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-rw-r--r--libavformat/rtp.c687
1 files changed, 687 insertions, 0 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c
new file mode 100644
index 0000000000..f36da6d41d
--- /dev/null
+++ b/libavformat/rtp.c
@@ -0,0 +1,687 @@
+/*
+ * RTP input/output format
+ * Copyright (c) 2002 Fabrice Bellard.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser 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
+ */
+#include "avformat.h"
+
+#include <unistd.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netinet/in.h>
+#ifndef __BEOS__
+# include <arpa/inet.h>
+#else
+# include "barpainet.h"
+#endif
+#include <netdb.h>
+
+//#define DEBUG
+
+
+/* TODO: - add RTCP statistics reporting (should be optional).
+
+ - add support for h263/mpeg4 packetized output : IDEA: send a
+ buffer to 'rtp_write_packet' contains all the packets for ONE
+ frame. Each packet should have a four byte header containing
+ the length in big endian format (same trick as
+ 'url_open_dyn_packet_buf')
+*/
+
+#define RTP_VERSION 2
+
+#define RTP_MAX_SDES 256 /* maximum text length for SDES */
+
+/* RTCP paquets use 0.5 % of the bandwidth */
+#define RTCP_TX_RATIO_NUM 5
+#define RTCP_TX_RATIO_DEN 1000
+
+typedef enum {
+ RTCP_SR = 200,
+ RTCP_RR = 201,
+ RTCP_SDES = 202,
+ RTCP_BYE = 203,
+ RTCP_APP = 204
+} rtcp_type_t;
+
+typedef enum {
+ RTCP_SDES_END = 0,
+ RTCP_SDES_CNAME = 1,
+ RTCP_SDES_NAME = 2,
+ RTCP_SDES_EMAIL = 3,
+ RTCP_SDES_PHONE = 4,
+ RTCP_SDES_LOC = 5,
+ RTCP_SDES_TOOL = 6,
+ RTCP_SDES_NOTE = 7,
+ RTCP_SDES_PRIV = 8,
+ RTCP_SDES_IMG = 9,
+ RTCP_SDES_DOOR = 10,
+ RTCP_SDES_SOURCE = 11
+} rtcp_sdes_type_t;
+
+enum RTPPayloadType {
+ RTP_PT_ULAW = 0,
+ RTP_PT_GSM = 3,
+ RTP_PT_G723 = 4,
+ RTP_PT_ALAW = 8,
+ RTP_PT_S16BE_STEREO = 10,
+ RTP_PT_S16BE_MONO = 11,
+ RTP_PT_MPEGAUDIO = 14,
+ RTP_PT_JPEG = 26,
+ RTP_PT_H261 = 31,
+ RTP_PT_MPEGVIDEO = 32,
+ RTP_PT_MPEG2TS = 33,
+ RTP_PT_H263 = 34, /* old H263 encapsulation */
+ RTP_PT_PRIVATE = 96,
+};
+
+typedef struct RTPContext {
+ int payload_type;
+ UINT32 ssrc;
+ UINT16 seq;
+ UINT32 timestamp;
+ UINT32 base_timestamp;
+ UINT32 cur_timestamp;
+ int max_payload_size;
+ /* rtcp sender statistics receive */
+ INT64 last_rtcp_ntp_time;
+ UINT32 last_rtcp_timestamp;
+ /* rtcp sender statistics */
+ unsigned int packet_count;
+ unsigned int octet_count;
+ unsigned int last_octet_count;
+ int first_packet;
+ /* buffer for output */
+ UINT8 buf[RTP_MAX_PACKET_LENGTH];
+ UINT8 *buf_ptr;
+} RTPContext;
+
+int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
+{
+ switch(payload_type) {
+ case RTP_PT_ULAW:
+ codec->codec_id = CODEC_ID_PCM_MULAW;
+ codec->channels = 1;
+ codec->sample_rate = 8000;
+ break;
+ case RTP_PT_ALAW:
+ codec->codec_id = CODEC_ID_PCM_ALAW;
+ codec->channels = 1;
+ codec->sample_rate = 8000;
+ break;
+ case RTP_PT_S16BE_STEREO:
+ codec->codec_id = CODEC_ID_PCM_S16BE;
+ codec->channels = 2;
+ codec->sample_rate = 44100;
+ break;
+ case RTP_PT_S16BE_MONO:
+ codec->codec_id = CODEC_ID_PCM_S16BE;
+ codec->channels = 1;
+ codec->sample_rate = 44100;
+ break;
+ case RTP_PT_MPEGAUDIO:
+ codec->codec_id = CODEC_ID_MP2;
