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1238 lines
36 KiB
C
1238 lines
36 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., 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 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 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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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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#define GST_TYPE_H264_SCAN_MODE (gst_h264_scan_mode_get_type())
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static GType
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gst_h264_scan_mode_get_type (void)
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{
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static GType h264_scan_mode_type = 0;
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static const GEnumValue h264_scan_modes[] = {
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{GST_H264_SCAN_MODE_BYTESTREAM,
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"Scan complete bytestream for NALUs",
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"bytestream"},
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{GST_H264_SCAN_MODE_MULTI_NAL, "Buffers contain multiple complete NALUs",
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"multiple"},
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{GST_H264_SCAN_MODE_SINGLE_NAL, "Buffers contain a single complete NALU",
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"single"},
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{0, NULL, NULL},
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};
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if (!h264_scan_mode_type) {
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h264_scan_mode_type =
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g_enum_register_static ("GstH264PayScanMode", h264_scan_modes);
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}
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return h264_scan_mode_type;
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}
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#define DEFAULT_PROFILE_LEVEL_ID NULL
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#define DEFAULT_SPROP_PARAMETER_SETS NULL
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#define DEFAULT_SCAN_MODE GST_H264_SCAN_MODE_MULTI_NAL
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#define DEFAULT_BUFFER_LIST FALSE
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#define DEFAULT_CONFIG_INTERVAL 0
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enum
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{
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PROP_0,
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PROP_PROFILE_LEVEL_ID,
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PROP_SPROP_PARAMETER_SETS,
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PROP_SCAN_MODE,
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PROP_BUFFER_LIST,
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PROP_CONFIG_INTERVAL,
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PROP_LAST
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};
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#define IS_ACCESS_UNIT(x) (((x) > 0x00) && ((x) < 0x06))
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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 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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static gboolean gst_rtp_h264_pay_handle_event (GstPad * pad, GstEvent * event);
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static GstStateChangeReturn gst_basertppayload_change_state (GstElement *
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element, GstStateChange transition);
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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_simple (element_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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}
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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->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_PROFILE_LEVEL_ID, g_param_spec_string ("profile-level-id",
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"profile-level-id",
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"The base64 profile-level-id to set in the sink caps (deprecated)",
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DEFAULT_PROFILE_LEVEL_ID,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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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));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SCAN_MODE,
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g_param_spec_enum ("scan-mode", "Scan Mode",
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"How to scan the input buffers for NAL units. Performance can be "
