/*

 * Opus Payloader Gst Element

 *

 *   @author: Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>

 *

 * 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 <string.h>

#include <gst/rtp/gstrtpbuffer.h>

#include <gst/audio/audio.h>

#include "gstrtpopuspay.h"

#include "gstrtputils.h"

GST_DEBUG_CATEGORY_STATIC (rtpopuspay_debug);

#define GST_CAT_DEFAULT (rtpopuspay_debug)

static GstStaticPadTemplate gst_rtp_opus_pay_sink_template =

GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS

    ("audio/x-opus, channels = (int) [1, 2], channel-mapping-family = (int) 0")

    );

static GstStaticPadTemplate gst_rtp_opus_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) 48000, "

        "encoding-params = (string) \"2\", "

        "encoding-name = (string) { \"OPUS\", \"X-GST-OPUS-DRAFT-SPITTKA-00\" }")

    );

static gboolean gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload,

    GstCaps * caps);

static GstCaps *gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,

    GstPad * pad, GstCaps * filter);

static GstFlowReturn gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload *

    payload, GstBuffer * buffer);

G_DEFINE_TYPE (GstRtpOPUSPay, gst_rtp_opus_pay, GST_TYPE_RTP_BASE_PAYLOAD);

static void

gst_rtp_opus_pay_class_init (GstRtpOPUSPayClass * klass)

{

  GstRTPBasePayloadClass *gstbasertppayload_class;

  GstElementClass *element_class;

  gstbasertppayload_class = (GstRTPBasePayloadClass *) klass;

  element_class = GST_ELEMENT_CLASS (klass);

  gstbasertppayload_class->set_caps = gst_rtp_opus_pay_setcaps;

  gstbasertppayload_class->get_caps = gst_rtp_opus_pay_getcaps;

  gstbasertppayload_class->handle_buffer = gst_rtp_opus_pay_handle_buffer;

  gst_element_class_add_static_pad_template (element_class,

      &gst_rtp_opus_pay_src_template);

  gst_element_class_add_static_pad_template (element_class,

      &gst_rtp_opus_pay_sink_template);

  gst_element_class_set_static_metadata (element_class,

      "RTP Opus payloader",

      "Codec/Payloader/Network/RTP",

      "Puts Opus audio in RTP packets",

      "Danilo Cesar Lemes de Paula <danilo.cesar@collabora.co.uk>");

  GST_DEBUG_CATEGORY_INIT (rtpopuspay_debug, "rtpopuspay", 0,

      "Opus RTP Payloader");

}

static void

gst_rtp_opus_pay_init (GstRtpOPUSPay * rtpopuspay)

{

}

static gboolean

gst_rtp_opus_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)

{

  gboolean res;

  GstCaps *src_caps;

  GstStructure *s;

  const char *encoding_name = "OPUS";

  gint channels, rate;

  const char *sprop_stereo = NULL;

  char *sprop_maxcapturerate = NULL;

  src_caps = gst_pad_get_allowed_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));

  if (src_caps) {

    GstStructure *s;

    const GValue *value;

    s = gst_caps_get_structure (src_caps, 0);

    if (gst_structure_has_field (s, "encoding-name")) {

      GValue default_value = G_VALUE_INIT;

      g_value_init (&default_value, G_TYPE_STRING);

      g_value_set_static_string (&default_value, encoding_name);

      value = gst_structure_get_value (s, "encoding-name");

      if (!gst_value_can_intersect (&default_value, value))

        encoding_name = "X-GST-OPUS-DRAFT-SPITTKA-00";

    }

    gst_caps_unref (src_caps);

  }

  s = gst_caps_get_structure (caps, 0);

  if (gst_structure_get_int (s, "channels", &channels)) {

    if (channels > 2) {

      GST_ERROR_OBJECT (payload,

          "More than 2 channels with channel-mapping-family=0 is invalid");

      return FALSE;

    } else if (channels == 2) {

      sprop_stereo = "1";

    } else {

      sprop_stereo = "0";

