/*
* RTP input/output format
- * Copyright (c) 2002 Fabrice Bellard.
+ * Copyright (c) 2002 Fabrice Bellard
*
- * This library is free software; you can redistribute it and/or
+ * This file is part of Libav.
+ *
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * License along with Libav; 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 <unistd.h>
-#include <sys/types.h>
-#include <sys/socket.h>
-#include <netinet/in.h>
-#ifndef __BEOS__
-# include <arpa/inet.h>
-#else
-# include "barpainet.h"
-#endif
-#include <netdb.h>
-
-//#define DEBUG
-
-
-/* TODO: - add RTCP statistics reporting (should be optional).
-
- - add support for h263/mpeg4 packetized output : IDEA: send a
- buffer to 'rtp_write_packet' contains all the packets for ONE
- frame. Each packet should have a four byte header containing
- the length in big endian format (same trick as
- 'url_open_dyn_packet_buf')
-*/
-
-#define RTP_VERSION 2
-#define RTP_MAX_SDES 256 /* maximum text length for SDES */
-
-/* RTCP paquets use 0.5 % of the bandwidth */
-#define RTCP_TX_RATIO_NUM 5
-#define RTCP_TX_RATIO_DEN 1000
-
-typedef enum {
- RTCP_SR = 200,
- RTCP_RR = 201,
- RTCP_SDES = 202,
- RTCP_BYE = 203,
- RTCP_APP = 204
-} rtcp_type_t;
-
-typedef enum {
- RTCP_SDES_END = 0,
- RTCP_SDES_CNAME = 1,
- RTCP_SDES_NAME = 2,
- RTCP_SDES_EMAIL = 3,
- RTCP_SDES_PHONE = 4,
- RTCP_SDES_LOC = 5,
- RTCP_SDES_TOOL = 6,
- RTCP_SDES_NOTE = 7,
- RTCP_SDES_PRIV = 8,
- RTCP_SDES_IMG = 9,
- RTCP_SDES_DOOR = 10,
- RTCP_SDES_SOURCE = 11
-} rtcp_sdes_type_t;
-
-struct RTPDemuxContext {
- AVFormatContext *ic;
- AVStream *st;
- int payload_type;
- uint32_t ssrc;
- uint16_t seq;
- uint32_t timestamp;
- uint32_t base_timestamp;
- uint32_t cur_timestamp;
- int max_payload_size;
- MpegTSContext *ts; /* only used for RTP_PT_MPEG2TS payloads */
- int read_buf_index;
- int read_buf_size;
-
- /* rtcp sender statistics receive */
- int64_t last_rtcp_ntp_time;
- int64_t first_rtcp_ntp_time;
- uint32_t last_rtcp_timestamp;
- /* rtcp sender statistics */
- unsigned int packet_count;
- unsigned int octet_count;
- unsigned int last_octet_count;
- int first_packet;
- /* buffer for output */
- uint8_t buf[RTP_MAX_PACKET_LENGTH];
- uint8_t *buf_ptr;
-};
-
-int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
-{
- switch(payload_type) {
- case RTP_PT_ULAW:
- codec->codec_type = CODEC_TYPE_AUDIO;
- codec->codec_id = CODEC_ID_PCM_MULAW;
- codec->channels = 1;
- codec->sample_rate = 8000;
- break;
- case RTP_PT_ALAW:
- codec->codec_type = CODEC_TYPE_AUDIO;
- codec->codec_id = CODEC_ID_PCM_ALAW;
- codec->channels = 1;
- codec->sample_rate = 8000;
- break;
- case RTP_PT_S16BE_STEREO:
- codec->codec_type = CODEC_TYPE_AUDIO;
- codec->codec_id = CODEC_ID_PCM_S16BE;
- codec->channels = 2;
- codec->sample_rate = 44100;
- break;
- case RTP_PT_S16BE_MONO:
- codec->codec_type = CODEC_TYPE_AUDIO;
- codec->codec_id = CODEC_ID_PCM_S16BE;
- codec->channels = 1;
- codec->sample_rate = 44100;
- break;
- case RTP_PT_MPEGAUDIO:
- codec->codec_type = CODEC_TYPE_AUDIO;
- codec->codec_id = CODEC_ID_MP2;
- break;
- case RTP_PT_JPEG:
- codec->codec_type = CODEC_TYPE_VIDEO;
- codec->codec_id = CODEC_ID_MJPEG;
- break;
- case RTP_PT_MPEGVIDEO:
- codec->codec_type = CODEC_TYPE_VIDEO;
- codec->codec_id = CODEC_ID_MPEG1VIDEO;
- break;
- case RTP_PT_MPEG2TS:
- codec->codec_type = CODEC_TYPE_DATA;
