mirror of https://github.com/FFmpeg/FFmpeg.git
Originally committed as revision 11408 to svn://svn.ffmpeg.org/ffmpeg/trunkpull/126/head
Luca Abeni
17 years ago
parent
9389e63c83
commit
83a0d3878c
3 changed files with 357 additions and 330 deletions
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/*
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* RTP output format |
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* Copyright (c) 2002 Fabrice Bellard. |
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* |
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* This file is part of FFmpeg. |
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* |
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* 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 |
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* 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 |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* 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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*/ |
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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_mpv.h" |
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#include "rtp_aac.h" |
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//#define DEBUG
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#define RTCP_SR_SIZE 28 |
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static int rtp_write_header(AVFormatContext *s1) |
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{ |
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RTPDemuxContext *s = s1->priv_data; |
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int payload_type, max_packet_size, n; |
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AVStream *st; |
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if (s1->nb_streams != 1) |
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return -1; |
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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 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
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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->cur_timestamp = 0; |
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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); |
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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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s->max_frames_per_packet = 0; |
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if (s1->max_delay) { |
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if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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if (st->codec->frame_size == 0) { |
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av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); |
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} else { |
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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); |
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} |
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} |
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if (st->codec->codec_type == CODEC_TYPE_VIDEO) { |
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/* FIXME: We should round down here... */ |
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s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base); |
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} |
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} |
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av_set_pts_info(st, 32, 1, 90000); |
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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; |
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break; |
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case CODEC_ID_MPEG1VIDEO: |
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case CODEC_ID_MPEG2VIDEO: |
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break; |
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case CODEC_ID_MPEG2TS: |
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n = s->max_payload_size / TS_PACKET_SIZE; |
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if (n < 1) |
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n = 1; |
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s->max_payload_size = n * TS_PACKET_SIZE; |
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s->buf_ptr = s->buf; |
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break; |
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case CODEC_ID_AAC: |
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s->read_buf_index = 0; |
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default: |
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if (st->codec->codec_type == CODEC_TYPE_AUDIO) { |
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av_set_pts_info(st, 32, 1, st->codec->sample_rate); |
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} |
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s->buf_ptr = s->buf; |
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break; |
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} |
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return 0; |
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} |
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/* 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; |
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s->last_rtcp_ntp_time = ntp_time; |
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rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q, |
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s1->streams[0]->time_base) + s->base_timestamp; |
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put_byte(s1->pb, (RTP_VERSION << 6)); |
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put_byte(s1->pb, 200); |
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put_be16(s1->pb, 6); /* length in words - 1 */ |
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put_be32(s1->pb, s->ssrc); |
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put_be32(s1->pb, ntp_time / 1000000); |
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put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); |
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put_be32(s1->pb, rtp_ts); |
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put_be32(s1->pb, s->packet_count); |
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put_be32(s1->pb, s->octet_count); |
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put_flush_packet(s1->pb); |
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} |
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/* send an rtp packet. sequence number is incremented, but the caller
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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 |
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printf("rtp_send_data size=%d\n", len); |
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#endif |
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/* build the RTP header */ |
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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); |
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put_be32(s1->pb, s->timestamp); |
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put_be32(s1->pb, s->ssrc); |
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put_buffer(s1->pb, buf1, len); |
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put_flush_packet(s1->pb); |
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s->seq++; |
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s->octet_count += len; |
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s->packet_count++; |
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} |
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/* send an integer number of samples and compute time stamp and fill
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the rtp send buffer before sending. */ |
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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; |
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max_packet_size = (s->max_payload_size / sample_size) * sample_size; |
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/* not needed, but who nows */ |
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if ((size % sample_size) != 0) |
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av_abort(); |
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n = 0; |
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while (size > 0) { |
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s->buf_ptr = s->buf; |
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len = FFMIN(max_packet_size, size); |
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/* copy data */ |
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memcpy(s->buf_ptr, buf1, len); |
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s->buf_ptr += len; |
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buf1 += len; |
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size -= len; |
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s->timestamp = s->cur_timestamp + n / sample_size; |
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ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); |
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n += (s->buf_ptr - s->buf); |
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} |
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} |
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/* NOTE: we suppose that exactly one frame is given as argument here */ |
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/* XXX: test it */ |
