/***
  This file is part of PulseAudio.

  Copyright 2016 Arun Raghavan <mail@arunraghavan.net>

  PulseAudio is free software; you can redistribute it and/or modify
  it under the terms of the GNU Lesser General Public License as published
  by the Free Software Foundation; either version 2.1 of the License,
  or (at your option) any later version.

  PulseAudio is distributed in the hope that it will be useful, but
  WITHOUT ANY WARRANTY; without even the implied warranty of
  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  General Public License for more details.

  You should have received a copy of the GNU Lesser General Public License
  along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
***/

#ifdef HAVE_CONFIG_H
#include <config.h>
#endif

#include <pulse/timeval.h>
#include <pulsecore/fdsem.h>
#include <pulsecore/core-rtclock.h>

#include "rtp.h"

#include <gio/gio.h>

#include <gst/gst.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include <gst/rtp/gstrtpbuffer.h>

#define MAKE_ELEMENT_NAMED(v, e, n)                     \
    v = gst_element_factory_make(e, n);                 \
    if (!v) {                                           \
        pa_log("Could not create %s element", e);       \
        goto fail;                                      \
    }

#define MAKE_ELEMENT(v, e) MAKE_ELEMENT_NAMED((v), (e), NULL)

struct pa_rtp_context {
    pa_fdsem *fdsem;
    pa_sample_spec ss;

    GstElement *pipeline;
    GstElement *appsrc;
    GstElement *appsink;

    uint32_t last_timestamp;
};

static GstCaps* caps_from_sample_spec(const pa_sample_spec *ss) {
    if (ss->format != PA_SAMPLE_S16BE)
        return NULL;

    return gst_caps_new_simple("audio/x-raw",
            "format", G_TYPE_STRING, "S16BE",
            "rate", G_TYPE_INT, (int) ss->rate,
            "channels", G_TYPE_INT, (int) ss->channels,
            "layout", G_TYPE_STRING, "interleaved",
            NULL);
}
static bool init_send_pipeline(pa_rtp_context *c, int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss) {
    GstElement *appsrc = NULL, *pay = NULL, *capsf = NULL, *rtpbin = NULL, *sink = NULL;
    GstCaps *caps;

    MAKE_ELEMENT(appsrc, "appsrc");
    MAKE_ELEMENT(pay, "rtpL16pay");
    MAKE_ELEMENT(capsf, "capsfilter");
    MAKE_ELEMENT(rtpbin, "rtpbin");
    MAKE_ELEMENT(sink, "fdsink");

    c->pipeline = gst_pipeline_new(NULL);

    gst_bin_add_many(GST_BIN(c->pipeline), appsrc, pay, capsf, rtpbin, sink, NULL);

    caps = caps_from_sample_spec(ss);
    if (!caps) {
        pa_log("Unsupported format to payload");
        goto fail;
    }

    g_object_set(appsrc, "caps", caps, "is-live", TRUE, "blocksize", mtu, "format", 3 /* time */, NULL);
    g_object_set(pay, "mtu", mtu, NULL);
    g_object_set(sink, "fd", fd, "enable-last-sample", FALSE, NULL);

    gst_caps_unref(caps);

    /* Force the payload type that we want */
    caps = gst_caps_new_simple("application/x-rtp", "payload", G_TYPE_INT, (int) payload, NULL);
    g_object_set(capsf, "caps", caps, NULL);
    gst_caps_unref(caps);

    if (!gst_element_link(appsrc, pay) ||
        !gst_element_link(pay, capsf) ||
        !gst_element_link_pads(capsf, "src", rtpbin, "send_rtp_sink_0") ||
        !gst_element_link_pads(rtpbin, "send_rtp_src_0", sink, "sink")) {

        pa_log("Could not set up send pipeline");
        goto fail;
    }

    if (gst_element_set_state(c->pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
        pa_log("Could not start pipeline");
        goto fail;
    }

    c->appsrc = gst_object_ref(appsrc);

    return true;

