rtp.c 36.5 KB
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/*
 * RTP input/output format
 * Copyright (c) 2002 Fabrice Bellard.
 *
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 * This file is part of FFmpeg.
 *
 * 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
 * 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
 * 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
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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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 */
#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_h264.h"
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#include "rtp_mpv.h"
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//#define DEBUG


/* TODO: - add RTCP statistics reporting (should be optional).

         - add support for h263/mpeg4 packetized output : IDEA: send a
         buffer to 'rtp_write_packet' contains all the packets for ONE
         frame. Each packet should have a four byte header containing
         the length in big endian format (same trick as
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         'url_open_dyn_packet_buf')
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*/

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/* 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},
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  {12, "QCELP",      CODEC_TYPE_AUDIO,   CODEC_ID_QCELP, 8000, 1},
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  {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},
  {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}
};

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/* statistics functions */
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;

static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
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static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
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static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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{
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    handler->next= RTPFirstDynamicPayloadHandler;
    RTPFirstDynamicPayloadHandler= handler;
}
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void av_register_rtp_dynamic_payload_handlers(void)
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{
    register_dynamic_payload_handler(&mp4v_es_handler);
    register_dynamic_payload_handler(&mpeg4_generic_handler);
    register_dynamic_payload_handler(&ff_h264_dynamic_handler);
}
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int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
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    if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
        codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
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        codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
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        if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
            codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
        if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
            codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
        return 0;
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    }
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    return -1;
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}

int rtp_get_payload_type(AVCodecContext *codec)
{
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    int i, payload_type;
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    /* compute the payload type */
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    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;
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        }
    return payload_type;
}

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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
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{
    if (buf[1] != 200)
        return -1;
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    s->last_rtcp_ntp_time = AV_RB64(buf + 8);
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    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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    s->last_rtcp_timestamp = AV_RB32(buf + 16);
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    return 0;
}

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#define RTP_SEQ_MOD (1<<16)

/**
* called on parse open packet
*/
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
{
    memset(s, 0, sizeof(RTPStatistics));
    s->max_seq= base_sequence;
    s->probation= 1;
}

/**
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
*/
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
{
    s->max_seq= seq;
    s->cycles= 0;
    s->base_seq= seq -1;
    s->bad_seq= RTP_SEQ_MOD + 1;
    s->received= 0;
    s->expected_prior= 0;
    s->received_prior= 0;
    s->jitter= 0;
    s->transit= 0;
}

/**
* returns 1 if we should handle this packet.
*/
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
{
    uint16_t udelta= seq - s->max_seq;
    const int MAX_DROPOUT= 3000;
    const int MAX_MISORDER = 100;
    const int MIN_SEQUENTIAL = 2;

    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
    if(s->probation)
    {
        if(seq==s->max_seq + 1) {
            s->probation--;
            s->max_seq= seq;
            if(s->probation==0) {
                rtp_init_sequence(s, seq);
                s->received++;
                return 1;
            }
        } else {
            s->probation= MIN_SEQUENTIAL - 1;
            s->max_seq = seq;
        }
    } else if (udelta < MAX_DROPOUT) {
        // in order, with permissible gap
        if(seq < s->max_seq) {
            //sequence number wrapped; count antother 64k cycles
            s->cycles += RTP_SEQ_MOD;
        }
        s->max_seq= seq;
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
        // sequence made a large jump...
        if(seq==s->bad_seq) {
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
            rtp_init_sequence(s, seq);
        } else {
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
            return 0;
        }
    } else {
        // duplicate or reordered packet...
    }
    s->received++;
    return 1;
}

#if 0
/**
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
* never change.  I left this in in case someone else can see a way. (rdm)
*/
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
{
    uint32_t transit= arrival_timestamp - sent_timestamp;
    int d;
    s->transit= transit;
    d= FFABS(transit - s->transit);
    s->jitter += d - ((s->jitter + 8)>>4);
}
#endif

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int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
    ByteIOContext pb;
    uint8_t *buf;
    int len;
    int rtcp_bytes;
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    RTPStatistics *stats= &s->statistics;
    uint32_t lost;
    uint32_t extended_max;
    uint32_t expected_interval;
    uint32_t received_interval;
    uint32_t lost_interval;
    uint32_t expected;
    uint32_t fraction;
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
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    if (!s->rtp_ctx || (count < 1))
        return -1;

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    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
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    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
    s->octet_count += count;
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
        RTCP_TX_RATIO_DEN;
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
    if (rtcp_bytes < 28)
        return -1;
    s->last_octet_count = s->octet_count;

    if (url_open_dyn_buf(&pb) < 0)
        return -1;

