| /* |
| * RTP output format |
| * Copyright (c) 2002 Fabrice Bellard |
| * |
| * This file is part of FFmpeg. |
| * |
| * FFmpeg is free software; you can redistribute it and/or |
| * modify it under the terms of the GNU Lesser General Public |
| * License as published by the Free Software Foundation; either |
| * version 2.1 of the License, or (at your option) any later version. |
| * |
| * FFmpeg is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
| * Lesser General Public License for more details. |
| * |
| * You should have received a copy of the GNU Lesser General Public |
| * License along with FFmpeg; if not, write to the Free Software |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
| */ |
| |
| #include "libavcodec/bitstream.h" |
| #include "avformat.h" |
| #include "mpegts.h" |
| |
| #include <unistd.h> |
| #include "network.h" |
| |
| #include "rtpenc.h" |
| |
| //#define DEBUG |
| |
| #define RTCP_SR_SIZE 28 |
| #define NTP_OFFSET 2208988800ULL |
| #define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL) |
| |
| static uint64_t ntp_time(void) |
| { |
| return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US; |
| } |
| |
| static int rtp_write_header(AVFormatContext *s1) |
| { |
| RTPMuxContext *s = s1->priv_data; |
| int payload_type, max_packet_size, n; |
| AVStream *st; |
| |
| if (s1->nb_streams != 1) |
| return -1; |
| st = s1->streams[0]; |
| |
| payload_type = ff_rtp_get_payload_type(st->codec); |
| if (payload_type < 0) |
| payload_type = RTP_PT_PRIVATE; /* private payload type */ |
| s->payload_type = payload_type; |
| |
| // following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately |
| s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ |
| s->timestamp = s->base_timestamp; |
| s->cur_timestamp = 0; |
| s->ssrc = 0; /* FIXME: was random(), what should this be? */ |
| s->first_packet = 1; |
| s->first_rtcp_ntp_time = AV_NOPTS_VALUE; |
| |
| max_packet_size = url_fget_max_packet_size(s1->pb); |
| if (max_packet_size <= 12) |
| return AVERROR(EIO); |
| s->buf = av_malloc(max_packet_size); |
| if (s->buf == NULL) { |
| return AVERROR(ENOMEM); |
| } |
| s->max_payload_size = max_packet_size - 12; |
| |
| s->max_frames_per_packet = 0; |
| if (s1->max_delay) { |
| if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
| if (st->codec->frame_size == 0) { |
| av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
| } else { |
| s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN); |
| } |
| } |
| if (st->codec->codec_type == CODEC_TYPE_VIDEO) { |
| /* FIXME: We should round down here... */ |
| s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); |
| } |
| } |
| |
| av_set_pts_info(st, 32, 1, 90000); |
| switch(st->codec->codec_id) { |
| case CODEC_ID_MP2: |
| case CODEC_ID_MP3: |
| s->buf_ptr = s->buf + 4; |
| break; |
| case CODEC_ID_MPEG1VIDEO: |
| case CODEC_ID_MPEG2VIDEO: |
| break; |
| case CODEC_ID_MPEG2TS: |
| n = s->max_payload_size / TS_PACKET_SIZE; |
| if (n < 1) |
| n = 1; |
| s->max_payload_size = n * TS_PACKET_SIZE; |
| s->buf_ptr = s->buf; |
| break; |
| case CODEC_ID_AAC: |
| s->num_frames = 0; |
| default: |
| if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
| av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
| } |
| s->buf_ptr = s->buf; |
| break; |
| } |
| |
| return 0; |
| } |
| |
| /* send an rtcp sender report packet */ |
