#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/video/gstvideometa.h>
#include <gst/app/app.h>
#include <cstring>
#include <iostream>
#include <cassert>
#include <sstream>

#define CHUNK_SIZE 1024   /* Amount of bytes we are sending in each buffer */
#define SAMPLE_RATE 44100 /* Samples per second we are sending */

/* Structure to contain all our information, so we can pass it to callbacks */
typedef struct _CustomData {
    GstElement *pipeline, *app_source, *tee, *audio_queue, *audio_convert1, *audio_resample, *audio_sink;
    GstElement *video_queue, *audio_convert2, *visual, *video_convert, *video_sink;
    GstElement *app_queue, *app_sink;

    guint64 num_samples;   /* Number of samples generated so far (for timestamp generation) */
    gfloat a, b, c, d;     /* For waveform generation */

    guint sourceid;        /* To control the GSource */

    GMainLoop *main_loop;  /* GLib's Main Loop */
} CustomData;

/* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc.
 * The idle handler is added to the mainloop when appsrc requests us to start sending data (need-data signal)
 * and is removed when appsrc has enough data (enough-data signal).
 */
static gboolean push_data(CustomData *data) {
    GstBuffer *buffer = nullptr;
    GstFlowReturn ret;
    int i;
    GstMapInfo map;
    gint16 *raw;
    gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */
    gfloat freq;

    /* Create a new empty buffer */
    buffer = gst_buffer_new_and_alloc (CHUNK_SIZE);

    /* Set its timestamp and duration */
            GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale(data->num_samples, GST_SECOND, SAMPLE_RATE);
    GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale(num_samples, GST_SECOND, SAMPLE_RATE);

    static int my_counter_ = 0;
    my_counter_ += 1;
    std::string dont_blame_me = "abc fuck this " + std::to_string(my_counter_) + "X";
    auto my_meta = gst_buffer_add_video_caption_meta(
            buffer, GST_VIDEO_CAPTION_TYPE_CEA608_RAW, (unsigned char *) dont_blame_me.c_str(), dont_blame_me.size());
    // std::cout << "My GstMeta: " << my_meta << std::endl;

    auto tc = gst_video_time_code_new(
            30, 1, nullptr, GST_VIDEO_TIME_CODE_FLAGS_NONE, 12, 12, 12, 12, 0);
    auto my_meta_2 = gst_buffer_add_video_time_code_meta(buffer, tc);

    /* Generate some psychodelic waveforms */
    gst_buffer_map(buffer, &map, GST_MAP_WRITE);
    raw = (gint16 *) map.data;
    data->c += data->d;
    data->d -= data->c / 1000;
    freq = 1100 + 1000 * data->d;
    for (i = 0; i < num_samples; i++) {
        data->a += data->b;
        data->b -= data->a / freq;
        raw[i] = (gint16) (500 * data->a);
    }
    gst_buffer_unmap(buffer, &map);
    data->num_samples += num_samples;

    /* Push the buffer into the appsrc */
    g_signal_emit_by_name(data->app_source, "push-buffer", buffer, &ret);

    /* Free the buffer now that we are done with it */
    gst_buffer_unref(buffer);

    if (ret != GST_FLOW_OK) {
        /* We got some error, stop sending data */
        return FALSE;
    }

    return TRUE;
}

/* This signal callback triggers when appsrc needs data. Here, we add an idle handler
 * to the mainloop to start pushing data into the appsrc */
static void start_feed(GstElement *source, guint size, CustomData *data) {
    if (data->sourceid == 0) {
        g_print("Start feeding\n");
        data->sourceid = g_idle_add((GSourceFunc) push_data, data);
    }
}

/* This callback triggers when appsrc has enough data and we can stop sending.
 * We remove the idle handler from the mainloop */
static void stop_feed(GstElement *source, CustomData *data) {
    if (data->sourceid != 0) {
        g_print("Stop feeding\n");
        g_source_remove(data->sourceid);
        data->sourceid = 0;
    }
}

/* The appsink has received a buffer */
static GstFlowReturn new_sample(GstElement *sink, CustomData *data) {
    GstSample *sample;

    /* Retrieve the buffer */
    g_signal_emit_by_name(sink, "pull-sample", &sample);
    if (sample) {
        /* The only thing we do in this example is print a * to indicate a received buffer */
        g_print("*");

