# =============================================================================
# GENERIC ESP32-S3 VOIP FULL-DUPLEX - native ESPHome audio
# =============================================================================
# Standalone voip-only profile for native ESPHome audio.
#
# Purpose:
#   - No esp_audio_stack.
#   - No esp_aec / esp_afe software processing.
#   - voip_stack talks directly to standard ESPHome microphone/speaker.
#   - Good starting point for hardware codecs/front-ends that already include
#     echo cancellation or DSP, such as XMOS-based boards.
#   - Also useful for simple native-I2S builds where you intentionally accept
#     no software echo cancellation, and for testing VoIP audio directions.
#
# Endpoint mode: full_duplex. Mic-only and speaker-only variants live in this
# same folder and remove the unrelated compile-time pieces.
#
# Tested hardware example:
#   - Generic ESP32-S3 with PSRAM.
#   - INMP441 microphone on one I2S controller.
#   - MAX98357A amplifier on a second I2S controller.
# This is only the tested wiring example; native ESPHome audio can also be
# adapted to hardware that exposes mic/speaker on a different bus layout.
# ESPHome native I2S does not add software AEC by itself.
#
# Tested wiring target: Freenove ESP32-S3-WROOM CAM / FNK0085, camera removed.
# Change pins for your board.
#
# INMP441:
#   SCK/BCLK -> GPIO9
#   WS/LRCLK -> GPIO10
#   SD/DOUT  -> GPIO11
#   L/R      -> 3V3, tied locally with VDD so the mic outputs right channel
#   VDD      -> 3V3
#   GND      -> GND
#
# MAX98357A:
#   BCLK     -> GPIO12
#   LRC/WS   -> GPIO13
#   DIN      -> GPIO14
#   SD/EN    -> GPIO18 (optional; tie high instead if your module prefers)
#   VIN      -> 5V or 3V3 according to your MAX98357A module
#   GND      -> GND
# =============================================================================

# =============================================================================
# PROJECT SETUP
# =============================================================================

substitutions:
  name: generic-native-int-fd
  friendly_name: Generic S3 Native VoIP Full Duplex

  ext_components_source: "github://n-IA-hane/esphome-intercom@main"
  voip_stack_components_source: "github://n-IA-hane/esphome-voip-stack@main"
  audio_stack_components_source: "github://n-IA-hane/esphome-audio-stack@main"

  # ---------------------------------------------------------------------------
  # Pinout: native microphone I2S bus
  # ---------------------------------------------------------------------------
  rx_i2s_bclk_pin: GPIO9
  rx_i2s_lrclk_pin: GPIO10
  rx_i2s_din_pin: GPIO11

  # ---------------------------------------------------------------------------
  # Pinout: native speaker I2S bus
  # ---------------------------------------------------------------------------
  tx_i2s_bclk_pin: GPIO12
  tx_i2s_lrclk_pin: GPIO13
  tx_i2s_dout_pin: GPIO14

  # ---------------------------------------------------------------------------
  # Pinout: speaker amplifier and optional status LED
  # ---------------------------------------------------------------------------
  speaker_enable_pin: GPIO18
  status_led_pin: GPIO48
  status_led_chipset: WS2812
  status_led_rgb_order: GRB

# Shared voip-only packages: SIP transport, HA phonebook sync, API actions,
# basic buttons and native diagnostics.

packages:
  core: github://n-IA-hane/esphome-intercom/packages/voip_only.yaml@main
  voip_transport: github://n-IA-hane/esphome-intercom/packages/voip/transport.yaml@main
  phonebook_subscribe: github://n-IA-hane/esphome-intercom/packages/voip/phonebook_subscribe.yaml@main
  ha_api: github://n-IA-hane/esphome-intercom/packages/voip/ha_api.yaml@main
  voip_entities_full_duplex: github://n-IA-hane/esphome-intercom/packages/voip/entities_full_duplex.yaml@main
  call_settings: github://n-IA-hane/esphome-intercom/packages/voip/call_settings.yaml@main
  speaker_mute: github://n-IA-hane/esphome-intercom/packages/voip/speaker_mute.yaml@main
  s3_base: github://n-IA-hane/esphome-intercom/packages/platform/esp32s3_base.yaml@main
  native_diagnostics: github://n-IA-hane/esphome-intercom/packages/diagnostics/native.yaml@main
  status_led: github://n-IA-hane/esphome-intercom/packages/led/status_ws2812_single.yaml@main

# =============================================================================
# BOARD AND MEMORY
# =============================================================================

# Device identity exposed to ESPHome, HA and the shared VoIP phonebook.

esphome:
  name: ${name}
  friendly_name: ${friendly_name}

# ESP32-S3 native-audio target. This profile is intentionally independent from
# esp_audio_stack, so it validates the standard ESPHome audio contracts.

