[FFmpeg-cvslog] r10201 - in trunk/libavformat: Makefile rtp.c rtp_internal.h rtp_mpv.c rtp_mpv.h
lucabe
subversion
Fri Aug 24 09:13:34 CEST 2007
Author: lucabe
Date: Fri Aug 24 09:13:34 2007
New Revision: 10201
Log:
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
Added:
trunk/libavformat/rtp_mpv.c
- copied, changed from r10200, /trunk/libavformat/rtp.c
trunk/libavformat/rtp_mpv.h
Modified:
trunk/libavformat/Makefile
trunk/libavformat/rtp.c
trunk/libavformat/rtp_internal.h
Modified: trunk/libavformat/Makefile
==============================================================================
--- trunk/libavformat/Makefile (original)
+++ trunk/libavformat/Makefile Fri Aug 24 09:13:34 2007
@@ -119,7 +119,7 @@ OBJS-$(CONFIG_RM_DEMUXER)
OBJS-$(CONFIG_RM_MUXER) += rmenc.o
OBJS-$(CONFIG_ROQ_DEMUXER) += idroq.o
OBJS-$(CONFIG_ROQ_MUXER) += raw.o
-OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtp_h264.o
+OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtp_h264.o rtp_mpv.o
OBJS-$(CONFIG_RTSP_DEMUXER) += rtsp.o
OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o
OBJS-$(CONFIG_SEGAFILM_DEMUXER) += segafilm.o
Modified: trunk/libavformat/rtp.c
==============================================================================
--- trunk/libavformat/rtp.c (original)
+++ trunk/libavformat/rtp.c Fri Aug 24 09:13:34 2007
@@ -27,6 +27,7 @@
#include "rtp_internal.h"
#include "rtp_h264.h"
+#include "rtp_mpv.h"
//#define DEBUG
@@ -788,7 +789,7 @@ static void rtcp_send_sr(AVFormatContext
/* 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, int m)
+void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
RTPDemuxContext *s = s1->priv_data;
@@ -836,7 +837,7 @@ static void rtp_send_samples(AVFormatCon
n = (s->buf_ptr - s->buf);
/* if buffer full, then send it */
if (n >= max_packet_size) {
- rtp_send_data(s1, s->buf, n, 0);
+ ff_rtp_send_data(s1, s->buf, n, 0);
s->buf_ptr = s->buf;
/* update timestamp */
s->timestamp += n / sample_size;
@@ -859,7 +860,7 @@ static void rtp_send_mpegaudio(AVFormatC
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, 0);
+ ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
s->buf_ptr = s->buf + 4;
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
@@ -881,7 +882,7 @@ static void rtp_send_mpegaudio(AVFormatC
s->buf[2] = count >> 8;
s->buf[3] = count;
memcpy(s->buf + 4, buf1, len);
- rtp_send_data(s1, s->buf, len + 4, 0);
+ ff_rtp_send_data(s1, s->buf, len + 4, 0);
size -= len;
buf1 += len;
count += len;
@@ -900,55 +901,6 @@ static void rtp_send_mpegaudio(AVFormatC
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;
-
- /* 90 KHz time stamp */
- s->timestamp = s->base_timestamp +
- av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
- rtp_send_data(s1, s->buf, q - s->buf, (len == size));
-
- buf1 += len;
- size -= len;
- }
- s->cur_timestamp++;
-}
-
static void rtp_send_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
@@ -966,7 +918,7 @@ static void rtp_send_raw(AVFormatContext
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
- rtp_send_data(s1, buf1, len, (len == size));
+ ff_rtp_send_data(s1, buf1, len, (len == size));
buf1 += len;
size -= len;
@@ -992,7 +944,7 @@ static void rtp_send_mpegts_raw(AVFormat
out_len = s->buf_ptr - s->buf;
if (out_len >= s->max_payload_size) {
- rtp_send_data(s1, s->buf, out_len, 0);
+ ff_rtp_send_data(s1, s->buf, out_len, 0);
s->buf_ptr = s->buf;
}
}
@@ -1042,7 +994,7 @@ static int rtp_write_packet(AVFormatCont
rtp_send_mpegaudio(s1, buf1, size);
break;
case CODEC_ID_MPEG1VIDEO:
- rtp_send_mpegvideo(s1, buf1, size);
+ ff_rtp_send_mpegvideo(s1, buf1, size);
break;
case CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, buf1, size);
Modified: trunk/libavformat/rtp_internal.h
==============================================================================
--- trunk/libavformat/rtp_internal.h (original)
+++ trunk/libavformat/rtp_internal.h Fri Aug 24 09:13:34 2007
@@ -110,5 +110,7 @@ struct RTPDemuxContext {
extern RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler;
int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size); ///< from rtsp.c, but used by rtp dynamic protocol handlers.
