Add start/stop tones and rotate WAV sessions

bridge/.env.example

@@ -20,3 +20,6 @@ HA_WEBHOOK_URL=
 SAVE_SESSIONS=true
 SESSIONS_DIR=/data/sessions
 PCM_SAMPLE_RATE=16000
+MAX_SESSION_BYTES=16000000
+WAV_SEGMENT_MAX_BYTES=2097152
+WAV_KEEP_FILES=10

bridge/README.md

@@ -80,6 +80,10 @@ You can build automations on these events (for STT/TTS pipelines or Node-RED han
 
 - Audio format: PCM16LE, mono, 16 kHz
 - `SAVE_SESSIONS=true` stores `.wav` files in `bridge/data/sessions`
+- Recording is written during `start`..`stop` and rotated automatically:
+  - `WAV_SEGMENT_MAX_BYTES` max size per `.wav` file (default: `2097152` = 2 MB)
+  - `WAV_KEEP_FILES` max number of `.wav` files to keep (default: `10`)
+- `MAX_SESSION_BYTES` is only used if session file saving is disabled
 - MQTT is recommended for control/events, WebSocket for streaming audio
 
 ## 7) Build and push to Gitea registry
@@ -127,6 +131,8 @@ services:
       SAVE_SESSIONS: "true"
       SESSIONS_DIR: "/data/sessions"
       PCM_SAMPLE_RATE: "16000"
+      WAV_SEGMENT_MAX_BYTES: "2097152"
+      WAV_KEEP_FILES: "10"
     volumes:
       - evs_bridge_data:/data
 

bridge/app.py

@@ -4,9 +4,10 @@ import json
 import logging
 import os
 import time
+import wave
 from dataclasses import dataclass, field
 from pathlib import Path
-from typing import Dict, Optional
+from typing import Dict, List, Optional
 
 import aiohttp
 import paho.mqtt.client as mqtt
@@ -46,6 +47,10 @@ HA_WEBHOOK_URL = os.getenv("HA_WEBHOOK_URL", "").strip()
 SAVE_SESSIONS = getenv_bool("SAVE_SESSIONS", True)
 SESSIONS_DIR = Path(os.getenv("SESSIONS_DIR", "/data/sessions"))
 PCM_SAMPLE_RATE = int(os.getenv("PCM_SAMPLE_RATE", "16000"))
+MAX_SESSION_BYTES = int(os.getenv("MAX_SESSION_BYTES", "16000000"))
+WAV_SEGMENT_MAX_BYTES = int(os.getenv("WAV_SEGMENT_MAX_BYTES", str(2 * 1024 * 1024)))
+WAV_KEEP_FILES = int(os.getenv("WAV_KEEP_FILES", "10"))
+WAV_HEADER_BYTES = 44
 
 
 @dataclass
@@ -56,6 +61,12 @@ class DeviceSession:
     ptt_active: bool = False
     pcm_bytes: bytearray = field(default_factory=bytearray)
     last_rx_ts: float = field(default_factory=time.time)
+    rx_bytes_total: int = 0
+    segment_index: int = 0
+    segment_pcm_bytes: int = 0
+    current_wav_path: Optional[str] = None
+    current_wav: Optional[wave.Wave_write] = None
+    saved_wavs: List[str] = field(default_factory=list)
 
 
 class BridgeState:
@@ -89,12 +100,12 @@ state = BridgeState()
 
 
 def build_metrics(device_id: str, session: DeviceSession) -> dict:
-    samples = len(session.pcm_bytes) // 2
+    samples = session.rx_bytes_total // 2
     seconds = samples / float(PCM_SAMPLE_RATE)
     return {
         "device_id": device_id,
         "ptt_active": session.ptt_active,
-        "rx_bytes": len(session.pcm_bytes),
+        "rx_bytes": session.rx_bytes_total,
         "duration_s": round(seconds, 3),
         "last_rx_ts": session.last_rx_ts,
     }
@@ -113,26 +124,95 @@ async def call_ha_webhook(event: str, payload: dict) -> None:
         log.exception("ha webhook call failed")
 
