/* GStreamer * Copyright (C) <2005> Wim Taymans * * 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 "gstrtpelements.h" #include "gstrtpmp4gdepay.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpmp4gdepay_debug); #define GST_CAT_DEFAULT (rtpmp4gdepay_debug) static GstStaticPadTemplate gst_rtp_mp4g_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/mpeg," "mpegversion=(int) 4," "systemstream=(boolean)false;" "audio/mpeg," "mpegversion=(int) 4, " "stream-format=(string)raw") ); static GstStaticPadTemplate gst_rtp_mp4g_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) { \"video\", \"audio\", \"application\" }, " "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MPEG4-GENERIC\", " /* required string params */ /* "streamtype = (string) { \"4\", \"5\" }, " Not set by Wowza 4 = video, 5 = audio */ /* "profile-level-id = (string) [1,MAX], " */ /* "config = (string) [1,MAX]" */ "mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\", \"aac-hbr\" } " /* Optional general parameters */ /* "objecttype = (string) [1,MAX], " */ /* "constantsize = (string) [1,MAX], " *//* constant size of each AU */ /* "constantduration = (string) [1,MAX], " *//* constant duration of each AU */ /* "maxdisplacement = (string) [1,MAX], " */ /* "de-interleavebuffersize = (string) [1,MAX], " */ /* Optional configuration parameters */ /* "sizelength = (string) [1, 32], " */ /* "indexlength = (string) [1, 32], " */ /* "indexdeltalength = (string) [1, 32], " */ /* "ctsdeltalength = (string) [1, 32], " */ /* "dtsdeltalength = (string) [1, 32], " */ /* "randomaccessindication = (string) {0, 1}, " */ /* "streamstateindication = (string) [0, 32], " */ /* "auxiliarydatasizelength = (string) [0, 32]" */ ) ); /* simple bitstream parser */ typedef struct { const guint8 *data; const guint8 *end; gint head; /* bitpos in the cache of next bit */ guint64 cache; /* cached bytes */ } GstBsParse; static void gst_bs_parse_init (GstBsParse * bs, const guint8 * data, guint size) { bs->data = data; bs->end = data + size; bs->head = 0; bs->cache = 0xffffffff; } static guint32 gst_bs_parse_read (GstBsParse * bs, guint n) { guint32 res = 0; gint shift; if (n == 0) return res; /* fill up the cache if we need to */ while (bs->head < n) { if (bs->data >= bs->end) { /* we're at the end, can't produce more than head number of bits */ n = bs->head; break; } /* shift bytes in cache, moving the head bits of the cache left */ bs->cache = (bs->cache << 8) | *bs->data++; bs->head += 8; } /* bring the required bits down and truncate */ if ((shift = bs->head - n) > 0) res = bs->cache >> shift; else res = bs->cache; /* mask out required bits */ if (n < 32) res &= (1 << n) - 1; bs->head = shift; return res; } #define gst_rtp_mp4g_depay_parent_class parent_class G_DEFINE_TYPE (GstRtpMP4GDepay, gst_rtp_mp4g_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmp4gdepay, "rtpmp4gdepay", GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_DEPAY, rtp_element_init (plugin)); static void gst_rtp_mp4g_depay_finalize (GObject * object); static gboolean gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps); static GstBuffer *gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp); static gboolean gst_rtp_mp4g_depay_handle_event (GstRTPBaseDepayload * filter, GstEvent * event); static GstStateChangeReturn gst_rtp_mp4g_depay_change_state (GstElement * element, GstStateChange transition); static void gst_rtp_mp4g_depay_class_init (GstRtpMP4GDepayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass; gobject_class->finalize = gst_rtp_mp4g_depay_finalize; gstelement_class->change_state = gst_rtp_mp4g_depay_change_state; gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mp4g_depay_process; gstrtpbasedepayload_class->set_caps = gst_rtp_mp4g_depay_setcaps; gstrtpbasedepayload_class->handle_event = gst_rtp_mp4g_depay_handle_event; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_mp4g_depay_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_mp4g_depay_sink_template); gst_element_class_set_static_metadata (gstelement_class, "RTP MPEG4 ES depayloader", "Codec/Depayloader/Network/RTP", "Extracts MPEG4 elementary streams from RTP packets (RFC 3640)", "Wim Taymans "); GST_DEBUG_CATEGORY_INIT (rtpmp4gdepay_debug, "rtpmp4gdepay", 0, "MP4-generic RTP Depayloader"); } static void gst_rtp_mp4g_depay_init (GstRtpMP4GDepay * rtpmp4gdepay) { gst_rtp_base_depayload_set_aggregate_hdrext_enabled (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), TRUE); rtpmp4gdepay->adapter = gst_adapter_new (); rtpmp4gdepay->packets = g_queue_new (); } static void gst_rtp_mp4g_depay_finalize (GObject * object) { GstRtpMP4GDepay *rtpmp4gdepay; rtpmp4gdepay = GST_RTP_MP4G_DEPAY (object); g_object_unref (rtpmp4gdepay->adapter); rtpmp4gdepay->adapter = NULL; g_queue_free (rtpmp4gdepay->packets); rtpmp4gdepay->packets = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } static gint gst_rtp_mp4g_depay_parse_int (GstStructure * structure, const gchar * field, gint def) { const gchar *str; gint res; if ((str = gst_structure_get_string (structure, field))) return atoi (str); if (gst_structure_get_int (structure, field, &res)) return res; return def; } static gboolean gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps) { GstStructure *structure; GstRtpMP4GDepay *rtpmp4gdepay; GstCaps *srccaps = NULL; const gchar *str; gint clock_rate; gint someint; gboolean res; rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload); structure = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (structure, "clock-rate", &clock_rate)) clock_rate = 90000; /* default */ depayload->clock_rate = clock_rate; rtpmp4gdepay->check_adts = FALSE; if ((str = gst_structure_get_string (structure, "media"))) { if (strcmp (str, "audio") == 0) { srccaps = gst_caps_new_simple ("audio/mpeg", "mpegversion", G_TYPE_INT, 4, "stream-format", G_TYPE_STRING, "raw", NULL); rtpmp4gdepay->check_adts = TRUE; rtpmp4gdepay->warn_adts = TRUE; } else if (strcmp (str, "video") == 0) { srccaps = gst_caps_new_simple ("video/mpeg", "mpegversion", G_TYPE_INT, 4, "systemstream", G_TYPE_BOOLEAN, FALSE, NULL); } } if (srccaps == NULL) goto unknown_media; /* these values are optional and have a default value of 0 (no header) */ rtpmp4gdepay->sizelength = gst_rtp_mp4g_depay_parse_int (structure, "sizelength", 0); rtpmp4gdepay->indexlength = gst_rtp_mp4g_depay_parse_int (structure, "indexlength", 0); rtpmp4gdepay->indexdeltalength = gst_rtp_mp4g_depay_parse_int (structure, "indexdeltalength", 0); rtpmp4gdepay->ctsdeltalength = gst_rtp_mp4g_depay_parse_int (structure, "ctsdeltalength", 0); rtpmp4gdepay->dtsdeltalength = gst_rtp_mp4g_depay_parse_int (structure, "dtsdeltalength", 0); someint = gst_rtp_mp4g_depay_parse_int (structure, "randomaccessindication", 0); rtpmp4gdepay->randomaccessindication = someint > 0 ? 1 : 0; rtpmp4gdepay->streamstateindication = gst_rtp_mp4g_depay_parse_int (structure, "streamstateindication", 0); rtpmp4gdepay->auxiliarydatasizelength = gst_rtp_mp4g_depay_parse_int (structure, "auxiliarydatasizelength", 0); rtpmp4gdepay->constantSize = gst_rtp_mp4g_depay_parse_int (structure, "constantsize", 0); rtpmp4gdepay->constantDuration = gst_rtp_mp4g_depay_parse_int (structure, "constantduration", 0); rtpmp4gdepay->maxDisplacement = gst_rtp_mp4g_depay_parse_int (structure, "maxdisplacement", 0); /* get config string */ if ((str = gst_structure_get_string (structure, "config"))) { GValue v = { 0 }; g_value_init (&v, GST_TYPE_BUFFER); if (gst_value_deserialize (&v, str)) { GstBuffer *buffer; buffer = gst_value_get_buffer (&v); gst_caps_set_simple (srccaps, "codec_data", GST_TYPE_BUFFER, buffer, NULL); g_value_unset (&v); } else { g_warning ("cannot convert config to buffer"); } } res = gst_pad_set_caps (depayload->srcpad, srccaps); gst_caps_unref (srccaps); return res; /* ERRORS */ unknown_media: { GST_DEBUG_OBJECT (rtpmp4gdepay, "Unknown media type"); return FALSE; } } static void gst_rtp_mp4g_depay_clear_queue (GstRtpMP4GDepay * rtpmp4gdepay) { GstBuffer *outbuf; while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets))) gst_buffer_unref (outbuf); } static void gst_rtp_mp4g_depay_reset (GstRtpMP4GDepay * rtpmp4gdepay) { gst_adapter_clear (rtpmp4gdepay->adapter); rtpmp4gdepay->max_AU_index = -1; rtpmp4gdepay->next_AU_index = -1; rtpmp4gdepay->prev_AU_index = -1; rtpmp4gdepay->prev_rtptime = -1; rtpmp4gdepay->last_AU_index = -1; gst_rtp_mp4g_depay_clear_queue (rtpmp4gdepay); } static void gst_rtp_mp4g_depay_push_outbuf (GstRtpMP4GDepay * rtpmp4gdepay, GstBuffer * outbuf, guint AU_index) { gboolean discont = FALSE; if (AU_index != rtpmp4gdepay->next_AU_index) { GST_DEBUG_OBJECT (rtpmp4gdepay, "discont, expected AU_index %u", rtpmp4gdepay->next_AU_index); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); discont = TRUE; } GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing %sAU_index %u", discont ? "" : "expected ", AU_index); gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpmp4gdepay), outbuf, 0); if (!rtpmp4gdepay->outbufs) { rtpmp4gdepay->outbufs = gst_buffer_list_new_sized (g_queue_get_length (rtpmp4gdepay->packets)); } gst_buffer_list_add (rtpmp4gdepay->outbufs, outbuf); rtpmp4gdepay->next_AU_index = AU_index + 1; } static void gst_rtp_mp4g_depay_flush_queue (GstRtpMP4GDepay * rtpmp4gdepay) { GstBuffer *outbuf; guint AU_index; while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets))) { AU_index = GST_BUFFER_OFFSET (outbuf); GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index); gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index); } } static void gst_rtp_mp4g_depay_queue (GstRtpMP4GDepay * rtpmp4gdepay, GstBuffer * outbuf) { guint AU_index = GST_BUFFER_OFFSET (outbuf); if (rtpmp4gdepay->next_AU_index == -1) { GST_DEBUG_OBJECT (rtpmp4gdepay, "Init AU counter %u", AU_index); rtpmp4gdepay->next_AU_index = AU_index; } if (rtpmp4gdepay->next_AU_index == AU_index) { GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u", AU_index); /* we received the expected packet, push it and flush as much as we can from * the queue */ gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index); while ((outbuf = g_queue_peek_head (rtpmp4gdepay->packets))) { AU_index = GST_BUFFER_OFFSET (outbuf); GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index); if (rtpmp4gdepay->next_AU_index == AU_index) { outbuf = g_queue_pop_head (rtpmp4gdepay->packets); gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index); } else { GST_DEBUG_OBJECT (rtpmp4gdepay, "waiting for next AU_index %u", rtpmp4gdepay->next_AU_index); break; } } } else { GList *list; GST_DEBUG_OBJECT (rtpmp4gdepay, "queueing AU_index %u", AU_index); /* loop the list to skip strictly smaller AU_index buffers */ for (list = rtpmp4gdepay->packets->head; list; list = g_list_next (list)) { guint idx; gint gap; idx = GST_BUFFER_OFFSET (GST_BUFFER_CAST (list->data)); /* compare the new seqnum to the one in the buffer */ gap = (gint) (idx - AU_index); GST_DEBUG_OBJECT (rtpmp4gdepay, "compare with AU_index %u, gap %d", idx, gap); /* AU_index <= idx, we can stop looking */ if (G_LIKELY (gap > 0)) break; } if (G_LIKELY (list)) g_queue_insert_before (rtpmp4gdepay->packets, list, outbuf); else g_queue_push_tail (rtpmp4gdepay->packets, outbuf); } } static