feat(M2): UDP voice/media plane -- SFU relay, Opus, AEAD, jitter buffer
Adds the full voice pipeline: 14-byte binary frame header, ChaCha20-Poly1305
AEAD keyed from the TLS exporter, libopus encode/decode with FEC/PLC/DTX,
an adaptive per-ssrc jitter buffer, a miniaudio capture/playback engine, an
APM passthrough stub, and the UdpBinding/StreamAnnounce signaling chain
wired through ConnSession/SessionRegistry into a new server-side SFU
(MediaRelay) that decrypts and re-encrypts frames per channel member.
Exit criterion verified: test_m2_voice — two headless clients relay 50
encrypted Opus frames through the server; ctest --preset m1-dev is 9/9
green. Also corrects protocol.md's UdpBinding diagram, which described the
UDP-side binding packet as AEAD-sealed when it is in fact a plaintext
bootstrap frame (separate from the TCP/TLS UdpBinding ack).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 01:31:14 +02:00
|
|
|
#include "audio/apm_processor.h"
|
|
|
|
|
|
feat(audio): real noise suppression via vendored RNNoise (send + receive)
The two-sided NR plumbing (RemoteStream::recv_ns + the per-listener
vc_set_remote_stream noise_reduction toggle) was wired but inert:
ApmProcessor::create() returned a no-op passthrough, because the
originally-planned webrtc-audio-processing has no working Windows/macOS
build. Drop in RNNoise as the real backend behind the same ApmProcessor
interface, lighting up both NR paths.
- Vendor RNNoise (BSD-3 + CC0) at third_party/rnnoise/ — the vcpkg port
is !windows !arm, so it can't cover our primary targets. Shrunk int8
model (78MB -> 11.7MB via upstream scripts/shrink_model.sh), built as a
standalone C static lib with no RTCD (portable scalar path on x86,
auto-NEON on arm64) under -DDISABLE_DEBUG_FLOAT. Model is baked in
(rnnoise_create(NULL)); no runtime file.
- New RnnoiseProcessor (core/src/audio/apm_processor.cpp) selected by
ApmProcessor::create() when VOICECAT_HAS_NS. Mono/48kHz/480-sample;
our clock is fixed 48kHz and Opus frame sizes are multiples of 480, so
no resampling. RT-safe: allocates at construction, lock-free in the
capture/playback callbacks.
- Receive-side: lit up via the factory; gated to mono streams (a stereo
stream is a screen-audio share, not voice).
- Send-side (new): vc_set_input_noise_reduction(client, enable) ABI +
vc_client::mic_ns_, run before input gain/VAD in on_capture_frame. A
stereo mic is downmixed to mono ONLY when NR is on — with NR off a
stereo mic keeps full stereo (never collapse mic quality unasked).
- Enable C as a project language for the vendored lib.
- New noise_suppression test: white noise through ApmProcessor::create()
drops ~99.9% RMS. ctest --preset dev green, 28/28. windows-client DLL
builds clean with vc_set_input_noise_reduction exported, system-only deps.
- Docs synced: voice.md §10, tech-stack.md §1/§5, third_party/README.md,
vcpkg.json note, PROGRESS.md, CLAUDE.md.
