os = "android")]
|
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let mut audio_mode_stack = crate::mode_stack::ModeStack::new();
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#[cfg(target_os = "android")]
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let _android_voice_unit = {
|
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if cfg.mobile_voice_preset {
|
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// Step 2: foreground service.
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if crate::android_voice_unit::chanora_android_start_voice_service() {
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info!(
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target: "chanora_audio",
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"android: voice foreground service start dispatched (SDD-115)"
|
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);
|
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} else {
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warn!(
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target: "chanora_audio",
|
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"android: foreground service start failed; capture may be denied in background (SDD-115)"
|
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);
|
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}
|
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// Step 4 (mode engage) BEFORE Step 5 (effect bind);
|
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// hardware-effect routing only engages reliably under
|
||
// MODE_IN_COMMUNICATION (SDD-113 item 6 / SDD-115).
|
||
//
|
||
// SDD-108 §1/§2: route through `ModeStack` so the
|
||
// 0 → 1 transition snapshots the prior platform mode
|
||
// (via `android_get_audio_mode`) and only that
|
||
// transition writes `MODE_IN_COMMUNICATION` via
|
||
// `android_set_audio_mode`. P0 only ever observes
|
||
// refcount {0, 1} per SRS-189 but the composition
|
||
// model is in place for P1.
|
||
match android_get_audio_mode() {
|
||
Ok(prior_now) => {
|
||
let outcome = audio_mode_stack.acquire(prior_now);
|
||
if let crate::mode_stack::ModeAcquire::FirstAcquire { prior } = outcome {
|
||
match android_set_audio_mode(ANDROID_MODE_IN_COMMUNICATION) {
|
||
Ok(()) => info!(
|
||
target: "chanora_audio",
|
||
prior_mode = prior,
|
||
"android: AudioManager mode set to MODE_IN_COMMUNICATION (SDD-108)"
|
||
),
|
||
Err(e) => {
|
||
// SDD-108 §5: setMode failed AFTER
|
||
// the 0 → 1 ModeStack transition.
|
||
// Roll the stack back so refcount
|
||
// returns to 0 and the snapshot is
|
||
// cleared; otherwise a future
|
||
// release would issue an
|
||
// unmatched setMode(prior) against
|
||
// a system that never had its mode
|
||
// changed by us.
|
||
warn!(
|
||
target: "chanora_audio",
|
||
error = %e,
|
||
prior_mode = prior,
|
||
"android: setMode failed; rolling back ModeStack acquire (SDD-108 §5)"
|
||
);
|
||
let _ = audio_mode_stack.release();
|
||
}
|
||
}
|
||
}
|
||
}
|
||
Err(e) => warn!(
|
||
target: "chanora_audio",
|
||
error = %e,
|
||
"android: AudioManager.getMode failed; skipping mode engage (SDD-108)"
|
||
),
|
||
}
|
||
// Step 3 + 5: open streams (SDD-111/112) and bind
|
||
// hardware effects (SDD-113). Failure here is logged
|
||
// and the engine continues with software AEC/NS/AGC
|
||
// via the existing engine path; the cpal data path
|
||
// remains the in-flight carrier.
|
||
//
|
||
// The prior silent-no-op log line at this site
|
||
// ("engagement depends on device AEC/NS support
|
||
// under MODE_IN_COMMUNICATION") is removed: the
|
||
// AndroidVoiceUnit either succeeds in engaging
|
||
// hardware effects (SDD-113) or logs the per-effect
|
||
// fallback, so engagement is now observable rather
|
||
// than rationalised.
|
||
let cfg_av = crate::mobile_voice_backend::AndroidVoiceStreamConfig {
|
||
effects: cfg.effects,
|
||
..Default::default()
|
||
};
|
||
match crate::android_voice_unit::AndroidVoiceUnit::open(&cfg_av) {
|
||
Ok(mut unit) => {
|
||
use crate::mobile_voice_backend::MobileVoiceAudioBackend;
|
||
if let Err(e) = unit.start() {
|
||
warn!(
|
||
target: "chanora_audio",
|
||
error = %e,
|
||
"android: AndroidVoiceUnit::start failed — cpal path remains active (SDD-115)"
|
||
);
|
||
}
|
||
Some(unit)
|
||
}
|
||
Err(e) => {
|
||
warn!(
|
||
target: "chanora_audio",
|
||
error = %e,
|
||
"android: AndroidVoiceUnit::open failed — software AEC/NS/AGC fallback engages (SDD-111/SDD-113)"
|
||
);
|
||
None
|
||
}
|
||
}
|
||
} else {
|
||
None
|
||
}
|
||
};
|
||
#[cfg(target_os = "ios")]
|
||
{
|
||
if cfg.mobile_voice_preset {
|
||
// iOS AVAudioSession configuration is performed
|
||
// Swift-side in `apps/chanora_flutter/ios/Runner/
|
||
// AppDelegate.swift::application(_:didFinishLaunching\
|
||
// WithOptions:)` BEFORE Flutter starts its audio
|
||
// pipeline. The category/mode set there
|
||
// (`.playAndRecord` + `.voiceChat`,
|
||
// `defaultToSpeaker | allowBluetooth |
|
||
// allowBluetoothA2DP`) is the recommended iOS shape
|
||
// for voice clients and engages on-device AEC / NS
|
||
// routing where supported. cpal's CoreAudio
|
||
// backend then opens its streams against that
|
||
// session and inherits the routing. Logging here
|
||
// just records that the engine-start path
|
||
// acknowledges the request; the actual session
|
||
// mutation lives in Swift because it must happen
|
||
// before Dart loads.
|
||
info!(
|
||
target: "chanora_audio",
|
||
"ios: voice-chat session mode requested — AVAudioSession configured in AppDelegate"
|
||
);
|
||
}
|
||
}
|
||
|
||
let transmit_flag_for_capture = transmit_gate.flag_arc();
|
||
let frames_sent = Arc::new(AtomicU32::new(0));
|
||
let frames_received = Arc::new(AtomicU32::new(0));
|
||
let output_gain = Arc::new(AtomicU32::new(1.0_f32.to_bits()));
|
||
let output_muted = Arc::new(AtomicBool::new(false));
|
||
|
||
// ---------- Capture ----------
|
||
// Capture is best-effort. If the platform default input
|
||
// device refuses any supported config (typical for
|
||
// headless null sources or for users who deny the mic
|
||
// permission) we log and continue — playback alone is
|
||
// still useful. PTT becomes a no-op in that case.
|
||
let capture_result = try_open_capture(
|
||
&in_dev,
|
||
voice_out_tx,
|
||
transmit_flag_for_capture,
|
||
frames_sent.clone(),
|
||
cfg.mic_gain,
|
||
);
|
||
let (input_stream, capture_active) = match capture_result {
|
||
Ok(s) => (Some(s), true),
|
||
Err(e) => {
|
||
warn!(
|
||
target: "chanora_audio",
|
||
error = %e,
|
||
"capture stream unavailable; continuing with playback only"
|
||
);
|
||
(None, false)
|
||
}
|
||
};
|
||
if let Some(s) = &input_stream {
|
||
s.play()
|
||
.map_err(|e| AudioError::Backend(format!("input play: {e}")))?;
|
||
}
|
||
|
||
// ---------- Playback ----------
|
||
let audio_handler: Arc<Mutex<AudioHandler<SessionAudioId>>> =
|
||
Arc::new(Mutex::new(AudioHandler::new()));
|
||
|
||
// Linux uses SDL2 for output (Qint / upstream tsclientlib
|
||
// pattern). cpal's Linux backend opens raw ALSA which routes
|
||
// through `dmix`+`plug` and produces audible crackling /
|
||
// popping on the 48 kHz → device-rate step. SDL2 on the same
|
||
// box routes through PipeWire's PA bridge (or PulseAudio)
|
||
// whose resampler is high-quality. We keep cpal on Windows
|
||
// and macOS — both have native backends (WASAPI / CoreAudio)
|
||
// without this problem. See `crates/chanora_audio/src/sdl_output.rs`
|
||
// for the full rationale.