GstBuffer * gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp) { GstRtpMP4GDepay *rtpmp4gdepay; GstBuffer *outbuf = NULL; GstClockTime timestamp; rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload); /* flush remaining data on discont */ if (GST_BUFFER_IS_DISCONT (rtp->buffer)) { GST_DEBUG_OBJECT (rtpmp4gdepay, "received DISCONT"); gst_adapter_clear (rtpmp4gdepay->adapter); } timestamp = GST_BUFFER_PTS (rtp->buffer); { gint payload_len, payload_AU; guint8 *payload; guint32 rtptime; guint AU_headers_len; guint AU_size, AU_index, AU_index_delta, payload_AU_size; gboolean M; payload_len = gst_rtp_buffer_get_payload_len (rtp); payload = gst_rtp_buffer_get_payload (rtp); GST_DEBUG_OBJECT (rtpmp4gdepay, "received payload of %d", payload_len); rtptime = gst_rtp_buffer_get_timestamp (rtp); M = gst_rtp_buffer_get_marker (rtp); if (rtpmp4gdepay->sizelength > 0) { gint num_AU_headers, AU_headers_bytes, i; GstBsParse bs; if (payload_len < 2) goto short_payload; /* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+ * |AU-headers-length|AU-header|AU-header| |AU-header|padding| * | | (1) | (2) | | (n) * | bits | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+ * * The length is 2 bytes and contains the length of the following * AU-headers in bits. */ AU_headers_len = (payload[0] << 8) | payload[1]; AU_headers_bytes = (AU_headers_len + 7) / 8; num_AU_headers = AU_headers_len / 16; GST_DEBUG_OBJECT (rtpmp4gdepay, "AU headers len %d, bytes %d, num %d", AU_headers_len, AU_headers_bytes, num_AU_headers); /* skip header */ payload += 2; payload_len -= 2; if (payload_len < AU_headers_bytes) goto short_payload; /* skip special headers, point to first payload AU */ payload_AU = 2 + AU_headers_bytes; payload_AU_size = payload_len - AU_headers_bytes; if (G_UNLIKELY (rtpmp4gdepay->auxiliarydatasizelength)) { gint aux_size; /* point the bitstream parser to the first auxiliary data bit */ gst_bs_parse_init (&bs, payload + AU_headers_bytes, payload_len - AU_headers_bytes); aux_size = gst_bs_parse_read (&bs, rtpmp4gdepay->auxiliarydatasizelength); /* convert to bytes */ aux_size = (aux_size + 7) / 8; /* AU data then follows auxiliary data */ if (payload_AU_size < aux_size) goto short_payload; payload_AU += aux_size; payload_AU_size -= aux_size; } /* point the bitstream parser to the first AU header bit */ gst_bs_parse_init (&bs, payload, payload_len); AU_index = AU_index_delta = 0; for (i = 0; i < num_AU_headers && payload_AU_size > 0; i++) { /* parse AU header * +---------------------------------------+ * | AU-size | * +---------------------------------------+ * | AU-Index / AU-Index-delta | * +---------------------------------------+ * | CTS-flag | * +---------------------------------------+ * | CTS-delta | * +---------------------------------------+ * | DTS-flag | * +---------------------------------------+ * | DTS-delta | * +---------------------------------------+ * | RAP-flag | * +---------------------------------------+ * | Stream-state | * +---------------------------------------+ */ AU_size = gst_bs_parse_read (&bs, rtpmp4gdepay->sizelength); /* calculate the AU_index, which is only on the first AU of the packet * and the AU_index_delta on the other AUs. This will be used to * reconstruct the AU ordering when interleaving. */ if (i == 0) { AU_index = gst_bs_parse_read (&bs, rtpmp4gdepay->indexlength); GST_DEBUG_OBJECT (rtpmp4gdepay, "AU index %u", AU_index); if (AU_index == 0 && rtpmp4gdepay->prev_AU_index == 0) { gint diff; gint cd; /* if we see two consecutive packets with AU_index of 0, we can * assume we have constantDuration packets. Since we don't have * the index we must use the AU duration to calculate the * index. Get the diff between the timestamps first, this