/* ex: set tabstop=2 shiftwidth=2 expandtab: */ /* GStreamer * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #include /* Included to not duplicate gst_rtp_h264_add_sps_pps () */ #include "gstrtph264depay.h" #include "gstrtph264pay.h" #include "gstrtputils.h" #include "gstbuffermemory.h" #define IDR_TYPE_ID 5 #define SPS_TYPE_ID 7 #define PPS_TYPE_ID 8 #define AUD_TYPE_ID 9 #define STAP_A_TYPE_ID 24 #define FU_A_TYPE_ID 28 GST_DEBUG_CATEGORY_STATIC (rtph264pay_debug); #define GST_CAT_DEFAULT (rtph264pay_debug) #define GST_TYPE_RTP_H264_AGGREGATE_MODE \ (gst_rtp_h264_aggregate_mode_get_type ()) static GType gst_rtp_h264_aggregate_mode_get_type (void) { static GType type = 0; static const GEnumValue values[] = { {GST_RTP_H264_AGGREGATE_NONE, "Do not aggregate NAL units", "none"}, {GST_RTP_H264_AGGREGATE_ZERO_LATENCY, "Aggregate NAL units until a VCL unit is included", "zero-latency"}, {GST_RTP_H264_AGGREGATE_MAX_STAP, "Aggregate all NAL units with the same timestamp (adds one frame of" " latency)", "max-stap"}, {0, NULL, NULL}, }; if (!type) { type = g_enum_register_static ("GstRtpH264AggregateMode", values); } return type; } /* references: * * RFC 3984 */ static GstStaticPadTemplate gst_rtp_h264_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/x-h264, " "stream-format = (string) avc, alignment = (string) au;" "video/x-h264, " "stream-format = (string) byte-stream, alignment = (string) { nal, au }") ); static GstStaticPadTemplate gst_rtp_h264_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"video\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 90000, " "encoding-name = (string) \"H264\"") ); #define DEFAULT_SPROP_PARAMETER_SETS NULL #define DEFAULT_CONFIG_INTERVAL 0 #define DEFAULT_AGGREGATE_MODE GST_RTP_H264_AGGREGATE_NONE enum { PROP_0, PROP_SPROP_PARAMETER_SETS, PROP_CONFIG_INTERVAL, PROP_AGGREGATE_MODE, }; static void gst_rtp_h264_pay_finalize (GObject * object); static void gst_rtp_h264_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_h264_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstCaps *gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad, GstCaps * filter); static gboolean gst_rtp_h264_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps); static GstFlowReturn gst_rtp_h264_pay_handle_buffer (GstRTPBasePayload * pad, GstBuffer * buffer); static gboolean gst_rtp_h264_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event); static GstStateChangeReturn gst_rtp_h264_pay_change_state (GstElement * element, GstStateChange transition); static gboolean gst_rtp_h264_pay_src_query (GstPad * pad, GstObject * parent, GstQuery * query); static void gst_rtp_h264_pay_reset_bundle (GstRtpH264Pay * rtph264pay); #define gst_rtp_h264_pay_parent_class parent_class G_DEFINE_TYPE (GstRtpH264Pay, gst_rtp_h264_pay, GST_TYPE_RTP_BASE_PAYLOAD); static void gst_rtp_h264_pay_class_init (GstRtpH264PayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstRTPBasePayloadClass *gstrtpbasepayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gobject_class->set_property = gst_rtp_h264_pay_set_property; gobject_class->get_property = gst_rtp_h264_pay_get_property; g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SPROP_PARAMETER_SETS, g_param_spec_string ("sprop-parameter-sets", "sprop-parameter-sets", "The base64 sprop-parameter-sets to set in out caps (set to NULL to " "extract from stream)", DEFAULT_SPROP_PARAMETER_SETS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CONFIG_INTERVAL, g_param_spec_int ("config-interval", "SPS PPS Send Interval", "Send SPS and PPS Insertion Interval in seconds (sprop parameter sets " "will be multiplexed in the data stream when detected.) " "(0 = disabled, -1 = send with every IDR frame)", -1, 