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libavformat/rtpenc.c

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00001 /*
00002  * RTP output format
00003  * Copyright (c) 2002 Fabrice Bellard
00004  *
00005  * This file is part of FFmpeg.
00006  *
00007  * FFmpeg is free software; you can redistribute it and/or
00008  * modify it under the terms of the GNU Lesser General Public
00009  * License as published by the Free Software Foundation; either
00010  * version 2.1 of the License, or (at your option) any later version.
00011  *
00012  * FFmpeg is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00015  * Lesser General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU Lesser General Public
00018  * License along with FFmpeg; if not, write to the Free Software
00019  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00020  */
00021 
00022 #include "avformat.h"
00023 #include "mpegts.h"
00024 #include "internal.h"
00025 #include "libavutil/random_seed.h"
00026 #include "libavutil/opt.h"
00027 
00028 #include "rtpenc.h"
00029 
00030 //#define DEBUG
00031 
00032 static const AVOption options[] = {
00033     FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
00034     { NULL },
00035 };
00036 
00037 static const AVClass rtp_muxer_class = {
00038     .class_name = "RTP muxer",
00039     .item_name  = av_default_item_name,
00040     .option     = options,
00041     .version    = LIBAVUTIL_VERSION_INT,
00042 };
00043 
00044 #define RTCP_SR_SIZE 28
00045 
00046 static int is_supported(enum CodecID id)
00047 {
00048     switch(id) {
00049     case CODEC_ID_H263:
00050     case CODEC_ID_H263P:
00051     case CODEC_ID_H264:
00052     case CODEC_ID_MPEG1VIDEO:
00053     case CODEC_ID_MPEG2VIDEO:
00054     case CODEC_ID_MPEG4:
00055     case CODEC_ID_AAC:
00056     case CODEC_ID_MP2:
00057     case CODEC_ID_MP3:
00058     case CODEC_ID_PCM_ALAW:
00059     case CODEC_ID_PCM_MULAW:
00060     case CODEC_ID_PCM_S8:
00061     case CODEC_ID_PCM_S16BE:
00062     case CODEC_ID_PCM_S16LE:
00063     case CODEC_ID_PCM_U16BE:
00064     case CODEC_ID_PCM_U16LE:
00065     case CODEC_ID_PCM_U8:
00066     case CODEC_ID_MPEG2TS:
00067     case CODEC_ID_AMR_NB:
00068     case CODEC_ID_AMR_WB:
00069     case CODEC_ID_VORBIS:
00070     case CODEC_ID_THEORA:
00071     case CODEC_ID_VP8:
00072     case CODEC_ID_ADPCM_G722:
00073         return 1;
00074     default:
00075         return 0;
00076     }
00077 }
00078 
00079 static int rtp_write_header(AVFormatContext *s1)
00080 {
00081     RTPMuxContext *s = s1->priv_data;
00082     int max_packet_size, n;
00083     AVStream *st;
00084 
00085     if (s1->nb_streams != 1)
00086         return -1;
00087     st = s1->streams[0];
00088     if (!is_supported(st->codec->codec_id)) {
00089         av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
00090 
00091         return -1;
00092     }
00093 
00094     s->payload_type = ff_rtp_get_payload_type(st->codec);
00095     if (s->payload_type < 0)
00096         s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO);
00097 
00098     s->base_timestamp = av_get_random_seed();
00099     s->timestamp = s->base_timestamp;
00100     s->cur_timestamp = 0;
00101     s->ssrc = av_get_random_seed();
00102     s->first_packet = 1;
00103     s->first_rtcp_ntp_time = ff_ntp_time();
00104     if (s1->start_time_realtime)
00105         /* Round the NTP time to whole milliseconds. */
00106         s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
00107                                  NTP_OFFSET_US;
00108 
00109     max_packet_size = s1->pb->max_packet_size;
00110     if (max_packet_size <= 12)
00111         return AVERROR(EIO);
00112     s->buf = av_malloc(max_packet_size);
00113     if (s->buf == NULL) {
00114         return AVERROR(ENOMEM);
00115     }
00116     s->max_payload_size = max_packet_size - 12;
00117 
00118     s->max_frames_per_packet = 0;
00119     if (s1->max_delay) {
00120         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
00121             if (st->codec->frame_size == 0) {
00122                 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
00123             } else {
