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72f0385606
Original commit message from CVS: 2008-04-07 Julien Moutte <julien@fluendo.com> * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps): Fix build because of a bad argument number.
706 lines
19 KiB
C
706 lines
19 KiB
C
/* GStreamer
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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 "gstrtph264pay.h"
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#define SPS_TYPE_ID 7
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#define PPS_TYPE_ID 8
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#define USE_MEMCMP
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GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
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#define GST_CAT_DEFAULT (rtph264pay_debug)
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/* references:
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*
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* RFC 3984
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*/
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/* elementfactory information */
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static const GstElementDetails gst_rtp_h264pay_details =
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GST_ELEMENT_DETAILS ("RTP packet payloader",
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"Codec/Payloader/Network",
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"Payload-encode H264 video into RTP packets (RFC 3984)",
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"Laurent Glayal <spglegle@yahoo.fr>");
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static GstStaticPadTemplate gst_rtp_h264_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-h264")
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);
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static GstStaticPadTemplate gst_rtp_h264_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) 90000, " "encoding-name = (string) \"H264\"")
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);
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static void gst_rtp_h264_pay_finalize (GObject * object);
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static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * pad,
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GstBuffer * buffer);
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GST_BOILERPLATE (GstRtpH264Pay, gst_rtp_h264_pay, GstBaseRTPPayload,
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GST_TYPE_BASE_RTP_PAYLOAD);
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static void
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gst_rtp_h264_pay_base_init (gpointer 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_h264_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_h264_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_h264pay_details);
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}
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static void
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gst_rtp_h264_pay_class_init (GstRtpH264PayClass * 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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gobject_class->finalize = gst_rtp_h264_pay_finalize;
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gstelement_class->change_state = gst_rtp_h264_pay_change_state;
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gstbasertppayload_class->set_caps = gst_rtp_h264_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0,
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"H264 RTP Payloader");
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}
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static void
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gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay, GstRtpH264PayClass * klass)
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{
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rtph264pay->profile = 0;
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rtph264pay->sps = NULL;
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rtph264pay->pps = NULL;
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}
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static void
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gst_rtp_h264_pay_finalize (GObject * object)
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{
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GstRtpH264Pay *rtph264pay;
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rtph264pay = GST_RTP_H264_PAY (object);
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if (rtph264pay->sps)
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g_free (rtph264pay->sps);
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if (rtph264pay->pps)
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g_free (rtph264pay->pps);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gchar *
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encode_base64 (const guint8 * in, guint size, guint * len)
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{
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gchar *ret, *d;
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static const gchar *v =
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"ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
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*len = ((size + 2) / 3) * 4;
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d = ret = (gchar *) g_malloc (*len + 1);
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for (; size; in += 3) { /* process tuplets */
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*d++ = v[in[0] >> 2]; /* byte 1: high 6 bits (1) */
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/* byte 2: low 2 bits (1), high 4 bits (2) */
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*d++ = v[((in[0] << 4) + (--size ? (in[1] >> 4) : 0)) & 0x3f];
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/* byte 3: low 4 bits (2), high 2 bits (3) */
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*d++ = size ? v[((in[1] << 2) + (--size ? (in[2] >> 6) : 0)) & 0x3f] : '=';
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/* byte 4: low 6 bits (3) */
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*d++ = size ? v[in[2] & 0x3f] : '=';
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if (size)
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size--; /* count third character if processed */
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}
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*d = '\0'; /* tie off string */
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return ret; /* return the resulting string */
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}
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static gboolean
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gst_rtp_h264_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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{
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GstRtpH264Pay *rtph264pay;
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GstStructure *str;
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const GValue *value;
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guint8 *data;
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guint size;
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rtph264pay = GST_RTP_H264_PAY (basepayload);
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str = gst_caps_get_structure (caps, 0);
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/* we can only set the output caps when we found the sprops and profile
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* NALs */
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gst_basertppayload_set_options (basepayload, "video", TRUE, "H264", 90000);
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/* packetized AVC video has a codec_data */
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if ((value = gst_structure_get_value (str, "codec_data"))) {
