mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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329 lines
9.4 KiB
C
329 lines
9.4 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpmpapay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpmpapay_debug);
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#define GST_CAT_DEFAULT (rtpmpapay_debug)
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static GstStaticPadTemplate gst_rtp_mpa_pay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/mpeg, " "mpegversion = (int) 1")
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);
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static GstStaticPadTemplate gst_rtp_mpa_pay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_MPA_STRING ", "
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"clock-rate = (int) 90000; "
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) 90000, " "encoding-name = (string) \"MPA\"")
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);
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static void gst_rtp_mpa_pay_finalize (GObject * object);
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static GstStateChangeReturn gst_rtp_mpa_pay_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static gboolean gst_rtp_mpa_pay_sink_event (GstRTPBasePayload * payload,
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GstEvent * event);
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static GstFlowReturn gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay);
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static GstFlowReturn gst_rtp_mpa_pay_handle_buffer (GstRTPBasePayload * payload,
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GstBuffer * buffer);
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#define gst_rtp_mpa_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpMPAPay, gst_rtp_mpa_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_mpa_pay_class_init (GstRtpMPAPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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GST_DEBUG_CATEGORY_INIT (rtpmpapay_debug, "rtpmpapay", 0,
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"MPEG Audio RTP Depayloader");
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
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gobject_class->finalize = gst_rtp_mpa_pay_finalize;
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gstelement_class->change_state = gst_rtp_mpa_pay_change_state;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_mpa_pay_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_mpa_pay_sink_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP MPEG audio payloader", "Codec/Payloader/Network/RTP",
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"Payload MPEG audio as RTP packets (RFC 2038)",
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"Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasepayload_class->set_caps = gst_rtp_mpa_pay_setcaps;
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gstrtpbasepayload_class->sink_event = gst_rtp_mpa_pay_sink_event;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_mpa_pay_handle_buffer;
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}
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static void
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gst_rtp_mpa_pay_init (GstRtpMPAPay * rtpmpapay)
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{
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rtpmpapay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_mpa_pay_finalize (GObject * object)
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{
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GstRtpMPAPay *rtpmpapay;
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rtpmpapay = GST_RTP_MPA_PAY (object);
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g_object_unref (rtpmpapay->adapter);
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rtpmpapay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_rtp_mpa_pay_reset (GstRtpMPAPay * pay)
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{
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pay->first_ts = -1;
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pay->duration = 0;
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gst_adapter_clear (pay->adapter);
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GST_DEBUG_OBJECT (pay, "reset depayloader");
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}
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static gboolean
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gst_rtp_mpa_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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gboolean res;
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gst_rtp_base_payload_set_options (payload, "audio", TRUE, "MPA", 90000);
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res = gst_rtp_base_payload_set_outcaps (payload, NULL);
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return res;
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}
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static gboolean
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gst_rtp_mpa_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
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{
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gboolean ret;
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GstRtpMPAPay *rtpmpapay;
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rtpmpapay = GST_RTP_MPA_PAY (payload);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:
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/* make sure we push the last packets in the adapter on EOS */
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gst_rtp_mpa_pay_flush (rtpmpapay);
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break;
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case GST_EVENT_FLUSH_STOP:
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gst_rtp_mpa_pay_reset (rtpmpapay);
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break;
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default:
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break;
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}
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ret = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
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return ret;
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}
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static GstFlowReturn
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gst_rtp_mpa_pay_flush (GstRtpMPAPay * rtpmpapay)
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{
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guint avail;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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guint16 frag_offset;
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/* the data available in the adapter is either smaller
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* than the MTU or bigger. In the case it is smaller, the complete
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* adapter contents can be put in one packet. In the case the
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* adapter has more than one MTU, we need to split the MPA data
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* over multiple packets. The frag_offset in each packet header
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* needs to be updated with the position in the MPA frame. */
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avail = gst_adapter_available (rtpmpapay->adapter);
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ret = GST_FLOW_OK;
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frag_offset = 0;
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while (avail > 0) {
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guint towrite;
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guint8 *payload;
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guint payload_len;
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guint packet_len;
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GstRTPBuffer rtp = { NULL };
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GstBuffer *paybuf;
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/* this will be the total length of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (4 + avail, 0, 0);
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/* fill one MTU or all available bytes */
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towrite = MIN (packet_len, GST_RTP_BASE_PAYLOAD_MTU (rtpmpapay));
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/* this is the payload length */
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payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
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/* create buffer to hold the payload */
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outbuf = gst_rtp_buffer_new_allocate (4, 0, 0);
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gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
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payload_len -= 4;
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gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_PAYLOAD_MPA);
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/*
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* 0 1 2 3
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* 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
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* | MBZ | Frag_offset |
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*/
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payload = gst_rtp_buffer_get_payload (&rtp);
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payload[0] = 0;
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payload[1] = 0;
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payload[2] = frag_offset >> 8;
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payload[3] = frag_offset & 0xff;
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avail -= payload_len;
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frag_offset += payload_len;
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if (avail == 0)
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gst_rtp_buffer_set_marker (&rtp, TRUE);
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gst_rtp_buffer_unmap (&rtp);
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paybuf = gst_adapter_take_buffer_fast (rtpmpapay->adapter, payload_len);
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outbuf = gst_buffer_append (outbuf, paybuf);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpmpapay->first_ts;
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GST_BUFFER_DURATION (outbuf) = rtpmpapay->duration;
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ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmpapay), outbuf);
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}
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return ret;
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}
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static GstFlowReturn
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gst_rtp_mpa_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpMPAPay *rtpmpapay;
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GstFlowReturn ret;
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guint size, avail;
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guint packet_len;
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GstClockTime duration, timestamp;
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rtpmpapay = GST_RTP_MPA_PAY (basepayload);
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size = gst_buffer_get_size (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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timestamp = GST_BUFFER_TIMESTAMP (buffer);
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if (GST_BUFFER_IS_DISCONT (buffer)) {
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GST_DEBUG_OBJECT (rtpmpapay, "DISCONT");
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gst_rtp_mpa_pay_reset (rtpmpapay);
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}
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avail = gst_adapter_available (rtpmpapay->adapter);
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/* get packet length of previous data and this new data,
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* payload length includes a 4 byte header */
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packet_len = gst_rtp_buffer_calc_packet_len (4 + avail + size, 0, 0);
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/* if this buffer is going to overflow the packet, flush what we
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* have. */
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if (gst_rtp_base_payload_is_filled (basepayload,
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packet_len, rtpmpapay->duration + duration)) {
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ret = gst_rtp_mpa_pay_flush (rtpmpapay);
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avail = 0;
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} else {
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ret = GST_FLOW_OK;
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}
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if (avail == 0) {
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GST_DEBUG_OBJECT (rtpmpapay,
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"first packet, save timestamp %" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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rtpmpapay->first_ts = timestamp;
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rtpmpapay->duration = 0;
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}
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gst_adapter_push (rtpmpapay->adapter, buffer);
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rtpmpapay->duration = duration;
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return ret;
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}
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static GstStateChangeReturn
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gst_rtp_mpa_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstRtpMPAPay *rtpmpapay;
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GstStateChangeReturn ret;
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rtpmpapay = GST_RTP_MPA_PAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_rtp_mpa_pay_reset (rtpmpapay);
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_rtp_mpa_pay_reset (rtpmpapay);
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break;
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default:
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break;
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}
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return ret;
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}
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gboolean
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gst_rtp_mpa_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpmpapay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_MPA_PAY);
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}
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