audioparsers: move 'amrparse' into it

gst/audioparsers/Makefile.am

@@ -1,11 +1,11 @@
 plugin_LTLIBRARIES = libgstaudioparsersbad.la
 
 libgstaudioparsersbad_la_SOURCES = \
-	gstaacparse.c gstbaseparse.c plugin.c
+	gstaacparse.c gstamrparse.c gstbaseparse.c plugin.c
 
 libgstaudioparsersbad_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
 libgstaudioparsersbad_la_LIBADD = $(GST_BASE_LIBS)
 libgstaudioparsersbad_la_LDFLAGS = $(PACKAGE_LIBS) $(GST_PLUGIN_LDFLAGS)
 libgstaudioparsersbad_la_LIBTOOLFLAGS = --tag=disable-static
 
-noinst_HEADERS = gstaacparse.h gstbaseparse.h
+noinst_HEADERS = gstaacparse.h gstamrparse.h gstbaseparse.h

gst/audioparsers/gstamrparse.c

@@ -0,0 +1,578 @@
+/* GStreamer Adaptive Multi-Rate parser plugin
+ * Copyright (C) 2006 Edgard Lima <edgard.lima@indt.org.br>
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:gstamrparse
+ * @short_description: AMR parser
+ * @see_also: #GstAmrnbDec, #GstAmrnbEnc
+ *
+ * <refsect2>
+ * <para>
+ * This is an AMR parser capable of handling both narrow-band and wideband
+ * formats.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch filesrc location=abc.amr ! amrparse ! amrdec ! audioresample ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </refsect2>
+ */
+
+#include <string.h>
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstamrparse.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 (gst_amrparse_debug);
+#define GST_CAT_DEFAULT gst_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, 5, 0, 0, 0, 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 void gst_amrparse_finalize (GObject * object);
+
+gboolean gst_amrparse_start (GstBaseParse * parse);
+gboolean gst_amrparse_stop (GstBaseParse * parse);
+
+static gboolean gst_amrparse_sink_setcaps (GstBaseParse * parse,
+    GstCaps * caps);
+
+gboolean gst_amrparse_check_valid_frame (GstBaseParse * parse,
+    GstBuffer * buffer, guint * framesize, gint * skipsize);
+
+GstFlowReturn gst_amrparse_parse_frame (GstBaseParse * parse,
+    GstBuffer * buffer);
+
+gboolean gst_amrparse_convert (GstBaseParse * parse,
+    GstFormat src_format,
+    gint64 src_value, GstFormat dest_format, gint64 * dest_value);
+
+gboolean gst_amrparse_event (GstBaseParse * parse, GstEvent * event);
+
+#define _do_init(bla) \
+    GST_DEBUG_CATEGORY_INIT (gst_amrparse_debug, "amrparse", 0, \
+                             "AMR-NB audio stream parser");
+
+GST_BOILERPLATE_FULL (GstAmrParse, gst_amrparse, GstBaseParse,
+    GST_TYPE_BASE_PARSE, _do_init);
+
+
+/**
+ * gst_amrparse_base_init:
+ * @klass: #GstElementClass.
+ *
+ */
+static void
+gst_amrparse_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstElementDetails details = GST_ELEMENT_DETAILS ("AMR audio stream parser",
+      "Codec/Parser/Audio",
+      "Adaptive Multi-Rate audio parser",
+      "Ronald Bultje <rbultje@ronald.bitfreak.net>");
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_template));
+
+  gst_element_class_set_details (element_class, &details);
+}
+
+
+/**
+ * gst_amrparse_class_init:
+ * @klass: GstAmrParseClass.
+ *
+ */
+static void
+gst_amrparse_class_init (GstAmrParseClass * klass)
+{
+  GObjectClass *object_class = G_OBJECT_CLASS (klass);
+  GstBaseParseClass *parse_class = GST_BASE_PARSE_CLASS (klass);
+
+  object_class->finalize = gst_amrparse_finalize;
+
+  parse_class->start = GST_DEBUG_FUNCPTR (gst_amrparse_start);
+  parse_class->stop = GST_DEBUG_FUNCPTR (gst_amrparse_stop);
+  parse_class->event = GST_DEBUG_FUNCPTR (gst_amrparse_event);
+  parse_class->convert = GST_DEBUG_FUNCPTR (gst_amrparse_convert);
+  parse_class->set_sink_caps = GST_DEBUG_FUNCPTR (gst_amrparse_sink_setcaps);
+  parse_class->parse_frame = GST_DEBUG_FUNCPTR (gst_amrparse_parse_frame);
+  parse_class->check_valid_frame =
+      GST_DEBUG_FUNCPTR (gst_amrparse_check_valid_frame);
+}
+
+
+/**
+ * gst_amrparse_init:
+ * @amrparse: #GstAmrParse
+ * @klass: #GstAmrParseClass.
+ *
+ */
+static void
+gst_amrparse_init (GstAmrParse * amrparse, GstAmrParseClass * klass)