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GstBuffer *buffer;
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GString *sprops;
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guint num_sps, num_pps;
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gint i, count, nal_size;
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gint profile;
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gchar *profile_str;
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GST_DEBUG_OBJECT (rtph264pay, "have packetized h264");
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rtph264pay->packetized = TRUE;
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buffer = gst_value_get_buffer (value);
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data = GST_BUFFER_DATA (buffer);
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size = GST_BUFFER_SIZE (buffer);
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/* parse the avcC data */
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if (size < 7)
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goto avcc_too_small;
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/* parse the version, this must be 1 */
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if (data[0] != 1)
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goto wrong_version;
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/* AVCProfileIndication */
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/* profile_compat */
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/* AVCLevelIndication */
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profile = (data[1] << 16) | (data[2] << 8) | data[3];
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GST_DEBUG_OBJECT (rtph264pay, "profile %06x", profile);
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/* 6 bits reserved | 2 bits lengthSizeMinusOne */
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rtph264pay->nal_length_size = (data[4] & 0x03) + 1;
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GST_DEBUG_OBJECT (rtph264pay, "nal length %u", rtph264pay->nal_length_size);
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/* 3 bits reserved | 5 bits numOfSequenceParameterSets */
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num_sps = data[5] & 0x1f;
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GST_DEBUG_OBJECT (rtph264pay, "num SPS %u", num_sps);
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data += 6;
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size -= 6;
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/* create the sprop-parameter-sets */
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sprops = g_string_new ("");
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count = 0;
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for (i = 0; i < num_sps; i++) {
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gchar *set;
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guint len;
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if (size < 2)
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goto avcc_error;
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nal_size = (data[0] << 8) | data[1];
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data += 2;
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size -= 2;
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if (size < nal_size)
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goto avcc_error;
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set = encode_base64 (data, nal_size, &len);
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g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
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count++;
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g_free (set);
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data += nal_size;
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size -= nal_size;
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}
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if (size < 1)
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goto avcc_error;
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num_pps = data[0];
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data += 1;
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size -= 1;
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GST_DEBUG_OBJECT (rtph264pay, "num PPS %u", num_pps);
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for (i = 0; i < num_pps; i++) {
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gchar *set;
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guint len;
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if (size < 2)
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goto avcc_error;
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nal_size = (data[0] << 8) | data[1];
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data += 2;
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size -= 2;
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if (size < nal_size)
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goto avcc_error;
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set = encode_base64 (data, nal_size, &len);
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g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
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count++;
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g_free (set);
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data += nal_size;
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size -= nal_size;
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}
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GST_DEBUG_OBJECT (rtph264pay, "sprops %s", sprops->str);
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profile_str = g_strdup_printf ("%06x", profile);
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gst_basertppayload_set_outcaps (basepayload, "profile-level-id",
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G_TYPE_STRING, profile_str,
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"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
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g_free (profile_str);
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g_string_free (sprops, TRUE);
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} else {
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GST_DEBUG_OBJECT (rtph264pay, "have bytestream h264");
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rtph264pay->packetized = FALSE;
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}
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return TRUE;
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avcc_too_small:
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{
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GST_ERROR_OBJECT (rtph264pay, "avcC size %u < 7", size);
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return FALSE;
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}
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wrong_version:
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{
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GST_ERROR_OBJECT (rtph264pay, "wrong avcC version");
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return FALSE;
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}
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avcc_error:
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{
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GST_ERROR_OBJECT (rtph264pay, "avcC too small ");
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return FALSE;
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}
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}
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static guint
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next_start_code (guint8 * data, guint size)
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{
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/* Boyer-Moore string matching algorithm, in a degenerative
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* sense because our search 'alphabet' is binary - 0 & 1 only.