3600, DEFAULT_CONFIG_INTERVAL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) ); /** * GstRtpH264Pay:aggregate-mode * * Bundle suitable SPS/PPS NAL units into STAP-A aggregate packets. * * This can potentially reduce RTP packetization overhead but not all * RTP implementations handle it correctly. * * For best compatibility, it is recommended to set this to "none" (the * default) for RTSP and for WebRTC to "zero-latency". * * Since: 1.18 */ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_AGGREGATE_MODE, g_param_spec_enum ("aggregate-mode", "Attempt to use aggregate packets", "Bundle suitable SPS/PPS NAL units into STAP-A " "aggregate packets", GST_TYPE_RTP_H264_AGGREGATE_MODE, DEFAULT_AGGREGATE_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) ); gobject_class->finalize = gst_rtp_h264_pay_finalize; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_h264_pay_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_h264_pay_sink_template); gst_element_class_set_static_metadata (gstelement_class, "RTP H264 payloader", "Codec/Payloader/Network/RTP", "Payload-encode H264 video into RTP packets (RFC 3984)", "Laurent Glayal "); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_h264_pay_change_state); gstrtpbasepayload_class->get_caps = gst_rtp_h264_pay_getcaps; gstrtpbasepayload_class->set_caps = gst_rtp_h264_pay_setcaps; gstrtpbasepayload_class->handle_buffer = gst_rtp_h264_pay_handle_buffer; gstrtpbasepayload_class->sink_event = gst_rtp_h264_pay_sink_event; GST_DEBUG_CATEGORY_INIT (rtph264pay_debug, "rtph264pay", 0, "H264 RTP Payloader"); gst_type_mark_as_plugin_api (GST_TYPE_RTP_H264_AGGREGATE_MODE, 0); } static void gst_rtp_h264_pay_init (GstRtpH264Pay * rtph264pay) { rtph264pay->queue = g_array_new (FALSE, FALSE, sizeof (guint)); rtph264pay->profile = 0; rtph264pay->sps = g_ptr_array_new_with_free_func ( (GDestroyNotify) gst_buffer_unref); rtph264pay->pps = g_ptr_array_new_with_free_func ( (GDestroyNotify) gst_buffer_unref); rtph264pay->last_spspps = -1; rtph264pay->spspps_interval = DEFAULT_CONFIG_INTERVAL; rtph264pay->aggregate_mode = DEFAULT_AGGREGATE_MODE; rtph264pay->delta_unit = FALSE; rtph264pay->discont = FALSE; rtph264pay->adapter = gst_adapter_new (); gst_pad_set_query_function (GST_RTP_BASE_PAYLOAD_SRCPAD (rtph264pay), gst_rtp_h264_pay_src_query); } static void gst_rtp_h264_pay_clear_sps_pps (GstRtpH264Pay * rtph264pay) { g_ptr_array_set_size (rtph264pay->sps, 0); g_ptr_array_set_size (rtph264pay->pps, 0); } static void gst_rtp_h264_pay_finalize (GObject * object) { GstRtpH264Pay *rtph264pay; rtph264pay = GST_RTP_H264_PAY (object); g_array_free (rtph264pay->queue, TRUE); g_ptr_array_free (rtph264pay->sps, TRUE); g_ptr_array_free (rtph264pay->pps, TRUE); g_free (rtph264pay->sprop_parameter_sets); g_object_unref (rtph264pay->adapter); gst_rtp_h264_pay_reset_bundle (rtph264pay); G_OBJECT_CLASS (parent_class)->finalize (object); } static const gchar all_levels[][4] = { "1", "1b", "1.1", "1.2", "1.3", "2", "2.1", "2.2", "3", "3.1", "3.2", "4", "4.1", "4.2", "5", "5.1" }; static GstCaps * gst_rtp_h264_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad, GstCaps * filter) { GstCaps *template_caps; GstCaps *allowed_caps; GstCaps *caps, *icaps; gboolean append_unrestricted; guint i; allowed_caps = gst_pad_peer_query_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), NULL); if (allowed_caps == NULL) return NULL; template_caps = gst_static_pad_template_get_caps (&gst_rtp_h264_pay_sink_template); if (gst_caps_is_any (allowed_caps)) { caps = gst_caps_ref (template_caps); goto done; } if (gst_caps_is_empty (allowed_caps)) { caps = gst_caps_ref (allowed_caps); goto done; } caps = gst_caps_new_empty (); append_unrestricted = FALSE; for (i = 0; i < gst_caps_get_size (allowed_caps); i++) { GstStructure *s = gst_caps_get_structure (allowed_caps, i); GstStructure *new_s = gst_structure_new_empty ("video/x-h264"); const gchar *profile_level_id; profile_level_id = gst_structure_get_string (s, "profile-level-id"); if (profile_level_id && strlen (profile_level_id) == 6) { const gchar *profile; const gchar *level; long int spsint; guint8 sps[3]; spsint = strtol (profile_level_id, NULL, 16); sps[0] = spsint >> 16; sps[1] = spsint >> 8; sps[2] = spsint; profile = gst_codec_utils_h264_get_profile (sps, 3); level = gst_codec_utils_h264_get_level (sps, 3); if (profile && level) { GST_LOG_OBJECT (payload, "In caps, have profile %s and level %s", profile, level); if (!strcmp (profile, "constrained-baseline")) gst_structure_set (new_s, "profile", G_TYPE_STRING, profile, NULL); else { GValue val = { 0, }; GValue profiles = { 0, }; g_value_init (&profiles, GST_TYPE_LIST); g_value_init (&val, G_TYPE_STRING); g_value_set_static_string (&val, profile); gst_value_list_append_value (&profiles, &val); g_value_set_static_string (&val, "constrained-baseline"); gst_value_list_append_value (&profiles, &val); gst_structure_take_value (new_s, "profile", &profiles); } if (!strcmp (level, "1")) gst_structure_set (new_s, "level", G_TYPE_STRING, level, NULL); else { GValue levels = { 0, }; GValue val = { 0, }; int j; g_value_init (&levels, GST_TYPE_LIST); g_value_init (&val, G_TYPE_STRING); for (j = 0; j < G_N_ELEMENTS (all_levels); j++) { g_value_set_static_string (&val, all_levels[j]); gst_value_list_prepend_value (&levels, &val); if (!strcmp (level, all_levels[j])) break; } gst_structure_take_value (new_s, "level", &levels); } } else { /* Invalid profile-level-id means baseline */ gst_structure_set (new_s, "profile", G_TYPE_STRING, "constrained-baseline", NULL); } } else { /* No profile-level-id means baseline or unrestricted */ gst_structure_set (new_s, "profile", G_TYPE_STRING, "constrained-baseline", NULL); append_unrestricted = TRUE; } caps = gst_caps_merge_structure (caps, new_s); } if (append_unrestricted) { caps = gst_caps_merge_structure (caps, gst_structure_new ("video/x-h264", NULL, NULL)); } icaps = gst_caps_intersect (caps, template_caps); gst_caps_unref (caps); caps = icaps; done: if (filter) { GST_DEBUG_OBJECT (payload, "Intersect %" GST_PTR_FORMAT " and filter %" GST_PTR_FORMAT, caps, filter); icaps = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); caps = icaps; } gst_caps_unref (template_caps); gst_caps_unref (allowed_caps); GST_LOG_OBJECT (payload, "returning caps %" GST_PTR_FORMAT, caps); return caps; } static gboolean gst_rtp_h264_pay_src_query (GstPad * pad, GstObject * parent, GstQuery * query) { GstRtpH264Pay *rtph264pay = GST_RTP_H264_PAY (parent); if (GST_QUERY_TYPE (query) == GST_QUERY_LATENCY) { gboolean retval; gboolean live; GstClockTime min_latency, max_latency; retval = gst_pad_query_default (pad, parent, query); if (!retval) return retval; if (rtph264pay->stream_format == GST_H264_STREAM_FORMAT_UNKNOWN || rtph264pay->alignment == GST_H264_ALIGNMENT_UNKNOWN) return FALSE; gst_query_parse_latency (query, &live, &min_latency, &max_latency); if (rtph264pay->aggregate_mode == GST_RTP_H264_AGGREGATE_MAX_STAP && rtph264pay->alignment != GST_H264_ALIGNMENT_AU && rtph264pay->fps_num) { GstClockTime one_frame = gst_util_uint64_scale_int (GST_SECOND, rtph264pay->fps_denum, rtph264pay->fps_num); min_latency += one_frame; max_latency += one_frame; gst_query_set_latency (query, live, min_latency, max_latency); } return TRUE; } return gst_pad_query_default (pad, parent, query); } /* 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_buffer_new_allocate (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_buffer_new_allocate (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); }