Client on/off UI toggles (Windows/macOS/iOS) are the remaining follow-up.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-23 13:30:54 +02:00
|
|
|
#include <algorithm>
|
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode
Core ABI extensions (voicecat.h):
- vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters
for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads
- VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password
- VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_
until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519)
- vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only
- VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate
- vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores
it atomically so the audio RT path reads without a lock
C++ implementation:
- SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id,
password_protected, and max_users (were permanently zeroed)
- TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS
- TofuStore split into peek (read-only) + pin (write) so first-connect only persists
after user approval; tofu_store_path in vc_config for per-user pin file location
- TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows)
- windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests
- New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green)
Windows client (clients/windows/ — .NET 10 WinForms):
- VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks,
Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer
- VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity-
Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user
ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode,
per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar)
- PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation)
- PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams
- Accessibility: explicit AccessibleName/Description on every control, & mnemonics,
Activity log ListBox as durable screen-reader record, AutomationNotification for
curated live announcements
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-17 00:35:16 +02:00
|
|
|
#include <atomic>
|
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback
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>
2026-06-16 16:11:52 +02:00
|
|
|
#include <chrono>
|
|
|
|
|
#include <cmath>
|
|
|
|
|
|
feat(audio): real noise suppression via vendored RNNoise (send + receive)
The two-sided NR plumbing (RemoteStream::recv_ns + the per-listener
vc_set_remote_stream noise_reduction toggle) was wired but inert:
ApmProcessor::create() returned a no-op passthrough, because the
originally-planned webrtc-audio-processing has no working Windows/macOS
build. Drop in RNNoise as the real backend behind the same ApmProcessor
interface, lighting up both NR paths.
- Vendor RNNoise (BSD-3 + CC0) at third_party/rnnoise/ — the vcpkg port
is !windows !arm, so it can't cover our primary targets. Shrunk int8
model (78MB -> 11.7MB via upstream scripts/shrink_model.sh), built as a
standalone C static lib with no RTCD (portable scalar path on x86,
auto-NEON on arm64) under -DDISABLE_DEBUG_FLOAT. Model is baked in
(rnnoise_create(NULL)); no runtime file.
- New RnnoiseProcessor (core/src/audio/apm_processor.cpp) selected by
ApmProcessor::create() when VOICECAT_HAS_NS. Mono/48kHz/480-sample;
our clock is fixed 48kHz and Opus frame sizes are multiples of 480, so
no resampling. RT-safe: allocates at construction, lock-free in the
capture/playback callbacks.
- Receive-side: lit up via the factory; gated to mono streams (a stereo
stream is a screen-audio share, not voice).
- Send-side (new): vc_set_input_noise_reduction(client, enable) ABI +
vc_client::mic_ns_, run before input gain/VAD in on_capture_frame. A
stereo mic is downmixed to mono ONLY when NR is on — with NR off a
stereo mic keeps full stereo (never collapse mic quality unasked).
- Enable C as a project language for the vendored lib.
- New noise_suppression test: white noise through ApmProcessor::create()
drops ~99.9% RMS. ctest --preset dev green, 28/28. windows-client DLL
builds clean with vc_set_input_noise_reduction exported, system-only deps.
- Docs synced: voice.md §10, tech-stack.md §1/§5, third_party/README.md,
vcpkg.json note, PROGRESS.md, CLAUDE.md.
Client on/off UI toggles (Windows/macOS/iOS) are the remaining follow-up.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-23 13:30:54 +02:00
|
|
|
#ifdef VOICECAT_HAS_NS
|
|
|
|
|
#include "rnnoise.h"
|
|
|
|
|
#endif
|
|
|
|
|
|
feat(M2): UDP voice/media plane -- SFU relay, Opus, AEAD, jitter buffer
Adds the full voice pipeline: 14-byte binary frame header, ChaCha20-Poly1305
AEAD keyed from the TLS exporter, libopus encode/decode with FEC/PLC/DTX,
an adaptive per-ssrc jitter buffer, a miniaudio capture/playback engine, an
APM passthrough stub, and the UdpBinding/StreamAnnounce signaling chain
wired through ConnSession/SessionRegistry into a new server-side SFU
(MediaRelay) that decrypts and re-encrypts frames per channel member.