+ break;
+ case RTP_PT_JPEG:
+ codec->codec_id = CODEC_ID_MJPEG;
+ break;
+ case RTP_PT_MPEGVIDEO:
+ codec->codec_id = CODEC_ID_MPEG1VIDEO;
+ break;
+ default:
+ return -1;
+ }
+ return 0;
+}
+
+/* return < 0 if unknown payload type */
+int rtp_get_payload_type(AVCodecContext *codec)
+{
+ int payload_type;
+
+ /* compute the payload type */
+ payload_type = -1;
+ switch(codec->codec_id) {
+ case CODEC_ID_PCM_MULAW:
+ payload_type = RTP_PT_ULAW;
+ break;
+ case CODEC_ID_PCM_ALAW:
+ payload_type = RTP_PT_ALAW;
+ break;
+ case CODEC_ID_PCM_S16BE:
+ if (codec->channels == 1) {
+ payload_type = RTP_PT_S16BE_MONO;
+ } else if (codec->channels == 2) {
+ payload_type = RTP_PT_S16BE_STEREO;
+ }
+ break;
+ case CODEC_ID_MP2:
+ case CODEC_ID_MP3LAME:
+ payload_type = RTP_PT_MPEGAUDIO;
+ break;
+ case CODEC_ID_MJPEG:
+ payload_type = RTP_PT_JPEG;
+ break;
+ case CODEC_ID_MPEG1VIDEO:
+ payload_type = RTP_PT_MPEGVIDEO;
+ break;
+ default:
+ break;
+ }
+ return payload_type;
+}
+
+static inline UINT32 decode_be32(const UINT8 *p)
+{
+ return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
+}
+
+static inline UINT32 decode_be64(const UINT8 *p)
+{
+ return ((UINT64)decode_be32(p) << 32) | decode_be32(p + 4);
+}
+
+static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
+{
+ RTPContext *s = s1->priv_data;
+
+ if (buf[1] != 200)
+ return -1;
+ s->last_rtcp_ntp_time = decode_be64(buf + 8);
+ s->last_rtcp_timestamp = decode_be32(buf + 16);
+ return 0;
+}
+
+/**
+ * Parse an RTP packet directly sent as raw data. Can only be used if
+ * 'raw' is given as input file
+ * @param s1 media file context
+ * @param pkt returned packet
+ * @param buf input buffer
+ * @param len buffer len
+ * @return zero if no error.
+ */
+int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt,
+ const unsigned char *buf, int len)
+{
+ RTPContext *s = s1->priv_data;
+ unsigned int ssrc, h;
+ int payload_type, seq, delta_timestamp;
+ AVStream *st;
+ UINT32 timestamp;
+
+ if (len < 12)
+ return -1;
+
+ if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
+ return -1;
+ if (buf[1] >= 200 && buf[1] <= 204) {
+ rtcp_parse_packet(s1, buf, len);
+ return -1;
+ }
+ payload_type = buf[1] & 0x7f;
+ seq = (buf[2] << 8) | buf[3];
+ timestamp = decode_be32(buf + 4);
+ ssrc = decode_be32(buf + 8);
+
+ if (s->payload_type < 0) {
+ s->payload_type = payload_type;
+
+ if (payload_type == RTP_PT_MPEG2TS) {
+ /* XXX: special case : not a single codec but a whole stream */
+ return -1;
+ } else {
+ st = av_new_stream(s1, 0);
+ if (!st)
+ return -1;
+ rtp_get_codec_info(&st->codec, payload_type);
+ }
+ }
+
+ /* NOTE: we can handle only one payload type */
+ if (s->payload_type != payload_type)
+ return -1;
+#if defined(DEBUG) || 1
+ if (seq != ((s->seq + 1) & 0xffff)) {
+ printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n",
+ payload_type, seq, ((s->seq + 1) & 0xffff));
+ }
+ s->seq = seq;
+#endif
+ len -= 12;
+ buf += 12;
+ st = s1->streams[0];
+ switch(st->codec.codec_id) {
+ case CODEC_ID_MP2:
+ /* better than nothing: skip mpeg audio RTP header */
+ if (len <= 4)
+ return -1;
+ h = decode_be32(buf);
+ len -= 4;
+ buf += 4;
+ av_new_packet(pkt, len);