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"increased when certain assumptions are made about the input buffers",
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GST_TYPE_H264_SCAN_MODE, DEFAULT_SCAN_MODE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BUFFER_LIST,
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g_param_spec_boolean ("buffer-list", "Buffer List",
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"Use Buffer Lists",
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DEFAULT_BUFFER_LIST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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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_uint ("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.) (0 = disabled)",
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0, 3600, DEFAULT_CONFIG_INTERVAL,
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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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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_basertppayload_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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gstbasertppayload_class->handle_event = gst_rtp_h264_pay_handle_event;
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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->queue = g_array_new (FALSE, FALSE, sizeof (guint));
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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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rtph264pay->last_spspps = -1;
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rtph264pay->scan_mode = GST_H264_SCAN_MODE_MULTI_NAL;
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rtph264pay->buffer_list = DEFAULT_BUFFER_LIST;
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rtph264pay->spspps_interval = DEFAULT_CONFIG_INTERVAL;
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rtph264pay->adapter = gst_adapter_new ();
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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_list_foreach (rtph264pay->sps, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (rtph264pay->sps);
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rtph264pay->sps = NULL;
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g_list_foreach (rtph264pay->pps, (GFunc) gst_mini_object_unref, NULL);
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g_list_free (rtph264pay->pps);
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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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g_array_free (rtph264pay->queue, TRUE);
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gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
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g_free (rtph264pay->sprop_parameter_sets);
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g_object_unref (rtph264pay->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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/* take the currently configured SPS and PPS lists and set them on the caps as
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* sprop-parameter-sets */
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static gboolean
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gst_rtp_h264_pay_set_sps_pps (GstBaseRTPPayload * basepayload)
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{
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GstRtpH264Pay *payloader = GST_RTP_H264_PAY (basepayload);
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gchar *profile;
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gchar *set;
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GList *walk;
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GString *sprops;
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guint count;
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gboolean res;
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sprops = g_string_new ("");
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count = 0;
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/* build the sprop-parameter-sets */
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for (walk = payloader->sps; walk; walk = g_list_next (walk)) {
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GstBuffer *sps_buf = GST_BUFFER_CAST (walk->data);
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set =
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g_base64_encode (GST_BUFFER_DATA (sps_buf), GST_BUFFER_SIZE (sps_buf));
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g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
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g_free (set);
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count++;
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}
