/* 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 <gst/rtp/gstrtpbuffer.h>
+#include <stdlib.h>
#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 parser",
- "Codec/Depayr/Network",
- "Extracts AMR audio from RTP packets (RFC 3267)",
- "Wim Taymans <wim@fluendo.com>");
-
/* RtpAMRDepay signals and args */
enum
{
* params see RFC 3267, section 8.1
*/
static GstStaticPadTemplate gst_rtp_amr_depay_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
+ "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\""
);
static GstStaticPadTemplate gst_rtp_amr_depay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+ 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 gboolean gst_rtp_amr_depay_setcaps (GstBaseRTPDepayload * depayload,
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
depayload = GST_BASE_RTP_DEPAYLOAD (rtpamrdepay);
- depayload->clock_rate = 8000;
gst_pad_use_fixed_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload));
}
GstCaps *srccaps;
GstRtpAMRDepay *rtpamrdepay;
const gchar *params;
- const gchar *str;
- gint clock_rate;
+ const gchar *str, *type;
+ gint clock_rate, need_clock_rate;
+ gboolean res;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
+ /* 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
}
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
- clock_rate = 8000;
+ clock_rate = need_clock_rate;
+ depayload->clock_rate = clock_rate;
/* we require 1 channel, 8000 Hz, octet aligned, no CRC,
* no robust sorting, no interleaving for now */
if (rtpamrdepay->channels != 1)
return FALSE;
- if (clock_rate != 8000)
+ if (clock_rate != need_clock_rate)
return FALSE;
if (rtpamrdepay->octet_align != TRUE)
return FALSE;
if (rtpamrdepay->interleaving != FALSE)
return FALSE;
- srccaps = gst_caps_new_simple ("audio/AMR",
+ 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 res;
- return TRUE;
+ /* ERRORS */
+invalid_mode:
+ {
+ GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
+ return FALSE;
+ }
}
/* -1 is invalid */
-static gint 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 const gint wb_frame_size[16] = {
+ 17, 23, 32, 36, 40, 46, 50, 58,
+ 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;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
- if (!rtpamrdepay->negotiated)
- goto not_negotiated;
-
- if (!gst_rtp_buffer_validate (buf)) {
- GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
- (NULL), ("AMR RTP packet did not validate"));
- goto bad_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 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 */
{
- gint payload_len;
guint8 *payload, *p, *dp;
- guint32 timestamp;
guint8 CMR;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
payload_len = gst_rtp_buffer_get_payload_len (buf);
/* need at least 2 bytes for the header */
- if (payload_len < 2) {
- GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
- (NULL), ("AMR RTP payload too small (%d)", payload_len));
- goto bad_packet;
- }
+ if (payload_len < 2)
+ goto too_small;
payload = gst_rtp_buffer_get_payload (buf);
/* 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|
* +-+-+-+-+-+-+-+-+
payload_len -= 1;
payload += 1;
- if (ILP > ILL) {
- GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
- (NULL), ("AMR RTP wrong interleaving"));
- goto bad_packet;
- }
+ if (ILP > ILL)
+ 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..
* +-+-+-+-+-+-+-+-+..
fr_size = frame_size[FT];
GST_DEBUG_OBJECT (rtpamrdepay, "frame size %d", fr_size);
- if (fr_size == -1) {
- GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
- (NULL), ("AMR RTP frame size == -1"));
- goto bad_packet;
- }
+ if (fr_size == -1)
+ goto wrong_framesize;
if (fr_size > 0) {
amr_len += fr_size;
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) {
- GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
- (NULL), ("AMR RTP wrong length 1"));
- goto bad_packet;
- }
+ 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) {
- GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
- (NULL), ("AMR RTP wrong length 2"));
- goto bad_packet;
- }
+ if (num_packets + amr_len > payload_len)
+ goto wrong_length_2;
}
- timestamp = gst_rtp_buffer_get_timestamp (buf);
-
outbuf = gst_buffer_new_and_alloc (payload_len);
- GST_BUFFER_TIMESTAMP (outbuf) =
- gst_util_uint64_scale_int (timestamp, GST_SECOND,
- depayload->clock_rate);
/* point to destination */
p = GST_BUFFER_DATA (outbuf);
dp += fr_size;
}
}
- gst_buffer_set_caps (outbuf,
- GST_PAD_CAPS (GST_BASE_RTP_DEPAYLOAD_SRCPAD (depayload)));
+ /* we can set the duration because each packet is 20 milliseconds */
+ GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
+
+ 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 */
-not_negotiated:
+too_small:
{
- GST_ELEMENT_ERROR (rtpamrdepay, STREAM, NOT_IMPLEMENTED,
- (NULL), ("not negotiated"));
- return NULL;
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP payload too small (%d)", payload_len));
+ goto bad_packet;
+ }
+wrong_interleaving:
+ {
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP wrong interleaving"));
+ goto bad_packet;
+ }
+wrong_framesize:
+ {
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP frame size == -1"));
+ goto bad_packet;
+ }
+wrong_length_1:
+ {
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP wrong length 1"));
+ goto bad_packet;
+ }
+wrong_length_2:
+ {
+ GST_ELEMENT_WARNING (rtpamrdepay, STREAM, DECODE,
+ (NULL), ("AMR RTP wrong length 2"));
+ goto bad_packet;
}
bad_packet:
{
gst_rtp_amr_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpamrdepay",
- GST_RANK_NONE, GST_TYPE_RTP_AMR_DEPAY);
+ GST_RANK_MARGINAL, GST_TYPE_RTP_AMR_DEPAY);
}
projects / platform / upstream / gst-plugins-good.git / blobdiff