Mercurial > libavformat.hg
view rtp.c @ 2891:a6c922b05571 libavformat
Split the RTP demuxing functions out of rtp.c, to simplify the RTP muxer's dependencies
| author | lucabe |
|---|---|
| date | Fri, 04 Jan 2008 19:33:50 +0000 |
| parents | dd1fbe36d103 |
| children | 0d82fdf4fa94 |
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/* * RTP input/output format * Copyright (c) 2002 Fabrice Bellard. * * This file is part of FFmpeg. * * FFmpeg 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 2.1 of the License, or (at your option) any later version. * * FFmpeg 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "avformat.h" #include "mpegts.h" #include "bitstream.h" #include <unistd.h> #include "network.h" #include "rtp_internal.h" #include "rtp_mpv.h" #include "rtp_aac.h" //#define DEBUG #define RTCP_SR_SIZE 28 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */ AVRtpPayloadType_t AVRtpPayloadTypes[]= { {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1}, {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1}, {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1}, {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2}, {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1}, {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1}, {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1}, {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1}, {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1}, {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1}, {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1}, {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1}, {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG2VIDEO, 90000, -1}, {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1}, {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1}, {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1} }; int rtp_get_codec_info(AVCodecContext *codec, int payload_type) { int i = 0; for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++) if (AVRtpPayloadTypes[i].pt == payload_type) { if (AVRtpPayloadTypes[i].codec_id != CODEC_ID_NONE) { codec->codec_type = AVRtpPayloadTypes[i].codec_type; codec->codec_id = AVRtpPayloadTypes[i].codec_id; if (AVRtpPayloadTypes[i].audio_channels > 0) codec->channels = AVRtpPayloadTypes[i].audio_channels; if (AVRtpPayloadTypes[i].clock_rate > 0) codec->sample_rate = AVRtpPayloadTypes[i].clock_rate; return 0; } } return -1; } int rtp_get_payload_type(AVCodecContext *codec) { int i, payload_type; /* compute the payload type */ for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) { if (codec->codec_id == CODEC_ID_PCM_S16BE) if (codec->channels != AVRtpPayloadTypes[i].audio_channels) continue; payload_type = AVRtpPayloadTypes[i].pt; } return payload_type; } const char *ff_rtp_enc_name(int payload_type) { int i; for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++) if (AVRtpPayloadTypes[i].pt == payload_type) { return AVRtpPayloadTypes[i].enc_name; } return ""; } enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type) { int i; for (i = 0; AVRtpPayloadTypes[i].pt >= 0; i++) if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec_type == AVRtpPayloadTypes[i].codec_type)){ return AVRtpPayloadTypes[i].codec_id; } return CODEC_ID_NONE; } /* rtp output */ static int rtp_write_header(AVFormatContext *s1) { RTPDemuxContext *s = s1->priv_data; int payload_type, max_packet_size, n; 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; // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ s->timestamp = s->base_timestamp; s->cur_timestamp = 0; s->ssrc = 0; /* FIXME: was random(), what should this be? */ s->first_packet = 1; s->first_rtcp_ntp_time = AV_NOPTS_VALUE; max_packet_size = url_fget_max_packet_size(s1->pb); if (max_packet_size <= 12) return AVERROR(EIO); s->max_payload_size = max_packet_size - 12; s->max_frames_per_packet = 0; if (s1->max_delay) { if (st->codec->codec_type == CODEC_TYPE_AUDIO) { if (st->codec->frame_size == 0) { av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); } else { 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); } } if (st->codec->codec_type == CODEC_TYPE_VIDEO) { /* FIXME: We should round down here... */ s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base); } } av_set_pts_info(st, 32, 1, 90000); switch(st->codec->codec_id) { case CODEC_ID_MP2: case CODEC_ID_MP3: s->buf_ptr = s->buf + 4; break; case CODEC_ID_MPEG1VIDEO: case CODEC_ID_MPEG2VIDEO: break; case CODEC_ID_MPEG2TS: n = s->max_payload_size / TS_PACKET_SIZE; if (n < 1) n = 1; s->max_payload_size = n * TS_PACKET_SIZE; s->buf_ptr = s->buf; break; case CODEC_ID_AAC: s->read_buf_index = 0; default: if (st->codec->codec_type == CODEC_TYPE_AUDIO) { av_set_pts_info(st, 32, 1, st->codec->sample_rate); } 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) { RTPDemuxContext *s = s1->priv_data; uint32_t rtp_ts; #if defined(DEBUG) printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); #endif if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time; s->last_rtcp_ntp_time = ntp_time; rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q, s1->streams[0]->time_base) + s->base_timestamp; 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_be32(s1->pb, ntp_time / 1000000); put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); put_be32(s1->pb, rtp_ts); 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 */ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) { RTPDemuxContext *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) | ((m & 0x01) << 7)); 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) { RTPDemuxContext *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(); n = 0; while (size > 0) { s->buf_ptr = s->buf; len = FFMIN(max_packet_size, size); /* copy data */ memcpy(s->buf_ptr, buf1, len); s->buf_ptr += len; buf1 += len; size -= len; s->timestamp = s->cur_timestamp + n / sample_size; ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); n += (s->buf_ptr - s->buf); } } /* 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) { RTPDemuxContext *s = s1->priv_data; 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) { ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); s->buf_ptr = s->buf + 4; } } if (s->buf_ptr == s->buf + 4) { s->timestamp = s->cur_timestamp; } /* 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); ff_rtp_send_data(s1, s->buf, len + 4, 0); 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; } } static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size) { RTPDemuxContext *s = s1->priv_data; int len, max_packet_size; max_packet_size = s->max_payload_size; while (size > 0) { len = max_packet_size; if (len > size) len = size; s->timestamp = s->cur_timestamp; ff_rtp_send_data(s1, buf1, len, (len == size)); buf1 += len; size -= len; } } /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ static void rtp_send_mpegts_raw(AVFormatContext *s1, const uint8_t *buf1, int size) { RTPDemuxContext *s = s1->priv_data; int len, out_len; while (size >= TS_PACKET_SIZE) { len = s->max_payload_size - (s->buf_ptr - s->buf); if (len > size) len = size; memcpy(s->buf_ptr, buf1, len); buf1 += len; size -= len; s->buf_ptr += len; out_len = s->buf_ptr - s->buf; if (out_len >= s->max_payload_size) { ff_rtp_send_data(s1, s->buf, out_len, 0); s->buf_ptr = s->buf; } } } /* write an RTP packet. 'buf1' must contain a single specific frame. */ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) { RTPDemuxContext *s = s1->priv_data; AVStream *st = s1->streams[0]; int rtcp_bytes; int size= pkt->size; uint8_t *buf1= pkt->data; #ifdef DEBUG printf("%d: write len=%d\n", pkt->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 >= RTCP_SR_SIZE) && (av_gettime() - s->last_rtcp_ntp_time > 5000000))) { rtcp_send_sr(s1, av_gettime()); s->last_octet_count = s->octet_count; s->first_packet = 0; } s->cur_timestamp = s->base_timestamp + pkt->pts; 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: case CODEC_ID_MPEG2VIDEO: ff_rtp_send_mpegvideo(s1, buf1, size); break; case CODEC_ID_AAC: ff_rtp_send_aac(s1, buf1, size); break; case CODEC_ID_MPEG2TS: rtp_send_mpegts_raw(s1, buf1, size); break; default: /* better than nothing : send the codec raw data */ rtp_send_raw(s1, buf1, size); break; } return 0; } AVOutputFormat rtp_muxer = { "rtp", "RTP output format", NULL, NULL, sizeof(RTPDemuxContext), CODEC_ID_PCM_MULAW, CODEC_ID_NONE, rtp_write_header, rtp_write_packet, };