diff options
Diffstat (limited to 'gst/rtp/gstrtpvorbispay.c')
| -rw-r--r-- | gst/rtp/gstrtpvorbispay.c | 1002 |
1 files changed, 1002 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpvorbispay.c b/gst/rtp/gstrtpvorbispay.c new file mode 100644 index 0000000000..e54e2a7cde --- /dev/null +++ b/gst/rtp/gstrtpvorbispay.c @@ -0,0 +1,1002 @@ +/* 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 "gstrtpelements.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); +GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpvorbispay, "rtpvorbispay", + GST_RANK_SECONDARY, GST_TYPE_RTP_VORBIS_PAY, rtp_element_init (plugin)); + +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; + } +} |