/* 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
* 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
+ * 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.
*/
#ifdef HAVE_CONFIG_H
#include <gst/rtp/gstrtpbuffer.h>
+#include <stdlib.h>
#include <string.h>
-#include "gstrtpamrdec.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 Storage Format
- * for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio
- * Codecs.
+ * RFC 3267 - Real-Time Transport Protocol (RTP) Payload Format and File
+ * Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate
+ * Wideband (AMR-WB) Audio Codecs.
*/
-/* elementfactory information */
-static GstElementDetails gst_rtp_amrdec_details = {
- "RTP packet parser",
- "Codec/Parser/Network",
- "Extracts AMR audio from RTP packets (RFC 3267)",
- "Wim Taymans <wim@fluendo.com>"
-};
-
-/* RtpAMRDec signals and args */
+/* RtpAMRDepay signals and args */
enum
{
/* FILL ME */
enum
{
- ARG_0,
- ARG_FREQUENCY
+ ARG_0
};
-/* input is an RTP packet
+/* input is an RTP packet
*
* params see RFC 3267, section 8.1
*/
-static GstStaticPadTemplate gst_rtpamrdec_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
+static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
+ GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) [ 96, 255 ], "
- "clock_rate = (int) 8000, "
- "encoding_name = (string) \"AMR\", "
- "encoding_params = (string) \"1\", "
- "octet-align = (boolean) TRUE, "
- "crc = (boolean) FALSE, "
- "robust-sorting = (boolean) FALSE, " "interleaving = (boolean) FALSE"
- /* following options are not needed for a decoder
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 8000, "
+ "encoding-name = (string) \"AMR\", "
+ "encoding-params = (string) \"1\", "
+ /* NOTE that all values must be strings in orde to be able to do SDP <->
+ * GstCaps mapping. */
+ "octet-align = (string) \"1\", "
+ "crc = (string) { \"0\", \"1\" }, "
+ "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\";"
+ /* following options are not needed for a decoder
+ *
+ "mode-set = (int) [ 0, 7 ], "
+ "mode-change-period = (int) [ 1, MAX ], "
+ "mode-change-neighbor = (boolean) { TRUE, FALSE }, "
+ "maxptime = (int) [ 20, MAX ], "
+ "ptime = (int) [ 20, MAX ]"
+ */
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 16000, "
+ "encoding-name = (string) \"AMR-WB\", "
+ "encoding-params = (string) \"1\", "
+ /* NOTE that all values must be strings in orde to be able to do SDP <->
+ * GstCaps mapping. */
+ "octet-align = (string) \"1\", "
+ "crc = (string) { \"0\", \"1\" }, "
+ "robust-sorting = (string) \"0\", " "interleaving = (string) \"0\""
+ /* following options are not needed for a decoder
*
"mode-set = (int) [ 0, 7 ], "
"mode-change-period = (int) [ 1, MAX ], "
)
);
-static GstStaticPadTemplate gst_rtpamrdec_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
+ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000")
+ GST_STATIC_CAPS ("audio/AMR, " "channels = (int) 1," "rate = (int) 8000;"
+ "audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
);
-static void gst_rtpamrdec_class_init (GstRtpAMRDecClass * klass);
-static void gst_rtpamrdec_base_init (GstRtpAMRDecClass * klass);
-static void gst_rtpamrdec_init (GstRtpAMRDec * rtpamrdec);
-
-static gboolean gst_rtpamrdec_sink_setcaps (GstPad * pad, GstCaps * caps);
-static GstFlowReturn gst_rtpamrdec_chain (GstPad * pad, GstBuffer * buffer);
-
