clement
aa7ae277ed
- hook polarity active-HIGH + auto-resync (was LOW) - ring cadence FT 1.5s ON / 3.5s OFF - DTMF Goertzel decoder (dtmf.c/h) + rotary debounce - LISTEN half-duplex: capture → /v1/voice/reply → play - WAV playback buffered PSRAM + mono→stereo upmix - SPIFFS mount at boot for pre-loaded greetings - ES8388: DAC digital vol + mic PGA + GPIO INPUT_OUTPUT - turn_client multipart + 90s timeout + fixed routing - debug endpoints: vol/dacvol/offhook/getfile/hookmon
94 lines
3.6 KiB
C
94 lines
3.6 KiB
C
#pragma once
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/*
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* audio.h — I2S + ES8388 audio interface for the PLIP voice annex.
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*
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* Provides:
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* - es8388 + I2S initialisation (called once from app_main)
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* - Blocking 440 Hz tone (Phase A proof)
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* - WAV file playback from SD or embedded asset
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* - Ring tone (400/450 Hz cadence) for Phase D
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* - Async playback command queue (used by cmd_exec / ESP-NOW)
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*/
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#include <stdint.h>
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#include <stddef.h>
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#include "esp_err.h"
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#include "driver/i2s_std.h"
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#ifdef __cplusplus
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extern "C" {
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#endif
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/* Initialise ES8388 and I2S channels. Must be called before any other
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* audio_* function. Returns ESP_OK on success. */
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esp_err_t audio_init(void);
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/* Return the speaker I2S TX channel handle (used by cmd_exec for WAV play). */
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i2s_chan_handle_t audio_spk_handle(void);
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/* Play a pure sine tone for duration_ms (blocking). */
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void audio_play_tone(float frequency_hz, int duration_ms);
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/* Enqueue an async play command. path may be:
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* - "/sdcard/<file>.wav" — read from SD
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* - "embedded://" — built-in C5-E5-G5 cue
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* - "" or NULL — same as embedded://
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* Returns ESP_OK if the command was enqueued (non-blocking from any task). */
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esp_err_t audio_play_async(const char *path);
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/* Stop current playback immediately. */
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esp_err_t audio_stop(void);
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/* True while the audio worker is playing a clip (tone/WAV). The conversation
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* LISTEN loop polls this to stay half-duplex: never capture while playing
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* (avoids the earpiece→mic feedback that saturated the line). */
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bool audio_is_playing(void);
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/* Start ring tone cadence (ON 1s / OFF 2s) at ~440 Hz. Continues until
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* audio_stop() is called. Non-blocking — spawns an internal task. */
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esp_err_t audio_ring_start(void);
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/* Enable or disable the power amplifier (GPIO21 PA_ENABLE).
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* Call audio_pa_set(true) on off-hook, audio_pa_set(false) on on-hook.
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* ring_start forces PA on internally; ring_stop calls audio_pa_set(false)
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* only if the handset is on-hook. */
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void audio_pa_set(bool enable);
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/*
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* Capture microphone audio and encode as WAV (16 kHz, mono, S16-LE).
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*
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* out : caller-provided buffer (must be >= 44 + PCM bytes)
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* out_max : size of out in bytes
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* max_ms : hard cap on capture duration in milliseconds
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* silence_ms: milliseconds of silence after voice onset to trigger stop
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*
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* Returns total bytes written (44-byte WAV header + PCM), or -1 on error.
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* Uses half-duplex: disables TX speaker during capture and re-enables it
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* on return. Safe to call from any task; not reentrant.
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*/
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int audio_capture_wav(uint8_t *out, size_t out_max, int max_ms, int silence_ms);
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/*
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* Streaming capture API — avoids a large output buffer by letting the caller
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* write chunks incrementally (e.g. directly into a TCP socket).
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*
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* Usage:
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* audio_capture_begin(max_ms, silence_ms) — enable RX, returns 0 on OK
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* audio_capture_read_frame(mono_out, n_samples, rms_sq_out)
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* — read one 20 ms frame (mono S16)
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* fills n_samples mono samples,
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* sets *rms_sq_out (energy metric)
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* returns actual samples read, 0=timeout, -1=error
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* audio_capture_end() — disable RX, re-enable TX
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*
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* n_samples must be 320 (= SAMPLE_RATE/1000*20, one 20 ms frame).
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* The caller is responsible for VAD and stop logic.
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*/
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int audio_capture_begin(int max_ms, int silence_ms);
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int audio_capture_read_frame(int16_t *mono_out, int n_samples, int64_t *rms_sq_out);
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void audio_capture_end(void);
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#ifdef __cplusplus
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}
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#endif
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