+ memcpy(pkt->data, buf, len);
+ break;
+ case CODEC_ID_MPEG1VIDEO:
+ /* better than nothing: skip mpeg audio RTP header */
+ if (len <= 4)
+ return -1;
+ h = decode_be32(buf);
+ buf += 4;
+ len -= 4;
+ if (h & (1 << 26)) {
+ /* mpeg2 */
+ if (len <= 4)
+ return -1;
+ buf += 4;
+ len -= 4;
+ }
+ av_new_packet(pkt, len);
+ memcpy(pkt->data, buf, len);
+ break;
+ default:
+ av_new_packet(pkt, len);
+ memcpy(pkt->data, buf, len);
+ break;
+ }
+
+ if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
+ /* compute pts from timestamp with received ntp_time */
+ delta_timestamp = timestamp - s->last_rtcp_timestamp;
+ /* XXX: do conversion, but not needed for mpeg at 90 KhZ */
+ pkt->pts = s->last_rtcp_ntp_time + delta_timestamp;
+ }
+ return 0;
+}
+
+static int rtp_read_header(AVFormatContext *s1,
+ AVFormatParameters *ap)
+{
+ RTPContext *s = s1->priv_data;
+ s->payload_type = -1;
+ s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
+ return 0;
+}
+
+static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
+{
+ char buf[RTP_MAX_PACKET_LENGTH];
+ int ret;
+
+ /* XXX: needs a better API for packet handling ? */
+ for(;;) {
+ ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
+ if (ret < 0)
+ return AVERROR_IO;
+ if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
+ break;
+ }
+ return 0;
+}
+
+static int rtp_read_close(AVFormatContext *s1)
+{
+ // RTPContext *s = s1->priv_data;
+ return 0;
+}
+
+static int rtp_probe(AVProbeData *p)
+{
+ if (strstart(p->filename, "rtp://", NULL))
+ return AVPROBE_SCORE_MAX;
+ return 0;
+}
+
+/* rtp output */
+
+static int rtp_write_header(AVFormatContext *s1)
+{
+ RTPContext *s = s1->priv_data;
+ int payload_type, max_packet_size;
+ AVStream *st;
+
+ if (s1->nb_streams != 1)
+ return -1;
+ st = s1->streams[0];
+
+ payload_type = rtp_get_payload_type(&st->codec);
+ if (payload_type < 0)
+ payload_type = RTP_PT_PRIVATE; /* private payload type */
+ s->payload_type = payload_type;
+
+ s->base_timestamp = random();
+ s->timestamp = s->base_timestamp;
+ s->ssrc = random();
+ s->first_packet = 1;
+
+ max_packet_size = url_fget_max_packet_size(&s1->pb);
+ if (max_packet_size <= 12)
+ return AVERROR_IO;
+ s->max_payload_size = max_packet_size - 12;
+
+ switch(st->codec.codec_id) {
+ case CODEC_ID_MP2:
+ case CODEC_ID_MP3LAME:
+ s->buf_ptr = s->buf + 4;
+ s->cur_timestamp = 0;
+ break;
+ case CODEC_ID_MPEG1VIDEO:
+ s->cur_timestamp = 0;
+ break;
+ default:
+ s->buf_ptr = s->buf;
+ break;
+ }
+
+ return 0;
+}
+
+/* send an rtcp sender report packet */
+static void rtcp_send_sr(AVFormatContext *s1, INT64 ntp_time)
+{
+ RTPContext *s = s1->priv_data;
+#if defined(DEBUG)
+ printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
+#endif
+ put_byte(&s1->pb, (RTP_VERSION << 6));
+ put_byte(&s1->pb, 200);
+ put_be16(&s1->pb, 6); /* length in words - 1 */
+ put_be32(&s1->pb, s->ssrc);
+ put_be64(&s1->pb, ntp_time);
+ put_be32(&s1->pb, s->timestamp);
+ put_be32(&s1->pb, s->packet_count);
+ put_be32(&s1->pb, s->octet_count);
+ put_flush_packet(&s1->pb);
+}
+
+/* send an rtp packet. sequence number is incremented, but the caller
+ must update the timestamp itself */