| static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) |
| { |
| RTPMuxContext *s = s1->priv_data; |
| uint32_t rtp_ts; |
| |
| dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); |
| |
| if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time; |
| s->last_rtcp_ntp_time = ntp_time; |
| rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, |
| s1->streams[0]->time_base) + s->base_timestamp; |
| put_byte(s1->pb, (RTP_VERSION << 6)); |
| put_byte(s1->pb, 200); |
| put_be16(s1->pb, 6); /* length in words - 1 */ |
| put_be32(s1->pb, s->ssrc); |
| put_be32(s1->pb, ntp_time / 1000000); |
| put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
| put_be32(s1->pb, rtp_ts); |
| put_be32(s1->pb, s->packet_count); |
| put_be32(s1->pb, s->octet_count); |
| put_flush_packet(s1->pb); |
| } |
| |
| /* send an rtp packet. sequence number is incremented, but the caller |
| must update the timestamp itself */ |
| void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) |
| { |
| RTPMuxContext *s = s1->priv_data; |
| |
| dprintf(s1, "rtp_send_data size=%d\n", len); |
| |
| /* build the RTP header */ |
| put_byte(s1->pb, (RTP_VERSION << 6)); |
| put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); |
| put_be16(s1->pb, s->seq); |
| put_be32(s1->pb, s->timestamp); |
| put_be32(s1->pb, s->ssrc); |
| |
| put_buffer(s1->pb, buf1, len); |
| put_flush_packet(s1->pb); |
| |
| s->seq++; |
| s->octet_count += len; |
| s->packet_count++; |
| } |
| |
| /* send an integer number of samples and compute time stamp and fill |
| the rtp send buffer before sending. */ |
| static void rtp_send_samples(AVFormatContext *s1, |
| const uint8_t *buf1, int size, int sample_size) |
| { |
| RTPMuxContext *s = s1->priv_data; |
| int len, max_packet_size, n; |
| |
| max_packet_size = (s->max_payload_size / sample_size) * sample_size; |
| /* not needed, but who nows */ |
| if ((size % sample_size) != 0) |
| av_abort(); |
| n = 0; |
| while (size > 0) { |
| s->buf_ptr = s->buf; |
| len = FFMIN(max_packet_size, size); |
| |
| /* copy data */ |
| memcpy(s->buf_ptr, buf1, len); |
| s->buf_ptr += len; |
| buf1 += len; |
| size -= len; |
| s->timestamp = s->cur_timestamp + n / sample_size; |
| ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
| n += (s->buf_ptr - s->buf); |
| } |
| } |
| |
| /* NOTE: we suppose that exactly one frame is given as argument here */ |
| /* XXX: test it */ |
| static void rtp_send_mpegaudio(AVFormatContext *s1, |
| const uint8_t *buf1, int size) |
| { |
| RTPMuxContext *s = s1->priv_data; |
| int len, count, max_packet_size; |
| |
| max_packet_size = s->max_payload_size; |
| |
| /* test if we must flush because not enough space */ |
| len = (s->buf_ptr - s->buf); |
| if ((len + size) > max_packet_size) { |
| if (len > 4) { |
| ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
| s->buf_ptr = s->buf + 4; |
| } |
| } |
| if (s->buf_ptr == s->buf + 4) { |
| s->timestamp = s->cur_timestamp; |
| } |
| |
| /* add the packet */ |
| if (size > max_packet_size) { |
| /* big packet: fragment */ |
| count = 0; |
| while (size > 0) { |
| len = max_packet_size - 4; |
| if (len > size) |
| len = size; |
| /* build fragmented packet */ |
| s->buf[0] = 0; |
| s->buf[1] = 0; |
| s->buf[2] = count >> 8; |
| s->buf[3] = count; |
| memcpy(s->buf + 4, buf1, len); |
| ff_rtp_send_data(s1, s->buf, len + 4, 0); |
| size -= len; |
| buf1 += len; |
| count += len; |
| } |
| } else { |
