mirror of
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>
1818 lines
54 KiB
C
1818 lines
54 KiB
C
/* ex: set tabstop=2 shiftwidth=2 expandtab: */
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/* 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <stdlib.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/pbutils/pbutils.h>
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#include <gst/video/video.h>
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/* Included to not duplicate gst_rtp_h264_add_sps_pps () */
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#include "gstrtph264depay.h"
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#include "gstrtph264pay.h"
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#include "gstrtputils.h"
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#include "gstbuffermemory.h"
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#define IDR_TYPE_ID 5
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#define SPS_TYPE_ID 7
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#define PPS_TYPE_ID 8
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#define AUD_TYPE_ID 9
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#define STAP_A_TYPE_ID 24
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#define FU_A_TYPE_ID 28
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GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug);
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#define GST_CAT_DEFAULT (rtph264pay_debug)
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#define GST_TYPE_RTP_H264_AGGREGATE_MODE \
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(gst_rtp_h264_aggregate_mode_get_type ())
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static GType
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gst_rtp_h264_aggregate_mode_get_type (void)
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{
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static GType type = 0;
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static const GEnumValue values[] = {
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{GST_RTP_H264_AGGREGATE_NONE, "Do not aggregate NAL units", "none"},
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{GST_RTP_H264_AGGREGATE_ZERO_LATENCY,
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"Aggregate NAL units until a VCL unit is included", "zero-latency"},
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{GST_RTP_H264_AGGREGATE_MAX_STAP,
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"Aggregate all NAL units with the same timestamp (adds one frame of"
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" latency)", "max-stap"},
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{0, NULL, NULL},
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};
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if (!type) {
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type = g_enum_register_static ("GstRtpH264AggregateMode", values);
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}
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return type;
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}
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/* references:
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*
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* RFC 3984
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*/
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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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"stream-format = (string) avc, alignment = (string) au;"
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"video/x-h264, "
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"stream-format = (string) byte-stream, alignment = (string) { nal, au }")
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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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#define DEFAULT_SPROP_PARAMETER_SETS NULL
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#define DEFAULT_CONFIG_INTERVAL 0
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#define DEFAULT_AGGREGATE_MODE GST_RTP_H264_AGGREGATE_NONE
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enum
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{
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PROP_0,
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PROP_SPROP_PARAMETER_SETS,
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PROP_CONFIG_INTERVAL,
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PROP_AGGREGATE_MODE,
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};
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static void gst_rtp_h264_pay_finalize (GObject * object);
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static void gst_rtp_h264_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_h264_pay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstCaps *gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload,
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GstPad * pad, GstCaps * filter);
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static gboolean gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * pad,
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GstBuffer * buffer);
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static gboolean gst_rtp_h264_pay_sink_event (GstRTPBasePayload * payload,
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GstEvent * event);
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static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement *
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element, GstStateChange transition);
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static gboolean gst_rtp_h264_pay_src_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static void gst_rtp_h264_pay_reset_bundle (GstRtpH264Pay * rtph264pay);
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#define gst_rtp_h264_pay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpH264Pay, gst_rtp_h264_pay, GST_TYPE_RTP_BASE_PAYLOAD);
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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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GstRTPBasePayloadClass *gstrtpbasepayload_class;
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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_rtp_h264_pay_set_property;
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gobject_class->get_property = gst_rtp_h264_pay_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_SPROP_PARAMETER_SETS, g_param_spec_string ("sprop-parameter-sets",
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"sprop-parameter-sets",
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"The base64 sprop-parameter-sets to set in out caps (set to NULL to "
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"extract from stream)",
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DEFAULT_SPROP_PARAMETER_SETS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_CONFIG_INTERVAL,
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g_param_spec_int ("config-interval",
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"SPS PPS Send Interval",
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"Send SPS and PPS Insertion Interval in seconds (sprop parameter sets "
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"will be multiplexed in the data stream when detected.) "
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"(0 = disabled, -1 = send with every IDR frame)",
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-1, 3600, DEFAULT_CONFIG_INTERVAL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
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);
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/**
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* GstRtpH264Pay:aggregate-mode
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*
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* Bundle suitable SPS/PPS NAL units into STAP-A aggregate packets.
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*
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* This can potentially reduce RTP packetization overhead but not all
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* RTP implementations handle it correctly.
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*
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* For best compatibility, it is recommended to set this to "none" (the
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* default) for RTSP and for WebRTC to "zero-latency".
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*
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* Since: 1.18
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*/
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g_object_class_install_property (G_OBJECT_CLASS (klass),
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PROP_AGGREGATE_MODE,
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g_param_spec_enum ("aggregate-mode",
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"Attempt to use aggregate packets",
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"Bundle suitable SPS/PPS NAL units into STAP-A "
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"aggregate packets",
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GST_TYPE_RTP_H264_AGGREGATE_MODE,
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DEFAULT_AGGREGATE_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)
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);
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gobject_class->finalize = gst_rtp_h264_pay_finalize;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_h264_pay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_h264_pay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class, "RTP H264 payloader",
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"Codec/Payloader/Network/RTP",
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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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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_h264_pay_change_state);
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gstrtpbasepayload_class->get_caps = gst_rtp_h264_pay_getcaps;
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gstrtpbasepayload_class->set_caps = gst_rtp_h264_pay_setcaps;
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gstrtpbasepayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer;
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gstrtpbasepayload_class->sink_event = gst_rtp_h264_pay_sink_event;
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GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0,
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"H264 RTP Payloader");
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gst_type_mark_as_plugin_api (GST_TYPE_RTP_H264_AGGREGATE_MODE, 0);
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}
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static void
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gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay)
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{
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rtph264pay->queue = g_array_new (FALSE, FALSE, sizeof (guint));
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rtph264pay->profile = 0;
