yading@11: /*
yading@11:  * RTP output format
yading@11:  * Copyright (c) 2002 Fabrice Bellard
yading@11:  *
yading@11:  * This file is part of FFmpeg.
yading@11:  *
yading@11:  * FFmpeg is free software; you can redistribute it and/or
yading@11:  * modify it under the terms of the GNU Lesser General Public
yading@11:  * License as published by the Free Software Foundation; either
yading@11:  * version 2.1 of the License, or (at your option) any later version.
yading@11:  *
yading@11:  * FFmpeg is distributed in the hope that it will be useful,
yading@11:  * but WITHOUT ANY WARRANTY; without even the implied warranty of
yading@11:  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
yading@11:  * Lesser General Public License for more details.
yading@11:  *
yading@11:  * You should have received a copy of the GNU Lesser General Public
yading@11:  * License along with FFmpeg; if not, write to the Free Software
yading@11:  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
yading@11:  */
yading@11: 
yading@11: #include "avformat.h"
yading@11: #include "mpegts.h"
yading@11: #include "internal.h"
yading@11: #include "libavutil/mathematics.h"
yading@11: #include "libavutil/random_seed.h"
yading@11: #include "libavutil/opt.h"
yading@11: 
yading@11: #include "rtpenc.h"
yading@11: 
yading@11: //#define DEBUG
yading@11: 
yading@11: static const AVOption options[] = {
yading@11:     FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
yading@11:     { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
yading@11:     { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
yading@11:     { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
yading@11:     { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
yading@11:     { NULL },
yading@11: };
yading@11: 
yading@11: static const AVClass rtp_muxer_class = {
yading@11:     .class_name = "RTP muxer",
yading@11:     .item_name  = av_default_item_name,
yading@11:     .option     = options,
yading@11:     .version    = LIBAVUTIL_VERSION_INT,
yading@11: };
yading@11: 
yading@11: #define RTCP_SR_SIZE 28
yading@11: 
yading@11: static int is_supported(enum AVCodecID id)
yading@11: {
yading@11:     switch(id) {
yading@11:     case AV_CODEC_ID_H263:
yading@11:     case AV_CODEC_ID_H263P:
yading@11:     case AV_CODEC_ID_H264:
yading@11:     case AV_CODEC_ID_MPEG1VIDEO:
yading@11:     case AV_CODEC_ID_MPEG2VIDEO:
yading@11:     case AV_CODEC_ID_MPEG4:
yading@11:     case AV_CODEC_ID_AAC:
yading@11:     case AV_CODEC_ID_MP2:
yading@11:     case AV_CODEC_ID_MP3:
yading@11:     case AV_CODEC_ID_PCM_ALAW:
yading@11:     case AV_CODEC_ID_PCM_MULAW:
yading@11:     case AV_CODEC_ID_PCM_S8:
yading@11:     case AV_CODEC_ID_PCM_S16BE:
yading@11:     case AV_CODEC_ID_PCM_S16LE:
yading@11:     case AV_CODEC_ID_PCM_U16BE:
yading@11:     case AV_CODEC_ID_PCM_U16LE:
yading@11:     case AV_CODEC_ID_PCM_U8:
yading@11:     case AV_CODEC_ID_MPEG2TS:
yading@11:     case AV_CODEC_ID_AMR_NB:
yading@11:     case AV_CODEC_ID_AMR_WB:
yading@11:     case AV_CODEC_ID_VORBIS:
yading@11:     case AV_CODEC_ID_THEORA:
yading@11:     case AV_CODEC_ID_VP8:
yading@11:     case AV_CODEC_ID_ADPCM_G722:
yading@11:     case AV_CODEC_ID_ADPCM_G726:
yading@11:     case AV_CODEC_ID_ILBC:
yading@11:     case AV_CODEC_ID_MJPEG:
yading@11:     case AV_CODEC_ID_SPEEX:
yading@11:     case AV_CODEC_ID_OPUS:
yading@11:         return 1;
yading@11:     default:
yading@11:         return 0;
yading@11:     }
yading@11: }
yading@11: 
yading@11: static int rtp_write_header(AVFormatContext *s1)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11:     int n;
yading@11:     AVStream *st;
yading@11: 
yading@11:     if (s1->nb_streams != 1) {
yading@11:         av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
yading@11:         return AVERROR(EINVAL);
yading@11:     }
yading@11:     st = s1->streams[0];
yading@11:     if (!is_supported(st->codec->codec_id)) {
yading@11:         av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
yading@11: 
yading@11:         return -1;
yading@11:     }
yading@11: 
yading@11:     if (s->payload_type < 0) {
yading@11:         /* Re-validate non-dynamic payload types */
yading@11:         if (st->id < RTP_PT_PRIVATE)
yading@11:             st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