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for (walk = payloader->pps; walk; walk = g_list_next (walk)) {
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GstBuffer *pps_buf = GST_BUFFER_CAST (walk->data);
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set =
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g_base64_encode (GST_BUFFER_DATA (pps_buf), GST_BUFFER_SIZE (pps_buf));
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g_string_append_printf (sprops, "%s%s", count ? "," : "", set);
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g_free (set);
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count++;
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}
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/* profile is 24 bit. Force it to respect the limit */
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profile = g_strdup_printf ("%06x", payloader->profile & 0xffffff);
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/* combine into output caps */
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res = gst_basertppayload_set_outcaps (basepayload,
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"sprop-parameter-sets", G_TYPE_STRING, sprops->str, NULL);
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g_string_free (sprops, TRUE);
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g_free (profile);
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return res;
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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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guint num_sps, num_pps;
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gint i, nal_size;
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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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rtph264pay->profile = (data[1] << 16) | (data[2] << 8) | data[3];
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GST_DEBUG_OBJECT (rtph264pay, "profile %06x", rtph264pay->profile);
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/* 6 bits reserved | 2 bits lengthSizeMinusOne */
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/* this is the number of bytes in front of the NAL units to mark their
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* length */
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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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for (i = 0; i < num_sps; i++) {
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GstBuffer *sps_buf;
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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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GST_LOG_OBJECT (rtph264pay, "SPS %d size %d", i, nal_size);
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if (size < nal_size)
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goto avcc_error;
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/* make a buffer out of it and add to SPS list */
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sps_buf = gst_buffer_new_and_alloc (nal_size);
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memcpy (GST_BUFFER_DATA (sps_buf), data, nal_size);
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rtph264pay->sps = g_list_append (rtph264pay->sps, sps_buf);
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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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/* 8 bits numOfPictureParameterSets */
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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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GstBuffer *pps_buf;
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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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GST_LOG_OBJECT (rtph264pay, "PPS %d size %d", i, nal_size);
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if (size < nal_size)
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goto avcc_error;
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/* make a buffer out of it and add to PPS list */
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pps_buf = gst_buffer_new_and_alloc (nal_size);
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memcpy (GST_BUFFER_DATA (pps_buf), data, nal_size);
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rtph264pay->pps = g_list_append (rtph264pay->pps, pps_buf);
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data += nal_size;
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size -= nal_size;
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}
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/* and update the caps with the collected data */
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if (!gst_rtp_h264_pay_set_sps_pps (basepayload))
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return FALSE;
