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GST-Tensordecoder-ov_ep/subprojects/gst-plugins-good/gst/rtp/gstrtpamrdepay.c

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/* GStreamer
* 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
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpamrdepay
* @title: rtpamrdepay
* @see_also: rtpamrpay
*
* Extract AMR audio from RTP packets according to RFC 3267.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
*
* ## Example pipeline
* |[
* gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1, payload=(int)96' ! rtpamrdepay ! amrnbdec ! pulsesink
* ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to
* the rtpamrpay example to create the RTP stream.
*
*/
/*
* 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.
*
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include <stdlib.h>
#include <string.h>
#include "gstrtpelements.h"
#include "gstrtpamrdepay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpamrdepay_debug);
#define GST_CAT_DEFAULT (rtpamrdepay_debug)
/* RtpAMRDepay signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0
};
/* input is an RTP packet
*
* params see RFC 3267, section 8.1
*/
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\", "
"clock-rate = (int) 8000, " "encoding-name = (string) \"AMR\", "
/* This is the default, so the peer doesn't have to specify it
* "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\";"
/* following options are not needed for a decoder
*
"crc = (string) { \"0\", \"1\" }, "
"robust-sorting = (string) \"0\", "
"interleaving = (string) \"0\";"
"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\", "
"clock-rate = (int) 16000, " "encoding-name = (string) \"AMR-WB\", "
/* This is the default, so the peer doesn't have to specify it
* "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\";"
/* following options are not needed for a decoder
*
"crc = (string) { \"0\", \"1\" }, "
"robust-sorting = (string) \"0\", "
"interleaving = (string) \"0\""
"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 ]"
*/
)
);
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;"
"audio/AMR-WB, " "channels = (int) 1," "rate = (int) 16000")
);
static gboolean gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_amr_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
#define gst_rtp_amr_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpAMRDepay, gst_rtp_amr_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpamrdepay, "rtpamrdepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_DEPAY, rtp_element_init (plugin));
static void
gst_rtp_amr_depay_class_init (GstRtpAMRDepayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_amr_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_amr_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP AMR depayloader", "Codec/Depayloader/Network/RTP",
"Extracts AMR or AMR-WB audio from RTP packets (RFC 3267)",
"Wim Taymans <wim.taymans@gmail.com>");
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_amr_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_amr_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpamrdepay_debug, "rtpamrdepay", 0,
"AMR/AMR-WB RTP Depayloader");
}
static void
gst_rtp_amr_depay_init (GstRtpAMRDepay * rtpamrdepay)
{
GstRTPBaseDepayload *depayload;
depayload = GST_RTP_BASE_DEPAYLOAD (rtpamrdepay);
gst_pad_use_fixed_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload));
}
static gboolean
gst_rtp_amr_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstCaps *srccaps;
GstRtpAMRDepay *rtpamrdepay;
const gchar *params;
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
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 */
rtpamrdepay->octet_align = TRUE;
}
if (!(str = gst_structure_get_string (structure, "robust-sorting")))
rtpamrdepay->robust_sorting = FALSE;
else
rtpamrdepay->robust_sorting = (atoi (str) == 1);
if (rtpamrdepay->robust_sorting) {
/* robust_sorting mode implies octet aligned mode */
rtpamrdepay->octet_align = TRUE;
}
if (!(str = gst_structure_get_string (structure, "interleaving")))
rtpamrdepay->interleaving = FALSE;
else
rtpamrdepay->interleaving = (atoi (str) == 1);
if (rtpamrdepay->interleaving) {
/* interleaving mode implies octet aligned mode */
rtpamrdepay->octet_align = TRUE;
}
if (!(params = gst_structure_get_string (structure, "encoding-params")))
rtpamrdepay->channels = 1;
else {
rtpamrdepay->channels = atoi (params);
}
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,
* no robust sorting, no interleaving for now */
if (rtpamrdepay->channels != 1)
return FALSE;
if (clock_rate != need_clock_rate)
return FALSE;
if (rtpamrdepay->octet_align != TRUE)
return FALSE;
if (rtpamrdepay->robust_sorting != FALSE)
return FALSE;
if (rtpamrdepay->interleaving != FALSE)
return FALSE;
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_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), srccaps);
gst_caps_unref (srccaps);
return res;
/* ERRORS */
invalid_mode:
{
GST_ERROR_OBJECT (rtpamrdepay, "invalid encoding-name");
return FALSE;
}
}
/* -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
};
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 (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpAMRDepay *rtpamrdepay;
const gint *frame_size;
GstBuffer *outbuf = NULL;
gint payload_len;
GstMapInfo map;
rtpamrdepay = GST_RTP_AMR_DEPAY (depayload);
/* 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 */
{
guint8 *payload, *p, *dp;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
gint ILL, ILP;
payload_len = gst_rtp_buffer_get_payload_len (rtp);
/* need at least 2 bytes for the header */
if (payload_len < 2)
goto too_small;
payload = gst_rtp_buffer_get_payload (rtp);
/* 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; */
/* strip CMR header now, pack FT and the data for the decoder */
payload_len -= 1;
payload += 1;
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);
/* point to destination */
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
/* point to first data packet */
p = map.data;
dp = payload + num_packets;
if (rtpamrdepay->crc) {
/* skip CRC if present */
dp += num_nonempty_packets;
}
for (i = 0; i < num_packets; i++) {
gint fr_size;
/* copy FT, clear F bit */
*p++ = payload[i] & 0x7f;
fr_size = frame_size[(payload[i] & 0x78) >> 3];
if (fr_size > 0) {
/* copy data packet, FIXME, calc CRC here. */
memcpy (p, dp, fr_size);
p += fr_size;
dp += fr_size;
}
}
gst_buffer_unmap (outbuf, &map);
/* 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 (rtp)) {
/* marker bit marks a buffer after a talkspurt. */
GST_DEBUG_OBJECT (depayload, "marker bit was set");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
}
GST_DEBUG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (outbuf));
gst_rtp_copy_audio_meta (rtpamrdepay, outbuf, rtp->buffer);
}
return outbuf;
/* ERRORS */
too_small:
{
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:
{
/* no fatal error */
return NULL;
}
}
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