[Ffmpeg-cvslog] CVS: ffmpeg/libavformat rtp.c, 1.16, 1.17 rtp.h, 1.2, 1.3 rtsp.c, 1.21, 1.22 rtsp.h, 1.5, 1.6
Michael Niedermayer CVS
michael
Thu May 26 09:47:54 CEST 2005
Update of /cvsroot/ffmpeg/ffmpeg/libavformat
In directory mail:/var2/tmp/cvs-serv8986
Modified Files:
rtp.c rtp.h rtsp.c rtsp.h
Log Message:
RTP/RTSP and MPEG4-AAC audio
- preliminary support for mpeg4-aac rtp payload (no interleaving support)
- use udp transport as default (makes more sense with rtp, doesn't it ?)
- some code factorization, so adding support for new rtp payload will be easier
(I hope ;-)
patch by (Romain DEGEZ: romain degez, smartjog com)
Index: rtp.c
===================================================================
RCS file: /cvsroot/ffmpeg/ffmpeg/libavformat/rtp.c,v
retrieving revision 1.16
retrieving revision 1.17
diff -u -d -r1.16 -r1.17
--- rtp.c 30 Apr 2005 21:43:59 -0000 1.16
+++ rtp.c 26 May 2005 07:47:51 -0000 1.17
@@ -18,6 +18,7 @@
*/
#include "avformat.h"
#include "mpegts.h"
+#include "bitstream.h"
#include <unistd.h>
#include <sys/types.h>
@@ -42,36 +43,146 @@
'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;
+/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
+AVRtpPayloadType_t AVRtpPayloadTypes[]=
+{
+ {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1},
+ {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1},
+ {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1},
+ {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2},
+ {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1},
+ {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1},
+ {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1},
+ {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1},
+ {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
+ {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
+ {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
+ {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
+ {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
+ {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
+ {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
+ {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1},
+ {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
+ {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
+ {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
+ {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
+ {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1},
+ {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1},
+ {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1},
+ {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1},
+ {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
+ {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
+};
-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;
+AVRtpDynamicPayloadType_t AVRtpDynamicPayloadTypes[]=
+{
+ {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4},
+ {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_MPEG4AAC},
+ {"", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE}
+};
struct RTPDemuxContext {
AVFormatContext *ic;
@@ -83,7 +194,7 @@
uint32_t base_timestamp;
uint32_t cur_timestamp;
int max_payload_size;
- MpegTSContext *ts; /* only used for RTP_PT_MPEG2TS payloads */
+ MpegTSContext *ts; /* only used for MP2T payloads */
int read_buf_index;
int read_buf_size;
@@ -99,94 +210,37 @@
/* buffer for output */
uint8_t buf[RTP_MAX_PACKET_LENGTH];
uint8_t *buf_ptr;
+ /* special infos for au headers parsing */
+ rtp_payload_data_t *rtp_payload_data;
};