esp32:
  board: freenove_esp32_s3_wroom
  variant: esp32s3
  flash_size: 8MB
  framework:
    type: esp-idf
    advanced:
      compiler_optimization: PERF
    sdkconfig_options:
      CONFIG_MBEDTLS_EXTERNAL_MEM_ALLOC: "y"
      CONFIG_ESP_CONSOLE_USB_SERIAL_JTAG: "y"
      CONFIG_ESP_CONSOLE_SECONDARY_NONE: "y"

# PSRAM gives VoIP tasks and buffers headroom even on the native ESPHome path.

psram:
  mode: octal
  speed: 80MHz

# Persist user settings without excessive flash writes.

preferences:
  flash_write_interval: 5min

# =============================================================================
# EXTERNAL COMPONENTS
# =============================================================================

# Project components used by the native path: voip_stack plus the shared
# internal VoIP task/ring-buffer helpers, without esp_audio_stack.

external_components:
  - source: ${voip_stack_components_source}
    components: [voip_stack]

# =============================================================================
# CONNECTIVITY
# =============================================================================

# Normal Wi-Fi client mode. Power save is disabled for low-latency SIP/RTP and
# reliable HA API updates.

wifi:
  ssid: !secret wifi_ssid
  password: !secret wifi_password
  power_save_mode: NONE

# OTA update endpoint used by ESPHome Dashboard and `esphome upload`.

ota:
  - platform: esphome

# USB Serial/JTAG logger. VoIP domains stay visible at INFO; noisy entity
# domains stay WARN for normal field use.

logger:
  hardware_uart: USB_SERIAL_JTAG
  level: INFO
  logs:
    voip_stack: INFO
    voip_stack.fsm: INFO
    voip_stack.tcp: INFO
    voip_stack.udp: INFO
    voip_stack.audio: INFO
    voip_stack.settings: INFO
    sensor: WARN
    text_sensor: WARN
    binary_sensor: WARN
    switch: WARN
    number: WARN
    button: WARN
    api: WARN
    api.connection: WARN
    component: WARN
    i2s_audio: INFO
    microphone: INFO
    speaker: INFO

# =============================================================================
# BUSES
# =============================================================================

# Two native ESPHome I2S buses: RX for microphone and TX for speaker.

i2s_audio:
  - id: rx_i2s
    i2s_bclk_pin: ${rx_i2s_bclk_pin}
    i2s_lrclk_pin: ${rx_i2s_lrclk_pin}

  - id: tx_i2s
    i2s_bclk_pin: ${tx_i2s_bclk_pin}
    i2s_lrclk_pin: ${tx_i2s_lrclk_pin}

# =============================================================================
# OUTPUTS AND LIGHTS
# =============================================================================

# Optional amplifier enable pin for MAX98357A-style boards.

output:
  - platform: gpio
    id: speaker_enable
    pin: ${speaker_enable_pin}

# =============================================================================
# AUDIO PIPELINE
# =============================================================================

# Native ESPHome microphone. INMP441 is strapped to the right slot. Keep the
# wire as stereo and let MicrophoneSource select channel 1; this matches the
# stable esp_audio_stack path and avoids mono-slot silence on some IDF/board
# combinations.

microphone:
  - platform: i2s_audio
    id: mic_main
    adc_type: external
    i2s_audio_id: rx_i2s
    i2s_din_pin: ${rx_i2s_din_pin}
    channel: stereo
    sample_rate: 48000
    bits_per_sample: 32bit
    correct_dc_offset: true

# Native ESPHome speaker. Keep it at VoIP wire format to avoid a
# resampler in this regression profile.

speaker:
  - platform: i2s_audio
    id: hw_speaker
    dac_type: external
    i2s_audio_id: tx_i2s
    i2s_dout_pin: ${tx_i2s_dout_pin}
    channel: mono
    sample_rate: 48000
    bits_per_sample: 16bit
    buffer_duration: 128ms
    timeout: 500ms

# =============================================================================
# MEDIA, ASSIST AND VOIP
# =============================================================================

# Local SIP phone in full-duplex mode. It advertises 48 kHz/10 ms PCM because
# both native mic and speaker are configured at that rate.

voip_stack:
  id: phone
  task_stacks_in_psram: true
  buffers_in_psram: true
  audio:
    tx:
      sample_rate: 48000
      pcm_format: s16le
      channels: 1
      frame_ms: 10
    rx:
      sample_rate: 48000
      pcm_format: s16le
      channels: 1
      frame_ms: 10
  microphone_source:
    microphone: mic_main
    bits_per_sample: 16
    channels: [1]
    gain_factor: 1
  speaker: hw_speaker
  dc_offset_removal: false
  ringing_timeout: 30s
  delete_contact_missing_from:
    updates_number: 1

# =============================================================================
# RUNTIME LOGIC AND ENTITIES
# =============================================================================

# User-facing amp power switch. It only gates the hardware enable pin.

switch:
  - platform: output
    id: speaker_power
    name: Speaker Power
    output: speaker_enable
    restore_mode: RESTORE_DEFAULT_ON