// This file is part of the 64k demo project.
// It manages the low-level audio device and high-level audio state.
// Uses backend abstraction for testability.

#include "audio.h"
#include "audio_backend.h"
#include "backend/miniaudio_backend.h"
#include "ring_buffer.h"
#include "synth.h"
#include "util/asset_manager.h"

#define MINIAUDIO_IMPLEMENTATION
#include "miniaudio.h"

#include <stdio.h>

// Global ring buffer for audio streaming
static AudioRingBuffer g_ring_buffer;

// Pending write buffer for partially written samples
// Maximum size: one chunk (533 frames @ 60fps = 1066 samples stereo)
#define MAX_PENDING_SAMPLES 2048
static float g_pending_buffer[MAX_PENDING_SAMPLES];
static int g_pending_samples = 0; // How many samples are waiting to be written

// Global backend pointer for audio abstraction
static AudioBackend* g_audio_backend = nullptr;
static MiniaudioBackend g_default_backend;
static bool g_using_default_backend = false;

#if !defined(STRIP_ALL)
// Allow tests to inject a custom backend
void audio_set_backend(AudioBackend* backend) {
  g_audio_backend = backend;
}

// Get current backend (for tests)
AudioBackend* audio_get_backend() {
  return g_audio_backend;
}
#endif /* !defined(STRIP_ALL) */

void audio_init() {
  // Note: synth_init() must be called separately before using audio system.
  // In production code, use AudioEngine::init() which manages initialization
  // order.

  // Clear pending buffer
  g_pending_samples = 0;

  // Use default backend if none set
  if (g_audio_backend == nullptr) {
    g_audio_backend = &g_default_backend;
    g_using_default_backend = true;
  }

  g_audio_backend->init();
}

void audio_start() {
  if (g_audio_backend == nullptr) {
    printf("Cannot start: audio not initialized.\n");
    return;
  }
  g_audio_backend->start();
}

void audio_render_ahead(float music_time, float dt) {
  // Target: maintain look-ahead buffer
  const float target_lookahead = (float)RING_BUFFER_LOOKAHEAD_MS / 1000.0f;

  // Render in small chunks to keep synth time synchronized with tracker
  // Chunk size: one frame's worth of audio (~16.6ms @ 60fps)
  const int chunk_frames = (int)(dt * RING_BUFFER_SAMPLE_RATE);
  const int chunk_samples = chunk_frames * RING_BUFFER_CHANNELS;

  if (chunk_frames <= 0)
    return;

  // Keep rendering small chunks until buffer is full enough
  while (true) {
    // First, try to flush any pending samples from previous partial writes
    if (g_pending_samples > 0) {
      const int written =
          g_ring_buffer.write(g_pending_buffer, g_pending_samples);

      if (written > 0) {
        // Some or all samples were written
        // Move remaining samples to front of buffer
        const int remaining = g_pending_samples - written;
        if (remaining > 0) {
          for (int i = 0; i < remaining; ++i) {
            g_pending_buffer[i] = g_pending_buffer[written + i];
          }
        }
        g_pending_samples = remaining;

        // Notify backend (for testing/tracking)
#if !defined(STRIP_ALL)
        if (g_audio_backend != nullptr) {
          g_audio_backend->on_frames_rendered(written / RING_BUFFER_CHANNELS);
        }
#endif
      }

      // If still have pending samples, buffer is full - wait for consumption
      if (g_pending_samples > 0)
        break;
    }

    // Check current buffer state
    const int buffered_samples = g_ring_buffer.available_read();
    const float buffered_time =
        (float)buffered_samples /
        (RING_BUFFER_SAMPLE_RATE * RING_BUFFER_CHANNELS);

    // Stop if buffer is full enough
    if (buffered_time >= target_lookahead)
      break;

    // Check available space and render chunk that fits
    const int available_space = g_ring_buffer.available_write();
    if (available_space == 0) {
      // Buffer is completely full, wait for audio callback to consume
      break;
    }