- codec->codec_id = CODEC_ID_MPEG2TS;
- break;
- default:
- return -1;
- }
- return 0;
-}
-
-/* return < 0 if unknown payload type */
-int rtp_get_payload_type(AVCodecContext *codec)
-{
- int payload_type;
-
- /* compute the payload type */
- payload_type = -1;
- switch(codec->codec_id) {
- case CODEC_ID_PCM_MULAW:
- payload_type = RTP_PT_ULAW;
- break;
- case CODEC_ID_PCM_ALAW:
- payload_type = RTP_PT_ALAW;
- break;
- case CODEC_ID_PCM_S16BE:
- if (codec->channels == 1) {
- payload_type = RTP_PT_S16BE_MONO;
- } else if (codec->channels == 2) {
- payload_type = RTP_PT_S16BE_STEREO;
- }
- break;
- case CODEC_ID_MP2:
- case CODEC_ID_MP3:
- payload_type = RTP_PT_MPEGAUDIO;
- break;
- case CODEC_ID_MJPEG:
- payload_type = RTP_PT_JPEG;
- break;
- case CODEC_ID_MPEG1VIDEO:
- payload_type = RTP_PT_MPEGVIDEO;
- break;
- case CODEC_ID_MPEG2TS:
- payload_type = RTP_PT_MPEG2TS;
- break;
- default:
- break;
- }
- return payload_type;
-}
-
-static inline uint32_t decode_be32(const uint8_t *p)
-{
- return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
-}
-
-static inline uint64_t decode_be64(const uint8_t *p)
-{
- return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
-}
+#include "libavutil/opt.h"
+#include "avformat.h"
-static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
-{
- if (buf[1] != 200)
- return -1;
- s->last_rtcp_ntp_time = decode_be64(buf + 8);
- if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
- s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
- s->last_rtcp_timestamp = decode_be32(buf + 16);
- return 0;
-}
+#include "rtp.h"
-/**
- * open a new RTP parse context for stream 'st'. 'st' can be NULL for
- * MPEG2TS streams to indicate that they should be demuxed inside the
- * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
+/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
+/* payload types >= 96 are dynamic;
+ * payload types between 72 and 76 are reserved for RTCP conflict avoidance;
+ * all the other payload types not present in the table are unassigned or
+ * reserved
*/
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type)
-{
- RTPDemuxContext *s;
-
- s = av_mallocz(sizeof(RTPDemuxContext));
- if (!s)
- return NULL;
- s->payload_type = payload_type;
- s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
- s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
- s->ic = s1;
- s->st = st;
- if (payload_type == RTP_PT_MPEG2TS) {
- s->ts = mpegts_parse_open(s->ic);
- if (s->ts == NULL) {
- av_free(s);
- return NULL;
- }
- } else {
- switch(st->codec.codec_id) {
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
- case CODEC_ID_MP2:
- case CODEC_ID_MP3:
- case CODEC_ID_MPEG4:
- st->need_parsing = 1;
- break;
- default:
- break;
- }
- }
- return s;
-}
+static const struct {
+ int pt;
+ const char enc_name[6];
+ enum AVMediaType codec_type;
+ enum AVCodecID codec_id;
+ int clock_rate;
+ int audio_channels;
+} rtp_payload_types[] = {
+ {0, "PCMU", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_PCM_MULAW, 8000, 1},
+ {3, "GSM", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1},
+ {4, "G723", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_G723_1, 8000, 1},
+ {5, "DVI4", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1},
+ {6, "DVI4", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 16000, 1},