+
+void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m);
#endif /* RTP_INTERNAL_H */
Copied: trunk/libavformat/rtp_mpv.c (from r10200, /trunk/libavformat/rtp.c)
==============================================================================
--- /trunk/libavformat/rtp.c (original)
+++ trunk/libavformat/rtp_mpv.c Fri Aug 24 09:13:34 2007
@@ -1,5 +1,5 @@
/*
- * RTP input/output format
+ * RTP packetization for MPEG video
* Copyright (c) 2002 Fabrice Bellard.
*
* This file is part of FFmpeg.
@@ -19,891 +19,11 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avformat.h"
-#include "mpegts.h"
-#include "bitstream.h"
-
-#include <unistd.h>
-#include "network.h"
-
#include "rtp_internal.h"
-#include "rtp_h264.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')
-*/
-
-/* 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_QCELP, 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}
-};
-
-/* statistics functions */
-RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
-
-static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
-static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
-
-static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
-{
- handler->next= RTPFirstDynamicPayloadHandler;
- RTPFirstDynamicPayloadHandler= handler;
-}
-
-void av_register_rtp_dynamic_payload_handlers(void)
-{
- register_dynamic_payload_handler(&mp4v_es_handler);
- register_dynamic_payload_handler(&mpeg4_generic_handler);
- register_dynamic_payload_handler(&ff_h264_dynamic_handler);
-}
-
-int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
-{
- if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
- codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
- codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
- 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 -1;
-}
-
-int rtp_get_payload_type(AVCodecContext *codec)
-{
- int i, payload_type;
-
- /* compute the payload type */
- 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;
- }
- return payload_type;
-}
-
-static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
-{
- if (buf[1] != 200)
- return -1;
- s->last_rtcp_ntp_time = AV_RB64(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 = AV_RB32(buf + 16);
- return 0;
-}
-
-#define RTP_SEQ_MOD (1<<16)
-
-/**
-* called on parse open packet
-*/
-static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
-{
- memset(s, 0, sizeof(RTPStatistics));
- s->max_seq= base_sequence;
- s->probation= 1;
-}
-
-/**
-* called whenever there is a large jump in sequence numbers, or when they get out of probation...
-*/
-static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
-{
- s->max_seq= seq;
- s->cycles= 0;
- s->base_seq= seq -1;
- s->bad_seq= RTP_SEQ_MOD + 1;
- s->received= 0;
- s->expected_prior= 0;
- s->received_prior= 0;
- s->jitter= 0;
- s->transit= 0;
-}
-
-/**
-* returns 1 if we should handle this packet.