yading@11: 
yading@11:         s->payload_type = st->id;
yading@11:     } else {
yading@11:         /* private option takes priority */
yading@11:         st->id = s->payload_type;
yading@11:     }
yading@11: 
yading@11:     s->base_timestamp = av_get_random_seed();
yading@11:     s->timestamp = s->base_timestamp;
yading@11:     s->cur_timestamp = 0;
yading@11:     if (!s->ssrc)
yading@11:         s->ssrc = av_get_random_seed();
yading@11:     s->first_packet = 1;
yading@11:     s->first_rtcp_ntp_time = ff_ntp_time();
yading@11:     if (s1->start_time_realtime)
yading@11:         /* Round the NTP time to whole milliseconds. */
yading@11:         s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
yading@11:                                  NTP_OFFSET_US;
yading@11:     // Pick a random sequence start number, but in the lower end of the
yading@11:     // available range, so that any wraparound doesn't happen immediately.
yading@11:     // (Immediate wraparound would be an issue for SRTP.)
yading@11:     if (s->seq < 0) {
yading@11:         if (st->codec->flags & CODEC_FLAG_BITEXACT) {
yading@11:             s->seq = 0;
yading@11:         } else
yading@11:             s->seq = av_get_random_seed() & 0x0fff;
yading@11:     } else
yading@11:         s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
yading@11: 
yading@11:     if (s1->packet_size) {
yading@11:         if (s1->pb->max_packet_size)
yading@11:             s1->packet_size = FFMIN(s1->packet_size,
yading@11:                                     s1->pb->max_packet_size);
yading@11:     } else
yading@11:         s1->packet_size = s1->pb->max_packet_size;
yading@11:     if (s1->packet_size <= 12) {
yading@11:         av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
yading@11:         return AVERROR(EIO);
yading@11:     }
yading@11:     s->buf = av_malloc(s1->packet_size);
yading@11:     if (s->buf == NULL) {
yading@11:         return AVERROR(ENOMEM);
yading@11:     }
yading@11:     s->max_payload_size = s1->packet_size - 12;
yading@11: 
yading@11:     s->max_frames_per_packet = 0;
yading@11:     if (s1->max_delay > 0) {
yading@11:         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
yading@11:             int frame_size = av_get_audio_frame_duration(st->codec, 0);
yading@11:             if (!frame_size)
yading@11:                 frame_size = st->codec->frame_size;
yading@11:             if (frame_size == 0) {
yading@11:                 av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
yading@11:             } else {
yading@11:                 s->max_frames_per_packet =
yading@11:                         av_rescale_q_rnd(s1->max_delay,
yading@11:                                          AV_TIME_BASE_Q,
yading@11:                                          (AVRational){ frame_size, st->codec->sample_rate },
yading@11:                                          AV_ROUND_DOWN);
yading@11:             }
yading@11:         }
yading@11:         if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
yading@11:             /* FIXME: We should round down here... */
yading@11:             s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
yading@11:         }
yading@11:     }
yading@11: 
yading@11:     avpriv_set_pts_info(st, 32, 1, 90000);
yading@11:     switch(st->codec->codec_id) {
yading@11:     case AV_CODEC_ID_MP2:
yading@11:     case AV_CODEC_ID_MP3:
yading@11:         s->buf_ptr = s->buf + 4;
yading@11:         break;
yading@11:     case AV_CODEC_ID_MPEG1VIDEO:
yading@11:     case AV_CODEC_ID_MPEG2VIDEO:
yading@11:         break;
yading@11:     case AV_CODEC_ID_MPEG2TS:
yading@11:         n = s->max_payload_size / TS_PACKET_SIZE;
yading@11:         if (n < 1)
yading@11:             n = 1;
yading@11:         s->max_payload_size = n * TS_PACKET_SIZE;
yading@11:         s->buf_ptr = s->buf;
yading@11:         break;
yading@11:     case AV_CODEC_ID_H264:
yading@11:         /* check for H.264 MP4 syntax */
yading@11:         if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
yading@11:             s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
yading@11:         }
yading@11:         break;
yading@11:     case AV_CODEC_ID_VORBIS:
yading@11:     case AV_CODEC_ID_THEORA:
yading@11:         if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
yading@11:         s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
yading@11:         s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
yading@11:         s->num_frames = 0;
yading@11:         goto defaultcase;
yading@11:     case AV_CODEC_ID_ADPCM_G722:
yading@11:         /* Due to a historical error, the clock rate for G722 in RTP is
yading@11:          * 8000, even if the sample rate is 16000. See RFC 3551. */
yading@11:         avpriv_set_pts_info(st, 32, 1, 8000);
yading@11:         break;
yading@11:     case AV_CODEC_ID_OPUS:
yading@11:         if (st->codec->channels > 2) {
yading@11:             av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
yading@11:             goto fail;
yading@11:         }
yading@11:         /* The opus RTP RFC says that all opus streams should use 48000 Hz
yading@11:          * as clock rate, since all opus sample rates can be expressed in
yading@11:          * this clock rate, and sample rate changes on the fly are supported. */
yading@11:         avpriv_set_pts_info(st, 32, 1, 48000);
yading@11:         break;
yading@11:     case AV_CODEC_ID_ILBC:
yading@11:         if (st->codec->block_align != 38 && st->codec->block_align != 50) {
yading@11:             av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
yading@11:             goto fail;
yading@11:         }
yading@11:         if (!s->max_frames_per_packet)
yading@11:             s->max_frames_per_packet = 1;
yading@11:         s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
yading@11:                                          s->max_payload_size / st->codec->block_align);
yading@11:         goto defaultcase;
yading@11:     case AV_CODEC_ID_AMR_NB:
yading@11:     case AV_CODEC_ID_AMR_WB:
yading@11:         if (!s->max_frames_per_packet)
yading@11:             s->max_frames_per_packet = 12;
yading@11:         if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
yading@11:             n = 31;
yading@11:         else
yading@11:             n = 61;
yading@11:         /* max_header_toc_size + the largest AMR payload must fit */
yading@11:         if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
yading@11:             av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
yading@11:             goto fail;
yading@11:         }
yading@11:         if (st->codec->channels != 1) {
yading@11:             av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
yading@11:             goto fail;
yading@11:         }
yading@11:     case AV_CODEC_ID_AAC:
yading@11:         s->num_frames = 0;
yading@11:     default:
yading@11: defaultcase:
yading@11:         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
yading@11:             avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
yading@11:         }
yading@11:         s->buf_ptr = s->buf;
yading@11:         break;
yading@11:     }
yading@11: 
yading@11:     return 0;
yading@11: 
yading@11: fail:
yading@11:     av_freep(&s->buf);
yading@11:     return AVERROR(EINVAL);
yading@11: }
yading@11: 
yading@11: /* send an rtcp sender report packet */
yading@11: static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11:     uint32_t rtp_ts;
yading@11: 
yading@11:     av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
yading@11: 
yading@11:     s->last_rtcp_ntp_time = ntp_time;
yading@11:     rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
yading@11:                           s1->streams[0]->time_base) + s->base_timestamp;
yading@11:     avio_w8(s1->pb, (RTP_VERSION << 6));
yading@11:     avio_w8(s1->pb, RTCP_SR);
yading@11:     avio_wb16(s1->pb, 6); /* length in words - 1 */
yading@11:     avio_wb32(s1->pb, s->ssrc);
yading@11:     avio_wb32(s1->pb, ntp_time / 1000000);
yading@11:     avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
yading@11:     avio_wb32(s1->pb, rtp_ts);
yading@11:     avio_wb32(s1->pb, s->packet_count);
yading@11:     avio_wb32(s1->pb, s->octet_count);
yading@11: 
yading@11:     if (s->cname) {
yading@11:         int len = FFMIN(strlen(s->cname), 255);
yading@11:         avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
yading@11:         avio_w8(s1->pb, RTCP_SDES);
yading@11:         avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