+ {7, "LPC", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1},
+ {8, "PCMA", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_PCM_ALAW, 8000, 1},
+ {9, "G722", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_ADPCM_G722, 8000, 1},
+ {10, "L16", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_PCM_S16BE, 44100, 2},
+ {11, "L16", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_PCM_S16BE, 44100, 1},
+ {12, "QCELP", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_QCELP, 8000, 1},
+ {13, "CN", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1},
+ {14, "MPA", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_MP2, -1, -1},
+ {14, "MPA", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_MP3, -1, -1},
+ {15, "G728", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1},
+ {16, "DVI4", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 11025, 1},
+ {17, "DVI4", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 22050, 1},
+ {18, "G729", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1},
+ {25, "CelB", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_NONE, 90000, -1},
+ {26, "JPEG", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_MJPEG, 90000, -1},
+ {28, "nv", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_NONE, 90000, -1},
+ {31, "H261", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_H261, 90000, -1},
+ {32, "MPV", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_MPEG1VIDEO, 90000, -1},
+ {32, "MPV", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_MPEG2VIDEO, 90000, -1},
+ {33, "MP2T", AVMEDIA_TYPE_DATA, AV_CODEC_ID_MPEG2TS, 90000, -1},
+ {34, "H263", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_H263, 90000, -1},
+ {-1, "", AVMEDIA_TYPE_UNKNOWN, AV_CODEC_ID_NONE, -1, -1}
+};
-/**
- * Parse an RTP or RTCP packet directly sent as a buffer.
- * @param s RTP parse context.
- * @param pkt returned packet
- * @param buf input buffer or NULL to read the next packets
- * @param len buffer len
- * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
- * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
- */
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
- const uint8_t *buf, int len)
+int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
- unsigned int ssrc, h;
- int payload_type, seq, delta_timestamp, ret;
- AVStream *st;
- uint32_t timestamp;
-
- if (!buf) {
- /* return the next packets, if any */
- if (s->read_buf_index >= s->read_buf_size)
- return -1;
- ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
- s->read_buf_size - s->read_buf_index);
- if (ret < 0)
- return -1;
- s->read_buf_index += ret;
- if (s->read_buf_index < s->read_buf_size)
- return 1;
- else
- return 0;
- }
+ int i = 0;
- if (len < 12)
- return -1;
-
- if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
- return -1;
- if (buf[1] >= 200 && buf[1] <= 204) {
- rtcp_parse_packet(s, buf, len);
- return -1;
- }
- payload_type = buf[1] & 0x7f;
- seq = (buf[2] << 8) | buf[3];
- timestamp = decode_be32(buf + 4);
- ssrc = decode_be32(buf + 8);
-
- /* NOTE: we can handle only one payload type */
- if (s->payload_type != payload_type)
- return -1;
-#if defined(DEBUG) || 1
- if (seq != ((s->seq + 1) & 0xffff)) {
- printf("RTP: PT=%02x: bad cseq %04x expected=%04x\n",
- payload_type, seq, ((s->seq + 1) & 0xffff));
- }
- s->seq = seq;
-#endif