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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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|
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static void
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gst_rtp_h264_pay_parse_sprop_parameter_sets (GstRtpH264Pay * rtph264pay)
|
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{
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const gchar *ps;
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gchar **params;
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guint len, num_sps, num_pps;
|
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gint i;
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GstBuffer *buf;
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ps = rtph264pay->sprop_parameter_sets;
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if (ps == NULL)
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return;
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|
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gst_rtp_h264_pay_clear_sps_pps (rtph264pay);
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|
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params = g_strsplit (ps, ",", 0);
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len = g_strv_length (params);
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|
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GST_DEBUG_OBJECT (rtph264pay, "we have %d params", len);
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|
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num_sps = num_pps = 0;
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|
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for (i = 0; params[i]; i++) {
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gsize nal_len;
|
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guint8 *nalp;
|
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guint save = 0;
|
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gint state = 0;
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|
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nal_len = strlen (params[i]);
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buf = gst_buffer_new_and_alloc (nal_len);
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nalp = GST_BUFFER_DATA (buf);
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|
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nal_len = g_base64_decode_step (params[i], nal_len, nalp, &state, &save);
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GST_BUFFER_SIZE (buf) = nal_len;
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|
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if (!nal_len) {
|
|
gst_buffer_unref (buf);
|
|
continue;
|
|
}
|
|
|
|
/* append to the right list */
|
|
if ((nalp[0] & 0x1f) == 7) {
|
|
GST_DEBUG_OBJECT (rtph264pay, "adding param %d as SPS %d", i, num_sps);
|
|
rtph264pay->sps = g_list_append (rtph264pay->sps, buf);
|
|
num_sps++;
|
|
} else {
|
|
GST_DEBUG_OBJECT (rtph264pay, "adding param %d as PPS %d", i, num_pps);
|
|
rtph264pay->pps = g_list_append (rtph264pay->pps, buf);
|
|
num_pps++;
|
|
}
|
|
}
|
|
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 timestamp)
|
|
{
|
|
const guint8 *sps = NULL, *pps = NULL;
|
|
guint sps_len = 0, pps_len = 0;
|
|
guint8 header, type;
|
|
guint len;
|
|
gboolean updated;
|
|
|
|
/* default is no update */
|
|
updated = FALSE;
|
|
|
|
GST_DEBUG ("NAL payload len=%u", size);
|
|
|
|
len = size;
|
|
header = data[0];
|
|
type = header & 0x1f;
|
|
|
|
/* keep sps & pps separately so that we can update either one
|
|
* independently. We also 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) {
|
|
/* encode the entire SPS NAL in base64 */
|
|
GST_DEBUG ("Found SPS %x %x %x Len=%u", (header >> 7),
|
|
(header >> 5) & 3, type, len);
|
|
|
|
sps = data;
|
|
sps_len = len;
|
|
/* remember when we last saw SPS */
|
|
if (timestamp != -1)
|
|
payloader->last_spspps = timestamp;
|
|
} else if (PPS_TYPE_ID == type) {
|
|
/* encoder the entire PPS NAL in base64 */
|
|
GST_DEBUG ("Found PPS %x %x %x Len = %u",
|
|
(header >> 7), (header >> 5) & 3, type, len);
|
|
|
|
pps = data;
|
|
pps_len = len;
|
|
/* remember when we last saw PPS */
|
|
if (timestamp != -1)
|
|
payloader->last_spspps = timestamp;
|
|
} else {
|
|
GST_DEBUG ("NAL: %x %x %x Len = %u", (header >> 7),
|
|
(header >> 5) & 3, type, len);
|
|
}
|
|
|
|
/* If we encountered an SPS and/or a PPS, check if it's the
|
|
* same as the one we have. If not, update our version and
|
|
* set updated to TRUE
|
|
*/
|
|
if (sps_len > 0) {
|
|
GstBuffer *sps_buf;
|
|
|
|
if (payloader->sps != NULL) {
|
|
sps_buf = GST_BUFFER_CAST (payloader->sps->data);
|
|
|
|
if ((GST_BUFFER_SIZE (sps_buf) != sps_len)
|
|
|| memcmp (GST_BUFFER_DATA (sps_buf), sps, sps_len)) {
|
|
/* something changed, update */
|
|
payloader->profile = (sps[1] << 16) + (sps[2] << 8) + sps[3];
|
|
GST_DEBUG ("Profile level IDC = %06x", payloader->profile);
|
|
