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/mathematics.h"
00026 #include "libavutil/random_seed.h"
00027 #include "libavutil/opt.h"
00028 
00029 #include "rtpenc.h"
00030 
00031 //#define DEBUG
00032 
00033 static const AVOption options[] = {
00034     FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
00035     { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
00036     { NULL },
00037 };
00038 
00039 static const AVClass rtp_muxer_class = {
00040     .class_name = "RTP muxer",
00041     .item_name  = av_default_item_name,
00042     .option     = options,
00043     .version    = LIBAVUTIL_VERSION_INT,
00044 };
00045 
00046 #define RTCP_SR_SIZE 28
00047 
00048 static int is_supported(enum CodecID id)
00049 {
00050     switch(id) {
00051     case CODEC_ID_H263:
00052     case CODEC_ID_H263P:
00053     case CODEC_ID_H264:
00054     case CODEC_ID_MPEG1VIDEO:
00055     case CODEC_ID_MPEG2VIDEO:
00056     case CODEC_ID_MPEG4:
00057     case CODEC_ID_AAC:
00058     case CODEC_ID_MP2:
00059     case CODEC_ID_MP3:
00060     case CODEC_ID_PCM_ALAW:
00061     case CODEC_ID_PCM_MULAW:
00062     case CODEC_ID_PCM_S8:
00063     case CODEC_ID_PCM_S16BE:
00064     case CODEC_ID_PCM_S16LE:
00065     case CODEC_ID_PCM_U16BE:
00066     case CODEC_ID_PCM_U16LE:
00067     case CODEC_ID_PCM_U8:
00068     case CODEC_ID_MPEG2TS:
00069     case CODEC_ID_AMR_NB:
00070     case CODEC_ID_AMR_WB:
00071     case CODEC_ID_VORBIS:
00072     case CODEC_ID_THEORA:
00073     case CODEC_ID_VP8:
00074     case CODEC_ID_ADPCM_G722:
00075     case CODEC_ID_ADPCM_G726:
00076         return 1;
00077     default:
00078         return 0;
00079     }
00080 }
00081 
00082 static int rtp_write_header(AVFormatContext *s1)
00083 {
00084     RTPMuxContext *s = s1->priv_data;
00085     int max_packet_size, n;
00086     AVStream *st;
00087 
00088     if (s1->nb_streams != 1)
00089         return -1;
00090     st = s1->streams[0];
00091     if (!is_supported(st->codec->codec_id)) {
00092         av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
00093 
00094         return -1;
00095     }
00096 
00097     if (s->payload_type < 0)
00098         s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
00099     s->base_timestamp = av_get_random_seed();
00100     s->timestamp = s->base_timestamp;
00101     s->cur_timestamp = 0;
00102     s->ssrc = av_get_random_seed();
00103     s->first_packet = 1;
00104     s->first_rtcp_ntp_time = ff_ntp_time();
00105     if (s1->start_time_realtime)
00106         /* Round the NTP time to whole milliseconds. */
00107         s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
00108                                  NTP_OFFSET_US;
00109 
00110     max_packet_size = s1->pb->max_packet_size;
00111     if (max_packet_size <= 12)
00112         return AVERROR(EIO);
00113     s->buf = av_malloc(max_packet_size);
00114     if (s->buf == NULL) {
00115         return AVERROR(ENOMEM);
00116     }
00117     s->max_payload_size = max_packet_size - 12;
00118 
00119     s->max_frames_per_packet = 0;
00120     if (s1->max_delay) {
00121         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
00122             if (st->codec->frame_size == 0) {
00123                 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
00124             } else {
00125                 s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * (int64_t)st->codec->frame_size, AV_ROUND_DOWN);
00126             }
00127         }
00128         if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
00129             /* FIXME: We should round down here... */
00130             s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
00131         }
00132     }
00133 
00134     avpriv_set_pts_info(st, 32, 1, 90000);
00135     switch(st->codec->codec_id) {
00136     case CODEC_ID_MP2:
00137     case CODEC_ID_MP3:
00138         s->buf_ptr = s->buf + 4;
00139         break;
00140     case CODEC_ID_MPEG1VIDEO:
00141     case CODEC_ID_MPEG2VIDEO:
00142         break;
00143     case CODEC_ID_MPEG2TS:
00144         n = s->max_payload_size / TS_PACKET_SIZE;
00145         if (n < 1)