    // Receiver Report
    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(&pb, 201);
    put_be16(&pb, 7); /* length in words - 1 */
    put_be32(&pb, s->ssrc); // our own SSRC
    put_be32(&pb, s->ssrc); // XXX: should be the server's here!
    // some placeholders we should really fill...
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    // RFC 1889/p64
    extended_max= stats->cycles + stats->max_seq;
    expected= extended_max - stats->base_seq + 1;
    lost= expected - stats->received;
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
    expected_interval= expected - stats->expected_prior;
    stats->expected_prior= expected;
    received_interval= stats->received - stats->received_prior;
    stats->received_prior= stats->received;
    lost_interval= expected_interval - received_interval;
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
    else fraction = (lost_interval<<8)/expected_interval;

    fraction= (fraction<<24) | lost;

    put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
    put_be32(&pb, extended_max); /* max sequence received */
    put_be32(&pb, stats->jitter>>4); /* jitter */

    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
    {
        put_be32(&pb, 0); /* last SR timestamp */
        put_be32(&pb, 0); /* delay since last SR */
    } else {
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;

        put_be32(&pb, middle_32_bits); /* last SR timestamp */
        put_be32(&pb, delay_since_last); /* delay since last SR */
    }
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    // CNAME
    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(&pb, 202);
    len = strlen(s->hostname);
    put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
    put_be32(&pb, s->ssrc);
    put_byte(&pb, 0x01);
    put_byte(&pb, len);
    put_buffer(&pb, s->hostname, len);
    // padding
    for (len = (6 + len) % 4; len % 4; len++) {
        put_byte(&pb, 0);
    }

    put_flush_packet(&pb);
    len = url_close_dyn_buf(&pb, &buf);
    if ((len > 0) && buf) {
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        int result;
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#if defined(DEBUG)
        printf("sending %d bytes of RR\n", len);
#endif
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        result= url_write(s->rtp_ctx, buf, len);
#if defined(DEBUG)
        printf("result from url_write: %d\n", result);
#endif
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        av_free(buf);
    }
    return 0;
}

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/**
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 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
 * MPEG2TS streams to indicate that they should be demuxed inside the
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 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
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 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
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 */
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
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{
    RTPDemuxContext *s;

    s = av_mallocz(sizeof(RTPDemuxContext));
    if (!s)
        return NULL;
    s->payload_type = payload_type;
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
    s->ic = s1;
    s->st = st;
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    s->rtp_payload_data = rtp_payload_data;
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    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
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    if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
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        s->ts = mpegts_parse_open(s->ic);
        if (s->ts == NULL) {
            av_free(s);
            return NULL;
        }
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    } else {
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        switch(st->codec->codec_id) {
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        case CODEC_ID_MPEG1VIDEO:
        case CODEC_ID_MPEG2VIDEO:
        case CODEC_ID_MP2:
        case CODEC_ID_MP3:
        case CODEC_ID_MPEG4:
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        case CODEC_ID_H264:
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            st->need_parsing = AVSTREAM_PARSE_FULL;
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            break;
        default:
            break;
        }
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    }
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    // needed to send back RTCP RR in RTSP sessions
    s->rtp_ctx = rtpc;
    gethostname(s->hostname, sizeof(s->hostname));
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    return s;
}

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static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
{
    int au_headers_length, au_header_size, i;
    GetBitContext getbitcontext;
    rtp_payload_data_t *infos;

    infos = s->rtp_payload_data;

    if (infos == NULL)
        return -1;

    /* decode the first 2 bytes where are stored the AUHeader sections
       length in bits */
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    au_headers_length = AV_RB16(buf);
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    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
      return -1;

    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;

    /* skip AU headers length section (2 bytes) */
    buf += 2;

    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);

    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
    au_header_size = infos->sizelength + infos->indexlength;
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
        return -1;

    infos->nb_au_headers = au_headers_length / au_header_size;
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);

    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
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       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
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       but does when sending the whole as one big packet...  */
    infos->au_headers[0].size = 0;
    infos->au_headers[0].index = 0;
    for (i = 0; i < infos->nb_au_headers; ++i) {
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
    }

    infos->nb_au_headers = 1;

    return 0;
}

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/**
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
 */
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
    switch(s->st->codec->codec_id) {
        case CODEC_ID_MP2:
        case CODEC_ID_MPEG1VIDEO:
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
                int64_t addend;

                int delta_timestamp;
                /* XXX: is it really necessary to unify the timestamp base ? */
                /* compute pts from timestamp with received ntp_time */
                delta_timestamp = timestamp - s->last_rtcp_timestamp;
                /* convert to 90 kHz without overflow */
                addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
                addend = (addend * 5625) >> 14;
                pkt->pts = addend + delta_timestamp;
            }
            break;
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        case CODEC_ID_AAC:
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        case CODEC_ID_H264:
        case CODEC_ID_MPEG4:
            pkt->pts = timestamp;
            break;
        default:
            /* no timestamp info yet */
            break;
    }
    pkt->stream_index = s->st->index;
}

