/* GStreamer Adaptive Multi-Rate parser plugin

 * Copyright (C) 2006 Edgard Lima <edgard.lima@gmail.com>

 * Copyright (C) 2008 Nokia Corporation. All rights reserved.

 *

 * Contact: Stefan Kost <stefan.kost@nokia.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.

 */

/**

 * SECTION:element-amrparse

 * @short_description: AMR parser

 * @see_also: #GstAmrnbDec, #GstAmrnbEnc

 *

 * This is an AMR parser capable of handling both narrow-band and wideband

 * formats.

 *

 * <refsect2>

 * <title>Example launch line</title>

 * |[

 * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrdec ! audioresample ! audioconvert ! alsasink

 * ]|

 * </refsect2>

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <string.h>

#include "gstamrparse.h"

#include <gst/pbutils/pbutils.h>

static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",

    GST_PAD_SRC,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1;"

        "audio/AMR-WB, " "rate = (int) 16000, " "channels = (int) 1;")

    );

static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",

    GST_PAD_SINK,

    GST_PAD_ALWAYS,

    GST_STATIC_CAPS ("audio/x-amr-nb-sh; audio/x-amr-wb-sh"));

GST_DEBUG_CATEGORY_STATIC (amrparse_debug);

#define GST_CAT_DEFAULT amrparse_debug

static const gint block_size_nb[16] =

    { 12, 13, 15, 17, 19, 20, 26, 31, 5, 0, 0, 0, 0, 0, 0, 0 };

static const gint block_size_wb[16] =

    { 17, 23, 32, 36, 40, 46, 50, 58, 60, 5, -1, -1, -1, -1, 0, 0 };

/* AMR has a "hardcoded" framerate of 50fps */

#define AMR_FRAMES_PER_SECOND 50

#define AMR_FRAME_DURATION (GST_SECOND/AMR_FRAMES_PER_SECOND)

#define AMR_MIME_HEADER_SIZE 9

static gboolean gst_amr_parse_start (GstBaseParse * parse);

static gboolean gst_amr_parse_stop (GstBaseParse * parse);

static gboolean gst_amr_parse_sink_setcaps (GstBaseParse * parse,

    GstCaps * caps);

static GstCaps *gst_amr_parse_sink_getcaps (GstBaseParse * parse,

    GstCaps * filter);

static GstFlowReturn gst_amr_parse_handle_frame (GstBaseParse * parse,

    GstBaseParseFrame * frame, gint * skipsize);

static GstFlowReturn gst_amr_parse_pre_push_frame (GstBaseParse * parse,

    GstBaseParseFrame * frame);

G_DEFINE_TYPE (GstAmrParse, gst_amr_parse, GST_TYPE_BASE_PARSE);

/**

 * gst_amr_parse_class_init:

 * @klass: GstAmrParseClass.

 *

 */

static void

gst_amr_parse_class_init (GstAmrParseClass * klass)

{

  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);

  GST_DEBUG_CATEGORY_INIT (amrparse_debug, "amrparse", 0,

      "AMR-NB audio stream parser");

  gst_element_class_add_static_pad_template (element_class, &sink_template);

  gst_element_class_add_static_pad_template (element_class, &src_template);

  gst_element_class_set_static_metadata (element_class,

      "AMR audio stream parser", "Codec/Parser/Audio",

      "Adaptive Multi-Rate audio parser",

      "Ronald Bultje <rbultje@ronald.bitfreak.net>");

  parse_class->start = GST_DEBUG_FUNCPTR (gst_amr_parse_start);

  parse_class->stop = GST_DEBUG_FUNCPTR (gst_amr_parse_stop);

  parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_setcaps);

  parse_class->get_sink_caps = GST_DEBUG_FUNCPTR (gst_amr_parse_sink_getcaps);

  parse_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amr_parse_handle_frame);

  parse_class->pre_push_frame =

      GST_DEBUG_FUNCPTR (gst_amr_parse_pre_push_frame);

}

/**

 * gst_amr_parse_init:

 * @amrparse: #GstAmrParse

 * @klass: #GstAmrParseClass.

 *

 */

static void

gst_amr_parse_init (GstAmrParse * amrparse)

{

  /* init rest */

  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 62);

  GST_DEBUG ("initialized");

  GST_PAD_SET_ACCEPT_INTERSECT (GST_BASE_PARSE_SINK_PAD (amrparse));

  GST_PAD_SET_ACCEPT_TEMPLATE (GST_BASE_PARSE_SINK_PAD (amrparse));

}

/**

 * gst_amr_parse_set_src_caps:

 * @amrparse: #GstAmrParse.

 *

 * Set source pad caps according to current knowledge about the

 * audio stream.

 *

 * Returns: TRUE if caps were successfully set.