fail:
    if (c->pipeline) {
        gst_object_unref(c->pipeline);
    } else {
        /* These weren't yet added to pipeline, so we still have a ref */
        if (appsrc)
            gst_object_unref(appsrc);
        if (pay)
            gst_object_unref(pay);
        if (capsf)
            gst_object_unref(capsf);
        if (rtpbin)
            gst_object_unref(rtpbin);
        if (sink)
            gst_object_unref(sink);
    }

    return false;
}

pa_rtp_context* pa_rtp_context_new_send(int fd, uint8_t payload, size_t mtu, const pa_sample_spec *ss) {
    pa_rtp_context *c = NULL;
    GError *error = NULL;

    pa_assert(fd >= 0);

    c = pa_xnew0(pa_rtp_context, 1);

    c->ss = *ss;

    if (!gst_init_check(NULL, NULL, &error)) {
        pa_log_error("Could not initialise GStreamer: %s", error->message);
        g_error_free(error);
        goto fail;
    }

    if (!init_send_pipeline(c, fd, payload, mtu, ss))
        goto fail;

    return c;

fail:
    pa_rtp_context_free(c);
    return NULL;
}

/* Called from I/O thread context */
static bool process_bus_messages(pa_rtp_context *c) {
    GstBus *bus;
    GstMessage *message;
    bool ret = true;

    bus = gst_pipeline_get_bus(GST_PIPELINE(c->pipeline));

    while (ret && (message = gst_bus_pop(bus))) {
        if (GST_MESSAGE_TYPE(message) == GST_MESSAGE_ERROR) {
            GError *error = NULL;

            ret = false;

            gst_message_parse_error(message, &error, NULL);
            pa_log("Got an error: %s", error->message);

            g_error_free(error);
        }

        gst_message_unref(message);
    }

    gst_object_unref(bus);

    return ret;
}

static void free_buffer(pa_memblock *memblock) {
    pa_memblock_release(memblock);
    pa_memblock_unref(memblock);
}

/* Called from I/O thread context */
int pa_rtp_send(pa_rtp_context *c, pa_memblockq *q) {
    pa_memchunk chunk = { 0, };
    GstBuffer *buf;
    void *data;
    bool stop = false;
    int ret = 0;

    pa_assert(c);
    pa_assert(q);

    if (!process_bus_messages(c))
        return -1;

    while (!stop && pa_memblockq_peek(q, &chunk) == 0) {
        pa_assert(chunk.memblock);

        data = pa_memblock_acquire(chunk.memblock);

        buf = gst_buffer_new_wrapped_full(GST_MEMORY_FLAG_READONLY | GST_MEMORY_FLAG_PHYSICALLY_CONTIGUOUS,
                                          data, chunk.length, chunk.index, chunk.length, chunk.memblock,
                                          (GDestroyNotify) free_buffer);

        if (gst_app_src_push_buffer(GST_APP_SRC(c->appsrc), buf) != GST_FLOW_OK) {
            pa_log_error("Could not push buffer");
            stop = true;
            ret = -1;
        }

        pa_memblockq_drop(q, chunk.length);
    }

    return ret;
}

static GstCaps* rtp_caps_from_sample_spec(const pa_sample_spec *ss) {
    if (ss->format != PA_SAMPLE_S16BE)
        return NULL;

    return gst_caps_new_simple("application/x-rtp",
            "media", G_TYPE_STRING, "audio",
            "encoding-name", G_TYPE_STRING, "L16",
            "clock-rate", G_TYPE_INT, (int) ss->rate,
            "payload", G_TYPE_INT, (int) pa_rtp_payload_from_sample_spec(ss),
            "layout", G_TYPE_STRING, "interleaved",
            NULL);
}

static void on_pad_added(GstElement *element, GstPad *pad, gpointer userdata) {
    pa_rtp_context *c = (pa_rtp_context *) userdata;
    GstElement *depay;
    GstPad *sinkpad;
    GstPadLinkReturn ret;

    depay = gst_bin_get_by_name(GST_BIN(c->pipeline), "depay");
    pa_assert(depay);