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/**
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 * Parse an RTP or RTCP packet directly sent as a buffer.
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 * @param s RTP parse context.
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 * @param pkt returned packet
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 * @param buf input buffer or NULL to read the next packets
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 * @param len buffer len
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 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
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 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
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 */
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int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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                     const uint8_t *buf, int len)
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{
    unsigned int ssrc, h;
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    int payload_type, seq, ret;
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    AVStream *st;
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    uint32_t timestamp;
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    int rv= 0;
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    if (!buf) {
        /* return the next packets, if any */
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        if(s->st && s->parse_packet) {
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            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
            rv= s->parse_packet(s, pkt, &timestamp, NULL, 0);
            finalize_packet(s, pkt, timestamp);
            return rv;
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        } else {
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            // TODO: Move to a dynamic packet handler (like above)
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            if (s->read_buf_index >= s->read_buf_size)
                return -1;
            ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
                                      s->read_buf_size - s->read_buf_index);
            if (ret < 0)
                return -1;
            s->read_buf_index += ret;
            if (s->read_buf_index < s->read_buf_size)
                return 1;
            else
                return 0;
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        }
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    }

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    if (len < 12)
        return -1;

    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
        return -1;
    if (buf[1] >= 200 && buf[1] <= 204) {
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        rtcp_parse_packet(s, buf, len);
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        return -1;
    }
    payload_type = buf[1] & 0x7f;
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    seq  = AV_RB16(buf + 2);
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    timestamp = AV_RB32(buf + 4);
    ssrc = AV_RB32(buf + 8);
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    /* store the ssrc in the RTPDemuxContext */
    s->ssrc = ssrc;
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    /* NOTE: we can handle only one payload type */
    if (s->payload_type != payload_type)
        return -1;
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    st = s->st;
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    // only do something with this if all the rtp checks pass...
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
    {
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        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
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               payload_type, seq, ((s->seq + 1) & 0xffff));
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        return -1;
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    }
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    s->seq = seq;
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    len -= 12;
    buf += 12;
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    if (!st) {
        /* specific MPEG2TS demux support */
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
        if (ret < 0)
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            return -1;
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        if (ret < len) {
            s->read_buf_size = len - ret;
            memcpy(s->buf, buf + ret, s->read_buf_size);
            s->read_buf_index = 0;
            return 1;
        }
    } else {
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        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
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        switch(st->codec->codec_id) {
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        case CODEC_ID_MP2:
            /* better than nothing: skip mpeg audio RTP header */
            if (len <= 4)
                return -1;
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            h = AV_RB32(buf);
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            len -= 4;
            buf += 4;
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        case CODEC_ID_MPEG1VIDEO:
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            /* better than nothing: skip mpeg video RTP header */
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            if (len <= 4)
                return -1;
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            h = AV_RB32(buf);
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            buf += 4;
            len -= 4;
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            if (h & (1 << 26)) {
                /* mpeg2 */
                if (len <= 4)
                    return -1;
                buf += 4;
                len -= 4;
            }
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
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            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
            // timestamps.
            // TODO: Put this into a dynamic packet handler...
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        case CODEC_ID_AAC:
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            if (rtp_parse_mp4_au(s, buf))
                return -1;
            {
                rtp_payload_data_t *infos = s->rtp_payload_data;
                if (infos == NULL)
                    return -1;
                buf += infos->au_headers_length_bytes + 2;
                len -= infos->au_headers_length_bytes + 2;

                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
                    one au_header */
                av_new_packet(pkt, infos->au_headers[0].size);
                memcpy(pkt->data, buf, infos->au_headers[0].size);
                buf += infos->au_headers[0].size;
                len -= infos->au_headers[0].size;
            }
            s->read_buf_size = len;
            s->buf_ptr = buf;
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            rv= 0;
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            break;
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        default:
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            if(s->parse_packet) {
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                rv= s->parse_packet(s, pkt, &timestamp, buf, len);
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            } else {
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                av_new_packet(pkt, len);
                memcpy(pkt->data, buf, len);
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            }
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            break;
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        }
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        // now perform timestamp things....
        finalize_packet(s, pkt, timestamp);
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    }
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    return rv;
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}

