feature: alarm

This commit is contained in:
Mikkeli Matlock
2026-02-15 21:11:33 +09:00
parent 12dbbd8942
commit dca989a01b
14 changed files with 543 additions and 1 deletions

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idf_component_register(
SRCS "audio_client.cpp"
REQUIRES espressif__esp_websocket_client port_bsp
PRIV_REQUIRES esp_event json
INCLUDE_DIRS "./")

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#include "audio_client.h"
#include "codec_bsp.h"
#include "esp_websocket_client.h"
#include "esp_log.h"
#include "cJSON.h"
#include <string.h>
#include <freertos/FreeRTOS.h>
#include <freertos/queue.h>
#include <freertos/task.h>
#include <esp_heap_caps.h>
static const char *TAG = "audio_client";
#define AUDIO_CHUNK_SIZE 4096
#define PCM_QUEUE_DEPTH 10
#define PLAYBACK_STACK_SIZE (4 * 1024)
#define PLAYBACK_PRIORITY 4
#define WS_BUFFER_SIZE 8192
static esp_websocket_client_handle_t s_client = NULL;
static CodecPort *s_codec = NULL;
static QueueHandle_t s_pcm_queue = NULL;
static TaskHandle_t s_playback_task = NULL;
static volatile audio_state_t s_state = AUDIO_IDLE;
static volatile bool s_playing = false;
/* Forward declarations */
static void playback_task(void *arg);
static void ws_event_handler(void *arg, esp_event_base_t event_base,
int32_t event_id, void *event_data);
/* ---------- Queue helpers ---------- */
static void flush_queue(void)
{
uint8_t *chunk;
while (xQueueReceive(s_pcm_queue, &chunk, 0) == pdTRUE) {
heap_caps_free(chunk);
}
}
/* ---------- WebSocket event handler ---------- */
static void handle_text_frame(const char *data, int len)
{
cJSON *root = cJSON_ParseWithLength(data, len);
if (!root) {
ESP_LOGW(TAG, "JSON parse failed");
return;
}
cJSON *type = cJSON_GetObjectItem(root, "type");
if (!cJSON_IsString(type)) {
cJSON_Delete(root);
return;
}
if (strcmp(type->valuestring, "alarm_start") == 0) {
int sr = 24000;
int ch = 2;
int bits = 16;
cJSON *item;
item = cJSON_GetObjectItem(root, "sample_rate");
if (cJSON_IsNumber(item)) sr = item->valueint;
item = cJSON_GetObjectItem(root, "channels");
if (cJSON_IsNumber(item)) ch = item->valueint;
item = cJSON_GetObjectItem(root, "bits");
if (cJSON_IsNumber(item)) bits = item->valueint;
ESP_LOGI(TAG, "Alarm start: %dHz %dch %dbit", sr, ch, bits);
/* Flush any stale data */
flush_queue();
/* Open codec for playback */
s_codec->CodecPort_SetInfo("es8311", 1, sr, ch, bits);
s_codec->CodecPort_SetSpeakerVol(70);
s_playing = true;
s_state = AUDIO_PLAYING;
} else if (strcmp(type->valuestring, "alarm_stop") == 0) {
ESP_LOGI(TAG, "Alarm stop");
s_playing = false;
/* Let playback task drain remaining chunks, then close */
vTaskDelay(pdMS_TO_TICKS(100));
flush_queue();
s_codec->CodecPort_CloseSpeaker();
s_state = AUDIO_CONNECTED;
}
cJSON_Delete(root);
}
static void handle_binary_frame(const uint8_t *data, int len)
{
if (!s_playing) return;
uint8_t *chunk = heap_caps_malloc(len, MALLOC_CAP_SPIRAM);
if (!chunk) {
ESP_LOGW(TAG, "PSRAM alloc failed (%d bytes)", len);
return;
}
memcpy(chunk, data, len);
if (xQueueSend(s_pcm_queue, &chunk, 0) != pdTRUE) {
ESP_LOGW(TAG, "PCM queue full, dropping chunk");
heap_caps_free(chunk);
}
}
static void ws_event_handler(void *arg, esp_event_base_t event_base,
int32_t event_id, void *event_data)
{
esp_websocket_event_data_t *ev = (esp_websocket_event_data_t *)event_data;
switch (event_id) {
case WEBSOCKET_EVENT_CONNECTED:
ESP_LOGI(TAG, "Audio WS connected");
s_state = AUDIO_CONNECTED;
break;
case WEBSOCKET_EVENT_DISCONNECTED:
ESP_LOGW(TAG, "Audio WS disconnected");
s_playing = false;
flush_queue();
s_state = AUDIO_IDLE;
break;
case WEBSOCKET_EVENT_DATA:
if (ev->op_code == 0x01 && ev->data_len > 0) {
handle_text_frame(ev->data_ptr, ev->data_len);
} else if (ev->op_code == 0x02 && ev->data_len > 0) {
handle_binary_frame((const uint8_t *)ev->data_ptr, ev->data_len);
}
break;
case WEBSOCKET_EVENT_ERROR:
ESP_LOGE(TAG, "Audio WS error");
s_playing = false;
s_state = AUDIO_ERROR;
break;
default:
break;
}
}
/* ---------- Playback task ---------- */
static void playback_task(void *arg)
{
uint8_t *chunk;
for (;;) {
if (xQueueReceive(s_pcm_queue, &chunk, pdMS_TO_TICKS(500)) == pdTRUE) {
if (s_playing) {
s_codec->CodecPort_PlayWrite(chunk, AUDIO_CHUNK_SIZE);
}
heap_caps_free(chunk);
}
}
}
/* ---------- Public API ---------- */
void audio_client_init(const char *uri, void *codec)
{
s_codec = (CodecPort *)codec;
s_pcm_queue = xQueueCreate(PCM_QUEUE_DEPTH, sizeof(uint8_t *));
if (!s_pcm_queue) {
ESP_LOGE(TAG, "Failed to create PCM queue");
return;
}
esp_websocket_client_config_t config = {};
config.uri = uri;
config.reconnect_timeout_ms = 5000;
config.buffer_size = WS_BUFFER_SIZE;
s_client = esp_websocket_client_init(&config);
esp_websocket_register_events(s_client, WEBSOCKET_EVENT_ANY, ws_event_handler, NULL);
ESP_LOGI(TAG, "Audio client initialized: %s", uri);
}
void audio_client_start(void)
{
if (!s_client) return;
/* Create playback task pinned to Core 1 */
xTaskCreatePinnedToCore(playback_task, "audio_play", PLAYBACK_STACK_SIZE,
NULL, PLAYBACK_PRIORITY, &s_playback_task, 1);
esp_websocket_client_start(s_client);
ESP_LOGI(TAG, "Audio client started");
}
void audio_client_stop(void)
{
if (!s_client) return;
s_playing = false;
esp_websocket_client_stop(s_client);
flush_queue();
if (s_playback_task) {
vTaskDelete(s_playback_task);
s_playback_task = NULL;
}
s_state = AUDIO_IDLE;
ESP_LOGI(TAG, "Audio client stopped");
}
audio_state_t audio_client_get_state(void)
{
return s_state;
}