-*/
-static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
-{
- uint16_t udelta= seq - s->max_seq;
- const int MAX_DROPOUT= 3000;
- const int MAX_MISORDER = 100;
- const int MIN_SEQUENTIAL = 2;
-
- /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
- if(s->probation)
- {
- if(seq==s->max_seq + 1) {
- s->probation--;
- s->max_seq= seq;
- if(s->probation==0) {
- rtp_init_sequence(s, seq);
- s->received++;
- return 1;
- }
- } else {
- s->probation= MIN_SEQUENTIAL - 1;
- s->max_seq = seq;
- }
- } else if (udelta < MAX_DROPOUT) {
- // in order, with permissible gap
- if(seq < s->max_seq) {
- //sequence number wrapped; count antother 64k cycles
- s->cycles += RTP_SEQ_MOD;
- }
- s->max_seq= seq;
- } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
- // sequence made a large jump...
- if(seq==s->bad_seq) {
- // two sequential packets-- assume that the other side restarted without telling us; just resync.
- rtp_init_sequence(s, seq);
- } else {
- s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
- return 0;
- }
- } else {
- // duplicate or reordered packet...
- }
- s->received++;
- return 1;
-}
-
-#if 0
-/**
-* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
-* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
-* never change. I left this in in case someone else can see a way. (rdm)
-*/
-static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
-{
- uint32_t transit= arrival_timestamp - sent_timestamp;
- int d;
- s->transit= transit;
- d= FFABS(transit - s->transit);
- s->jitter += d - ((s->jitter + 8)>>4);
-}
-#endif
-
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
-{
- ByteIOContext pb;
- uint8_t *buf;
- int len;
- int rtcp_bytes;
- RTPStatistics *stats= &s->statistics;
- uint32_t lost;
- uint32_t extended_max;
- uint32_t expected_interval;
- uint32_t received_interval;
- uint32_t lost_interval;
- uint32_t expected;
- uint32_t fraction;
- uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
-
- if (!s->rtp_ctx || (count < 1))
- return -1;
-
- /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
- /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
- s->octet_count += count;
- rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
- RTCP_TX_RATIO_DEN;
- rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
- if (rtcp_bytes < 28)
- return -1;
- s->last_octet_count = s->octet_count;
-
- if (url_open_dyn_buf(&pb) < 0)
- return -1;
-
- // Receiver Report
- put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
- put_byte(&pb, 201);
- put_be16(&pb, 7); /* length in words - 1 */
- put_be32(&pb, s->ssrc); // our own SSRC
- put_be32(&pb, s->ssrc); // XXX: should be the server's here!
- // some placeholders we should really fill...
- // RFC 1889/p64
- extended_max= stats->cycles + stats->max_seq;
- expected= extended_max - stats->base_seq + 1;
- lost= expected - stats->received;
- lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
- expected_interval= expected - stats->expected_prior;
- stats->expected_prior= expected;
- received_interval= stats->received - stats->received_prior;
- stats->received_prior= stats->received;
- lost_interval= expected_interval - received_interval;
- if (expected_interval==0 || lost_interval<=0) fraction= 0;
- else fraction = (lost_interval<<8)/expected_interval;
-
- fraction= (fraction<<24) | lost;
-
- put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
- put_be32(&pb, extended_max); /* max sequence received */
- put_be32(&pb, stats->jitter>>4); /* jitter */
-
- if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
- {
- put_be32(&pb, 0); /* last SR timestamp */
- put_be32(&pb, 0); /* delay since last SR */
- } else {
- uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
- uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
-
- put_be32(&pb, middle_32_bits); /* last SR timestamp */
- put_be32(&pb, delay_since_last); /* delay since last SR */
- }
-
- // CNAME
- put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
- put_byte(&pb, 202);
- len = strlen(s->hostname);
- put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
- put_be32(&pb, s->ssrc);
- put_byte(&pb, 0x01);
- put_byte(&pb, len);
- put_buffer(&pb, s->hostname, len);
- // padding