updated = TRUE;
|
|
}
|
|
} else {
|
|
/* no previous SPS, update */
|
|
updated = TRUE;
|
|
}
|
|
|
|
if (updated) {
|
|
sps_buf = gst_buffer_new_and_alloc (sps_len);
|
|
memcpy (GST_BUFFER_DATA (sps_buf), sps, sps_len);
|
|
|
|
if (payloader->sps) {
|
|
/* replace old buffer */
|
|
gst_buffer_unref (payloader->sps->data);
|
|
payloader->sps->data = sps_buf;
|
|
} else {
|
|
/* add new buffer */
|
|
payloader->sps = g_list_prepend (payloader->sps, sps_buf);
|
|
}
|
|
}
|
|
}
|
|
|
|
if (pps_len > 0) {
|
|
GstBuffer *pps_buf;
|
|
|
|
if (payloader->pps != NULL) {
|
|
pps_buf = GST_BUFFER_CAST (payloader->pps->data);
|
|
|
|
if ((GST_BUFFER_SIZE (pps_buf) != pps_len)
|
|
|| memcmp (GST_BUFFER_DATA (pps_buf), pps, pps_len)) {
|
|
/* something changed, update */
|
|
updated = TRUE;
|
|
}
|
|
} else {
|
|
/* no previous SPS, update */
|
|
updated = TRUE;
|
|
}
|
|
|
|
if (updated) {
|
|
pps_buf = gst_buffer_new_and_alloc (pps_len);
|
|
memcpy (GST_BUFFER_DATA (pps_buf), pps, pps_len);
|
|
|
|
if (payloader->pps) {
|
|
/* replace old buffer */
|
|
gst_buffer_unref (payloader->pps->data);
|
|
payloader->pps->data = pps_buf;
|
|
} else {
|
|
/* add new buffer */
|
|
payloader->pps = g_list_prepend (payloader->pps, pps_buf);
|
|
}
|
|
}
|
|
}
|
|
return updated;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
|
|
const guint8 * data, guint size, GstClockTime timestamp,
|
|
GstBuffer * buffer_orig);
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_send_sps_pps (GstBaseRTPPayload * basepayload,
|
|
GstRtpH264Pay * rtph264pay, GstClockTime timestamp)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
GList *walk;
|
|
|
|
for (walk = rtph264pay->sps; walk; walk = g_list_next (walk)) {
|
|
GstBuffer *sps_buf = GST_BUFFER_CAST (walk->data);
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay, "inserting SPS in the stream");
|
|
/* resend SPS */
|
|
ret = gst_rtp_h264_pay_payload_nal (basepayload,
|
|
GST_BUFFER_DATA (sps_buf), GST_BUFFER_SIZE (sps_buf), timestamp,
|
|
sps_buf);
|
|
/* Not critical here; but throw a warning */
|
|
if (ret != GST_FLOW_OK)
|
|
GST_WARNING ("Problem pushing SPS");
|
|
}
|
|
for (walk = rtph264pay->pps; walk; walk = g_list_next (walk)) {
|
|
GstBuffer *pps_buf = GST_BUFFER_CAST (walk->data);
|
|
|
|
GST_DEBUG_OBJECT (rtph264pay, "inserting PPS in the stream");
|
|
/* resend PPS */
|
|
ret = gst_rtp_h264_pay_payload_nal (basepayload,
|
|
GST_BUFFER_DATA (pps_buf), GST_BUFFER_SIZE (pps_buf), timestamp,
|
|
pps_buf);
|
|
/* Not critical here; but throw a warning */
|
|
if (ret != GST_FLOW_OK)
|
|
GST_WARNING ("Problem pushing PPS");
|
|
}
|
|
|
|
if (timestamp != -1)
|
|
rtph264pay->last_spspps = timestamp;
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_payload_nal (GstBaseRTPPayload * basepayload,
|
|
const guint8 * data, guint size, GstClockTime timestamp,
|
|
GstBuffer * buffer_orig)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
GstFlowReturn ret;
|
|
guint8 nalType;
|
|
guint packet_len, payload_len, mtu;
|
|
GstBuffer *outbuf;
|
|
guint8 *payload;
|
|
GstBufferList *list = NULL;
|
|
GstBufferListIterator *it = NULL;
|
|
gboolean send_spspps;
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
mtu = GST_BASE_RTP_PAYLOAD_MTU (rtph264pay);
|
|
|
|
nalType = data[0] & 0x1f;
|
|
GST_DEBUG_OBJECT (rtph264pay, "Processing Buffer with NAL TYPE=%d", nalType);
|
|
|
|
send_spspps = FALSE;
|
|
|
|
/* check if we need to emit an SPS/PPS now */
|
|
if (nalType == IDR_TYPE_ID && rtph264pay->spspps_interval > 0) {
|
|
if (rtph264pay->last_spspps != -1) {
|
|
guint64 diff;
|
|
|
|
GST_LOG_OBJECT (rtph264pay,
|
|
"now %" GST_TIME_FORMAT ", last SPS/PPS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (timestamp), GST_TIME_ARGS (rtph264pay->last_spspps));
|
|
|
|
/* calculate diff between last SPS/PPS in milliseconds */
|
|
if (timestamp > rtph264pay->last_spspps)
|
|
diff = timestamp - 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;
|
|
}
|
|
}
|
|
|
|
if (send_spspps || rtph264pay->send_spspps) {
|
|
/* we need to send SPS/PPS now first. FIXME, don't use the timestamp for
|
|
* checking when we need to send SPS/PPS but convert to running_time first. */
|
|
rtph264pay->send_spspps = FALSE;
|
|
ret = gst_rtp_h264_pay_send_sps_pps (basepayload, rtph264pay, timestamp);
|
|
if (ret != GST_FLOW_OK)
|
|
return ret;
|
|
}
|
|
|
|
packet_len = gst_rtp_buffer_calc_packet_len (size, 0, 0);
|
|
|
|
if (packet_len < mtu) {
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"NAL Unit fit in one packet datasize=%d mtu=%d", size, mtu);
|
|
/* will fit in one packet */
|
|
|
|
if (rtph264pay->buffer_list) {
|
|
/* use buffer lists
|
|
* first create buffer without payload containing only the RTP header
|
|
* and then another buffer containing the payload. both buffers will
|
|
* be then added to the list */
|
|
outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
|
|
} else {
|
|
/* use the old-fashioned way with a single buffer and memcpy */
|
|
outbuf = gst_rtp_buffer_new_allocate (size, 0, 0);
|
|
}
|
|
|
|