|
||
#[cfg(target_os = "linux")]
|
||
let output_stream = crate::sdl_output::SdlOutput::start(
|
||
audio_handler.clone(),
|
||
output_gain.clone(),
|
||
output_muted.clone(),
|
||
)?;
|
||
|
||
#[cfg(not(target_os = "linux"))]
|
||
let output_stream = {
|
||
let out_cfg = out_dev
|
||
.default_output_config()
|
||
.map_err(|e| AudioError::StreamConfig(format!("output default: {e}")))?;
|
||
let out_format = out_cfg.sample_format();
|
||
let dev_sample_rate = out_cfg.sample_rate();
|
||
let dev_channels = out_cfg.channels() as usize;
|
||
// Buffer-size rationale:
|
||
// * Windows (WASAPI via cpal): the default period is
|
||
// small enough to expose audio-thread scheduler
|
||
// jitter on shared-mode endpoints. Pinning at 2048
|
||
// frames (~46 ms @ 44.1 kHz) gives the Opus decode
|
||
// callback enough headroom while still being well
|
||
// under voice-chat latency tolerance.
|
||
// * macOS (CoreAudio via cpal): the default period
|
||
// is fine and the OS picks a HAL-friendly size.
|
||
// * iOS (CoreAudio via cpal): RemoteIO units reject
|
||
// arbitrary buffer-size requests and surface them
|
||
// as `build_output_stream: The requested stream
|
||
// configuration is not supported by the device`.
|
||
// Must use BufferSize::Default.
|
||
#[cfg(target_os = "windows")]
|
||
let buffer_size = cpal::BufferSize::Fixed(2048);
|
||
#[cfg(not(target_os = "windows"))]
|
||
let buffer_size = cpal::BufferSize::Default;
|
||
let out_stream_cfg = cpal::StreamConfig {
|
||
channels: out_cfg.channels(),
|
||
sample_rate: out_cfg.sample_rate(),
|
||
buffer_size,
|
||
};
|
||
info!(
|
||
target: "chanora_audio",
|
||
dev_sample_rate,
|
||
dev_channels,
|
||
buffer_size = ?buffer_size,
|
||
"output stream using device native config (no 48k force)"
|
||
);
|
||
|
||
let stream = match out_format {
|
||
SampleFormat::F32 => build_output_stream::<f32>(
|
||
&out_dev,
|
||
&out_stream_cfg,
|
||
audio_handler.clone(),
|
||
output_gain.clone(),
|
||
output_muted.clone(),
|
||
dev_sample_rate,
|
||
dev_channels,
|
||
)?,
|
||
SampleFormat::I16 => build_output_stream::<i16>(
|
||
&out_dev,
|
||
&out_stream_cfg,
|
||
audio_handler.clone(),
|
||
output_gain.clone(),
|
||
output_muted.clone(),
|
||
dev_sample_rate,
|
||
dev_channels,
|
||
)?,
|
||
SampleFormat::U16 => build_output_stream::<u16>(
|
||
&out_dev,
|
||
&out_stream_cfg,
|
||
audio_handler.clone(),
|
||
output_gain.clone(),
|
||
output_muted.clone(),
|
||
dev_sample_rate,
|
||
dev_channels,
|
||
)?,
|
||
other => {
|
||
return Err(AudioError::StreamConfig(format!(
|
||
"unsupported output format: {other:?}"
|
||
)))
|
||
}
|
||
};
|
||
stream
|
||
.play()
|
||
.map_err(|e| AudioError::Backend(format!("output play: {e}")))?;
|
||
stream
|
||
};
|
||
|
||
// ---------- Inbound forwarder ----------
|
||
let (shutdown_tx, mut shutdown_rx) = tokio::sync::oneshot::channel();
|
||
let handler_for_task = audio_handler.clone();
|
||
let frames_received_for_task = frames_received.clone();
|
||
tokio::spawn(async move {
|
||
loop {
|
||
tokio::select! {
|
||
_ = &mut shutdown_rx => {
|
||
debug!(target: "chanora_audio", "inbound forwarder shutting down");
|
||
break;
|
||
}
|
||
item = voice_in_rx.recv() => {
|
||
match item {
|
||
Some(v) => {
|
||
let id = SessionAudioId(v.from_client);
|
||
let mut h = handler_for_task.lock().unwrap();
|
||
if let Err(e) = h.handle_packet(id, v.packet) {
|
||
debug!(target: "chanora_audio", error = %e, "decode failed");
|
||
} else {
|
||
frames_received_for_task.fetch_add(1, Ordering::Relaxed);
|
||
}
|
||
}
|
||
None => break,
|
||
}
|
||
}
|
||
}
|
||
}
|
||
});
|
||
|
||
// Select and arm the desktop PTT backend is no longer the
|
||
// engine's job (SDD-088). The PTT controller lives in
|
||
// `chanora_core::ptt::PttController`; the engine is
|
||
// responsible only for the cpal streams and the
|
||
// missed-key-up watchdog (SDD-092).
|
||
|
||
// The engine no longer spawns a watchdog against the
|
||
// gate (see comment on the `ptt_watchdog` field for the
|
||
// rationale). The session spawns the watchdog against the
|
||
// selector's `ptt_held` signal instead, so it never fires
|
||
// in Continuous mode.
|
||
let ptt_watchdog: Option<crate::ptt::MissedKeyUpWatchdog> = None;
|
||
|
||
Ok(Self {
|
||
transmit_gate,
|
||
frames_sent,
|
||
frames_received,
|
||
output_gain,
|
||
output_muted,
|
||
_input_stream: Mutex::new(input_stream),
|
||
_output_stream: Mutex::new(Some(output_stream)),
|
||
#[cfg(target_os = "android")]
|
||
_android_voice_unit: Mutex::new(_android_voice_unit),
|
||
#[cfg(target_os = "android")]
|
||
audio_mode_stack: Mutex::new(audio_mode_stack),
|
||
shutdown_tx: Some(shutdown_tx),
|
||
capture_active,
|
||
ptt_watchdog,
|
||
})
|
||
}
|
||
|
||
/// iOS implementation: a single VoiceProcessingIO AudioUnit
|
||
/// drives mic capture + speaker playback (see
|
||
/// `ios_voice_unit.rs` for why cpal is unsuitable on iOS).
|
||
/// This mirrors `start_with_gate_cpal` in scaffolding —
|
||
/// atomics, audio handler, inbound forwarder task — but
|
||
/// replaces the two cpal stream constructions with a single
|
||
/// `IosVoiceUnit::start` call.
|
||
#[cfg(target_os = "ios")]
|
||
fn start_with_gate_ios(
|
||
cfg: AudioEngineConfig,
|
||
voice_out_tx: mpsc::Sender<OutPacket>,
|
||
mut voice_in_rx: mpsc::Receiver<InboundVoice>,
|
||
transmit_gate: crate::ptt::AudioTransmitGate,
|
||
) -> Result<Self, AudioError> {
|
||
info!(
|
||
target: "chanora_audio",
|
||
"starting audio engine: iOS VoiceProcessingIO backend"
|
||
);
|
||
|
||
// iOS AVAudioSession configuration is performed Swift-side
|
||
// in `apps/chanora_flutter/ios/Runner/AppDelegate.swift`
|
||
// BEFORE Flutter starts its audio pipeline. The category +
|
||
// mode pair set there (`.playAndRecord` + `.default`) is
|
||
// what VPIO binds against. Logging here just records that
|
||
// the engine-start path acknowledges the request; the
|
||
// actual session mutation lives in Swift because it must
|
||
// happen before Dart loads.
|
||
if cfg.mobile_voice_preset {
|
||
info!(
|
||
target: "chanora_audio",
|
||
"ios: voice-chat session mode requested — AVAudioSession configured in AppDelegate"
|
||
);
|
||
}
|
||
|
||
let transmit_flag_for_capture = transmit_gate.flag_arc();
|
||
let frames_sent = Arc::new(AtomicU32::new(0));
|
||
let frames_received = Arc::new(AtomicU32::new(0));
|
||
let output_gain = Arc::new(AtomicU32::new(1.0_f32.to_bits()));
|
||
let output_muted = Arc::new(AtomicBool::new(false));
|
||
|
||
let audio_handler: Arc<Mutex<AudioHandler<SessionAudioId>>> =
|
||
Arc::new(Mutex::new(AudioHandler::new()));
|
||
|
||
// Construct the VPIO unit. Commit 1 ships a no-op callback
|
||
// pair; commits 3 + 4 land the real capture + playback
|
||
// wiring. Construction failure here is fatal (mirrors how
|
||
// the cpal output-stream construction failure is fatal in
|
||
// the non-iOS path).
