/* GStreamer

 * Copyright (C) <2009> Wim Taymans <wim.taymans@gmail.com>

 *

 * This library is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Library General Public

 * License as published by the Free Software Foundation; either

 * version 2 of the License, or (at your option) any later version.

 *

 * This library 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

 * Library General Public License for more details.

 *

 * You should have received a copy of the GNU Library General Public

 * License along with this library; if not, write to the

 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,

 * Boston, MA 02110-1301, USA.

 */

#ifdef HAVE_CONFIG_H

#  include "config.h"

#endif

#include <stdlib.h>

#include <string.h>

#include <gst/rtp/gstrtpbuffer.h>

#include <gst/audio/audio.h>

#include "gstrtpceltpay.h"

#include "gstrtputils.h"

GST_DEBUG_CATEGORY_STATIC (rtpceltpay_debug);

#define GST_CAT_DEFAULT (rtpceltpay_debug)

static GstStaticPadTemplate gst_rtp_celt_pay_sink_template =

GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("audio/x-celt, "

        "rate = (int) [ 32000, 64000 ], "

        "channels = (int) [1, 2], " "frame-size = (int) [ 64, 512 ]")

    );

static GstStaticPadTemplate gst_rtp_celt_pay_src_template =

GST_STATIC_PAD_TEMPLATE ("src",

    GST_PAD_SRC,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("application/x-rtp, "

        "media = (string) \"audio\", "

        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "

        "clock-rate =  (int) [ 32000, 48000 ], "

        "encoding-name = (string) \"CELT\"")

    );

static void gst_rtp_celt_pay_finalize (GObject * object);

static GstStateChangeReturn gst_rtp_celt_pay_change_state (GstElement *

    element, GstStateChange transition);

static gboolean gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload,

    GstCaps * caps);

static GstCaps *gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload,

    GstPad * pad, GstCaps * filter);

static GstFlowReturn gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload *

    payload, GstBuffer * buffer);

#define gst_rtp_celt_pay_parent_class parent_class

G_DEFINE_TYPE (GstRtpCELTPay, gst_rtp_celt_pay, GST_TYPE_RTP_BASE_PAYLOAD);

static void

gst_rtp_celt_pay_class_init (GstRtpCELTPayClass * klass)

{

  GObjectClass *gobject_class;

  GstElementClass *gstelement_class;

  GstRTPBasePayloadClass *gstrtpbasepayload_class;

  GST_DEBUG_CATEGORY_INIT (rtpceltpay_debug, "rtpceltpay", 0,

      "CELT RTP Payloader");

  gobject_class = (GObjectClass *) klass;

  gstelement_class = (GstElementClass *) klass;

  gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;

  gobject_class->finalize = gst_rtp_celt_pay_finalize;

  gstelement_class->change_state = gst_rtp_celt_pay_change_state;

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_celt_pay_sink_template);

  gst_element_class_add_static_pad_template (gstelement_class,

      &gst_rtp_celt_pay_src_template);

  gst_element_class_set_static_metadata (gstelement_class, "RTP CELT payloader",

      "Codec/Payloader/Network/RTP",

      "Payload-encodes CELT audio into a RTP packet",

      "Wim Taymans <wim.taymans@gmail.com>");

  gstrtpbasepayload_class->set_caps = gst_rtp_celt_pay_setcaps;

  gstrtpbasepayload_class->get_caps = gst_rtp_celt_pay_getcaps;

  gstrtpbasepayload_class->handle_buffer = gst_rtp_celt_pay_handle_buffer;

}

static void

gst_rtp_celt_pay_init (GstRtpCELTPay * rtpceltpay)

{

  rtpceltpay->queue = g_queue_new ();

}

static void

gst_rtp_celt_pay_finalize (GObject * object)

{

  GstRtpCELTPay *rtpceltpay;

  rtpceltpay = GST_RTP_CELT_PAY (object);

  g_queue_free (rtpceltpay->queue);