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rtph264pay->sps = g_ptr_array_new_with_free_func (
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(GDestroyNotify) gst_buffer_unref);
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rtph264pay->pps = g_ptr_array_new_with_free_func (
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(GDestroyNotify) gst_buffer_unref);
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rtph264pay->last_spspps = -1;
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rtph264pay->spspps_interval = DEFAULT_CONFIG_INTERVAL;
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rtph264pay->aggregate_mode = DEFAULT_AGGREGATE_MODE;
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rtph264pay->delta_unit = FALSE;
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rtph264pay->discont = FALSE;
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rtph264pay->adapter = gst_adapter_new ();
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gst_pad_set_query_function (GST_RTP_BASE_PAYLOAD_SRCPAD (rtph264pay),
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gst_rtp_h264_pay_src_query);
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}
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static void
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gst_rtp_h264_pay_clear_sps_pps (GstRtpH264Pay * rtph264pay)
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{
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g_ptr_array_set_size (rtph264pay->sps, 0);
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g_ptr_array_set_size (rtph264pay->pps, 0);
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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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g_array_free (rtph264pay->queue, TRUE);
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g_ptr_array_free (rtph264pay->sps, TRUE);
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g_ptr_array_free (rtph264pay->pps, TRUE);
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g_free (rtph264pay->sprop_parameter_sets);
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g_object_unref (rtph264pay->adapter);
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gst_rtp_h264_pay_reset_bundle (rtph264pay);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static const gchar all_levels[][4] = {
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"1",
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"1b",
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"1.1",
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"1.2",
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"1.3",
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"2",
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"2.1",
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"2.2",
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"3",
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"3.1",
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"3.2",
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"4",
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"4.1",
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"4.2",
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"5",
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"5.1"
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};
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static GstCaps *
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gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
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GstCaps * filter)
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{
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GstCaps *template_caps;
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GstCaps *allowed_caps;
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GstCaps *caps, *icaps;
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gboolean append_unrestricted;
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guint i;
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allowed_caps =
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gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), NULL);
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if (allowed_caps == NULL)
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return NULL;
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template_caps =
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gst_static_pad_template_get_caps (&gst_rtp_h264_pay_sink_template);
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if (gst_caps_is_any (allowed_caps)) {
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caps = gst_caps_ref (template_caps);
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goto done;
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}
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if (gst_caps_is_empty (allowed_caps)) {
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caps = gst_caps_ref (allowed_caps);
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goto done;
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}
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caps = gst_caps_new_empty ();
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append_unrestricted = FALSE;
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for (i = 0; i < gst_caps_get_size (allowed_caps); i++) {
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GstStructure *s = gst_caps_get_structure (allowed_caps, i);
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GstStructure *new_s = gst_structure_new_empty ("video/x-h264");
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const gchar *profile_level_id;
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profile_level_id = gst_structure_get_string (s, "profile-level-id");
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if (profile_level_id && strlen (profile_level_id) == 6) {
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const gchar *profile;
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const gchar *level;
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long int spsint;
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guint8 sps[3];
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spsint = strtol (profile_level_id, NULL, 16);
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sps[0] = spsint >> 16;
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sps[1] = spsint >> 8;
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sps[2] = spsint;
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profile = gst_codec_utils_h264_get_profile (sps, 3);
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level = gst_codec_utils_h264_get_level (sps, 3);
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if (profile && level) {
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GST_LOG_OBJECT (payload, "In caps, have profile %s and level %s",
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profile, level);
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if (!strcmp (profile, "constrained-baseline"))
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gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL);
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else {
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GValue val = { 0, };
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GValue profiles = { 0, };
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g_value_init (&profiles, GST_TYPE_LIST);
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g_value_init (&val, G_TYPE_STRING);
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g_value_set_static_string (&val, profile);
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gst_value_list_append_value (&profiles, &val);
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g_value_set_static_string (&val, "constrained-baseline");
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gst_value_list_append_value (&profiles, &val);
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gst_structure_take_value (new_s, "profile", &profiles);
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}
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if (!strcmp (level, "1"))
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gst_structure_set (new_s, "level", G_TYPE_STRING, level, NULL);
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else {
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GValue levels = { 0, };
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GValue val = { 0, };
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int j;
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g_value_init (&levels, GST_TYPE_LIST);
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g_value_init (&val, G_TYPE_STRING);
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for (j = 0; j < G_N_ELEMENTS (all_levels); j++) {
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g_value_set_static_string (&val, all_levels[j]);
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gst_value_list_prepend_value (&levels, &val);
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if (!strcmp (level, all_levels[j]))
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break;
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}
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gst_structure_take_value (new_s, "level", &levels);
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}
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} else {
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/* Invalid profile-level-id means baseline */
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gst_structure_set (new_s,
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"profile", G_TYPE_STRING, "constrained-baseline", NULL);
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}
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} else {
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/* No profile-level-id means baseline or unrestricted */
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gst_structure_set (new_s,
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"profile", G_TYPE_STRING, "constrained-baseline", NULL);
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append_unrestricted = TRUE;
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}
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caps = gst_caps_merge_structure (caps, new_s);
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}
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if (append_unrestricted) {
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caps =
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gst_caps_merge_structure (caps, gst_structure_new ("video/x-h264", NULL,
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NULL));
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}
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icaps = gst_caps_intersect (caps, template_caps);
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gst_caps_unref (caps);
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caps = icaps;
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done:
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if (filter) {
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GST_DEBUG_OBJECT (payload, "Intersect %" GST_PTR_FORMAT " and filter %"