-static void gst_rtpamrdec_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec);
-static void gst_rtpamrdec_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec);
-
-static GstStateChangeReturn gst_rtpamrdec_change_state (GstElement * element,
- GstStateChange transition);
+static gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload,
+ GstCaps * caps);
+static GstBuffer *gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload,
+ GstBuffer * buf);
-static GstElementClass *parent_class = NULL;
-
-static GType
-gst_rtpamrdec_get_type (void)
-{
- static GType rtpamrdec_type = 0;
-
- if (!rtpamrdec_type) {
- static const GTypeInfo rtpamrdec_info = {
- sizeof (GstRtpAMRDecClass),
- (GBaseInitFunc) gst_rtpamrdec_base_init,
- NULL,
- (GClassInitFunc) gst_rtpamrdec_class_init,
- NULL,
- NULL,
- sizeof (GstRtpAMRDec),
- 0,
- (GInstanceInitFunc) gst_rtpamrdec_init,
- };
-
- rtpamrdec_type =
- g_type_register_static (GST_TYPE_ELEMENT, "GstRtpAMRDec",
- &rtpamrdec_info, 0);
- }
- return rtpamrdec_type;
-}
+GST_BOILERPLATE (GstRtpAMRDepay, gst_rtp_amr_depay, GstBaseRTPDepayload,
+ GST_TYPE_BASE_RTP_DEPAYLOAD);
static void
-gst_rtpamrdec_base_init (GstRtpAMRDecClass * klass)
+gst_rtp_amr_depay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtpamrdec_src_template));
+ gst_static_pad_template_get (&gst_rtp_amr_depay_src_template));
gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtpamrdec_sink_template));
+ gst_static_pad_template_get (&gst_rtp_amr_depay_sink_template));
- gst_element_class_set_details (element_class, &gst_rtp_amrdec_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_rtpamrdec_class_init (GstRtpAMRDecClass * klass)
+gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
{
- GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
-
- gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
+ GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
- parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+ gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
- gobject_class->set_property = gst_rtpamrdec_set_property;
- gobject_class->get_property = gst_rtpamrdec_get_property;
+ gstbasertpdepayload_class->process = gst_rtp_amr_depay_process;
+ gstbasertpdepayload_class->set_caps = gst_rtp_amr_depay_setcaps;
- gstelement_class->change_state = gst_rtpamrdec_change_state;
+ GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0,
+ "AMR/AMR-WB RTP Depayloader");
}
static void
-gst_rtpamrdec_init (GstRtpAMRDec * rtpamrdec)
+gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay,
+ GstRtpAMRDepayClass * klass)
{
- rtpamrdec->srcpad =
- gst_pad_new_from_template (gst_static_pad_template_get
- (&gst_rtpamrdec_src_template), "src");
-
- gst_element_add_pad (GST_ELEMENT (rtpamrdec), rtpamrdec->srcpad);
-
- rtpamrdec->sinkpad =
- gst_pad_new_from_template (gst_static_pad_template_get
- (&gst_rtpamrdec_sink_template), "sink");
- gst_pad_set_setcaps_function (rtpamrdec->sinkpad, gst_rtpamrdec_sink_setcaps);
- gst_pad_set_chain_function (rtpamrdec->sinkpad, gst_rtpamrdec_chain);
- gst_element_add_pad (GST_ELEMENT (rtpamrdec), rtpamrdec->sinkpad);
+ GstBaseRTPDepayload *depayload;
+
+ depayload = GST_BASE_RTP_DEPAYLOAD (rtpamrdepay);
+
+ gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
}
static gboolean
-gst_rtpamrdec_sink_setcaps (GstPad * pad, GstCaps * caps)
+gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstCaps *srccaps;
- GstRtpAMRDec *rtpamrdec;
