/* GStreamer * Copyright (C) <2006> 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 <gst/rtp/gstrtpbuffer.h> #include <gst/audio/audio.h> #include "fnv1hash.h" #include "gstrtpvorbispay.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpvorbispay_debug); #define GST_CAT_DEFAULT (rtpvorbispay_debug) /* references: * http://www.rfc-editor.org/rfc/rfc5215.txt */ static GstStaticPadTemplate gst_rtp_vorbis_pay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"VORBIS\"" /* All required parameters * * "encoding-params = (string) <num channels>" * "configuration = (string) ANY" */ ) ); static GstStaticPadTemplate gst_rtp_vorbis_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-vorbis") ); #define DEFAULT_CONFIG_INTERVAL 0 enum { PROP_0, PROP_CONFIG_INTERVAL }; #define gst_rtp_vorbis_pay_parent_class parent_class G_DEFINE_TYPE (GstRtpVorbisPay, gst_rtp_vorbis_pay, GST_TYPE_RTP_BASE_PAYLOAD); static gboolean gst_rtp_vorbis_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps); static GstStateChangeReturn gst_rtp_vorbis_pay_change_state (GstElement * element, GstStateChange transition); static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstRTPBasePayload * pad, GstBuffer * buffer); static gboolean gst_rtp_vorbis_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event); static gboolean gst_rtp_vorbis_pay_parse_id (GstRTPBasePayload * basepayload, guint8 * data, guint size); static gboolean gst_rtp_vorbis_pay_finish_headers (GstRTPBasePayload * basepayload); static void gst_rtp_vorbis_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_vorbis_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstRTPBasePayloadClass *gstrtpbasepayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gstelement_class->change_state = gst_rtp_vorbis_pay_change_state; gstrtpbasepayload_class->set_caps = gst_rtp_vorbis_pay_setcaps; gstrtpbasepayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer; gstrtpbasepayload_class->sink_event = gst_rtp_vorbis_pay_sink_event; gobject_class->set_property = gst_rtp_vorbis_pay_set_property; gobject_class->get_property = gst_rtp_vorbis_pay_get_property; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_vorbis_pay_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_vorbis_pay_sink_template); gst_element_class_set_static_metadata (gstelement_class, "RTP Vorbis payloader", "Codec/Payloader/Network/RTP", "Payload-encode Vorbis audio into RTP packets (RFC 5215)", "Wim Taymans <wim.taymans@gmail.com>"); GST_DEBUG_CATEGORY_INIT (rtpvorbispay_debug, "rtpvorbispay", 0, "Vorbis RTP Payloader"); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CONFIG_INTERVAL, g_param_spec_uint ("config-interval", "Config Send Interval", "Send Config Insertion Interval in seconds (configuration headers " "will be multiplexed in the data stream when detected.) (0 = disabled)", 0, 3600, DEFAULT_CONFIG_INTERVAL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) ); } static void gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay) { rtpvorbispay->last_config = GST_CLOCK_TIME_NONE; } static void gst_rtp_vorbis_pay_clear_packet (GstRtpVorbisPay * rtpvorbispay) { if (rtpvorbispay->packet) gst_buffer_unref (rtpvorbispay->packet); rtpvorbispay->packet = NULL; g_list_free_full (rtpvorbispay->packet_buffers, (GDestroyNotify) gst_buffer_unref); rtpvorbispay->packet_buffers = NULL; } static void gst_rtp_vorbis_pay_cleanup (GstRtpVorbisPay * rtpvorbispay) { gst_rtp_vorbis_pay_clear_packet (rtpvorbispay); g_list_free_full (rtpvorbispay->headers, (GDestroyNotify) gst_buffer_unref); rtpvorbispay->headers = NULL; g_free (rtpvorbispay->config_data); rtpvorbispay->config_data = NULL; rtpvorbispay->last_config = GST_CLOCK_TIME_NONE; } static gboolean gst_rtp_vorbis_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps) { GstRtpVorbisPay *rtpvorbispay; GstStructure *s; const GValue *array; gint asize, i; GstBuffer *buf; GstMapInfo map; rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload); s = gst_caps_get_structure (caps, 0); rtpvorbispay->need_headers = TRUE; if ((array = gst_structure_get_value (s, "streamheader")) == NULL) goto done; if (G_VALUE_TYPE (array) != GST_TYPE_ARRAY) goto done; if ((asize = gst_value_array_get_size (array)) < 3) goto done; for (i = 0; i < asize; i++) { const GValue *value; value = gst_value_array_get_value (array, i); if ((buf = gst_value_get_buffer (value)) == NULL) goto null_buffer; gst_buffer_map (buf, &map, GST_MAP_READ); if (map.size < 1) goto invalid_streamheader; /* no data packets allowed */ if ((map.data[0] & 1) == 0) goto invalid_streamheader; /* we need packets with id 1, 3, 5 */ if (map.data[0] != (i * 2) + 1) goto invalid_streamheader; if (i == 0) { /* identification, we need to parse this in order to get the clock rate. */ if (G_UNLIKELY (!gst_rtp_vorbis_pay_parse_id (basepayload, map.data, map.size))) goto parse_id_failed; } GST_DEBUG_OBJECT (rtpvorbispay, "collecting header %d", i); rtpvorbispay->headers = g_list_append (rtpvorbispay->headers, gst_buffer_ref (buf)); gst_buffer_unmap (buf, &map); } if (!gst_rtp_vorbis_pay_finish_headers (basepayload)) goto finish_failed; done: return TRUE; /* ERRORS */ null_buffer: { GST_WARNING_OBJECT (rtpvorbispay, "streamheader with null buffer received"); return FALSE; } invalid_streamheader: { GST_WARNING_OBJECT (rtpvorbispay, "unable to parse initial header"); gst_buffer_unmap (buf, &map); return FALSE; } parse_id_failed: { GST_WARNING_OBJECT (rtpvorbispay, "unable to parse initial header"); gst_buffer_unmap (buf, &map); return FALSE; } finish_failed: { GST_WARNING_OBJECT (rtpvorbispay, "unable to finish headers"); return FALSE; } } static void gst_rtp_vorbis_pay_reset_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT) { guint payload_len; GstRTPBuffer rtp = { NULL }; GST_LOG_OBJECT (rtpvorbispay, "reset packet"); rtpvorbispay->payload_pos = 4; gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_READ, &rtp); payload_len = gst_rtp_buffer_get_payload_len (&rtp); gst_rtp_buffer_unmap (&rtp); rtpvorbispay->payload_left = payload_len - 4; rtpvorbispay->payload_duration = 0; rtpvorbispay->payload_F = 0; rtpvorbispay->payload_VDT = VDT; rtpvorbispay->payload_pkts = 0; } static void gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay, guint8 VDT, GstClockTime timestamp) { guint len; GST_LOG_OBJECT (rtpvorbispay, "starting new packet, VDT: %d", VDT); gst_rtp_vorbis_pay_clear_packet (rtpvorbispay); /* new packet allocate max packet size */ len = gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (rtpvorbispay), 0, 0); rtpvorbispay->packet = gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (rtpvorbispay), len, 0, 0); gst_rtp_vorbis_pay_reset_packet (rtpvorbispay, VDT); GST_BUFFER_PTS (rtpvorbispay->packet) = timestamp; } static GstFlowReturn gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay) { GstFlowReturn ret; guint8 *payload; guint hlen; GstRTPBuffer rtp = { NULL }; GList *l; /* check for empty packet */ if (!rtpvorbispay->packet || rtpvorbispay->payload_pos <= 4) return GST_FLOW_OK; GST_LOG_OBJECT (rtpvorbispay, "flushing packet"); gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_WRITE, &rtp); /* fix header */ payload = gst_rtp_buffer_get_payload (&rtp); /* * 0 1 2 3 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | Ident | F |VDT|# pkts.| * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * * F: Fragment type (0=none, 1=start, 2=cont, 3=end) * VDT: Vorbis data type (0=vorbis, 1=config, 2=comment, 3=reserved) * pkts: number of packets. */ payload[0] = (rtpvorbispay->payload_ident >> 16) & 0xff; payload[1] = (rtpvorbispay->payload_ident >> 8) & 0xff; payload[2] = (rtpvorbispay->payload_ident) & 0xff; payload[3] = (rtpvorbispay->payload_F & 0x3) << 6 | (rtpvorbispay->payload_VDT & 0x3) << 4 | (rtpvorbispay->payload_pkts & 0xf); gst_rtp_buffer_unmap (&rtp); /* shrink the buffer size to the last written byte */ hlen = gst_rtp_buffer_calc_header_len (0); gst_buffer_resize (rtpvorbispay->packet, 0, hlen + rtpvorbispay->payload_pos); GST_BUFFER_DURATION (rtpvorbispay->packet) = rtpvorbispay->payload_duration; for (l = g_list_last (rtpvorbispay->packet_buffers); l; l = l->prev) { GstBuffer *buf = GST_BUFFER_CAST (l->data); gst_rtp_copy_audio_meta (rtpvorbispay, rtpvorbispay->packet, buf); gst_buffer_unref (buf); } g_list_free (rtpvorbispay->packet_buffers); rtpvorbispay->packet_buffers = NULL; /* push, this gives away our ref to the packet, so clear it. */ ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpvorbispay), rtpvorbispay->packet); rtpvorbispay->packet = NULL; return ret; } static gboolean gst_rtp_vorbis_pay_finish_headers (GstRTPBasePayload * basepayload) { GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload); GList *walk; guint length, size, n_headers, configlen, extralen; gchar *cstr, *configuration; guint8 *data, *config; guint32 ident; gboolean res; GST_DEBUG_OBJECT (rtpvorbispay, "finish headers"); if (!rtpvorbispay->headers) goto no_headers; /* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | Number of packed headers | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | Packed header | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | Packed header | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | .... | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * * We only construct a config containing 1 packed header like this: * * 0 1 2 3 * 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | Ident | length .. * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * .. | n. of headers | length1 | length2 .. * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * .. | Identification Header .. * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * ................................................................. * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * .. | Comment Header .. * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * ................................................................. * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * .. Comment Header | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * | Setup Header .. * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * ................................................................. * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ * .. Setup Header | * +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ */ /* we need 4 bytes for the number of headers (which is always 1), 3 bytes for * the ident, 2 bytes for length, 1 byte for n. of headers. */ size = 4 + 3 + 2 + 1; /* count the size of the headers first and update the hash */ length = 0; n_headers = 0; ident = fnv1_hash_32_new (); extralen = 1; for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) { GstBuffer *buf = GST_BUFFER_CAST (walk->data); GstMapInfo map; guint bsize; bsize = gst_buffer_get_size (buf); length += bsize; n_headers++; /* count number of bytes needed for length fields, we don't need this for * the last header. */ if (g_list_next (walk)) { do { size++; extralen++; bsize >>= 7; } while (bsize); } /* update hash */ gst_buffer_map (buf, &map, GST_MAP_READ); ident = fnv1_hash_32_update (ident, map.data, map.size); gst_buffer_unmap (buf, &map); } /* packet length is header size + packet length */ configlen = size + length; config = data = g_malloc (configlen); /* number of packed headers, we only pack 1 header */ data[0] = 0; data[1] = 0; data[2] = 0; data[3] = 1; ident = fnv1_hash_32_to_24 (ident); rtpvorbispay->payload_ident = ident; GST_DEBUG_OBJECT (rtpvorbispay, "ident 0x%08x", ident); /* take lower 3 bytes */ data[4] = (ident >> 16) & 0xff; data[5] = (ident >> 8) & 0xff; data[6] = ident & 0xff; /* store length of all vorbis headers */ data[7] = ((length) >> 8) & 0xff; data[8] = (length) & 0xff; /* store number of headers minus one. */ data[9] = n_headers - 1; data += 10; /* store length for each header */ for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) { GstBuffer *buf = GST_BUFFER_CAST (walk->data); guint bsize, size, temp; guint flag; /* only need to store the length when it's not the last header */ if (!g_list_next (walk)) break; bsize = gst_buffer_get_size (buf); /* calc size */ size = 0; do { size++; bsize >>= 7; } while (bsize); temp = size; bsize = gst_buffer_get_size (buf); /* write the size backwards */ flag = 0; while (size) { size--; data[size] = (bsize & 0x7f) | flag; bsize >>= 7; flag = 0x80; /* Flag bit on all bytes of the length