libavformat/rtpenc.c - third_party/ffmpeg - Git at Google

 /*
  * 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 "avformat.h"
 #include "mpegts.h"
 #include "internal.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/random_seed.h"
 #include "libavutil/opt.h"

 #include "rtpenc.h"

 //#define DEBUG

 static const AVOption options[] = {
     FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
     { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
     { NULL },
 };

 static const AVClass rtp_muxer_class = {
     .class_name = "RTP muxer",
     .item_name  = av_default_item_name,
     .option     = options,
     .version    = LIBAVUTIL_VERSION_INT,
 };

 #define RTCP_SR_SIZE 28

 static int is_supported(enum CodecID id)
 {
     switch(id) {
     case CODEC_ID_H263:
     case CODEC_ID_H263P:
     case CODEC_ID_H264:
     case CODEC_ID_MPEG1VIDEO:
     case CODEC_ID_MPEG2VIDEO:
     case CODEC_ID_MPEG4:
     case CODEC_ID_AAC:
     case CODEC_ID_MP2:
     case CODEC_ID_MP3:
     case CODEC_ID_PCM_ALAW:
     case CODEC_ID_PCM_MULAW:
     case CODEC_ID_PCM_S8:
     case CODEC_ID_PCM_S16BE:
     case CODEC_ID_PCM_S16LE:
     case CODEC_ID_PCM_U16BE:
     case CODEC_ID_PCM_U16LE:
     case CODEC_ID_PCM_U8:
     case CODEC_ID_MPEG2TS:
     case CODEC_ID_AMR_NB:
     case CODEC_ID_AMR_WB:
     case CODEC_ID_VORBIS:
     case CODEC_ID_THEORA:
     case CODEC_ID_VP8:
     case CODEC_ID_ADPCM_G722:
     case CODEC_ID_ADPCM_G726:
         return 1;
     default:
         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 %s\n", avcodec_get_name(st->codec->codec_id));

         return -1;
     }

     if (s->payload_type < 0)
         s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
     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 = s1->pb->max_packet_size;
     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 * (int64_t)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);
         }
     }

     avpriv_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_VORBIS:
     case CODEC_ID_THEORA:
         if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
         s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
         s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
         s->num_frames = 0;
         goto defaultcase;
     case CODEC_ID_VP8:
         av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
                                  "incompatible with the latest spec drafts.\n");
         break;
     case CODEC_ID_ADPCM_G722:
         /* Due to a historical error, the clock rate for G722 in RTP is
          * 8000, even if the sample rate is 16000. See RFC 3551. */
         avpriv_set_pts_info(st, 32, 1, 8000);
         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:
 defaultcase:
         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
             avpriv_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;

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

     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;
     avio_w8(s1->pb, (RTP_VERSION << 6));
     avio_w8(s1->pb, RTCP_SR);
     avio_wb16(s1->pb, 6); /* length in words - 1 */
     avio_wb32(s1->pb, s->ssrc);
     avio_wb32(s1->pb, ntp_time / 1000000);
     avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
     avio_wb32(s1->pb, rtp_ts);
     avio_wb32(s1->pb, s->packet_count);
     avio_wb32(s1->pb, s->octet_count);
     avio_flush(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;

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

     /* build the RTP header */
     avio_w8(s1->pb, (RTP_VERSION << 6));
     avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
     avio_wb16(s1->pb, s->seq);
     avio_wb32(s1->pb, s->timestamp);
     avio_wb32(s1->pb, s->ssrc);

     avio_write(s1->pb, buf1, len);
     avio_flush(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_bits)
 {
     RTPMuxContext *s = s1->priv_data;
     int len, max_packet_size, n;
     /* Calculate the number of bytes to get samples aligned on a byte border */
     int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);

     max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
     /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
     if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 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 * 8 / sample_size_bits;
         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
         n += (s->buf_ptr - s->buf);
     }
 }