        GstBuffer *buffer = gst_sample_get_buffer(sample);
        const auto &n_memory = gst_buffer_n_memory(buffer);
        std::cout << "n_memory = " << n_memory << std::endl;
        std::cout << "buffer->pts = " << buffer->pts << std::endl;
        std::cout << "buffer->dts = " << buffer->dts << std::endl;
        std::cout << "buffer->duration = " << buffer->duration << std::endl;
        std::cout << "buffer->offset = " << buffer->offset << std::endl;
        std::cout << "buffer->offset_end = " << buffer->offset_end << std::endl;
        GstMeta *gst_meta;
        gpointer state = nullptr;
        while ((gst_meta = gst_buffer_iterate_meta(buffer, &state))) {
            if (gst_meta->info == gst_video_caption_meta_get_info()) {
                auto specific_meta = (GstVideoCaptionMeta *) gst_meta;
                if (specific_meta) {
                    auto x = (const char *) (specific_meta->data);
                    std::cout << "MetaInfo is recognized to be [GstVideoCaptionMeta]"
                              << "caption = " << std::string(x, specific_meta->size)
                              << std::endl;
                }
            } else if (gst_meta->info == gst_video_time_code_meta_get_info()) {
                auto specific_meta = (GstVideoTimeCodeMeta *) gst_meta;
                if (specific_meta) {
                    std::cout << "MetaInfo is recognized to be [GstVideoTimeCodeMeta]"
                              << " h = " << specific_meta->tc.hours
                              << " m = " << specific_meta->tc.minutes
                              << " s = " << specific_meta->tc.seconds
                              << " f = " << specific_meta->tc.frames
                              << std::endl;
                }
            } else {
                std::cout << "GstMetaInfo is not recognized."
                          << " info = " << gst_meta->info
                          << " api = " << gst_meta->info->api
                          << std::endl;
            }
        }
        gst_sample_unref(sample);
        return GST_FLOW_OK;
    }
    return GST_FLOW_ERROR;
}

/* This function is called when an error message is posted on the bus */
static void error_cb(GstBus *bus, GstMessage *msg, CustomData *data) {
    GError *err;
    gchar *debug_info;

    /* Print error details on the screen */
    gst_message_parse_error(msg, &err, &debug_info);
    g_printerr("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message);
    g_printerr("Debugging information: %s\n", debug_info ? debug_info : "none");
    g_clear_error(&err);
    g_free(debug_info);

    g_main_loop_quit(data->main_loop);
}

int main(int argc, char *argv[]) {
    CustomData data;
    GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad;
    GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
    GstAudioInfo info;
    GstCaps *audio_caps;
    GstBus *bus;

    /* Initialize custom data structure */
    memset(&data, 0, sizeof(data));
    data.b = 1; /* For waveform generation */
    data.d = 1;

    /* Initialize GStreamer */
    gst_init(&argc, &argv);

    /* Create the elements */
    data.app_source = gst_element_factory_make("appsrc", "audio_source");
    data.tee = gst_element_factory_make("tee", "tee");
    data.audio_queue = gst_element_factory_make("queue", "audio_queue");
    data.audio_convert1 = gst_element_factory_make("audioconvert", "audio_convert1");
    data.audio_resample = gst_element_factory_make("audioresample", "audio_resample");
    data.audio_sink = gst_element_factory_make("autoaudiosink", "audio_sink");
    data.video_queue = gst_element_factory_make("queue", "video_queue");
    data.audio_convert2 = gst_element_factory_make("audioconvert", "audio_convert2");
    data.visual = gst_element_factory_make("wavescope", "visual");
    data.video_convert = gst_element_factory_make("videoconvert", "video_convert");
    data.video_sink = gst_element_factory_make("autovideosink", "video_sink");
    data.app_queue = gst_element_factory_make("queue", "app_queue");
    data.app_sink = gst_element_factory_make("appsink", "app_sink");

    /* Create the empty pipeline */
    data.pipeline = gst_pipeline_new("test-pipeline");

    if (!data.pipeline || !data.app_source || !data.tee || !data.audio_queue || !data.audio_convert1 ||
        !data.audio_resample || !data.audio_sink || !data.video_queue || !data.audio_convert2 || !data.visual ||
        !data.video_convert || !data.video_sink || !data.app_queue || !data.app_sink) {
        g_printerr("Not all elements could be created.\n");
        return -1;
    }