    // Get direct write pointer from ring buffer
    int available_for_write = 0;
    float* write_ptr = g_ring_buffer.get_write_region(&available_for_write);

    if (available_for_write == 0) {
      break; // Buffer full, wait for consumption
    }

    // Clamp to desired chunk size
    const int actual_samples =
        (available_for_write < chunk_samples) ? available_for_write
                                              : chunk_samples;
    const int actual_frames = actual_samples / RING_BUFFER_CHANNELS;

    // Render directly to ring buffer (NO COPY, NO ALLOCATION)
    synth_render(write_ptr, actual_frames);

    // Apply clipping in-place (Phase 2: ensure samples stay in [-1.0, 1.0])
    for (int i = 0; i < actual_samples; ++i) {
      if (write_ptr[i] > 1.0f)
        write_ptr[i] = 1.0f;
      if (write_ptr[i] < -1.0f)
        write_ptr[i] = -1.0f;
    }

    // Commit written data atomically
    g_ring_buffer.commit_write(actual_samples);

    // Notify backend of frames rendered
#if !defined(STRIP_ALL)
    if (g_audio_backend != nullptr) {
      g_audio_backend->on_frames_rendered(actual_frames);
    }
#endif

    // Handle wrap-around: if we wanted more samples but ring wrapped,
    // get a second region and render remaining chunk
    if (actual_samples < chunk_samples) {
      int second_avail = 0;
      float* second_ptr = g_ring_buffer.get_write_region(&second_avail);
      if (second_avail > 0) {
        const int remaining_samples = chunk_samples - actual_samples;
        const int second_samples =
            (second_avail < remaining_samples) ? second_avail
                                               : remaining_samples;
        const int second_frames = second_samples / RING_BUFFER_CHANNELS;

        synth_render(second_ptr, second_frames);

        // Apply clipping to wrap-around region
        for (int i = 0; i < second_samples; ++i) {
          if (second_ptr[i] > 1.0f)
            second_ptr[i] = 1.0f;
          if (second_ptr[i] < -1.0f)
            second_ptr[i] = -1.0f;
        }

        g_ring_buffer.commit_write(second_samples);

        // Notify backend of additional frames
#if !defined(STRIP_ALL)
        if (g_audio_backend != nullptr) {
          g_audio_backend->on_frames_rendered(second_frames);
        }
#endif
      }
    }
  }
}

float audio_get_playback_time() {
  const int64_t total_samples = g_ring_buffer.get_total_read();
  return (float)total_samples /
         (RING_BUFFER_SAMPLE_RATE * RING_BUFFER_CHANNELS);
}

float audio_get_render_time() {
  const int64_t total_samples = g_ring_buffer.get_total_written();
  return (float)total_samples /
         (RING_BUFFER_SAMPLE_RATE * RING_BUFFER_CHANNELS);
}

float audio_get_realtime_peak() {
  if (g_audio_backend == nullptr) {
    return 0.0f;
  }
  return g_audio_backend->get_realtime_peak();
}

// Expose ring buffer for backends
AudioRingBuffer* audio_get_ring_buffer() {
  return &g_ring_buffer;
}

#if !defined(STRIP_ALL)
void audio_render_silent(float duration_sec) {
  const int sample_rate = 32000;
  const int chunk_size = 512;
  int total_frames = (int)(duration_sec * sample_rate);
  float buffer[chunk_size * 2]; // Stereo

  while (total_frames > 0) {
    int frames_to_render =
        (total_frames > chunk_size) ? chunk_size : total_frames;
    synth_render(buffer, frames_to_render);
    total_frames -= frames_to_render;

    // Notify backend of frames rendered (for mock tracking)
    if (g_audio_backend != nullptr) {
      g_audio_backend->on_frames_rendered(frames_to_render);
    }
  }
}
#endif /* !defined(STRIP_ALL) */

void audio_update() {
}

void audio_shutdown() {
  if (g_audio_backend != nullptr) {
    g_audio_backend->shutdown();
  }
  synth_shutdown();

  // Clear backend pointer if using default
  if (g_using_default_backend) {
    g_audio_backend = nullptr;
    g_using_default_backend = false;
  }
}