except the last */ } data += temp; } /* copy header data */ for (walk = rtpvorbispay->headers; walk; walk = g_list_next (walk)) { GstBuffer *buf = GST_BUFFER_CAST (walk->data); gst_buffer_extract (buf, 0, data, gst_buffer_get_size (buf)); data += gst_buffer_get_size (buf); } rtpvorbispay->need_headers = FALSE; /* serialize to base64 */ configuration = g_base64_encode (config, configlen); /* store for later re-sending */ g_free (rtpvorbispay->config_data); rtpvorbispay->config_size = configlen - 4 - 3 - 2; rtpvorbispay->config_data = g_malloc (rtpvorbispay->config_size); rtpvorbispay->config_extra_len = extralen; memcpy (rtpvorbispay->config_data, config + 4 + 3 + 2, rtpvorbispay->config_size); g_free (config); /* configure payloader settings */ cstr = g_strdup_printf ("%d", rtpvorbispay->channels); gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "VORBIS", rtpvorbispay->rate); res = gst_rtp_base_payload_set_outcaps (basepayload, "encoding-params", G_TYPE_STRING, cstr, "configuration", G_TYPE_STRING, configuration, NULL); g_free (cstr); g_free (configuration); return res; /* ERRORS */ no_headers: { GST_DEBUG_OBJECT (rtpvorbispay, "finish headers"); return FALSE; } } static gboolean gst_rtp_vorbis_pay_parse_id (GstRTPBasePayload * basepayload, guint8 * data, guint size) { GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload); guint8 channels; gint32 rate, version; if (G_UNLIKELY (size < 16)) goto too_short; if (G_UNLIKELY (memcmp (data, "\001vorbis", 7))) goto invalid_start; data += 7; if (G_UNLIKELY ((version = GST_READ_UINT32_LE (data)) != 0)) goto invalid_version; data += 4; if (G_UNLIKELY ((channels = *data++) < 1)) goto invalid_channels; if (G_UNLIKELY ((rate = GST_READ_UINT32_LE (data)) < 1)) goto invalid_rate; /* all fine, store the values */ rtpvorbispay->channels = channels; rtpvorbispay->rate = rate; return TRUE; /* ERRORS */ too_short: { GST_ELEMENT_ERROR (basepayload, STREAM, DECODE, ("Identification packet is too short, need at least 16, got %d", size), (NULL)); return FALSE; } invalid_start: { GST_ELEMENT_ERROR (basepayload, STREAM, DECODE, ("Invalid header start in identification packet"), (NULL)); return FALSE; } invalid_version: { GST_ELEMENT_ERROR (basepayload, STREAM, DECODE, ("Invalid version, expected 0, got %d", version), (NULL)); return FALSE; } invalid_rate: { GST_ELEMENT_ERROR (basepayload, STREAM, DECODE, ("Invalid rate %d", rate), (NULL)); return FALSE; } invalid_channels: { GST_ELEMENT_ERROR (basepayload, STREAM, DECODE, ("Invalid channels %d", channels), (NULL)); return FALSE; } } static GstFlowReturn gst_rtp_vorbis_pay_payload_buffer (GstRtpVorbisPay * rtpvorbispay, guint8 VDT, GstBuffer * buffer, guint8 * data, guint size, GstClockTime timestamp, GstClockTime duration, guint not_in_length) { GstFlowReturn ret = GST_FLOW_OK; guint newsize; guint packet_len; GstClockTime newduration; gboolean flush; guint plen; guint8 *ppos, *payload; gboolean fragmented; GstRTPBuffer rtp = { NULL }; /* size increases with packet length and 2 bytes size eader. */ newduration = rtpvorbispay->payload_duration; if (duration != GST_CLOCK_TIME_NONE) newduration += duration; newsize = rtpvorbispay->payload_pos + 2 + size; packet_len = gst_rtp_buffer_calc_packet_len (newsize, 0, 0); /* check buffer filled against length and max latency */ flush = gst_rtp_base_payload_is_filled (GST_RTP_BASE_PAYLOAD (rtpvorbispay), packet_len, newduration); /* we can store up to 15 vorbis packets in one RTP packet. */ flush |= (rtpvorbispay->payload_pkts == 15); /* flush if we have a new VDT */ if (rtpvorbispay->packet) flush |= (rtpvorbispay->payload_VDT != VDT); if (flush) ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay); if (ret != GST_FLOW_OK) goto done; /* create new packet if we must */ if (!rtpvorbispay->packet) { gst_rtp_vorbis_pay_init_packet (rtpvorbispay, VDT, timestamp); } gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_WRITE, &rtp); payload = gst_rtp_buffer_get_payload (&rtp); ppos = payload + rtpvorbispay->payload_pos; fragmented = FALSE; /* put buffer in packet, it either fits completely or needs to be fragmented * over multiple RTP packets. */ do { plen = MIN (rtpvorbispay->payload_left - 2, size); GST_LOG_OBJECT (rtpvorbispay, "append %u bytes", plen); /* data is copied in the payload with a 2 byte length header */ ppos[0] = ((plen - not_in_length) >> 8) & 0xff; ppos[1] = ((plen - not_in_length) & 0xff); if (plen) memcpy (&ppos[2], data, plen); if (buffer) { if (!rtpvorbispay->packet_buffers || rtpvorbispay->packet_buffers->data != (gpointer) buffer) rtpvorbispay->packet_buffers = g_list_prepend (rtpvorbispay->packet_buffers, gst_buffer_ref (buffer)); } else { GList *l; for (l = rtpvorbispay->headers; l; l = l->next) rtpvorbispay->packet_buffers = g_list_prepend (rtpvorbispay->packet_buffers, gst_buffer_ref (l->data)); } /* only first (only) configuration cuts length field */ /* NOTE: spec (if any) is not clear on this ... */ not_in_length = 0; size -= plen; data += plen; rtpvorbispay->payload_pos += plen + 2; rtpvorbispay->payload_left -= plen + 2; if (fragmented) { if (size == 0) /* last fragment, set F to 0x3. */ rtpvorbispay->payload_F = 0x3; else /* fragment continues, set F to 0x2. */ rtpvorbispay->payload_F = 0x2; } else { if (size > 0) { /* fragmented packet starts, set F to 0x1, mark ourselves as * fragmented. */ rtpvorbispay->payload_F = 0x1; fragmented = TRUE; } } if (fragmented) { gst_rtp_buffer_unmap (&rtp); /* fragmented packets are always flushed and have ptks of 0 */ rtpvorbispay->payload_pkts = 0; ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay); if (size > 0) { /* start new packet and get pointers. VDT stays the same. */ gst_rtp_vorbis_pay_init_packet (rtpvorbispay, rtpvorbispay->payload_VDT, timestamp); gst_rtp_buffer_map (rtpvorbispay->packet, GST_MAP_WRITE, &rtp); payload = gst_rtp_buffer_get_payload (&rtp); ppos = payload + rtpvorbispay->payload_pos; } } else { /* unfragmented packet, update stats for next packet, size == 0 and we * exit the while loop */ rtpvorbispay->payload_pkts++; if (duration != GST_CLOCK_TIME_NONE) rtpvorbispay->payload_duration += duration; } } while (size && ret == GST_FLOW_OK); if (rtp.buffer) gst_rtp_buffer_unmap (&rtp); done: return ret; } static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer) { GstRtpVorbisPay *rtpvorbispay; GstFlowReturn ret; GstMapInfo map; gsize size; guint8 *data; GstClockTime duration, timestamp; guint8 VDT; rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload); gst_buffer_map (buffer, &map, GST_MAP_READ); data = map.data; size = map.size; duration = GST_BUFFER_DURATION (buffer); timestamp = GST_BUFFER_PTS (buffer); GST_LOG_OBJECT (rtpvorbispay, "size %" G_GSIZE_FORMAT ", duration %" GST_TIME_FORMAT, size, GST_TIME_ARGS (duration)); if (G_UNLIKELY (size < 1)) goto wrong_size; /* find packet type */ if (data[0] & 1) { /* header */ if (data[0] == 1) { /* identification, we need to parse this in order to get the clock rate. */ if (G_UNLIKELY (!gst_rtp_vorbis_pay_parse_id (basepayload, data, size))) goto parse_id_failed; VDT = 1; } else if (data[0] == 3) { /* comment */ VDT = 2; } else if (data[0] == 5) { /* setup */ VDT = 1; } else goto unknown_header; } else /* data */ VDT = 0; /* we need to collect the headers and construct a config string from them */ if (VDT != 0) { rtpvorbispay->need_headers = TRUE; if (!rtpvorbispay->need_headers && VDT == 1) { GST_INFO_OBJECT (rtpvorbispay, "getting new headers, replace existing"); g_list_free_full (rtpvorbispay->headers, (GDestroyNotify) gst_buffer_unref); rtpvorbispay->headers = NULL; } GST_DEBUG_OBJECT (rtpvorbispay, "collecting header"); /* append header to the list of headers, or replace * if the same type of header was already in there. * * This prevents storing an infinite amount of e.g. comment headers, there * must only be one */ gst_buffer_unmap (buffer, &map); if (rtpvorbispay->headers) { gboolean found = FALSE; GList *l; guint8 new_header_type; gst_buffer_extract (buffer, 0, &new_header_type, 1); for (l = rtpvorbispay->headers; l; l = l->next) { GstBuffer *header = l->data; guint8 header_type; if (gst_buffer_extract (header, 0, &header_type, 1) && header_type == new_header_type) { found = TRUE; gst_buffer_unref (header); l->data = buffer; break; } } if (!found) rtpvorbispay->headers = g_list_append (rtpvorbispay->headers, buffer); } else { rtpvorbispay->headers = g_list_append (rtpvorbispay->headers, buffer); } ret = GST_FLOW_OK; goto done; } else if (rtpvorbispay->headers && rtpvorbispay->need_headers) { if (!gst_rtp_vorbis_pay_finish_headers (basepayload)) goto header_error; } /* there is a config request, see if we need to insert it */ if (rtpvorbispay->config_interval > 0 && rtpvorbispay->config_data) { gboolean send_config = FALSE; GstClockTime running_time = gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME, timestamp); if (rtpvorbispay->last_config != -1) { guint64 diff; GST_LOG_OBJECT (rtpvorbispay, "now %" GST_TIME_FORMAT ", last config %" GST_TIME_FORMAT, GST_TIME_ARGS (running_time), GST_TIME_ARGS (rtpvorbispay->last_config)); /* calculate diff between last config in milliseconds */ if (running_time > rtpvorbispay->last_config) { diff = running_time - rtpvorbispay->last_config; } else { diff = 0; } GST_DEBUG_OBJECT (rtpvorbispay, "interval since last config %" GST_TIME_FORMAT, GST_TIME_ARGS (diff)); /* bigger than interval, queue config */ if (GST_TIME_AS_SECONDS (diff) >= rtpvorbispay->config_interval) { GST_DEBUG_OBJECT (rtpvorbispay, "time to send config"); send_config = TRUE; } } else { /* no known previous config time, send now */ GST_DEBUG_OBJECT (rtpvorbispay, "no previous config time, send now"); send_config = TRUE; } if (send_config) { /* we need to send config now first */ /* different TDT type forces flush */ gst_rtp_vorbis_pay_payload_buffer (rtpvorbispay, 1, NULL, rtpvorbispay->config_data, rtpvorbispay->config_size, timestamp, GST_CLOCK_TIME_NONE, rtpvorbispay->config_extra_len); if (running_time != -1) { rtpvorbispay->last_config = running_time; } } } ret = gst_rtp_vorbis_pay_payload_buffer (rtpvorbispay, VDT, buffer, data, size, timestamp, duration, 0); gst_buffer_unmap (buffer, &map); gst_buffer_unref (buffer); done: return ret; /* ERRORS */ wrong_size: { GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE, ("Invalid packet size (1 < %" G_GSIZE_FORMAT ")", size), (NULL)); gst_buffer_unmap (buffer, &map); gst_buffer_unref (buffer); return GST_FLOW_OK; } parse_id_failed: { gst_buffer_unmap (buffer, &map); gst_buffer_unref (buffer); return GST_FLOW_ERROR; } unknown_header: { GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE, (NULL), ("Ignoring unknown header received")); gst_buffer_unmap (buffer, &map); gst_buffer_unref (buffer); return GST_FLOW_OK; } header_error: { GST_ELEMENT_WARNING (rtpvorbispay, STREAM, DECODE, (NULL), ("Error initializing header config")); gst_buffer_unmap (buffer, &map); gst_buffer_unref (buffer); return GST_FLOW_OK; } } static gboolean gst_rtp_vorbis_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event) { GstRtpVorbisPay *rtpvorbispay = GST_RTP_VORBIS_PAY (payload); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_STOP: gst_rtp_vorbis_pay_clear_packet (rtpvorbispay); break; default: break; } /* false to let parent handle event as well */ return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event); } static GstStateChangeReturn gst_rtp_vorbis_pay_change_state (GstElement * element, GstStateChange transition) { GstRtpVorbisPay *rtpvorbispay; GstStateChangeReturn ret; rtpvorbispay = GST_RTP_VORBIS_PAY (element); switch (transition) { default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_rtp_vorbis_pay_cleanup (rtpvorbispay); break; default: break; } return ret; } static void gst_rtp_vorbis_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpVorbisPay *rtpvorbispay; rtpvorbispay = GST_RTP_VORBIS_PAY (object); switch (prop_id) { case PROP_CONFIG_INTERVAL: rtpvorbispay->config_interval = g_value_get_uint (value); break; default: break; } } static void gst_rtp_vorbis_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpVorbisPay *rtpvorbispay; rtpvorbispay = GST_RTP_VORBIS_PAY (object); switch (prop_id) { case PROP_CONFIG_INTERVAL: g_value_set_uint (value, rtpvorbispay->config_interval); break; default: break; } } gboolean gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpvorbispay", GST_RANK_SECONDARY, GST_TYPE_RTP_VORBIS_PAY); }