+static void rtp_send_data(AVFormatContext *s1, UINT8 *buf1, int len)
+{
+ RTPContext *s = s1->priv_data;
+
+#ifdef DEBUG
+ printf("rtp_send_data size=%d\n", len);
+#endif
+
+ /* build the RTP header */
+ put_byte(&s1->pb, (RTP_VERSION << 6));
+ put_byte(&s1->pb, s->payload_type & 0x7f);
+ put_be16(&s1->pb, s->seq);
+ put_be32(&s1->pb, s->timestamp);
+ put_be32(&s1->pb, s->ssrc);
+
+ put_buffer(&s1->pb, buf1, len);
+ put_flush_packet(&s1->pb);
+
+ s->seq++;
+ s->octet_count += len;
+ s->packet_count++;
+}
+
+/* send an integer number of samples and compute time stamp and fill
+ the rtp send buffer before sending. */
+static void rtp_send_samples(AVFormatContext *s1,
+ UINT8 *buf1, int size, int sample_size)
+{
+ RTPContext *s = s1->priv_data;
+ int len, max_packet_size, n;
+
+ max_packet_size = (s->max_payload_size / sample_size) * sample_size;
+ /* not needed, but who nows */
+ if ((size % sample_size) != 0)
+ av_abort();
+ while (size > 0) {
+ len = (max_packet_size - (s->buf_ptr - s->buf));
+ if (len > size)
+ len = size;
+
+ /* copy data */
+ memcpy(s->buf_ptr, buf1, len);
+ s->buf_ptr += len;
+ buf1 += len;
+ size -= len;
+ n = (s->buf_ptr - s->buf);
+ /* if buffer full, then send it */
+ if (n >= max_packet_size) {
+ rtp_send_data(s1, s->buf, n);
+ s->buf_ptr = s->buf;
+ /* update timestamp */
+ s->timestamp += n / sample_size;
+ }
+ }
+}
+
+/* NOTE: we suppose that exactly one frame is given as argument here */
+/* XXX: test it */
+static void rtp_send_mpegaudio(AVFormatContext *s1,
+ UINT8 *buf1, int size)
+{
+ RTPContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
+ int len, count, max_packet_size;
+
+ max_packet_size = s->max_payload_size;
+
+ /* test if we must flush because not enough space */
+ len = (s->buf_ptr - s->buf);
+ if ((len + size) > max_packet_size) {
+ if (len > 4) {
+ rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
+ s->buf_ptr = s->buf + 4;
+ /* 90 KHz time stamp */
+ s->timestamp = s->base_timestamp +
+ (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
+ }
+ }
+
+ /* add the packet */
+ if (size > max_packet_size) {
+ /* big packet: fragment */
+ count = 0;
+ while (size > 0) {
+ len = max_packet_size - 4;
+ if (len > size)
+ len = size;
+ /* build fragmented packet */
+ s->buf[0] = 0;
+ s->buf[1] = 0;
+ s->buf[2] = count >> 8;
+ s->buf[3] = count;
+ memcpy(s->buf + 4, buf1, len);
+ rtp_send_data(s1, s->buf, len + 4);
+ size -= len;
+ buf1 += len;
+ count += len;
+ }
+ } else {
+ if (s->buf_ptr == s->buf + 4) {
+ /* no fragmentation possible */
+ s->buf[0] = 0;
+ s->buf[1] = 0;
+ s->buf[2] = 0;
+ s->buf[3] = 0;
+ }
+ memcpy(s->buf_ptr, buf1, size);
+ s->buf_ptr += size;
+ }
+ s->cur_timestamp += st->codec.frame_size;
+}
+
+/* NOTE: a single frame must be passed with sequence header if
+ needed. XXX: use slices. */
+static void rtp_send_mpegvideo(AVFormatContext *s1,
+ UINT8 *buf1, int size)
+{
+ RTPContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
+ int len, h, max_packet_size;
+ UINT8 *q;
+
+ max_packet_size = s->max_payload_size;
+
+ while (size > 0) {
+ /* XXX: more correct headers */
+ h = 0;
+ if (st->codec.sub_id == 2)
+ h |= 1 << 26; /* mpeg 2 indicator */