+{
+  /* init rest */
+  gst_base_parse_set_min_frame_size (GST_BASE_PARSE (amrparse), 62);
+  amrparse->ts = 0;
+  GST_DEBUG ("initialized");
+
+}
+
+
+/**
+ * gst_amrparse_finalize:
+ * @object:
+ *
+ */
+static void
+gst_amrparse_finalize (GObject * object)
+{
+  GstAmrParse *amrparse;
+
+  amrparse = GST_AMRPARSE (object);
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+
+/**
+ * gst_amrparse_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_amrparse_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_amrparse_sink_setcaps:
+ * @sinkpad: GstPad
+ * @caps: GstCaps
+ *
+ * Returns: TRUE on success.
+ */
+static gboolean
+gst_amrparse_sink_setcaps (GstBaseParse * parse, GstCaps * caps)
+{
+  GstAmrParse *amrparse;
+  GstStructure *structure;
+  const gchar *name;
+
+  amrparse = GST_AMRPARSE (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_amrparse_set_src_caps (amrparse);
+  return TRUE;
+}
+
+
+/**
+ * gst_amrparse_update_duration:
+ * @amrparse: #GstAmrParse.
+ *
+ * Send duration information to base class.
+ */
+static void
+gst_amrparse_update_duration (GstAmrParse * amrparse)
+{
+  GstPad *peer;
+  GstBaseParse *parse;
+
+  parse = GST_BASE_PARSE (amrparse);
+
+  /* Cannot estimate duration. No data has been passed to us yet */
+  if (!amrparse->framecount) {
+    return;
+  }
+
+  peer = gst_pad_get_peer (parse->sinkpad);
+  if (peer) {
+    GstFormat pformat = GST_FORMAT_BYTES;
+    guint64 bpf = amrparse->bytecount / amrparse->framecount;
+    gboolean qres = FALSE;
+    gint64 ptot;
+
+    qres = gst_pad_query_duration (peer, &pformat, &ptot);
+    gst_object_unref (GST_OBJECT (peer));
+    if (qres && bpf) {
+      gst_base_parse_set_duration (parse, GST_FORMAT_TIME,
+          AMR_FRAME_DURATION * ptot / bpf);
+    }
+  }
+}
+
+
+/**
+ * gst_amrparse_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_amrparse_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_amrparse_set_src_caps (amrparse);
+  return TRUE;
+}
+
+
+/**
+ * gst_amrparse_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.
+ */
+gboolean
+gst_amrparse_check_valid_frame (GstBaseParse * parse,
+    GstBuffer * buffer, guint * framesize, gint * skipsize)
+{
+  const guint8 *data;
+  gint fsize, mode, dsize;
+  GstAmrParse *amrparse;
+
+  amrparse = GST_AMRPARSE (parse);
+  data = GST_BUFFER_DATA (buffer);
+  dsize = GST_BUFFER_SIZE (buffer);
+
+  GST_LOG ("buffer: %d bytes", dsize);
+
+  if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
+    /* Discontinuous stream -> drop the sync */
+    amrparse->sync = FALSE;
+  }
+
+  if (amrparse->need_header) {
+    if (dsize >= AMR_MIME_HEADER_SIZE &&
+        gst_amrparse_parse_header (amrparse, data, skipsize)) {
+      amrparse->need_header = FALSE;
+    } 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. */
+    return FALSE;
+  }
+
+  /* Does this look like a possible frame header candidate? */
+  if ((data[0] & 0x83) == 0) {
+    /* Yep. Retrieve the frame size */
+    mode = (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 (amrparse->sync || amrparse->eos ||
+        (dsize >= fsize && (data[fsize] & 0x83) == 0)) {
+      amrparse->sync = TRUE;
+      *framesize = fsize;
+      return TRUE;
+    }
+  }
+
+  GST_LOG ("sync lost");
+  amrparse->sync = FALSE;
+  return FALSE;
+}
+
+
+/**
+ * gst_amrparse_parse_frame:
+ * @parse: #GstBaseParse.
+ * @buffer: #GstBuffer.
+ *
+ * Implementation of "parse" vmethod in #GstBaseParse class.
+ *
+ * Returns: #GstFlowReturn defining the parsing status.
+ */
+GstFlowReturn
+gst_amrparse_parse_frame (GstBaseParse * parse, GstBuffer * buffer)
+{
+  GstAmrParse *amrparse;
+
+  amrparse = GST_AMRPARSE (parse);
+
+  if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
+    gint64 btime;
+    gboolean r = gst_amrparse_convert (parse, GST_FORMAT_BYTES,
+        GST_BUFFER_OFFSET (buffer),
+        GST_FORMAT_TIME, &btime);
+    if (r) {
+      /* FIXME: What to do if the conversion fails? */