Exit criterion verified: test_m2_voice — two headless clients relay 50
encrypted Opus frames through the server; ctest --preset m1-dev is 9/9
green. Also corrects protocol.md's UdpBinding diagram, which described the
UDP-side binding packet as AEAD-sealed when it is in fact a plaintext
bootstrap frame (separate from the TCP/TLS UdpBinding ack).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 01:31:14 +02:00
|
|
|
namespace voicecat::audio {
|
|
|
|
|
|
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback
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>
2026-06-16 16:11:52 +02:00
|
|
|
namespace {
|
|
|
|
|
int64_t steady_now_ms() {
|
|
|
|
|
return std::chrono::duration_cast<std::chrono::milliseconds>(
|
|
|
|
|
std::chrono::steady_clock::now().time_since_epoch())
|
|
|
|
|
.count();
|
|
|
|
|
}
|
|
|
|
|
} // namespace
|
|
|
|
|
|
feat(M2): UDP voice/media plane -- SFU relay, Opus, AEAD, jitter buffer
Adds the full voice pipeline: 14-byte binary frame header, ChaCha20-Poly1305
AEAD keyed from the TLS exporter, libopus encode/decode with FEC/PLC/DTX,
an adaptive per-ssrc jitter buffer, a miniaudio capture/playback engine, an
APM passthrough stub, and the UdpBinding/StreamAnnounce signaling chain
wired through ConnSession/SessionRegistry into a new server-side SFU
(MediaRelay) that decrypts and re-encrypts frames per channel member.
Exit criterion verified: test_m2_voice — two headless clients relay 50
encrypted Opus frames through the server; ctest --preset m1-dev is 9/9
green. Also corrects protocol.md's UdpBinding diagram, which described the
UDP-side binding packet as AEAD-sealed when it is in fact a plaintext
bootstrap frame (separate from the TCP/TLS UdpBinding ack).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 01:31:14 +02:00
|
|
|
// ── ApmPassthrough ────────────────────────────────────────────────────────────
|
|
|
|
|
// No-op: returns true (VAD always open), does not modify PCM.
|
|
|
|
|
// Replaced by WebrtcApmProcessor when VOICECAT_HAS_APM is defined.
|
|
|
|
|
class ApmPassthrough final : public ApmProcessor {
|
|
|
|
|
public:
|
|
|
|
|
void process_render(const int16_t*, int, int) override {}
|
|
|
|
|
bool process_capture(int16_t*, int, int) override { return true; }
|
|
|
|
|
};
|
|
|
|
|
|
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback
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>
2026-06-16 16:11:52 +02:00
|
|
|
// ── EnergyVadProcessor ──────────────────────────────────────────────────────
|
|
|
|
|
// Lightweight, dependency-free energy/RMS VAD — see apm_processor.h's create_vad() doc comment
|
|
|
|
|
// for why this exists instead of a real APM. No AEC (process_render is a no-op); doesn't modify
|
|
|
|
|
// the PCM it's given, only inspects it.
|
|
|
|
|
class EnergyVadProcessor final : public ApmProcessor {
|
|
|
|
|
public:
|
|
|
|
|
EnergyVadProcessor(float rms_threshold, int64_t hang_time_ms)
|
|
|
|
|
: threshold_(rms_threshold), hang_time_ms_(hang_time_ms) {}
|
|
|
|
|
|
|
|
|
|
void process_render(const int16_t*, int, int) override {}
|
|
|
|
|
|
|
|
|
|
bool process_capture(int16_t* pcm, int samples, int /*sample_rate*/) override {
|
|
|
|
|
if (samples > 0) {
|
|
|
|
|
double sum_sq = 0.0;
|
|
|
|
|
for (int i = 0; i < samples; ++i) {
|
|
|
|
|
double s = static_cast<double>(pcm[i]) / 32768.0;
|
|
|
|
|
sum_sq += s * s;
|
|
|
|
|
}
|
|
|
|
|
double rms = std::sqrt(sum_sq / samples);
|
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode
Core ABI extensions (voicecat.h):
- vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters
for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads
- VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password
- VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_
until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519)
- vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only
- VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate
- vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores
it atomically so the audio RT path reads without a lock
C++ implementation:
- SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id,
password_protected, and max_users (were permanently zeroed)
- TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS
- TofuStore split into peek (read-only) + pin (write) so first-connect only persists
after user approval; tofu_store_path in vc_config for per-user pin file location
- TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows)
- windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests
- New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green)
Windows client (clients/windows/ — .NET 10 WinForms):
- VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks,
Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer
- VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity-
Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user
ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode,
per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar)
- PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation)
- PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams
- Accessibility: explicit AccessibleName/Description on every control, & mnemonics,
Activity log ListBox as durable screen-reader record, AutomationNotification for
curated live announcements
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-17 00:35:16 +02:00
|
|
|
if (rms >= threshold_.load(std::memory_order_relaxed)) last_voice_ms_ = steady_now_ms();
|
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback
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>
2026-06-16 16:11:52 +02:00
|
|
|
}
|
|
|
|
|
return (steady_now_ms() - last_voice_ms_) < hang_time_ms_;
|
|
|
|
|
}
|
|
|
|
|
|
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode
Core ABI extensions (voicecat.h):
- vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters
for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads
- VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password
- VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_
until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519)
- vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only
- VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate
- vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores
it atomically so the audio RT path reads without a lock
C++ implementation:
- SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id,
password_protected, and max_users (were permanently zeroed)
- TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS
- TofuStore split into peek (read-only) + pin (write) so first-connect only persists
after user approval; tofu_store_path in vc_config for per-user pin file location
- TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows)
- windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests
- New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green)
Windows client (clients/windows/ — .NET 10 WinForms):
- VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks,
Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer
- VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity-
Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user
ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode,
per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar)
- PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation)
- PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams
- Accessibility: explicit AccessibleName/Description on every control, & mnemonics,
Activity log ListBox as durable screen-reader record, AutomationNotification for
curated live announcements
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-17 00:35:16 +02:00
|
|
|
void set_threshold(float t) override {
|
|
|
|
|
threshold_.store(t, std::memory_order_relaxed);
|
|
|
|
|
}
|
|
|
|
|
|
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback
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>
2026-06-16 16:11:52 +02:00
|
|
|
private:
|
feat(M4): Windows WinForms client, TOFU identity pinning, VAD threshold + always-on mode
Core ABI extensions (voicecat.h):
- vc_list_channels / vc_list_users / vc_list_user_streams — pull-based snapshot getters
for the channel-tree and user-list UI; session_model_mu_ guards cross-thread reads
- VC_EVENT_JOIN_RESULT / vc_join_channel — channel join with optional password
- VC_EVENT_SERVER_IDENTITY + vc_confirm_server_identity — TOFU gate that blocks io_thread_
until the UI approves or rejects; pins TLS leaf-cert SHA-256 (not declared Ed25519)
- vc_get_server_identity_display — Ed25519 fingerprint for human-readable display only
- VC_INPUT_ALWAYS_ON = 2 in vc_input_mode — transmit unconditionally, no VAD gate
- vc_set_vad_threshold — live RMS threshold update (0.0–1.0); EnergyVadProcessor stores
it atomically so the audio RT path reads without a lock
C++ implementation:
- SessionModel::apply_snapshot / apply_channel_event fixed to populate parent_id,
password_protected, and max_users (were permanently zeroed)