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GST_PTR_FORMAT, caps, filter);
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icaps = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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caps = icaps;
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}
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gst_caps_unref (template_caps);
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gst_caps_unref (allowed_caps);
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GST_LOG_OBJECT (payload, "returning caps %" GST_PTR_FORMAT, caps);
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return caps;
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}
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|
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static gboolean
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gst_rtp_h264_pay_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
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{
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GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (parent);
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if (GST_QUERY_TYPE (query) == GST_QUERY_LATENCY) {
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gboolean retval;
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gboolean live;
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GstClockTime min_latency, max_latency;
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retval = gst_pad_query_default (pad, parent, query);
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if (!retval)
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return retval;
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if (rtph264pay->stream_format == GST_H264_STREAM_FORMAT_UNKNOWN ||
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rtph264pay->alignment == GST_H264_ALIGNMENT_UNKNOWN)
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return FALSE;
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gst_query_parse_latency (query, &live, &min_latency, &max_latency);
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|
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if (rtph264pay->aggregate_mode == GST_RTP_H264_AGGREGATE_MAX_STAP &&
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rtph264pay->alignment != GST_H264_ALIGNMENT_AU && rtph264pay->fps_num) {
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GstClockTime one_frame = gst_util_uint64_scale_int (GST_SECOND,
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rtph264pay->fps_denum, rtph264pay->fps_num);
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min_latency += one_frame;
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max_latency += one_frame;
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gst_query_set_latency (query, live, min_latency, max_latency);
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}
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return TRUE;
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}
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return gst_pad_query_default (pad, parent, query);
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}
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|
|
/* take the currently configured SPS and PPS lists and set them on the caps as
|
|
* sprop-parameter-sets */
|
|
static gboolean
|
|
gst_rtp_h264_pay_set_sps_pps (GstRTPBasePayload * basepayload)
|
|
{
|
|
GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
|
|
gchar *profile;
|
|
gchar *set;
|
|
GString *sprops;
|
|
guint count;
|
|
gboolean res;
|
|
GstMapInfo map;
|
|
guint i;
|
|
|
|
sprops = g_string_new ("");
|
|
count = 0;
|
|
|
|
/* build the sprop-parameter-sets */
|
|
for (i = 0; i < payloader->sps->len; i++) {
|
|
GstBuffer *sps_buf =
|
|
GST_BUFFER_CAST (g_ptr_array_index (payloader->sps, i));
|
|
|
|
gst_buffer_map (sps_buf, &map, GST_MAP_READ);
|
|
set = g_base64_encode (map.data, map.size);
|
|
gst_buffer_unmap (sps_buf, &map);
|
|
|
|
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
|
|
g_free (set);
|
|
count++;
|
|
}
|
|
for (i = 0; i < payloader->pps->len; i++) {
|
|
GstBuffer *pps_buf =
|
|
GST_BUFFER_CAST (g_ptr_array_index (payloader->pps, i));
|
|
|
|
gst_buffer_map (pps_buf, &map, GST_MAP_READ);
|
|
set = g_base64_encode (map.data, map.size);
|
|
gst_buffer_unmap (pps_buf, &map);
|
|
|
|
g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
|
|
g_free (set);
|
|
count++;
|
|
}
|
|
|
|
if (G_LIKELY (count)) {
|
|
if (payloader->profile != 0) {
|
|
/* profile is 24 bit. Force it to respect the limit */
|
|
profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
|
|
/* combine into output caps */
|
|
res = gst_rtp_base_payload_set_outcaps (basepayload,
|
|
"packetization-mode", G_TYPE_STRING, "1",
|
|
"profile-level-id", G_TYPE_STRING, profile,
|
|
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
|
|
g_free (profile);
|
|
} else {
|
|
res = gst_rtp_base_payload_set_outcaps (basepayload,
|
|
"packetization-mode", G_TYPE_STRING, "1",
|
|
"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
|
|
}
|
|
|
|
} else {
|
|
res = gst_rtp_base_payload_set_outcaps (basepayload, NULL);
|
|
}
|
|
g_string_free (sprops, TRUE);
|
|
|
|
return res;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
GstStructure *str;
|
|
const GValue *value;
|
|
GstMapInfo map;
|
|
guint8 *data;
|
|
gsize size;
|
|
GstBuffer *buffer;
|
|
const gchar *alignment, *stream_format;
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
|
|
str = gst_caps_get_structure (caps, 0);
|
|
|
|
/* we can only set the output caps when we found the sprops and profile
|
|
* NALs */
|
|
gst_rtp_base_payload_set_options (basepayload, "video", TRUE, "H264", 90000);
|
|
|
|
rtph264pay->alignment = GST_H264_ALIGNMENT_UNKNOWN;
|
|
alignment = gst_structure_get_string (str, "alignment");
|
|
if (alignment) {
|
|
if (g_str_equal (alignment, "au"))
|
|
rtph264pay->alignment = GST_H264_ALIGNMENT_AU;
|
|
if (g_str_equal (alignment, "nal"))
|
|
rtph264pay->alignment = GST_H264_ALIGNMENT_NAL;
|
|
}
|
|
|
|
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_UNKNOWN;
|
|
stream_format = gst_structure_get_string (str, "stream-format");
|
|
if (stream_format) {
|
|
if (g_str_equal (stream_format, "avc"))
|
|
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_AVC;
|
|
if (g_str_equal (stream_format, "byte-stream"))
|
|
rtph264pay->stream_format = GST_H264_STREAM_FORMAT_BYTESTREAM;
|
|
}
|
|
|
|
if (!gst_structure_get_fraction (str, "framerate", &rtph264pay->fps_num,
|
|
&rtph264pay->fps_denum))
|
|
rtph264pay->fps_num = rtph264pay->fps_denum = 0;
|
|
|
|
/* packetized AVC video has a codec_data */
|
|
if ((value = gst_structure_get_value (str, "codec_data"))) {
|
|
guint num_sps, num_pps;
|
|
gint i, nal_size;
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay, "have packetized h264");
|
|
|
|
buffer = gst_value_get_buffer (value);
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READ);
|
|
data = map.data;
|
|
size = map.size;
|
|
|
|
/* parse the avcC data */
|
|
if (size < 7)
|
|
goto avcc_too_small;
|
|
/* parse the version, this must be 1 */
|
|
if (data[0] != 1)
|
|
goto wrong_version;
|
|
|
|
/* AVCProfileIndication */
|
|
/* profile_compat */
|
|
/* AVCLevelIndication */
|
|
rtph264pay->profile = (data[1] << 16) | (data[2] << 8) | data[3];
|
|
GST_DEBUG_OBJECT (rtph264pay, "profile %06x", rtph264pay->profile);
|
|
|
|
/* 6 bits reserved | 2 bits lengthSizeMinusOne */
|
|
/* this is the number of bytes in front of the NAL units to mark their
|
|
* length */
|
|
rtph264pay->nal_length_size = (data[4] & 0x03) + 1;
|
|
GST_DEBUG_OBJECT (rtph264pay, "nal length %u", rtph264pay->nal_length_size);
|
|
/* 3 bits reserved | 5 bits numOfSequenceParameterSets */
|
|
num_sps = data[5] & 0x1f;
|
|
GST_DEBUG_OBJECT (rtph264pay, "num SPS %u", num_sps);
|
|
|
|
data += 6;
|
|
size -= 6;
|
|
|
|
/* create the sprop-parameter-sets */
|
|
for (i = 0; i < num_sps; i++) {
|
|
GstBuffer *sps_buf;
|
|
|
|
if (size < 2)
|
|
goto avcc_error;
|
|
|
|
nal_size = (data[0] << 8) | data[1];
|
|
data += 2;
|
|
size -= 2;
|
|
|
|
GST_LOG_OBJECT (rtph264pay, "SPS %d size %d", i, nal_size);
|
|
|
|
if (size < nal_size)
|
|
goto avcc_error;
|
|
|
|
/* make a buffer out of it and add to SPS list */
|
|
sps_buf = gst_buffer_new_and_alloc (nal_size);
|
|
gst_buffer_fill (sps_buf, 0, data, nal_size);
|
|
gst_rtp_h264_add_sps_pps (GST_ELEMENT (rtph264pay), rtph264pay->sps,
|
|
rtph264pay->pps, sps_buf);
|
|
data += nal_size;
|
|
size -= nal_size;
|
|
}
|
|
if (size < 1)
|
|
goto avcc_error;
|
|
|
|
/* 8 bits numOfPictureParameterSets */
|
|
num_pps = data[0];
|
|
data += 1;
|
|
size -= 1;
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay, "num PPS %u", num_pps);
|
|
for (i = 0; i < num_pps; i++) {
|
|
GstBuffer *pps_buf;
|
|
|
|
if (size < 2)
|
|
goto avcc_error;
|
|
|
|
nal_size = (data[0] << 8) | data[1];
|
|
data += 2;
|
|
size -= 2;
|
|
|
|
GST_LOG_OBJECT (rtph264pay, "PPS %d size %d", i, nal_size);
|
|
|
|
if (size < nal_size)
|
|
goto avcc_error;
|
|
|
|
/* make a buffer out of it and add to PPS list */
|
|
pps_buf = gst_buffer_new_and_alloc (nal_size);
|
|
gst_buffer_fill (pps_buf, 0, data, nal_size);
|
|
gst_rtp_h264_add_sps_pps (GST_ELEMENT (rtph264pay), rtph264pay->sps,
|
|
rtph264pay->pps, pps_buf);
|
|
|
|
data += nal_size;
|
|
size -= nal_size;
|
|
}
|
|
|
|
/* and update the caps with the collected data */
|
|
if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
|
|
goto set_sps_pps_failed;
|
|
|
|
gst_buffer_unmap (buffer, &map);
|
|
} else {
|
|
GST_DEBUG_OBJECT (rtph264pay, "have bytestream h264");
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
avcc_too_small:
|
|
{
|
|
GST_ERROR_OBJECT (rtph264pay, "avcC size %" G_GSIZE_FORMAT " < 7", size);
|
|
goto error;
|
|
}
|
|
wrong_version:
|
|
{
|
|
GST_ERROR_OBJECT (rtph264pay, "wrong avcC version");
|
|
goto error;
|
|
}
|
|
avcc_error:
|
|
{
|
|
GST_ERROR_OBJECT (rtph264pay, "avcC too small ");
|
|
goto error;
|
|
}
|
|
set_sps_pps_failed:
|
|
{
|
|
GST_ERROR_OBJECT (rtph264pay, "failed to set sps/pps");
|
|
goto error;
|
|
}
|
|
error:
|
|
{
|
|
gst_buffer_unmap (buffer, &map);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_h264_pay_parse_sprop_parameter_sets (GstRtpH264Pay * rtph264pay)
|
|
{
|
|
const gchar *ps;
|
|
gchar **params;
|
|
guint len;
|
|
gint i;
|
|
GstBuffer *buf;
|
|
|
|
ps = rtph264pay->sprop_parameter_sets;
|
|
if (ps == NULL)
|
|
return;
|
|
|
|
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
|
|
|
|
params = g_strsplit (ps, ",", 0);
|
|
len = g_strv_length (params);
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay, "we have %d params", len);
|
|
|
|
for (i = 0; params[i]; i++) {
|
|
gsize nal_len;
|
|
GstMapInfo map;
|
|
guint8 *nalp;
|
|
guint save = 0;
|
|
gint state = 0;
|
|
|
|
nal_len = strlen (params[i]);
|
|
buf = gst_buffer_new_and_alloc (nal_len);
|
|
|
|
gst_buffer_map (buf, &map, GST_MAP_WRITE);
|
|
nalp = map.data;
|
|
nal_len = g_base64_decode_step (params[i], nal_len, nalp, &state, &save);
|
|
gst_buffer_unmap (buf, &map);
|
|
gst_buffer_resize (buf, 0, nal_len);
|
|
|
|
if (!nal_len) {
|
|
gst_buffer_unref (buf);
|
|
continue;
|
|
}
|
|
|
|
gst_rtp_h264_add_sps_pps (GST_ELEMENT (rtph264pay), rtph264pay->sps,
|
|
rtph264pay->pps, buf);
|
|
}
|
|
g_strfreev (params);
|
|
}
|
|
|
|
static guint
|
|
next_start_code (const guint8 * data, guint size)
|
|
{
|
|
/* Boyer-Moore string matching algorithm, in a degenerative
|
|
* sense because our search 'alphabet' is binary - 0 & 1 only.