 
-def save_session_wav(device_id: str, pcm: bytes) -> Optional[str]:
+def enforce_wav_retention() -> None:
-    if not SAVE_SESSIONS or not pcm:
+    if not SAVE_SESSIONS:
+        return
+    try:
+        SESSIONS_DIR.mkdir(parents=True, exist_ok=True)
+        wavs = sorted(
+            [p for p in SESSIONS_DIR.glob("*.wav") if p.is_file()],
+            key=lambda p: p.stat().st_mtime,
+        )
+        while len(wavs) > WAV_KEEP_FILES:
+            oldest = wavs.pop(0)
+            try:
+                oldest.unlink()
+                log.info("deleted old wav: %s", oldest)
+            except Exception:
+                log.exception("failed to delete old wav: %s", oldest)
+    except Exception:
+        log.exception("failed to enforce wav retention")
+
+
+def close_active_wav(session: DeviceSession) -> None:
+    if not session.current_wav:
+        return
+    try:
+        session.current_wav.close()
+    except Exception:
+        log.exception("failed closing wav for %s", session.device_id)
+    finally:
+        session.current_wav = None
+        session.current_wav_path = None
+        session.segment_pcm_bytes = 0
+
+
+def open_new_wav_segment(session: DeviceSession) -> Optional[str]:
+    if not SAVE_SESSIONS:
         return None
     try:
         SESSIONS_DIR.mkdir(parents=True, exist_ok=True)
-        ts = int(time.time())
+        ts_ms = int(time.time() * 1000)
-        path = SESSIONS_DIR / f"{device_id}_{ts}.wav"
+        name = f"{session.device_id}_{ts_ms}_part{session.segment_index:03d}.wav"
-        import wave
+        path = SESSIONS_DIR / name
-
+        wf = wave.open(str(path), "wb")
-        with wave.open(str(path), "wb") as wf:
         wf.setnchannels(1)
         wf.setsampwidth(2)
         wf.setframerate(PCM_SAMPLE_RATE)
-            wf.writeframes(pcm)
+        session.current_wav = wf
+        session.current_wav_path = str(path)
+        session.segment_pcm_bytes = 0
+        session.saved_wavs.append(str(path))
+        enforce_wav_retention()
+        session.segment_index += 1
         return str(path)
     except Exception:
-        log.exception("failed to save wav")
+        log.exception("failed to open wav segment for %s", session.device_id)
+        close_active_wav(session)
         return None
 
 
+def append_pcm_with_rotation(session: DeviceSession, data: bytes) -> None:
+    if not SAVE_SESSIONS or not data:
+        return
+    if WAV_SEGMENT_MAX_BYTES <= WAV_HEADER_BYTES:
+        log.warning("WAV_SEGMENT_MAX_BYTES too small, minimum is %s", WAV_HEADER_BYTES + 2)
+        return
+
+    max_pcm_per_file = WAV_SEGMENT_MAX_BYTES - WAV_HEADER_BYTES
+    offset = 0
+    total = len(data)
+    while offset < total:
+        if not session.current_wav:
+            opened = open_new_wav_segment(session)
+            if not opened:
+                return
+        free_bytes = max_pcm_per_file - session.segment_pcm_bytes
+        if free_bytes <= 0:
+            close_active_wav(session)
+            continue
+        chunk_len = min(free_bytes, total - offset)
+        chunk = data[offset : offset + chunk_len]
+        try:
+            session.current_wav.writeframesraw(chunk)
+            session.segment_pcm_bytes += chunk_len
+        except Exception:
+            log.exception("failed writing wav segment for %s", session.device_id)
+            close_active_wav(session)
+            return
+        offset += chunk_len
+
+
 async def handle_text_message(device_id: str, session: DeviceSession, raw: str) -> None:
     try:
         msg = json.loads(raw)
@@ -144,33 +224,77 @@ async def handle_text_message(device_id: str, session: DeviceSession, raw: str)
     if msg_type == "start":
         session.ptt_active = True
         session.pcm_bytes.clear()
+        session.rx_bytes_total = 0
+        session.saved_wavs.clear()
+        session.segment_index = 0
+        close_active_wav(session)
+        first_path = open_new_wav_segment(session)
         payload = {"type": "start", "ts": time.time(), "device_id": device_id}
+        if first_path:
+            payload["wav_path"] = first_path
         state.publish_status(device_id, payload)
         await call_ha_webhook("start", payload)
+        log.info("start: device=%s wav=%s", device_id, first_path or "-")
         return
 
