static int rtp_write_header(AVFormatContext *s1) { RTPMuxContext *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 = ff_rtp_get_payload_type(st->codec); if (payload_type < 0) payload_type = RTP_PT_PRIVATE; /* private payload type */ if (s1->rtp_pt > 0) payload_type = s1->rtp_pt; /* 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 if (random_state.index == 0) av_init_random(av_gettime() + (getpid() << 16), &random_state); //s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ s->base_timestamp = av_random(&random_state); s->timestamp = s->base_timestamp; s->cur_timestamp = 0; //s->ssrc = 0; /* FIXME: was random(), what should this be? */ s->ssrc = av_random(&random_state); 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->buf = av_malloc(max_packet_size); if (s->buf == NULL) { return AVERROR(ENOMEM); } 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, (AVRational){1, 1000000}, 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->num_frames = 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; } memset(s->stock_buf, 0, sizeof(s->stock_buf)); s->stock_len = 0; return 0; }
static int rtp_write_header(AVFormatContext *s1) { RTPMuxContext *s = s1->priv_data; int max_packet_size, n; AVStream *st; if (s1->nb_streams != 1) return -1; st = s1->streams[0]; if (!is_supported(st->codec->codec_id)) { av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id); return -1; } s->payload_type = ff_rtp_get_payload_type(st->codec); if (s->payload_type < 0) s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO); s->base_timestamp = av_get_random_seed(); s->timestamp = s->base_timestamp; s->cur_timestamp = 0; s->ssrc = av_get_random_seed(); s->first_packet = 1; s->first_rtcp_ntp_time = ff_ntp_time(); if (s1->start_time_realtime) /* Round the NTP time to whole milliseconds. */ s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 + NTP_OFFSET_US; max_packet_size = url_fget_max_packet_size(s1->pb); if (max_packet_size <= 12) return AVERROR(EIO); s->buf = av_malloc(max_packet_size); if (s->buf == NULL) { return AVERROR(ENOMEM); } s->max_payload_size = max_packet_size - 12; s->max_frames_per_packet = 0; if (s1->max_delay) { if (st->codec->codec_type == AVMEDIA_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 == AVMEDIA_TYPE_VIDEO) { /* FIXME: We should round down here... */ s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, 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_H264: /* check for H.264 MP4 syntax */ if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) { s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1; } break; case CODEC_ID_AMR_NB: case CODEC_ID_AMR_WB: if (!s->max_frames_per_packet) s->max_frames_per_packet = 12; if (st->codec->codec_id == CODEC_ID_AMR_NB) n = 31; else n = 61; /* max_header_toc_size + the largest AMR payload must fit */ if (1 + s->max_frames_per_packet + n > s->max_payload_size) { av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); return -1; } if (st->codec->channels != 1) { av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); return -1; } case CODEC_ID_AAC: s->num_frames = 0; default: if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { av_set_pts_info(st, 32, 1, st->codec->sample_rate); } s->buf_ptr = s->buf; break; } return 0; }
static int rtp_write_header(AVFormatContext *s1) { RTPMuxContext *s = s1->priv_data; int payload_type, max_packet_size, n; AVStream *st; if (s1->nb_streams != 1) return -1; st = s1->streams[0]; if (!is_supported(st->codec->codec_id)) { av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id); return -1; } payload_type = ff_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->buf = av_malloc(max_packet_size); if (s->buf == NULL) { return AVERROR(ENOMEM); } 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, (AVRational){1, 1000000}, 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_AMR_NB: case CODEC_ID_AMR_WB: if (!s->max_frames_per_packet) s->max_frames_per_packet = 12; if (st->codec->codec_id == CODEC_ID_AMR_NB) n = 31; else n = 61; /* max_header_toc_size + the largest AMR payload must fit */ if (1 + s->max_frames_per_packet + n > s->max_payload_size) { av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); return -1; } if (st->codec->channels != 1) { av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); return -1; } case CODEC_ID_AAC: s->num_frames = 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; }