|
|
* This allow us to simplify the general BM algorithm to a very
|
|
* simple form. */
|
|
/* assume 1 is in the 3th byte */
|
|
guint offset = 2;
|
|
|
|
while (offset < size) {
|
|
if (1 == data[offset]) {
|
|
unsigned int shift = offset;
|
|
|
|
if (0 == data[--shift]) {
|
|
if (0 == data[--shift]) {
|
|
return shift;
|
|
}
|
|
}
|
|
/* The jump is always 3 because of the 1 previously matched.
|
|
* All the 0's must be after this '1' matched at offset */
|
|
offset += 3;
|
|
} else if (0 == data[offset]) {
|
|
/* maybe next byte is 1? */
|
|
offset++;
|
|
} else {
|
|
/* can jump 3 bytes forward */
|
|
offset += 3;
|
|
}
|
|
/* at each iteration, we rescan in a backward manner until
|
|
* we match 0.0.1 in reverse order. Since our search string
|
|
* has only 2 'alpabets' (i.e. 0 & 1), we know that any
|
|
* mismatch will force us to shift a fixed number of steps */
|
|
}
|
|
GST_DEBUG ("Cannot find next NAL start code. returning %u", size);
|
|
|
|
return size;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_h264_pay_decode_nal (GstRtpH264Pay * payloader,
|
|
const guint8 * data, guint size, GstClockTime dts, GstClockTime pts)
|
|
{
|
|
guint8 header, type;
|
|
gboolean updated;
|
|
|
|
/* default is no update */
|
|
updated = FALSE;
|
|
|
|
GST_DEBUG ("NAL payload len=%u", size);
|
|
|
|
header = data[0];
|
|
type = header & 0x1f;
|
|
|
|
/* We record the timestamp of the last SPS/PPS so
|
|
* that we can insert them at regular intervals and when needed. */
|
|
if (SPS_TYPE_ID == type || PPS_TYPE_ID == type) {
|
|
GstBuffer *nal;
|
|
|
|
/* trailing 0x0 are not part of the SPS/PPS */
|
|
while (size > 0 && data[size - 1] == 0x0)
|
|
size--;
|
|
|
|
/* encode the entire SPS NAL in base64 */
|
|
GST_DEBUG ("Found %s %x %x %x Len=%u", type == SPS_TYPE_ID ? "SPS" : "PPS",
|
|
(header >> 7), (header >> 5) & 3, type, size);
|
|
|
|
nal = gst_buffer_new_allocate (NULL, size, NULL);
|
|
gst_buffer_fill (nal, 0, data, size);
|
|
|
|
updated = gst_rtp_h264_add_sps_pps (GST_ELEMENT (payloader),
|
|
payloader->sps, payloader->pps, nal);
|
|
|
|
/* remember when we last saw SPS */
|
|
if (pts != -1)
|
|
payloader->last_spspps =
|
|
gst_segment_to_running_time (&GST_RTP_BASE_PAYLOAD_CAST
|
|
(payloader)->segment, GST_FORMAT_TIME, pts);
|
|
} else {
|
|
GST_DEBUG ("NAL: %x %x %x Len = %u", (header >> 7),
|
|
(header >> 5) & 3, type, size);
|
|
}
|
|
|
|
return updated;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_payload_nal (GstRTPBasePayload * basepayload,
|
|
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
|
|
gboolean delta_unit, gboolean discont);
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_payload_nal_single (GstRTPBasePayload * basepayload,
|
|
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
|
|
gboolean delta_unit, gboolean discont);
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_payload_nal_fragment (GstRTPBasePayload * basepayload,
|
|
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
|
|
gboolean delta_unit, gboolean discont, guint8 nal_header);
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_payload_nal_bundle (GstRTPBasePayload * basepayload,
|
|
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
|
|
gboolean delta_unit, gboolean discont, guint8 nal_header);
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_send_sps_pps (GstRTPBasePayload * basepayload,
|
|
GstClockTime dts, GstClockTime pts, gboolean delta_unit, gboolean discont)
|
|
{
|
|
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
gboolean sent_all_sps_pps = TRUE;
|
|
guint i;
|
|
|
|
for (i = 0; i < rtph264pay->sps->len; i++) {
|
|
GstBuffer *sps_buf =
|
|
GST_BUFFER_CAST (g_ptr_array_index (rtph264pay->sps, i));
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay, "inserting SPS in the stream");
|
|
/* resend SPS */
|
|
ret = gst_rtp_h264_pay_payload_nal (basepayload, gst_buffer_ref (sps_buf),
|
|
dts, pts, FALSE, delta_unit, discont);
|
|
/* Not critical here; but throw a warning */
|
|
if (ret != GST_FLOW_OK) {
|
|
sent_all_sps_pps = FALSE;
|
|
GST_WARNING_OBJECT (basepayload, "Problem pushing SPS");
|
|
}
|
|
}
|
|
for (i = 0; i < rtph264pay->pps->len; i++) {
|
|
GstBuffer *pps_buf =
|
|
GST_BUFFER_CAST (g_ptr_array_index (rtph264pay->pps, i));
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay, "inserting PPS in the stream");
|
|
/* resend PPS */
|
|
ret = gst_rtp_h264_pay_payload_nal (basepayload, gst_buffer_ref (pps_buf),
|
|
dts, pts, FALSE, TRUE, FALSE);
|
|
/* Not critical here; but throw a warning */
|
|
if (ret != GST_FLOW_OK) {
|
|
sent_all_sps_pps = FALSE;
|
|
GST_WARNING_OBJECT (basepayload, "Problem pushing PPS");
|
|
}
|
|
}
|
|
|
|
if (pts != -1 && sent_all_sps_pps)
|
|
rtph264pay->last_spspps =
|
|
gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME,
|
|
pts);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* @delta_unit: if %FALSE the first packet sent won't have the
|
|
* GST_BUFFER_FLAG_DELTA_UNIT flag.