  G_OBJECT_CLASS (parent_class)->finalize (object);

}

static void

gst_rtp_celt_pay_clear_queued (GstRtpCELTPay * rtpceltpay)

{

  GstBuffer *buf;

  while ((buf = g_queue_pop_head (rtpceltpay->queue)))

    gst_buffer_unref (buf);

  rtpceltpay->bytes = 0;

  rtpceltpay->sbytes = 0;

  rtpceltpay->qduration = 0;

}

static void

gst_rtp_celt_pay_add_queued (GstRtpCELTPay * rtpceltpay, GstBuffer * buffer,

    guint ssize, guint size, GstClockTime duration)

{

  g_queue_push_tail (rtpceltpay->queue, buffer);

  rtpceltpay->sbytes += ssize;

  rtpceltpay->bytes += size;

  /* only add durations when we have a valid previous duration */

  if (rtpceltpay->qduration != -1) {

    if (duration != -1)

      /* only add valid durations */

      rtpceltpay->qduration += duration;

    else

      /* if we add a buffer without valid duration, our total queued duration

       * becomes unknown */

      rtpceltpay->qduration = -1;

  }

}

static gboolean

gst_rtp_celt_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)

{

  /* don't configure yet, we wait for the ident packet */

  return TRUE;

}

static GstCaps *

gst_rtp_celt_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,

    GstCaps * filter)

{

  GstCaps *otherpadcaps;

  GstCaps *caps;

  const gchar *params;

  caps = gst_pad_get_pad_template_caps (pad);

  otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);

  if (otherpadcaps) {

    if (!gst_caps_is_empty (otherpadcaps)) {

      GstStructure *ps;

      GstStructure *s;

      gint clock_rate = 0, frame_size = 0, channels = 1;

      caps = gst_caps_make_writable (caps);

      ps = gst_caps_get_structure (otherpadcaps, 0);

      s = gst_caps_get_structure (caps, 0);

      if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) {

        gst_structure_fixate_field_nearest_int (s, "rate", clock_rate);

      }

      if ((params = gst_structure_get_string (ps, "frame-size")))

        frame_size = atoi (params);

      if (frame_size)

        gst_structure_set (s, "frame-size", G_TYPE_INT, frame_size, NULL);

      if ((params = gst_structure_get_string (ps, "encoding-params"))) {

        channels = atoi (params);

        gst_structure_fixate_field_nearest_int (s, "channels", channels);

      }

      GST_DEBUG_OBJECT (payload, "clock-rate=%d frame-size=%d channels=%d",

          clock_rate, frame_size, channels);

    }

    gst_caps_unref (otherpadcaps);

  }

  if (filter) {

    GstCaps *tmp;

    GST_DEBUG_OBJECT (payload, "Intersect %" GST_PTR_FORMAT " and filter %"

        GST_PTR_FORMAT, caps, filter);

    tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);

    gst_caps_unref (caps);

    caps = tmp;

  }

  return caps;

}

static gboolean

gst_rtp_celt_pay_parse_ident (GstRtpCELTPay * rtpceltpay,

    const guint8 * data, guint size)

{

  guint32 version, header_size, rate, nb_channels, frame_size, overlap;

  guint32 bytes_per_packet;

  GstRTPBasePayload *payload;

  gchar *cstr, *fsstr;

  gboolean res;

  /* we need the header string (8), the version string (20), the version

   * and the header length. */

  if (size < 36)

    goto too_small;

  if (!g_str_has_prefix ((const gchar *) data, "CELT    "))

    goto wrong_header;

  /* skip header and version string */

  data += 28;

  version = GST_READ_UINT32_LE (data);

  GST_DEBUG_OBJECT (rtpceltpay, "version %08x", version);

#if 0

  if (version != 1)

    goto wrong_version;

#endif

  data += 4;