 */

static gboolean

gst_amr_parse_set_src_caps (GstAmrParse * amrparse)

{

  GstCaps *src_caps = NULL;

  gboolean res = FALSE;

  if (amrparse->wide) {

    GST_DEBUG_OBJECT (amrparse, "setting srcpad caps to AMR-WB");

    src_caps = gst_caps_new_simple ("audio/AMR-WB",

        "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 16000, NULL);

  } else {

    GST_DEBUG_OBJECT (amrparse, "setting srcpad caps to AMR-NB");

    /* Max. size of NB frame is 31 bytes, so we can set the min. frame

       size to 32 (+1 for next frame header) */

    gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 32);

    src_caps = gst_caps_new_simple ("audio/AMR",

        "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 8000, NULL);

  }

  gst_pad_use_fixed_caps (GST_BASE_PARSE (amrparse)->srcpad);

  res = gst_pad_set_caps (GST_BASE_PARSE (amrparse)->srcpad, src_caps);

  gst_caps_unref (src_caps);

  return res;

}

/**

 * gst_amr_parse_sink_setcaps:

 * @sinkpad: GstPad

 * @caps: GstCaps

 *

 * Returns: TRUE on success.

 */

static gboolean

gst_amr_parse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)

{

  GstAmrParse *amrparse;

  GstStructure *structure;

  const gchar *name;

  amrparse = GST_AMR_PARSE (parse);

  structure = gst_caps_get_structure (caps, 0);

  name = gst_structure_get_name (structure);

  GST_DEBUG_OBJECT (amrparse, "setcaps: %s", name);

  if (!strncmp (name, "audio/x-amr-wb-sh", 17)) {

    amrparse->block_size = block_size_wb;

    amrparse->wide = 1;

  } else if (!strncmp (name, "audio/x-amr-nb-sh", 17)) {

    amrparse->block_size = block_size_nb;

    amrparse->wide = 0;

  } else {

    GST_WARNING ("Unknown caps");

    return FALSE;

  }

  amrparse->need_header = FALSE;

  gst_base_parse_set_frame_rate (GST_BASE_PARSE (amrparse), 50, 1, 2, 2);

  gst_amr_parse_set_src_caps (amrparse);

  return TRUE;

}

/**

 * gst_amr_parse_parse_header:

 * @amrparse: #GstAmrParse

 * @data: Header data to be parsed.

 * @skipsize: Output argument where the frame size will be stored.

 *

 * Check if the given data contains an AMR mime header.

 *

 * Returns: TRUE on success.

 */

static gboolean

gst_amr_parse_parse_header (GstAmrParse * amrparse,

    const guint8 * data, gint * skipsize)

{

  GST_DEBUG_OBJECT (amrparse, "Parsing header data");

  if (!memcmp (data, "#!AMR-WB\n", 9)) {

    GST_DEBUG_OBJECT (amrparse, "AMR-WB detected");

    amrparse->block_size = block_size_wb;

    amrparse->wide = TRUE;

    *skipsize = amrparse->header = 9;

  } else if (!memcmp (data, "#!AMR\n", 6)) {

    GST_DEBUG_OBJECT (amrparse, "AMR-NB detected");

    amrparse->block_size = block_size_nb;

    amrparse->wide = FALSE;

    *skipsize = amrparse->header = 6;

  } else

    return FALSE;

  gst_amr_parse_set_src_caps (amrparse);

  return TRUE;

}

/**

 * gst_amr_parse_check_valid_frame:

 * @parse: #GstBaseParse.

 * @buffer: #GstBuffer.

 * @framesize: Output variable where the found frame size is put.

 * @skipsize: Output variable which tells how much data needs to be skipped

 *            until a frame header is found.

 *

 * Implementation of "check_valid_frame" vmethod in #GstBaseParse class.

 *

 * Returns: TRUE if the given data contains valid frame.

 */

static GstFlowReturn

gst_amr_parse_handle_frame (GstBaseParse * parse,

    GstBaseParseFrame * frame, gint * skipsize)

{

  GstBuffer *buffer;

  GstMapInfo map;

  gint fsize = 0, mode, dsize;

  GstAmrParse *amrparse;

  GstFlowReturn ret = GST_FLOW_OK;

  gboolean found = FALSE;

  amrparse = GST_AMR_PARSE (parse);

  buffer = frame->buffer;

  gst_buffer_map (buffer, &map, GST_MAP_READ);

  dsize = map.size;

  GST_LOG ("buffer: %d bytes", dsize);

  if (amrparse->need_header) {

    if (dsize >= AMR_MIME_HEADER_SIZE &&

        gst_amr_parse_parse_header (amrparse, map.data, skipsize)) {

      amrparse->need_header = FALSE;

      gst_base_parse_set_frame_rate (GST_BASE_PARSE (amrparse), 50, 1, 2, 2);

    } else {

      GST_WARNING ("media doesn't look like a AMR format");

    }

    /* We return FALSE, so this frame won't get pushed forward. Instead,

       the "skip" value is set, so next time we will receive a valid frame. */

    goto done;

  }

  *skipsize = 1;

  /* Does this look like a possible frame header candidate? */

  if ((map.data[0] & 0x83) == 0) {

    /* Yep. Retrieve the frame size */

    mode = (map.data[0] >> 3) & 0x0F;

    fsize = amrparse->block_size[mode] + 1;     /* +1 for the header byte */

    /* We recognize this data as a valid frame when:

