/* 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-rtpamrpay
* @see_also: rtpamrdepay
*
* Payload AMR audio into RTP packets according to RFC 3267.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt
*
* <refsect2>
* <title>Example pipeline</title>
* |[
* gst-launch-1.0 -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink
* ]| This example pipeline will encode and payload an AMR stream. Refer to
* the rtpamrdepay example to depayload and decode the RTP stream.
* </refsect2>
*/
/* 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.
*
* ETSI TS 126 201 V6.0.0 (2004-12) - Digital cellular telecommunications system (Phase 2+);
* Universal Mobile Telecommunications System (UMTS);
* AMR speech codec, wideband;
* Frame structure
* (3GPP TS 26.201 version 6.0.0 Release 6)
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpamrpay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpamrpay_debug);
#define GST_CAT_DEFAULT (rtpamrpay_debug)
static GstStaticPadTemplate gst_rtp_amr_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/AMR, channels=(int)1, rate=(int)8000; "
"audio/AMR-WB, channels=(int)1, rate=(int)16000")
);
static GstStaticPadTemplate gst_rtp_amr_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
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\", "
"octet-align = (string) \"1\", "
"crc = (string) \"0\", "
"robust-sorting = (string) \"0\", "
"interleaving = (string) \"0\", "
"mode-set = (int) [ 0, 7 ], "
"mode-change-period = (int) [ 1, MAX ], "
"mode-change-neighbor = (string) { \"0\", \"1\" }, "
"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\", "
"octet-align = (string) \"1\", "
"crc = (string) \"0\", "
"robust-sorting = (string) \"0\", "
"interleaving = (string) \"0\", "
"mode-set = (int) [ 0, 7 ], "
"mode-change-period = (int) [ 1, MAX ], "
"mode-change-neighbor = (string) { \"0\", \"1\" }, "
"maxptime = (int) [ 20, MAX ], " "ptime = (int) [ 20, MAX ]")
);
static gboolean gst_rtp_amr_pay_setcaps (GstRTPBasePayload * basepayload,
GstCaps * caps);
static GstFlowReturn gst_rtp_amr_pay_handle_buffer (GstRTPBasePayload * pad,
GstBuffer * buffer);
static GstStateChangeReturn
gst_rtp_amr_pay_change_state (GstElement * element, GstStateChange transition);
#define gst_rtp_amr_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpAMRPay, gst_rtp_amr_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_amr_pay_class_init (GstRtpAMRPayClass * klass)
{
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gstelement_class->change_state = gst_rtp_amr_pay_change_state;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_amr_pay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_amr_pay_sink_template);
gst_element_class_set_static_metadata (gstelement_class, "RTP AMR payloader",
"Codec/Payloader/Network/RTP",
"Payload-encode AMR or AMR-WB audio into RTP packets (RFC 3267)",
"Wim Taymans <wim.taymans@gmail.com>");
gstrtpbasepayload_class->set_caps = gst_rtp_amr_pay_setcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_amr_pay_handle_buffer;
GST_DEBUG_CATEGORY_INIT (rtpamrpay_debug, "rtpamrpay", 0,
"AMR/AMR-WB RTP Payloader");
}
static void
gst_rtp_amr_pay_init (GstRtpAMRPay * rtpamrpay)
{
}
static void
gst_rtp_amr_pay_reset (GstRtpAMRPay * pay)
{
pay->next_rtp_time = 0;
pay->first_ts = GST_CLOCK_TIME_NONE;
pay->first_rtp_time = 0;
}
static gboolean
gst_rtp_amr_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpAMRPay *rtpamrpay;
gboolean res;
const GstStructure *s;
const gchar *str;
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
/* figure out the mode Narrow or Wideband */
s = gst_caps_get_structure (caps, 0);
if ((str = gst_structure_get_name (s))) {
if (strcmp (str, "audio/AMR") == 0)
rtpamrpay->mode = GST_RTP_AMR_P_MODE_NB;
else if (strcmp (str, "audio/AMR-WB") == 0)
rtpamrpay->mode = GST_RTP_AMR_P_MODE_WB;
else
goto wrong_type;
} else
goto wrong_type;
if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "AMR", 8000);
else
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "AMR-WB",
16000);
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, "1", "octet-align", G_TYPE_STRING, "1",
/* don't set the defaults
*
* "crc", G_TYPE_STRING, "0",
* "robust-sorting", G_TYPE_STRING, "0",
* "interleaving", G_TYPE_STRING, "0",
*/
NULL);
return res;
/* ERRORS */
wrong_type:
{
GST_ERROR_OBJECT (rtpamrpay, "unsupported media type '%s'",
GST_STR_NULL (str));
return FALSE;
}
}
static void
gst_rtp_amr_pay_recalc_rtp_time (GstRtpAMRPay * rtpamrpay,
GstClockTime timestamp)
{
/* re-sync rtp time */
if (GST_CLOCK_TIME_IS_VALID (rtpamrpay->first_ts) &&
GST_CLOCK_TIME_IS_VALID (timestamp) && timestamp >= rtpamrpay->first_ts) {
GstClockTime diff;
guint32 rtpdiff;
/* interpolate to reproduce gap from start, rather than intermediate
* intervals to avoid roundup accumulation errors */
diff = timestamp - rtpamrpay->first_ts;
rtpdiff = ((diff / GST_MSECOND) * 8) <<
(rtpamrpay->mode == GST_RTP_AMR_P_MODE_WB);
rtpamrpay->next_rtp_time = rtpamrpay->first_rtp_time + rtpdiff;
GST_DEBUG_OBJECT (rtpamrpay,
"elapsed time %" GST_TIME_FORMAT ", rtp %" G_GUINT32_FORMAT ", "