00146             n = 1;
00147         s->max_payload_size = n * TS_PACKET_SIZE;
00148         s->buf_ptr = s->buf;
00149         break;
00150     case CODEC_ID_H264:
00151         /* check for H.264 MP4 syntax */
00152         if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
00153             s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
00154         }
00155         break;
00156     case CODEC_ID_VORBIS:
00157     case CODEC_ID_THEORA:
00158         if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
00159         s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
00160         s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
00161         s->num_frames = 0;
00162         goto defaultcase;
00163     case CODEC_ID_VP8:
00164         av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
00165                                  "incompatible with the latest spec drafts.\n");
00166         break;
00167     case CODEC_ID_ADPCM_G722:
00168         /* Due to a historical error, the clock rate for G722 in RTP is
00169          * 8000, even if the sample rate is 16000. See RFC 3551. */
00170         avpriv_set_pts_info(st, 32, 1, 8000);
00171         break;
00172     case CODEC_ID_AMR_NB:
00173     case CODEC_ID_AMR_WB:
00174         if (!s->max_frames_per_packet)
00175             s->max_frames_per_packet = 12;
00176         if (st->codec->codec_id == CODEC_ID_AMR_NB)
00177             n = 31;
00178         else
00179             n = 61;
00180         /* max_header_toc_size + the largest AMR payload must fit */
00181         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
00182             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
00183             return -1;
00184         }
00185         if (st->codec->channels != 1) {
00186             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
00187             return -1;
00188         }
00189     case CODEC_ID_AAC:
00190         s->num_frames = 0;
00191     default:
00192 defaultcase:
00193         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
00194             avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
00195         }
00196         s->buf_ptr = s->buf;
00197         break;
00198     }
00199 
00200     return 0;
00201 }
00202 
00203 /* send an rtcp sender report packet */
00204 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
00205 {
00206     RTPMuxContext *s = s1->priv_data;
00207     uint32_t rtp_ts;
00208 
00209     av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
00210 
00211     s->last_rtcp_ntp_time = ntp_time;
00212     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
00213                           s1->streams[0]->time_base) + s->base_timestamp;
00214     avio_w8(s1->pb, (RTP_VERSION << 6));
00215     avio_w8(s1->pb, RTCP_SR);
00216     avio_wb16(s1->pb, 6); /* length in words - 1 */
00217     avio_wb32(s1->pb, s->ssrc);
00218     avio_wb32(s1->pb, ntp_time / 1000000);
00219     avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
00220     avio_wb32(s1->pb, rtp_ts);
00221     avio_wb32(s1->pb, s->packet_count);
00222     avio_wb32(s1->pb, s->octet_count);
00223     avio_flush(s1->pb);
00224 }
00225 
00226 /* send an rtp packet. sequence number is incremented, but the caller
00227    must update the timestamp itself */
00228 void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
00229 {
00230     RTPMuxContext *s = s1->priv_data;
00231 
00232     av_dlog(s1, "rtp_send_data size=%d\n", len);
00233 
00234     /* build the RTP header */
00235     avio_w8(s1->pb, (RTP_VERSION << 6));
00236     avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
00237     avio_wb16(s1->pb, s->seq);
00238     avio_wb32(s1->pb, s->timestamp);
00239     avio_wb32(s1->pb, s->ssrc);
00240 
00241     avio_write(s1->pb, buf1, len);
00242     avio_flush(s1->pb);
00243 
00244     s->seq++;
00245     s->octet_count += len;
00246     s->packet_count++;
00247 }
00248 
00249 /* send an integer number of samples and compute time stamp and fill
00250    the rtp send buffer before sending. */
00251 static void rtp_send_samples(AVFormatContext *s1,