/* only set the marker bit on packets containing access units */
|
|
if (IS_ACCESS_UNIT (nalType)) {
|
|
gst_rtp_buffer_set_marker (outbuf, 1);
|
|
}
|
|
|
|
/* timestamp the outbuffer */
|
|
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
|
|
|
|
if (rtph264pay->buffer_list) {
|
|
GstBuffer *paybuf;
|
|
|
|
/* create another buffer with the payload. */
|
|
if (buffer_orig)
|
|
paybuf = gst_buffer_create_sub (buffer_orig, data -
|
|
GST_BUFFER_DATA (buffer_orig), size);
|
|
else {
|
|
paybuf = gst_buffer_new_and_alloc (size);
|
|
memcpy (GST_BUFFER_DATA (paybuf), data, size);
|
|
}
|
|
|
|
list = gst_buffer_list_new ();
|
|
it = gst_buffer_list_iterate (list);
|
|
|
|
/* add both buffers to the buffer list */
|
|
gst_buffer_list_iterator_add_group (it);
|
|
gst_buffer_list_iterator_add (it, outbuf);
|
|
gst_buffer_list_iterator_add (it, paybuf);
|
|
|
|
gst_buffer_list_iterator_free (it);
|
|
|
|
/* push the list to the next element in the pipe */
|
|
ret = gst_basertppayload_push_list (basepayload, list);
|
|
} else {
|
|
payload = gst_rtp_buffer_get_payload (outbuf);
|
|
GST_DEBUG_OBJECT (basepayload, "Copying %d bytes to outbuf", size);
|
|
memcpy (payload, data, size);
|
|
|
|
ret = gst_basertppayload_push (basepayload, outbuf);
|
|
}
|
|
} else {
|
|
/* fragmentation Units FU-A */
|
|
guint8 nalHeader;
|
|
guint limitedSize;
|
|
int ii = 0, start = 1, end = 0, pos = 0;
|
|
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"NAL Unit DOES NOT fit in one packet datasize=%d mtu=%d", size, mtu);
|
|
|
|
nalHeader = *data;
|
|
pos++;
|
|
size--;
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
GST_DEBUG_OBJECT (basepayload, "Using FU-A fragmentation for data size=%d",
|
|
size);
|
|
|
|
/* We keep 2 bytes for FU indicator and FU Header */
|
|
payload_len = gst_rtp_buffer_calc_payload_len (mtu - 2, 0, 0);
|
|
|
|
if (rtph264pay->buffer_list) {
|
|
list = gst_buffer_list_new ();
|
|
it = gst_buffer_list_iterate (list);
|
|
}
|
|
|
|
while (end == 0) {
|
|
limitedSize = size < payload_len ? size : payload_len;
|
|
GST_DEBUG_OBJECT (basepayload,
|
|
"Inside FU-A fragmentation limitedSize=%d iteration=%d", limitedSize,
|
|
ii);
|
|
|
|
if (rtph264pay->buffer_list) {
|
|
/* use buffer lists
|
|
* first create buffer without payload containing only the RTP header
|
|
* and then another buffer containing the payload. both buffers will
|
|
* be then added to the list */
|
|
outbuf = gst_rtp_buffer_new_allocate (2, 0, 0);
|
|
} else {
|
|
/* use the old-fashioned way with a single buffer and memcpy
|
|
* first create buffer to hold the payload */
|
|
outbuf = gst_rtp_buffer_new_allocate (limitedSize + 2, 0, 0);
|
|
}
|
|
|
|
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
|
|
payload = gst_rtp_buffer_get_payload (outbuf);
|
|
|
|
if (limitedSize == size) {
|
|
GST_DEBUG_OBJECT (basepayload, "end size=%d iteration=%d", size, ii);
|
|
end = 1;
|
|
}
|
|
if (IS_ACCESS_UNIT (nalType)) {
|
|
gst_rtp_buffer_set_marker (outbuf, end);
|
|
}
|
|
|
|
/* FU indicator */
|
|
payload[0] = (nalHeader & 0x60) | 28;
|
|
|
|
/* FU Header */
|
|
payload[1] = (start << 7) | (end << 6) | (nalHeader & 0x1f);
|
|
|
|
if (rtph264pay->buffer_list) {
|
|
GstBuffer *paybuf;
|
|
|
|
/* create another buffer to hold the payload */
|
|
if (buffer_orig)
|
|
paybuf = gst_buffer_create_sub (buffer_orig, data -
|
|
GST_BUFFER_DATA (buffer_orig) + pos, limitedSize);
|
|
else {
|
|
paybuf = gst_buffer_new_and_alloc (limitedSize);
|
|
memcpy (GST_BUFFER_DATA (paybuf), data + pos, limitedSize);
|
|
}
|
|
|
|
/* create a new group to hold the header and the payload */
|
|
gst_buffer_list_iterator_add_group (it);
|
|
|
|
/* add both buffers to the buffer list */
|
|
gst_buffer_list_iterator_add (it, outbuf);
|
|
gst_buffer_list_iterator_add (it, paybuf);
|
|
|
|
} else {
|
|
memcpy (&payload[2], data + pos, 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;
|
|
}
|
|
|
|
size -= limitedSize;
|
|
pos += limitedSize;
|
|
ii++;
|
|
start = 0;
|
|
}
|
|
|
|
if (rtph264pay->buffer_list) {
|
|
/* free iterator and push the whole buffer list at once */
|
|
gst_buffer_list_iterator_free (it);
|
|
ret = gst_basertppayload_push_list (basepayload, list);
|
|
}
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_h264_pay_handle_buffer (GstBaseRTPPayload * basepayload,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpH264Pay *rtph264pay;
|
|
GstFlowReturn ret;
|
|
guint size, nal_len, i;
|
|
const guint8 *data, *nal_data;
|
|
GstClockTime timestamp;
|
|
GArray *nal_queue;
|
|
guint pushed = 0;
|
|
|
|
rtph264pay = GST_RTP_H264_PAY (basepayload);
|
|
|
|
/* the input buffer contains one or more NAL units */
|
|
|
|
if (rtph264pay->scan_mode == GST_H264_SCAN_MODE_BYTESTREAM) {
|
|
timestamp = gst_adapter_prev_timestamp (rtph264pay->adapter, NULL);
|
|
gst_adapter_push (rtph264pay->adapter, buffer);
|
|
size = gst_adapter_available (rtph264pay->adapter);
|
|
data = gst_adapter_peek (rtph264pay->adapter, size);
|
|
GST_DEBUG_OBJECT (basepayload, "got %d bytes (%d)", size,
|
