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3df533de2c
Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtpilbcdepay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init): * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process): * gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process): * gst/rtp/gstrtppcmadepay.c: * gst/rtp/gstrtppcmudepay.c: * gst/rtp/gstrtpspeexdepay.c: * gst/rtp/gstrtpspeexpay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Fix caps with payload numbers. Add some fixed payload numbers to caps when possible.
496 lines
14 KiB
C
496 lines
14 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim@fluendo.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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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 "gstrtpmp4vpay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpmp4vpay_debug);
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#define GST_CAT_DEFAULT (rtpmp4vpay_debug)
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/* elementfactory information */
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static const GstElementDetails gst_rtp_mp4vpay_details =
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GST_ELEMENT_DETAILS ("RTP packet payloader",
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"Codec/Payloader/Network",
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"Payload MPEG4 video as RTP packets (RFC 3016)",
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"Wim Taymans <wim@fluendo.com>");
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static GstStaticPadTemplate gst_rtp_mp4v_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 ("video/mpeg,"
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"mpegversion=(int) 4," "systemstream=(boolean)false")
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);
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static GstStaticPadTemplate gst_rtp_mp4v_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) \"video\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP4V-ES\""
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/* two string params
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*
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"profile-level-id = (string) [1,MAX]"
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"config = (string) [1,MAX]"
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*/
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)
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);
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#define DEFAULT_SEND_CONFIG FALSE
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enum
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{
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ARG_0,
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ARG_SEND_CONFIG
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};
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static void gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass);
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static void gst_rtp_mp4v_pay_base_init (GstRtpMP4VPayClass * klass);
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static void gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay);
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static void gst_rtp_mp4v_pay_finalize (GObject * object);
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static void gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_rtp_mp4v_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload *
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payload, GstBuffer * buffer);
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static GstBaseRTPPayloadClass *parent_class = NULL;
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static GType
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gst_rtp_mp4v_pay_get_type (void)
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{
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static GType rtpmp4vpay_type = 0;
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if (!rtpmp4vpay_type) {
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static const GTypeInfo rtpmp4vpay_info = {
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sizeof (GstRtpMP4VPayClass),
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(GBaseInitFunc) gst_rtp_mp4v_pay_base_init,
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NULL,
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(GClassInitFunc) gst_rtp_mp4v_pay_class_init,
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NULL,
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NULL,
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sizeof (GstRtpMP4VPay),
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0,
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(GInstanceInitFunc) gst_rtp_mp4v_pay_init,
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};
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rtpmp4vpay_type =
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g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpMP4VPay",
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&rtpmp4vpay_info, 0);
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}
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return rtpmp4vpay_type;
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}
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static void
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gst_rtp_mp4v_pay_base_init (GstRtpMP4VPayClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_mp4v_pay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_mp4v_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_mp4vpay_details);
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}
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static void
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gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->set_property = gst_rtp_mp4v_pay_set_property;
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gobject_class->get_property = gst_rtp_mp4v_pay_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SEND_CONFIG,
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g_param_spec_boolean ("send-config", "Send Config",
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"Send the config parameters in RTP packets as well",
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DEFAULT_SEND_CONFIG, G_PARAM_READWRITE));
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gobject_class->finalize = gst_rtp_mp4v_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_mp4v_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_mp4v_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpmp4vpay_debug, "rtpmp4vpay", 0,
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"MP4 video RTP Payloader");
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}
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static void
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gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay)
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{
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rtpmp4vpay->adapter = gst_adapter_new ();
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rtpmp4vpay->rate = 90000;
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rtpmp4vpay->profile = 1;
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rtpmp4vpay->send_config = DEFAULT_SEND_CONFIG;
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}
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static void
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gst_rtp_mp4v_pay_finalize (GObject * object)
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{
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GstRtpMP4VPay *rtpmp4vpay;
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rtpmp4vpay = GST_RTP_MP4V_PAY (object);
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g_object_unref (rtpmp4vpay->adapter);
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rtpmp4vpay->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_mp4v_pay_new_caps (GstRtpMP4VPay * rtpmp4vpay)
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{
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gchar *profile, *config;
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GValue v = { 0 };