     if msg_type == "stop":
         session.ptt_active = False
+        close_active_wav(session)
         metrics = build_metrics(device_id, session)
-        wav_path = save_session_wav(device_id, bytes(session.pcm_bytes))
         payload = {"type": "stop", "ts": time.time(), "device_id": device_id, **metrics}
-        if wav_path:
+        if session.saved_wavs:
-            payload["wav_path"] = wav_path
+            payload["wav_path"] = session.saved_wavs[-1]
+            payload["wav_paths"] = session.saved_wavs
         state.publish_status(device_id, payload)
         await call_ha_webhook("stop", payload)
+        log.info(
+            "stop: device=%s bytes=%s duration_s=%s wav_count=%s last_wav=%s",
+            device_id,
+            metrics.get("rx_bytes", 0),
+            metrics.get("duration_s", 0),
+            len(session.saved_wavs),
+            session.saved_wavs[-1] if session.saved_wavs else "-",
+        )
         return
 
     if msg_type == "ping":
         await session.ws.send(json.dumps({"type": "pong", "ts": time.time()}))
         return
 
+    if msg_type == "mic_level":
+        payload = {
+            "type": "mic_level",
+            "ts": time.time(),
+            "device_id": device_id,
+            "peak": msg.get("peak", 0),
+            "avg_abs": msg.get("avg_abs", 0),
+            "samples": msg.get("samples", 0),
+        }
+        state.publish_status(device_id, payload)
+        log.info(
+            "mic_level: device=%s peak=%s avg_abs=%s samples=%s",
+            device_id,
+            payload["peak"],
+            payload["avg_abs"],
+            payload["samples"],
+        )
+        return
+
     log.info("text msg from %s: %s", device_id, msg)
 
 
 async def handle_binary_message(device_id: str, session: DeviceSession, data: bytes) -> None:
     session.last_rx_ts = time.time()
     if session.ptt_active:
+        session.rx_bytes_total += len(data)
+        if SAVE_SESSIONS:
+            append_pcm_with_rotation(session, data)
+        else:
             session.pcm_bytes.extend(data)
+            if len(session.pcm_bytes) > MAX_SESSION_BYTES:
+                # Keep newest data within cap to avoid unbounded memory growth.
+                drop = len(session.pcm_bytes) - MAX_SESSION_BYTES
+                del session.pcm_bytes[:drop]
     if ECHO_ENABLED:
         await session.ws.send(data)
 
@@ -211,6 +335,7 @@ async def ws_handler(ws: WebSocketServerProtocol, path: str) -> None:
     except websockets.ConnectionClosed:
         pass
     finally:
+        close_active_wav(session)
         if state.devices.get(device_id) is session:
             del state.devices[device_id]
         state.publish_status(device_id, {"type": "disconnected", "ts": time.time(), "device_id": device_id})

include/secrets.example.h

@@ -17,5 +17,19 @@ static const char* EVS_DEVICE_ID = "esp32-room-name";
 
 // Connectivity behavior
 static constexpr uint32_t EVS_RECONNECT_MS = 5000;
+static constexpr bool EVS_DEFAULT_STREAM_MODE = true;
+static constexpr bool EVS_SERIAL_COMMAND_ECHO = true;
+
+// INMP441 tuning
+// L/R pin on mic:
+// - GND usually = left channel
+// - 3V3 usually = right channel
+static constexpr bool EVS_MIC_USE_RIGHT_CHANNEL = false;
+// 24-bit sample to 16-bit conversion shift. Start with 8.
+static constexpr int EVS_MIC_S24_TO_S16_SHIFT = 8;
+// Digital mic gain (avoid clipping). Typical 0.2 .. 0.8.
+static constexpr float EVS_MIC_GAIN = 0.35f;
+// Extra gain only for local loopback monitor path.
+static constexpr float EVS_LOOPBACK_MONITOR_GAIN = 0.35f;
 