00124                 s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN);
00125             }
00126         }
00127         if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
00128             /* FIXME: We should round down here... */
00129             s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
00130         }
00131     }
00132 
00133     av_set_pts_info(st, 32, 1, 90000);
00134     switch(st->codec->codec_id) {
00135     case CODEC_ID_MP2:
00136     case CODEC_ID_MP3:
00137         s->buf_ptr = s->buf + 4;
00138         break;
00139     case CODEC_ID_MPEG1VIDEO:
00140     case CODEC_ID_MPEG2VIDEO:
00141         break;
00142     case CODEC_ID_MPEG2TS:
00143         n = s->max_payload_size / TS_PACKET_SIZE;
00144         if (n < 1)
00145             n = 1;
00146         s->max_payload_size = n * TS_PACKET_SIZE;
00147         s->buf_ptr = s->buf;
00148         break;
00149     case CODEC_ID_H264:
00150         /* check for H.264 MP4 syntax */
00151         if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
00152             s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
00153         }
00154         break;
00155     case CODEC_ID_VORBIS:
00156     case CODEC_ID_THEORA:
00157         if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
00158         s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
00159         s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
00160         s->num_frames = 0;
00161         goto defaultcase;
00162     case CODEC_ID_VP8:
00163         av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
00164                                  "incompatible with the latest spec drafts.\n");
00165         break;
00166     case CODEC_ID_ADPCM_G722:
00167         /* Due to a historical error, the clock rate for G722 in RTP is
00168          * 8000, even if the sample rate is 16000. See RFC 3551. */
00169         av_set_pts_info(st, 32, 1, 8000);
00170         break;
00171     case CODEC_ID_AMR_NB:
00172     case CODEC_ID_AMR_WB:
00173         if (!s->max_frames_per_packet)
00174             s->max_frames_per_packet = 12;
00175         if (st->codec->codec_id == CODEC_ID_AMR_NB)
00176             n = 31;
00177         else
00178             n = 61;
00179         /* max_header_toc_size + the largest AMR payload must fit */
00180         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
00181             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
00182             return -1;
00183         }
00184         if (st->codec->channels != 1) {
00185             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
00186             return -1;
00187         }
00188     case CODEC_ID_AAC:
00189         s->num_frames = 0;
00190     default:
00191 defaultcase:
00192         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
00193             av_set_pts_info(st, 32, 1, st->codec->sample_rate);
00194         }
00195         s->buf_ptr = s->buf;
00196         break;
00197     }
00198 
00199     return 0;
00200 }
00201 
00202 /* send an rtcp sender report packet */
00203 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
00204 {
00205     RTPMuxContext *s = s1->priv_data;
00206     uint32_t rtp_ts;
00207 
00208     av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
00209 
00210     s->last_rtcp_ntp_time = ntp_time;
00211     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
00212                           s1->streams[0]->time_base) + s->base_timestamp;
00213     avio_w8(s1->pb, (RTP_VERSION << 6));
00214     avio_w8(s1->pb, RTCP_SR);
00215     avio_wb16(s1->pb, 6); /* length in words - 1 */
00216     avio_wb32(s1->pb, s->ssrc);
00217     avio_wb32(s1->pb, ntp_time / 1000000);
00218     avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
00219     avio_wb32(s1->pb, rtp_ts);
00220     avio_wb32(s1->pb, s->packet_count);
00221     avio_wb32(s1->pb, s->octet_count);
00222     avio_flush(s1->pb);
00223 }
00224 
00225 /* send an rtp packet. sequence number is incremented, but the caller
00226    must update the timestamp itself */
00227 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
00228 {
00229     RTPMuxContext *s = s1->priv_data;
00230 
00231     av_dlog(s1, "rtp_send_data size=%d\n", len);