    }

  }

  if (gst_structure_get_int (s, "rate", &rate)) {

    sprop_maxcapturerate = g_strdup_printf ("%d", rate);

  }

  gst_rtp_base_payload_set_options (payload, "audio", FALSE,

      encoding_name, 48000);

  if (sprop_maxcapturerate && sprop_stereo) {

    res =

        gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",

        G_TYPE_STRING, sprop_maxcapturerate, "sprop-stereo", G_TYPE_STRING,

        sprop_stereo, NULL);

  } else if (sprop_maxcapturerate) {

    res =

        gst_rtp_base_payload_set_outcaps (payload, "sprop-maxcapturerate",

        G_TYPE_STRING, sprop_maxcapturerate, NULL);

  } else if (sprop_stereo) {

    res =

        gst_rtp_base_payload_set_outcaps (payload, "sprop-stereo",

        G_TYPE_STRING, sprop_stereo, NULL);

  } else {

    res = gst_rtp_base_payload_set_outcaps (payload, NULL);

  }

  g_free (sprop_maxcapturerate);

  return res;

}

static GstFlowReturn

gst_rtp_opus_pay_handle_buffer (GstRTPBasePayload * basepayload,

    GstBuffer * buffer)

{

  GstBuffer *outbuf;

  GstClockTime pts, dts, duration;

  pts = GST_BUFFER_PTS (buffer);

  dts = GST_BUFFER_DTS (buffer);

  duration = GST_BUFFER_DURATION (buffer);

  outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);

  gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);

  outbuf = gst_buffer_append (outbuf, buffer);

  GST_BUFFER_PTS (outbuf) = pts;

  GST_BUFFER_DTS (outbuf) = dts;

  GST_BUFFER_DURATION (outbuf) = duration;

  /* Push out */

  return gst_rtp_base_payload_push (basepayload, outbuf);

}

static GstCaps *

gst_rtp_opus_pay_getcaps (GstRTPBasePayload * payload,

    GstPad * pad, GstCaps * filter)

{

  GstCaps *caps, *peercaps, *tcaps;

  GstStructure *s;

  const gchar *stereo;

  if (pad == GST_RTP_BASE_PAYLOAD_SRCPAD (payload))

    return

        GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps

        (payload, pad, filter);

  tcaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));

  peercaps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload),

      tcaps);

  gst_caps_unref (tcaps);

  if (!peercaps)

    return

        GST_RTP_BASE_PAYLOAD_CLASS (gst_rtp_opus_pay_parent_class)->get_caps

        (payload, pad, filter);

  if (gst_caps_is_empty (peercaps))

    return peercaps;

  caps = gst_pad_get_pad_template_caps (GST_RTP_BASE_PAYLOAD_SINKPAD (payload));

  s = gst_caps_get_structure (peercaps, 0);

  stereo = gst_structure_get_string (s, "stereo");

  if (stereo != NULL) {

    caps = gst_caps_make_writable (caps);

    if (!strcmp (stereo, "1")) {

      GstCaps *caps2 = gst_caps_copy (caps);

      gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);

      gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 1, NULL);

      caps = gst_caps_merge (caps, caps2);

    } else if (!strcmp (stereo, "0")) {

      GstCaps *caps2 = gst_caps_copy (caps);

      gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);

      gst_caps_set_simple (caps2, "channels", G_TYPE_INT, 2, NULL);

      caps = gst_caps_merge (caps, caps2);

    }

  }

  gst_caps_unref (peercaps);

  if (filter) {

    GstCaps *tmp = gst_caps_intersect_full (caps, filter,

        GST_CAPS_INTERSECT_FIRST);

    gst_caps_unref (caps);

    caps = tmp;

  }

  GST_DEBUG_OBJECT (payload, "Returning caps: %" GST_PTR_FORMAT, caps);

  return caps;

}

gboolean

gst_rtp_opus_pay_plugin_init (GstPlugin * plugin)

{

  return gst_element_register (plugin, "rtpopuspay",

      GST_RANK_PRIMARY, GST_TYPE_RTP_OPUS_PAY);

}