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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Use gst_rtp_base_payload_allocate_output_buffer() instead of gst_rtp_buffer_new_allocate() in order to allocate RTP buffer with correct number of CSRCs according to the meta. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
401 lines
12 KiB
C
401 lines
12 KiB
C
/* Farsight
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* Copyright (C) 2006 Marcel Moreaux <marcelm@spacelabs.nl>
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* (C) 2008 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 <stdlib.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpdvpay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY (rtpdvpay_debug);
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#define GST_CAT_DEFAULT (rtpdvpay_debug)
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#define DEFAULT_MODE GST_DV_PAY_MODE_VIDEO
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enum
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{
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PROP_0,
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PROP_MODE
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};
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/* takes both system and non-system streams */
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static GstStaticPadTemplate gst_rtp_dv_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/x-dv")
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);
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static GstStaticPadTemplate gst_rtp_dv_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\", \"audio\" } ,"
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"encoding-name = (string) \"DV\", "
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"clock-rate = (int) 90000,"
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"encode = (string) { \"SD-VCR/525-60\", \"SD-VCR/625-50\", \"HD-VCR/1125-60\","
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"\"HD-VCR/1250-50\", \"SDL-VCR/525-60\", \"SDL-VCR/625-50\","
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"\"306M/525-60\", \"306M/625-50\", \"314M-25/525-60\","
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"\"314M-25/625-50\", \"314M-50/525-60\", \"314M-50/625-50\" }"
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/* optional parameters can't go in the template
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* "audio = (string) { \"bundled\", \"none\" }"
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*/
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)
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);
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static gboolean gst_rtp_dv_pay_setcaps (GstRTPBasePayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_dv_pay_handle_buffer (GstRTPBasePayload * payload,
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GstBuffer * buffer);
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#define GST_TYPE_DV_PAY_MODE (gst_dv_pay_mode_get_type())
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static GType
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gst_dv_pay_mode_get_type (void)
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{
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static GType dv_pay_mode_type = 0;
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static const GEnumValue dv_pay_modes[] = {
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{GST_DV_PAY_MODE_VIDEO, "Video only", "video"},
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{GST_DV_PAY_MODE_BUNDLED, "Video and Audio bundled", "bundled"},
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{GST_DV_PAY_MODE_AUDIO, "Audio only", "audio"},
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{0, NULL, NULL},
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};
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if (!dv_pay_mode_type) {
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dv_pay_mode_type = g_enum_register_static ("GstDVPayMode", dv_pay_modes);
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}
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return dv_pay_mode_type;
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}
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static void gst_dv_pay_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_dv_pay_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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#define gst_rtp_dv_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRTPDVPay, gst_rtp_dv_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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static void
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gst_rtp_dv_pay_class_init (GstRTPDVPayClass * 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 (rtpdvpay_debug, "rtpdvpay", 0, "DV RTP Payloader");
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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->set_property = gst_dv_pay_set_property;
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gobject_class->get_property = gst_dv_pay_get_property;
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g_object_class_install_property (gobject_class, PROP_MODE,
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g_param_spec_enum ("mode", "Mode",
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"The payload mode of payloading",
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GST_TYPE_DV_PAY_MODE, DEFAULT_MODE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_dv_pay_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_dv_pay_src_template);
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gst_element_class_set_static_metadata (gstelement_class, "RTP DV Payloader",
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"Codec/Payloader/Network/RTP",
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"Payloads DV into RTP packets (RFC 3189)",
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"Marcel Moreaux <marcelm@spacelabs.nl>, Wim Taymans <wim.taymans@gmail.com>");
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gstrtpbasepayload_class->set_caps = gst_rtp_dv_pay_setcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_dv_pay_handle_buffer;
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gst_type_mark_as_plugin_api (GST_TYPE_DV_PAY_MODE, 0);
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}
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static void
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gst_rtp_dv_pay_init (GstRTPDVPay * rtpdvpay)
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{
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}
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static void
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gst_dv_pay_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec)
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{
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GstRTPDVPay *rtpdvpay = GST_RTP_DV_PAY (object);
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switch (prop_id) {
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case PROP_MODE:
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rtpdvpay->mode = g_value_get_enum (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_dv_pay_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec)
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{
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GstRTPDVPay *rtpdvpay = GST_RTP_DV_PAY (object);
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switch (prop_id) {
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case PROP_MODE:
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g_value_set_enum (value, rtpdvpay->mode);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static gboolean
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gst_rtp_dv_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
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{
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/* We don't do anything here, but we could check if it's a system stream and if
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* it's not, default to sending the video only. We will negotiate downstream
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* caps when we get to see the first frame. */
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return TRUE;
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}
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static gboolean
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gst_dv_pay_negotiate (GstRTPDVPay * rtpdvpay, guint8 * data, gsize size)
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{
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const gchar *encode, *media;
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gboolean audio_bundled, res;
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if ((data[3] & 0x80) == 0) { /* DSF flag */
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/* it's an NTSC format */
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if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
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/* NTSC 50Mbps */
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encode = "314M-25/525-60";
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} else { /* 4:1:1 sampling */
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/* NTSC 25Mbps */
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encode = "SD-VCR/525-60";
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}
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} else {
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/* it's a PAL format */
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if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
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/* PAL 50Mbps */
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encode = "314M-50/625-50";
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} else if ((data[5] & 0x07) == 0) { /* APT flag */
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/* PAL 25Mbps 4:2:0 */
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encode = "SD-VCR/625-50";
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} else
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/* PAL 25Mbps 4:1:1 */
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encode = "314M-25/625-50";
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}
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media = "video";
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audio_bundled = FALSE;
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switch (rtpdvpay->mode) {
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case GST_DV_PAY_MODE_AUDIO:
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media = "audio";
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break;
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case GST_DV_PAY_MODE_BUNDLED:
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audio_bundled = TRUE;
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break;
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default:
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break;
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}
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gst_rtp_base_payload_set_options (GST_RTP_BASE_PAYLOAD (rtpdvpay), media,
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TRUE, "DV", 90000);
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if (audio_bundled) {
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res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpdvpay),
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"encode", G_TYPE_STRING, encode,
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"audio", G_TYPE_STRING, "bundled", NULL);
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} else {
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res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpdvpay),
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"encode", G_TYPE_STRING, encode, NULL);
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}
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return res;
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}
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static gboolean
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include_dif (GstRTPDVPay * rtpdvpay, guint8 * data)
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{
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gint block_type;
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gboolean res;
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block_type = data[0] >> 5;
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switch (block_type) {
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case 0: /* Header block */
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case 1: /* Subcode block */
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case 2: /* VAUX block */
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/* always include these blocks */
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res = TRUE;
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break;
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case 3: /* Audio block */
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/* never include audio if we are doing video only */
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if (rtpdvpay->mode == GST_DV_PAY_MODE_VIDEO)
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res = FALSE;
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else
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res = TRUE;
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break;
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case 4: /* Video block */
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/* never include video if we are doing audio only */
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if (rtpdvpay->mode == GST_DV_PAY_MODE_AUDIO)
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res = FALSE;
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else
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res = TRUE;
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break;
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default: /* Something bogus, just ignore */
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res = FALSE;
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break;
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}
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return res;
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}
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/* Get a DV frame, chop it up in pieces, and push the pieces to the RTP layer.
