gstreamer/gst/rtp/gstrtpmpapay.c
2021-03-29 12:45:22 +02:00

341 lines
9.8 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_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;
}