|
GST_BUFFER_SIZE (buffer));
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (timestamp))
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
} else {
|
|
size = GST_BUFFER_SIZE (buffer);
|
|
data = GST_BUFFER_DATA (buffer);
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
GST_DEBUG_OBJECT (basepayload, "got %d bytes", size);
|
|
}
|
|
|
|
ret = GST_FLOW_OK;
|
|
|
|
/* now loop over all NAL units and put them in a packet
|
|
* FIXME, we should really try to pack multiple NAL units into one RTP packet
|
|
* if we can, especially for the config packets that wont't cause decoder
|
|
* latency. */
|
|
if (rtph264pay->packetized) {
|
|
guint nal_length_size;
|
|
|
|
nal_length_size = rtph264pay->nal_length_size;
|
|
|
|
while (size > nal_length_size) {
|
|
gint i;
|
|
|
|
nal_len = 0;
|
|
for (i = 0; i < nal_length_size; i++) {
|
|
nal_len = ((nal_len << 8) + data[i]);
|
|
}
|
|
|
|
/* skip the length bytes, make sure we don't run past the buffer size */
|
|
data += nal_length_size;
|
|
size -= nal_length_size;
|
|
|
|
if (size >= nal_len) {
|
|
GST_DEBUG_OBJECT (basepayload, "got NAL of size %u", nal_len);
|
|
} else {
|
|
nal_len = size;
|
|
GST_DEBUG_OBJECT (basepayload, "got incomplete NAL of size %u",
|
|
nal_len);
|
|
}
|
|
|
|
ret =
|
|
gst_rtp_h264_pay_payload_nal (basepayload, data, nal_len, timestamp,
|
|
buffer);
|
|
if (ret != GST_FLOW_OK)
|
|
break;
|
|
|
|
data += nal_len;
|
|
size -= nal_len;
|
|
}
|
|
} 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_data = data;
|
|
nal_queue = rtph264pay->queue;
|
|
|
|
/* 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 %u",
|
|
next, size);
|
|
|
|
/* first pass to locate NALs and parse SPS/PPS */
|
|
while (size > 4) {
|
|
/* skip start code */
|
|
data += 3;
|
|
size -= 3;
|
|
|
|
if (rtph264pay->scan_mode == GST_H264_SCAN_MODE_SINGLE_NAL) {
|
|
/* we are told that there is only a single NAL in this packet so that we
|
|
* can avoid scanning for the next NAL. */
|
|
next = size;
|
|
} else {
|
|
/* 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);
|
|
|
|
if (next == size
|
|
&& rtph264pay->scan_mode == GST_H264_SCAN_MODE_BYTESTREAM) {
|
|
/* Didn't find the start of next NAL, 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_basertppayload_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, timestamp)
|
|
|| 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 */
|
|
data = nal_data;
|
|
pushed = 0;
|
|
|
|
for (i = 0; i < nal_queue->len; i++) {
|
|
guint size;
|
|
|
|
nal_len = g_array_index (nal_queue, guint, i);
|
|
/* skip start code */
|
|
data += 3;
|
|
|
|
/* Trim the end unless we're the last NAL in the buffer.
|
|
* 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;
|
|
if (i + 1 != nal_queue->len
|
|
|| rtph264pay->scan_mode == GST_H264_SCAN_MODE_BYTESTREAM)
|
|
for (; size > 1 && data[size - 1] == 0x0; size--)
|
|
/* skip */ ;
|
|
|
|
/* put the data in one or more RTP packets */
|
|
ret =
|
|
gst_rtp_h264_pay_payload_nal (basepayload, data, size, timestamp,
|
|
buffer);
|
|
if (ret != GST_FLOW_OK) {
|
|
break;
|
|
}
|
|
|
|
/* move to next NAL packet */
|
|
data += nal_len;
|
|
size -= nal_len;
|
|
pushed += nal_len + 3;
|
|
}
|
|
g_array_set_size (nal_queue, 0);
|
|
}
|
|
|
|
if (rtph264pay->scan_mode == GST_H264_SCAN_MODE_BYTESTREAM)
|
|
gst_adapter_flush (rtph264pay->adapter, pushed);
|
|
else
|
|
gst_buffer_unref (buffer);
|
|
|
|
return ret;
|
|
|
|
caps_rejected:
|
|
{
|
|
GST_WARNING_OBJECT (basepayload, "Could not set outcaps");
|
|
g_array_set_size (nal_queue, 0);
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_h264_pay_handle_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
const GstStructure *s;
|
|
GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (GST_PAD_PARENT (pad));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_adapter_clear (rtph264pay->adapter);
|
|
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;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_basertppayload_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);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
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_PROFILE_LEVEL_ID:
|
|
break;
|
|
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_SCAN_MODE:
|
|
rtph264pay->scan_mode = g_value_get_enum (value);
|
|
break;
|
|
case PROP_BUFFER_LIST:
|
|
rtph264pay->buffer_list = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_CONFIG_INTERVAL:
|
|
rtph264pay->spspps_interval = g_value_get_uint (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_PROFILE_LEVEL_ID:
|
|
break;
|
|
case PROP_SPROP_PARAMETER_SETS:
|
|
g_value_set_string (value, rtph264pay->sprop_parameter_sets);
|
|
break;
|
|
case PROP_SCAN_MODE:
|
|
g_value_set_enum (value, rtph264pay->scan_mode);
|
|
break;
|
|
case PROP_BUFFER_LIST:
|
|
g_value_set_boolean (value, rtph264pay->buffer_list);
|
|
break;
|
|
case PROP_CONFIG_INTERVAL:
|
|
g_value_set_uint (value, rtph264pay->spspps_interval);
|
|
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);
|
|
}
|