+ GstRtpAMRDepay *rtpamrdepay;
const gchar *params;
+ const gchar *str, *type;
+ gint clock_rate, need_clock_rate;
+ gboolean res;
- rtpamrdec = GST_RTP_AMR_DEC (GST_OBJECT_PARENT (pad));
+ rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_boolean (structure, "octet-align",
- &rtpamrdec->octet_align))
- rtpamrdec->octet_align = FALSE;
-
- if (!gst_structure_get_boolean (structure, "crc", &rtpamrdec->crc))
- rtpamrdec->crc = FALSE;
-
- if (rtpamrdec->crc) {
+ /* figure out the mode first and set the clock rates */
+ if ((str = gst_structure_get_string (structure, "encoding-name"))) {
+ if (strcmp (str, "AMR") == 0) {
+ rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_NB;
+ need_clock_rate = 8000;
+ type = "audio/AMR";
+ } else if (strcmp (str, "AMR-WB") == 0) {
+ rtpamrdepay->mode = GST_RTP_AMR_DP_MODE_WB;
+ need_clock_rate = 16000;
+ type = "audio/AMR-WB";
+ } else
+ goto invalid_mode;
+ } else
+ goto invalid_mode;
+
+ if (!(str = gst_structure_get_string (structure, "octet-align")))
+ rtpamrdepay->octet_align = FALSE;
+ else
+ rtpamrdepay->octet_align = (atoi (str) == 1);
+
+ if (!(str = gst_structure_get_string (structure, "crc")))
+ rtpamrdepay->crc = FALSE;
+ else
+ rtpamrdepay->crc = (atoi (str) == 1);
+
+ if (rtpamrdepay->crc) {
/* crc mode implies octet aligned mode */
- rtpamrdec->octet_align = TRUE;
+ rtpamrdepay->octet_align = TRUE;
}
- if (!gst_structure_get_boolean (structure, "robust-sorting",
- &rtpamrdec->robust_sorting))
- rtpamrdec->robust_sorting = FALSE;
+ if (!(str = gst_structure_get_string (structure, "robust-sorting")))
+ rtpamrdepay->robust_sorting = FALSE;
+ else
+ rtpamrdepay->robust_sorting = (atoi (str) == 1);
- if (rtpamrdec->robust_sorting) {
+ if (rtpamrdepay->robust_sorting) {
/* robust_sorting mode implies octet aligned mode */
- rtpamrdec->octet_align = TRUE;
+ rtpamrdepay->octet_align = TRUE;
}
- if (!gst_structure_get_boolean (structure, "interleaving",
- &rtpamrdec->interleaving))
- rtpamrdec->interleaving = FALSE;
+ if (!(str = gst_structure_get_string (structure, "interleaving")))
+ rtpamrdepay->interleaving = FALSE;
+ else
+ rtpamrdepay->interleaving = (atoi (str) == 1);
- if (rtpamrdec->interleaving) {
+ if (rtpamrdepay->interleaving) {
/* interleaving mode implies octet aligned mode */
- rtpamrdec->octet_align = TRUE;
+ rtpamrdepay->octet_align = TRUE;
}
- if (!(params = gst_structure_get_string (structure, "encoding_params")))
- rtpamrdec->channels = 1;
+ if (!(params = gst_structure_get_string (structure, "encoding-params")))
+ rtpamrdepay->channels = 1;
else {
- rtpamrdec->channels = atoi (params);
+ rtpamrdepay->channels = atoi (params);
}
- if (!gst_structure_get_int (structure, "clock_rate", &rtpamrdec->rate))
- rtpamrdec->rate = 8000;
+ 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,
+ /* we require 1 channel, 8000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving for now */
- if (rtpamrdec->channels != 1)
- return FALSE;
- if (rtpamrdec->rate != 8000)
+ if (rtpamrdepay->channels != 1)
return FALSE;
- if (rtpamrdec->octet_align != TRUE)
+ if (clock_rate != need_clock_rate)
return FALSE;
- if (rtpamrdec->crc != FALSE)
+ if (rtpamrdepay->octet_align != TRUE)
return FALSE;
- if (rtpamrdec->robust_sorting != FALSE)
+ if (rtpamrdepay->robust_sorting != FALSE)
return FALSE;
- if (rtpamrdec->interleaving != FALSE)
+ if (rtpamrdepay->interleaving != FALSE)
return FALSE;
