gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpmpapay.c
Sebastian Dröge b0afaffc5d rtp: In payloaders map the RTP marker flag to the corresponding buffer flag
This allows downstream of a payloader to know the RTP header's marker
flag without first having to map the buffer and parse the RTP header.

Especially inside RTP header extension implementations this can be
useful to decide which packet corresponds to e.g. the last packet of a
video frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1776>
2022-02-28 10:13:11 +00:00

343 lines
9.9 KiB
C

/* 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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/audio/audio.h>
#include "gstrtpelements.h"
#include "gstrtpmpapay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpmpapay_debug);
#define GST_CAT_DEFAULT (rtpmpapay_debug)
static GstStaticPadTemplate gst_rtp_mpa_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1")
);
static GstStaticPadTemplate gst_rtp_mpa_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_MPA_STRING ", "
"clock-rate = (int) 90000; "
"application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\"")
);
static void gst_rtp_mpa_pay_finalize (GObject * object);
static GstStateChangeReturn gst_rtp_mpa_pay_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static gboolean gst_rtp_mpa_pay_sink_event (GstRTPBasePayload * payload,
GstEvent * event);
static GstFlowReturn gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay);
static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstRTPBasePayload * payload,
GstBuffer * buffer);
#define gst_rtp_mpa_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpMPAPay, gst_rtp_mpa_pay, GST_TYPE_RTP_BASE_PAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmpapay, "rtpmpapay",
GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_PAY, rtp_element_init (plugin));
static void
gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpmpapay_debug, "rtpmpapay", 0,
"MPEG Audio RTP Depayloader");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->finalize = gst_rtp_mpa_pay_finalize;
gstelement_class->change_state = gst_rtp_mpa_pay_change_state;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_mpa_pay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_mpa_pay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP MPEG audio payloader", "Codec/Payloader/Network/RTP",
"Payload MPEG audio as RTP packets (RFC 2038)",
"Wim Taymans <wim.taymans@gmail.com>");
gstrtpbasepayload_class->set_caps = gst_rtp_mpa_pay_setcaps;
gstrtpbasepayload_class->sink_event = gst_rtp_mpa_pay_sink_event;
gstrtpbasepayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer;
}
static void
gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
{
rtpmpapay->adapter = gst_adapter_new ();
GST_RTP_BASE_PAYLOAD (rtpmpapay)->pt = GST_RTP_PAYLOAD_MPA;
}
static void
gst_rtp_mpa_pay_finalize (GObject * object)
{
GstRtpMPAPay *rtpmpapay;
rtpmpapay = GST_RTP_MPA_PAY (object);
g_object_unref (rtpmpapay->adapter);
rtpmpapay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_mpa_pay_reset (GstRtpMPAPay * pay)
{
pay->first_ts = -1;
pay->duration = 0;
gst_adapter_clear (pay->adapter);
GST_DEBUG_OBJECT (pay, "reset depayloader");
}
static gboolean
gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gst_rtp_base_payload_set_options (payload, "audio",
payload->pt != GST_RTP_PAYLOAD_MPA, "MPA", 90000);
res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static gboolean
gst_rtp_mpa_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
gboolean ret;
GstRtpMPAPay *rtpmpapay;
rtpmpapay = GST_RTP_MPA_PAY (payload);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
/* make sure we push the last packets in the adapter on EOS */
gst_rtp_mpa_pay_flush (rtpmpapay);
break;
case GST_EVENT_FLUSH_STOP:
gst_rtp_mpa_pay_reset (rtpmpapay);
break;
default:
break;
}
ret = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
return ret;
}
#define RTP_HEADER_LEN 12
static GstFlowReturn
gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint16 frag_offset;
GstBufferList *list;
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
* adapter has more than one MTU, we need to split the MPA data
* over multiple packets. The frag_offset in each packet header
* needs to be updated with the position in the MPA frame. */
avail = gst_adapter_available (rtpmpapay->adapter);
ret = GST_FLOW_OK;
list =
gst_buffer_list_new_sized (avail / (GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay) -
RTP_HEADER_LEN) + 1);
frag_offset = 0;
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
GstRTPBuffer rtp = { NULL };
GstBuffer *paybuf;
/* this will be the total length of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
/* fill one MTU or all available bytes */
towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay));
/* this is the payload length */
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
outbuf =
gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
(rtpmpapay), 4, 0, 0);
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload_len -= 4;
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_PAYLOAD_MPA);
/*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | MBZ | Frag_offset |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
payload = gst_rtp_buffer_get_payload (&rtp);
payload[0] = 0;
payload[1] = 0;
payload[2] = frag_offset >> 8;
payload[3] = frag_offset & 0xff;
avail -= payload_len;
frag_offset += payload_len;
if (avail == 0) {
gst_rtp_buffer_set_marker (&rtp, TRUE);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER);
}
gst_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (rtpmpapay->adapter, payload_len);
gst_rtp_copy_audio_meta (rtpmpapay, outbuf, paybuf);
outbuf = gst_buffer_append (outbuf, paybuf);
GST_BUFFER_PTS (outbuf) = rtpmpapay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
gst_buffer_list_add (list, outbuf);
}
ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmpapay), list);
return ret;
}
static GstFlowReturn
gst_rtp_mpa_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpMPAPay *rtpmpapay;
GstFlowReturn ret;
guint size, avail;
guint packet_len;
GstClockTime duration, timestamp;
rtpmpapay = GST_RTP_MPA_PAY (basepayload);
size = gst_buffer_get_size (buffer);
duration = GST_BUFFER_DURATION (buffer);
timestamp = GST_BUFFER_PTS (buffer);
if (GST_BUFFER_IS_DISCONT (buffer)) {
GST_DEBUG_OBJECT (rtpmpapay, "DISCONT");
gst_rtp_mpa_pay_reset (rtpmpapay);
}
avail = gst_adapter_available (rtpmpapay->adapter);
/* get packet length of previous data and this new data,
* payload length includes a 4 byte header */
packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_rtp_base_payload_is_filled (basepayload,
packet_len, rtpmpapay->duration + duration)) {
ret = gst_rtp_mpa_pay_flush (rtpmpapay);
avail = 0;
} else {
ret = GST_FLOW_OK;
}
if (avail == 0) {
GST_DEBUG_OBJECT (rtpmpapay,
"first packet, save timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
rtpmpapay->first_ts = timestamp;
rtpmpapay->duration = 0;
}
gst_adapter_push (rtpmpapay->adapter, buffer);
rtpmpapay->duration = duration;
return ret;
}
static GstStateChangeReturn
gst_rtp_mpa_pay_change_state (GstElement * element, GstStateChange transition)
{
GstRtpMPAPay *rtpmpapay;
GstStateChangeReturn ret;
rtpmpapay = GST_RTP_MPA_PAY (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_rtp_mpa_pay_reset (rtpmpapay);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtp_mpa_pay_reset (rtpmpapay);
break;
default:
break;
}
return ret;
}