|
|
* @discont: if %TRUE the first packet sent will have the
|
|
* GST_BUFFER_FLAG_DISCONT flag.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_payload_nal (GstRTPBasePayload * basepayload,
|
|
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
|
|
gboolean delta_unit, gboolean discont)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
guint8 nal_header, nal_type;
|
|
gboolean send_spspps;
|
|
guint size;
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
size = gst_buffer_get_size (paybuf);
|
|
|
|
gst_buffer_extract (paybuf, 0, &nal_header, 1);
|
|
nal_type = nal_header & 0x1f;
|
|
|
|
/* These payload type are reserved for STAP-A, STAP-B, MTAP16, and MTAP24
|
|
* as internally used NAL types */
|
|
switch (nal_type) {
|
|
case 24:
|
|
case 25:
|
|
case 26:
|
|
case 27:
|
|
GST_WARNING_OBJECT (rtph264pay, "Ignoring reserved NAL TYPE=%d",
|
|
nal_type);
|
|
gst_buffer_unref (paybuf);
|
|
return GST_FLOW_OK;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay,
|
|
"payloading NAL Unit: datasize=%u type=%d pts=%" GST_TIME_FORMAT,
|
|
size, nal_type, GST_TIME_ARGS (pts));
|
|
|
|
/* should set src caps before pushing stuff,
|
|
* and if we did not see enough SPS/PPS, that may not be the case */
|
|
if (G_UNLIKELY (!gst_pad_has_current_caps (GST_RTP_BASE_PAYLOAD_SRCPAD
|
|
(basepayload))))
|
|
gst_rtp_h264_pay_set_sps_pps (basepayload);
|
|
|
|
send_spspps = FALSE;
|
|
|
|
/* check if we need to emit an SPS/PPS now */
|
|
if (nal_type == IDR_TYPE_ID && rtph264pay->spspps_interval > 0) {
|
|
if (rtph264pay->last_spspps != -1) {
|
|
guint64 diff;
|
|
GstClockTime running_time =
|
|
gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME,
|
|
pts);
|
|
|
|
GST_LOG_OBJECT (rtph264pay,
|
|
"now %" GST_TIME_FORMAT ", last SPS/PPS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (running_time),
|
|
GST_TIME_ARGS (rtph264pay->last_spspps));
|
|
|
|
/* calculate diff between last SPS/PPS in milliseconds */
|
|
if (running_time > rtph264pay->last_spspps)
|
|
diff = running_time - rtph264pay->last_spspps;
|
|
else
|
|
diff = 0;
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay,
|
|
"interval since last SPS/PPS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (diff));
|
|
|
|
/* bigger than interval, queue SPS/PPS */
|
|
if (GST_TIME_AS_SECONDS (diff) >= rtph264pay->spspps_interval) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "time to send SPS/PPS");
|
|
send_spspps = TRUE;
|
|
}
|
|
} else {
|
|
/* no know previous SPS/PPS time, send now */
|
|
GST_DEBUG_OBJECT (rtph264pay, "no previous SPS/PPS time, send now");
|
|
send_spspps = TRUE;
|
|
}
|
|
} else if (nal_type == IDR_TYPE_ID && rtph264pay->spspps_interval == -1) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "sending SPS/PPS before current IDR frame");
|
|
/* send SPS/PPS before every IDR frame */
|
|
send_spspps = TRUE;
|
|
}
|
|
|
|
if (send_spspps || rtph264pay->send_spspps) {
|
|
/* we need to send SPS/PPS now first. FIXME, don't use the pts for
|
|
* checking when we need to send SPS/PPS but convert to running_time first. */
|
|
GstFlowReturn ret;
|
|
|
|
rtph264pay->send_spspps = FALSE;
|
|
|
|
ret = gst_rtp_h264_pay_send_sps_pps (basepayload, dts, pts, delta_unit,
|
|
discont);
|
|
if (ret != GST_FLOW_OK) {
|
|
gst_buffer_unref (paybuf);
|
|
return ret;
|
|
}
|
|
|
|
delta_unit = TRUE;
|
|
discont = FALSE;
|
|
}
|
|
|
|
if (rtph264pay->aggregate_mode != GST_RTP_H264_AGGREGATE_NONE)
|
|
return gst_rtp_h264_pay_payload_nal_bundle (basepayload, paybuf, dts, pts,
|
|
end_of_au, delta_unit, discont, nal_header);
|
|
|
|
return gst_rtp_h264_pay_payload_nal_fragment (basepayload, paybuf, dts, pts,
|
|
end_of_au, delta_unit, discont, nal_header);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_payload_nal_fragment (GstRTPBasePayload * basepayload,
|
|
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
|
|
gboolean delta_unit, gboolean discont, guint8 nal_header)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
guint mtu, size, max_fragment_size, max_fragments, ii, pos;
|
|
GstBuffer *outbuf;
|
|
guint8 *payload;
|
|
GstBufferList *list = NULL;
|
|
GstRTPBuffer rtp = { NULL };
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtph264pay);
|
|
size = gst_buffer_get_size (paybuf);
|
|
|
|
if (gst_rtp_buffer_calc_packet_len (size, 0, 0) <= mtu) {
|
|
/* We don't need to fragment this packet */
|
|
GST_DEBUG_OBJECT (rtph264pay,
|
|
"sending NAL Unit: datasize=%u mtu=%u", size, mtu);
|
|
return gst_rtp_h264_pay_payload_nal_single (basepayload, paybuf, dts, pts,
|
|
end_of_au, delta_unit, discont);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"using FU-A fragmentation for NAL Unit: datasize=%u mtu=%u", size, mtu);
|
|
|
|
/* We keep 2 bytes for FU indicator and FU Header */
|
|
max_fragment_size = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
|
|
max_fragments = (size + max_fragment_size - 2) / max_fragment_size;
|
|
list = gst_buffer_list_new_sized (max_fragments);
|
|
|
|
/* Start at the NALU payload */
|
|
for (pos = 1, ii = 0; pos < size; pos += max_fragment_size, ii++) {
|
|
guint remaining, fragment_size;
|
|
gboolean first_fragment, last_fragment;
|
|
|
|
remaining = size - pos;
|
|
fragment_size = MIN (remaining, max_fragment_size);
|
|
first_fragment = (pos == 1);
|
|
last_fragment = (remaining <= max_fragment_size);
|
|
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"creating FU-A packet %u/%u, size %u",
|
|
ii + 1, max_fragments, fragment_size);
|
|
|
|
/* use buffer lists
|
|
* create buffer without payload containing only the RTP header
|
|
* (memory block at index 0) */
|
|
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 2, 0, 0);
|
|
|
|
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
|
|
|
|
GST_BUFFER_DTS (outbuf) = dts;
|
|
GST_BUFFER_PTS (outbuf) = pts;
|
|
payload = gst_rtp_buffer_get_payload (&rtp);
|
|
|
|
/* If it's the last fragment and the end of this au, mark the end of
|
|
* slice */
|
|
gst_rtp_buffer_set_marker (&rtp, last_fragment && end_of_au);
|
|
|
|
/* FU indicator */
|
|
payload[0] = (nal_header & 0x60) | FU_A_TYPE_ID;
|
|
|
|
/* FU Header */
|
|
payload[1] = (first_fragment << 7) | (last_fragment << 6) |
|
|
(nal_header & 0x1f);
|
|
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
/* insert payload memory block */
|
|
gst_rtp_copy_video_meta (rtph264pay, outbuf, paybuf);
|
|
gst_buffer_copy_into (outbuf, paybuf, GST_BUFFER_COPY_MEMORY, pos,
|
|
fragment_size);
|
|
|
|
if (!delta_unit)
|
|
/* Only the first packet sent should not have the flag */
|
|
delta_unit = TRUE;
|
|
else
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
|
|
|
|
if (discont) {
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
|
/* Only the first packet sent should have the flag */
|
|
discont = FALSE;
|
|
}
|
|
|
|
/* add the buffer to the buffer list */
|