can be * positive or negative. */ if (rtpmp4gdepay->prev_rtptime <= rtptime) diff = rtptime - rtpmp4gdepay->prev_rtptime; else diff = -(rtpmp4gdepay->prev_rtptime - rtptime); /* if no constantDuration was given, make one */ if (rtpmp4gdepay->constantDuration != 0) { cd = rtpmp4gdepay->constantDuration; GST_DEBUG_OBJECT (depayload, "using constantDuration %d", cd); } else if (rtpmp4gdepay->prev_AU_num > 0) { /* use number of packets and of previous frame */ cd = diff / rtpmp4gdepay->prev_AU_num; GST_DEBUG_OBJECT (depayload, "guessing constantDuration %d", cd); if (!GST_BUFFER_IS_DISCONT (rtp->buffer)) { /* rfc3640 - 3.2.3.2 * if we see two consecutive packets with AU_index of 0 and * there has been no discontinuity, we must conclude that this * value of constantDuration is correct from now on. */ GST_DEBUG_OBJECT (depayload, "constantDuration of %d detected", cd); rtpmp4gdepay->constantDuration = cd; } } else { /* assume this frame has the same number of packets as the * previous one */ cd = diff / num_AU_headers; GST_DEBUG_OBJECT (depayload, "guessing constantDuration %d", cd); } if (cd > 0) { /* get the number of packets by dividing with the duration */ diff /= cd; } else { diff = 0; } rtpmp4gdepay->last_AU_index += diff; rtpmp4gdepay->prev_AU_index = AU_index; AU_index = rtpmp4gdepay->last_AU_index; GST_DEBUG_OBJECT (rtpmp4gdepay, "diff %d, AU index %u", diff, AU_index); } else { rtpmp4gdepay->prev_AU_index = AU_index; rtpmp4gdepay->last_AU_index = AU_index; } /* keep track of the highest AU_index */ if (rtpmp4gdepay->max_AU_index != -1 && rtpmp4gdepay->max_AU_index <= AU_index) { GST_DEBUG_OBJECT (rtpmp4gdepay, "new interleave group, flushing"); /* a new interleave group started, flush */ gst_rtp_mp4g_depay_flush_queue (rtpmp4gdepay); } if (G_UNLIKELY (!rtpmp4gdepay->maxDisplacement && rtpmp4gdepay->max_AU_index != -1 && rtpmp4gdepay->max_AU_index >= AU_index)) { GstBuffer *outbuf; /* some broken non-interleaved streams have AU-index jumping around * all over the place, apparently assuming receiver disregards */ GST_DEBUG_OBJECT (rtpmp4gdepay, "non-interleaved broken AU indices;" " forcing continuous flush"); /* reset AU to avoid repeated DISCONT in such case */ outbuf = g_queue_peek_head (rtpmp4gdepay->packets); if (G_LIKELY (outbuf)) { rtpmp4gdepay->next_AU_index = GST_BUFFER_OFFSET (outbuf); gst_rtp_mp4g_depay_flush_queue (rtpmp4gdepay); } /* rebase next_AU_index to current rtp's first AU_index */ rtpmp4gdepay->next_AU_index = AU_index; } rtpmp4gdepay->prev_rtptime = rtptime; rtpmp4gdepay->prev_AU_num = num_AU_headers; } else { AU_index_delta = gst_bs_parse_read (&bs, rtpmp4gdepay->indexdeltalength); AU_index += AU_index_delta + 1; } /* keep track of highest AU_index */ if (rtpmp4gdepay->max_AU_index == -1 || AU_index > rtpmp4gdepay->max_AU_index) rtpmp4gdepay->max_AU_index = AU_index; /* the presentation time offset, a 2s-complement value, we need this to * calculate the timestamp on the output packet. */ if (rtpmp4gdepay->ctsdeltalength > 0) { if (gst_bs_parse_read (&bs, 1)) gst_bs_parse_read (&bs, rtpmp4gdepay->ctsdeltalength); } /* the decoding time offset, a 2s-complement value */ if (rtpmp4gdepay->dtsdeltalength > 0) { if (gst_bs_parse_read (&bs, 1)) gst_bs_parse_read (&bs, rtpmp4gdepay->dtsdeltalength); } /* RAP-flag to indicate that the AU contains a keyframe */ if (rtpmp4gdepay->randomaccessindication) gst_bs_parse_read (&bs, 1); /* stream-state */ if (rtpmp4gdepay->streamstateindication > 0) gst_bs_parse_read (&bs, rtpmp4gdepay->streamstateindication); GST_DEBUG_OBJECT (rtpmp4gdepay, "size %d, index %d, delta %d", AU_size, AU_index, AU_index_delta); /* fragmented pakets have the AU_size set to the size of the * unfragmented AU. */ if (AU_size > payload_AU_size) AU_size = payload_AU_size; /* collect stuff in the adapter, strip header from payload and push in * the adapter */ outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, payload_AU, AU_size); gst_adapter_push (rtpmp4gdepay->adapter, outbuf); if (M) { guint32 v = 0; guint avail; /* packet is complete, flush */ avail = gst_adapter_available (rtpmp4gdepay->adapter); /* Some broken senders send ADTS headers (e.g. some Sony cameras). * Try to detect those and skip them (still needs config set), but * don't check every frame, only the first (unless we detect ADTS) */ if (rtpmp4gdepay->check_adts && avail >= 7) { if (gst_adapter_masked_scan_uint32_peek (rtpmp4gdepay->adapter, 0xfffe0000, 0xfff00000, 0, 4, &v) == 0) { guint adts_hdr_len = (((v >> 16) & 0x1) == 0) ? 9 : 7; if (avail > adts_hdr_len) { if (rtpmp4gdepay->warn_adts) { GST_WARNING_OBJECT (rtpmp4gdepay, "Detected ADTS header of " "%u bytes, skipping", adts_hdr_len); rtpmp4gdepay->warn_adts = FALSE; } gst_adapter_flush (rtpmp4gdepay->adapter, adts_hdr_len); avail -= adts_hdr_len; } } else { rtpmp4gdepay->check_adts = FALSE; rtpmp4gdepay->warn_adts = TRUE; } } outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail); /* copy some of the fields we calculated above on the buffer. We also * copy the AU_index so that we can sort the packets in our queue. */ GST_BUFFER_PTS (outbuf) = timestamp; GST_BUFFER_OFFSET (outbuf) = AU_index; if (rtpmp4gdepay->constantDuration != 0) { /* if we have constantDuration, calculate timestamp for next AU * in this RTP packet. */ timestamp += (rtpmp4gdepay->constantDuration * GST_SECOND) / depayload->clock_rate; /* Set the duration for the outgoing buffer */ GST_BUFFER_DURATION (outbuf) = timestamp - GST_BUFFER_PTS (outbuf); } else { /* otherwise, make sure we don't use the timestamp again for other * AUs. */ timestamp = GST_CLOCK_TIME_NONE; } GST_DEBUG_OBJECT (depayload, "pushing buffer of size %" G_GSIZE_FORMAT, gst_buffer_get_size (outbuf)); gst_rtp_mp4g_depay_queue (rtpmp4gdepay, outbuf); } payload_AU += AU_size; payload_AU_size -= AU_size; } if (rtpmp4gdepay->outbufs) { gst_rtp_base_depayload_push_list (depayload, g_steal_pointer (&rtpmp4gdepay->outbufs)); } } else { /* push complete buffer in adapter */ outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 0, payload_len); gst_adapter_push (rtpmp4gdepay->adapter, outbuf); /* if this was the last packet of the VOP, create and push a buffer */ if (M) { guint avail; avail = gst_adapter_available (rtpmp4gdepay->adapter); outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail); GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %" G_GSIZE_FORMAT, gst_buffer_get_size (outbuf)); return outbuf; } } } return NULL; /* ERRORS */ short_payload: { GST_ELEMENT_WARNING (rtpmp4gdepay, STREAM, DECODE, ("Packet payload was too short."), (NULL)); gst_rtp_base_depayload_dropped (depayload); return NULL; } } static gboolean gst_rtp_mp4g_depay_handle_event (GstRTPBaseDepayload * filter, GstEvent * event) { gboolean ret; GstRtpMP4GDepay *rtpmp4gdepay; rtpmp4gdepay = GST_RTP_MP4G_DEPAY (filter); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_STOP: gst_rtp_mp4g_depay_reset (rtpmp4gdepay); break; default: break; } ret = GST_RTP_BASE_DEPAYLOAD_CLASS (parent_class)->handle_event (filter, event); return ret; } static GstStateChangeReturn gst_rtp_mp4g_depay_change_state (GstElement * element, GstStateChange transition) { GstRtpMP4GDepay *rtpmp4gdepay; GstStateChangeReturn ret; rtpmp4gdepay = GST_RTP_MP4G_DEPAY (element); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: gst_rtp_mp4g_depay_reset (rtpmp4gdepay); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_rtp_mp4g_depay_reset (rtpmp4gdepay); break; default: break; } return ret; }