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* This allow us to simplify the general BM algorithm to a very
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* simple form. */
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/* assume 1 is in the 4th byte */
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guint offset = 3;
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while (offset < size) {
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if (1 == data[offset]) {
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unsigned int shift = offset;
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if (0 == data[--shift]) {
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if (0 == data[--shift]) {
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if (0 == data[--shift]) {
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return shift;
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}
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}
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}
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/* The jump is always 4 because of the 1 previously matched.
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* All the 0's must be after this '1' matched at offset */
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offset += 4;
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} else if (0 == data[offset]) {
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/* maybe next byte is 1? */
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offset++;
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} else {
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/* can jump 4 bytes forward */
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offset += 4;
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}
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/* at each iteration, we rescan in a backward manner until
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* we match 0.0.0.1 in reverse order. Since our search string
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* has only 2 'alpabets' (i.e. 0 & 1), we know that any
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* mismatch will force us to shift a fixed number of steps */
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}
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GST_DEBUG ("Cannot find next NAL start code. returning %u", size);
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return size;
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}
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/* we don't use memcpy but this faster version (around 20%) because we need to
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* perform it on all data. */
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static gboolean
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is_nal_equal (const guint8 * nal1, const guint8 * nal2, guint len)
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{
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/* if we have a 64-bit processor, we may use guint64 to make
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* this go faster. Otherwise with 32 bits, we are already
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* going faster than byte to byte compare.
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*/
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guint remainder = len & 0x3;
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guint num_int = len >> 2;
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guint32 *pu1 = (guint32 *) nal1, *pu2 = (guint32 *) nal2;
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guint i;
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/* compare by groups of sizeof(guint32) bytes */
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for (i = 0; i < num_int; i++) {
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/* XOR is faster than CMP?... */
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if (pu1[i] ^ pu2[i])
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return FALSE;
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}
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/* check that the remaining bytes are still equal */
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if (!remainder) {
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return TRUE;
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} else if (1 == remainder) {
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return (nal1[--len] == nal2[len]);
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} else { /* 2 or 3 */
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if (remainder & 1) { /* -1 if 3 bytes left */
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if (nal1[--len] != nal2[len])
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return FALSE;
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}
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/* last 2 bytes */
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return ((nal1[--len] == nal2[len]) /* -1 */
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&&(nal1[--len] == nal2[len])); /* -2 */
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}
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}
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static void
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gst_rtp_h264_pay_decode_nal (GstRtpH264Pay * payloader,
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guint8 * data, guint size, gboolean * updated)
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{
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guint8 *sps = NULL, *pps = NULL;
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guint sps_len = 0, pps_len = 0;
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/* default is no update */
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*updated = FALSE;
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if (size <= 3) {
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GST_WARNING ("Encoded buffer len %u <= 3", size);
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} else {
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GST_DEBUG ("NAL payload len=%u", size);
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/* loop through all NAL units and save the locations of any
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* SPS / PPS for later processing. Only the last seen SPS
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* or PPS will be considered */
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while (size > 5) {
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guint8 header, type;
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guint len;
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len = next_start_code (data, size);
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header = data[0];
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type = header & 0x1f;
