#include "player.h"

#include <string.h>

#include "aes67_cfg.h"
#include "aes67_tx.h"
#include "aes67_web.h"
#include "audio_ring.h"
#include "decoder.h"
#include "hls.h"
#include "esp_log.h"
#include "freertos/FreeRTOS.h"
#include "freertos/task.h"

#define RATE            48000
#define CHANNELS        2
#define RING_FRAMES     (4 * RATE)      // 4 s in PSRAM
#define PREFILL_FRAMES  (RATE)          // 1 s before (re)starting output

static const char *TAG = "player";

typedef enum { SRC_TONE, SRC_OFF, SRC_HLS, SRC_SPOTIFY } src_t;
static const char *const SRC_NAME[] = { "tone", "off", "hls", "spotify" };

static volatile src_t s_src = SRC_OFF;
static volatile bool s_playing;         // ring output running (after prefill)
static volatile bool s_flush;           // consumer drops buffered audio on the next read

static src_t mode_of(const char *m)
{
    return !strcmp(m, "tone") ? SRC_TONE : !strcmp(m, "hls") ? SRC_HLS :
           !strcmp(m, "spotify") ? SRC_SPOTIFY : !strcmp(m, "auto") ? SRC_HLS /* failover: step 7 */ : SRC_OFF;
}

// AES67 TX pull callback (TX task, must not block). Silence while buffering or off: not an underrun.
static size_t player_read(int32_t *buf, size_t frames)
{
    if (s_flush) {
        audio_ring_flush();
        s_flush = false;
        s_playing = false;
    }
    if (s_src == SRC_OFF || s_src == SRC_SPOTIFY) {     // Spotify: step 7 (cspot)
        memset(buf, 0, frames * CHANNELS * sizeof(int32_t));
        return frames;
    }
    if (!s_playing) {
        if (audio_ring_level() < PREFILL_FRAMES) {
            memset(buf, 0, frames * CHANNELS * sizeof(int32_t));
            return frames;
        }
        s_playing = true;
    }
    size_t got = audio_ring_read(buf, frames);
    if (got < frames) {
        s_playing = false;          // ran dry: underrun (counted by TX), prefill again
    }
    return got;
}

void player_apply(const cJSON *source)
{
    src_t src = mode_of(cJSON_GetObjectItemCaseSensitive(source, "mode")->valuestring);
    if (src == s_src) {
        return;
    }
    s_src = src;
    s_flush = true;
    // The test tone is TX's own (phase-locked to PTP); everything else goes through player_read.
    aes67_tx_set_source(src == SRC_TONE ? NULL : player_read);
    ESP_LOGI(TAG, "source: %s", SRC_NAME[src]);
}

// Source side (HLS task): write converted 48 kHz frames, waiting for space at playback speed.
// Gives up when the source is no longer HLS.
size_t player_write(const int32_t *frames, size_t n)
{
    size_t done = 0;
    while (done < n) {
        if (s_src != SRC_HLS) {
            return done;
        }
        size_t w = audio_ring_write(frames + done * CHANNELS, n - done);
        if (!w) {
            vTaskDelay(pdMS_TO_TICKS(20));
        }
        done += w;
    }
    return done;
}

static void player_status(cJSON *st)
{
    const char *state = s_src == SRC_TONE ? "playing" : s_src == SRC_OFF ? "idle" :
                        s_src == SRC_SPOTIFY ? "not implemented" : s_playing ? "playing" : "buffering";
    cJSON_AddStringToObject(st, "active_source", SRC_NAME[s_src]);
    cJSON_AddStringToObject(st, "source_state", state);
    cJSON_AddStringToObject(st, "spotify_state", "not implemented");
    cJSON_AddNumberToObject(st, "buffer_ms", (double)(audio_ring_level() * 1000 / RATE));
}

esp_err_t player_init(void)
{
    esp_err_t err = audio_ring_init(RING_FRAMES, CHANNELS);
    if (err != ESP_OK) {
        ESP_LOGE(TAG, "ring buffer: %s", esp_err_to_name(err));
        return err;
    }
    cJSON *src = cfg_get("source");
    s_src = (src_t)-1;
    player_apply(src);
    cJSON_Delete(src);
    err = decoder_init();
    if (err != ESP_OK) {
        return err;
    }
    err = hls_start(decoder_feed);
    if (err != ESP_OK) {
        return err;
    }
    return status_register(player_status);
}