view rtpenc.c @ 4723:a2390c6a35e6 libavformat

Fix index generation in the way that it was supposed to be used. See the discussion in the ML thread "[PATCH] rmdec.c: merge old/new packet reading code". Over time, this code broke somewhat, e.g. seq was never actually written into (and was thus always 1, therefore the seq condition was always true), whereas it was supposed to be set to the sequence number of the video slice in case the video frame is divided over multiple RM packets (slices). The problem of this is that packets other than those containing the beginning of a video frame would be indexed as well. Secondly, flags&2 is supposed to be true for video keyframes and for these audio packets containing the start of a block. For some codecs (e.g. AAC), that is every single packet, whereas for others (e.g. cook), that is the packet containing the first of a series of scrambled packets that are to be descrambled together. Indexing any of the following would lead to incomplete and thus useless frames. Problem here is that flags would be reset to 2 to indicate that the first packet is ready to be returned, and in addition if no data was left to be returned (which is always true for the first packet), then we wouldn't actually write the index entry anyway. All in all, the idea was good and it probably worked at some point, but that is long ago. This patch should at the very least make it likely for this code to be executed again at the right times, i.e. the way it was originally intended to be used.

/*
 * RTP 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 "libavcodec/bitstream.h"
#include "avformat.h"
#include "mpegts.h"

#include <unistd.h>
#include "network.h"

#include "rtpenc.h"

//#define DEBUG

#define RTCP_SR_SIZE 28
#define NTP_OFFSET 2208988800ULL
#define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)

static uint64_t ntp_time(void)
{
  return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
}

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 */
    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_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;
}

/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{
    RTPMuxContext *s = s1->priv_data;
    uint32_t rtp_ts;

    dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);

    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, (AVRational){1, 1000000},
                          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)
{
    RTPMuxContext *s = s1->priv_data;

    dprintf(s1, "rtp_send_data size=%d\n", len);

    /* 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)
{
    RTPMuxContext *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)
{
    RTPMuxContext *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)
{
    RTPMuxContext *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)
{
    RTPMuxContext *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)
{
    RTPMuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int rtcp_bytes;
    int size= pkt->size;
    uint8_t *buf1= pkt->data;

    dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);

    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) &&
                           (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
        rtcp_send_sr(s1, ntp_time());
        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;
    case CODEC_ID_H264:
        ff_rtp_send_h264(s1, buf1, size);
        break;
    default:
        /* better than nothing : send the codec raw data */
        rtp_send_raw(s1, buf1, size);
        break;
    }
    return 0;
}

static int rtp_write_trailer(AVFormatContext *s1)
{
    RTPMuxContext *s = s1->priv_data;

    av_freep(&s->buf);

    return 0;
}

AVOutputFormat rtp_muxer = {
    "rtp",
    NULL_IF_CONFIG_SMALL("RTP output format"),
    NULL,
    NULL,
    sizeof(RTPMuxContext),
    CODEC_ID_PCM_MULAW,
    CODEC_ID_NONE,
    rtp_write_header,
    rtp_write_packet,
    rtp_write_trailer,
};