 #endif  // EVS_SECRETS_H

src/main.cpp

@@ -1,10 +1,18 @@
 #include <Arduino.h>
 #include <WiFi.h>
 #include <ArduinoWebsockets.h>
+#include <math.h>
 #include "driver/i2s.h"
 #include "secrets.h"
 using namespace websockets;
 
+static constexpr bool kDefaultStreamMode = EVS_DEFAULT_STREAM_MODE;
+static constexpr bool kSerialCommandEcho = EVS_SERIAL_COMMAND_ECHO;
+static constexpr bool kMicUseRightChannel = EVS_MIC_USE_RIGHT_CHANNEL;
+static constexpr int kMicS24ToS16Shift = EVS_MIC_S24_TO_S16_SHIFT;
+static constexpr float kMicGain = EVS_MIC_GAIN;
+static constexpr float kLoopbackMonitorGain = EVS_LOOPBACK_MONITOR_GAIN;
+
 // ---------------------------
 // Project config
 // ---------------------------
@@ -24,6 +32,8 @@ static constexpr int PWM_CHANNEL = 0;
 static constexpr uint32_t PWM_FREQ = 22050;
 static constexpr uint8_t PWM_RES_BITS = 8;
 static constexpr uint32_t SPEAKER_SAMPLE_RATE = 16000;
+static constexpr uint32_t MIC_TELEMETRY_INTERVAL_MS = 1000;
+static constexpr float PI_F = 3.14159265358979323846f;
 
 // WiFi / WebSocket
 
@@ -40,6 +50,7 @@ static bool g_wsConnected = false;
 static uint32_t g_lastConnectTryMs = 0;
 static uint32_t g_nextOutUs = 0;
 static bool g_streamingActive = false;
+static uint32_t g_lastMicTelemetryMs = 0;
 
 static constexpr size_t RX_SAMPLES_CAP = 16000;
 static int16_t g_rxSamples[RX_SAMPLES_CAP];
@@ -52,7 +63,7 @@ static bool initMicI2s() {
       .mode = (i2s_mode_t)(I2S_MODE_MASTER | I2S_MODE_RX),
       .sample_rate = MIC_SAMPLE_RATE,
       .bits_per_sample = MIC_BITS,
-      .channel_format = I2S_CHANNEL_FMT_ONLY_LEFT,
+      .channel_format = kMicUseRightChannel ? I2S_CHANNEL_FMT_ONLY_RIGHT : I2S_CHANNEL_FMT_ONLY_LEFT,
       .communication_format = I2S_COMM_FORMAT_STAND_I2S,
       .intr_alloc_flags = 0,
       .dma_buf_count = 8,
@@ -129,6 +140,49 @@ static void enqueuePcmFrame(const int16_t* frame, size_t count) {
   }
 }
 