00252                              const uint8_t *buf1, int size, int sample_size_bits)
00253 {
00254     RTPMuxContext *s = s1->priv_data;
00255     int len, max_packet_size, n;
00256     /* Calculate the number of bytes to get samples aligned on a byte border */
00257     int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
00258 
00259     max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
00260     /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
00261     if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
00262         av_abort();
00263     n = 0;
00264     while (size > 0) {
00265         s->buf_ptr = s->buf;
00266         len = FFMIN(max_packet_size, size);
00267 
00268         /* copy data */
00269         memcpy(s->buf_ptr, buf1, len);
00270         s->buf_ptr += len;
00271         buf1 += len;
00272         size -= len;
00273         s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
00274         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
00275         n += (s->buf_ptr - s->buf);
00276     }
00277 }
00278 
00279 static void rtp_send_mpegaudio(AVFormatContext *s1,
00280                                const uint8_t *buf1, int size)
00281 {
00282     RTPMuxContext *s = s1->priv_data;
00283     int len, count, max_packet_size;
00284 
00285     max_packet_size = s->max_payload_size;
00286 
00287     /* test if we must flush because not enough space */
00288     len = (s->buf_ptr - s->buf);
00289     if ((len + size) > max_packet_size) {
00290         if (len > 4) {
00291             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
00292             s->buf_ptr = s->buf + 4;
00293         }
00294     }
00295     if (s->buf_ptr == s->buf + 4) {
00296         s->timestamp = s->cur_timestamp;
00297     }
00298 
00299     /* add the packet */
00300     if (size > max_packet_size) {
00301         /* big packet: fragment */
00302         count = 0;
00303         while (size > 0) {
00304             len = max_packet_size - 4;
00305             if (len > size)
00306                 len = size;
00307             /* build fragmented packet */
00308             s->buf[0] = 0;
00309             s->buf[1] = 0;
00310             s->buf[2] = count >> 8;
00311             s->buf[3] = count;
00312             memcpy(s->buf + 4, buf1, len);
00313             ff_rtp_send_data(s1, s->buf, len + 4, 0);
00314             size -= len;
00315             buf1 += len;
00316             count += len;
00317         }
00318     } else {
00319         if (s->buf_ptr == s->buf + 4) {
00320             /* no fragmentation possible */
00321             s->buf[0] = 0;
00322             s->buf[1] = 0;
00323             s->buf[2] = 0;
00324             s->buf[3] = 0;
00325         }
00326         memcpy(s->buf_ptr, buf1, size);
00327         s->buf_ptr += size;
00328     }
00329 }
00330 
00331 static void rtp_send_raw(AVFormatContext *s1,
00332                          const uint8_t *buf1, int size)
00333 {
00334     RTPMuxContext *s = s1->priv_data;
00335     int len, max_packet_size;
00336 
00337     max_packet_size = s->max_payload_size;
00338 
00339     while (size > 0) {
00340         len = max_packet_size;
00341         if (len > size)
00342             len = size;
00343 
00344         s->timestamp = s->cur_timestamp;
00345         ff_rtp_send_data(s1, buf1, len, (len == size));
00346 
00347         buf1 += len;
00348         size -= len;
00349     }
00350 }
00351 
00352 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
00353 static void rtp_send_mpegts_raw(AVFormatContext *s1,
00354                                 const uint8_t *buf1, int size)
00355 {
00356     RTPMuxContext *s = s1->priv_data;
00357     int len, out_len;
00358 
00359     while (size >= TS_PACKET_SIZE) {
00360         len = s->max_payload_size - (s->buf_ptr - s->buf);
00361         if (len > size)
00362             len = size;
00363         memcpy(s->buf_ptr, buf1, len);
00364         buf1 += len;
00365         size -= len;
00366         s->buf_ptr += len;
00367 
00368         out_len = s->buf_ptr - s->buf;
00369         if (out_len >= s->max_payload_size) {