"new offset %" G_GUINT32_FORMAT, GST_TIME_ARGS (diff), rtpdiff,
rtpamrpay->next_rtp_time);
}
}
/* -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 GstFlowReturn
gst_rtp_amr_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpAMRPay *rtpamrpay;
const gint *frame_size;
GstFlowReturn ret;
guint payload_len;
GstMapInfo map;
GstBuffer *outbuf;
guint8 *payload, *ptr, *payload_amr;
GstClockTime timestamp, duration;
guint packet_len, mtu;
gint i, num_packets, num_nonempty_packets;
gint amr_len;
gboolean sid = FALSE;
GstRTPBuffer rtp = { NULL };
rtpamrpay = GST_RTP_AMR_PAY (basepayload);
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpamrpay);
gst_buffer_map (buffer, &map, GST_MAP_READ);
timestamp = GST_BUFFER_PTS (buffer);
duration = GST_BUFFER_DURATION (buffer);
/* setup frame size pointer */
if (rtpamrpay->mode == GST_RTP_AMR_P_MODE_NB)
frame_size = nb_frame_size;
else
frame_size = wb_frame_size;
GST_DEBUG_OBJECT (basepayload, "got %" G_GSIZE_FORMAT " bytes", map.size);
/* FIXME, only
* octet aligned, no interleaving, single channel, no CRC,
* no robust-sorting. To fix this you need to implement the downstream
* negotiation function. */
/* first count number of packets and total amr frame size */
amr_len = num_packets = num_nonempty_packets = 0;
for (i = 0; i < map.size; i++) {
guint8 FT;
gint fr_size;
FT = (map.data[i] & 0x78) >> 3;
fr_size = frame_size[FT];
GST_DEBUG_OBJECT (basepayload, "frame type %d, frame size %d", FT, fr_size);
/* FIXME, we don't handle this yet.. */
if (fr_size <= 0)
goto wrong_size;
if (fr_size == 5)
sid = TRUE;
amr_len += fr_size;
num_nonempty_packets++;
num_packets++;
i += fr_size;
}
if (amr_len > map.size)
goto incomplete_frame;
/* we need one extra byte for the CMR, the ToC is in the input
* data */
payload_len = map.size + 1;
/* get packet len to check against MTU */
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
if (packet_len > mtu)
goto too_big;
/* now alloc output buffer */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* copy timestamp */
GST_BUFFER_PTS (outbuf) = timestamp;
if (duration != GST_CLOCK_TIME_NONE)
GST_BUFFER_DURATION (outbuf) = duration;
else {
GST_BUFFER_DURATION (outbuf) = num_packets * 20 * GST_MSECOND;
}
if (GST_BUFFER_IS_DISCONT (buffer)) {
GST_DEBUG_OBJECT (basepayload, "discont, setting marker bit");
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
gst_rtp_buffer_set_marker (&rtp, TRUE);
gst_rtp_amr_pay_recalc_rtp_time (rtpamrpay, timestamp);
}
if (G_UNLIKELY (sid)) {
gst_rtp_amr_pay_recalc_rtp_time (rtpamrpay, timestamp);
}
/* perfect rtptime */
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (rtpamrpay->first_ts))) {
rtpamrpay->first_ts = timestamp;
rtpamrpay->first_rtp_time = rtpamrpay->next_rtp_time;
}
GST_BUFFER_OFFSET (outbuf) = rtpamrpay->next_rtp_time;
rtpamrpay->next_rtp_time +=
(num_packets * 160) << (rtpamrpay->mode == GST_RTP_AMR_P_MODE_WB);
/* get payload, this is now writable */
payload = gst_rtp_buffer_get_payload (&rtp);
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* | CMR |R|R|R|R|
* +-+-+-+-+-+-+-+-+
*/
payload[0] = 0xF0; /* CMR, no specific mode requested */
/* this is where we copy the AMR data, after num_packets FTs and the
* CMR. */
payload_amr = payload + num_packets + 1;
/* copy data in payload, first we copy all the FTs then all
* the AMR data. The last FT has to have the F flag cleared. */
ptr = map.data;
for (i = 1; i <= num_packets; i++) {
guint8 FT;
gint fr_size;
/* 0 1 2 3 4 5 6 7
* +-+-+-+-+-+-+-+-+
* |F| FT |Q|P|P| more FT...
* +-+-+-+-+-+-+-+-+
*/
FT = (*ptr & 0x78) >> 3;
fr_size = frame_size[FT];
if (i == num_packets)
/* last packet, clear F flag */
payload[i] = *ptr & 0x7f;
else
/* set F flag */
payload[i] = *ptr | 0x80;
memcpy (payload_amr, &ptr[1], fr_size);
/* all sizes are > 0 since we checked for that above */
ptr += fr_size + 1;
payload_amr += fr_size;
}
gst_buffer_unmap (buffer, &map);
gst_rtp_buffer_unmap (&rtp);
gst_rtp_copy_audio_meta (rtpamrpay, outbuf, buffer);
gst_buffer_unref (buffer);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
return ret;
/* ERRORS */
wrong_size:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received AMR frame with size <= 0"));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
incomplete_frame:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received incomplete AMR frames"));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
too_big:
{
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
(NULL), ("received too many AMR frames for MTU"));
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
static GstStateChangeReturn
gst_rtp_amr_pay_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
/* handle upwards state changes here */
switch (transition) {
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
/* handle downwards state changes */
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtp_amr_pay_reset (GST_RTP_AMR_PAY (element));
break;
default:
break;
}
return ret;
}
gboolean
gst_rtp_amr_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpamrpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_AMR_PAY);
}