 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;
         }
     }
 }

 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;

     av_dlog(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) &&
                            (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
         rtcp_send_sr(s1, ff_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, pkt->data, size, 8 * 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, pkt->data, size, 16 * st->codec->channels);
         break;
     case CODEC_ID_ADPCM_G722:
         /* The actual sample size is half a byte per sample, but since the
          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
          * the correct parameter for send_samples_bits is 8 bits per stream
          * clock. */
         rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
         break;
     case CODEC_ID_ADPCM_G726:
         rtp_send_samples(s1, pkt->data, size,
                          st->codec->bits_per_coded_sample * st->codec->channels);
         break;
     case CODEC_ID_MP2:
     case CODEC_ID_MP3:
         rtp_send_mpegaudio(s1, pkt->data, size);
         break;
     case CODEC_ID_MPEG1VIDEO:
     case CODEC_ID_MPEG2VIDEO:
         ff_rtp_send_mpegvideo(s1, pkt->data, size);
         break;
     case CODEC_ID_AAC:
         if (s->flags & FF_RTP_FLAG_MP4A_LATM)
             ff_rtp_send_latm(s1, pkt->data, size);
         else
             ff_rtp_send_aac(s1, pkt->data, size);
         break;
     case CODEC_ID_AMR_NB:
     case CODEC_ID_AMR_WB:
         ff_rtp_send_amr(s1, pkt->data, size);
         break;
     case CODEC_ID_MPEG2TS:
         rtp_send_mpegts_raw(s1, pkt->data, size);
         break;
     case CODEC_ID_H264:
         ff_rtp_send_h264(s1, pkt->data, size);
         break;
     case CODEC_ID_H263:
     case CODEC_ID_H263P:
         ff_rtp_send_h263(s1, pkt->data, size);
         break;
     case CODEC_ID_VORBIS:
     case CODEC_ID_THEORA:
         ff_rtp_send_xiph(s1, pkt->data, size);
         break;
     case CODEC_ID_VP8:
         ff_rtp_send_vp8(s1, pkt->data, size);
         break;
     default:
         /* better than nothing : send the codec raw data */
         rtp_send_raw(s1, pkt->data, size);
         break;
     }
     return 0;
 }

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

     av_freep(&s->buf);

     return 0;
 }

 AVOutputFormat ff_rtp_muxer = {
     .name              = "rtp",
     .long_name         = NULL_IF_CONFIG_SMALL("RTP output format"),
     .priv_data_size    = sizeof(RTPMuxContext),
     .audio_codec       = CODEC_ID_PCM_MULAW,
     .video_codec       = CODEC_ID_MPEG4,
     .write_header      = rtp_write_header,
     .write_packet      = rtp_write_packet,
     .write_trailer     = rtp_write_trailer,
     .priv_class = &rtp_muxer_class,
 };
	/*
	* 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 "avformat.h"
	#include "mpegts.h"
	#include "internal.h"
	#include "libavutil/mathematics.h"
	#include "libavutil/random_seed.h"
	#include "libavutil/opt.h"

	#include "rtpenc.h"

	//#define DEBUG

	static const AVOption options[] = {
	FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
	{ "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
	{ NULL },
	};

	static const AVClass rtp_muxer_class = {
	.class_name = "RTP muxer",
	.item_name = av_default_item_name,
	.option = options,
	.version = LIBAVUTIL_VERSION_INT,
	};

	#define RTCP_SR_SIZE 28

	static int is_supported(enum CodecID id)
	{
	switch(id) {
	case CODEC_ID_H263:
	case CODEC_ID_H263P:
	case CODEC_ID_H264:
	case CODEC_ID_MPEG1VIDEO:
	case CODEC_ID_MPEG2VIDEO:
	case CODEC_ID_MPEG4:
	case CODEC_ID_AAC:
	case CODEC_ID_MP2:
	case CODEC_ID_MP3:
	case CODEC_ID_PCM_ALAW:
	case CODEC_ID_PCM_MULAW:
	case CODEC_ID_PCM_S8:
	case CODEC_ID_PCM_S16BE:
	case CODEC_ID_PCM_S16LE:
	case CODEC_ID_PCM_U16BE:
	case CODEC_ID_PCM_U16LE:
	case CODEC_ID_PCM_U8:
	case CODEC_ID_MPEG2TS:
	case CODEC_ID_AMR_NB:
	case CODEC_ID_AMR_WB:
	case CODEC_ID_VORBIS:
	case CODEC_ID_THEORA:
	case CODEC_ID_VP8:
	case CODEC_ID_ADPCM_G722:
	case CODEC_ID_ADPCM_G726:
	return 1;
	default:
	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 %s\n", avcodec_get_name(st->codec->codec_id));

	return -1;
	}

	if (s->payload_type < 0)
	s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
	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 = s1->pb->max_packet_size;
	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 * (int64_t)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);
	}
	}

	avpriv_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_VORBIS:
	case CODEC_ID_THEORA:
	if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
	s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
	s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
	s->num_frames = 0;
	goto defaultcase;
	case CODEC_ID_VP8:
	av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
	"incompatible with the latest spec drafts.\n");
	break;
	case CODEC_ID_ADPCM_G722:
	/* Due to a historical error, the clock rate for G722 in RTP is
	* 8000, even if the sample rate is 16000. See RFC 3551. */
	avpriv_set_pts_info(st, 32, 1, 8000);
	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:
	defaultcase:
	if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
	avpriv_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;