+ q = s->buf;
+ *q++ = h >> 24;
+ *q++ = h >> 16;
+ *q++ = h >> 8;
+ *q++ = h;
+
+ if (st->codec.sub_id == 2) {
+ h = 0;
+ *q++ = h >> 24;
+ *q++ = h >> 16;
+ *q++ = h >> 8;
+ *q++ = h;
+ }
+
+ len = max_packet_size - (q - s->buf);
+ if (len > size)
+ len = size;
+
+ memcpy(q, buf1, len);
+ q += len;
+
+ /* 90 KHz time stamp */
+ /* XXX: overflow */
+ s->timestamp = s->base_timestamp +
+ (s->cur_timestamp * 90000LL * FRAME_RATE_BASE) / st->codec.frame_rate;
+ rtp_send_data(s1, s->buf, q - s->buf);
+
+ buf1 += len;
+ size -= len;
+ }
+ s->cur_timestamp++;
+}
+
+static void rtp_send_raw(AVFormatContext *s1,
+ UINT8 *buf1, int size)
+{
+ RTPContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
+ int len, max_packet_size;
+
+ max_packet_size = s->max_payload_size;
+
+ while (size > 0) {
+ len = max_packet_size;
+ if (len > size)
+ len = size;
+
+ /* 90 KHz time stamp */
+ /* XXX: overflow */
+ s->timestamp = s->base_timestamp +
+ (s->cur_timestamp * 90000LL * FRAME_RATE_BASE) / st->codec.frame_rate;
+ rtp_send_data(s1, buf1, len);
+
+ buf1 += len;
+ size -= len;
+ }
+ s->cur_timestamp++;
+}
+
+/* write an RTP packet. 'buf1' must contain a single specific frame. */
+static int rtp_write_packet(AVFormatContext *s1, int stream_index,
+ UINT8 *buf1, int size, int force_pts)
+{
+ RTPContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
+ int rtcp_bytes;
+ INT64 ntp_time;
+
+#ifdef DEBUG
+ printf("%d: write len=%d\n", stream_index, size);
+#endif
+
+ /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
+ rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
+ RTCP_TX_RATIO_DEN;
+ if (s->first_packet || rtcp_bytes >= 28) {
+ /* compute NTP time */
+ ntp_time = force_pts; // ((INT64)force_pts << 28) / 5625
+ rtcp_send_sr(s1, ntp_time);
+ s->last_octet_count = s->octet_count;
+ s->first_packet = 0;
+ }
+
+ switch(st->codec.codec_id) {
+ case CODEC_ID_PCM_MULAW:
+ case CODEC_ID_PCM_ALAW:
+ case CODEC_ID_PCM_U8:
+ case CODEC_ID_PCM_S8:
+ rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
+ break;
+ case CODEC_ID_PCM_U16BE:
+ case CODEC_ID_PCM_U16LE:
+ case CODEC_ID_PCM_S16BE:
+ case CODEC_ID_PCM_S16LE:
+ rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
+ break;
+ case CODEC_ID_MP2:
+ case CODEC_ID_MP3LAME:
+ rtp_send_mpegaudio(s1, buf1, size);
+ break;
+ case CODEC_ID_MPEG1VIDEO:
+ rtp_send_mpegvideo(s1, buf1, size);
+ break;
+ default:
+ /* better than nothing : send the codec raw data */
+ rtp_send_raw(s1, buf1, size);
+ break;
+ }
+ return 0;
+}
+
+static int rtp_write_trailer(AVFormatContext *s1)
+{
+ // RTPContext *s = s1->priv_data;
+ return 0;
+}
+
+AVInputFormat rtp_demux = {
+ "rtp",
+ "RTP input format",
+ sizeof(RTPContext),
+ rtp_probe,
+ rtp_read_header,
+ rtp_read_packet,
+ rtp_read_close,
+ .flags = AVFMT_NOHEADER,
+};
+
+AVOutputFormat rtp_mux = {
+ "rtp",
+ "RTP output format",
+ NULL,
+ NULL,
+ sizeof(RTPContext),
+ CODEC_ID_PCM_MULAW,
+ CODEC_ID_NONE,
+ rtp_write_header,
+ rtp_write_packet,
+ rtp_write_trailer,
+};
+
+int rtp_init(void)
+{
+ av_register_output_format(&rtp_mux);
+ av_register_input_format(&rtp_demux);
+ return 0;
+}
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