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/* keep sps & pps separately so that we can update either one
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* independently */
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if (SPS_TYPE_ID == type) {
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/* encode the entire SPS NAL in base64 */
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GST_DEBUG ("Found SPS %x %x %x Len=%u\n", (header >> 7),
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(header >> 5) & 3, type, len);
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sps = data;
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sps_len = len;
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} else if (PPS_TYPE_ID == type) {
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/* encoder the entire PPS NAL in base64 */
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GST_DEBUG ("Found PPS %x %x %x Len = %u\n",
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(header >> 7), (header >> 5) & 3, type, len);
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pps = data;
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pps_len = len;
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} else {
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GST_DEBUG ("NAL: %x %x %x Len = %u\n", (header >> 7),
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(header >> 5) & 3, type, len);
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}
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/* end of loop */
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if (len >= size - 4) {
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break;
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}
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/* next NAL start */
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data += len + 4;
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size -= len + 4;
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}
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/* If we encountered an SPS and/or a PPS, check if it's the
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* same as the one we have. If not, update our version and
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* set *updated to TRUE
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*/
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if (sps_len > 0) {
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if ((payloader->sps_len != sps_len)
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|| !is_nal_equal (payloader->sps, sps, sps_len)) {
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payloader->profile = (sps[1] << 16) + (sps[2] << 8) + sps[3];
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GST_DEBUG ("Profile level IDC = %06x", payloader->profile);
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if (payloader->sps_len)
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g_free (payloader->sps);
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payloader->sps = sps_len ? g_new (guint8, sps_len) : NULL;
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memcpy (payloader->sps, sps, sps_len);
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payloader->sps_len = sps_len;
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*updated = TRUE;
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}
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}
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if (pps_len > 0) {
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if ((payloader->pps_len != pps_len)
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|| !is_nal_equal (payloader->pps, pps, pps_len)) {
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if (payloader->pps_len)
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g_free (payloader->pps);
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payloader->pps = pps_len ? g_new (guint8, pps_len) : NULL;
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memcpy (payloader->pps, pps, pps_len);
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payloader->pps_len = pps_len;
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*updated = TRUE;
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}
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}
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}
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}
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static void
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gst_rtp_h264_pay_parse_sps_pps (GstBaseRTPPayload * basepayload,
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guint8 * data, guint size)
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{
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gboolean update = FALSE;
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GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
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gst_rtp_h264_pay_decode_nal (payloader, data, size, &update);
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|
|
/* if has new SPS & PPS, update the output caps */
|
|
if (update) {
|
|
gchar *profile;
|
|
gchar *sps;
|
|
gchar *pps;
|
|
gchar *sprops;
|
|
guint len;
|
|
|
|
/* profile is 24 bit. Force it to respect the limit */
|
|
profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
|
|
|
|
/* build the sprop-parameter-sets */
|
|
sps = (payloader->sps_len > 0)
|
|
? encode_base64 (payloader->sps, payloader->sps_len, &len) : NULL;
|
|
pps = (payloader->pps_len > 0)
|
|
? encode_base64 (payloader->pps, payloader->pps_len, &len) : NULL;
|
|
|
|
if (sps)
|
|
sprops = g_strjoin (",", sps, pps, NULL);
|
|
else
|
|
sprops = g_strdup (pps);
|
|
|
|
gst_basertppayload_set_outcaps (basepayload, "profile-level-id",
|
|
G_TYPE_STRING, profile,
|
|
"sprop-parameter-sets", G_TYPE_STRING, sprops, NULL);
|
|
|
|
GST_DEBUG ("outcaps udpate: profile=%s, sps=%s, pps=%s\n",
|
|
profile, sps, pps);
|
|
|
|
g_free (sprops);
|
|
g_free (profile);
|
|
g_free (sps);
|
|
g_free (pps);
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
GstFlowReturn ret;
|
|
guint size, idxdata;
|
|
GstBuffer *outbuf;
|
|
guint8 *payload, *data, *pdata;
|
|
guint8 nalType;
|
|
GstClockTime timestamp;
|
|
guint packet_len, payload_len, mtu;
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtph264pay);
|
|
|
|
size = GST_BUFFER_SIZE (buffer);
|
|
data = GST_BUFFER_DATA (buffer);
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
|
|
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
|
|
|
|
/* we don't support AVC input yet */
|
|
if (rtph264pay->packetized)
|
|
goto not_supported;
|
|
|
|
/* H264 stream analysis */
|
|
pdata = data;
|
|
|
|
/* use next_start_code() to scan buffer.
|
|
* next_start_code() returns the offset in data,
|
|
* starting from zero to the first byte of 0.0.0.1
|
|
* If no start code is found, it returns the value of the
|
|
* 'size' parameter.
