diff options
author | Wim Taymans <wim.taymans@gmail.com> | 2006-09-22 12:08:14 +0000 |
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committer | Wim Taymans <wim.taymans@gmail.com> | 2006-09-22 12:08:14 +0000 |
commit | 8dbf0334202a4af22453ce50513bce26f51a3075 (patch) | |
tree | 97cd9d6169c2b948653a431206b7b692e3c99bef /gst/rtp/gstrtpvorbispay.c | |
parent | 3b5584f8d128b57f7c6dcbde21df044511b9fd99 (diff) |
gst/rtp/: Small cleanups.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
Small cleanups.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_process),
(gst_rtp_vorbis_depay_set_property),
(gst_rtp_vorbis_depay_get_property),
(gst_rtp_vorbis_depay_change_state),
(gst_rtp_vorbis_depay_plugin_init):
* gst/rtp/gstrtpvorbisdepay.h:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_flush_packet),
(gst_rtp_vorbis_pay_append_buffer),
(gst_rtp_vorbis_pay_handle_buffer),
(gst_rtp_vorbis_pay_plugin_init):
* gst/rtp/gstrtpvorbispay.h:
Add experimental vorbis pay and depayloaders.
Diffstat (limited to 'gst/rtp/gstrtpvorbispay.c')
-rw-r--r-- | gst/rtp/gstrtpvorbispay.c | 333 |
1 files changed, 333 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpvorbispay.c b/gst/rtp/gstrtpvorbispay.c new file mode 100644 index 00000000..30336425 --- /dev/null +++ b/gst/rtp/gstrtpvorbispay.c @@ -0,0 +1,333 @@ +/* GStreamer + * Copyright (C) <2006> Wim Taymans <wim@fluendo.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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +# include "config.h" +#endif + +#include <string.h> + +#include <gst/rtp/gstrtpbuffer.h> + +#include "gstrtpvorbispay.h" + +GST_DEBUG_CATEGORY_STATIC (rtpvorbispay_debug); +#define GST_CAT_DEFAULT (rtpvorbispay_debug) + +/* references: + * http://svn.xiph.org/trunk/vorbis/doc/draft-ietf-avt-rtp-vorbis-01.txt + */ + +/* elementfactory information */ +static const GstElementDetails gst_rtp_vorbispay_details = +GST_ELEMENT_DETAILS ("RTP packet parser", + "Codec/Payloader/Network", + "Payload-encode Vorbis audio into RTP packets (draft-01 RFC XXXX)", + "Wim Taymans <wim@fluendo.com>"); + +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) [ 96, 127 ], " + "clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"vorbis\"" + /* All required parameters + * + * "encoding-params = (string) <num channels>" + * "delivery-method = (string) { inline, in_band, out_band/<specific_name> } " + * "configuration = (string) ANY" + */ + /* All optional parameters + * + * "configuration-uri =" + */ + ) + ); + +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") + ); + +GST_BOILERPLATE (GstRtpVorbisPay, gst_rtp_vorbis_pay, GstBaseRTPPayload, + GST_TYPE_BASE_RTP_PAYLOAD); + +static gboolean gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload, + GstCaps * caps); +static GstFlowReturn gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * pad, + GstBuffer * buffer); + +static void +gst_rtp_vorbis_pay_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_vorbis_pay_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&gst_rtp_vorbis_pay_sink_template)); + + gst_element_class_set_details (element_class, &gst_rtp_vorbispay_details); +} + +static void +gst_rtp_vorbis_pay_class_init (GstRtpVorbisPayClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstBaseRTPPayloadClass *gstbasertppayload_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass; + + parent_class = g_type_class_peek_parent (klass); + + gstbasertppayload_class->set_caps = gst_rtp_vorbis_pay_setcaps; + gstbasertppayload_class->handle_buffer = gst_rtp_vorbis_pay_handle_buffer; + + GST_DEBUG_CATEGORY_INIT (rtpvorbispay_debug, "rtpvorbispay", 0, + "Vorbis RTP Payloader"); +} + +static void +gst_rtp_vorbis_pay_init (GstRtpVorbisPay * rtpvorbispay, + GstRtpVorbisPayClass * klass) +{ +} + +static gboolean +gst_rtp_vorbis_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps) +{ + GstRtpVorbisPay *rtpvorbispay; + + rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload); + + gst_basertppayload_set_options (basepayload, "audio", TRUE, "vorbis", 8000); + gst_basertppayload_set_outcaps (basepayload, + "encoding-params", G_TYPE_STRING, "1", + /* don't set the defaults + */ + NULL); + + return TRUE; +} + +static void +gst_rtp_vorbis_pay_init_packet (GstRtpVorbisPay * rtpvorbispay) +{ + guint payload_len; + + if (rtpvorbispay->packet) + gst_buffer_unref (rtpvorbispay->packet); + + GST_DEBUG_OBJECT (rtpvorbispay, "starting new packet"); + + /* new packet allocate max packet size */ + rtpvorbispay->packet = + gst_rtp_buffer_new_allocate_len (GST_BASE_RTP_PAYLOAD_MTU + (rtpvorbispay), 0, 0); + rtpvorbispay->payload_pos = 4; + payload_len = gst_rtp_buffer_get_payload_len (rtpvorbispay->packet); + rtpvorbispay->payload_left = payload_len - 4; + rtpvorbispay->payload_duration = 0; + rtpvorbispay->payload_ident = 0; + rtpvorbispay->payload_F = 0; + rtpvorbispay->payload_VDT = 0; + rtpvorbispay->payload_pkts = 0; +} + +static GstFlowReturn +gst_rtp_vorbis_pay_flush_packet (GstRtpVorbisPay * rtpvorbispay) +{ + GstFlowReturn ret; + guint8 *payload; + guint hlen; + + /* check for empty packet */ + if (!rtpvorbispay || rtpvorbispay->payload_pos <= 4) + return GST_FLOW_OK; + + GST_DEBUG_OBJECT (rtpvorbispay, "flushing packet"); + + /* fix header */ + payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet); + /* + * 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); + + /* shrink the buffer size to the last written byte */ + hlen = gst_rtp_buffer_calc_header_len (0); + GST_BUFFER_SIZE (rtpvorbispay->packet) = hlen + rtpvorbispay->payload_pos; + + /* push, this gives away our ref to the packet, so clear it. */ + ret = + gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpvorbispay), + rtpvorbispay->packet); + rtpvorbispay->packet = NULL; + + /* prepare new packet */ + gst_rtp_vorbis_pay_init_packet (rtpvorbispay); + + return ret; +} + +static GstFlowReturn +gst_rtp_vorbis_pay_append_buffer (GstRtpVorbisPay * rtpvorbispay, + GstBuffer * buffer) +{ + GstFlowReturn res; + guint size; + GstClockTime duration; + guint plen; + guint8 *ppos, *payload, *data; + gboolean fragmented; + + res = GST_FLOW_OK; + + if (rtpvorbispay->payload_left < 2) + return res; + + size = GST_BUFFER_SIZE (buffer); + /* skip packets that are too big */ + if (size > 0xffff) + return res; + + data = GST_BUFFER_DATA (buffer); + duration = GST_BUFFER_DURATION (buffer); + payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet); + ppos = payload + rtpvorbispay->payload_pos; + fragmented = FALSE; + + while (size) { + plen = MIN (rtpvorbispay->payload_left - 2, size); + + GST_DEBUG_OBJECT (rtpvorbispay, "append %u bytes", plen); + + ppos[0] = (plen >> 8) & 0xff; + ppos[1] = (plen & 0xff); + memcpy (&ppos[2], data, plen); + + 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) { + /* unfragmented packet, update stats for next packet */ + rtpvorbispay->payload_pkts++; + if (duration != GST_CLOCK_TIME_NONE) + rtpvorbispay->payload_duration += duration; + } else { + /* fragmented packet starts, set F to 0x1, mark ourselves as + * fragmented. */ + rtpvorbispay->payload_F = 0x1; + fragmented = TRUE; + } + } + if (fragmented) { + /* fragmented packets are always flushed and have ptks of 0 */ + rtpvorbispay->payload_pkts = 0; + res = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay); + /* get new pointers */ + payload = gst_rtp_buffer_get_payload (rtpvorbispay->packet); + ppos = payload + rtpvorbispay->payload_pos; + } + } + + return res; +} + +static GstFlowReturn +gst_rtp_vorbis_pay_handle_buffer (GstBaseRTPPayload * basepayload, + GstBuffer * buffer) +{ + GstRtpVorbisPay *rtpvorbispay; + GstFlowReturn ret; + guint size, newsize; + guint packet_len; + GstClockTime duration, newduration; + gboolean flush; + + rtpvorbispay = GST_RTP_VORBIS_PAY (basepayload); + + size = GST_BUFFER_SIZE (buffer); + duration = GST_BUFFER_DURATION (buffer); + + GST_DEBUG_OBJECT (rtpvorbispay, "size %u, duration %" GST_TIME_FORMAT, + size, GST_TIME_ARGS (duration)); + + if (!rtpvorbispay->packet) + gst_rtp_vorbis_pay_init_packet (rtpvorbispay); + + /* 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_basertppayload_is_filled (basepayload, packet_len, newduration); + /* we can store up to 15 vorbis packets in one RTP packet. */ + flush |= (rtpvorbispay->payload_pkts == 15); + + if (flush) + ret = gst_rtp_vorbis_pay_flush_packet (rtpvorbispay); + + /* put buffer in packet */ + ret = gst_rtp_vorbis_pay_append_buffer (rtpvorbispay, buffer); + + return ret; +} + +gboolean +gst_rtp_vorbis_pay_plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rtpvorbispay", + GST_RANK_NONE, GST_TYPE_RTP_VORBIS_PAY); +} |