- TlsContext::peer_cert_fingerprint — SHA-256 of peer leaf cert DER via mbedTLS
- TofuStore split into peek (read-only) + pin (write) so first-connect only persists
after user approval; tofu_store_path in vc_config for per-user pin file location
- TcpAcceptor uses dual-stack IPv6+IPv4 fallback (fixes localhost → ::1 on Windows)
- windows-client CMake preset: Release shared DLL, static MinGW runtime, no tools/tests
- New C++ tests: test_channel_user_list_abi, test_tofu_flow (14/14 green)
Windows client (clients/windows/ — .NET 10 WinForms):
- VoiceCat.Interop: LibraryImport P/Invoke surface, UnmanagedCallersOnly callbacks,
Channel<VoiceCatEvent> event delivery drained by 30ms WinForms Timer
- VoiceCat.App: ConnectDialog (saved servers, DPAPI password storage), ServerIdentity-
Dialog (TOFU first-connect / mismatch warning), MainForm (channel TreeView, user
ListBox, RichTextBox chat, voice controls, device pickers, VAD/PTT/always-on mode,
per-user gain/mute/NR tuning, VAD sensitivity TrackBar, level meter ProgressBar)
- PttKeyCaptureDialog — focus-scoped PTT key capture (documented limitation)
- PerUserTuningDialog — real-time gain/mute/NR applied to all of a user's streams
- Accessibility: explicit AccessibleName/Description on every control, & mnemonics,
Activity log ListBox as durable screen-reader record, AutomationNotification for
curated live announcements
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-17 00:35:16 +02:00
|
|
|
std::atomic<float> threshold_;
|
|
|
|
|
int64_t hang_time_ms_;
|
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback
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>
2026-06-16 16:11:52 +02:00
|
|
|
int64_t last_voice_ms_ = 0; // epoch start -> gate begins closed until first loud frame
|
|
|
|
|
};
|
|
|
|
|
|
feat(audio): real noise suppression via vendored RNNoise (send + receive)
The two-sided NR plumbing (RemoteStream::recv_ns + the per-listener
vc_set_remote_stream noise_reduction toggle) was wired but inert:
ApmProcessor::create() returned a no-op passthrough, because the
originally-planned webrtc-audio-processing has no working Windows/macOS
build. Drop in RNNoise as the real backend behind the same ApmProcessor
interface, lighting up both NR paths.
- Vendor RNNoise (BSD-3 + CC0) at third_party/rnnoise/ — the vcpkg port
is !windows !arm, so it can't cover our primary targets. Shrunk int8
model (78MB -> 11.7MB via upstream scripts/shrink_model.sh), built as a
standalone C static lib with no RTCD (portable scalar path on x86,
auto-NEON on arm64) under -DDISABLE_DEBUG_FLOAT. Model is baked in
(rnnoise_create(NULL)); no runtime file.
- New RnnoiseProcessor (core/src/audio/apm_processor.cpp) selected by
ApmProcessor::create() when VOICECAT_HAS_NS. Mono/48kHz/480-sample;
our clock is fixed 48kHz and Opus frame sizes are multiples of 480, so
no resampling. RT-safe: allocates at construction, lock-free in the
capture/playback callbacks.
- Receive-side: lit up via the factory; gated to mono streams (a stereo
stream is a screen-audio share, not voice).
- Send-side (new): vc_set_input_noise_reduction(client, enable) ABI +
vc_client::mic_ns_, run before input gain/VAD in on_capture_frame. A
stereo mic is downmixed to mono ONLY when NR is on — with NR off a
stereo mic keeps full stereo (never collapse mic quality unasked).
- Enable C as a project language for the vendored lib.
- New noise_suppression test: white noise through ApmProcessor::create()
drops ~99.9% RMS. ctest --preset dev green, 28/28. windows-client DLL
builds clean with vc_set_input_noise_reduction exported, system-only deps.
- Docs synced: voice.md §10, tech-stack.md §1/§5, third_party/README.md,
vcpkg.json note, PROGRESS.md, CLAUDE.md.
Client on/off UI toggles (Windows/macOS/iOS) are the remaining follow-up.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-23 13:30:54 +02:00
|
|
|
#ifdef VOICECAT_HAS_NS
|
|
|
|
|
// ── RnnoiseProcessor ─────────────────────────────────────────────────────────
|
|
|
|
|
// Real noise suppression via vendored RNNoise (third_party/rnnoise; docs/voice.md §10-11).
|
|
|
|
|
// RNNoise is a mono, 48 kHz, fixed 480-sample (10 ms) speech denoiser; our engine clock is fixed
|
|
|
|
|
// at 48 kHz and every Opus frame size (480/960/1920/2880) is a multiple of 480, so we process
|
|
|
|
|
// whole 480-sample chunks with no resampling and no cross-call carry. Mono only — callers gate
|
|
|
|
|
// on a single channel (a stereo screen-audio share is never voice and isn't denoised).