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#pragma once
#include <stdbool.h>
#ifdef __cplusplus
extern "C" {
#endif
typedef enum {
AUDIO_IDLE = 0,
AUDIO_CONNECTED,
AUDIO_PLAYING,
AUDIO_ERROR,
} audio_state_t;
/**
* Initialize the audio streaming client.
* @param uri WebSocket URI (e.g. "ws://192.168.2.199:8766")
* @param codec Pointer to CodecPort instance (passed as void* for C linkage)
*/
void audio_client_init(const char *uri, void *codec);
/** Start the WebSocket connection and playback task. */
void audio_client_start(void);
/** Stop playback and disconnect. */
void audio_client_stop(void);
/** Get current audio client state. */
audio_state_t audio_client_get_state(void);
#ifdef __cplusplus
}
#endif

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@@ -4,5 +4,6 @@
#define WIFI_SSID "Novoyuuparosk_H3C"
#define WIFI_PASSWORD "northwich"
#define WS_SERVER_URI "ws://192.168.2.199:8765"
#define AUDIO_SERVER_URI "ws://192.168.2.199:8766"
#endif

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@@ -1,5 +1,5 @@
idf_component_register(
SRCS "user_app.cpp" "alert.cpp"
REQUIRES app_bsp
PRIV_REQUIRES esp_wifi_bsp ws_client dashboard_ui port_bsp esp_timer codec_board
PRIV_REQUIRES esp_wifi_bsp ws_client dashboard_ui port_bsp esp_timer codec_board audio_client
INCLUDE_DIRS "./")

