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ffmpeg / libavformat / rtpenc.c @ a23c9c4a

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1 83a0d387 Luca Abeni
/*
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 * RTP output format
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 * Copyright (c) 2002 Fabrice Bellard.
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "avformat.h"
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#include "mpegts.h"
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#include "bitstream.h"
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#include <unistd.h>
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#include "network.h"
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#include "rtp_internal.h"
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#include "rtp_mpv.h"
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#include "rtp_aac.h"
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#include "rtp_h264.h"
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//#define DEBUG
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#define RTCP_SR_SIZE 28
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static int rtp_write_header(AVFormatContext *s1)
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{
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    RTPDemuxContext *s = s1->priv_data;
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    int payload_type, max_packet_size, n;
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    AVStream *st;
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    if (s1->nb_streams != 1)
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        return -1;
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    st = s1->streams[0];
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    payload_type = rtp_get_payload_type(st->codec);
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    if (payload_type < 0)
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        payload_type = RTP_PT_PRIVATE; /* private payload type */
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    s->payload_type = payload_type;
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// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
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    s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
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    s->timestamp = s->base_timestamp;
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    s->cur_timestamp = 0;
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    s->ssrc = 0; /* FIXME: was random(), what should this be? */
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    s->first_packet = 1;
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    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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    max_packet_size = url_fget_max_packet_size(s1->pb);
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    if (max_packet_size <= 12)
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        return AVERROR(EIO);
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    s->max_payload_size = max_packet_size - 12;
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    s->max_frames_per_packet = 0;
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    if (s1->max_delay) {
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        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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            if (st->codec->frame_size == 0) {
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                av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
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            } else {
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                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);
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            }
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        }
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        if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
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            /* FIXME: We should round down here... */
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            s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
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        }
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    }
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    av_set_pts_info(st, 32, 1, 90000);
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    switch(st->codec->codec_id) {
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    case CODEC_ID_MP2:
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    case CODEC_ID_MP3:
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        s->buf_ptr = s->buf + 4;
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        break;
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    case CODEC_ID_MPEG1VIDEO:
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    case CODEC_ID_MPEG2VIDEO:
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        break;
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    case CODEC_ID_MPEG2TS:
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        n = s->max_payload_size / TS_PACKET_SIZE;
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        if (n < 1)
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            n = 1;
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        s->max_payload_size = n * TS_PACKET_SIZE;
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        s->buf_ptr = s->buf;
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        break;
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    case CODEC_ID_AAC:
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        s->read_buf_index = 0;
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    default:
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        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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            av_set_pts_info(st, 32, 1, st->codec->sample_rate);
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        }
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        s->buf_ptr = s->buf;
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        break;
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    }
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    return 0;
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}
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/* send an rtcp sender report packet */
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static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
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{
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    RTPDemuxContext *s = s1->priv_data;
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    uint32_t rtp_ts;
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#if defined(DEBUG)
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    printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
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#endif
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    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
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    s->last_rtcp_ntp_time = ntp_time;
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    rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
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                          s1->streams[0]->time_base) + s->base_timestamp;
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    put_byte(s1->pb, (RTP_VERSION << 6));
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    put_byte(s1->pb, 200);
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    put_be16(s1->pb, 6); /* length in words - 1 */
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    put_be32(s1->pb, s->ssrc);
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    put_be32(s1->pb, ntp_time / 1000000);
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    put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
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    put_be32(s1->pb, rtp_ts);
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    put_be32(s1->pb, s->packet_count);
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    put_be32(s1->pb, s->octet_count);
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    put_flush_packet(s1->pb);
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}
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/* send an rtp packet. sequence number is incremented, but the caller
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   must update the timestamp itself */
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void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
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{
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    RTPDemuxContext *s = s1->priv_data;
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#ifdef DEBUG
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    printf("rtp_send_data size=%d\n", len);
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#endif
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    /* build the RTP header */
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    put_byte(s1->pb, (RTP_VERSION << 6));
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    put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
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    put_be16(s1->pb, s->seq);
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    put_be32(s1->pb, s->timestamp);
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    put_be32(s1->pb, s->ssrc);
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    put_buffer(s1->pb, buf1, len);
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    put_flush_packet(s1->pb);
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    s->seq++;
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    s->octet_count += len;
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    s->packet_count++;
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}
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/* send an integer number of samples and compute time stamp and fill
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   the rtp send buffer before sending. */
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static void rtp_send_samples(AVFormatContext *s1,
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                             const uint8_t *buf1, int size, int sample_size)
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{
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    RTPDemuxContext *s = s1->priv_data;
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    int len, max_packet_size, n;
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    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
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    /* not needed, but who nows */
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    if ((size % sample_size) != 0)
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        av_abort();
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    n = 0;
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    while (size > 0) {
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        s->buf_ptr = s->buf;
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        len = FFMIN(max_packet_size, size);
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        /* copy data */
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        memcpy(s->buf_ptr, buf1, len);
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        s->buf_ptr += len;
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        buf1 += len;
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        size -= len;
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        s->timestamp = s->cur_timestamp + n / sample_size;
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        ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
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        n += (s->buf_ptr - s->buf);
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    }
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}
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/* NOTE: we suppose that exactly one frame is given as argument here */