|
|
* pdata is unchanged by the call to next_start_code()
|
|
*/
|
|
{
|
|
guint offset = next_start_code (pdata, size);
|
|
|
|
pdata += offset;
|
|
idxdata = size - offset;
|
|
}
|
|
|
|
if (idxdata < 5) {
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"Returning GST_FLOW_OK without creating RTP packet");
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
pdata += 4;
|
|
idxdata -= 4;
|
|
|
|
/* We know our stream is a valid H264 NAL packet,
|
|
* go parse it for SPS/PPS to enrich the caps */
|
|
gst_rtp_h264_pay_parse_sps_pps (basepayload, pdata, idxdata);
|
|
|
|
nalType = pdata[0] & 0x1f;
|
|
GST_DEBUG_OBJECT (basepayload, "Processing Buffer with NAL TYPE=%d", nalType);
|
|
|
|
packet_len = gst_rtp_buffer_calc_packet_len (idxdata, 0, 0);
|
|
|
|
if (packet_len < mtu) {
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"NAL Unit fit in one packet datasize=%d mtu=%d", idxdata, mtu);
|
|
/* will fit in one packet */
|
|
outbuf = gst_rtp_buffer_new_allocate (idxdata, 0, 0);
|
|
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
|
|
gst_rtp_buffer_set_marker (outbuf, 1);
|
|
|
|
payload = gst_rtp_buffer_get_payload (outbuf);
|
|
GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", idxdata);
|
|
memcpy (payload, pdata, idxdata);
|
|
gst_buffer_unref (buffer);
|
|
ret = gst_basertppayload_push (basepayload, outbuf);
|
|
return ret;
|
|
} else {
|
|
/* Fragmentation Units FU-A */
|
|
guint8 nalHeader;
|
|
guint limitedSize;
|
|
int ii = 0, start = 1, end = 0, first = 0;
|
|
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", idxdata, mtu);
|
|
|
|
nalHeader = *pdata;
|
|
pdata++;
|
|
idxdata--;
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d",
|
|
idxdata);
|
|
|
|
/* We keep 2 bytes for FU indicator and FU Header */
|
|
payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
|
|
|
|
while (end == 0) {
|
|
limitedSize = idxdata < payload_len ? idxdata : payload_len;
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize,
|
|
ii);
|
|
|
|
outbuf = gst_rtp_buffer_new_allocate (limitedSize + 2, 0, 0);
|
|
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
|
|
payload = gst_rtp_buffer_get_payload (outbuf);
|
|
|
|
if (limitedSize == idxdata) {
|
|
GST_DEBUG_OBJECT (basepayload, "end idxdata=%d iteration=%d", idxdata,
|
|
ii);
|
|
end = 1;
|
|
}
|
|
gst_rtp_buffer_set_marker (outbuf, end);
|
|
|
|
/* FU indicator */
|
|
payload[0] = (nalHeader & 0x60) | 28;
|
|
|
|
/* FU Header */
|
|
payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f);
|
|
|
|
memcpy (&payload[2], pdata + first, limitedSize);
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"recorded %d payload bytes into packet iteration=%d", limitedSize + 2,
|
|
ii);
|
|
|
|
ret = gst_basertppayload_push (basepayload, outbuf);
|
|
if (ret != GST_FLOW_OK)
|
|
break;
|
|
|
|
idxdata -= limitedSize;
|
|
first += limitedSize;
|
|
ii++;
|
|
start = 0;
|
|
}
|
|
|
|
gst_buffer_unref (buffer);
|
|
return ret;
|
|
}
|
|
|
|
/* ERRORS */
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
|
|
(NULL), ("Should not be there !!"));
|
|
gst_buffer_unref (buffer);
|
|
|
|
return GST_FLOW_ERROR;
|
|
|
|
not_supported:
|
|
{
|
|
GST_ELEMENT_ERROR (basepayload, STREAM, FORMAT,
|
|
(NULL), ("AVC H264 is not supported yet"));
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_NOT_SUPPORTED;
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (element);
|
|
|
|
switch (transition) {
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_h264_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtph264pay",
|
|
GST_RANK_NONE, GST_TYPE_RTP_H264_PAY);
|
|
}
|