+static void enqueueTone(uint16_t freqHz, uint16_t durationMs, int16_t amplitude) {
+  if (freqHz == 0 || durationMs == 0) {
+    return;
+  }
+  const uint32_t sampleCount = (uint32_t)SPEAKER_SAMPLE_RATE * durationMs / 1000U;
+  const float phaseStep = 2.0f * PI_F * (float)freqHz / (float)SPEAKER_SAMPLE_RATE;
+  float phase = 0.0f;
+  for (uint32_t i = 0; i < sampleCount; ++i) {
+    const float s = sinf(phase);
+    const int16_t sample = (int16_t)(s * (float)amplitude);
+    if (!enqueuePcmSample(sample)) {
+      return;
+    }
+    phase += phaseStep;
+    if (phase > 2.0f * PI_F) {
+      phase -= 2.0f * PI_F;
+    }
+  }
+}
+
+static void enqueueSilenceMs(uint16_t durationMs) {
+  const uint32_t sampleCount = (uint32_t)SPEAKER_SAMPLE_RATE * durationMs / 1000U;
+  for (uint32_t i = 0; i < sampleCount; ++i) {
+    if (!enqueuePcmSample(0)) {
+      return;
+    }
+  }
+}
+
+static void playStartTone() {
+  // Ascending double beep: A5 -> C6.
+  enqueueTone(880, 90, 5000);
+  enqueueSilenceMs(35);
+  enqueueTone(1047, 110, 6000);
+}
+
+static void playStopTone() {
+  // Descending double beep: C6 -> A5.
+  enqueueTone(1047, 90, 5000);
+  enqueueSilenceMs(35);
+  enqueueTone(880, 110, 6000);
+}
+
 static void onWsMessageCallback(WebsocketsMessage message) {
   if (!message.isBinary()) {
     return;
@@ -157,9 +211,13 @@ static void onWsEventCallback(WebsocketsEvent event, String) {
     if (g_mode == DeviceMode::StreamToServer && !g_streamingActive) {
       g_ws.send("{\"type\":\"start\"}");
       g_streamingActive = true;
+      playStartTone();
     }
     Serial.println("WS connected");
   } else if (event == WebsocketsEvent::ConnectionClosed) {
+    if (g_streamingActive) {
+      playStopTone();
+    }
     g_wsConnected = false;
     g_streamingActive = false;
     Serial.println("WS disconnected");
@@ -206,11 +264,13 @@ static void setMode(DeviceMode mode) {
   if (g_mode == DeviceMode::StreamToServer && g_wsConnected && g_streamingActive) {
     g_ws.send("{\"type\":\"stop\"}");
     g_streamingActive = false;
+    playStopTone();
   }
   g_mode = mode;
   if (g_mode == DeviceMode::StreamToServer && g_wsConnected && !g_streamingActive) {
     g_ws.send("{\"type\":\"start\"}");
     g_streamingActive = true;
+    playStartTone();
   }
 }
 
@@ -222,35 +282,93 @@ static void handleFrameForServer(const int16_t* frame, size_t count) {
   g_ws.sendBinary(reinterpret_cast<const char*>(frame), count * sizeof(int16_t));
 }
 
-static void serviceSpeaker() {
+static void publishMicTelemetryIfDue(const int16_t* frame, size_t count) {
-  const uint32_t periodUs = 1000000UL / SPEAKER_SAMPLE_RATE;
+  if (count == 0) {
-  const uint32_t now = micros();
-  if ((int32_t)(now - g_nextOutUs) < 0) {
     return;
   }
-  g_nextOutUs += periodUs;
 
+  int16_t peak = 0;
+  int64_t sum_abs = 0;
+  for (size_t i = 0; i < count; ++i) {
+    int16_t v = frame[i];
+    int16_t av = (v < 0) ? (int16_t)(-v) : v;
+    if (av > peak) peak = av;
+    sum_abs += av;
+  }
+  const uint32_t avg_abs = (uint32_t)(sum_abs / (int64_t)count);
+
+  const uint32_t now = millis();
+  if ((now - g_lastMicTelemetryMs) < MIC_TELEMETRY_INTERVAL_MS) {
+    return;
+  }
+  g_lastMicTelemetryMs = now;
+
+  Serial.print("MIC peak=");
+  Serial.print(peak);
+  Serial.print(" avg_abs=");
+  Serial.print(avg_abs);
+  Serial.print(" ws=");
+  Serial.println(g_wsConnected ? "connected" : "disconnected");
+
+  if (g_wsConnected) {
+    String msg = "{\"type\":\"mic_level\",\"peak\":";
+    msg += String(peak);
+    msg += ",\"avg_abs\":";
+    msg += String(avg_abs);
+    msg += ",\"samples\":";
+    msg += String((unsigned)count);
+    msg += "}";
+    g_ws.send(msg);
+  }
+}
+
+static inline int16_t convertMicSampleToPcm16(int32_t raw32) {
+  // INMP441 uses 24-bit signed samples packed into 32-bit I2S slots.
+  int32_t s24 = raw32 >> 8;
+  int32_t s16 = s24 >> kMicS24ToS16Shift;
+  float scaled = (float)s16 * kMicGain;
+  if (scaled > 32767.0f) scaled = 32767.0f;
+  if (scaled < -32768.0f) scaled = -32768.0f;
+  return (int16_t)scaled;
+}
+
+static void serviceSpeaker() {
+  const uint32_t periodUs = 1000000UL / SPEAKER_SAMPLE_RATE;
+  uint32_t now = micros();
+  int processed = 0;
+  // Catch up if we lagged behind, but cap work per loop iteration.
+  while ((int32_t)(now - g_nextOutUs) >= 0 && processed < 8) {
+    g_nextOutUs += periodUs;
     int16_t s = 0;
     if (dequeuePcmSample(s)) {
       pwmWrite(pcm16ToPwm8(s));
     } else {
       pwmWrite(128);
     }
+    now = micros();
+    ++processed;
+  }
 }
 