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

	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;
	avio_w8(s1->pb, (RTP_VERSION << 6));
	avio_w8(s1->pb, RTCP_SR);
	avio_wb16(s1->pb, 6); /* length in words - 1 */
	avio_wb32(s1->pb, s->ssrc);
	avio_wb32(s1->pb, ntp_time / 1000000);
	avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
	avio_wb32(s1->pb, rtp_ts);
	avio_wb32(s1->pb, s->packet_count);
	avio_wb32(s1->pb, s->octet_count);
	avio_flush(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;

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

	/* build the RTP header */
	avio_w8(s1->pb, (RTP_VERSION << 6));
	avio_w8(s1->pb, (s->payload_type & 0x7f) \| ((m & 0x01) << 7));
	avio_wb16(s1->pb, s->seq);
	avio_wb32(s1->pb, s->timestamp);
	avio_wb32(s1->pb, s->ssrc);

	avio_write(s1->pb, buf1, len);
	avio_flush(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_bits)
	{
	RTPMuxContext *s = s1->priv_data;
	int len, max_packet_size, n;
	/* Calculate the number of bytes to get samples aligned on a byte border */
	int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);

	max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
	/* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
	if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 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 * 8 / sample_size_bits;
	ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
	n += (s->buf_ptr - s->buf);
	}
	}

	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;
	}
	}
	}

	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;

	av_dlog(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) &&
	(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
	rtcp_send_sr(s1, ff_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, pkt->data, size, 8 * 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, pkt->data, size, 16 * st->codec->channels);
	break;
	case CODEC_ID_ADPCM_G722:
	/* The actual sample size is half a byte per sample, but since the
	* stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
	* the correct parameter for send_samples_bits is 8 bits per stream
	* clock. */
	rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
	break;
	case CODEC_ID_ADPCM_G726:
	rtp_send_samples(s1, pkt->data, size,
	st->codec->bits_per_coded_sample * st->codec->channels);
	break;
	case CODEC_ID_MP2:
	case CODEC_ID_MP3:
	rtp_send_mpegaudio(s1, pkt->data, size);
	break;
	case CODEC_ID_MPEG1VIDEO:
	case CODEC_ID_MPEG2VIDEO:
	ff_rtp_send_mpegvideo(s1, pkt->data, size);
	break;
	case CODEC_ID_AAC:
	if (s->flags & FF_RTP_FLAG_MP4A_LATM)
	ff_rtp_send_latm(s1, pkt->data, size);
	else
	ff_rtp_send_aac(s1, pkt->data, size);
	break;
	case CODEC_ID_AMR_NB:
	case CODEC_ID_AMR_WB:
	ff_rtp_send_amr(s1, pkt->data, size);
	break;
	case CODEC_ID_MPEG2TS:
	rtp_send_mpegts_raw(s1, pkt->data, size);
	break;
	case CODEC_ID_H264:
	ff_rtp_send_h264(s1, pkt->data, size);
	break;
	case CODEC_ID_H263:
	case CODEC_ID_H263P:
	ff_rtp_send_h263(s1, pkt->data, size);
	break;
	case CODEC_ID_VORBIS:
	case CODEC_ID_THEORA:
	ff_rtp_send_xiph(s1, pkt->data, size);
	break;
	case CODEC_ID_VP8:
	ff_rtp_send_vp8(s1, pkt->data, size);
	break;
	default:
	/* better than nothing : send the codec raw data */
	rtp_send_raw(s1, pkt->data, size);
	break;
	}
	return 0;
	}

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

	av_freep(&s->buf);

	return 0;
	}

	AVOutputFormat ff_rtp_muxer = {
	.name = "rtp",
	.long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
	.priv_data_size = sizeof(RTPMuxContext),
	.audio_codec = CODEC_ID_PCM_MULAW,
	.video_codec = CODEC_ID_MPEG4,
	.write_header = rtp_write_header,
	.write_packet = rtp_write_packet,
	.write_trailer = rtp_write_trailer,
	.priv_class = &rtp_muxer_class,
	};