     *     - We are in sync. There is no need for extra checks then

     *     - We are in EOS. There might not be enough data to check next frame

     *     - Sync is lost, but the following data after this frame seem

     *       to contain a valid header as well (and there is enough data to

     *       perform this check)

     */

    if (fsize) {

      *skipsize = 0;

      /* in sync, no further check */

      if (!GST_BASE_PARSE_LOST_SYNC (parse)) {

        found = TRUE;

      } else if (dsize > fsize) {

        /* enough data, check for next sync */

        if ((map.data[fsize] & 0x83) == 0)

          found = TRUE;

      } else if (GST_BASE_PARSE_DRAINING (parse)) {

        /* not enough, but draining, so ok */

        found = TRUE;

      }

    }

  }

done:

  gst_buffer_unmap (buffer, &map);

  if (found && fsize <= map.size) {

    ret = gst_base_parse_finish_frame (parse, frame, fsize);

  }

  return ret;

}

/**

 * gst_amr_parse_start:

 * @parse: #GstBaseParse.

 *

 * Implementation of "start" vmethod in #GstBaseParse class.

 *

 * Returns: TRUE on success.

 */

static gboolean

gst_amr_parse_start (GstBaseParse * parse)

{

  GstAmrParse *amrparse;

  amrparse = GST_AMR_PARSE (parse);

  GST_DEBUG ("start");

  amrparse->need_header = TRUE;

  amrparse->header = 0;

  amrparse->sent_codec_tag = FALSE;

  return TRUE;

}

/**

 * gst_amr_parse_stop:

 * @parse: #GstBaseParse.

 *

 * Implementation of "stop" vmethod in #GstBaseParse class.

 *

 * Returns: TRUE on success.

 */

static gboolean

gst_amr_parse_stop (GstBaseParse * parse)

{

  GstAmrParse *amrparse;

  amrparse = GST_AMR_PARSE (parse);

  GST_DEBUG ("stop");

  amrparse->need_header = TRUE;

  amrparse->header = 0;

  return TRUE;

}

static GstCaps *

gst_amr_parse_sink_getcaps (GstBaseParse * parse, GstCaps * filter)

{

  GstCaps *peercaps, *templ;

  GstCaps *res;

  templ = gst_pad_get_pad_template_caps (GST_BASE_PARSE_SINK_PAD (parse));

  peercaps = gst_pad_peer_query_caps (GST_BASE_PARSE_SRC_PAD (parse), filter);

  if (peercaps) {

    guint i, n;

    /* Rename structure names */

    peercaps = gst_caps_make_writable (peercaps);

    n = gst_caps_get_size (peercaps);

    for (i = 0; i < n; i++) {

      GstStructure *s = gst_caps_get_structure (peercaps, i);

      if (gst_structure_has_name (s, "audio/AMR"))

        gst_structure_set_name (s, "audio/x-amr-nb-sh");

      else

        gst_structure_set_name (s, "audio/x-amr-wb-sh");

    }

    res = gst_caps_intersect_full (peercaps, templ, GST_CAPS_INTERSECT_FIRST);

    gst_caps_unref (peercaps);

    res = gst_caps_make_writable (res);

    /* Append the template caps because we still want to accept

     * caps without any fields in the case upstream does not

     * know anything.

     */

    gst_caps_append (res, templ);

  } else {

    res = templ;

  }

  if (filter) {

    GstCaps *intersection;

    intersection =

        gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);

    gst_caps_unref (res);

    res = intersection;

  }

  return res;

}

static GstFlowReturn

gst_amr_parse_pre_push_frame (GstBaseParse * parse, GstBaseParseFrame * frame)

{

  GstAmrParse *amrparse = GST_AMR_PARSE (parse);

  if (!amrparse->sent_codec_tag) {

    GstTagList *taglist;

    GstCaps *caps;

    /* codec tag */

    caps = gst_pad_get_current_caps (GST_BASE_PARSE_SRC_PAD (parse));

    if (G_UNLIKELY (caps == NULL)) {

      if (GST_PAD_IS_FLUSHING (GST_BASE_PARSE_SRC_PAD (parse))) {

        GST_INFO_OBJECT (parse, "Src pad is flushing");

        return GST_FLOW_FLUSHING;

      } else {

        GST_INFO_OBJECT (parse, "Src pad is not negotiated!");

        return GST_FLOW_NOT_NEGOTIATED;

      }

    }

    taglist = gst_tag_list_new_empty ();

    gst_pb_utils_add_codec_description_to_tag_list (taglist,

        GST_TAG_AUDIO_CODEC, caps);

    gst_caps_unref (caps);

    gst_base_parse_merge_tags (parse, taglist, GST_TAG_MERGE_REPLACE);

    gst_tag_list_unref (taglist);

    /* also signals the end of first-frame processing */

    amrparse->sent_codec_tag = TRUE;

  }

  return GST_FLOW_OK;

}