|
||
let ios_voice_unit = crate::ios_voice_unit::IosVoiceUnit::start(
|
||
audio_handler.clone(),
|
||
output_gain.clone(),
|
||
output_muted.clone(),
|
||
voice_out_tx,
|
||
transmit_flag_for_capture,
|
||
frames_sent.clone(),
|
||
cfg.mic_gain,
|
||
)?;
|
||
|
||
// Capture is always considered active on iOS — VPIO's
|
||
// input element is wired up by the AudioUnit itself, no
|
||
// separate "did the capture stream open" question to
|
||
// answer. If the user denied mic permission VPIO will
|
||
// simply hand us silence buffers.
|
||
let capture_active = true;
|
||
|
||
// Inbound forwarder: same shape as the non-iOS path. Pumps
|
||
// Opus packets from `voice_in_rx` into AudioHandler so the
|
||
// VPIO render callback (commit 4) finds decoded frames
|
||
// waiting.
|
||
let (shutdown_tx, mut shutdown_rx) = tokio::sync::oneshot::channel();
|
||
let handler_for_task = audio_handler.clone();
|
||
let frames_received_for_task = frames_received.clone();
|
||
tokio::spawn(async move {
|
||
loop {
|
||
tokio::select! {
|
||
_ = &mut shutdown_rx => {
|
||
debug!(target: "chanora_audio", "inbound forwarder shutting down");
|
||
break;
|
||
}
|
||
item = voice_in_rx.recv() => {
|
||
match item {
|
||
Some(v) => {
|
||
let id = SessionAudioId(v.from_client);
|
||
let mut h = handler_for_task.lock().unwrap();
|
||
if let Err(e) = h.handle_packet(id, v.packet) {
|
||
debug!(target: "chanora_audio", error = %e, "decode failed");
|
||
} else {
|
||
frames_received_for_task.fetch_add(1, Ordering::Relaxed);
|
||
}
|
||
}
|
||
None => break,
|
||
}
|
||
}
|
||
}
|
||
}
|
||
});
|
||
|
||
let ptt_watchdog: Option<crate::ptt::MissedKeyUpWatchdog> = None;
|
||
|
||
Ok(Self {
|
||
transmit_gate,
|
||
frames_sent,
|
||
frames_received,
|
||
output_gain,
|
||
output_muted,
|
||
_ios_voice_unit: Mutex::new(Some(ios_voice_unit)),
|
||
shutdown_tx: Some(shutdown_tx),
|
||
capture_active,
|
||
ptt_watchdog,
|
||
})
|
||
}
|
||
|
||
/// Stop the engine. Idempotent.
|
||
pub fn stop(&mut self) {
|
||
if let Some(tx) = self.shutdown_tx.take() {
|
||
let _ = tx.send(());
|
||
}
|
||
// The platform PTT backend is no longer owned by the
|
||
// engine (SDD-088); its lifecycle is managed by
|
||
// `chanora_core::ptt::PttController`. The engine only
|
||
// needs to abort its watchdog and drop the audio streams.
|
||
// Aborting the watchdog cancels its tokio task.
|
||
self.ptt_watchdog.take();
|
||
// Drop the streams, which stops their callback threads.
|
||
// Each platform has a slightly different backend; the
|
||
// common contract is that dropping the wrapper stops
|
||
// audio. iOS collapses input + output into one
|
||
// `IosVoiceUnit` (see `ios_voice_unit.rs`); every other
|
||
// platform has separate cpal input + cpal/SDL output.
|
||
#[cfg(not(target_os = "ios"))]
|
||
{
|
||
let _ = self._input_stream.lock().unwrap().take();
|
||
let _ = self._output_stream.lock().unwrap().take();
|
||
}
|
||
#[cfg(target_os = "ios")]
|
||
{
|
||
let _ = self._ios_voice_unit.lock().unwrap().take();
|
||
}
|
||
// SDD-115 reverse-order teardown on Android:
|
||
// 1) close the voice unit (releases SDD-113 hardware
|
||
// effects then stops + closes the Oboe streams);
|
||
// 2) restore the prior audio mode (SDD-108 §1 on 1 → 0
|
||
// transition);
|
||
// 3) stop the foreground service.
|
||
#[cfg(target_os = "android")]
|
||
{
|
||
if let Some(mut unit) = self._android_voice_unit.lock().unwrap().take() {
|
||
use crate::mobile_voice_backend::MobileVoiceAudioBackend;
|
||
if let Err(e) = unit.close() {
|
||
warn!(
|
||
target: "chanora_audio",
|
||
error = %e,
|
||
"android: AndroidVoiceUnit::close failed (SDD-115)"
|
||
);
|
||
}
|
||
}
|
||
// SDD-108 §1/§2: release the audio-mode stack. Only the
|
||
// 1 → 0 transition writes the platform; mid-stack
|
||
// releases stay engaged. Underflow (release without a
|
||
// matching acquire) is clamped without panic per
|
||
// SDD-108 §1.
|
||
// SDD-108 §5: tolerate a poisoned mutex on the teardown
|
||
// path — if a panicking thread held the lock, we still
|
||
// need to drive the release to completion (otherwise the
|
||
// platform stays in MODE_IN_COMMUNICATION).
|
||
let release = self
|
||
.audio_mode_stack
|
||
.lock()
|
||
.unwrap_or_else(|e| e.into_inner())
|
||
.release();
|
||
match release {
|
||
crate::mode_stack::ModeRelease::LastRelease { prior } => {
|
||
match android_set_audio_mode(prior) {
|
||
Ok(()) => info!(
|
||
target: "chanora_audio",
|
||
restored_mode = prior,
|
||
"android: AudioManager mode restored (SDD-108)"
|
||
),
|
||
Err(e) => warn!(
|
||
target: "chanora_audio",
|
||
error = %e,
|
||
restored_mode = prior,
|
||
"android: failed to restore prior AudioManager mode (SDD-108)"
|
||
),
|
||
}
|
||
}
|
||
crate::mode_stack::ModeRelease::StillHeld => {
|
||
info!(
|
||
target: "chanora_audio",
|
||
"android: audio mode still held by composed session (SDD-108)"
|
||
);
|
||
}
|
||
crate::mode_stack::ModeRelease::AlreadyReleased => {
|
||
// No engage ever happened (e.g. mobile_voice_preset
|
||
// was false, or getMode failed). Silent no-op.
|
||
}
|
||
}
|
||
if crate::android_voice_unit::chanora_android_stop_voice_service() {
|
||
info!(
|
||
target: "chanora_audio",
|
||
"android: voice foreground service stop dispatched (SDD-115)"
|
||
);
|
||
}
|
||
}
|
||
info!(target: "chanora_audio", "audio engine stopped");
|
||
}
|
||
|
||
/// iOS-only: restart the underlying VoiceProcessingIO unit after
|
||
/// route changes.
|
||
pub fn ios_restart_voice_unit(&self) -> Result<(), AudioError> {
|
||
#[cfg(target_os = "ios")]
|
||
{
|
||
let mut guard = self._ios_voice_unit.lock().unwrap();
|
||
let unit = guard
|
||
.as_mut()
|
||
.ok_or_else(|| AudioError::Backend("ios voice unit not running".to_string()))?;
|
||
return unit.restart();
|
||
}
|
||
#[cfg(not(target_os = "ios"))]
|
||
{
|
||
Ok(())
|
||
}
|
||
}
|
||
|
||
/// iOS-only: pause the underlying VoiceProcessingIO unit.