yading@11: 
yading@11:         avio_wb32(s1->pb, s->ssrc);
yading@11:         avio_w8(s1->pb, 0x01); /* CNAME */
yading@11:         avio_w8(s1->pb, len);
yading@11:         avio_write(s1->pb, s->cname, len);
yading@11:         avio_w8(s1->pb, 0); /* END */
yading@11:         for (len = (7 + len) % 4; len % 4; len++)
yading@11:             avio_w8(s1->pb, 0);
yading@11:     }
yading@11: 
yading@11:     avio_flush(s1->pb);
yading@11: }
yading@11: 
yading@11: /* send an rtp packet. sequence number is incremented, but the caller
yading@11:    must update the timestamp itself */
yading@11: void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11: 
yading@11:     av_dlog(s1, "rtp_send_data size=%d\n", len);
yading@11: 
yading@11:     /* build the RTP header */
yading@11:     avio_w8(s1->pb, (RTP_VERSION << 6));
yading@11:     avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
yading@11:     avio_wb16(s1->pb, s->seq);
yading@11:     avio_wb32(s1->pb, s->timestamp);
yading@11:     avio_wb32(s1->pb, s->ssrc);
yading@11: 
yading@11:     avio_write(s1->pb, buf1, len);
yading@11:     avio_flush(s1->pb);
yading@11: 
yading@11:     s->seq = (s->seq + 1) & 0xffff;
yading@11:     s->octet_count += len;
yading@11:     s->packet_count++;
yading@11: }
yading@11: 
yading@11: /* send an integer number of samples and compute time stamp and fill
yading@11:    the rtp send buffer before sending. */
yading@11: static int rtp_send_samples(AVFormatContext *s1,
yading@11:                             const uint8_t *buf1, int size, int sample_size_bits)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11:     int len, max_packet_size, n;
yading@11:     /* Calculate the number of bytes to get samples aligned on a byte border */
yading@11:     int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
yading@11: 
yading@11:     max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
yading@11:     /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
yading@11:     if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
yading@11:         return AVERROR(EINVAL);
yading@11:     n = 0;
yading@11:     while (size > 0) {
yading@11:         s->buf_ptr = s->buf;
yading@11:         len = FFMIN(max_packet_size, size);
yading@11: 
yading@11:         /* copy data */
yading@11:         memcpy(s->buf_ptr, buf1, len);
yading@11:         s->buf_ptr += len;
yading@11:         buf1 += len;
yading@11:         size -= len;
yading@11:         s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
yading@11:         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
yading@11:         n += (s->buf_ptr - s->buf);
yading@11:     }
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: static void rtp_send_mpegaudio(AVFormatContext *s1,
yading@11:                                const uint8_t *buf1, int size)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11:     int len, count, max_packet_size;
yading@11: 
yading@11:     max_packet_size = s->max_payload_size;
yading@11: 
yading@11:     /* test if we must flush because not enough space */
yading@11:     len = (s->buf_ptr - s->buf);
yading@11:     if ((len + size) > max_packet_size) {
yading@11:         if (len > 4) {
yading@11:             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
yading@11:             s->buf_ptr = s->buf + 4;
yading@11:         }
yading@11:     }
yading@11:     if (s->buf_ptr == s->buf + 4) {
yading@11:         s->timestamp = s->cur_timestamp;
yading@11:     }
yading@11: 
yading@11:     /* add the packet */
yading@11:     if (size > max_packet_size) {
yading@11:         /* big packet: fragment */
yading@11:         count = 0;
yading@11:         while (size > 0) {
yading@11:             len = max_packet_size - 4;
yading@11:             if (len > size)
yading@11:                 len = size;
yading@11:             /* build fragmented packet */
yading@11:             s->buf[0] = 0;
yading@11:             s->buf[1] = 0;
yading@11:             s->buf[2] = count >> 8;
yading@11:             s->buf[3] = count;
yading@11:             memcpy(s->buf + 4, buf1, len);
yading@11:             ff_rtp_send_data(s1, s->buf, len + 4, 0);
yading@11:             size -= len;
yading@11:             buf1 += len;
yading@11:             count += len;