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profile = g_strdup_printf ("%d", rtpmp4vpay->profile);
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g_value_init (&v, GST_TYPE_BUFFER);
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gst_value_set_buffer (&v, rtpmp4vpay->config);
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config = gst_value_serialize (&v);
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gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4vpay),
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"profile-level-id", G_TYPE_STRING, profile,
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"config", G_TYPE_STRING, config, NULL);
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g_value_unset (&v);
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g_free (profile);
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g_free (config);
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}
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static gboolean
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gst_rtp_mp4v_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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GstRtpMP4VPay *rtpmp4vpay;
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GstStructure *structure;
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const GValue *codec_data;
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rtpmp4vpay = GST_RTP_MP4V_PAY (payload);
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gst_basertppayload_set_options (payload, "video", TRUE, "MP4V-ES",
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rtpmp4vpay->rate);
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structure = gst_caps_get_structure (caps, 0);
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codec_data = gst_structure_get_value (structure, "codec_data");
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if (codec_data) {
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GST_LOG_OBJECT (rtpmp4vpay, "got codec_data");
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if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) {
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GstBuffer *buffer;
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guint8 *data;
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guint size;
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buffer = gst_value_get_buffer (codec_data);
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data = GST_BUFFER_DATA (buffer);
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size = GST_BUFFER_SIZE (buffer);
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if (size < 5)
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goto done;
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rtpmp4vpay->profile = data[4];
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GST_LOG_OBJECT (rtpmp4vpay, "configuring codec_data, profile %d",
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data[4]);
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if (rtpmp4vpay->config)
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gst_buffer_unref (rtpmp4vpay->config);
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rtpmp4vpay->config = gst_buffer_copy (buffer);
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gst_rtp_mp4v_pay_new_caps (rtpmp4vpay);
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}
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}
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done:
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return TRUE;
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}
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static GstFlowReturn
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gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay)
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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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/* 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 MP4V data
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* over multiple packets. */
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avail = gst_adapter_available (rtpmp4vpay->adapter);
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ret = GST_FLOW_OK;
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while (avail > 0) {
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guint towrite;
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guint8 *payload;
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guint8 *data;
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guint payload_len;
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guint packet_len;
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/* this will be the total lenght of the packet */
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packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
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/* fill one MTU or all available bytes */
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towrite = MIN (packet_len, GST_BASE_RTP_PAYLOAD_MTU (rtpmp4vpay));
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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 (payload_len, 0, 0);
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/* copy payload */
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payload = gst_rtp_buffer_get_payload (outbuf);
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data = (guint8 *) gst_adapter_peek (rtpmp4vpay->adapter, payload_len);
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memcpy (payload, data, payload_len);
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gst_adapter_flush (rtpmp4vpay->adapter, payload_len);
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avail -= payload_len;
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gst_rtp_buffer_set_marker (outbuf, avail == 0);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4vpay->first_ts;
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4vpay), outbuf);
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}
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return ret;
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}
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#define VOS_STARTCODE 0x000001B0
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#define VOS_ENDCODE 0x000001B1
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#define USER_DATA_STARTCODE 0x000001B2
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#define GOP_STARTCODE 0x000001B3
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#define VISUAL_OBJECT_STARTCODE 0x000001B5
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#define VOP_STARTCODE 0x000001B6
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static gboolean
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gst_rtp_mp4v_pay_depay_data (GstRtpMP4VPay * enc, guint8 * data, guint size,
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gint * strip)
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{
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guint32 code;
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gboolean result;
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*strip = 0;
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if (size < 5)
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return FALSE;
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code = GST_READ_UINT32_BE (data);
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GST_DEBUG_OBJECT (enc, "start code 0x%08x", code);
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switch (code) {
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case VOS_STARTCODE:
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{
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gint i;
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guint8 profile;
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gboolean newprofile = FALSE;
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gboolean equal;
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/* profile_and_level_indication */
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profile = data[4];
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GST_DEBUG_OBJECT (enc, "VOS profile 0x%08x", profile);
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if (profile != enc->profile) {