00232 
00233     /* build the RTP header */
00234     avio_w8(s1->pb, (RTP_VERSION << 6));
00235     avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
00236     avio_wb16(s1->pb, s->seq);
00237     avio_wb32(s1->pb, s->timestamp);
00238     avio_wb32(s1->pb, s->ssrc);
00239 
00240     avio_write(s1->pb, buf1, len);
00241     avio_flush(s1->pb);
00242 
00243     s->seq++;
00244     s->octet_count += len;
00245     s->packet_count++;
00246 }
00247 
00248 /* send an integer number of samples and compute time stamp and fill
00249    the rtp send buffer before sending. */
00250 static void rtp_send_samples(AVFormatContext *s1,
00251                              const uint8_t *buf1, int size, int sample_size)
00252 {
00253     RTPMuxContext *s = s1->priv_data;
00254     int len, max_packet_size, n;
00255 
00256     max_packet_size = (s->max_payload_size / sample_size) * sample_size;
00257     /* not needed, but who nows */
00258     if ((size % sample_size) != 0)
00259         av_abort();
00260     n = 0;
00261     while (size > 0) {
00262         s->buf_ptr = s->buf;
00263         len = FFMIN(max_packet_size, size);
00264 
00265         /* copy data */
00266         memcpy(s->buf_ptr, buf1, len);
00267         s->buf_ptr += len;
00268         buf1 += len;
00269         size -= len;
00270         s->timestamp = s->cur_timestamp + n / sample_size;
00271         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
00272         n += (s->buf_ptr - s->buf);
00273     }
00274 }
00275 
00276 static void rtp_send_mpegaudio(AVFormatContext *s1,
00277                                const uint8_t *buf1, int size)
00278 {
00279     RTPMuxContext *s = s1->priv_data;
00280     int len, count, max_packet_size;
00281 
00282     max_packet_size = s->max_payload_size;
00283 
00284     /* test if we must flush because not enough space */
00285     len = (s->buf_ptr - s->buf);
00286     if ((len + size) > max_packet_size) {
00287         if (len > 4) {
00288             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
00289             s->buf_ptr = s->buf + 4;
00290         }
00291     }
00292     if (s->buf_ptr == s->buf + 4) {
00293         s->timestamp = s->cur_timestamp;
00294     }
00295 
00296     /* add the packet */
00297     if (size > max_packet_size) {
00298         /* big packet: fragment */
00299         count = 0;
00300         while (size > 0) {
00301             len = max_packet_size - 4;
00302             if (len > size)
00303                 len = size;
00304             /* build fragmented packet */
00305             s->buf[0] = 0;
00306             s->buf[1] = 0;
00307             s->buf[2] = count >> 8;
00308             s->buf[3] = count;
00309             memcpy(s->buf + 4, buf1, len);
00310             ff_rtp_send_data(s1, s->buf, len + 4, 0);
00311             size -= len;
00312             buf1 += len;
00313             count += len;
00314         }
00315     } else {
00316         if (s->buf_ptr == s->buf + 4) {
00317             /* no fragmentation possible */
00318             s->buf[0] = 0;
00319             s->buf[1] = 0;
00320             s->buf[2] = 0;
00321             s->buf[3] = 0;
00322         }
00323         memcpy(s->buf_ptr, buf1, size);
00324         s->buf_ptr += size;
00325     }
00326 }
00327 
00328 static void rtp_send_raw(AVFormatContext *s1,
00329                          const uint8_t *buf1, int size)
00330 {
00331     RTPMuxContext *s = s1->priv_data;
00332     int len, max_packet_size;
00333 
00334     max_packet_size = s->max_payload_size;
00335 
00336     while (size > 0) {
00337         len = max_packet_size;
00338         if (len > size)
00339             len = size;
00340 
00341         s->timestamp = s->cur_timestamp;
00342         ff_rtp_send_data(s1, buf1, len, (len == size));
00343 
00344         buf1 += len;
00345         size -= len;
00346     }
00347 }
00348 
00349 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
00350 static void rtp_send_mpegts_raw(AVFormatContext *s1,
00351                                 const uint8_t *buf1, int size)
00352 {
00353     RTPMuxContext *s = s1->priv_data;
00354     int len, out_len;
00355 