| if (s->buf_ptr == s->buf + 4) { |
| /* no fragmentation possible */ |
| s->buf[0] = 0; |
| s->buf[1] = 0; |
| s->buf[2] = 0; |
| s->buf[3] = 0; |
| } |
| memcpy(s->buf_ptr, buf1, size); |
| s->buf_ptr += size; |
| } |
| } |
| |
| static void rtp_send_raw(AVFormatContext *s1, |
| const uint8_t *buf1, int size) |
| { |
| RTPMuxContext *s = s1->priv_data; |
| int len, max_packet_size; |
| |
| max_packet_size = s->max_payload_size; |
| |
| while (size > 0) { |
| len = max_packet_size; |
| if (len > size) |
| len = size; |
| |
| s->timestamp = s->cur_timestamp; |
| ff_rtp_send_data(s1, buf1, len, (len == size)); |
| |
| buf1 += len; |
| size -= len; |
| } |
| } |
| |
| /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
| static void rtp_send_mpegts_raw(AVFormatContext *s1, |
| const uint8_t *buf1, int size) |
| { |
| RTPMuxContext *s = s1->priv_data; |
| int len, out_len; |
| |
| while (size >= TS_PACKET_SIZE) { |
| len = s->max_payload_size - (s->buf_ptr - s->buf); |
| if (len > size) |
| len = size; |
| memcpy(s->buf_ptr, buf1, len); |
| buf1 += len; |
| size -= len; |
| s->buf_ptr += len; |
| |
| out_len = s->buf_ptr - s->buf; |
| if (out_len >= s->max_payload_size) { |
| ff_rtp_send_data(s1, s->buf, out_len, 0); |
| s->buf_ptr = s->buf; |
| } |
| } |
| } |
| |
| /* write an RTP packet. 'buf1' must contain a single specific frame. */ |
| static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) |
| { |
| RTPMuxContext *s = s1->priv_data; |
| AVStream *st = s1->streams[0]; |
| int rtcp_bytes; |
| int size= pkt->size; |
| uint8_t *buf1= pkt->data; |
| |
| dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size); |
| |
| rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / |
| RTCP_TX_RATIO_DEN; |
| if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
| (ntp_time() - s->last_rtcp_ntp_time > 5000000))) { |
| rtcp_send_sr(s1, ntp_time()); |
| s->last_octet_count = s->octet_count; |
| s->first_packet = 0; |
| } |
| s->cur_timestamp = s->base_timestamp + pkt->pts; |
| |
| switch(st->codec->codec_id) { |
| case CODEC_ID_PCM_MULAW: |
| case CODEC_ID_PCM_ALAW: |
| case CODEC_ID_PCM_U8: |
| case CODEC_ID_PCM_S8: |
| rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
| break; |
| case CODEC_ID_PCM_U16BE: |
| case CODEC_ID_PCM_U16LE: |
| case CODEC_ID_PCM_S16BE: |
| case CODEC_ID_PCM_S16LE: |
| rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
| break; |
| case CODEC_ID_MP2: |
| case CODEC_ID_MP3: |
| rtp_send_mpegaudio(s1, buf1, size); |
| break; |
| case CODEC_ID_MPEG1VIDEO: |
| case CODEC_ID_MPEG2VIDEO: |
| ff_rtp_send_mpegvideo(s1, buf1, size); |
| break; |
| case CODEC_ID_AAC: |
| ff_rtp_send_aac(s1, buf1, size); |
| break; |
| case CODEC_ID_MPEG2TS: |
| rtp_send_mpegts_raw(s1, buf1, size); |
| break; |
| case CODEC_ID_H264: |
| ff_rtp_send_h264(s1, buf1, size); |
| break; |
| default: |
| /* better than nothing : send the codec raw data */ |
| rtp_send_raw(s1, buf1, size); |
| break; |
| } |
| return 0; |
| } |
| |
| static int rtp_write_trailer(AVFormatContext *s1) |
| { |
| RTPMuxContext *s = s1->priv_data; |
| |
| av_freep(&s->buf); |
| |
| return 0; |
| } |
| |
| AVOutputFormat rtp_muxer = { |
| "rtp", |
| NULL_IF_CONFIG_SMALL("RTP output format"), |
| NULL, |
| NULL, |
| sizeof(RTPMuxContext), |
| CODEC_ID_PCM_MULAW, |
| CODEC_ID_NONE, |
| rtp_write_header, |
| rtp_write_packet, |
| rtp_write_trailer, |
| }; |