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;
+ if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
+ codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
+ codec->codec_id = AVRtpPayloadTypes[payload_type].codec_type;
+ if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
+ codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
+ if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
+ codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
+ return 0;
}
- return 0;
+ return -1;
}
/* return < 0 if unknown payload type */
int rtp_get_payload_type(AVCodecContext *codec)
{
- int payload_type;
+ int i, 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;
+ for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
+ if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
+ if (codec->codec_id == CODEC_ID_PCM_S16BE)
+ if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
+ continue;
+ payload_type = AVRtpPayloadTypes[i].pt;
}
- 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;
}
@@ -216,7 +270,7 @@
* MPEG2TS streams to indicate that they should be demuxed inside the
* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
*/
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type)
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_t *rtp_payload_data)
{
RTPDemuxContext *s;
@@ -228,7 +282,8 @@
s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
s->ic = s1;
s->st = st;
- if (payload_type == RTP_PT_MPEG2TS) {
+ s->rtp_payload_data = rtp_payload_data;
+ if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
s->ts = mpegts_parse_open(s->ic);
if (s->ts == NULL) {
av_free(s);
@@ -250,6 +305,57 @@
return s;
}
+static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
+{
+ AVCodecContext codec;
+ int au_headers_length, au_header_size, i;
+ GetBitContext getbitcontext;
+ rtp_payload_data_t *infos;
+
+ infos = s->rtp_payload_data;
+
+ if (infos == NULL)
+ return -1;
+
+ codec = s->st->codec;
+
+ /* decode the first 2 bytes where are stored the AUHeader sections
+ length in bits */
+ au_headers_length = BE_16(buf);
+
+ if (au_headers_length > RTP_MAX_PACKET_LENGTH)
+ return -1;
+
+ infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
+
+ /* skip AU headers length section (2 bytes) */
+ buf += 2;
+
+ init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
+
+ /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
+ au_header_size = infos->sizelength + infos->indexlength;
+ if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
+ return -1;
+
+ infos->nb_au_headers = au_headers_length / au_header_size;
+ infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
+
+ /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
+ In my test, the faad decoder doesnt behave correctly when sending each AU one by one
+ but does when sending the whole as one big packet... */
+ infos->au_headers[0].size = 0;
+ infos->au_headers[0].index = 0;
+ for (i = 0; i < infos->nb_au_headers; ++i) {
+ infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
+ infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
+ }
+
+ infos->nb_au_headers = 1;
+
+ return 0;
+}
+
/**
* Parse an RTP or RTCP packet directly sent as a buffer.
* @param s RTP parse context.
@@ -304,8 +410,8 @@
av_log(&s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
payload_type, seq, ((s->seq + 1) & 0xffff));
}
- s->seq = seq;
#endif
+ s->seq = seq;
len -= 12;
buf += 12;
@@ -370,6 +476,28 @@
pkt->pts = addend + delta_timestamp;
}
break;
+ case CODEC_ID_MPEG4:
+ pkt->pts = timestamp;
+ break;
+ case CODEC_ID_MPEG4AAC:
+ if (rtp_parse_mp4_au(s, buf))
+ return -1;
+ rtp_payload_data_t *infos = s->rtp_payload_data;
+ if (infos == NULL)
+ return -1;
+ buf += infos->au_headers_length_bytes + 2;
+ len -= infos->au_headers_length_bytes + 2;
+
+ /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
+ one au_header */
+ av_new_packet(pkt, infos->au_headers[0].size);
+ memcpy(pkt->data, buf, infos->au_headers[0].size);
+ buf += infos->au_headers[0].size;
+ len -= infos->au_headers[0].size;
+ s->read_buf_size = len;
+ s->buf_ptr = (char *)buf;
+ pkt->stream_index = s->st->index;
+ return 0;
default:
/* no timestamp info yet */
break;
@@ -381,7 +509,7 @@
void rtp_parse_close(RTPDemuxContext *s)
{
- if (s->payload_type == RTP_PT_MPEG2TS) {
+ if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
mpegts_parse_close(s->ts);
}
av_free(s);
Index: rtp.h
===================================================================
RCS file: /cvsroot/ffmpeg/ffmpeg/libavformat/rtp.h,v
retrieving revision 1.2
retrieving revision 1.3
diff -u -d -r1.2 -r1.3
--- rtp.h 29 Oct 2003 14:25:27 -0000 1.2
+++ rtp.h 26 May 2005 07:47:51 -0000 1.3
@@ -19,22 +19,6 @@
#ifndef RTP_H
#define RTP_H
-enum RTPPayloadType {
- RTP_PT_ULAW = 0,
- RTP_PT_GSM = 3,
- RTP_PT_G723 = 4,
- RTP_PT_ALAW = 8,
- RTP_PT_S16BE_STEREO = 10,
- RTP_PT_S16BE_MONO = 11,
- RTP_PT_MPEGAUDIO = 14,
- RTP_PT_JPEG = 26,
- RTP_PT_H261 = 31,
- RTP_PT_MPEGVIDEO = 32,
- RTP_PT_MPEG2TS = 33,
- RTP_PT_H263 = 34, /* old H263 encapsulation */
- RTP_PT_PRIVATE = 96,
-};
-
#define RTP_MIN_PACKET_LENGTH 12
#define RTP_MAX_PACKET_LENGTH 1500 /* XXX: suppress this define */
@@ -43,8 +27,8 @@
int rtp_get_payload_type(AVCodecContext *codec);
typedef struct RTPDemuxContext RTPDemuxContext;
-
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type);
+typedef struct rtp_payload_data_s rtp_payload_data_s;
+RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_s *rtp_payload_data);
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len);
void rtp_parse_close(RTPDemuxContext *s);
@@ -58,4 +42,82 @@
extern URLProtocol rtp_protocol;
+#define RTP_PT_PRIVATE 96
+#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
+
+/* Structure listing usefull vars to parse RTP packet payload*/
+typedef struct rtp_payload_data_s
+{
+ int sizelength;
+ int indexlength;
+ int indexdeltalength;
+ int profile_level_id;
+ int streamtype;
+ int objecttype;
+ char *mode;
+
+ /* mpeg 4 AU headers */
+ struct AUHeaders {
+ int size;
+ int index;
+ int cts_flag;
+ int cts;
+ int dts_flag;
+ int dts;
+ int rap_flag;
+ int streamstate;
+ } *au_headers;
+ int nb_au_headers;
+ int au_headers_length_bytes;
+ int cur_au_index;
+} rtp_payload_data_t;
+
+typedef struct AVRtpPayloadType_s
+{
+ int pt;
+ const char enc_name[50]; /* XXX: why 50 ? */
+ enum CodecType codec_type;
+ enum CodecID codec_id;
+ int clock_rate;
+ int audio_channels;
+} AVRtpPayloadType_t;
+
+typedef struct AVRtpDynamicPayloadType_s /* payload type >= 96 */
+{
+ const char enc_name[50]; /* XXX: still why 50 ? ;-) */
+ enum CodecType codec_type;
+ enum CodecID codec_id;
+} AVRtpDynamicPayloadType_t;
+
+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;
+
+extern AVRtpPayloadType_t AVRtpPayloadTypes[];
+extern AVRtpDynamicPayloadType_t AVRtpDynamicPayloadTypes[];
+
#endif /* RTP_H */
Index: rtsp.c
===================================================================
RCS file: /cvsroot/ffmpeg/ffmpeg/libavformat/rtsp.c,v
retrieving revision 1.21
retrieving revision 1.22
diff -u -d -r1.21 -r1.22
--- rtsp.c 16 Mar 2005 19:06:34 -0000 1.21