  /* ensure sizes */

  header_size = GST_READ_UINT32_LE (data);

  if (header_size < 56)

    goto header_too_small;

  if (size < header_size)

    goto payload_too_small;

  data += 4;

  rate = GST_READ_UINT32_LE (data);

  data += 4;

  nb_channels = GST_READ_UINT32_LE (data);

  data += 4;

  frame_size = GST_READ_UINT32_LE (data);

  data += 4;

  overlap = GST_READ_UINT32_LE (data);

  data += 4;

  bytes_per_packet = GST_READ_UINT32_LE (data);

  GST_DEBUG_OBJECT (rtpceltpay, "rate %d, nb_channels %d, frame_size %d",

      rate, nb_channels, frame_size);

  GST_DEBUG_OBJECT (rtpceltpay, "overlap %d, bytes_per_packet %d",

      overlap, bytes_per_packet);

  payload = GST_RTP_BASE_PAYLOAD (rtpceltpay);

  gst_rtp_base_payload_set_options (payload, "audio", FALSE, "CELT", rate);

  cstr = g_strdup_printf ("%d", nb_channels);

  fsstr = g_strdup_printf ("%d", frame_size);

  res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",

      G_TYPE_STRING, cstr, "frame-size", G_TYPE_STRING, fsstr, NULL);

  g_free (cstr);

  g_free (fsstr);

  return res;

  /* ERRORS */

too_small:

  {

    GST_DEBUG_OBJECT (rtpceltpay,

        "ident packet too small, need at least 32 bytes");

    return FALSE;

  }

wrong_header:

  {

    GST_DEBUG_OBJECT (rtpceltpay,

        "ident packet does not start with \"CELT    \"");

    return FALSE;

  }

#if 0

wrong_version:

  {

    GST_DEBUG_OBJECT (rtpceltpay, "can only handle version 1, have version %d",

        version);

    return FALSE;

  }

#endif

header_too_small:

  {

    GST_DEBUG_OBJECT (rtpceltpay,

        "header size too small, need at least 80 bytes, " "got only %d",

        header_size);

    return FALSE;

  }

payload_too_small:

  {

    GST_DEBUG_OBJECT (rtpceltpay,

        "payload too small, need at least %d bytes, got only %d", header_size,

        size);

    return FALSE;

  }

}

static GstFlowReturn

gst_rtp_celt_pay_flush_queued (GstRtpCELTPay * rtpceltpay)

{

  GstFlowReturn ret;

  GstBuffer *buf, *outbuf;

  guint8 *payload, *spayload;

  guint payload_len;

  GstClockTime duration;

  GstRTPBuffer rtp = { NULL, };

  payload_len = rtpceltpay->bytes + rtpceltpay->sbytes;

  duration = rtpceltpay->qduration;

  GST_DEBUG_OBJECT (rtpceltpay, "flushing out %u, duration %" GST_TIME_FORMAT,

      payload_len, GST_TIME_ARGS (rtpceltpay->qduration));

  /* get a big enough packet for the sizes + payloads */

  outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);

  GST_BUFFER_DURATION (outbuf) = duration;

  gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp;;

  /* point to the payload for size headers and data */

  spayload = gst_rtp_buffer_get_payload (&rtp;;

  payload = spayload + rtpceltpay->sbytes;

  while ((buf = g_queue_pop_head (rtpceltpay->queue))) {

    guint size;

    /* copy first timestamp to output */

    if (GST_BUFFER_PTS (outbuf) == -1)

      GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buf);

    /* write the size to the header */

    size = gst_buffer_get_size (buf);

    while (size > 0xff) {

      *spayload++ = 0xff;

      size -= 0xff;

    }

    *spayload++ = size;

    /* copy payload */

    size = gst_buffer_get_size (buf);

    gst_buffer_extract (buf, 0, payload, size);

    payload += size;

    gst_rtp_copy_audio_meta (rtpceltpay, outbuf, buf);

    gst_buffer_unref (buf);

  }

  gst_rtp_buffer_unmap (&rtp;;