    /* Configure wavescope */
    g_object_set(data.visual, "shader", 0, "style", 0, NULL);

    /* Configure appsrc */
    gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
    audio_caps = gst_audio_info_to_caps(&info);
    g_object_set(data.app_source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
    g_signal_connect (data.app_source, "need-data", G_CALLBACK(start_feed), &data);
    g_signal_connect (data.app_source, "enough-data", G_CALLBACK(stop_feed), &data);

    /* Configure appsink */
    g_object_set(data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
    g_signal_connect (data.app_sink, "new-sample", G_CALLBACK(new_sample), &data);
    gst_caps_unref(audio_caps);

    /* Link all elements that can be automatically linked because they have "Always" pads */
    gst_bin_add_many(GST_BIN (data.pipeline), data.app_source, data.tee, data.audio_queue, data.audio_convert1,
                     data.audio_resample,
                     data.audio_sink, data.video_queue, data.audio_convert2, data.visual, data.video_convert,
                     data.video_sink, data.app_queue,
                     data.app_sink, NULL);
    if (gst_element_link_many(data.app_source, data.tee, NULL) != TRUE ||
        gst_element_link_many(data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, NULL) !=
        TRUE ||
        gst_element_link_many(data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink,
                              NULL) != TRUE ||
        gst_element_link_many(data.app_queue, data.app_sink, NULL) != TRUE) {
        g_printerr("Elements could not be linked.\n");
        gst_object_unref(data.pipeline);
        return -1;
    }

    /* Manually link the Tee, which has "Request" pads */
    tee_audio_pad = gst_element_get_request_pad(data.tee, "src_%u");
    g_print("Obtained request pad %s for audio branch.\n", gst_pad_get_name (tee_audio_pad));
    queue_audio_pad = gst_element_get_static_pad(data.audio_queue, "sink");
    tee_video_pad = gst_element_get_request_pad(data.tee, "src_%u");
    g_print("Obtained request pad %s for video branch.\n", gst_pad_get_name (tee_video_pad));
    queue_video_pad = gst_element_get_static_pad(data.video_queue, "sink");
    tee_app_pad = gst_element_get_request_pad(data.tee, "src_%u");
    g_print("Obtained request pad %s for app branch.\n", gst_pad_get_name (tee_app_pad));
    queue_app_pad = gst_element_get_static_pad(data.app_queue, "sink");
    if (gst_pad_link(tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK ||
        gst_pad_link(tee_video_pad, queue_video_pad) != GST_PAD_LINK_OK ||
        gst_pad_link(tee_app_pad, queue_app_pad) != GST_PAD_LINK_OK) {
        g_printerr("Tee could not be linked\n");
        gst_object_unref(data.pipeline);
        return -1;
    }
    gst_object_unref(queue_audio_pad);
    gst_object_unref(queue_video_pad);
    gst_object_unref(queue_app_pad);

    /* Instruct the bus to emit signals for each received message, and connect to the interesting signals */
    bus = gst_element_get_bus(data.pipeline);
    gst_bus_add_signal_watch(bus);
    g_signal_connect (G_OBJECT(bus), "message::error", (GCallback) error_cb, &data);
    gst_object_unref(bus);

    /* Start playing the pipeline */
    gst_element_set_state(data.pipeline, GST_STATE_PLAYING);

    /* Create a GLib Main Loop and set it to run */
    data.main_loop = g_main_loop_new(NULL, FALSE);
    g_main_loop_run(data.main_loop);

    /* Release the request pads from the Tee, and unref them */
    gst_element_release_request_pad(data.tee, tee_audio_pad);
    gst_element_release_request_pad(data.tee, tee_video_pad);
    gst_element_release_request_pad(data.tee, tee_app_pad);
    gst_object_unref(tee_audio_pad);
    gst_object_unref(tee_video_pad);
    gst_object_unref(tee_app_pad);

    /* Free resources */
    gst_element_set_state(data.pipeline, GST_STATE_NULL);
    gst_object_unref(data.pipeline);
    return 0;
}