yading@11:         }
yading@11:     } else {
yading@11:         if (s->buf_ptr == s->buf + 4) {
yading@11:             /* no fragmentation possible */
yading@11:             s->buf[0] = 0;
yading@11:             s->buf[1] = 0;
yading@11:             s->buf[2] = 0;
yading@11:             s->buf[3] = 0;
yading@11:         }
yading@11:         memcpy(s->buf_ptr, buf1, size);
yading@11:         s->buf_ptr += size;
yading@11:     }
yading@11: }
yading@11: 
yading@11: static void rtp_send_raw(AVFormatContext *s1,
yading@11:                          const uint8_t *buf1, int size)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11:     int len, max_packet_size;
yading@11: 
yading@11:     max_packet_size = s->max_payload_size;
yading@11: 
yading@11:     while (size > 0) {
yading@11:         len = max_packet_size;
yading@11:         if (len > size)
yading@11:             len = size;
yading@11: 
yading@11:         s->timestamp = s->cur_timestamp;
yading@11:         ff_rtp_send_data(s1, buf1, len, (len == size));
yading@11: 
yading@11:         buf1 += len;
yading@11:         size -= len;
yading@11:     }
yading@11: }
yading@11: 
yading@11: /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
yading@11: static void rtp_send_mpegts_raw(AVFormatContext *s1,
yading@11:                                 const uint8_t *buf1, int size)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11:     int len, out_len;
yading@11: 
yading@11:     while (size >= TS_PACKET_SIZE) {
yading@11:         len = s->max_payload_size - (s->buf_ptr - s->buf);
yading@11:         if (len > size)
yading@11:             len = size;
yading@11:         memcpy(s->buf_ptr, buf1, len);
yading@11:         buf1 += len;
yading@11:         size -= len;
yading@11:         s->buf_ptr += len;
yading@11: 
yading@11:         out_len = s->buf_ptr - s->buf;
yading@11:         if (out_len >= s->max_payload_size) {
yading@11:             ff_rtp_send_data(s1, s->buf, out_len, 0);
yading@11:             s->buf_ptr = s->buf;
yading@11:         }
yading@11:     }
yading@11: }
yading@11: 
yading@11: static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11:     AVStream *st = s1->streams[0];
yading@11:     int frame_duration = av_get_audio_frame_duration(st->codec, 0);
yading@11:     int frame_size = st->codec->block_align;
yading@11:     int frames = size / frame_size;
yading@11: 
yading@11:     while (frames > 0) {
yading@11:         int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
yading@11: 
yading@11:         if (!s->num_frames) {
yading@11:             s->buf_ptr = s->buf;
yading@11:             s->timestamp = s->cur_timestamp;
yading@11:         }
yading@11:         memcpy(s->buf_ptr, buf, n * frame_size);
yading@11:         frames           -= n;
yading@11:         s->num_frames    += n;
yading@11:         s->buf_ptr       += n * frame_size;
yading@11:         buf              += n * frame_size;
yading@11:         s->cur_timestamp += n * frame_duration;
yading@11: 
yading@11:         if (s->num_frames == s->max_frames_per_packet) {
yading@11:             ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
yading@11:             s->num_frames = 0;
yading@11:         }
yading@11:     }
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11:     AVStream *st = s1->streams[0];
yading@11:     int rtcp_bytes;
yading@11:     int size= pkt->size;
yading@11: 
yading@11:     av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
yading@11: 
yading@11:     rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
yading@11:         RTCP_TX_RATIO_DEN;
yading@11:     if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
yading@11:                             (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
yading@11:         !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
yading@11:         rtcp_send_sr(s1, ff_ntp_time());
yading@11:         s->last_octet_count = s->octet_count;
yading@11:         s->first_packet = 0;
yading@11:     }
yading@11:     s->cur_timestamp = s->base_timestamp + pkt->pts;
yading@11: 
yading@11:     switch(st->codec->codec_id) {
yading@11:     case AV_CODEC_ID_PCM_MULAW:
yading@11:     case AV_CODEC_ID_PCM_ALAW:
yading@11:     case AV_CODEC_ID_PCM_U8:
yading@11:     case AV_CODEC_ID_PCM_S8:
yading@11:         return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
yading@11:     case AV_CODEC_ID_PCM_U16BE:
yading@11:     case AV_CODEC_ID_PCM_U16LE:
yading@11:     case AV_CODEC_ID_PCM_S16BE:
yading@11:     case AV_CODEC_ID_PCM_S16LE:
yading@11:         return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
yading@11:     case AV_CODEC_ID_ADPCM_G722:
yading@11:         /* The actual sample size is half a byte per sample, but since the
yading@11:          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
yading@11:          * the correct parameter for send_samples_bits is 8 bits per stream
yading@11:          * clock. */
yading@11:         return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
yading@11:     case AV_CODEC_ID_ADPCM_G726:
yading@11:         return rtp_send_samples(s1, pkt->data, size,
yading@11:                                 st->codec->bits_per_coded_sample * st->codec->channels);
yading@11:     case AV_CODEC_ID_MP2:
yading@11:     case AV_CODEC_ID_MP3:
yading@11:         rtp_send_mpegaudio(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_MPEG1VIDEO:
yading@11:     case AV_CODEC_ID_MPEG2VIDEO:
yading@11:         ff_rtp_send_mpegvideo(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_AAC:
yading@11:         if (s->flags & FF_RTP_FLAG_MP4A_LATM)
yading@11:             ff_rtp_send_latm(s1, pkt->data, size);
yading@11:         else
yading@11:             ff_rtp_send_aac(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_AMR_NB:
yading@11:     case AV_CODEC_ID_AMR_WB:
yading@11:         ff_rtp_send_amr(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_MPEG2TS:
yading@11:         rtp_send_mpegts_raw(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_H264:
yading@11:         ff_rtp_send_h264(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_H263:
yading@11:         if (s->flags & FF_RTP_FLAG_RFC2190) {
yading@11:             int mb_info_size = 0;
yading@11:             const uint8_t *mb_info =
yading@11:                 av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
yading@11:                                         &mb_info_size);
yading@11:             ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
yading@11:             break;
yading@11:         }
yading@11:         /* Fallthrough */
yading@11:     case AV_CODEC_ID_H263P:
yading@11:         ff_rtp_send_h263(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_VORBIS:
yading@11:     case AV_CODEC_ID_THEORA:
yading@11:         ff_rtp_send_xiph(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_VP8:
yading@11:         ff_rtp_send_vp8(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_ILBC:
yading@11:         rtp_send_ilbc(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_MJPEG:
yading@11:         ff_rtp_send_jpeg(s1, pkt->data, size);
yading@11:         break;
yading@11:     case AV_CODEC_ID_OPUS:
yading@11:         if (size > s->max_payload_size) {
yading@11:             av_log(s1, AV_LOG_ERROR,
yading@11:                    "Packet size %d too large for max RTP payload size %d\n",
yading@11:                    size, s->max_payload_size);
yading@11:             return AVERROR(EINVAL);
yading@11:         }
yading@11:         /* Intentional fallthrough */
yading@11:     default:
yading@11:         /* better than nothing : send the codec raw data */
yading@11:         rtp_send_raw(s1, pkt->data, size);
yading@11:         break;
yading@11:     }
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: static int rtp_write_trailer(AVFormatContext *s1)
yading@11: {
yading@11:     RTPMuxContext *s = s1->priv_data;
yading@11: 
yading@11:     av_freep(&s->buf);
yading@11: 
yading@11:     return 0;
yading@11: }
yading@11: 
yading@11: AVOutputFormat ff_rtp_muxer = {
yading@11:     .name              = "rtp",
yading@11:     .long_name         = NULL_IF_CONFIG_SMALL("RTP output"),
yading@11:     .priv_data_size    = sizeof(RTPMuxContext),
yading@11:     .audio_codec       = AV_CODEC_ID_PCM_MULAW,
yading@11:     .video_codec       = AV_CODEC_ID_MPEG4,
yading@11:     .write_header      = rtp_write_header,
yading@11:     .write_packet      = rtp_write_packet,
yading@11:     .write_trailer     = rtp_write_trailer,
yading@11:     .priv_class        = &rtp_muxer_class,
yading@11: };