- len -= 12;
- buf += 12;
-
- st = s->st;
- if (!st) {
- /* specific MPEG2TS demux support */
- ret = mpegts_parse_packet(s->ts, pkt, buf, len);
- if (ret < 0)
- return -1;
- if (ret < len) {
- s->read_buf_size = len - ret;
- memcpy(s->buf, buf + ret, s->read_buf_size);
- s->read_buf_index = 0;
- return 1;
- }
- } else {
- switch(st->codec.codec_id) {
- case CODEC_ID_MP2:
- /* better than nothing: skip mpeg audio RTP header */
- if (len <= 4)
- return -1;
- h = decode_be32(buf);
- len -= 4;
- buf += 4;
- av_new_packet(pkt, len);
- memcpy(pkt->data, buf, len);
- break;
- case CODEC_ID_MPEG1VIDEO:
- /* better than nothing: skip mpeg video RTP header */
- if (len <= 4)
- return -1;
- h = decode_be32(buf);
- buf += 4;
- len -= 4;
- if (h & (1 << 26)) {
- /* mpeg2 */
- if (len <= 4)
- return -1;
- buf += 4;
- len -= 4;
- }
- av_new_packet(pkt, len);
- memcpy(pkt->data, buf, len);
- break;
- default:
- av_new_packet(pkt, len);
- memcpy(pkt->data, buf, len);
- break;
- }
-
- switch(st->codec.codec_id) {
- case CODEC_ID_MP2:
- case CODEC_ID_MPEG1VIDEO:
- if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
- int64_t addend;
- /* XXX: is it really necessary to unify the timestamp base ? */
- /* compute pts from timestamp with received ntp_time */
- delta_timestamp = timestamp - s->last_rtcp_timestamp;
- /* convert to 90 kHz without overflow */
- addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
- addend = (addend * 5625) >> 14;
- pkt->pts = addend + delta_timestamp;
+ for (i = 0; rtp_payload_types[i].pt >= 0; i++)
+ if (rtp_payload_types[i].pt == payload_type) {
+ if (rtp_payload_types[i].codec_id != AV_CODEC_ID_NONE) {
+ codec->codec_type = rtp_payload_types[i].codec_type;
+ codec->codec_id = rtp_payload_types[i].codec_id;
+ if (rtp_payload_types[i].audio_channels > 0)
+ codec->channels = rtp_payload_types[i].audio_channels;
+ if (rtp_payload_types[i].clock_rate > 0)
+ codec->sample_rate = rtp_payload_types[i].clock_rate;
+ return 0;
}
- break;
- default:
- /* no timestamp info yet */
- break;
- }
- pkt->stream_index = s->st->index;
- }
- return 0;
-}
-
-void rtp_parse_close(RTPDemuxContext *s)
-{
- if (s->payload_type == RTP_PT_MPEG2TS) {
- mpegts_parse_close(s->ts);
- }
- av_free(s);
-}
-
-/* rtp output */
-
-static int rtp_write_header(AVFormatContext *s1)
-{
- RTPDemuxContext *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 = rtp_get_payload_type(&st->codec);
- if (payload_type < 0)
- payload_type = RTP_PT_PRIVATE; /* private payload type */
- s->payload_type = payload_type;
-
- s->base_timestamp = random();
- s->timestamp = s->base_timestamp;
- s->ssrc = random();
- s->first_packet = 1;
-
- max_packet_size = url_fget_max_packet_size(&s1->pb);
- if (max_packet_size <= 12)
- return AVERROR_IO;
- s->max_payload_size = max_packet_size - 12;
-
- switch(st->codec.codec_id) {
- case CODEC_ID_MP2:
- case CODEC_ID_MP3:
- s->buf_ptr = s->buf + 4;
- s->cur_timestamp = 0;
- break;
- case CODEC_ID_MPEG1VIDEO:
- s->cur_timestamp = 0;
- 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;
- default:
- 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)
-{
- RTPDemuxContext *s = s1->priv_data;
-#if defined(DEBUG)
- printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
-#endif
- 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_be64(&s1->pb, ntp_time);
- put_be32(&s1->pb, s->timestamp);