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/* XXX: test it */
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static void rtp_send_mpegaudio(AVFormatContext *s1,
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                               const uint8_t *buf1, int size)
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{
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    RTPDemuxContext *s = s1->priv_data;
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    int len, count, max_packet_size;
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    max_packet_size = s->max_payload_size;
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    /* test if we must flush because not enough space */
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    len = (s->buf_ptr - s->buf);
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    if ((len + size) > max_packet_size) {
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        if (len > 4) {
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            ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
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            s->buf_ptr = s->buf + 4;
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        }
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    }
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    if (s->buf_ptr == s->buf + 4) {
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        s->timestamp = s->cur_timestamp;
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    }
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    /* add the packet */
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    if (size > max_packet_size) {
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        /* big packet: fragment */
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        count = 0;
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        while (size > 0) {
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            len = max_packet_size - 4;
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            if (len > size)
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                len = size;
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            /* build fragmented packet */
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            s->buf[0] = 0;
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            s->buf[1] = 0;
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            s->buf[2] = count >> 8;
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            s->buf[3] = count;
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            memcpy(s->buf + 4, buf1, len);
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            ff_rtp_send_data(s1, s->buf, len + 4, 0);
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            size -= len;
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            buf1 += len;
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            count += len;
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        }
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    } else {
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        if (s->buf_ptr == s->buf + 4) {
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            /* no fragmentation possible */
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            s->buf[0] = 0;
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            s->buf[1] = 0;
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            s->buf[2] = 0;
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            s->buf[3] = 0;
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        }
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        memcpy(s->buf_ptr, buf1, size);
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        s->buf_ptr += size;
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    }
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}
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static void rtp_send_raw(AVFormatContext *s1,
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                         const uint8_t *buf1, int size)
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{
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    RTPDemuxContext *s = s1->priv_data;
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    int len, max_packet_size;
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    max_packet_size = s->max_payload_size;
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    while (size > 0) {
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        len = max_packet_size;
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        if (len > size)
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            len = size;
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        s->timestamp = s->cur_timestamp;
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        ff_rtp_send_data(s1, buf1, len, (len == size));
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        buf1 += len;
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        size -= len;
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    }
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}
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/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
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static void rtp_send_mpegts_raw(AVFormatContext *s1,
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                                const uint8_t *buf1, int size)
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{
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    RTPDemuxContext *s = s1->priv_data;
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    int len, out_len;
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    while (size >= TS_PACKET_SIZE) {
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        len = s->max_payload_size - (s->buf_ptr - s->buf);
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        if (len > size)
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            len = size;
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        memcpy(s->buf_ptr, buf1, len);
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        buf1 += len;
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        size -= len;
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        s->buf_ptr += len;
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        out_len = s->buf_ptr - s->buf;
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        if (out_len >= s->max_payload_size) {
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            ff_rtp_send_data(s1, s->buf, out_len, 0);
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            s->buf_ptr = s->buf;
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        }
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    }
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}
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/* write an RTP packet. 'buf1' must contain a single specific frame. */
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static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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    RTPDemuxContext *s = s1->priv_data;
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    AVStream *st = s1->streams[0];
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    int rtcp_bytes;
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    int size= pkt->size;
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    uint8_t *buf1= pkt->data;
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#ifdef DEBUG
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    printf("%d: write len=%d\n", pkt->stream_index, size);
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#endif
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    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
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    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
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        RTCP_TX_RATIO_DEN;
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    if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
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                           (av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
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        rtcp_send_sr(s1, av_gettime());
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        s->last_octet_count = s->octet_count;
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        s->first_packet = 0;
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    }
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    s->cur_timestamp = s->base_timestamp + pkt->pts;
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    switch(st->codec->codec_id) {
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    case CODEC_ID_PCM_MULAW:
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    case CODEC_ID_PCM_ALAW:
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    case CODEC_ID_PCM_U8:
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    case CODEC_ID_PCM_S8:
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        rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
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        break;
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    case CODEC_ID_PCM_U16BE:
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    case CODEC_ID_PCM_U16LE:
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    case CODEC_ID_PCM_S16BE:
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    case CODEC_ID_PCM_S16LE:
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        rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
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        break;
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    case CODEC_ID_MP2:
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    case CODEC_ID_MP3:
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        rtp_send_mpegaudio(s1, buf1, size);
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        break;
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    case CODEC_ID_MPEG1VIDEO:
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    case CODEC_ID_MPEG2VIDEO:
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        ff_rtp_send_mpegvideo(s1, buf1, size);
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        break;
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    case CODEC_ID_AAC:
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        ff_rtp_send_aac(s1, buf1, size);
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        break;
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    case CODEC_ID_MPEG2TS:
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        rtp_send_mpegts_raw(s1, buf1, size);
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        break;
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    case CODEC_ID_H264:
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        ff_rtp_send_h264(s1, buf1, size);
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        break;
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    default:
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        /* better than nothing : send the codec raw data */
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        rtp_send_raw(s1, buf1, size);
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        break;
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    }
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    return 0;
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}
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AVOutputFormat rtp_muxer = {
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    "rtp",
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    "RTP output format",
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    NULL,
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    NULL,
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    sizeof(RTPDemuxContext),
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    CODEC_ID_PCM_MULAW,
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    CODEC_ID_NONE,
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    rtp_write_header,
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    rtp_write_packet,
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};