+++ rtsp.c 26 May 2005 07:47:51 -0000 1.22
@@ -66,22 +66,15 @@
struct in_addr sdp_ip; /* IP address (from SDP content - not used in RTSP) */
int sdp_ttl; /* IP TTL (from SDP content - not used in RTSP) */
int sdp_payload_type; /* payload type - only used in SDP */
+ rtp_payload_data_t rtp_payload_data; /* rtp payload parsing infos from SDP */
} RTSPStream;
static int rtsp_read_play(AVFormatContext *s);
/* XXX: currently, the only way to change the protocols consists in
changing this variable */
-#if 0
-int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_TCP) | (1 << RTSP_PROTOCOL_RTP_UDP) | (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST);
-#else
-/* try it if a proxy is used */
-int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_TCP);
-#endif
-/* if non zero, then set a range for RTP ports */
-int rtsp_rtp_port_min = 0;
-int rtsp_rtp_port_max = 0;
+int rtsp_default_protocols = (1 << RTSP_PROTOCOL_RTP_UDP);
FFRTSPCallback *ff_rtsp_callback = NULL;
@@ -113,6 +106,8 @@
char *q;
p = *pp;
+ if (*p == '/')
+ p++;
skip_spaces(&p);
q = buf;
while (!strchr(sep, *p) && *p != '\0') {
@@ -145,18 +140,67 @@
/* parse the rtpmap description: <codec_name>/<clock_rate>[/<other
params>] */
-static int sdp_parse_rtpmap(AVCodecContext *codec, const char *p)
+static int sdp_parse_rtpmap(AVCodecContext *codec, int payload_type, const char *p)
{
char buf[256];
+ int i;
+ AVCodec *c;
+ char *c_name;
- /* codec name */
+ /* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and
+ see if we can handle this kind of payload */
get_word_sep(buf, sizeof(buf), "/", &p);
- if (!strcmp(buf, "MP4V-ES")) {
- codec->codec_id = CODEC_ID_MPEG4;
- return 0;
+ if (payload_type >= RTP_PT_PRIVATE) {
+ /* We are in dynmaic payload type case ... search into AVRtpDynamicPayloadTypes[] */
+ for (i = 0; AVRtpDynamicPayloadTypes[i].codec_id != CODEC_ID_NONE; ++i)
+ if (!strcmp(buf, AVRtpDynamicPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpDynamicPayloadTypes[i].codec_type)) {
+ codec->codec_id = AVRtpDynamicPayloadTypes[i].codec_id;
+ break;
+ }
} else {
- return -1;
+ /* We are in a standard case ( from http://www.iana.org/assignments/rtp-parameters) */
+ /* search into AVRtpPayloadTypes[] */
+ for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
+ if (!strcmp(buf, AVRtpPayloadTypes[i].enc_name) && (codec->codec_type == AVRtpPayloadTypes[i].codec_type)){
+ codec->codec_id = AVRtpPayloadTypes[i].codec_id;
+ break;
+ }
+ }
+
+ c = avcodec_find_decoder(codec->codec_id);
+ if (c && c->name)
+ c_name = (char *)c->name;
+ else
+ c_name = (char *)NULL;
+
+ if (c_name) {
+ get_word_sep(buf, sizeof(buf), "/", &p);
+ i = atoi(buf);
+ switch (codec->codec_type) {
+ case CODEC_TYPE_AUDIO:
+ av_log(codec, AV_LOG_DEBUG, " audio codec set to : %s\n", c_name);
+ codec->sample_rate = RTSP_DEFAULT_AUDIO_SAMPLERATE;
+ codec->channels = RTSP_DEFAULT_NB_AUDIO_CHANNELS;
+ if (i > 0) {
+ codec->sample_rate = i;
+ get_word_sep(buf, sizeof(buf), "/", &p);
+ i = atoi(buf);
+ if (i > 0)
+ codec->channels = i;
+ }
+ av_log(codec, AV_LOG_DEBUG, " audio samplerate set to : %i\n", codec->sample_rate);
+ av_log(codec, AV_LOG_DEBUG, " audio channels set to : %i\n", codec->channels);
+ break;
+ case CODEC_TYPE_VIDEO:
+ av_log(codec, AV_LOG_DEBUG, " video codec set to : %s\n", c_name);
+ break;
+ default:
+ break;
+ }
+ return 0;
}
+
+ return -1;
}
/* return the length and optionnaly the data */
@@ -188,11 +232,58 @@
return len;
}
-static void sdp_parse_fmtp(AVCodecContext *codec, const char *p)
+static void sdp_parse_fmtp_config(AVCodecContext *codec, char *attr, char *value)