- srccaps = gst_caps_new_simple ("audio/AMR",
- "channels", G_TYPE_INT, rtpamrdec->channels,
- "rate", G_TYPE_INT, rtpamrdec->rate, NULL);
- gst_pad_set_caps (rtpamrdec->srcpad, srccaps);
+ srccaps = gst_caps_new_simple (type,
+ "channels", G_TYPE_INT, rtpamrdepay->channels,
+ "rate", G_TYPE_INT, clock_rate, NULL);
+ res = gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
- rtpamrdec->negotiated = TRUE;
+ return res;
- return TRUE;
+ /* ERRORS */
+invalid_mode:
+ {
+ GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
+ return FALSE;
+ }
}
-static GstFlowReturn
-gst_rtpamrdec_chain (GstPad * pad, GstBuffer * buf)
-{
- GstRtpAMRDec *rtpamrdec;
- GstBuffer *outbuf;
- GstFlowReturn ret;
+/* -1 is invalid */
+static const gint nb_frame_size[16] = {
+ 12, 13, 15, 17, 19, 20, 26, 31,
+ 5, -1, -1, -1, -1, -1, -1, 0
+};
- rtpamrdec = GST_RTP_AMR_DEC (GST_OBJECT_PARENT (pad));
+static const gint wb_frame_size[16] = {
+ 17, 23, 32, 36, 40, 46, 50, 58,
+ 60, 5, -1, -1, -1, -1, -1, 0
+};
- if (!rtpamrdec->negotiated)
- goto not_negotiated;
+static GstBuffer *
+gst_rtp_amr_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+ GstRtpAMRDepay *rtpamrdepay;
+ const gint *frame_size;
+ GstBuffer *outbuf = NULL;
+ gint payload_len;
- if (!gst_rtpbuffer_validate (buf))
- goto bad_packet;
+ rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
- /* when we get here, 1 channel, 8000 Hz, octet aligned, no CRC,
- * no robust sorting, no interleaving data is to be parsed */
+ /* 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,
+ * no robust sorting, no interleaving data is to be depayloaded */
{
- gint payload_len;
- guint8 *payload;
- guint32 timestamp;
- guint8 CMR, F, FT, Q;
+ guint8 *payload, *p, *dp;
+ guint8 CMR;
+ gint i, num_packets, num_nonempty_packets;
+ gint amr_len;
+ gint ILL, ILP;
- payload_len = gst_rtpbuffer_get_payload_len (buf);
+ payload_len = gst_rtp_buffer_get_payload_len (buf);
/* need at least 2 bytes for the header */
if (payload_len < 2)
- goto bad_packet;
+ goto too_small;
- payload = gst_rtpbuffer_get_payload (buf);
+ payload = gst_rtp_buffer_get_payload (buf);
- /* parse header
- * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
- * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+..
- * | CMR |R|R|R|R|F| FT |Q|P|P|
- * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+..
+ /* depay CMR. The CMR is used by the sender to request
+ * a new encoding mode.
+ *
+ * 0 1 2 3 4 5 6 7
+ * +-+-+-+-+-+-+-+-+
+ * | CMR |R|R|R|R|
+ * +-+-+-+-+-+-+-+-+
*/
CMR = (payload[0] & 0xf0) >> 4;
- F = (payload[1] & 0x80) >> 7;
- /* we only support 1 packet per RTP packet for now */
- if (F != 0)
- goto one_packet_only;
-
- FT = (payload[1] & 0x78) >> 3;
- Q = (payload[1] & 0x04) >> 2;
-
- /* skip packet */
- if (FT > 9 && FT < 15) {
- ret = GST_FLOW_OK;
- goto skip;
- }
- /* strip header now, leave FT in the data for the decoder */
+ /* strip CMR header now, pack FT and the data for the decoder */
payload_len -= 1;
payload += 1;
- timestamp = gst_rtpbuffer_get_timestamp (buf);
+ GST_DEBUG_OBJECT (rtpamrdepay, "payload len %d", payload_len);
+
+ if (rtpamrdepay->interleaving) {
+ ILL = (payload[0] & 0xf0) >> 4;
+ ILP = (payload[0] & 0x0f);
+
+ payload_len -= 1;
+ payload += 1;
+
+ if (ILP > ILL)
+ goto wrong_interleaving;
+ }
+
+ /*
+ * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6
+ * +-+-+-+-+-+-+-+-+..
+ * |F| FT |Q|P|P| more FT..
+ * +-+-+-+-+-+-+-+-+..
