/* 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
#include <string.h>
#include "gstrtpamrdepay.h"
+GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
+#define GST_CAT_DEFAULT (rtpamrdepay_debug)
+
/* references:
*
* RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
* 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;
+
+ GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0,
+ "AMR/AMR-WB RTP Depayloader");
}
static void
const gchar *params;
const gchar *str, *type;
gint clock_rate, need_clock_rate;
+ gboolean res;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
if ((str = gst_structure_get_string (structure, "encoding-name"))) {
if (strcmp (str, "AMR") == 0) {
rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
- clock_rate = need_clock_rate = 8000;
+ need_clock_rate = 8000;
type = "audio/AMR";
} else if (strcmp (str, "AMR-WB") == 0) {
rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
- clock_rate = need_clock_rate = 16000;
+ need_clock_rate = 16000;
type = "audio/AMR-WB";
} else
goto invalid_mode;
rtpamrdepay->channels = atoi (params);
}
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = need_clock_rate;
depayload->clock_rate = clock_rate;
/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
srccaps = gst_caps_new_simple (type,
"channels", G_TYPE_INT, rtpamrdepay->channels,
"rate", G_TYPE_INT, clock_rate, NULL);
-
- gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
+ res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
- rtpamrdepay->negotiated = TRUE;
-
- return TRUE;
+ return res;
/* ERRORS */
invalid_mode:
}
/* -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);
- if (!rtpamrdepay->negotiated)
- goto not_negotiated;
-
- if (!gst_rtp_buffer_validate (buf))
- goto invalid_packet;
-
/* setup frame size pointer */
if (rtpamrdepay->mode == GST_RTP_AMR_DP_MODE_NB)
frame_size = nb_frame_size;
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;
/* 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;
- gst_buffer_set_caps (outbuf,
- GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload)));
+ 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);
+ }
- GST_DEBUG ("gst_rtp_amr_depay_chain: pushing buffer of size %d",
+ GST_DEBUG_OBJECT (depayload, "pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
}
return outbuf;
/* ERRORS */
-invalid_packet:
- {
- GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
- (NULL), ("AMR RTP packet did not validate"));
- goto bad_packet;
- }
-not_negotiated:
- {
- GST_ELEMENT_ERROR (rtpamrdepay, STREAM, NOT_IMPLEMENTED,
- (NULL), ("not negotiated"));
- return NULL;
- }
too_small:
{
GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
projects / platform / upstream / gst-plugins-good.git / blobdiff