+      amrparse->ts = btime;
+    }
+  }
+
+  GST_BUFFER_DURATION (buffer) = AMR_FRAME_DURATION;
+  GST_BUFFER_TIMESTAMP (buffer) = amrparse->ts;
+
+  if (GST_CLOCK_TIME_IS_VALID (amrparse->ts)) {
+    amrparse->ts += AMR_FRAME_DURATION;
+  }
+
+  if (!(++amrparse->framecount % 50)) {
+    gst_amrparse_update_duration (amrparse);
+  }
+  amrparse->bytecount += GST_BUFFER_SIZE (buffer);
+
+  gst_buffer_set_caps (buffer, GST_PAD_CAPS (parse->srcpad));
+  return GST_FLOW_OK;
+}
+
+
+/**
+ * gst_amrparse_start:
+ * @parse: #GstBaseParse.
+ *
+ * Implementation of "start" vmethod in #GstBaseParse class.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_amrparse_start (GstBaseParse * parse)
+{
+  GstAmrParse *amrparse;
+
+  amrparse = GST_AMRPARSE (parse);
+  GST_DEBUG ("start");
+  amrparse->need_header = TRUE;
+  amrparse->header = 0;
+  amrparse->sync = TRUE;
+  amrparse->eos = FALSE;
+  amrparse->framecount = 0;
+  amrparse->bytecount = 0;
+  amrparse->ts = 0;
+  return TRUE;
+}
+
+
+/**
+ * gst_amrparse_stop:
+ * @parse: #GstBaseParse.
+ *
+ * Implementation of "stop" vmethod in #GstBaseParse class.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_amrparse_stop (GstBaseParse * parse)
+{
+  GstAmrParse *amrparse;
+
+  amrparse = GST_AMRPARSE (parse);
+  GST_DEBUG ("stop");
+  amrparse->need_header = TRUE;
+  amrparse->header = 0;
+  amrparse->ts = -1;
+  return TRUE;
+}
+
+
+/**
+ * gst_amrparse_event:
+ * @parse: #GstBaseParse.
+ * @event: #GstEvent.
+ *
+ * Implementation of "event" vmethod in #GstBaseParse class.
+ *
+ * Returns: TRUE if the event was handled and can be dropped.
+ */
+gboolean
+gst_amrparse_event (GstBaseParse * parse, GstEvent * event)
+{
+  GstAmrParse *amrparse;
+
+  amrparse = GST_AMRPARSE (parse);
+  GST_DEBUG ("event");
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_EOS:
+      amrparse->eos = TRUE;
+      GST_DEBUG ("EOS event");
+      break;
+    default:
+      break;
+  }
+
+  return parent_class->event (parse, event);
+}
+
+
+/**
+ * gst_amrparse_convert:
+ * @parse: #GstBaseParse.
+ * @src_format: #GstFormat describing the source format.
+ * @src_value: Source value to be converted.
+ * @dest_format: #GstFormat defining the converted format.
+ * @dest_value: Pointer where the conversion result will be put.
+ *
+ * Implementation of "convert" vmethod in #GstBaseParse class.
+ *
+ * Returns: TRUE if conversion was successful.
+ */
+gboolean
+gst_amrparse_convert (GstBaseParse * parse,
+    GstFormat src_format,
+    gint64 src_value, GstFormat dest_format, gint64 * dest_value)
+{
+  gboolean ret = FALSE;
+  GstAmrParse *amrparse;
+  gfloat bpf;
+
+  amrparse = GST_AMRPARSE (parse);
+
+  /* We are not able to do any estimations until some data has been passed */
+  if (!amrparse->framecount)
+    return FALSE;
+
+  bpf = (gfloat) amrparse->bytecount / amrparse->framecount;
+
+  if (src_format == GST_FORMAT_BYTES) {
+    if (dest_format == GST_FORMAT_TIME) {
+      /* BYTES -> TIME conversion */
+      GST_DEBUG ("converting bytes -> time");
+
+      if (amrparse->framecount) {
+        *dest_value = AMR_FRAME_DURATION * (src_value - amrparse->header) / bpf;
+        GST_DEBUG ("conversion result: %" G_GINT64_FORMAT " ms",
+            *dest_value / GST_MSECOND);
+        ret = TRUE;
+      }
+    }
+  } else if (src_format == GST_FORMAT_TIME) {
+    GST_DEBUG ("converting time -> bytes");
+    if (dest_format == GST_FORMAT_BYTES) {
+      if (amrparse->framecount) {
+        *dest_value = bpf * src_value / AMR_FRAME_DURATION + amrparse->header;
+        GST_DEBUG ("time %" G_GINT64_FORMAT " ms in bytes = %" G_GINT64_FORMAT,
+            src_value / GST_MSECOND, *dest_value);
+        ret = TRUE;
+      }
+    }
+  } else if (src_format == GST_FORMAT_DEFAULT) {
+    /* DEFAULT == frame-based */
+    if (dest_format == GST_FORMAT_TIME) {
+      *dest_value = src_value * AMR_FRAME_DURATION;
+      ret = TRUE;
+    } else if (dest_format == GST_FORMAT_BYTES) {
+    }
+  }
+
+  return ret;
+}