+{
+ switch (codec->codec_id) {
+ case CODEC_ID_MPEG4:
+ case CODEC_ID_MPEG4AAC:
+ if (!strcmp(attr, "config")) {
+ /* decode the hexa encoded parameter */
+ int len = hex_to_data(NULL, value);
+ codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
+ if (!codec->extradata)
+ return;
+ codec->extradata_size = len;
+ hex_to_data(codec->extradata, value);
+ }
+ break;
+ default:
+ break;
+ }
+ return;
+}
+
+typedef struct attrname_map
+{
+ char *str;
+ uint16_t type;
+ uint32_t offset;
+} attrname_map_t;
+
+/* All known fmtp parmeters and the corresping RTPAttrTypeEnum */
+#define ATTR_NAME_TYPE_INT 0
+#define ATTR_NAME_TYPE_STR 1
+static attrname_map_t attr_names[]=
+{
+ {"SizeLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, sizelength)},
+ {"IndexLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexlength)},
+ {"IndexDeltaLength", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, indexdeltalength)},
+ {"profile-level-id", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, profile_level_id)},
+ {"StreamType", ATTR_NAME_TYPE_INT, offsetof(rtp_payload_data_t, streamtype)},
+ {"mode", ATTR_NAME_TYPE_STR, offsetof(rtp_payload_data_t, mode)},
+ {NULL, -1, -1},
+};
+
+/* parse a SDP line and save stream attributes */
+static void sdp_parse_fmtp(AVStream *st, const char *p)
{
char attr[256];
char value[4096];
- int len;
+ int i;
+
+ RTSPStream *rtsp_st = st->priv_data;
+ AVCodecContext *codec = &st->codec;
+ rtp_payload_data_t *rtp_payload_data = &rtsp_st->rtp_payload_data;
/* loop on each attribute */
for(;;) {
@@ -205,25 +296,17 @@
get_word_sep(value, sizeof(value), ";", &p);
if (*p == ';')
p++;
- /* handle MPEG4 video */
- switch(codec->codec_id) {
- case CODEC_ID_MPEG4:
- if (!strcmp(attr, "config")) {
- /* decode the hexa encoded parameter */
- len = hex_to_data(NULL, value);
- codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
- if (!codec->extradata)
- goto fail;
- codec->extradata_size = len;
- hex_to_data(codec->extradata, value);
- }
- break;
- default:
- /* ignore data for other codecs */
- break;
+ /* grab the codec extra_data from the config parameter of the fmtp line */
+ sdp_parse_fmtp_config(codec, attr, value);
+ /* Looking for a known attribute */
+ for (i = 0; attr_names[i].str; ++i) {
+ if (!strcasecmp(attr, attr_names[i].str)) {
+ if (attr_names[i].type == ATTR_NAME_TYPE_INT)
+ *(int *)((char *)rtp_payload_data + attr_names[i].offset) = atoi(value);
+ else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
+ *(char **)((char *)rtp_payload_data + attr_names[i].offset) = av_strdup(value);
+ }
}
- fail: ;
- // printf("'%s' = '%s'\n", attr, value);
}
}
@@ -314,7 +397,7 @@
get_word(buf1, sizeof(buf1), &p); /* format list */
rtsp_st->sdp_payload_type = atoi(buf1);
- if (rtsp_st->sdp_payload_type == RTP_PT_MPEG2TS) {
+ if (!strcmp(AVRtpPayloadTypes[rtsp_st->sdp_payload_type].enc_name, "MP2T")) {
/* no corresponding stream */
} else {
st = av_new_stream(s, 0);
@@ -323,7 +406,7 @@
st->priv_data = rtsp_st;
rtsp_st->stream_index = st->index;
st->codec.codec_type = codec_type;
- if (rtsp_st->sdp_payload_type < 96) {
+ if (rtsp_st->sdp_payload_type < RTP_PT_PRIVATE) {
/* if standard payload type, we can find the codec right now */
rtp_get_codec_info(&st->codec, rtsp_st->sdp_payload_type);
}
@@ -355,7 +438,7 @@
st = s->streams[i];
rtsp_st = st->priv_data;
if (rtsp_st->sdp_payload_type == payload_type) {
- sdp_parse_rtpmap(&st->codec, p);
+ sdp_parse_rtpmap(&st->codec, payload_type, p);
}
}
} else if (strstart(p, "fmtp:", &p)) {
@@ -366,7 +449,7 @@