|
gst_buffer_list_add (list, outbuf);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay,
|
|
"sending FU-A fragments: n=%u datasize=%u mtu=%u", ii, size, mtu);
|
|
|
|
gst_buffer_unref (paybuf);
|
|
return gst_rtp_base_payload_push_list (basepayload, list);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_payload_nal_single (GstRTPBasePayload * basepayload,
|
|
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
|
|
gboolean delta_unit, gboolean discont)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
GstBuffer *outbuf;
|
|
GstRTPBuffer rtp = { NULL };
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
|
|
/* create buffer without payload containing only the RTP header
|
|
* (memory block at index 0) */
|
|
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
|
|
|
|
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
|
|
|
|
/* Mark the end of a frame */
|
|
gst_rtp_buffer_set_marker (&rtp, end_of_au);
|
|
|
|
/* timestamp the outbuffer */
|
|
GST_BUFFER_PTS (outbuf) = pts;
|
|
GST_BUFFER_DTS (outbuf) = dts;
|
|
|
|
if (delta_unit)
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DELTA_UNIT);
|
|
|
|
if (discont)
|
|
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
|
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
/* insert payload memory block */
|
|
gst_rtp_copy_video_meta (rtph264pay, outbuf, paybuf);
|
|
outbuf = gst_buffer_append (outbuf, paybuf);
|
|
|
|
/* push the buffer to the next element */
|
|
return gst_rtp_base_payload_push (basepayload, outbuf);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_h264_pay_reset_bundle (GstRtpH264Pay * rtph264pay)
|
|
{
|
|
g_clear_pointer (&rtph264pay->bundle, gst_buffer_list_unref);
|
|
rtph264pay->bundle_size = 0;
|
|
rtph264pay->bundle_contains_vcl = FALSE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_send_bundle (GstRtpH264Pay * rtph264pay, gboolean end_of_au)
|
|
{
|
|
GstRTPBasePayload *basepayload;
|
|
GstBufferList *bundle;
|
|
guint length, bundle_size;
|
|
GstBuffer *first, *outbuf;
|
|
GstClockTime dts, pts;
|
|
gboolean delta, discont;
|
|
|
|
bundle_size = rtph264pay->bundle_size;
|
|
|
|
if (bundle_size == 0) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "no bundle, nothing to send");
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
basepayload = GST_RTP_BASE_PAYLOAD (rtph264pay);
|
|
bundle = rtph264pay->bundle;
|
|
length = gst_buffer_list_length (bundle);
|
|
|
|
first = gst_buffer_list_get (bundle, 0);
|
|
dts = GST_BUFFER_DTS (first);
|
|
pts = GST_BUFFER_PTS (first);
|
|
delta = GST_BUFFER_FLAG_IS_SET (first, GST_BUFFER_FLAG_DELTA_UNIT);
|
|
discont = GST_BUFFER_FLAG_IS_SET (first, GST_BUFFER_FLAG_DISCONT);
|
|
|
|
if (length == 1) {
|
|
/* Push unaggregated NALU */
|
|
outbuf = gst_buffer_ref (first);
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay,
|
|
"sending NAL Unit unaggregated: datasize=%u", bundle_size - 2);
|
|
} else {
|
|
guint8 stap_header;
|
|
guint i;
|
|
|
|
outbuf = gst_buffer_new_allocate (NULL, sizeof stap_header, NULL);
|
|
stap_header = STAP_A_TYPE_ID;
|
|
|
|
for (i = 0; i < length; i++) {
|
|
GstBuffer *buf = gst_buffer_list_get (bundle, i);
|
|
guint8 nal_header;
|
|
GstMemory *size_header;
|
|
GstMapInfo map;
|
|
|
|
gst_buffer_extract (buf, 0, &nal_header, sizeof nal_header);
|
|
|
|
/* Propagate F bit */
|
|
if ((nal_header & 0x80))
|
|
stap_header |= 0x80;
|
|
|
|
/* Select highest nal_ref_idc */
|
|
if ((nal_header & 0x60) > (stap_header & 0x60))
|
|
stap_header = (stap_header & 0x9f) | (nal_header & 0x60);
|
|
|
|
/* append NALU size */
|
|
size_header = gst_allocator_alloc (NULL, 2, NULL);
|
|
gst_memory_map (size_header, &map, GST_MAP_WRITE);
|
|
GST_WRITE_UINT16_BE (map.data, gst_buffer_get_size (buf));
|
|
gst_memory_unmap (size_header, &map);
|
|
gst_buffer_append_memory (outbuf, size_header);
|
|
|
|
/* append NALU data */
|
|
outbuf = gst_buffer_append (outbuf, gst_buffer_ref (buf));
|
|
}
|
|
|
|
gst_buffer_fill (outbuf, 0, &stap_header, sizeof stap_header);
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay,
|
|
"sending STAP-A bundle: n=%u header=%02x datasize=%u",
|
|
length, stap_header, bundle_size);
|
|
}
|
|
|
|
gst_rtp_h264_pay_reset_bundle (rtph264pay);
|
|
return gst_rtp_h264_pay_payload_nal_single (basepayload, outbuf, dts, pts,
|
|
end_of_au, delta, discont);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_h264_pay_payload_nal_bundle (GstRTPBasePayload * basepayload,
|
|
GstBuffer * paybuf, GstClockTime dts, GstClockTime pts, gboolean end_of_au,
|
|
gboolean delta_unit, gboolean discont, guint8 nal_header)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
GstFlowReturn ret;
|
|
guint mtu, pay_size, bundle_size;
|
|
GstBufferList *bundle;
|
|
guint8 nal_type;
|
|
gboolean start_of_au;
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
nal_type = nal_header & 0x1f;
|
|
mtu = GST_RTP_BASE_PAYLOAD_MTU (rtph264pay);
|
|
pay_size = 2 + gst_buffer_get_size (paybuf);
|
|
bundle = rtph264pay->bundle;
|
|
start_of_au = FALSE;
|
|
|
|
if (bundle) {
|
|
GstBuffer *first = gst_buffer_list_get (bundle, 0);
|
|
|
|
if (nal_type == AUD_TYPE_ID) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "found access delimiter");
|
|
start_of_au = TRUE;
|
|
} else if (discont) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "found discont");
|
|
start_of_au = TRUE;
|
|
} else if (GST_BUFFER_PTS (first) != pts || GST_BUFFER_DTS (first) != dts) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "found timestamp mismatch");
|
|
start_of_au = TRUE;
|
|
}
|
|
}
|
|
|
|
if (start_of_au) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "sending bundle before start of AU");
|
|
|
|
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
|
|
if (ret != GST_FLOW_OK)
|
|
goto out;
|
|
|
|
bundle = NULL;
|
|
}
|
|
|
|
bundle_size = 1 + pay_size;
|
|
|
|
if (gst_rtp_buffer_calc_packet_len (bundle_size, 0, 0) > mtu) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "NAL Unit cannot fit in a bundle");
|
|
|
|
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, FALSE);
|
|
if (ret != GST_FLOW_OK)
|
|
goto out;
|
|
|
|
return gst_rtp_h264_pay_payload_nal_fragment (basepayload, paybuf, dts, pts,
|
|
end_of_au, delta_unit, discont, nal_header);
|
|
}
|
|
|
|
bundle_size = rtph264pay->bundle_size + pay_size;
|
|
|
|
if (gst_rtp_buffer_calc_packet_len (bundle_size, 0, 0) > mtu) {
|
|
GST_DEBUG_OBJECT (rtph264pay,
|
|
"bundle overflows, sending: bundlesize=%u datasize=2+%u mtu=%u",
|
|
rtph264pay->bundle_size, pay_size - 2, mtu);
|
|
|
|
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, FALSE);
|
|
if (ret != GST_FLOW_OK)
|
|
goto out;
|
|
|
|
bundle = NULL;
|
|
}
|
|
|
|
if (!bundle) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "creating new STAP-A aggregate");
|
|
bundle = rtph264pay->bundle = gst_buffer_list_new ();
|
|