00356     while (size >= TS_PACKET_SIZE) {
00357         len = s->max_payload_size - (s->buf_ptr - s->buf);
00358         if (len > size)
00359             len = size;
00360         memcpy(s->buf_ptr, buf1, len);
00361         buf1 += len;
00362         size -= len;
00363         s->buf_ptr += len;
00364 
00365         out_len = s->buf_ptr - s->buf;
00366         if (out_len >= s->max_payload_size) {
00367             ff_rtp_send_data(s1, s->buf, out_len, 0);
00368             s->buf_ptr = s->buf;
00369         }
00370     }
00371 }
00372 
00373 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
00374 {
00375     RTPMuxContext *s = s1->priv_data;
00376     AVStream *st = s1->streams[0];
00377     int rtcp_bytes;
00378     int size= pkt->size;
00379 
00380     av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
00381 
00382     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
00383         RTCP_TX_RATIO_DEN;
00384     if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
00385                            (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
00386         rtcp_send_sr(s1, ff_ntp_time());
00387         s->last_octet_count = s->octet_count;
00388         s->first_packet = 0;
00389     }
00390     s->cur_timestamp = s->base_timestamp + pkt->pts;
00391 
00392     switch(st->codec->codec_id) {
00393     case CODEC_ID_PCM_MULAW:
00394     case CODEC_ID_PCM_ALAW:
00395     case CODEC_ID_PCM_U8:
00396     case CODEC_ID_PCM_S8:
00397         rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
00398         break;
00399     case CODEC_ID_PCM_U16BE:
00400     case CODEC_ID_PCM_U16LE:
00401     case CODEC_ID_PCM_S16BE:
00402     case CODEC_ID_PCM_S16LE:
00403         rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
00404         break;
00405     case CODEC_ID_ADPCM_G722:
00406         /* The actual sample size is half a byte per sample, but since the
00407          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
00408          * the correct parameter for send_samples is 1 byte per stream clock. */
00409         rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
00410         break;
00411     case CODEC_ID_MP2:
00412     case CODEC_ID_MP3:
00413         rtp_send_mpegaudio(s1, pkt->data, size);
00414         break;
00415     case CODEC_ID_MPEG1VIDEO:
00416     case CODEC_ID_MPEG2VIDEO:
00417         ff_rtp_send_mpegvideo(s1, pkt->data, size);
00418         break;
00419     case CODEC_ID_AAC:
00420         if (s->flags & FF_RTP_FLAG_MP4A_LATM)
00421             ff_rtp_send_latm(s1, pkt->data, size);
00422         else
00423             ff_rtp_send_aac(s1, pkt->data, size);
00424         break;
00425     case CODEC_ID_AMR_NB:
00426     case CODEC_ID_AMR_WB:
00427         ff_rtp_send_amr(s1, pkt->data, size);
00428         break;
00429     case CODEC_ID_MPEG2TS:
00430         rtp_send_mpegts_raw(s1, pkt->data, size);
00431         break;
00432     case CODEC_ID_H264:
00433         ff_rtp_send_h264(s1, pkt->data, size);
00434         break;
00435     case CODEC_ID_H263:
00436     case CODEC_ID_H263P:
00437         ff_rtp_send_h263(s1, pkt->data, size);
00438         break;
00439     case CODEC_ID_VORBIS:
00440     case CODEC_ID_THEORA:
00441         ff_rtp_send_xiph(s1, pkt->data, size);
00442         break;
00443     case CODEC_ID_VP8:
00444         ff_rtp_send_vp8(s1, pkt->data, size);
00445         break;
00446     default:
00447         /* better than nothing : send the codec raw data */
00448         rtp_send_raw(s1, pkt->data, size);
00449         break;
00450     }
00451     return 0;
00452 }
00453 
00454 static int rtp_write_trailer(AVFormatContext *s1)
00455 {
00456     RTPMuxContext *s = s1->priv_data;
00457 
00458     av_freep(&s->buf);
00459 
00460     return 0;
00461 }
00462 
00463 AVOutputFormat ff_rtp_muxer = {
00464     "rtp",
00465     NULL_IF_CONFIG_SMALL("RTP output format"),
00466     NULL,
00467     NULL,
00468     sizeof(RTPMuxContext),
00469     CODEC_ID_PCM_MULAW,
00470     CODEC_ID_NONE,
00471     rtp_write_header,
00472     rtp_write_packet,
00473     rtp_write_trailer,
00474     .priv_class = &rtp_muxer_class,
00475 };

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