|
|
|
|
|
//
|
|
|
|
|
// RT-safety (docs/architecture.md §3): the DenoiseState and the float scratch are allocated in the
|
|
|
|
|
// ctor; process_capture() does no allocation/locking. The state is owned per-stream (recv) or
|
|
|
|
|
// per-mic (send) so it persists across calls, which is exactly what RNNoise's overlap needs.
|
|
|
|
|
class RnnoiseProcessor final : public ApmProcessor {
|
|
|
|
|
public:
|
|
|
|
|
RnnoiseProcessor() : st_(rnnoise_create(nullptr)) {}
|
|
|
|
|
~RnnoiseProcessor() override {
|
|
|
|
|
if (st_) rnnoise_destroy(st_);
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
void process_render(const int16_t*, int, int) override {} // NS needs no AEC reference
|
|
|
|
|
|
|
|
|
|
bool process_capture(int16_t* pcm, int samples, int sample_rate) override {
|
|
|
|
|
// RNNoise is 48 kHz only; anything else passes through untouched (our clock is 48 kHz, so
|
|
|
|
|
// this guard never trips in practice — it's just a correctness backstop).
|
|
|
|
|
if (!st_ || sample_rate != 48000) return true;
|
|
|
|
|
for (int off = 0; off + kFrame <= samples; off += kFrame) {
|
|
|
|
|
for (int i = 0; i < kFrame; ++i) in_[i] = static_cast<float>(pcm[off + i]);
|
|
|
|
|
rnnoise_process_frame(st_, out_, in_);
|
|
|
|
|
for (int i = 0; i < kFrame; ++i) {
|
|
|
|
|
int32_t v = static_cast<int32_t>(std::lround(out_[i]));
|
|
|
|
|
pcm[off + i] = static_cast<int16_t>(std::clamp(v, -32768, 32767));
|
|
|
|
|
}
|
|
|
|
|
}
|
|
|
|
|
return true; // NS doesn't gate; the send path's VAD stays a separate stage
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
private:
|
|
|
|
|
static constexpr int kFrame = 480; // rnnoise_get_frame_size()
|
|
|
|
|
DenoiseState* st_;
|
|
|
|
|
float in_[kFrame];
|
|
|
|
|
float out_[kFrame];
|
|
|
|
|
};
|
|
|
|
|
#endif // VOICECAT_HAS_NS
|
|
|
|
|
|
feat(M2): UDP voice/media plane -- SFU relay, Opus, AEAD, jitter buffer
Adds the full voice pipeline: 14-byte binary frame header, ChaCha20-Poly1305
AEAD keyed from the TLS exporter, libopus encode/decode with FEC/PLC/DTX,
an adaptive per-ssrc jitter buffer, a miniaudio capture/playback engine, an
APM passthrough stub, and the UdpBinding/StreamAnnounce signaling chain
wired through ConnSession/SessionRegistry into a new server-side SFU
(MediaRelay) that decrypts and re-encrypts frames per channel member.
Exit criterion verified: test_m2_voice — two headless clients relay 50
encrypted Opus frames through the server; ctest --preset m1-dev is 9/9
green. Also corrects protocol.md's UdpBinding diagram, which described the
UDP-side binding packet as AEAD-sealed when it is in fact a plaintext
bootstrap frame (separate from the TCP/TLS UdpBinding ack).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 01:31:14 +02:00
|
|
|
std::unique_ptr<ApmProcessor> ApmProcessor::create() {
|
feat(audio): real noise suppression via vendored RNNoise (send + receive)
The two-sided NR plumbing (RemoteStream::recv_ns + the per-listener
vc_set_remote_stream noise_reduction toggle) was wired but inert:
ApmProcessor::create() returned a no-op passthrough, because the
originally-planned webrtc-audio-processing has no working Windows/macOS
build. Drop in RNNoise as the real backend behind the same ApmProcessor
interface, lighting up both NR paths.
