X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpenc.c;h=dfd7461d1836553af9b284a3a09af8a8bb153e1f;hb=f4b51d061f0f34e36be876b562b8abe47f4b9c1c;hp=0e129acea6f9d027cb733cc4e3142dfc48061dd5;hpb=d3536678dc87608c18c9aecde357173924c4a5fd;p=ffmpeg diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 0e129acea6f..dfd7461d183 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -2,67 +2,111 @@ * RTP output format * Copyright (c) 2002 Fabrice Bellard * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; 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 -#include "network.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 -#define NTP_OFFSET 2208988800ULL -#define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL) -static uint64_t ntp_time(void) +static int is_supported(enum CodecID id) { - return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US; + 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: + return 1; + default: + return 0; + } } static int rtp_write_header(AVFormatContext *s1) { RTPMuxContext *s = s1->priv_data; - int payload_type, max_packet_size, n; + 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 %x\n", st->codec->codec_id); - payload_type = rtp_get_payload_type(st->codec); - if (payload_type < 0) - payload_type = RTP_PT_PRIVATE; /* private payload type */ - s->payload_type = payload_type; + return -1; + } -// 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? */ + 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 = 0; /* FIXME: was random(), what should this be? */ + s->ssrc = av_get_random_seed(); s->first_packet = 1; - s->first_rtcp_ntp_time = AV_NOPTS_VALUE; + 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 = url_fget_max_packet_size(s1->pb); + max_packet_size = s1->pb->max_packet_size; if (max_packet_size <= 12) return AVERROR(EIO); s->buf = av_malloc(max_packet_size); @@ -73,14 +117,14 @@ static int rtp_write_header(AVFormatContext *s1) s->max_frames_per_packet = 0; if (s1->max_delay) { - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { + 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 * st->codec->frame_size, AV_ROUND_DOWN); } } - if (st->codec->codec_type == CODEC_TYPE_VIDEO) { + 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); } @@ -102,10 +146,50 @@ static int rtp_write_header(AVFormatContext *s1) 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. */ + av_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: - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { +defaultcase: + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { av_set_pts_info(st, 32, 1, st->codec->sample_rate); } s->buf_ptr = s->buf; @@ -121,22 +205,21 @@ 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); + av_dlog(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); + 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 @@ -145,17 +228,17 @@ 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); + av_dlog(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); + 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); - put_buffer(s1->pb, buf1, len); - put_flush_packet(s1->pb); + avio_write(s1->pb, buf1, len); + avio_flush(s1->pb); s->seq++; s->octet_count += len; @@ -190,8 +273,6 @@ static void rtp_send_samples(AVFormatContext *s1, } } -/* 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) { @@ -289,22 +370,20 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1, } } -/* 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); + 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) && - (ntp_time() - s->last_rtcp_ntp_time > 5000000))) { - rtcp_send_sr(s1, ntp_time()); + (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; } @@ -315,34 +394,58 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) 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); + rtp_send_samples(s1, pkt->data, 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); + rtp_send_samples(s1, pkt->data, size, 2 * 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 is 1 byte per stream clock. */ + rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels); break; case CODEC_ID_MP2: case CODEC_ID_MP3: - rtp_send_mpegaudio(s1, buf1, size); + rtp_send_mpegaudio(s1, pkt->data, size); break; case CODEC_ID_MPEG1VIDEO: case CODEC_ID_MPEG2VIDEO: - ff_rtp_send_mpegvideo(s1, buf1, size); + ff_rtp_send_mpegvideo(s1, pkt->data, size); break; case CODEC_ID_AAC: - ff_rtp_send_aac(s1, buf1, size); + 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, buf1, size); + rtp_send_mpegts_raw(s1, pkt->data, size); break; case CODEC_ID_H264: - ff_rtp_send_h264(s1, buf1, size); + 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, buf1, size); + rtp_send_raw(s1, pkt->data, size); break; } return 0; @@ -357,15 +460,14 @@ static int rtp_write_trailer(AVFormatContext *s1) 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, +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, };