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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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int len, count, max_packet_size; |
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max_packet_size = s->max_payload_size; |
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/* test if we must flush because not enough space */ |
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len = (s->buf_ptr - s->buf); |
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if ((len + size) > max_packet_size) { |
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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; |
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} |
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} |
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if (s->buf_ptr == s->buf + 4) { |
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s->timestamp = s->cur_timestamp; |
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} |
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/* add the packet */ |
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if (size > max_packet_size) { |
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/* big packet: fragment */ |
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count = 0; |
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while (size > 0) { |
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len = max_packet_size - 4; |
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if (len > size) |
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len = size; |
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/* build fragmented packet */ |
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s->buf[0] = 0; |
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s->buf[1] = 0; |
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s->buf[2] = count >> 8; |
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s->buf[3] = count; |
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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; |
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buf1 += len; |
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count += len; |
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} |
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} else { |
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if (s->buf_ptr == s->buf + 4) { |
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/* no fragmentation possible */ |
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s->buf[0] = 0; |
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s->buf[1] = 0; |
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s->buf[2] = 0; |
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s->buf[3] = 0; |
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} |
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memcpy(s->buf_ptr, buf1, size); |
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s->buf_ptr += size; |
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} |
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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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int len, max_packet_size; |
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max_packet_size = s->max_payload_size; |
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while (size > 0) { |
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len = max_packet_size; |
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if (len > size) |
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len = size; |
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s->timestamp = s->cur_timestamp; |
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ff_rtp_send_data(s1, buf1, len, (len == size)); |
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buf1 += len; |
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size -= len; |
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} |
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} |
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/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ |
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static void rtp_send_mpegts_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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int len, out_len; |
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while (size >= TS_PACKET_SIZE) { |
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len = s->max_payload_size - (s->buf_ptr - s->buf); |
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if (len > size) |
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len = size; |
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memcpy(s->buf_ptr, buf1, len); |
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buf1 += len; |
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size -= len; |
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s->buf_ptr += len; |
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out_len = s->buf_ptr - s->buf; |
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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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} |
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} |
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} |
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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]; |
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int rtcp_bytes; |
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int size= pkt->size; |
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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 |
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/* 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; |
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if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && |
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(av_gettime() - s->last_rtcp_ntp_time > 5000000))) { |
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rtcp_send_sr(s1, av_gettime()); |
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s->last_octet_count = s->octet_count; |
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s->first_packet = 0; |
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} |
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s->cur_timestamp = s->base_timestamp + pkt->pts; |
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switch(st->codec->codec_id) { |
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case CODEC_ID_PCM_MULAW: |
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case CODEC_ID_PCM_ALAW: |
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case CODEC_ID_PCM_U8: |
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case CODEC_ID_PCM_S8: |
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rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); |
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break; |
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case CODEC_ID_PCM_U16BE: |
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case CODEC_ID_PCM_U16LE: |
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case CODEC_ID_PCM_S16BE: |
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case CODEC_ID_PCM_S16LE: |
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rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); |
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break; |
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case CODEC_ID_MP2: |
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case CODEC_ID_MP3: |
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rtp_send_mpegaudio(s1, buf1, size); |
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break; |
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case CODEC_ID_MPEG1VIDEO: |
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case CODEC_ID_MPEG2VIDEO: |
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ff_rtp_send_mpegvideo(s1, buf1, size); |
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break; |
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case CODEC_ID_AAC: |
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ff_rtp_send_aac(s1, buf1, size); |
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break; |
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case CODEC_ID_MPEG2TS: |
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rtp_send_mpegts_raw(s1, buf1, size); |
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break; |
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default: |
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/* better than nothing : send the codec raw data */ |
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rtp_send_raw(s1, buf1, size); |
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break; |
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} |
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return 0; |
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} |
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AVOutputFormat rtp_muxer = { |
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"rtp", |
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"RTP output format", |
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NULL, |
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NULL, |
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sizeof(RTPDemuxContext), |
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CODEC_ID_PCM_MULAW, |
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CODEC_ID_NONE, |
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rtp_write_header, |
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rtp_write_packet, |
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}; |
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