- Vendor RNNoise (BSD-3 + CC0) at third_party/rnnoise/ — the vcpkg port
is !windows !arm, so it can't cover our primary targets. Shrunk int8
model (78MB -> 11.7MB via upstream scripts/shrink_model.sh), built as a
standalone C static lib with no RTCD (portable scalar path on x86,
auto-NEON on arm64) under -DDISABLE_DEBUG_FLOAT. Model is baked in
(rnnoise_create(NULL)); no runtime file.
- New RnnoiseProcessor (core/src/audio/apm_processor.cpp) selected by
ApmProcessor::create() when VOICECAT_HAS_NS. Mono/48kHz/480-sample;
our clock is fixed 48kHz and Opus frame sizes are multiples of 480, so
no resampling. RT-safe: allocates at construction, lock-free in the
capture/playback callbacks.
- Receive-side: lit up via the factory; gated to mono streams (a stereo
stream is a screen-audio share, not voice).
- Send-side (new): vc_set_input_noise_reduction(client, enable) ABI +
vc_client::mic_ns_, run before input gain/VAD in on_capture_frame. A
stereo mic is downmixed to mono ONLY when NR is on — with NR off a
stereo mic keeps full stereo (never collapse mic quality unasked).
- Enable C as a project language for the vendored lib.
- New noise_suppression test: white noise through ApmProcessor::create()
drops ~99.9% RMS. ctest --preset dev green, 28/28. windows-client DLL
builds clean with vc_set_input_noise_reduction exported, system-only deps.
- Docs synced: voice.md §10, tech-stack.md §1/§5, third_party/README.md,
vcpkg.json note, PROGRESS.md, CLAUDE.md.
Client on/off UI toggles (Windows/macOS/iOS) are the remaining follow-up.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-23 13:30:54 +02:00
|
|
|
#ifdef VOICECAT_HAS_NS
|
|
|
|
|
return std::make_unique<RnnoiseProcessor>();
|
|
|
|
|
#else
|
feat(M2): UDP voice/media plane -- SFU relay, Opus, AEAD, jitter buffer
Adds the full voice pipeline: 14-byte binary frame header, ChaCha20-Poly1305
AEAD keyed from the TLS exporter, libopus encode/decode with FEC/PLC/DTX,
an adaptive per-ssrc jitter buffer, a miniaudio capture/playback engine, an
APM passthrough stub, and the UdpBinding/StreamAnnounce signaling chain
wired through ConnSession/SessionRegistry into a new server-side SFU
(MediaRelay) that decrypts and re-encrypts frames per channel member.
Exit criterion verified: test_m2_voice — two headless clients relay 50
encrypted Opus frames through the server; ctest --preset m1-dev is 9/9
green. Also corrects protocol.md's UdpBinding diagram, which described the
UDP-side binding packet as AEAD-sealed when it is in fact a plaintext
bootstrap frame (separate from the TCP/TLS UdpBinding ack).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 01:31:14 +02:00
|
|
|
return std::make_unique<ApmPassthrough>();
|
feat(audio): real noise suppression via vendored RNNoise (send + receive)
The two-sided NR plumbing (RemoteStream::recv_ns + the per-listener
vc_set_remote_stream noise_reduction toggle) was wired but inert:
ApmProcessor::create() returned a no-op passthrough, because the
originally-planned webrtc-audio-processing has no working Windows/macOS
build. Drop in RNNoise as the real backend behind the same ApmProcessor
interface, lighting up both NR paths.
- Vendor RNNoise (BSD-3 + CC0) at third_party/rnnoise/ — the vcpkg port
is !windows !arm, so it can't cover our primary targets. Shrunk int8
model (78MB -> 11.7MB via upstream scripts/shrink_model.sh), built as a
standalone C static lib with no RTCD (portable scalar path on x86,
auto-NEON on arm64) under -DDISABLE_DEBUG_FLOAT. Model is baked in
(rnnoise_create(NULL)); no runtime file.