+ */
+ /* count number of packets by counting the FTs. Also
+ * count number of amr data bytes and number of non-empty
+ * packets (this is also the number of CRCs if present). */
+ amr_len = 0;
+ num_nonempty_packets = 0;
+ num_packets = 0;
+ for (i = 0; i < payload_len; i++) {
+ gint fr_size;
+ guint8 FT;
+
+ FT = (payload[i] & 0x78) >> 3;
+
+ fr_size = frame_size[FT];
+ GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
+ if (fr_size == -1)
+ goto wrong_framesize;
+
+ if (fr_size > 0) {
+ amr_len += fr_size;
+ num_nonempty_packets++;
+ }
+ num_packets++;
+
+ if ((payload[i] & 0x80) == 0)
+ break;
+ }
+
+ if (rtpamrdepay->crc) {
+ /* data len + CRC len + header bytes should be smaller than payload_len */
+ if (num_packets + num_nonempty_packets + amr_len > payload_len)
+ goto wrong_length_1;
+ } else {
+ /* data len + header bytes should be smaller than payload_len */
+ if (num_packets + amr_len > payload_len)
+ goto wrong_length_2;
+ }
outbuf = gst_buffer_new_and_alloc (payload_len);
- GST_BUFFER_TIMESTAMP (outbuf) = timestamp * GST_SECOND / rtpamrdec->rate;
+ /* point to destination */
+ p = GST_BUFFER_DATA (outbuf);
+ /* point to first data packet */
+ dp = payload + num_packets;
+ if (rtpamrdepay->crc) {
+ /* skip CRC if present */
+ dp += num_nonempty_packets;
+ }
- memcpy (GST_BUFFER_DATA (outbuf), payload, payload_len);
+ for (i = 0; i < num_packets; i++) {
+ gint fr_size;
- gst_buffer_set_caps (outbuf, GST_PAD_CAPS (rtpamrdec->srcpad));
+ /* copy FT, clear F bit */
+ *p++ = payload[i] & 0x7f;
- GST_DEBUG ("gst_rtpamrdec_chain: pushing buffer of size %d",
- GST_BUFFER_SIZE (outbuf));
- ret = gst_pad_push (rtpamrdec->srcpad, outbuf);
+ fr_size = frame_size[(payload[i] & 0x78) >> 3];
+ if (fr_size > 0) {
+ /* copy data packet, FIXME, calc CRC here. */
+ memcpy (p, dp, fr_size);
- skip:
- gst_buffer_unref (buf);
- }
+ p += fr_size;
+ dp += fr_size;
+ }
+ }
+ /* we can set the duration because each packet is 20 milliseconds */
+ GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
- return ret;
+ 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);
+ }
-not_negotiated:
- {
- GST_DEBUG ("not_negotiated");
- gst_buffer_unref (buf);
- return GST_FLOW_NOT_NEGOTIATED;
+ GST_DEBUG_OBJECT (depayload, "pushing buffer of size %d",
+ GST_BUFFER_SIZE (outbuf));
}
-bad_packet:
+ return outbuf;
+
+ /* ERRORS */
+too_small:
{
- GST_DEBUG ("Packet did not validate");
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP payload too small (%d)", payload_len));
+ goto bad_packet;
}
-one_packet_only:
+wrong_interleaving:
{
- GST_DEBUG ("One packet per RTP packet only");
- gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP wrong interleaving"));
+ goto bad_packet;
}
-}
-
-static void
-gst_rtpamrdec_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstRtpAMRDec *rtpamrdec;
-
- rtpamrdec = GST_RTP_AMR_DEC (object);
-
- switch (prop_id) {
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
+wrong_framesize:
+ {
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP frame size == -1"));
+ goto bad_packet;
}
-}
-
-static void
-gst_rtpamrdec_get_property (GObject * object, guint prop_id, GValue * value,
- GParamSpec * pspec)
-{
- GstRtpAMRDec *rtpamrdec;
-
- rtpamrdec = GST_RTP_AMR_DEC (object);
-
- switch (prop_id) {
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
+wrong_length_1:
+ {
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP wrong length 1"));
+ goto bad_packet;
}
-}
-
-static GstStateChangeReturn
-gst_rtpamrdec_change_state (GstElement * element, GstStateChange transition)
-{
- GstRtpAMRDec *rtpamrdec;
- GstStateChangeReturn ret;
-
- rtpamrdec = GST_RTP_AMR_DEC (element);
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- break;
- default:
- break;
+wrong_length_2:
+ {
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP wrong length 2"));
+ goto bad_packet;
}
-
- ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- switch (transition) {
- case GST_STATE_CHANGE_READY_TO_NULL:
- break;
- default:
- break;
+bad_packet:
+ {
+ /* no fatal error */
+ return NULL;
}
- return ret;
}
gboolean
-gst_rtpamrdec_plugin_init (GstPlugin * plugin)
+gst_rtp_amr_depay_plugin_init (GstPlugin * plugin)
{
- return gst_element_register (plugin, "rtpamrdec",
- GST_RANK_NONE, GST_TYPE_RTP_AMR_DEC);
+ return gst_element_register (plugin, "rtpamrdepay",
+ GST_RANK_MARGINAL, GST_TYPE_RTP_AMR_DEPAY);
}
projects / platform / upstream / gst-plugins-good.git / blobdiff