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@@ -1,4 +1,5 @@
#include "alert.h"
#include "audio_client.h"
#include "codec_bsp.h"
#include "i2c_bsp.h"
#include <esp_log.h>
@@ -84,6 +85,7 @@ void alert_trigger(alert_type_t type)
if (type >= ALERT_TYPE_COUNT) return;
if (s_muted) return;
if (!s_tone_buf || !s_codec) return;
if (audio_client_get_state() == AUDIO_PLAYING) return; /* don't fight over codec */
if (xSemaphoreTake(s_alert_mutex, pdMS_TO_TICKS(50)) != pdTRUE) return;

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@@ -9,6 +9,7 @@
#include "button_bsp.h"
#include "codec_bsp.h"
#include "alert.h"
#include "audio_client.h"
#include "lvgl_bsp.h"
#include <esp_log.h>
@@ -57,6 +58,9 @@ void UserApp_AppInit(void)
alert_init();
alert_set_codec(s_codec);
/* Audio streaming client */
audio_client_init(AUDIO_SERVER_URI, s_codec);
/* WebSocket client init (not started yet) */
ws_client_init(WS_SERVER_URI);
ws_client_set_data_callback(ws_data_cb);
@@ -78,6 +82,9 @@ void UserApp_TaskInit(void)
/* Start WebSocket client */
ws_client_start();
/* Start audio streaming client */
audio_client_start();
/* Sensor polling task - Core 1, 4KB stack */
xTaskCreatePinnedToCore(sensor_task, "sensor", 4 * 1024, NULL, 3, NULL, 1);

87
docs/ALARM_PROTOCOL.md Normal file
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# Alarm Audio Streaming Protocol
## Overview
The Raspberry Pi streams alarm audio to the ESP32-S3 over a dedicated WebSocket connection on **port 8766**. The protocol uses mixed text and binary frames — text for control messages, binary for raw PCM data.
## Connection
| Parameter | Value |
|-----------|-------|
| Port | 8766 |
| Transport | WebSocket |
| Direction | Pi (server) → ESP32 (client) |
The ESP32 connects and stays connected. The server initiates alarm playback when needed.
## Message Sequence
```
Pi ESP32
│ │
│◄──── WS connect ──────────│
│ │
│ (idle until alarm fires) │
│ │
├─ alarm_start (text) ──────►│ → open codec
├─ PCM chunk (binary) ──────►│ → queue + play
├─ PCM chunk (binary) ──────►│
│ ... │
├─ alarm_stop (text) ────────►│ → drain + close codec
│ │
```
## Control Messages (Text Frames)
### alarm_start
Sent before the first PCM chunk. The ESP32 uses these parameters to configure the DAC.
```json
{
"type": "alarm_start",
"sample_rate": 24000,
"channels": 2,
"bits": 16
}
```
### alarm_stop
Sent after the last PCM chunk.
```json
{
"type": "alarm_stop"
}
```
## PCM Data (Binary Frames)
- **Format:** Raw signed 16-bit little-endian, interleaved stereo (L, R, L, R, ...)
- **Chunk size:** 4096 bytes (1024 stereo samples at 16-bit)
- **Pacing:** Chunks sent at ~90% real-time speed to maintain buffer without overflow
- **Byte order:** Little-endian (native for both Pi and ESP32)
At 24kHz / 2ch / 16-bit:
- Bytes per second: 96,000
- Chunk duration: ~42.7 ms
- Send interval: ~38.4 ms (90% pacing)
## ESP32 Buffering
- FreeRTOS queue: 10 slots of pointer-to-chunk
- Total buffer capacity: ~427 ms of audio
- Overflow policy: drop chunk and log warning
## Future: Alarm Scheduling
Not yet implemented. Planned JSON format for alarm configuration:
```json
{
"alarm_time": "070000",
"alarm_days": ["Mon", "Tue", "Wed"],
"alarm_audio": "path/to/file.wav"
}
```

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0 700000 j
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3110000 3710000 ch
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12470000 13100000 ts
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35870000 36510000 u
36510000 36960000 t
36960000 38220000 o