|
||
pub fn ios_pause_voice_unit(&self) -> Result<(), AudioError> {
|
||
#[cfg(target_os = "ios")]
|
||
{
|
||
let mut guard = self._ios_voice_unit.lock().unwrap();
|
||
let unit = guard
|
||
.as_mut()
|
||
.ok_or_else(|| AudioError::Backend("ios voice unit not running".to_string()))?;
|
||
return unit.pause();
|
||
}
|
||
#[cfg(not(target_os = "ios"))]
|
||
{
|
||
Ok(())
|
||
}
|
||
}
|
||
|
||
/// iOS-only: resume the underlying VoiceProcessingIO unit.
|
||
pub fn ios_resume_voice_unit(&self) -> Result<(), AudioError> {
|
||
#[cfg(target_os = "ios")]
|
||
{
|
||
let mut guard = self._ios_voice_unit.lock().unwrap();
|
||
let unit = guard
|
||
.as_mut()
|
||
.ok_or_else(|| AudioError::Backend("ios voice unit not running".to_string()))?;
|
||
return unit.resume();
|
||
}
|
||
#[cfg(not(target_os = "ios"))]
|
||
{
|
||
Ok(())
|
||
}
|
||
}
|
||
|
||
/// Set the **transmission gate** (SRS-201). When true the
|
||
/// encoder feed is allowed to emit Opus frames; when false the
|
||
/// captured audio is discarded before encoding. This is the
|
||
/// only writer permitted on `transmit_active` (SAD-075 /
|
||
/// SDD-089). Push-to-Talk subsystems — focused PTT today,
|
||
/// per-platform global backends in a follow-up — call this
|
||
/// method exclusively. No-op when capture is inactive.
|
||
pub fn set_transmit_active(&self, active: bool) {
|
||
self.transmit_gate.set(active);
|
||
}
|
||
|
||
/// Current transmit gate state.
|
||
pub fn transmit_active(&self) -> bool {
|
||
self.transmit_gate.load()
|
||
}
|
||
|
||
/// Shared handle to the underlying transmit gate (SAD-075 /
|
||
/// SDD-089). Returned for diagnostics and integration tests
|
||
/// only; never mutate the underlying atomic directly — use
|
||
/// [`Self::set_transmit_active`] instead.
|
||
pub fn transmit_gate(&self) -> &crate::ptt::AudioTransmitGate {
|
||
&self.transmit_gate
|
||
}
|
||
|
||
/// Privacy-safe descriptor of the engine's PTT view. The
|
||
/// platform backend lives in `chanora_core::ptt::PttController`
|
||
/// (SDD-088); the engine itself no longer owns it. This getter
|
||
/// always returns the universal Focused fallback descriptor
|
||
/// and is retained only for legacy callers that constructed
|
||
/// engines directly without a controller (tests, headless
|
||
/// diagnostics).
|
||
pub fn ptt_descriptor(&self) -> crate::ptt::PttBackendDescriptor {
|
||
crate::ptt::PttBackendDescriptor::focused()
|
||
}
|
||
|
||
/// Legacy alias for [`Self::set_transmit_active`]. Retained so
|
||
/// the existing bridge `set_ptt` command and the existing
|
||
/// Flutter UI continue to compile during the v0.9.3 PTT
|
||
/// migration (SRS-201 splits the conceptual `ptt` flag into
|
||
/// `transmit_active` / `capture_active`).
|
||
#[doc(hidden)]
|
||
pub fn set_ptt(&self, active: bool) {
|
||
self.set_transmit_active(active);
|
||
}
|
||
|
||
/// Legacy alias for [`Self::transmit_active`].
|
||
#[doc(hidden)]
|
||
pub fn ptt(&self) -> bool {
|
||
self.transmit_active()
|
||
}
|
||
|
||
/// True if the capture stream opened. When false, the engine
|
||
/// runs in playback-only mode and the transmit gate is a
|
||
/// no-op (no frames will ever be encoded).
|
||
pub fn capture_active(&self) -> bool {
|
||
self.capture_active
|
||
}
|
||
|
||
/// Number of Opus frames sent since the engine started.
|
||
pub fn frames_sent(&self) -> u32 {
|
||
self.frames_sent.load(Ordering::Relaxed)
|
||
}
|
||
|
||
/// Number of inbound voice packets received and decoded.
|
||
pub fn frames_received(&self) -> u32 {
|
||
self.frames_received.load(Ordering::Relaxed)
|
||
}
|
||
|
||
/// Latest Android voice-audio diagnostics snapshot (SDD-112 item
|
||
/// 10 / SDD-113 item 7 / SDD-116 item 3). On non-Android targets
|
||
/// this always returns `None`. On Android it returns `Some(...)`
|
||
/// once `AndroidVoiceUnit::open()` has published a snapshot; the
|
||
/// slot is cleared on `close()` / `Drop`. Per SDD-090 the
|
||
/// snapshot contains only device-side technical scalars — no PII.
|
||
pub fn android_diagnostics(
|
||
&self,
|
||
) -> Option<crate::mobile_voice_backend::AndroidAudioDiagnostics> {
|
||
crate::mobile_voice_backend::current_android_audio_diagnostics()
|
||
}
|
||
|
||
/// Set master output mute. When true the output stream emits
|
||
/// silence regardless of incoming voice frames.
|
||
pub fn set_output_muted(&self, muted: bool) {
|
||
self.output_muted.store(muted, Ordering::Relaxed);
|
||
}
|
||
|
||
/// True if the master output is currently muted locally.
|
||
pub fn output_muted(&self) -> bool {
|
||
self.output_muted.load(Ordering::Relaxed)
|
||
}
|
||
|
||
/// Set master output gain. 1.0 is unity; 0.0 is silent. Values
|
||
/// above 1.0 amplify (and may clip downstream). Clamped to a
|
||
/// sensible range internally.
|
||
pub fn set_output_gain(&self, gain: f32) {
|
||
let clamped = gain.clamp(0.0, 4.0);
|
||
self.output_gain.store(clamped.to_bits(), Ordering::Relaxed);
|
||
}
|
||
|
||
/// Current master output gain.
|
||
pub fn output_gain(&self) -> f32 {
|
||
f32::from_bits(self.output_gain.load(Ordering::Relaxed))
|
||
}
|
||
}
|
||
|
||
impl Drop for AudioEngine {
|
||
fn drop(&mut self) {
|
||
self.stop();
|
||
}
|
||
}
|
||
|
||
// ---------- Capture pipeline ----------
|
||
|
||
#[cfg(not(target_os = "ios"))]
|
||
fn try_open_capture(
|
||
in_dev: &cpal::Device,
|
||
voice_out_tx: mpsc::Sender<OutPacket>,
|
||
transmit_active: Arc<AtomicBool>,
|
||
frames_sent: Arc<AtomicU32>,
|
||
mic_gain: f32,
|
||
) -> Result<cpal::Stream, AudioError> {
|
||
let in_cfg = in_dev
|
||
.default_input_config()
|
||
.map_err(|e| AudioError::StreamConfig(format!("input default: {e}")))?;
|
||
let in_sample_rate = in_cfg.sample_rate();
|
||
let in_channels = in_cfg.channels() as usize;
|
||
let in_format = in_cfg.sample_format();
|
||
// Buffer-size rationale (same shape as the output path):
|
||
// * Windows: pin to 2048 frames to avoid the small-period
|
||
// jitter of WASAPI shared mode.
|
||
// * macOS / iOS: CoreAudio picks a HAL-friendly default;
|
||
// iOS RemoteIO rejects arbitrary buffer-size requests.
|
||
// * Linux: same SDL2-vs-cpal split as the output path; we
|
||
// still use cpal for capture but leave Default since
|
||
// PipeWire's ALSA shim works well there.