- 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 */
-static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
-{
- RTPDemuxContext *s = s1->priv_data;
-
-#ifdef DEBUG
- printf("rtp_send_data size=%d\n", len);
-#endif
-
- /* build the RTP header */
- put_byte(&s1->pb, (RTP_VERSION << 6));
- put_byte(&s1->pb, s->payload_type & 0x7f);
- 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)
-{
- RTPDemuxContext *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();
- while (size > 0) {
- len = (max_packet_size - (s->buf_ptr - s->buf));
- if (len > size)
- len = size;
-
- /* copy data */
- memcpy(s->buf_ptr, buf1, len);
- s->buf_ptr += len;
- buf1 += len;
- size -= len;
- n = (s->buf_ptr - s->buf);
- /* if buffer full, then send it */
- if (n >= max_packet_size) {
- rtp_send_data(s1, s->buf, n);
- s->buf_ptr = s->buf;
- /* update timestamp */
- s->timestamp += n / sample_size;
- }
- }
-}
-
-/* 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)
-{
- RTPDemuxContext *s = s1->priv_data;
- AVStream *st = s1->streams[0];
- 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) {
- rtp_send_data(s1, s->buf, s->buf_ptr - s->buf);
- s->buf_ptr = s->buf + 4;
- /* 90 KHz time stamp */
- s->timestamp = s->base_timestamp +
- (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
- }
- }
-
- /* 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);
- rtp_send_data(s1, s->buf, len + 4);
- 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;
+ return -1;
+}
+
+int ff_rtp_get_payload_type(AVFormatContext *fmt,
+ AVCodecContext *codec, int idx)
+{
+ int i;
+ AVOutputFormat *ofmt = fmt ? fmt->oformat : NULL;
+
+ /* Was the payload type already specified for the RTP muxer? */
+ if (ofmt && ofmt->priv_class && fmt->priv_data) {
+ int64_t payload_type;
+ if (av_opt_get_int(fmt->priv_data, "payload_type", 0, &payload_type) >= 0 &&
+ payload_type >= 0)
+ return (int)payload_type;
+ }
+
+ /* static payload type */
+ for (i = 0; rtp_payload_types[i].pt >= 0; ++i)
+ if (rtp_payload_types[i].codec_id == codec->codec_id) {
+ if (codec->codec_id == AV_CODEC_ID_H263 && (!fmt || !fmt->oformat ||
+ !fmt->oformat->priv_class || !fmt->priv_data ||
+ !av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190")))
+ continue;
+ /* G722 has 8000 as nominal rate even if the sample rate is 16000,
+ * see section 4.5.2 in RFC 3551. */
+ if (codec->codec_id == AV_CODEC_ID_ADPCM_G722 &&
+ codec->sample_rate == 16000 && codec->channels == 1)
+ return rtp_payload_types[i].pt;
+ if (codec->codec_type == AVMEDIA_TYPE_AUDIO &&
+ ((rtp_payload_types[i].clock_rate > 0 &&
+ codec->sample_rate != rtp_payload_types[i].clock_rate) ||
+ (rtp_payload_types[i].audio_channels > 0 &&
+ codec->channels != rtp_payload_types[i].audio_channels)))
+ continue;
+ return rtp_payload_types[i].pt;
}
- memcpy(s->buf_ptr, buf1, size);
- s->buf_ptr += size;
- }
- s->cur_timestamp += st->codec.frame_size;
-}
-
-/* NOTE: a single frame must be passed with sequence header if
- needed. XXX: use slices. */
-static void rtp_send_mpegvideo(AVFormatContext *s1,
- const uint8_t *buf1, int size)
-{
- RTPDemuxContext *s = s1->priv_data;
- AVStream *st = s1->streams[0];
- int len, h, max_packet_size;
- uint8_t *q;
-
- max_packet_size = s->max_payload_size;
-
- while (size > 0) {
- /* XXX: more correct headers */
- h = 0;