    sinkpad = gst_element_get_static_pad(depay, "sink");

    ret = gst_pad_link(pad, sinkpad);
    if (ret != GST_PAD_LINK_OK) {
        GstBus *bus;
        GError *error;

        bus = gst_pipeline_get_bus(GST_PIPELINE(c->pipeline));
        error = g_error_new(GST_CORE_ERROR, GST_CORE_ERROR_PAD, "Could not link rtpbin to depayloader");
        gst_bus_post(bus, gst_message_new_error(GST_OBJECT(c->pipeline), error, NULL));

        /* Actually cause the I/O thread to wake up and process the error */
        pa_fdsem_post(c->fdsem);

        g_error_free(error);
        gst_object_unref(bus);
    }

    gst_object_unref(sinkpad);
    gst_object_unref(depay);
}

static bool init_receive_pipeline(pa_rtp_context *c, int fd, const pa_sample_spec *ss) {
    GstElement *udpsrc = NULL, *rtpbin = NULL, *depay = NULL, *appsink = NULL;
    GstCaps *caps;
    GSocket *socket;
    GError *error = NULL;

    MAKE_ELEMENT(udpsrc, "udpsrc");
    MAKE_ELEMENT(rtpbin, "rtpbin");
    MAKE_ELEMENT_NAMED(depay, "rtpL16depay", "depay");
    MAKE_ELEMENT(appsink, "appsink");

    c->pipeline = gst_pipeline_new(NULL);

    gst_bin_add_many(GST_BIN(c->pipeline), udpsrc, rtpbin, depay, appsink, NULL);

    socket = g_socket_new_from_fd(fd, &error);
    if (error) {
        pa_log("Could not create socket: %s", error->message);
        g_error_free(error);
        goto fail;
    }

    caps = rtp_caps_from_sample_spec(ss);
    if (!caps) {
        pa_log("Unsupported format to payload");
        goto fail;
    }

    g_object_set(udpsrc, "socket", socket, "caps", caps, "auto-multicast" /* caller handles this */, FALSE, NULL);
    g_object_set(rtpbin, "latency", 0, "buffer-mode", 0 /* none */, NULL);
    g_object_set(appsink, "sync", FALSE, "enable-last-sample", FALSE, NULL);

    gst_caps_unref(caps);
    g_object_unref(socket);

    if (!gst_element_link_pads(udpsrc, "src", rtpbin, "recv_rtp_sink_0") ||
        !gst_element_link(depay, appsink)) {

        pa_log("Could not set up receive pipeline");
        goto fail;
    }

    g_signal_connect(G_OBJECT(rtpbin), "pad-added", G_CALLBACK(on_pad_added), c);

    if (gst_element_set_state(c->pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
        pa_log("Could not start pipeline");
        goto fail;
    }

    c->appsink = gst_object_ref(appsink);

    return true;

fail:
    if (c->pipeline) {
        gst_object_unref(c->pipeline);
    } else {
        /* These weren't yet added to pipeline, so we still have a ref */
        if (udpsrc)
            gst_object_unref(udpsrc);
        if (depay)
            gst_object_unref(depay);
        if (rtpbin)
            gst_object_unref(rtpbin);
        if (appsink)
            gst_object_unref(appsink);
    }

    return false;
}

/* Called from the GStreamer streaming thread */
static void appsink_eos(GstAppSink *appsink, gpointer userdata) {
    pa_rtp_context *c = (pa_rtp_context *) userdata;

    pa_fdsem_post(c->fdsem);
}

/* Called from the GStreamer streaming thread */
static GstFlowReturn appsink_new_sample(GstAppSink *appsink, gpointer userdata) {
    pa_rtp_context *c = (pa_rtp_context *) userdata;

    pa_fdsem_post(c->fdsem);

    return GST_FLOW_OK;
}

pa_rtp_context* pa_rtp_context_new_recv(int fd, uint8_t payload, const pa_sample_spec *ss) {
    pa_rtp_context *c = NULL;
    GstAppSinkCallbacks callbacks = { 0, };
    GError *error = NULL;