 static void printHelp() {
   Serial.println();
   Serial.println("Commands:");
   Serial.println("  i = idle");
-  Serial.println("  s = stream mode (stub)");
+  Serial.println("  s = stream mode");
   Serial.println("  l = local loopback mode");
   Serial.println("  p = print network status");
   Serial.println("  h = help");
+  Serial.print("Default on boot: ");
+  Serial.println(kDefaultStreamMode ? "StreamToServer" : "LocalLoopback");
 }
 
 static void handleSerialCommands() {
   while (Serial.available()) {
     const char c = (char)Serial.read();
+    if (kSerialCommandEcho && c != '\r' && c != '\n') {
+      Serial.print("RX cmd: ");
+      Serial.println(c);
+    }
     if (c == 'i') {
       setMode(DeviceMode::Idle);
       Serial.println("Mode -> Idle");
@@ -269,6 +387,8 @@ static void handleSerialCommands() {
       Serial.println(g_wsConnected ? "connected" : "disconnected");
     } else if (c == 'h') {
       printHelp();
+    } else if (c != '\r' && c != '\n') {
+      Serial.println("Unknown command");
     }
   }
 }
@@ -291,7 +411,11 @@ void setup() {
   g_ws.onEvent(onWsEventCallback);
   g_nextOutUs = micros();
 
+  if (kDefaultStreamMode) {
+    setMode(DeviceMode::StreamToServer);
+  } else {
     setMode(DeviceMode::LocalLoopback);
+  }
   Serial.println("Audio init ok");
   Serial.println("Set local environment values in include/secrets.h");
   printHelp();
@@ -299,8 +423,10 @@ void setup() {
 
 void loop() {
   handleSerialCommands();
+  if (g_mode != DeviceMode::LocalLoopback) {
     ensureConnectivity();
     g_ws.poll();
+  }
   serviceSpeaker();
 
   size_t bytesRead = 0;
@@ -313,14 +439,19 @@ void loop() {
   const size_t sampleCount = bytesRead / sizeof(int32_t);
   static int16_t pcm16[MIC_FRAME_SAMPLES];
   for (size_t i = 0; i < sampleCount; ++i) {
-    // INMP441 delivers meaningful data in the high bits for 32-bit slot formats.
+    pcm16[i] = convertMicSampleToPcm16(g_micBuffer[i]);
-    pcm16[i] = (int16_t)(g_micBuffer[i] >> 14);
   }
 
   if (g_mode == DeviceMode::StreamToServer) {
     handleFrameForServer(pcm16, sampleCount);
+    publishMicTelemetryIfDue(pcm16, sampleCount);
   } else if (g_mode == DeviceMode::LocalLoopback) {
-    enqueuePcmFrame(pcm16, sampleCount);
+    for (size_t i = 0; i < sampleCount; ++i) {
+      float v = (float)pcm16[i] * kLoopbackMonitorGain;
+      if (v > 32767.0f) v = 32767.0f;
+      if (v < -32768.0f) v = -32768.0f;
+      enqueuePcmSample((int16_t)v);
+    }
   } else {
     // idle
   }