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*/
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static GstFlowReturn
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gst_rtp_dv_pay_handle_buffer (GstRTPBasePayload * basepayload,
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GstBuffer * buffer)
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{
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GstRTPDVPay *rtpdvpay;
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guint max_payload_size;
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GstBuffer *outbuf;
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GstFlowReturn ret = GST_FLOW_OK;
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gint hdrlen;
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gsize size;
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GstMapInfo map;
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guint8 *data;
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guint8 *dest;
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guint filled;
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GstRTPBuffer rtp = { NULL, };
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rtpdvpay = GST_RTP_DV_PAY (basepayload);
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hdrlen = gst_rtp_buffer_calc_header_len (0);
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/* DV frames are made up from a bunch of DIF blocks. DIF blocks are 80 bytes
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* each, and we should put an integral number of them in each RTP packet.
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* Therefore, we round the available room down to the nearest multiple of 80.
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*
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* The available room is just the packet MTU, minus the RTP header length. */
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max_payload_size = ((GST_RTP_BASE_PAYLOAD_MTU (rtpdvpay) - hdrlen) / 80) * 80;
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/* The length of the buffer to transmit. */
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if (!gst_buffer_map (buffer, &map, GST_MAP_READ)) {
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GST_ELEMENT_ERROR (rtpdvpay, CORE, FAILED,
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(NULL), ("Failed to map buffer"));
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gst_buffer_unref (buffer);
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return GST_FLOW_ERROR;
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}
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data = map.data;
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size = map.size;
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GST_DEBUG_OBJECT (rtpdvpay,
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"DV RTP payloader got buffer of %" G_GSIZE_FORMAT
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" bytes, splitting in %u byte " "payload fragments, at time %"
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GST_TIME_FORMAT, size, max_payload_size,
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GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
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if (!rtpdvpay->negotiated) {
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gst_dv_pay_negotiate (rtpdvpay, data, size);
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/* if we have not yet scanned the stream for its type, do so now */
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rtpdvpay->negotiated = TRUE;
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}
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outbuf = NULL;
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dest = NULL;
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filled = 0;
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/* while we have a complete DIF chunks left */
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while (size >= 80) {
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/* Allocate a new buffer, set the timestamp */
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if (outbuf == NULL) {
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outbuf =
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gst_rtp_base_payload_allocate_output_buffer (basepayload,
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max_payload_size, 0, 0);
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GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buffer);
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if (!gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp)) {
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gst_buffer_unref (outbuf);
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GST_ELEMENT_ERROR (rtpdvpay, CORE, FAILED,
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(NULL), ("Failed to map RTP buffer"));
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ret = GST_FLOW_ERROR;
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goto beach;
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}
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dest = gst_rtp_buffer_get_payload (&rtp);
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filled = 0;
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}
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/* inspect the DIF chunk, if we don't need to include it, skip to the next one. */
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if (include_dif (rtpdvpay, data)) {
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/* copy data in packet */
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memcpy (dest, data, 80);
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dest += 80;
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filled += 80;
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}
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/* go to next dif chunk */
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size -= 80;
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data += 80;
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/* push out the buffer if the next one would exceed the max packet size or
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* when we are pushing the last packet */
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if (filled + 80 > max_payload_size || size < 80) {
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if (size < 160) {
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guint hlen;
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/* set marker */
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gst_rtp_buffer_set_marker (&rtp, TRUE);
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/* shrink buffer to last packet */
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hlen = gst_rtp_buffer_get_header_len (&rtp);
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gst_rtp_buffer_set_packet_len (&rtp, hlen + filled);
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}
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/* Push out the created piece, and check for errors. */
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gst_rtp_buffer_unmap (&rtp);
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gst_rtp_copy_meta (GST_ELEMENT_CAST (basepayload), outbuf, buffer, 0);
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ret = gst_rtp_base_payload_push (basepayload, outbuf);
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if (ret != GST_FLOW_OK)
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break;
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outbuf = NULL;
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}
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}
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beach:
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gst_buffer_unmap (buffer, &map);
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gst_buffer_unref (buffer);
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return ret;
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}
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gboolean
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gst_rtp_dv_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpdvpay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_DV_PAY);
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}
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