bundle_size = rtph264pay->bundle_size = 1;
|
|
rtph264pay->bundle_contains_vcl = FALSE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay,
|
|
"bundling NAL Unit: bundlesize=%u datasize=2+%u mtu=%u",
|
|
rtph264pay->bundle_size, pay_size - 2, mtu);
|
|
|
|
paybuf = gst_buffer_make_writable (paybuf);
|
|
GST_BUFFER_PTS (paybuf) = pts;
|
|
GST_BUFFER_DTS (paybuf) = dts;
|
|
|
|
if (delta_unit)
|
|
GST_BUFFER_FLAG_SET (paybuf, GST_BUFFER_FLAG_DELTA_UNIT);
|
|
else
|
|
GST_BUFFER_FLAG_UNSET (paybuf, GST_BUFFER_FLAG_DELTA_UNIT);
|
|
|
|
if (discont)
|
|
GST_BUFFER_FLAG_SET (paybuf, GST_BUFFER_FLAG_DISCONT);
|
|
else
|
|
GST_BUFFER_FLAG_UNSET (paybuf, GST_BUFFER_FLAG_DISCONT);
|
|
|
|
gst_buffer_list_add (bundle, gst_buffer_ref (paybuf));
|
|
rtph264pay->bundle_size += pay_size;
|
|
ret = GST_FLOW_OK;
|
|
|
|
if ((nal_type >= 1 && nal_type <= 5) || nal_type == 14 ||
|
|
(nal_type >= 20 && nal_type <= 23))
|
|
rtph264pay->bundle_contains_vcl = TRUE;
|
|
|
|
if (end_of_au) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "sending bundle at end of AU");
|
|
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
|
|
}
|
|
|
|
out:
|
|
gst_buffer_unref (paybuf);
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
GstFlowReturn ret;
|
|
gsize size;
|
|
guint nal_len, i;
|
|
const guint8 *data;
|
|
GstClockTime dts, pts;
|
|
GArray *nal_queue;
|
|
gboolean avc;
|
|
GstBuffer *paybuf = NULL;
|
|
gsize skip;
|
|
gboolean delayed_not_delta_unit = FALSE;
|
|
gboolean delayed_discont = FALSE;
|
|
gboolean marker = FALSE;
|
|
gboolean draining = (buffer == NULL);
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
|
|
/* the input buffer contains one or more NAL units */
|
|
|
|
avc = rtph264pay->stream_format == GST_H264_STREAM_FORMAT_AVC;
|
|
|
|
if (avc) {
|
|
/* In AVC mode, there is no adapter, so nothing to drain */
|
|
if (draining)
|
|
return GST_FLOW_OK;
|
|
} else {
|
|
if (buffer) {
|
|
if (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT)) {
|
|
if (gst_adapter_available (rtph264pay->adapter) == 0)
|
|
rtph264pay->delta_unit = FALSE;
|
|
else
|
|
/* This buffer contains a key frame but the adapter isn't empty. So
|
|
* we'll purge it first by sending a first packet and then the second
|
|
* one won't have the DELTA_UNIT flag. */
|
|
delayed_not_delta_unit = TRUE;
|
|
}
|
|
|
|
if (GST_BUFFER_IS_DISCONT (buffer)) {
|
|
if (gst_adapter_available (rtph264pay->adapter) == 0)
|
|
rtph264pay->discont = TRUE;
|
|
else
|
|
/* This buffer has the DISCONT flag but the adapter isn't empty. So
|
|
* we'll purge it first by sending a first packet and then the second
|
|
* one will have the DISCONT flag set. */
|
|
delayed_discont = TRUE;
|
|
}
|
|
|
|
marker = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_MARKER);
|
|
gst_adapter_push (rtph264pay->adapter, buffer);
|
|
buffer = NULL;
|
|
}
|
|
|
|
/* We want to use the first TS used to construct the following NAL */
|
|
dts = gst_adapter_prev_dts (rtph264pay->adapter, NULL);
|
|
pts = gst_adapter_prev_pts (rtph264pay->adapter, NULL);
|
|
|
|
size = gst_adapter_available (rtph264pay->adapter);
|
|
/* Nothing to do here if the adapter is empty, e.g. on EOS */
|
|
if (size == 0)
|
|
return GST_FLOW_OK;
|
|
data = gst_adapter_map (rtph264pay->adapter, size);
|
|
GST_DEBUG_OBJECT (basepayload, "got %" G_GSIZE_FORMAT " bytes", size);
|
|
}
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
/* now loop over all NAL units and put them in a packet */
|
|
if (avc) {
|
|
GstBufferMemoryMap memory;
|
|
gsize remaining_buffer_size;
|
|
guint nal_length_size;
|
|
gsize offset = 0;
|
|
|
|
gst_buffer_memory_map (buffer, &memory);
|
|
remaining_buffer_size = gst_buffer_get_size (buffer);
|
|
|
|
pts = GST_BUFFER_PTS (buffer);
|
|
dts = GST_BUFFER_DTS (buffer);
|
|
rtph264pay->delta_unit = GST_BUFFER_FLAG_IS_SET (buffer,
|
|
GST_BUFFER_FLAG_DELTA_UNIT);
|
|
rtph264pay->discont = GST_BUFFER_IS_DISCONT (buffer);
|
|
marker = GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_MARKER);
|
|
GST_DEBUG_OBJECT (basepayload, "got %" G_GSIZE_FORMAT " bytes",
|
|
remaining_buffer_size);
|
|
|
|
nal_length_size = rtph264pay->nal_length_size;
|
|
|
|
while (remaining_buffer_size > nal_length_size) {
|
|
gint i;
|
|
gboolean end_of_au = FALSE;
|
|
|
|
nal_len = 0;
|
|
for (i = 0; i < nal_length_size; i++) {
|
|
nal_len = (nal_len << 8) + *memory.data;
|
|
if (!gst_buffer_memory_advance_bytes (&memory, 1))
|
|
break;
|
|
}
|
|
|
|
offset += nal_length_size;
|
|
remaining_buffer_size -= nal_length_size;
|
|
|
|
if (remaining_buffer_size >= nal_len) {
|
|
GST_DEBUG_OBJECT (basepayload, "got NAL of size %u", nal_len);
|
|
} else {
|
|
nal_len = remaining_buffer_size;
|
|
GST_DEBUG_OBJECT (basepayload, "got incomplete NAL of size %u",
|
|
nal_len);
|
|
}
|
|
|
|
/* If we're at the end of the buffer, then we're at the end of the
|
|
* access unit
|
|
*/
|
|
if (remaining_buffer_size - nal_len <= nal_length_size) {
|
|
if (rtph264pay->alignment == GST_H264_ALIGNMENT_AU || marker)
|
|
end_of_au = TRUE;
|
|
}
|
|
|
|
paybuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, offset,
|
|
nal_len);
|
|
ret =
|
|
gst_rtp_h264_pay_payload_nal (basepayload, paybuf, dts, pts,
|
|
end_of_au, rtph264pay->delta_unit, rtph264pay->discont);
|
|
|
|
if (!rtph264pay->delta_unit)
|
|
/* Only the first outgoing packet doesn't have the DELTA_UNIT flag */
|
|
rtph264pay->delta_unit = TRUE;
|
|
|
|
if (rtph264pay->discont)
|
|
/* Only the first outgoing packet have the DISCONT flag */
|
|
rtph264pay->discont = FALSE;
|
|
|
|
if (ret != GST_FLOW_OK)
|
|
break;
|
|
|
|
/* Skip current nal. If it is split over multiple GstMemory
|
|
* advance_bytes () will switch to the correct GstMemory. The payloader
|
|
* does not access those bytes directly but uses gst_buffer_copy_region ()
|
|
* to create a sub-buffer referencing the nal instead */
|
|
if (!gst_buffer_memory_advance_bytes (&memory, nal_len))
|
|
break;
|
|
|
|
offset += nal_len;
|
|
remaining_buffer_size -= nal_len;
|
|
}
|
|
|
|
gst_buffer_memory_unmap (&memory);
|
|
gst_buffer_unref (buffer);
|
|
} else {
|
|
guint next;
|
|
gboolean update = FALSE;
|
|
|
|
/* get offset of first start code */
|
|
next = next_start_code (data, size);
|
|
|
|
/* skip to start code, if no start code is found, next will be size and we
|
|
* will not collect data. */
|
|
data += next;
|
|
size -= next;
|
|
nal_queue = rtph264pay->queue;
|
|
skip = next;
|
|
|
|
/* array must be empty when we get here */
|
|
g_assert (nal_queue->len == 0);
|
|
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"found first start at %u, bytes left %" G_GSIZE_FORMAT, next, size);
|
|
|
|
/* first pass to locate NALs and parse SPS/PPS */
|
|
while (size > 4) {
|
|
/* skip start code */
|
|
data += 3;
|
|
size -= 3;
|
|
|
|
/* 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.
