The per-listener noise-reduction toggle (vc_set_remote_stream) did nothing
on Windows/macOS/iOS. The decode loop gated the RNNoise pass on
dec_channels == 1 as a proxy for "this stream is voice" (assuming
stereo => screen-share). The stereo-mic capture commit broke that: a stereo
mic with send-side NR off transmits stereo Opus, so the receiver decoded
two channels and skipped NR entirely. gain/mute have no channel guard, which
is why only NR appeared broken.
Thread the stream kind through init_recv_stream into RemoteStream::is_voice
(set from si.kind() == STREAM_MIC), gate receive NR on is_voice instead of
channel count, and fold a stereo voice frame to mono -> denoise -> duplicate
back across both channels in place (symmetric with the send-side downmix;
RNNoise is mono-only). Screen-audio shares are never denoised.
New test test_recv_noise_reduction drives AudioEngine and asserts a stereo
voice stream's noise floor collapses with NR on (RMS 1046 -> 0.1) while a
screen-audio share stays unchanged. ctest --preset dev green 29/29.
Docs: voice.md section 10. Shared-core fix; clients need only a rebuild.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Both desktop mics were hard-mono: the core defaults capture_channels=1 and
neither client ever called vc_set_capture_channels (only iOS did). Add a
persisted "Stereo microphone" toggle to each client's Audio settings, applied
when the mic stream starts and live via vc_set_capture_channels + vc_audio_restart.
Expose both ABI calls in the Windows interop; the macOS wrapper already had them.
Core fix: encode_and_send_frame now folds a stereo mic frame to mono on a mono
channel - previously the channels==2 branch encoded interleaved L/R directly even
on a mono channel, feeding a mono opus_encode 2x its samples (wrong pitch/garbage).
Real stereo still only reaches the wire on a stereo channel; on a mono channel the
mic is cleanly downmixed.
Test: test_stereo_mic_mono_channel. ctest --preset dev green (28/28). Docs: voice.md.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
Three iOS client problems fixed plus a new core stereo-mic capture ABI:
1. Channel-id sync bug (mic button permanently dimmed): SessionState never
synced currentChannelId from the self user's channelId on connect, so the
mic button (gated on currentChannelId == 0) stayed dimmed. Added
syncSelfChannel() (mirrors macOS MainWindowController.swift:461,491,522);
called from init/.channelList/.userJoined/.userLeft/.userUpdated/.joinResult.
Added applyServerMuteState() + serverMuted/serverDeafened to VoiceState.
2. Join/Leave Voice button: replaced icon-only mic toggle with explicit
text button (parity with macOS). Mute/deafen disable when not in voice.
3. IOSAudioRouter.swift (new): full AVAudioSession routing layer — input
port selection, built-in mic orientation/polar patterns, Bluetooth
HFP/A2DP/Off modes, Standard/Raw mic processing, stereo capture, AirPlay,
UserDefaults persistence. AudioSessionManager delegates to it.
4. Core stereo-mic capture (append-only ABI): vc_set_capture_channels()
lets the core open the mic device in stereo (2-ch interleaved). LocalStream
gains capture_channels; ensure_audio_running reads it; audio_engine.cpp
capture_accum_ + on_capture updated to channel-aware accumulation. Test
test_stereo_mic_capture (headless, L!=R stereo round-trip). Swift wrapper
VoiceCatClient.setCaptureChannels.
5. Settings UI rework: AVAudioSession-derived input/output tree replaces
miniaudio device picker.
6. iOS deployment target raised to 18.0 (Package.swift + project.pbxproj).
swift-tools-version 6.0 with swiftLanguageModes .v5.
Docs: tech-stack.md, architecture.md, voice.md, roadmap.md, building.md
updated; stale 'vc_audio_suspend/resume deferred' claims corrected.
Verified: ctest --preset dev 21/21 green; swift test 6/6 green;
xcodebuild -target VoiceCatiOS -sdk iphonesimulator BUILD SUCCEEDED.
start_loopback_capture hardcoded channels=1, forcing miniaudio to downmix the system's stereo mix to mono before the encoder saw it -- on_capture_frame then upmixed L=R to produce fake stereo. Now the loopback device opens in the channel's mode (stereo when the channel is stereo), CaptureCallback carries an explicit channels param so the encoder gets real interleaved L/R, and a mono fallback covers unusual render endpoints. New test_loopback_stereo_capture asserts L!=R end-to-end; 18/18 ctest green.