gst/audioparsers/gstamrparse.h

@@ -0,0 +1,87 @@
+/* GStreamer Adaptive Multi-Rate parser
+ * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.net>
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_AMRPARSE_H__
+#define __GST_AMRPARSE_H__
+
+#include <gst/gst.h>
+#include "gstbaseparse.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_AMRPARSE \
+  (gst_amrparse_get_type())
+#define GST_AMRPARSE(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_AMRPARSE, GstAmrParse))
+#define GST_AMRPARSE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_AMRPARSE, GstAmrParseClass))
+#define GST_IS_AMRPARSE(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_AMRPARSE))
+#define GST_IS_AMRPARSE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_AMRPARSE))
+
+
+typedef struct _GstAmrParse GstAmrParse;
+typedef struct _GstAmrParseClass GstAmrParseClass;
+
+/**
+ * GstAmrParse:
+ * @element: the parent element.
+ * @block_size: Pointer to frame size lookup table.
+ * @need_header: Tells whether the MIME header should be read in the beginning.
+ * @wide: Wideband mode.
+ * @eos: Indicates the EOS situation. Set when EOS event is received.
+ * @sync: Tells whether the parser is in sync.
+ * @framecount: Total amount of frames handled.
+ * @bytecount: Total amount of bytes handled.
+ * @ts: Timestamp of the current media.
+ *
+ * The opaque GstAacParse data structure.
+ */
+struct _GstAmrParse {
+  GstBaseParse element;
+  const gint *block_size;
+  gboolean need_header;
+  gint header;
+  gboolean wide;
+  gboolean eos;
+  gboolean sync;
+  guint64 framecount;
+  guint64 bytecount;
+  GstClockTime ts;
+};
+
+/**
+ * GstAmrParseClass:
+ * @parent_class: Element parent class.
+ *
+ * The opaque GstAmrParseClass data structure.
+ */
+struct _GstAmrParseClass {
+  GstBaseParseClass parent_class;
+};
+
+GType gst_amrparse_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_AMRPARSE_H__ */

gst/audioparsers/plugin.c

@@ -22,6 +22,7 @@
 #endif
 
 #include "gstaacparse.h"
+#include "gstamrparse.h"
 
 static gboolean
 plugin_init (GstPlugin * plugin)
@@ -30,6 +31,8 @@ plugin_init (GstPlugin * plugin)
 
   ret = gst_element_register (plugin, "aacparse",
       GST_RANK_NONE, GST_TYPE_AACPARSE);
+  ret &= gst_element_register (plugin, "amrparse",
+      GST_RANK_PRIMARY + 1, GST_TYPE_AMRPARSE);
 
   return ret;
 }