Mercurial > libavformat.hg
view rtp.c @ 251:300c94a708f7 libavformat
remove unused AMREXTRALIBS
author | bellard |
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date | Tue, 16 Sep 2003 20:29:19 +0000 |
parents | 3d92f793fd67 |
children | 6091b76cfc2a |
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/* * 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_t ssrc; uint16_t seq; uint32_t timestamp; uint32_t base_timestamp; uint32_t cur_timestamp; int max_payload_size; /* rtcp sender statistics receive */ int64_t last_rtcp_ntp_time; int64_t first_rtcp_ntp_time; uint32_t 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_t buf[RTP_MAX_PACKET_LENGTH]; uint8_t *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_MP3: 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_t decode_be32(const uint8_t *p) { return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3]; } static inline uint64_t decode_be64(const uint8_t *p) { return ((uint64_t)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); if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; 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_t 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; } switch(st->codec.codec_id) { case CODEC_ID_MP2: case CODEC_ID_MPEG1VIDEO: if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { int64_t addend; /* XXX: is it really necessary to unify the timestamp base ? */ /* compute pts from timestamp with received ntp_time */ delta_timestamp = timestamp - s->last_rtcp_timestamp; /* convert to 90 kHz without overflow */ addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; addend = (addend * 5625) >> 14; pkt->pts = addend + delta_timestamp; } break; default: /* no timestamp info yet */ break; } 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; s->first_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_MP3: 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_t 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, const uint8_t *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, const uint8_t *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, const uint8_t *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, const uint8_t *buf1, int size) { RTPContext *s = s1->priv_data; AVStream *st = s1->streams[0]; int len, h, max_packet_size; uint8_t *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 */ s->timestamp = s->base_timestamp + av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, 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, const uint8_t *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 */ s->timestamp = s->base_timestamp + av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, 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, const uint8_t *buf1, int size, int64_t pts) { RTPContext *s = s1->priv_data; AVStream *st = s1->streams[0]; int rtcp_bytes; int64_t 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 */ /* XXX: 90 kHz timestamp hardcoded */ ntp_time = (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_MP3: 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; }