RemoteStream::playout_ts was seeded to 0 and only advanced inside the
decode loop (including on every PLC iteration), so it free-ran at ~1x
wall-clock regardless of whether the sender was transmitting. The
sender's frame timestamps only advance while it actually sends (the
VAD/PTT gate returns before ls.timestamp += samples). Across a late join
or any VAD/PTT silence gap the two clocks diverged without bound; once
past the jitter buffer's 500 ms late-drop window every real frame was
dropped-as-late (clock ahead) or never-due (clock behind) -> permanent
silence, while the talk indicator (driven by push_recv_frame, independent
of the jitter buffer) stayed lit.
Add JitterBuffer::peek_front_ts() (try-lock, RT-safe) and seed/re-sync
playout_ts to the earliest buffered frame on the first frame and whenever
it has drifted past +/-200/500 ms. This seeds startup and recovers after
every silence gap.
New regression test test_playout_resync free-runs the clock ~2 s past the
drop window, pushes a ts=0 frame, and asserts audible output: fails
(energy=0) without the fix, passes with it. ctest --preset m1-dev: 14/14.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
AudioEngine::on_capture() was passing miniaudio's hardware callback period
(commonly 480 samples / 10 ms on WASAPI shared mode) directly to opus_encode(),
which requires exactly frame_samples_ (960 for 20 ms @ 48 kHz). The mismatch
returned OPUS_BAD_ARG and silently dropped every real mic frame, while screen
share and injected test frames happened to be correctly sized and worked fine.
Fix: accumulate PCM in a pre-allocated CaptureAccum buffer (mirroring the
existing RemoteStream::ring fix on the playback side) and only call capture_cb_
when a full frame_samples_ chunk is ready. Same pattern applied to on_loopback().
Add test_capture_frame_accumulation() to verify the accumulator fires exactly the
right number of callbacks for misaligned chunk sizes (480, 240+720, 1920 samples).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Closes the three items PROGRESS.md's M3 section explicitly carried forward as
out of scope:
- Device enumeration (vc_list_devices) + input device selection
(vc_set_input_device), backed by AudioEngine::enumerate_devices() via
miniaudio's ma_context_get_devices. Device ids are opaque hex-encoded
ma_device_id strings.
- VAD/PTT send-side input gate (vc_set_input_mode, vc_set_push_to_talk).
webrtc-audio-processing (the originally-planned APM) has no working
Windows/MSVC build upstream (GCC-only Meson, unfinished MinGW support, hard
abseil-cpp dependency), so VAD is a new lightweight, dependency-free
energy/RMS processor (EnergyVadProcessor) behind the existing ApmProcessor
interface. Gating is MIC-only; SCREEN_AUDIO/AUX_DEVICE always bypass it.
- True stereo playback: AudioEngine's mixer and output device now carry
stereo end-to-end (mono streams upmix L=R) instead of downmixing decoded
stereo streams to mono before mixing.
- Real WASAPI loopback capture for SCREEN_AUDIO (Windows-only, via
miniaudio's loopback device type), replacing test-only injection as the
production capture path.
Also: vccli gains --list-devices, --input-device, --input-mode, and
--share-screen-audio flags, plus a stdin command loop (ptt on/off, mode
vad/ptt) for manual verification. New test_vad_ptt_devices.cpp covers all
four items (ABI-level + a white-box AudioEngine stereo-mix check).
Docs updated to match: voice.md, roadmap.md (decision-log entry superseding
the original webrtc-audio-processing choice), tech-stack.md, README.md,
architecture.md, CLAUDE.md, PROGRESS.md.
Still explicitly out of scope, documented not silently dropped: real
webrtc-audio-processing/AEC (no AEC/NS/AGC exists at all yet), macOS/iOS
SCREEN_AUDIO capture, process-specific loopback, and a pre-existing
RT-thread rule violation in the capture path that predates this work.
Verified: ctest 12/12 green across 3 consecutive full-suite runs (both dev
and m1-dev presets build clean); test_vad_ptt_devices passed 5 consecutive
standalone runs; manually verified live (vccli --list-devices against real
hardware, vccli --voice --input-mode vad streaming without incident).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