- for (len = (6 + len) % 4; len % 4; len++) {
- put_byte(&pb, 0);
- }
-
- put_flush_packet(&pb);
- len = url_close_dyn_buf(&pb, &buf);
- if ((len > 0) && buf) {
- int result;
-#if defined(DEBUG)
- printf("sending %d bytes of RR\n", len);
-#endif
- result= url_write(s->rtp_ctx, buf, len);
-#if defined(DEBUG)
- printf("result from url_write: %d\n", result);
-#endif
- av_free(buf);
- }
- return 0;
-}
-
-/**
- * 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)
- * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
- */
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
-{
- 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;
- s->rtp_payload_data = rtp_payload_data;
- rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
- if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
- 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:
- case CODEC_ID_H264:
- st->need_parsing = AVSTREAM_PARSE_FULL;
- break;
- default:
- break;
- }
- }
- // needed to send back RTCP RR in RTSP sessions
- s->rtp_ctx = rtpc;
- gethostname(s->hostname, sizeof(s->hostname));
- return s;
-}
-
-static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
-{
- 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;
-
- /* decode the first 2 bytes where are stored the AUHeader sections
- length in bits */
- au_headers_length = AV_RB16(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 does not 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;
-}
-
-/**
- * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
- */
-static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
-{
- switch(s->st->codec->codec_id) {
- case CODEC_ID_MP2:
- case CODEC_ID_MPEG1VIDEO:
- if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
- int64_t addend;
-
- int delta_timestamp;
- /* 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;
- }
- break;
- case CODEC_ID_AAC:
- case CODEC_ID_H264:
- case CODEC_ID_MPEG4:
- pkt->pts = timestamp;
- break;
- default:
- /* no timestamp info yet */
- break;
- }
- pkt->stream_index = s->st->index;
-}
-
-/**
- * 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)
-{
- unsigned int ssrc, h;
- int payload_type, seq, ret;
- AVStream *st;
- uint32_t timestamp;
- int rv= 0;
-
- if (!buf) {
- /* return the next packets, if any */
- if(s->st && s->parse_packet) {
- timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
- rv= s->parse_packet(s, pkt, ×tamp, NULL, 0);
- finalize_packet(s, pkt, timestamp);
- return rv;
- } else {
- // TODO: Move to a dynamic packet handler (like above)
- 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;
- }
- }
-
- 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 = AV_RB16(buf + 2);
- timestamp = AV_RB32(buf + 4);
- ssrc = AV_RB32(buf + 8);
- /* store the ssrc in the RTPDemuxContext */
- s->ssrc = ssrc;
-
- /* NOTE: we can handle only one payload type */
- if (s->payload_type != payload_type)
- return -1;
-
- st = s->st;
- // only do something with this if all the rtp checks pass...
- if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
- {
- av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
- payload_type, seq, ((s->seq + 1) & 0xffff));
- return -1;
- }
-
- s->seq = seq;
- len -= 12;
- buf += 12;
-
- 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 {
- // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
- switch(st->codec->codec_id) {
- case CODEC_ID_MP2:
- /* better than nothing: skip mpeg audio RTP header */
- if (len <= 4)
- return -1;
- h = AV_RB32(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 = AV_RB32(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;
- // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
- // timestamps.
- // TODO: Put this into a dynamic packet handler...
- case CODEC_ID_AAC:
- 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 = buf;
- rv= 0;
- break;
- default:
- if(s->parse_packet) {
- rv= s->parse_packet(s, pkt, ×tamp, buf, len);
- } else {
- av_new_packet(pkt, len);
- memcpy(pkt->data, buf, len);
- }
- break;
- }
-
- // now perform timestamp things....
- finalize_packet(s, pkt, timestamp);
- }
- return rv;
-}
-
-void rtp_parse_close(RTPDemuxContext *s)
-{
- // TODO: fold this into the protocol specific data fields.