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newprofile = TRUE;
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enc->profile = profile;
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}
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/* up to the next GOP_STARTCODE or VOP_STARTCODE is
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* the config information */
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code = 0xffffffff;
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for (i = 5; i < size - 4; i++) {
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code = (code << 8) | data[i];
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if (code == GOP_STARTCODE || code == VOP_STARTCODE)
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break;
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}
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i -= 3;
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/* see if config changed */
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equal = FALSE;
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if (enc->config) {
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if (GST_BUFFER_SIZE (enc->config) == i) {
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equal = memcmp (GST_BUFFER_DATA (enc->config), data, i) == 0;
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}
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}
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/* if config string changed or new profile, make new caps */
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if (!equal || newprofile) {
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if (enc->config)
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gst_buffer_unref (enc->config);
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enc->config = gst_buffer_new_and_alloc (i);
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memcpy (GST_BUFFER_DATA (enc->config), data, i);
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gst_rtp_mp4v_pay_new_caps (enc);
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}
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*strip = i;
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/* we need to flush out the current packet. */
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result = TRUE;
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break;
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}
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case VOP_STARTCODE:
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GST_DEBUG_OBJECT (enc, "VOP");
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/* VOP startcode, we don't have to flush the packet */
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result = FALSE;
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break;
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default:
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GST_DEBUG_OBJECT (enc, "other startcode");
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/* all other startcodes need a flush */
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result = TRUE;
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break;
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}
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return result;
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}
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/* we expect buffers starting on startcodes.
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*/
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static GstFlowReturn
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gst_rtp_mp4v_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpMP4VPay *rtpmp4vpay;
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GstFlowReturn ret;
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guint size, avail;
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guint packet_len;
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guint8 *data;
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gboolean flush;
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gint strip;
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GstClockTime duration;
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ret = GST_FLOW_OK;
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rtpmp4vpay = GST_RTP_MP4V_PAY (basepayload);
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size = GST_BUFFER_SIZE (buffer);
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data = GST_BUFFER_DATA (buffer);
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duration = GST_BUFFER_DURATION (buffer);
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avail = gst_adapter_available (rtpmp4vpay->adapter);
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/* empty buffer, take timestamp */
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if (avail == 0) {
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rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtpmp4vpay->duration = 0;
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}
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/* depay incomming data and see if we need to start a new RTP
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* packet */
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flush = gst_rtp_mp4v_pay_depay_data (rtpmp4vpay, data, size, &strip);
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if (strip) {
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/* strip off config if requested */
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if (!rtpmp4vpay->send_config) {
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GstBuffer *subbuf;
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/* strip off header */
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subbuf = gst_buffer_create_sub (buffer, strip, size - strip);
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GST_BUFFER_TIMESTAMP (subbuf) = GST_BUFFER_TIMESTAMP (buffer);
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gst_buffer_unref (buffer);
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buffer = subbuf;
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size = GST_BUFFER_SIZE (buffer);
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data = GST_BUFFER_DATA (buffer);
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}
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}
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/* if we need to flush, do so now */
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if (flush) {
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ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
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rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtpmp4vpay->duration = 0;
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avail = 0;
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}
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/* get packet length of data and see if we exceeded MTU. */
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packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
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if (gst_basertppayload_is_filled (basepayload,
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packet_len, rtpmp4vpay->duration + duration)) {
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ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay);
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rtpmp4vpay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
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rtpmp4vpay->duration = 0;
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}
|
|
|
|
/* push new data */
|
|
gst_adapter_push (rtpmp4vpay->adapter, buffer);
|
|
rtpmp4vpay->duration += duration;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpMP4VPay *rtpmp4vpay;
|
|
|
|
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_SEND_CONFIG:
|
|
rtpmp4vpay->send_config = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpMP4VPay *rtpmp4vpay;
|
|
|
|
rtpmp4vpay = GST_RTP_MP4V_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
case ARG_SEND_CONFIG:
|
|
g_value_set_boolean (value, rtpmp4vpay->send_config);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_mp4v_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpmp4vpay",
|
|
GST_RANK_NONE, GST_TYPE_RTP_MP4V_PAY);
|
|
}
|