- New RnnoiseProcessor (core/src/audio/apm_processor.cpp) selected by
ApmProcessor::create() when VOICECAT_HAS_NS. Mono/48kHz/480-sample;
our clock is fixed 48kHz and Opus frame sizes are multiples of 480, so
no resampling. RT-safe: allocates at construction, lock-free in the
capture/playback callbacks.
- Receive-side: lit up via the factory; gated to mono streams (a stereo
stream is a screen-audio share, not voice).
- Send-side (new): vc_set_input_noise_reduction(client, enable) ABI +
vc_client::mic_ns_, run before input gain/VAD in on_capture_frame. A
stereo mic is downmixed to mono ONLY when NR is on — with NR off a
stereo mic keeps full stereo (never collapse mic quality unasked).
- Enable C as a project language for the vendored lib.
- New noise_suppression test: white noise through ApmProcessor::create()
drops ~99.9% RMS. ctest --preset dev green, 28/28. windows-client DLL
builds clean with vc_set_input_noise_reduction exported, system-only deps.
- Docs synced: voice.md §10, tech-stack.md §1/§5, third_party/README.md,
vcpkg.json note, PROGRESS.md, CLAUDE.md.
Client on/off UI toggles (Windows/macOS/iOS) are the remaining follow-up.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
2026-06-23 13:30:54 +02:00
|
|
|
#endif
|
feat(M2): UDP voice/media plane -- SFU relay, Opus, AEAD, jitter buffer
Adds the full voice pipeline: 14-byte binary frame header, ChaCha20-Poly1305
AEAD keyed from the TLS exporter, libopus encode/decode with FEC/PLC/DTX,
an adaptive per-ssrc jitter buffer, a miniaudio capture/playback engine, an
APM passthrough stub, and the UdpBinding/StreamAnnounce signaling chain
wired through ConnSession/SessionRegistry into a new server-side SFU
(MediaRelay) that decrypts and re-encrypts frames per channel member.
Exit criterion verified: test_m2_voice — two headless clients relay 50
encrypted Opus frames through the server; ctest --preset m1-dev is 9/9
green. Also corrects protocol.md's UdpBinding diagram, which described the
UDP-side binding packet as AEAD-sealed when it is in fact a plaintext
bootstrap frame (separate from the TCP/TLS UdpBinding ack).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 01:31:14 +02:00
|
|
|
}
|
|
|
|
|
|
feat: device enumeration, VAD/PTT input gate, stereo playback, WASAPI loopback
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>
2026-06-16 16:11:52 +02:00
|
|
|
std::unique_ptr<ApmProcessor> ApmProcessor::create_vad(float rms_threshold,
|
|
|
|
|
int64_t hang_time_ms) {
|
|
|
|
|
return std::make_unique<EnergyVadProcessor>(rms_threshold, hang_time_ms);
|
|
|
|
|
}
|
|
|
|
|
|
feat(M2): UDP voice/media plane -- SFU relay, Opus, AEAD, jitter buffer
Adds the full voice pipeline: 14-byte binary frame header, ChaCha20-Poly1305
AEAD keyed from the TLS exporter, libopus encode/decode with FEC/PLC/DTX,
an adaptive per-ssrc jitter buffer, a miniaudio capture/playback engine, an
APM passthrough stub, and the UdpBinding/StreamAnnounce signaling chain
wired through ConnSession/SessionRegistry into a new server-side SFU
(MediaRelay) that decrypts and re-encrypts frames per channel member.
Exit criterion verified: test_m2_voice — two headless clients relay 50
encrypted Opus frames through the server; ctest --preset m1-dev is 9/9
green. Also corrects protocol.md's UdpBinding diagram, which described the
UDP-side binding packet as AEAD-sealed when it is in fact a plaintext
bootstrap frame (separate from the TCP/TLS UdpBinding ack).
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
2026-06-16 01:31:14 +02:00
|
|
|
} // namespace voicecat::audio
|