st = s->streams[i];
rtsp_st = st->priv_data;
if (rtsp_st->sdp_payload_type == payload_type) {
- sdp_parse_fmtp(&st->codec, p);
+ sdp_parse_fmtp(st, p);
}
}
}
@@ -715,7 +798,7 @@
RTSPState *rt = s->priv_data;
char host[1024], path[1024], tcpname[1024], cmd[2048];
URLContext *rtsp_hd;
- int port, i, ret, err;
+ int port, i, j, ret, err;
RTSPHeader reply1, *reply = &reply1;
unsigned char *content = NULL;
RTSPStream *rtsp_st;
@@ -763,7 +846,8 @@
/* for each stream, make the setup request */
/* XXX: we assume the same server is used for the control of each
RTSP stream */
- for(i=0;i<rt->nb_rtsp_streams;i++) {
+
+ for(j = RTSP_RTP_PORT_MIN, i = 0; i < rt->nb_rtsp_streams; ++i) {
char transport[2048];
rtsp_st = rt->rtsp_streams[i];
@@ -774,22 +858,24 @@
/* RTP/UDP */
if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP)) {
char buf[256];
- int j;
/* first try in specified port range */
- if (rtsp_rtp_port_min != 0) {
- for(j=rtsp_rtp_port_min;j<=rtsp_rtp_port_max;j++) {
+ if (RTSP_RTP_PORT_MIN != 0) {
+ while(j <= RTSP_RTP_PORT_MAX) {
snprintf(buf, sizeof(buf), "rtp://?localport=%d", j);
- if (url_open(&rtsp_st->rtp_handle, buf, URL_RDONLY) == 0)
+ if (url_open(&rtsp_st->rtp_handle, buf, URL_RDONLY) == 0) {
+ j += 2; /* we will use two port by rtp stream (rtp and rtcp) */
goto rtp_opened;
+ }
}
}
- /* then try on any port */
- if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
- err = AVERROR_INVALIDDATA;
- goto fail;
- }
+/* then try on any port
+** if (url_open(&rtsp_st->rtp_handle, "rtp://", URL_RDONLY) < 0) {
+** err = AVERROR_INVALIDDATA;
+** goto fail;
+** }
+*/
rtp_opened:
port = rtp_get_local_port(rtsp_st->rtp_handle);
@@ -801,14 +887,14 @@
}
/* RTP/TCP */
- if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
+ else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_TCP)) {
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
snprintf(transport + strlen(transport), sizeof(transport) - strlen(transport) - 1,
"RTP/AVP/TCP");
}
- if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
+ else if (protocol_mask & (1 << RTSP_PROTOCOL_RTP_UDP_MULTICAST)) {
if (transport[0] != '\0')
pstrcat(transport, sizeof(transport), ",");
snprintf(transport + strlen(transport),
@@ -887,7 +973,8 @@
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
- rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type);
+ rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
+
if (!rtsp_st->rtp_ctx) {
err = AVERROR_NOMEM;
goto fail;
@@ -1233,7 +1320,7 @@
st = s->streams[rtsp_st->stream_index];
if (!st)
s->ctx_flags |= AVFMTCTX_NOHEADER;
- rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type);
+ rtsp_st->rtp_ctx = rtp_parse_open(s, st, rtsp_st->sdp_payload_type, &rtsp_st->rtp_payload_data);
if (!rtsp_st->rtp_ctx) {
err = AVERROR_NOMEM;
goto fail;
Index: rtsp.h
===================================================================
RCS file: /cvsroot/ffmpeg/ffmpeg/libavformat/rtsp.h,v
retrieving revision 1.5
retrieving revision 1.6
diff -u -d -r1.5 -r1.6
--- rtsp.h 10 Nov 2003 18:39:26 -0000 1.5
+++ rtsp.h 26 May 2005 07:47:51 -0000 1.6
@@ -35,6 +35,10 @@
#define RTSP_DEFAULT_PORT 554
#define RTSP_MAX_TRANSPORTS 8
#define RTSP_TCP_MAX_PACKET_SIZE 1472
+#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 2
+#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
+#define RTSP_RTP_PORT_MIN 5000
+#define RTSP_RTP_PORT_MAX 10000
typedef struct RTSPTransportField {
int interleaved_min, interleaved_max; /* interleave ids, if TCP transport */
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