- if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
- 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;
-
-// following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
- s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
- s->timestamp = s->base_timestamp;
- s->ssrc = 0; /* FIXME: was random(), what should this be? */
- s->first_packet = 1;
-
- max_packet_size = url_fget_max_packet_size(&s1->pb);
- if (max_packet_size <= 12)
- return AVERROR(EIO);
- 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 %"PRIx64" %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, int m)
-{
- 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) | ((m & 0x01) << 7));
- put_be16(&s1->pb, s->seq);
- put_be32(&s1->pb, s->timestamp);
- put_be32(&s1->pb, s->ssrc);
-
- put_buffer(&s1->pb, buf1, len);
- put_flush_packet(&s1->pb);
-
- s->seq++;
- s->octet_count += len;
- s->packet_count++;
-}
-
-/* send an integer number of samples and compute time stamp and fill
- the rtp send buffer before sending. */
-static void rtp_send_samples(AVFormatContext *s1,
- const uint8_t *buf1, int size, int sample_size)
-{
- 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, 0);
- 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, 0);
- 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, 0);
- size -= len;
- buf1 += len;
- count += len;
- }
- } else {
- if (s->buf_ptr == s->buf + 4) {
- /* no fragmentation possible */
- s->buf[0] = 0;
- s->buf[1] = 0;
- s->buf[2] = 0;
- s->buf[3] = 0;
- }
- memcpy(s->buf_ptr, buf1, size);
- s->buf_ptr += size;
- }
- 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)
+void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
{
RTPDemuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
@@ -941,32 +61,7 @@ static void rtp_send_mpegvideo(AVFormatC
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
- rtp_send_data(s1, s->buf, q - s->buf, (len == size));
-
- buf1 += len;
- size -= len;
- }
- s->cur_timestamp++;
-}
-
-static void rtp_send_raw(AVFormatContext *s1,
- const uint8_t *buf1, int size)
-{
- 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;
-
- /* 90 KHz time stamp */
- s->timestamp = s->base_timestamp +
- av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
- rtp_send_data(s1, buf1, len, (len == size));
+ ff_rtp_send_data(s1, s->buf, q - s->buf, (len == size));
buf1 += len;
size -= len;
@@ -974,95 +69,4 @@ static void rtp_send_raw(AVFormatContext
s->cur_timestamp++;
}
-/* 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)
-{
- RTPDemuxContext *s = s1->priv_data;
- int len, out_len;
- while (size >= TS_PACKET_SIZE) {
- len = s->max_payload_size - (s->buf_ptr - s->buf);
- if (len > size)
- len = size;
- memcpy(s->buf_ptr, buf1, len);
- buf1 += len;
- size -= len;
- s->buf_ptr += len;
-
- out_len = s->buf_ptr - s->buf;
- if (out_len >= s->max_payload_size) {
- rtp_send_data(s1, s->buf, out_len, 0);
- s->buf_ptr = s->buf;
- }
- }
-}
-
-/* write an RTP packet. 'buf1' must contain a single specific frame. */
-static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
-{
- RTPDemuxContext *s = s1->priv_data;
- AVStream *st = s1->streams[0];
- int rtcp_bytes;
- int64_t ntp_time;
- int size= pkt->size;
- uint8_t *buf1= pkt->data;
-
-#ifdef DEBUG
- printf("%d: write len=%d\n", pkt->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 = (pkt->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;
-}
-
-AVOutputFormat rtp_muxer = {
- "rtp",
- "RTP output format",
- NULL,
- NULL,
- sizeof(RTPDemuxContext),
- CODEC_ID_PCM_MULAW,
- CODEC_ID_NONE,
- rtp_write_header,
- rtp_write_packet,
-};
Added: trunk/libavformat/rtp_mpv.h
==============================================================================
--- (empty file)
+++ trunk/libavformat/rtp_mpv.h Fri Aug 24 09:13:34 2007
@@ -0,0 +1,26 @@
+/*
+ * RTP definitions
+ * Copyright (c) 2002 Fabrice Bellard.
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+#ifndef RTP_MPV_H
+#define RTP_MPV_H
+
+void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size);
+
+#endif /* RTP_MPV_H */
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