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void rtp_parse_close(RTPDemuxContext *s)
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{
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    // TODO: fold this into the protocol specific data fields.
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    if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
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        mpegts_parse_close(s->ts);
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    }
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    av_free(s);
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}

/* rtp output */

static int rtp_write_header(AVFormatContext *s1)
{
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    RTPDemuxContext *s = s1->priv_data;
    int payload_type, max_packet_size, n;
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    AVStream *st;

    if (s1->nb_streams != 1)
        return -1;
    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 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
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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->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);
    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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    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;
        s->cur_timestamp = 0;
        break;
    case CODEC_ID_MPEG1VIDEO:
        s->cur_timestamp = 0;
        break;
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    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;
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    default:
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        if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
            av_set_pts_info(st, 32, 1, st->codec->sample_rate);
        }
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        s->buf_ptr = s->buf;
        break;
    }

    return 0;
}

/* 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;
    rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
                          s1->streams[0]->time_base) + s->base_timestamp;
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    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);
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    put_be32(&s1->pb, ntp_time / 1000000);
    put_be32(&s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
    put_be32(&s1->pb, rtp_ts);
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    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 */
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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
    printf("rtp_send_data size=%d\n", len);
#endif

    /* build the RTP header */
    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);
    put_be32(&s1->pb, s->timestamp);
    put_be32(&s1->pb, s->ssrc);
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    put_buffer(&s1->pb, buf1, len);
    put_flush_packet(&s1->pb);
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    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,
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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;

    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
    /* not needed, but who nows */
    if ((size % sample_size) != 0)
        av_abort();
    while (size > 0) {
        len = (max_packet_size - (s->buf_ptr - s->buf));
        if (len > size)
            len = size;

        /* copy data */
        memcpy(s->buf_ptr, buf1, len);
        s->buf_ptr += len;
        buf1 += len;
        size -= len;
        n = (s->buf_ptr - s->buf);
        /* if buffer full, then send it */
        if (n >= max_packet_size) {
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            ff_rtp_send_data(s1, s->buf, n, 0);
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            s->buf_ptr = s->buf;
            /* update timestamp */
            s->timestamp += n / sample_size;
        }
    }
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}
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/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
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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    AVStream *st = s1->streams[0];
    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) {
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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;
            /* 90 KHz time stamp */
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            s->timestamp = s->base_timestamp +
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                (s->cur_timestamp * 90000LL) / st->codec->sample_rate;
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        }
    }

    /* 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);
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            ff_rtp_send_data(s1, s->buf, len + 4, 0);
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            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;
    }
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    s->cur_timestamp += st->codec->frame_size;
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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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    AVStream *st = s1->streams[0];
    int len, max_packet_size;

    max_packet_size = s->max_payload_size;

    while (size > 0) {
        len = max_packet_size;
        if (len > size)
            len = size;

        /* 90 KHz time stamp */
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        s->timestamp = s->base_timestamp +
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            av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
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        ff_rtp_send_data(s1, buf1, len, (len == size));
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        buf1 += len;
        size -= len;
    }
    s->cur_timestamp++;
}

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/* 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;
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        out_len = s->buf_ptr - s->buf;
        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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/* 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];
    int rtcp_bytes;
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    int size= pkt->size;
    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

    /* 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;
    if (s->first_packet || rtcp_bytes >= 28) {
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        rtcp_send_sr(s1, av_gettime());
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        s->last_octet_count = s->octet_count;
        s->first_packet = 0;
    }

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    switch(st->codec->codec_id) {
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    case CODEC_ID_PCM_MULAW:
    case CODEC_ID_PCM_ALAW:
    case CODEC_ID_PCM_U8:
    case CODEC_ID_PCM_S8:
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        rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
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        break;
    case CODEC_ID_PCM_U16BE:
    case CODEC_ID_PCM_U16LE:
    case CODEC_ID_PCM_S16BE:
    case CODEC_ID_PCM_S16LE:
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        rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
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        break;
    case CODEC_ID_MP2:
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    case CODEC_ID_MP3:
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        rtp_send_mpegaudio(s1, buf1, size);
        break;
    case CODEC_ID_MPEG1VIDEO:
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        ff_rtp_send_mpegvideo(s1, buf1, size);
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        break;
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    case CODEC_ID_MPEG2TS:
        rtp_send_mpegts_raw(s1, buf1, size);
        break;
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    default:
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        /* better than nothing : send the codec raw data */
        rtp_send_raw(s1, buf1, size);
        break;
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    }
    return 0;
}

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