- if (st->codec.sub_id == 2)
- h |= 1 << 26; /* mpeg 2 indicator */
- q = s->buf;
- *q++ = h >> 24;
- *q++ = h >> 16;
- *q++ = h >> 8;
- *q++ = h;
-
- if (st->codec.sub_id == 2) {
- h = 0;
- *q++ = h >> 24;
- *q++ = h >> 16;
- *q++ = h >> 8;
- *q++ = h;
- }
-
- len = max_packet_size - (q - s->buf);
- if (len > size)
- len = size;
- memcpy(q, buf1, len);
- q += len;
+ if (idx < 0)
+ idx = codec->codec_type == AVMEDIA_TYPE_AUDIO;
- /* 90 KHz time stamp */
- s->timestamp = s->base_timestamp +
- av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
- rtp_send_data(s1, s->buf, q - s->buf);
-
- buf1 += len;
- size -= len;
- }
- s->cur_timestamp++;
+ /* dynamic payload type */
+ return RTP_PT_PRIVATE + idx;
}
-static void rtp_send_raw(AVFormatContext *s1,
- const uint8_t *buf1, int size)
+const char *ff_rtp_enc_name(int payload_type)
{
- RTPDemuxContext *s = s1->priv_data;
- AVStream *st = s1->streams[0];
- int len, max_packet_size;
-
- max_packet_size = s->max_payload_size;
-
- while (size > 0) {
- len = max_packet_size;
- if (len > size)
- len = size;
+ int i;
- /* 90 KHz time stamp */
- s->timestamp = s->base_timestamp +
- av_rescale((int64_t)s->cur_timestamp * st->codec.frame_rate_base, 90000, st->codec.frame_rate);
- rtp_send_data(s1, buf1, len);
+ for (i = 0; rtp_payload_types[i].pt >= 0; i++)
+ if (rtp_payload_types[i].pt == payload_type)
+ return rtp_payload_types[i].enc_name;
- buf1 += len;
- size -= len;
- }
- s->cur_timestamp++;
+ return "";
}
-/* 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)
+enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type)
{
- RTPDemuxContext *s = s1->priv_data;
- int len, out_len;
+ int i;
- 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) {
- rtp_send_data(s1, s->buf, out_len);
- s->buf_ptr = s->buf;
- }
- }
-}
-
-/* write an RTP packet. 'buf1' must contain a single specific frame. */
-static int rtp_write_packet(AVFormatContext *s1, int stream_index,
- const uint8_t *buf1, int size, int64_t pts)
-{
- RTPDemuxContext *s = s1->priv_data;
- AVStream *st = s1->streams[0];
- int rtcp_bytes;
- int64_t ntp_time;
-
-#ifdef DEBUG
- printf("%d: write len=%d\n", stream_index, size);
-#endif
-
- /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
- rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
- RTCP_TX_RATIO_DEN;
- if (s->first_packet || rtcp_bytes >= 28) {
- /* compute NTP time */
- /* XXX: 90 kHz timestamp hardcoded */
- ntp_time = (pts << 28) / 5625;
- rtcp_send_sr(s1, ntp_time);
- s->last_octet_count = s->octet_count;
- s->first_packet = 0;
- }
-
- 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:
- rtp_send_mpegvideo(s1, buf1, size);
- break;
- case CODEC_ID_MPEG2TS:
- rtp_send_mpegts_raw(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)
-{
- // RTPDemuxContext *s = s1->priv_data;
- return 0;
-}
+ for (i = 0; rtp_payload_types[i].pt >= 0; i++)
+ if (!strcmp(buf, rtp_payload_types[i].enc_name) && (codec_type == rtp_payload_types[i].codec_type))
+ return rtp_payload_types[i].codec_id;
-AVOutputFormat rtp_mux = {
- "rtp",
- "RTP output format",
- NULL,
- NULL,
- sizeof(RTPDemuxContext),
- CODEC_ID_PCM_MULAW,
- CODEC_ID_NONE,
- rtp_write_header,
- rtp_write_packet,
- rtp_write_trailer,
-};
-
-int rtp_init(void)
-{
- av_register_output_format(&rtp_mux);
- return 0;
+ return AV_CODEC_ID_NONE;
}
projects / ffmpeg / blobdiff