feat(M3): multi-stream & per-channel tuning

Implements docs/roadmap.md M3: multiple concurrent streams per user (MIC +
SCREEN_AUDIO + AUX_DEVICE), independent per-stream receiver gain/mute/noise-
reduction, talk indicators, and enforced per-channel Opus configurability
(mono/stereo, bitrate, frame size, FEC/DTX, application).

Bugs fixed along the way (found while implementing, not pre-existing scope):
- Server hard-coded stream_id=1 for every announce, so a second stream from
  the same user silently overwrote the first in SessionRegistry::set_user_stream.
  Now a per-session counter (ConnSession::next_stream_id_); handle_stream_stop
  validates against announced_stream_ids_ before clearing.
- Client dropped mode/dtx/complexity/application from effective_audio even for
  the single M2 stream -- only sample_rate/bitrate_bps/frame_ms/fec were ever
  applied to OpusParams. Fixed on both the send (handle_stream_announce_result)
  and receive (sync_remote_streams) paths via a shared
  opus_params_from_audio_config() helper.
- OpusEncoder always used OPUS_APPLICATION_VOIP; added OpusParams::application
  and wired it through.
- on_playback's per-stream decode passed the wrong frame_size to opus_decode
  (total samples instead of samples-per-channel), which would have overflowed
  the decode buffer for any stereo stream.
- teardown_voice() raced when called concurrently from run_io()'s own cleanup
  and from disconnect() on a different thread -- both could see
  udp_thread_/talk_timer_thread_ as joinable() at once and race to join() the
  same std::thread (intermittent std::system_error under ctest). Fixed with a
  teardown_mu_ guard instead of carrying the flake forward.

New:
- Per-channel AudioConfig: SessionRegistry now seeds Lobby (mono/24kbps/VOIP/
  FEC+DTX) and a new "Music Room" channel (stereo/128kbps/AUDIO/no DTX);
  handle_stream_announce enforces the channel's config, clamping (not
  overriding) bitrate_bps to its ceiling.
- core/src/core/client.h/.cpp: local-stream state is now a
  std::unordered_map<int, LocalStream> keyed by vc_stream_kind, with
  request_id-correlated announce/result handling (request_id already
  round-tripped on the wire; just wasn't read before). on_capture_frame is
  kind-aware and upmixes mono capture to stereo when a stream's config calls
  for it. set_self_mute's mic_muted now only gates the MIC kind. NS is wired
  through set_remote_stream. New run_talk_timer() thread emits
  VC_EVENT_TALK_STATE from both remote and local edge detection.
- core/src/audio/audio_engine.h/.cpp: kind-keyed injection taps
  (inject_capture), stereo-to-mono downmix at the decode/mix boundary,
  RemoteStream gains recv_ns (lazy ApmProcessor) + noise_reduction_enabled
  and last_voice_ms/talking; new set_stream_noise_reduction() and
  poll_talk_transitions().
- core/src/session/session.h/.cpp: Stream now carries the full AudioConfig,
  not just sample_rate/frame_ms.
- New additive C ABI (core/include/voicecat.h): vc_audio_config +
  vc_get_stream_audio_config (effective Opus config for any stream you own or
  a peer's); vc_test_inject_capture (test-only synthetic PCM injection,
  clearly marked, mirrors AudioEngine::inject_capture).
- tests/test_m3_multistream.cpp: the M3 exit criterion through the real ABI
  (mirrors test_voice_client_abi.cpp's approach, not raw sockets) -- two
  concurrent local streams, independent gain/mute/NS control, per-channel
  config divergence via vc_get_stream_audio_config, talk indicators.

Explicitly out of scope for this pass (tracked in PROGRESS.md, not silently
dropped): VAD/PTT input gate + device enumeration; real WASAPI loopback
capture for SCREEN_AUDIO (synthetic injection only); true stereo playback
output (AudioEngine's mixer/output device stays mono -- Opus itself is fully
stereo-correct on the wire).

ctest --test-dir build/m1-dev: 11/11 green, verified across 3 consecutive
full-suite runs plus 8 standalone runs of the new test.

Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>

server/src/conn_session.cpp

@@ -2,6 +2,7 @@
 
 #ifdef VOICECAT_HAS_NET
 
+#include <algorithm>
 #include <chrono>
 #include <cstdio>
 #include <cstring>
@@ -329,15 +330,31 @@ void ConnSession::handle_udp_binding(uint64_t req_id, const voicecat::v1::UdpBin
 void ConnSession::handle_stream_announce(uint64_t req_id,
                                           const voicecat::v1::StreamAnnounce& msg) {
     uint32_t ssrc = registry_->assign_ssrc(session_id_);
+    uint32_t stream_id = next_stream_id_++;
 
     auto env = make_env(req_id);
     auto* res = env.mutable_stream_announce_result();
     res->set_ok(true);
-    res->set_stream_id(1);
+    res->set_stream_id(stream_id);
     res->set_ssrc(ssrc);
 
+    // Per-channel AudioConfig is authoritative (docs/voice.md §3): the channel's mode/
+    // frame_ms/application/fec/dtx/complexity/expected_packet_loss apply to every stream
+    // announced into it, regardless of kind. bitrate_bps is clamped (not overridden) to the
+    // channel's ceiling so a client may still request less. sample_rate stays
+    // client-requested-or-48000 — everything runs at 48kHz internally per voice.md §3.
     auto* eff = res->mutable_effective_audio();
-    if (msg.has_requested_audio()) {
+    auto chan_cfg = registry_->channel_audio_config(registry_->user_channel(user_id_.load()));
+    uint32_t requested_bps =
+        msg.has_requested_audio() ? msg.requested_audio().bitrate_bps() : 0;
+    uint32_t requested_rate =
+        msg.has_requested_audio() ? msg.requested_audio().sample_rate() : 0;
+    if (chan_cfg) {
+        *eff = *chan_cfg;
+        eff->set_bitrate_bps(requested_bps > 0 ? std::min(requested_bps, chan_cfg->bitrate_bps())
+                                                : chan_cfg->bitrate_bps());
+        eff->set_sample_rate(requested_rate > 0 ? requested_rate : 48000);
+    } else if (msg.has_requested_audio()) {
         *eff = msg.requested_audio();
     } else {
         eff->set_codec(0);         // OPUS
@@ -350,10 +367,10 @@ void ConnSession::handle_stream_announce(uint64_t req_id,
     if (eff->bitrate_bps() == 0)  eff->set_bitrate_bps(24000);
     if (eff->frame_ms() == 0)     eff->set_frame_ms(20);
 
-    announced_stream_id_ = res->stream_id();
+    announced_stream_ids_.push_back(stream_id);
 
     voicecat::v1::StreamInfo info;
-    info.set_stream_id(res->stream_id());
+    info.set_stream_id(stream_id);
     info.set_ssrc(ssrc);
     info.set_kind(msg.kind());
     *info.mutable_audio() = *eff;
@@ -372,8 +389,12 @@ void ConnSession::handle_stream_announce(uint64_t req_id,
 }
 
 void ConnSession::handle_stream_stop(const voicecat::v1::StreamStop& msg) {
+    auto it = std::find(announced_stream_ids_.begin(), announced_stream_ids_.end(),
+                        msg.stream_id());
+    if (it == announced_stream_ids_.end()) return;  // not ours — ignore (no spoofed stops)
+    announced_stream_ids_.erase(it);
+
     auto updated = registry_->clear_user_stream(user_id_.load(), msg.stream_id());
-    if (msg.stream_id() == announced_stream_id_) announced_stream_id_ = 0;
     if (updated) {
         auto bcast = make_env();
         auto* ue = bcast.mutable_user_event();