|
|
* data is unchanged by the call to next_start_code()
|
|
*/
|
|
next = next_start_code (data, size);
|
|
|
|
/* nal or au aligned input needs no delaying until next time */
|
|
if (next == size && !draining &&
|
|
rtph264pay->alignment == GST_H264_ALIGNMENT_UNKNOWN) {
|
|
/* Didn't find the start of next NAL and it's not EOS,
|
|
* handle it next time */
|
|
break;
|
|
}
|
|
|
|
/* nal length is distance to next start code */
|
|
nal_len = next;
|
|
|
|
GST_DEBUG_OBJECT (basepayload, "found next start at %u of size %u", next,
|
|
nal_len);
|
|
|
|
if (rtph264pay->sprop_parameter_sets != NULL) {
|
|
/* explicitly set profile and sprop, use those */
|
|
if (rtph264pay->update_caps) {
|
|
if (!gst_rtp_base_payload_set_outcaps (basepayload,
|
|
"sprop-parameter-sets", G_TYPE_STRING,
|
|
rtph264pay->sprop_parameter_sets, NULL))
|
|
goto caps_rejected;
|
|
|
|
/* parse SPS and PPS from provided parameter set (for insertion) */
|
|
gst_rtp_h264_pay_parse_sprop_parameter_sets (rtph264pay);
|
|
|
|
rtph264pay->update_caps = FALSE;
|
|
|
|
GST_DEBUG ("outcaps update: sprop-parameter-sets=%s",
|
|
rtph264pay->sprop_parameter_sets);
|
|
}
|
|
} else {
|
|
/* We know our stream is a valid H264 NAL packet,
|
|
* go parse it for SPS/PPS to enrich the caps */
|
|
/* order: make sure to check nal */
|
|
update =
|
|
gst_rtp_h264_pay_decode_nal (rtph264pay, data, nal_len, dts, pts)
|
|
|| update;
|
|
}
|
|
/* move to next NAL packet */
|
|
data += nal_len;
|
|
size -= nal_len;
|
|
|
|
g_array_append_val (nal_queue, nal_len);
|
|
}
|
|
|
|
/* if has new SPS & PPS, update the output caps */
|
|
if (G_UNLIKELY (update))
|
|
if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
|
|
goto caps_rejected;
|
|
|
|
/* second pass to payload and push */
|
|
|
|
if (nal_queue->len != 0)
|
|
gst_adapter_flush (rtph264pay->adapter, skip);
|
|
|
|
for (i = 0; i < nal_queue->len; i++) {
|
|
guint size;
|
|
gboolean end_of_au = FALSE;
|
|
|
|
nal_len = g_array_index (nal_queue, guint, i);
|
|
/* skip start code */
|
|
gst_adapter_flush (rtph264pay->adapter, 3);
|
|
|
|
/* Trim the end unless we're the last NAL in the stream.
|
|
* In case we're not at the end of the buffer we know the next block
|
|
* starts with 0x000001 so all the 0x00 bytes at the end of this one are
|
|
* trailing 0x0 that can be discarded */
|
|
size = nal_len;
|
|
data = gst_adapter_map (rtph264pay->adapter, size);
|
|
if (i + 1 != nal_queue->len || !draining)
|
|
for (; size > 1 && data[size - 1] == 0x0; size--)
|
|
/* skip */ ;
|
|
|
|
|
|
/* If it's the last nal unit we have in non-bytestream mode, we can
|
|
* assume it's the end of an access-unit
|
|
*
|
|
* FIXME: We need to wait until the next packet or EOS to
|
|
* actually payload the NAL so we can know if the current NAL is
|
|
* the last one of an access unit or not if we are in bytestream mode
|
|
*/
|
|
if (i == nal_queue->len - 1) {
|
|
if (rtph264pay->alignment == GST_H264_ALIGNMENT_AU ||
|
|
marker || draining)
|
|
end_of_au = TRUE;
|
|
}
|
|
paybuf = gst_adapter_take_buffer (rtph264pay->adapter, size);
|
|
g_assert (paybuf);
|
|
|
|
/* put the data in one or more RTP packets */
|
|
ret =
|
|
gst_rtp_h264_pay_payload_nal (basepayload, paybuf, dts, pts,
|
|
end_of_au, rtph264pay->delta_unit, rtph264pay->discont);
|
|
|
|
if (delayed_not_delta_unit) {
|
|
rtph264pay->delta_unit = FALSE;
|
|
delayed_not_delta_unit = FALSE;
|
|
} else {
|
|
/* Only the first outgoing packet doesn't have the DELTA_UNIT flag */
|
|
rtph264pay->delta_unit = TRUE;
|
|
}
|
|
|
|
if (delayed_discont) {
|
|
rtph264pay->discont = TRUE;
|
|
delayed_discont = FALSE;
|
|
} else {
|
|
/* Only the first outgoing packet have the DISCONT flag */
|
|
rtph264pay->discont = FALSE;
|
|
}
|
|
|
|
if (ret != GST_FLOW_OK) {
|
|
break;
|
|
}
|
|
|
|
/* move to next NAL packet */
|
|
/* Skips the trailing zeros */
|
|
gst_adapter_flush (rtph264pay->adapter, nal_len - size);
|
|
}
|
|
g_array_set_size (nal_queue, 0);
|
|
}
|
|
|
|
if (ret == GST_FLOW_OK && rtph264pay->bundle_size > 0 &&
|
|
rtph264pay->aggregate_mode == GST_RTP_H264_AGGREGATE_ZERO_LATENCY &&
|
|
rtph264pay->bundle_contains_vcl) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "sending bundle at end incoming packet");
|
|
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, FALSE);
|
|
}
|
|
|
|
|
|
done:
|
|
if (!avc) {
|
|
gst_adapter_unmap (rtph264pay->adapter);
|
|
}
|
|
|
|
return ret;
|
|
|
|
caps_rejected:
|
|
{
|
|
GST_WARNING_OBJECT (basepayload, "Could not set outcaps");
|
|
g_array_set_size (nal_queue, 0);
|
|
ret = GST_FLOW_NOT_NEGOTIATED;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_h264_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
|
|
{
|
|
gboolean res;
|
|
const GstStructure *s;
|
|
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (payload);
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_adapter_clear (rtph264pay->adapter);
|
|
gst_rtp_h264_pay_reset_bundle (rtph264pay);
|
|
break;
|
|
case GST_EVENT_CUSTOM_DOWNSTREAM:
|
|
s = gst_event_get_structure (event);
|
|
if (gst_structure_has_name (s, "GstForceKeyUnit")) {
|
|
gboolean resend_codec_data;
|
|
|
|
if (gst_structure_get_boolean (s, "all-headers",
|
|
&resend_codec_data) && resend_codec_data)
|
|
rtph264pay->send_spspps = TRUE;
|
|
}
|
|
break;
|
|
case GST_EVENT_EOS:
|
|
{
|
|
/* call handle_buffer with NULL to flush last NAL from adapter
|
|
* in byte-stream mode
|
|
*/
|
|
gst_rtp_h264_pay_handle_buffer (payload, NULL);
|
|
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
|
|
break;
|
|
}
|
|
case GST_EVENT_STREAM_START:
|
|
GST_DEBUG_OBJECT (rtph264pay, "New stream detected => Clear SPS and PPS");
|
|
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
|
|
ret = gst_rtp_h264_pay_send_bundle (rtph264pay, TRUE);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (ret != GST_FLOW_OK)
|
|
return FALSE;
|
|
|
|
res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
rtph264pay->send_spspps = FALSE;
|
|
gst_adapter_clear (rtph264pay->adapter);
|
|
gst_rtp_h264_pay_reset_bundle (rtph264pay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
rtph264pay->last_spspps = -1;
|
|
gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_h264_pay_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SPROP_PARAMETER_SETS:
|
|
g_free (rtph264pay->sprop_parameter_sets);
|
|
rtph264pay->sprop_parameter_sets = g_value_dup_string (value);
|
|
rtph264pay->update_caps = TRUE;
|
|
break;
|
|
case PROP_CONFIG_INTERVAL:
|
|
rtph264pay->spspps_interval = g_value_get_int (value);
|
|
break;
|
|
case PROP_AGGREGATE_MODE:
|
|
rtph264pay->aggregate_mode = g_value_get_enum (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_h264_pay_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SPROP_PARAMETER_SETS:
|
|
g_value_set_string (value, rtph264pay->sprop_parameter_sets);
|
|
break;
|
|
case PROP_CONFIG_INTERVAL:
|
|
g_value_set_int (value, rtph264pay->spspps_interval);
|
|
break;
|
|
case PROP_AGGREGATE_MODE:
|
|
g_value_set_enum (value, rtph264pay->aggregate_mode);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_h264_pay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtph264pay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_H264_PAY);
|
|
}
|