  /* we consumed it all */

  rtpceltpay->bytes = 0;

  rtpceltpay->sbytes = 0;

  rtpceltpay->qduration = 0;

  ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpceltpay), outbuf);

  return ret;

}

static GstFlowReturn

gst_rtp_celt_pay_handle_buffer (GstRTPBasePayload * basepayload,

    GstBuffer * buffer)

{

  GstFlowReturn ret;

  GstRtpCELTPay *rtpceltpay;

  gsize payload_len;

  GstMapInfo map;

  GstClockTime duration, packet_dur;

  guint i, ssize, packet_len;

  rtpceltpay = GST_RTP_CELT_PAY (basepayload);

  ret = GST_FLOW_OK;

  gst_buffer_map (buffer, &map, GST_MAP_READ);

  switch (rtpceltpay->packet) {

    case 0:

      /* ident packet. We need to parse the headers to construct the RTP

       * properties. */

      if (!gst_rtp_celt_pay_parse_ident (rtpceltpay, map.data, map.size))

        goto parse_error;

      goto cleanup;

    case 1:

      /* comment packet, we ignore it */

      goto cleanup;

    default:

      /* other packets go in the payload */

      break;

  }

  gst_buffer_unmap (buffer, &map);

  duration = GST_BUFFER_DURATION (buffer);

  GST_LOG_OBJECT (rtpceltpay,

      "got buffer of duration %" GST_TIME_FORMAT ", size %" G_GSIZE_FORMAT,

      GST_TIME_ARGS (duration), map.size);

  /* calculate the size of the size field and the payload */

  ssize = 1;

  for (i = map.size; i > 0xff; i -= 0xff)

    ssize++;

  GST_DEBUG_OBJECT (rtpceltpay, "bytes for size %u", ssize);

  /* calculate what the new size and duration would be of the packet */

  payload_len = ssize + map.size + rtpceltpay->bytes + rtpceltpay->sbytes;

  if (rtpceltpay->qduration != -1 && duration != -1)

    packet_dur = rtpceltpay->qduration + duration;

  else

    packet_dur = 0;

  packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);

  if (gst_rtp_base_payload_is_filled (basepayload, packet_len, packet_dur)) {

    /* size or duration would overflow the packet, flush the queued data */

    ret = gst_rtp_celt_pay_flush_queued (rtpceltpay);

  }

  /* queue the packet */

  gst_rtp_celt_pay_add_queued (rtpceltpay, buffer, ssize, map.size, duration);

done:

  rtpceltpay->packet++;

  return ret;

  /* ERRORS */

cleanup:

  {

    gst_buffer_unmap (buffer, &map);

    goto done;

  }

parse_error:

  {

    GST_ELEMENT_ERROR (rtpceltpay, STREAM, DECODE, (NULL),

        ("Error parsing first identification packet."));

    gst_buffer_unmap (buffer, &map);

    return GST_FLOW_ERROR;

  }

}

static GstStateChangeReturn

gst_rtp_celt_pay_change_state (GstElement * element, GstStateChange transition)

{

  GstRtpCELTPay *rtpceltpay;

  GstStateChangeReturn ret;

  rtpceltpay = GST_RTP_CELT_PAY (element);

  switch (transition) {

    case GST_STATE_CHANGE_NULL_TO_READY:

      break;

    case GST_STATE_CHANGE_READY_TO_PAUSED:

      rtpceltpay->packet = 0;

      break;

    default:

      break;

  }

  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

  switch (transition) {

    case GST_STATE_CHANGE_PAUSED_TO_READY:

      gst_rtp_celt_pay_clear_queued (rtpceltpay);

      break;

    case GST_STATE_CHANGE_READY_TO_NULL:

      break;

    default:

      break;

  }

  return ret;

}

gboolean

gst_rtp_celt_pay_plugin_init (GstPlugin * plugin)

{

  return gst_element_register (plugin, "rtpceltpay",

      GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_PAY);

}