|
||
let mut in_stream_cfg: cpal::StreamConfig = in_cfg.into();
|
||
#[cfg(target_os = "windows")]
|
||
{
|
||
in_stream_cfg.buffer_size = cpal::BufferSize::Fixed(2048);
|
||
}
|
||
#[cfg(not(target_os = "windows"))]
|
||
{
|
||
in_stream_cfg.buffer_size = cpal::BufferSize::Default;
|
||
}
|
||
|
||
let mut opus_enc = OpusEncoder::new(OpusSampleRate::Hz48000, OpusChannels::Mono, OpusApp::Voip)
|
||
.map_err(|e| AudioError::Opus(format!("encoder new: {e}")))?;
|
||
|
||
// Opus VOIP tuning. Defaults give us 'auto' bitrate (can drop
|
||
// to ~6 kbps during silence \u2014 which sounds garbled when
|
||
// talking resumes) and inband FEC disabled. On lossy mobile
|
||
// networks (cellular / iPhone WiFi roaming), packet loss
|
||
// without FEC produces audible clicks + cut-out frames.
|
||
//
|
||
// Settings derived from the Opus IETF VoIP recommendations
|
||
// (RFC 6716 \u00a7 7.1) and Discord's voice client tuning:
|
||
//
|
||
// * Bitrate 32 kbps : sweet spot for mono voice. Lower
|
||
// than 24 kbps starts to sound watery; higher than
|
||
// 64 kbps wastes bandwidth without perceptual gain on a
|
||
// human voice. Discord uses 64 kbps; mumble defaults to
|
||
// 40 kbps; we pick 32 kbps as a conservative VoIP value
|
||
// that survives 100 kbps uplinks comfortably.
|
||
// * Complexity 10 : max quality. The CPU cost on a modern
|
||
// iPhone (A14+) or any desktop is negligible (~0.5 % of
|
||
// a single core for 48 kHz mono).
|
||
// * Inband FEC on : opus inserts a low-bitrate redundancy
|
||
// copy of the previous frame inside the current packet
|
||
// so a single dropped packet can be reconstructed from
|
||
// the next one. Essential on lossy mobile.
|
||
// * Packet loss perc 5 % : tells the encoder to expect 5 %
|
||
// loss and pre-emptively budget bits for FEC. Higher
|
||
// values trade audio quality for resilience.
|
||
//
|
||
// Errors here are non-fatal: log + continue. The encoder
|
||
// works with defaults if any setter fails on an exotic
|
||
// libopus build.
|
||
if let Err(e) = opus_enc.set_bitrate(OpusBitrate::BitsPerSecond(32_000)) {
|
||
warn!(target: "chanora_audio", error = %e, "opus: set_bitrate(32000) failed");
|
||
}
|
||
if let Err(e) = opus_enc.set_complexity(10) {
|
||
warn!(target: "chanora_audio", error = %e, "opus: set_complexity(10) failed");
|
||
}
|
||
if let Err(e) = opus_enc.set_inband_fec(true) {
|
||
warn!(target: "chanora_audio", error = %e, "opus: set_inband_fec(true) failed");
|
||
}
|
||
if let Err(e) = opus_enc.set_packet_loss_perc(5) {
|
||
warn!(target: "chanora_audio", error = %e, "opus: set_packet_loss_perc(5) failed");
|
||
}
|
||
info!(
|
||
target: "chanora_audio",
|
||
bitrate_bps = 32_000,
|
||
complexity = 10,
|
||
inband_fec = true,
|
||
packet_loss_perc = 5,
|
||
"opus encoder tuned for VoIP"
|
||
);
|
||
|
||
let capture_state = Arc::new(Mutex::new(CaptureState::new(
|
||
opus_enc,
|
||
in_sample_rate,
|
||
in_channels,
|
||
mic_gain,
|
||
voice_out_tx,
|
||
transmit_active,
|
||
frames_sent,
|
||
)));
|
||
|
||
let stream = match in_format {
|
||
SampleFormat::F32 => build_input_stream::<f32>(in_dev, &in_stream_cfg, capture_state)?,
|
||
SampleFormat::I16 => build_input_stream::<i16>(in_dev, &in_stream_cfg, capture_state)?,
|
||
SampleFormat::U16 => build_input_stream::<u16>(in_dev, &in_stream_cfg, capture_state)?,
|
||
other => {
|
||
return Err(AudioError::StreamConfig(format!(
|
||
"unsupported input format: {other:?}"
|
||
)))
|
||
}
|
||
};
|
||
Ok(stream)
|
||
}
|
||
|
||
#[cfg(not(target_os = "ios"))]
|
||
struct CaptureState {
|
||
encoder: OpusEncoder,
|
||
in_sample_rate: u32,
|
||
in_channels: usize,
|
||
mic_gain: f32,
|
||
/// 48 kHz mono buffer accumulated to FRAME_SAMPLES before each encode.
|
||
pcm_accum: Vec<f32>,
|
||
/// Resampling state for non-48k sources (simple linear resampler).
|
||
resample_pos: f64,
|
||
/// Last input sample carried over from the previous cpal callback
|
||
/// so the resampler can interpolate across the buffer boundary
|
||
/// without dropping continuity. Without this, every cpal period
|
||
/// boundary produces a discontinuity → audible buzz / popping
|
||
/// roughly at the period rate (~100 Hz for a 10 ms period on
|
||
/// Linux ALSA defaults).
|
||
resample_last: f32,
|
||
opus_out: [u8; MAX_OPUS_FRAME],
|
||
voice_out_tx: mpsc::Sender<OutPacket>,
|
||
/// The PTT transmission gate. Read once per outbound frame; the
|
||
/// CaptureState never mutates this flag.
|
||
transmit_active: Arc<AtomicBool>,
|
||
frames_sent: Arc<AtomicU32>,
|
||
}
|
||
|
||
#[cfg(not(target_os = "ios"))]
|
||
impl CaptureState {
|
||
fn new(
|
||
encoder: OpusEncoder,
|
||
in_sample_rate: u32,
|
||
in_channels: usize,
|
||
mic_gain: f32,
|
||
voice_out_tx: mpsc::Sender<OutPacket>,
|
||
transmit_active: Arc<AtomicBool>,
|
||
frames_sent: Arc<AtomicU32>,
|
||
) -> Self {
|
||
Self {
|
||
encoder,
|
||
in_sample_rate,
|
||
in_channels,
|
||
mic_gain,
|
||
pcm_accum: Vec::with_capacity(FRAME_SAMPLES * 2),
|
||
resample_pos: 0.0,
|
||
resample_last: 0.0,
|
||
opus_out: [0u8; MAX_OPUS_FRAME],
|
||
voice_out_tx,
|
||
transmit_active,
|
||
frames_sent,
|
||
}
|
||
}
|
||
|
||
/// Consume an arbitrary-rate, multichannel cpal buffer; produce
|
||
/// 48 kHz mono frames; encode and send when `transmit_active`
|
||
/// is true (PTT engaged).
|
||
fn ingest<T: ToF32 + Copy>(&mut self, buf: &[T]) {
|
||
if !self.transmit_active.load(Ordering::Relaxed) {
|
||
// Drain accumulator while muted so we don't pop on PTT release.
|
||
self.pcm_accum.clear();
|
||
return;
|
||
}
|
||
|
||
// 1. Down-mix to mono + gain.
|
||
let mono: Vec<f32> = buf
|
||
.chunks(self.in_channels)
|
||
.map(|frame| {
|
||
let sum: f32 = frame.iter().map(|s| s.to_f32_sample()).sum();
|
||
(sum / frame.len() as f32) * self.mic_gain
|
||
})
|
||
.collect();
|
||
|
||
// 2. Resample to 48 kHz if needed.
|
||
if self.in_sample_rate == SAMPLE_RATE {
|
||
self.pcm_accum.extend_from_slice(&mono);
|
||
} else {
|
||
self.resample_into_accum(&mono);
|
||
}
|
||
|
||
// 3. Encode any complete frames. Clamp each sample to
|
||
// [-1.0, 1.0] before handing to libopus's float encoder —
|
||
// out-of-range samples are hard-clipped inside libopus,
|
||
// which produces audible distortion on transient peaks.
|
||
// Soft-clamping at the engine boundary preserves headroom
|
||
// and matches what every other VoIP client does.