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pi/audio_server.py Normal file
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"""
Alarm audio streaming test server.
Streams a WAV file as raw PCM chunks over WebSocket on port 8766.
Repeats every 30-60 seconds to exercise the ESP32 audio pipeline.
Protocol:
1. Text frame: {"type":"alarm_start","sample_rate":N,"channels":N,"bits":N}
2. Binary frames: raw PCM chunks (4096 bytes each, paced at ~90% real-time)
3. Text frame: {"type":"alarm_stop"}
"""
import asyncio
import json
import logging
import struct
import wave
from pathlib import Path
from random import randint
import websockets
logging.basicConfig(level=logging.INFO, format="%(asctime)s %(levelname)s %(message)s")
log = logging.getLogger("audio_server")
PORT = 8766
CHUNK_SIZE = 4096
AUDIO_DIR = Path(__file__).parent / "assets" / "alarm"
def find_wav() -> Path:
"""Find the first .wav file in the alarm assets directory."""
wavs = list(AUDIO_DIR.glob("*.wav"))
if not wavs:
raise FileNotFoundError(f"No .wav files found in {AUDIO_DIR}")
log.info("Using audio file: %s", wavs[0].name)
return wavs[0]
def read_wav(path: Path) -> tuple[bytes, int, int, int]:
"""Read WAV file and return (pcm_data, sample_rate, channels, bits_per_sample)."""
with wave.open(str(path), "rb") as wf:
sr = wf.getframerate()
ch = wf.getnchannels()
bits = wf.getsampwidth() * 8
pcm = wf.readframes(wf.getnframes())
log.info("WAV loaded: %dHz %dch %dbit, %.1fs, %d bytes",
sr, ch, bits, len(pcm) / (sr * ch * (bits // 8)), len(pcm))
return pcm, sr, ch, bits
def chunk_bytes(data: bytes, size: int):
"""Yield data in fixed-size chunks."""
for i in range(0, len(data), size):
yield data[i : i + size]
async def stream_alarm(ws, pcm: bytes, sr: int, ch: int, bits: int):
"""Stream one alarm cycle to the connected client."""
# Compute pacing: how long each chunk represents in seconds
bytes_per_sec = sr * ch * (bits // 8)
chunk_duration = CHUNK_SIZE / bytes_per_sec
pace_delay = chunk_duration * 0.9 # 90% real-time to avoid underrun
total_chunks = (len(pcm) + CHUNK_SIZE - 1) // CHUNK_SIZE
# Start
start_msg = json.dumps({
"type": "alarm_start",
"sample_rate": sr,
"channels": ch,
"bits": bits,
})
await ws.send(start_msg)
log.info("Sent alarm_start (%d chunks, pace %.1fms)", total_chunks, pace_delay * 1000)
# Stream PCM chunks
for i, chunk in enumerate(chunk_bytes(pcm, CHUNK_SIZE)):
await ws.send(chunk) # bytes → binary frame
await asyncio.sleep(pace_delay)
# Stop
await ws.send(json.dumps({"type": "alarm_stop"}))
log.info("Sent alarm_stop")
async def handler(ws):
"""Handle a single WebSocket connection."""
remote = ws.remote_address
log.info("Client connected: %s:%d", remote[0], remote[1])
wav_path = find_wav()
pcm, sr, ch, bits = read_wav(wav_path)
try:
while True:
delay = randint(30, 60)
log.info("Next alarm in %ds", delay)
await asyncio.sleep(delay)
await stream_alarm(ws, pcm, sr, ch, bits)
except websockets.exceptions.ConnectionClosed:
log.info("Client disconnected: %s:%d", remote[0], remote[1])
async def main():
log.info("Audio server starting on port %d", PORT)
async with websockets.serve(handler, "0.0.0.0", PORT):
await asyncio.Future() # run forever
if __name__ == "__main__":
main_loop = asyncio.run(main())

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pi/run_all.py Normal file
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#!/usr/bin/env python3
"""Launch stats_server and audio_server as child processes."""
import subprocess, sys, signal
from pathlib import Path
d = Path(__file__).parent
procs = [
subprocess.Popen([sys.executable, d / "stats_server.py"]),
subprocess.Popen([sys.executable, d / "audio_server.py"]),
]
signal.signal(signal.SIGINT, lambda *_: [p.terminate() for p in procs])
signal.signal(signal.SIGTERM, lambda *_: [p.terminate() for p in procs])
print(f"Running stats_server (PID {procs[0].pid}) + audio_server (PID {procs[1].pid})")
for p in procs:
p.wait()