00370             ff_rtp_send_data(s1, s->buf, out_len, 0);
00371             s->buf_ptr = s->buf;
00372         }
00373     }
00374 }
00375 
00376 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
00377 {
00378     RTPMuxContext *s = s1->priv_data;
00379     AVStream *st = s1->streams[0];
00380     int rtcp_bytes;
00381     int size= pkt->size;
00382 
00383     av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
00384 
00385     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
00386         RTCP_TX_RATIO_DEN;
00387     if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
00388                            (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
00389         rtcp_send_sr(s1, ff_ntp_time());
00390         s->last_octet_count = s->octet_count;
00391         s->first_packet = 0;
00392     }
00393     s->cur_timestamp = s->base_timestamp + pkt->pts;
00394 
00395     switch(st->codec->codec_id) {
00396     case CODEC_ID_PCM_MULAW:
00397     case CODEC_ID_PCM_ALAW:
00398     case CODEC_ID_PCM_U8:
00399     case CODEC_ID_PCM_S8:
00400         rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
00401         break;
00402     case CODEC_ID_PCM_U16BE:
00403     case CODEC_ID_PCM_U16LE:
00404     case CODEC_ID_PCM_S16BE:
00405     case CODEC_ID_PCM_S16LE:
00406         rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
00407         break;
00408     case CODEC_ID_ADPCM_G722:
00409         /* The actual sample size is half a byte per sample, but since the
00410          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
00411          * the correct parameter for send_samples_bits is 8 bits per stream
00412          * clock. */
00413         rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
00414         break;
00415     case CODEC_ID_ADPCM_G726:
00416         rtp_send_samples(s1, pkt->data, size,
00417                          st->codec->bits_per_coded_sample * st->codec->channels);
00418         break;
00419     case CODEC_ID_MP2:
00420     case CODEC_ID_MP3:
00421         rtp_send_mpegaudio(s1, pkt->data, size);
00422         break;
00423     case CODEC_ID_MPEG1VIDEO:
00424     case CODEC_ID_MPEG2VIDEO:
00425         ff_rtp_send_mpegvideo(s1, pkt->data, size);
00426         break;
00427     case CODEC_ID_AAC:
00428         if (s->flags & FF_RTP_FLAG_MP4A_LATM)
00429             ff_rtp_send_latm(s1, pkt->data, size);
00430         else
00431             ff_rtp_send_aac(s1, pkt->data, size);
00432         break;
00433     case CODEC_ID_AMR_NB:
00434     case CODEC_ID_AMR_WB:
00435         ff_rtp_send_amr(s1, pkt->data, size);
00436         break;
00437     case CODEC_ID_MPEG2TS:
00438         rtp_send_mpegts_raw(s1, pkt->data, size);
00439         break;
00440     case CODEC_ID_H264:
00441         ff_rtp_send_h264(s1, pkt->data, size);
00442         break;
00443     case CODEC_ID_H263:
00444     case CODEC_ID_H263P:
00445         ff_rtp_send_h263(s1, pkt->data, size);
00446         break;
00447     case CODEC_ID_VORBIS:
00448     case CODEC_ID_THEORA:
00449         ff_rtp_send_xiph(s1, pkt->data, size);
00450         break;
00451     case CODEC_ID_VP8:
00452         ff_rtp_send_vp8(s1, pkt->data, size);
00453         break;
00454     default:
00455         /* better than nothing : send the codec raw data */
00456         rtp_send_raw(s1, pkt->data, size);
00457         break;
00458     }
00459     return 0;
00460 }
00461 
00462 static int rtp_write_trailer(AVFormatContext *s1)
00463 {
00464     RTPMuxContext *s = s1->priv_data;
00465 
00466     av_freep(&s->buf);
00467 
00468     return 0;
00469 }
00470 
00471 AVOutputFormat ff_rtp_muxer = {
00472     .name              = "rtp",
00473     .long_name         = NULL_IF_CONFIG_SMALL("RTP output format"),
00474     .priv_data_size    = sizeof(RTPMuxContext),
00475     .audio_codec       = CODEC_ID_PCM_MULAW,
00476     .video_codec       = CODEC_ID_MPEG4,
00477     .write_header      = rtp_write_header,
00478     .write_packet      = rtp_write_packet,
00479     .write_trailer     = rtp_write_trailer,
00480     .priv_class = &rtp_muxer_class,
00481 };