    pa_assert(fd >= 0);

    c = pa_xnew0(pa_rtp_context, 1);

    c->fdsem = pa_fdsem_new();
    c->ss = *ss;

    if (!gst_init_check(NULL, NULL, &error)) {
        pa_log_error("Could not initialise GStreamer: %s", error->message);
        g_error_free(error);
        goto fail;
    }

    if (!init_receive_pipeline(c, fd, ss))
        goto fail;

    callbacks.eos = appsink_eos;
    callbacks.new_sample = appsink_new_sample;
    gst_app_sink_set_callbacks(GST_APP_SINK(c->appsink), &callbacks, c, NULL);

    return c;

fail:
    pa_rtp_context_free(c);
    return NULL;
}

/* Called from I/O thread context */
int pa_rtp_recv(pa_rtp_context *c, pa_memchunk *chunk, pa_mempool *pool, uint32_t *rtp_tstamp, struct timeval *tstamp) {
    GstSample *sample = NULL;
    GstBuffer *buf;
    GstMapInfo info;
    void *data;

    if (!process_bus_messages(c))
        goto fail;

    sample = gst_app_sink_pull_sample(GST_APP_SINK(c->appsink));
    if (!sample) {
        pa_log_warn("Could not get any more data");
        goto fail;
    }

    buf = gst_sample_get_buffer(sample);

    if (GST_BUFFER_IS_DISCONT(buf))
        pa_log_info("Discontinuity detected, possibly lost some packets");

    if (!gst_buffer_map(buf, &info, GST_MAP_READ))
        goto fail;

    pa_assert(pa_mempool_block_size_max(pool) >= info.size);

    chunk->memblock = pa_memblock_new(pool, info.size);
    chunk->index = 0;
    chunk->length = info.size;

    data = pa_memblock_acquire_chunk(chunk);
    /* TODO: we could probably just provide an allocator and avoid a memcpy */
    memcpy(data, info.data, info.size);
    pa_memblock_release(chunk->memblock);

    /* When buffer-mode = none, the buffer PTS is the RTP timestamp, converted
     * to time units (instead of clock-rate units as is in the header) and
     * wraparound-corrected, and the DTS is the pipeline clock timestamp from
     * when the buffer was acquired at the source (this is actually the running
     * time which is why we need to add base time). */
    *rtp_tstamp = gst_util_uint64_scale_int(GST_BUFFER_PTS(buf), c->ss.rate, GST_SECOND) & 0xFFFFFFFFU;
    pa_timeval_rtstore(tstamp, (GST_BUFFER_DTS(buf) + gst_element_get_base_time(c->pipeline)) / GST_USECOND, false);

    gst_buffer_unmap(buf, &info);
    gst_sample_unref(sample);

    return 0;

fail:
    if (sample)
        gst_sample_unref(sample);

    if (chunk->memblock)
        pa_memblock_unref(chunk->memblock);

    return -1;
}

void pa_rtp_context_free(pa_rtp_context *c) {
    pa_assert(c);

    if (c->appsrc) {
        gst_app_src_end_of_stream(GST_APP_SRC(c->appsrc));
        gst_object_unref(c->appsrc);
    }

    if (c->appsink)
        gst_object_unref(c->appsink);

    if (c->pipeline) {
        gst_element_set_state(c->pipeline, GST_STATE_NULL);
        gst_object_unref(c->pipeline);
    }

    if (c->fdsem)
        pa_fdsem_free(c->fdsem);

    pa_xfree(c);
}

pa_rtpoll_item* pa_rtp_context_get_rtpoll_item(pa_rtp_context *c, pa_rtpoll *rtpoll) {
    return pa_rtpoll_item_new_fdsem(rtpoll, PA_RTPOLL_LATE, c->fdsem);
}

size_t pa_rtp_context_get_frame_size(pa_rtp_context *c) {
    return pa_frame_size(&c->ss);
}