|
||
while self.pcm_accum.len() >= FRAME_SAMPLES {
|
||
let mut frame: Vec<f32> = self.pcm_accum.drain(..FRAME_SAMPLES).collect();
|
||
for s in frame.iter_mut() {
|
||
if *s > 1.0 {
|
||
*s = 1.0;
|
||
} else if *s < -1.0 {
|
||
*s = -1.0;
|
||
}
|
||
}
|
||
match self.encoder.encode_float(&frame, &mut self.opus_out[..]) {
|
||
Ok(len) => {
|
||
let packet = OutAudio::new(&AudioData::C2S {
|
||
id: 0,
|
||
codec: CodecType::OpusVoice,
|
||
data: &self.opus_out[..len],
|
||
});
|
||
match self.voice_out_tx.try_send(packet) {
|
||
Ok(()) => {
|
||
self.frames_sent.fetch_add(1, Ordering::Relaxed);
|
||
}
|
||
Err(mpsc::error::TrySendError::Full(_)) => {
|
||
warn!(target: "chanora_audio", "voice_out queue full; dropping frame");
|
||
}
|
||
Err(mpsc::error::TrySendError::Closed(_)) => {
|
||
warn!(target: "chanora_audio", "voice_out closed; stopping send");
|
||
}
|
||
}
|
||
}
|
||
Err(e) => {
|
||
error!(target: "chanora_audio", error = %e, "opus encode failed");
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
/// Simple linear resampler for `in_sample_rate → 48000`.
|
||
///
|
||
/// The resampler maintains continuity across cpal buffer
|
||
/// boundaries by treating `self.resample_last` as a virtual
|
||
/// sample at fractional index `0.0`, followed by the incoming
|
||
/// `mono` slice at indices `1.0..=mono.len()`. Without the
|
||
/// virtual anchor sample, the first interpolation point of
|
||
/// every new buffer collapses to `mono[0]` for both `a` and
|
||
/// `b`, producing a sample-and-hold step at every cpal period
|
||
/// boundary. On Linux ALSA defaults that's a ~100 Hz buzz /
|
||
/// popping. Production-quality work would use a windowed sinc
|
||
/// kernel; this carries the previous sample only and keeps the
|
||
/// CPU cost trivial.
|
||
fn resample_into_accum(&mut self, mono: &[f32]) {
|
||
if mono.is_empty() {
|
||
return;
|
||
}
|
||
let ratio = self.in_sample_rate as f64 / SAMPLE_RATE as f64;
|
||
let mut pos = self.resample_pos;
|
||
// The virtual buffer has length mono.len() + 1: index 0 is
|
||
// the carried-over last sample, indices 1..=mono.len() are
|
||
// the new buffer. We emit output samples while `pos` is
|
||
// strictly less than mono.len() so we always have a valid
|
||
// right-hand neighbour. The leftover sub-sample offset is
|
||
// carried over via `resample_pos` (rebased to the next
|
||
// buffer's virtual index 0 below).
|
||
while pos < mono.len() as f64 {
|
||
let i = pos.floor() as isize;
|
||
let frac = pos - i as f64;
|
||
let a = if i <= 0 {
|
||
self.resample_last
|
||
} else {
|
||
mono[(i - 1) as usize]
|
||
};
|
||
let b = if i < mono.len() as isize {
|
||
mono[i as usize]
|
||
} else {
|
||
// Should not happen given the while-condition, but
|
||
// guard for the boundary where ratio < 1.0 and `pos`
|
||
// can step past mono.len() in the last iteration.
|
||
a
|
||
};
|
||
self.pcm_accum
|
||
.push((a as f64 + frac * (b - a) as f64) as f32);
|
||
pos += ratio;
|
||
}
|
||
// Carry the leftover sub-sample offset, rebased so the next
|
||
// buffer's virtual index 0 is the new `resample_last`.
|
||
self.resample_pos = pos - mono.len() as f64;
|
||
// Anchor for the next buffer's interpolation.
|
||
self.resample_last = *mono.last().unwrap();
|
||
}
|
||
}
|
||
|
||
/// Per-sample format conversion to f32 in the range [-1.0, 1.0].
|
||
#[cfg(not(target_os = "ios"))]
|
||
trait ToF32 {
|
||
fn to_f32_sample(self) -> f32;
|
||
}
|
||
#[cfg(not(target_os = "ios"))]
|
||
impl ToF32 for f32 {
|
||
fn to_f32_sample(self) -> f32 {
|
||
self
|
||
}
|
||
}
|
||
#[cfg(not(target_os = "ios"))]
|
||
impl ToF32 for i16 {
|
||
fn to_f32_sample(self) -> f32 {
|
||
f32::from(self) / f32::from(i16::MAX)
|
||
}
|
||
}
|
||
#[cfg(not(target_os = "ios"))]
|
||
impl ToF32 for u16 {
|
||
fn to_f32_sample(self) -> f32 {
|
||
(f32::from(self) - f32::from(i16::MAX) - 1.0) / f32::from(i16::MAX)
|
||
}
|
||
}
|
||
|
||
#[cfg(not(target_os = "ios"))]
|
||
fn build_input_stream<T>(
|
||
device: &cpal::Device,
|
||
config: &cpal::StreamConfig,
|
||
state: Arc<Mutex<CaptureState>>,
|
||
) -> Result<cpal::Stream, AudioError>
|
||
where
|
||
T: SizedSample + ToF32 + Send + 'static,
|
||
{
|
||
let stream = device
|
||
.build_input_stream(
|
||
config,
|
||
move |data: &[T], _| {
|
||
let mut s = state.lock().unwrap();
|
||
s.ingest(data);
|
||
},
|
||
move |e| {
|
||
error!(target: "chanora_audio", error = %e, "input stream error");
|
||
},
|
||
None,
|
||
)
|
||
.map_err(|e| AudioError::Backend(format!("build_input_stream: {e}")))?;
|
||
Ok(stream)
|
||
}
|
||
|
||
// ---------- Playback pipeline ----------
|
||
|
||
#[cfg(not(target_os = "linux"))]
|
||
#[cfg(not(target_os = "ios"))]
|
||
fn build_output_stream<T>(
|
||
device: &cpal::Device,
|
||
config: &cpal::StreamConfig,
|
||
handler: Arc<Mutex<AudioHandler<SessionAudioId>>>,
|
||
output_gain: Arc<AtomicU32>,
|
||
output_muted: Arc<AtomicBool>,
|
||
dev_sample_rate: u32,
|
||
dev_channels: usize,
|
||
) -> Result<cpal::Stream, AudioError>
|
||
where
|
||
T: SizedSample + FromF32 + Send + 'static,
|
||
{
|
||
// Per-channel resampler state shared across cpal callbacks for
|
||
// continuity at buffer boundaries. AudioHandler produces 48 kHz
|
||
// stereo f32; we map the first two device channels to L/R and
|
||
// fill any extra channels with silence. `pos` carries the
|
||
// fractional source-sample offset; `last_l` / `last_r` are the
|
||
// anchor samples from the previous callback (avoid the sample-
|
||
// and-hold step that would otherwise pop at every period
|
||
// boundary — same trick as the capture-side resampler).
|
||
let resample_ratio = SAMPLE_RATE as f64 / dev_sample_rate as f64;
|
||
// `same_rate` is the common case where the device is already at
|
||
// 48 kHz — bypass the resampler entirely.
|
||
let same_rate = dev_sample_rate == SAMPLE_RATE;
|
||
let resample_state: Arc<Mutex<PlaybackResampleState>> =
|
||
Arc::new(Mutex::new(PlaybackResampleState {
|
||
pos: 0.0,
|
||
last_l: 0.0,
|
||
last_r: 0.0,
|
||
}));
|
||
// Reusable scratch buffer for 48 kHz stereo samples coming out
|
||
// of AudioHandler. Allocating a fresh `Vec` per cpal callback on
|
||
// glibc malloc was costing measurable time on the realtime audio
|
||
// thread and contributing to the popping users heard. We resize
|
||
// the buffer to the per-callback need and only grow the
|
||
// backing allocation when it must — typical buffer-size jitter
|
||
// stays under the high-water mark and skips the allocator
|
||
// entirely after the first few callbacks.
|
||
let mut scratch: Vec<f32> = Vec::with_capacity(8192);
|
||
// Diagnostic: log a warning if the cpal callback wall-clock
|
||
// exceeds the period budget so we can correlate user-perceived
|
||
// popping with measurable underruns. The threshold is half a
|
||
// period at 48 kHz / 2 ch / 1024-frame typical period — about
|
||
// 10 ms. We rate-limit the warning to once per second.
