| /* GStreamer |
| * Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com> |
| * |
| * 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 <string.h> |
| #include <stdlib.h> |
| #include <gst/rtp/gstrtpbuffer.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\" } " |
| /* 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); |
| |
| 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 <wim.taymans@gmail.com>"); |
| |
| GST_DEBUG_CATEGORY_INIT (rtpmp4gdepay_debug, "rtpmp4gdepay", 0, |
| "MP4-generic RTP Depayloader"); |
| } |
| |
| static void |
| gst_rtp_mp4g_depay_init (GstRtpMP4GDepay * rtpmp4gdepay) |
| { |
| 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; |
| |
| 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); |
| } 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_flush_queue (GstRtpMP4GDepay * rtpmp4gdepay) |
| { |
| GstBuffer *outbuf; |
| gboolean discont = FALSE; |
| 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); |
| |
| if (rtpmp4gdepay->next_AU_index != AU_index) { |
| GST_DEBUG_OBJECT (rtpmp4gdepay, "discont, expected AU_index %u", |
| rtpmp4gdepay->next_AU_index); |
| discont = TRUE; |
| } |
| |
| if (discont) { |
| GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); |
| discont = FALSE; |
| } |
| |
| GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing AU_index %u", AU_index); |
| gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpmp4gdepay), outbuf, 0); |
| gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), outbuf); |
| rtpmp4gdepay->next_AU_index = AU_index + 1; |
| } |
| } |
| |
| 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_drop_meta (GST_ELEMENT_CAST (rtpmp4gdepay), outbuf, 0); |
| gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), outbuf); |
| rtpmp4gdepay->next_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) { |
| GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u", |
| AU_index); |
| outbuf = g_queue_pop_head (rtpmp4gdepay->packets); |
| gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpmp4gdepay), outbuf, 0); |
| gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), |
| outbuf); |
| rtpmp4gdepay->next_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 higest 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) { |
| guint avail; |
| |
| /* packet is complete, flush */ |
| avail = gst_adapter_available (rtpmp4gdepay->adapter); |
| |
| 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; |
| } 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; |
| } |
| } 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)); |
| 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; |
| } |
| |
| gboolean |
| gst_rtp_mp4g_depay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpmp4gdepay", |
| GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_DEPAY); |
| } |
