/* GStreamer
- * Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
+ * Copyright (C) <2005> 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
* Wideband (AMR-WB) Audio Codecs.
*/
-/* elementfactory information */
-static const GstElementDetails gst_rtp_amrdepay_details =
-GST_ELEMENT_DETAILS ("RTP packet depayloader",
- "Codec/Depayloader/Network",
- "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
- "Wim Taymans <wim@fluendo.com>");
-
/* RtpAMRDepay signals and args */
enum
{
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_amr_depay_sink_template));
- gst_element_class_set_details (element_class, &gst_rtp_amrdepay_details);
+ gst_element_class_set_details_simple (element_class, "RTP AMR depayloader",
+ "Codec/Depayloader/Network",
+ "Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
+ "Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
{
- GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
- gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
- parent_class = g_type_class_peek_parent (klass);
-
gstbasertpdepayload_class->process = gst_rtp_amr_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_amr_depay_setcaps;
}
/* -1 is invalid */
-static gint nb_frame_size[16] = {
+static const gint nb_frame_size[16] = {
12, 13, 15, 17, 19, 20, 26, 31,
5, -1, -1, -1, -1, -1, -1, 0
};
-static gint wb_frame_size[16] = {
+
+static const gint wb_frame_size[16] = {
17, 23, 32, 36, 40, 46, 50, 58,
- 60, -1, -1, -1, -1, -1, -1, 0
+ 60, 5, -1, -1, -1, -1, -1, 0
};
static GstBuffer *
gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpAMRDepay *rtpamrdepay;
+ const gint *frame_size;
GstBuffer *outbuf = NULL;
gint payload_len;
- gint *frame_size;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
else
frame_size = wb_frame_size;
- /* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
+ /* when we get here, 1 channel, 8000/16000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving data is to be depayloaded */
{
guint8 *payload, *p, *dp;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
gint ILL, ILP;
- gboolean marker;
- marker = gst_rtp_buffer_get_marker (buf);
payload_len = gst_rtp_buffer_get_payload_len (buf);
/* need at least 2 bytes for the header */
/* depay CMR. The CMR is used by the sender to request
* a new encoding mode.
*
- * 0 1 2 3 4 5 6 7
+ * 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* | CMR |R|R|R|R|
* +-+-+-+-+-+-+-+-+
goto wrong_interleaving;
}
- /*
- * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
+ /*
+ * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
* +-+-+-+-+-+-+-+-+..
* |F| FT |Q|P|P| more FT..
* +-+-+-+-+-+-+-+-+..
/* we can set the duration because each packet is 20 milliseconds */
GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
- if (marker) {
+ if (gst_rtp_buffer_get_marker (buf)) {
/* marker bit marks a discont buffer after a talkspurt. */
GST_DEBUG_OBJECT (depayload, "marker bit was set");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- marker = FALSE;
}
GST_DEBUG_OBJECT (depayload, "pushing buffer of size %d",