|
||
let mut last_slow_warn = std::time::Instant::now()
|
||
.checked_sub(std::time::Duration::from_secs(2))
|
||
.unwrap_or_else(std::time::Instant::now);
|
||
let stream = device
|
||
.build_output_stream(
|
||
config,
|
||
move |out: &mut [T], _| {
|
||
let cb_start = std::time::Instant::now();
|
||
let muted = output_muted.load(Ordering::Relaxed);
|
||
let dev_frames = out.len() / dev_channels.max(1);
|
||
let src_frames = if same_rate {
|
||
dev_frames
|
||
} else {
|
||
// Ask for a few extra source frames so we never
|
||
// starve on the resample fractional boundary.
|
||
((dev_frames as f64 * resample_ratio).ceil() as usize) + 2
|
||
};
|
||
let needed = src_frames * 2;
|
||
if scratch.len() < needed {
|
||
scratch.resize(needed, 0.0);
|
||
}
|
||
// Zero the live slice; AudioHandler::fill_buffer
|
||
// writes silence into untouched samples, but
|
||
// resizing up from a smaller call leaves residual
|
||
// values from earlier callbacks. Use `fill` which
|
||
// optimises to memset on f32.
|
||
scratch[..needed].fill(0.0);
|
||
// Lock-and-decode. We deliberately hold the lock
|
||
// only for the duration of fill_buffer; the inbound
|
||
// forwarder uses handle_packet which is queue-fast.
|
||
{
|
||
let mut h = handler.lock().unwrap();
|
||
h.fill_buffer(&mut scratch[..needed]);
|
||
}
|
||
|
||
if muted {
|
||
for dst in out.iter_mut() {
|
||
*dst = T::from_f32_sample(0.0);
|
||
}
|
||
} else {
|
||
let gain = f32::from_bits(output_gain.load(Ordering::Relaxed));
|
||
if same_rate && dev_channels == 2 {
|
||
// Fast path: device is already 48 kHz stereo.
|
||
for (dst, s) in out.iter_mut().zip(scratch[..needed].iter().copied()) {
|
||
*dst = T::from_f32_sample(s * gain);
|
||
}
|
||
} else {
|
||
// Resample 48 kHz stereo → device-rate ×
|
||
// device-channels with continuity across
|
||
// callback boundaries.
|
||
let mut state = resample_state.lock().unwrap();
|
||
let mut pos = state.pos;
|
||
let mut last_l = state.last_l;
|
||
let mut last_r = state.last_r;
|
||
for frame_idx in 0..dev_frames {
|
||
let i = pos.floor() as isize;
|
||
let frac = pos - i as f64;
|
||
let (a_l, a_r) = if i <= 0 {
|
||
(last_l, last_r)
|
||
} else {
|
||
let idx = ((i - 1) as usize) * 2;
|
||
(scratch[idx], scratch[idx + 1])
|
||
};
|
||
let i_usize = i.max(0) as usize;
|
||
let (b_l, b_r) = if i_usize < src_frames {
|
||
let idx = i_usize * 2;
|
||
(scratch[idx], scratch[idx + 1])
|
||
} else {
|
||
(a_l, a_r)
|
||
};
|
||
let l = (a_l as f64 + frac * (b_l - a_l) as f64) as f32 * gain;
|
||
let r = (a_r as f64 + frac * (b_r - a_r) as f64) as f32 * gain;
|
||
let base = frame_idx * dev_channels;
|
||
if dev_channels == 1 {
|
||
// Mono output device (typical on iOS
|
||
// .voiceChat / phone-call audio path):
|
||
// downmix L+R to a single channel
|
||
// rather than dropping the R side.
|
||
// Without the downmix, anything panned
|
||
// right in the AudioHandler stereo mix
|
||
// is silently lost \u2014 on speakerphone
|
||
// this manifested as quiet remote
|
||
// speakers being inaudible.
|
||
out[base] = T::from_f32_sample((l + r) * 0.5);
|
||
} else {
|
||
out[base] = T::from_f32_sample(l);
|
||
if dev_channels >= 2 {
|
||
out[base + 1] = T::from_f32_sample(r);
|
||
}
|
||
for c in 2..dev_channels {
|
||
out[base + c] = T::from_f32_sample(0.0);
|
||
}
|
||
}
|
||
pos += resample_ratio;
|
||
}
|
||
let consumed = pos.floor() as usize;
|
||
state.pos = pos - consumed as f64;
|
||
if consumed > 0 && consumed <= src_frames {
|
||
let idx = (consumed - 1) * 2;
|
||
last_l = scratch[idx];
|
||
last_r = scratch[idx + 1];
|
||
state.last_l = last_l;
|
||
state.last_r = last_r;
|
||
}
|
||
}
|
||
}
|
||
|
||
// Period-budget diagnostic.
|
||
let elapsed = cb_start.elapsed();
|
||
let period_us = (dev_frames as u64 * 1_000_000) / dev_sample_rate as u64;
|
||
if elapsed.as_micros() as u64 > period_us / 2
|
||
&& last_slow_warn.elapsed() > std::time::Duration::from_secs(1)
|
||
{
|
||
last_slow_warn = std::time::Instant::now();
|
||
warn!(
|
||
target: "chanora_audio",
|
||
callback_us = elapsed.as_micros() as u64,
|
||
period_us,
|
||
dev_frames,
|
||
"output callback exceeded half the period budget — possible underrun cause"
|
||
);
|
||
}
|
||
},
|
||
move |e| {
|
||
error!(target: "chanora_audio", error = %e, "output stream error");
|
||
},
|
||
None,
|
||
)
|
||
.map_err(|e| {
|
||
error!(
|
||
target: "chanora_audio",
|
||
error = %e,
|
||
requested_channels = config.channels,
|
||
requested_sample_rate = config.sample_rate,
|
||
"build_output_stream FAILED"
|
||
);
|
||
AudioError::Backend(format!("build_output_stream: {e}"))
|
||
})?;
|
||
Ok(stream)
|
||
}
|
||
|
||
/// Resampler state carried across output cpal callbacks. See
|
||
/// `build_output_stream` for the rationale.
|
||
#[cfg(not(target_os = "linux"))]
|
||
#[cfg(not(target_os = "ios"))]
|
||
struct PlaybackResampleState {
|
||
pos: f64,
|
||
last_l: f32,
|
||
last_r: f32,
|
||
}
|
||
|
||
#[cfg(not(target_os = "linux"))]
|
||
#[cfg(not(target_os = "ios"))]
|
||
trait FromF32 {
|
||
fn from_f32_sample(v: f32) -> Self;
|
||
}
|
||
#[cfg(not(target_os = "linux"))]
|
||
#[cfg(not(target_os = "ios"))]
|
||
impl FromF32 for f32 {
|
||
fn from_f32_sample(v: f32) -> Self {
|
||
v
|
||
}
|
||
}
|
||
#[cfg(not(target_os = "linux"))]
|
||
#[cfg(not(target_os = "ios"))]
|
||
impl FromF32 for i16 {
|
||
fn from_f32_sample(v: f32) -> Self {
|
||
(v.clamp(-1.0, 1.0) * f32::from(i16::MAX)) as i16
|
||
}
|
||
}
|
||
#[cfg(not(target_os = "linux"))]
|
||
#[cfg(not(target_os = "ios"))]
|
||
impl FromF32 for u16 {
|
||
fn from_f32_sample(v: f32) -> Self {
|
||
let s = (v.clamp(-1.0, 1.0) * f32::from(i16::MAX)) as i32;
|
||
(s + i32::from(i16::MAX) + 1) as u16
|
||
}
|
||
}
|
||
|
||
// ---------- Android voice-communication routing ----------
|
||
//
|
||
// SDD-108 §1: `AudioManager.setMode(MODE_IN_COMMUNICATION)` engagement
|
||
// is the routing-level lever that tells Android "this is a voice
|
||
// call, please use the earpiece / engage hardware AEC / NS / AGC
|
||
// where the device supports it". The helpers here are the
|
||
// platform-write surface invoked by the engine's `ModeStack`
|
||
// (`crate::mode_stack`) — the stack owns refcount + snapshot
|
||
// semantics (SDD-108 §2), these helpers only perform the platform
|
||
// write / read.
|
||
//
|
||
// Placement rationale (SDD-108 §4): the JNI helpers for
|
||
// `AudioManager.getMode` / `AudioManager.setMode` live in
|
||
// `chanora_audio::engine` (not `chanora_bridge::android_init`) per
|
||
// SDD-108 §4, which assigns the AudioManager JNI surface to the Rust
|
||
// audio engine. This keeps the AudioManager interaction co-located
|
||
// with the engine state (`ModeStack`) that owns it, so the
|
||
// snapshot/restore lifecycle and the JNI calls evolve together.
|
||
//
|
||
// `AudioManager.MODE_IN_COMMUNICATION == 3` per the Android SDK.
|
||
#[cfg(target_os = "android")]
|
||
pub const ANDROID_MODE_IN_COMMUNICATION: i32 = 3;
|
||
|
||
/// SDD-108 §5: typed error for the Android `AudioManager` JNI surface.
|
||
///
|
||
/// Replaces the previous ad-hoc `Result<_, String>` so callers can
|
||
/// pattern-match on the failure category (attach vs. method call vs.
|
||
/// other) and log/route accordingly. `Display` renders a stable
|
||
/// human-readable form that is safe to feed into the existing
|
||
/// `tracing::warn!(error = %e, ...)` sites.
|
||
#[cfg(target_os = "android")]
|
||
#[derive(Debug, Clone)]
|
||
pub enum AudioModeError {
|
||
/// `JavaVM::from_raw` or `attach_current_thread` failed: the
|
||
/// engine could not reach the JVM at all.
|
||
JniAttachFailed(String),
|
||
/// A specific JNI method call failed (e.g. `getMode`, `setMode`,
|
||
/// `getSystemService`). `method` is a static string for grep-ability.
|
||
MethodCallFailed {
|
||
method: &'static str,
|
||
detail: String,
|
||
},
|
||
/// Catch-all for non-JNI-method failures (null context, panic in
|
||
/// the JNI body, etc.).
|
||
Other(String),
|
||
}
|
||
|
||
#[cfg(target_os = "android")]
|
||
impl std::fmt::Display for AudioModeError {
|
||
fn fmt(&self, f: &mut std::fmt::Formatter<'_>) -> std::fmt::Result {
|
||
match self {
|
||
Self::JniAttachFailed(d) => write!(f, "JNI attach failed: {d}"),
|
||
Self::MethodCallFailed { method, detail } => {
|
||
write!(f, "JNI method `{method}` failed: {detail}")
|
||
}
|
||
Self::Other(d) => write!(f, "{d}"),
|
||
}
|
||
}
|
||
}
|
||
|
||
#[cfg(target_os = "android")]
|
||
impl std::error::Error for AudioModeError {}
|
||
|
||
/// JNI helper shared by `android_get_audio_mode` and
|
||
/// `android_set_audio_mode`: attach to the current thread and return
|
||
/// the `AudioManager` jobject. Centralised so SDD-108's two platform
|
||
/// entry points share one bootstrapping path.
|
||
#[cfg(target_os = "android")]
|
||
fn android_audio_manager_call<F, R>(op: F) -> Result<R, AudioModeError>
|
||
where
|
||
F: FnOnce(&mut jni::JNIEnv, &jni::objects::JObject) -> Result<R, AudioModeError>
|
||
+ std::panic::UnwindSafe,
|
||
{
|
||
use jni::objects::{JObject, JString, JValue};
|
||
|
||
let result = std::panic::catch_unwind(|| -> Result<R, AudioModeError> {
|
||
let ctx = ndk_context::android_context();
|
||
let vm_ptr = ctx.vm();
|
||
if vm_ptr.is_null() {
|
||
return Err(AudioModeError::JniAttachFailed(
|
||
"ndk_context vm is null".to_string(),
|
||
));
|
||
}
|
||
// SAFETY: ndk_context::android_context guarantees `vm` points
|
||
// at a live JavaVM* set by our bridge_init JNI hook. The
|
||
// unsafe block contains only the cast required by
|
||
// `JavaVM::from_raw`.
|
||
let jvm = unsafe { jni::JavaVM::from_raw(vm_ptr as *mut _) }
|
||
.map_err(|e| AudioModeError::JniAttachFailed(format!("jvm from_raw: {e}")))?;
|
||
let mut env = jvm
|
||
.attach_current_thread()
|
||
.map_err(|e| AudioModeError::JniAttachFailed(format!("attach: {e}")))?;
|
||
|
||
let context_obj = unsafe { JObject::from_raw(ctx.context() as jni::sys::jobject) };
|
||
|
||
let service_name: JString = env.new_string("audio").map_err(|e| {
|
||
AudioModeError::MethodCallFailed {
|
||
method: "new_string",
|
||
detail: e.to_string(),
|
||
}
|
||
})?;
|
||
let audio_manager = env
|
||
.call_method(
|
||
&context_obj,
|
||
"getSystemService",
|
||
"(Ljava/lang/String;)Ljava/lang/Object;",
|
||
&[JValue::Object(&service_name.into())],
|
||
)
|
||
.map_err(|e| AudioModeError::MethodCallFailed {
|
||
method: "getSystemService",
|
||
detail: e.to_string(),
|
||
})?
|
||
.l()
|
||
.map_err(|e| AudioModeError::MethodCallFailed {
|
||
method: "getSystemService",
|
||
detail: format!("obj cast: {e}"),
|
||
})?;
|
||
if audio_manager.is_null() {
|
||
return Err(AudioModeError::Other(
|
||
"AudioManager service is null".to_string(),
|
||
));
|
||
}
|
||
op(&mut env, &audio_manager)
|
||
});
|
||
match result {
|
||
Ok(inner) => inner,
|
||
Err(_) => Err(AudioModeError::Other(
|
||
"panic in JNI audio_manager call".to_string(),
|
||
)),
|
||
}
|
||
}
|
||
|
||
/// SDD-108 §1 platform-read: `AudioManager.getMode()`.
|
||
///
|
||
/// Returns the integer mode constant currently active on the system.
|
||
/// Called by the engine on first acquire (0 → 1 transition) so that
|
||
/// `ModeStack` can snapshot the prior mode for restoration on last
|
||
/// release.
|
||
#[cfg(target_os = "android")]
|
||
pub fn android_get_audio_mode() -> Result<i32, AudioModeError> {
|
||
android_audio_manager_call(|env, audio_manager| {
|
||
env.call_method(audio_manager, "getMode", "()I", &[])
|
||
.map_err(|e| AudioModeError::MethodCallFailed {
|
||
method: "getMode",
|
||
detail: e.to_string(),
|
||
})?
|
||
.i()
|
||
.map_err(|e| AudioModeError::MethodCallFailed {
|
||
method: "getMode",
|
||
detail: format!("int cast: {e}"),
|
||
})
|
||
})
|
||
}
|
||
|
||
/// SDD-108 §1 platform-write: `AudioManager.setMode(mode)`.
|
||
///
|
||
/// `mode` is the Android `AudioManager.MODE_*` integer constant.
|
||
/// Use [`ANDROID_MODE_IN_COMMUNICATION`] for engagement; pass back
|
||
/// the snapshotted prior mode (from
|
||
/// [`crate::mode_stack::ModeRelease::LastRelease`]) for restoration.
|
||
#[cfg(target_os = "android")]
|
||
pub fn android_set_audio_mode(mode: i32) -> Result<(), AudioModeError> {
|
||
use jni::objects::JValue;
|
||
android_audio_manager_call(move |env, audio_manager| {
|
||
env.call_method(audio_manager, "setMode", "(I)V", &[JValue::Int(mode)])
|
||
.map_err(|e| AudioModeError::MethodCallFailed {
|
||
method: "setMode",
|
||
detail: e.to_string(),
|
||
})?;
|
||
Ok(())
|
||
})
|
||
}
|