rtpenc: Allow packetizing H263 according to the old RFC 2190

According to newer RFCs, this packetization scheme should only
be used for interfacing with legacy systems.

Implementing this packetization mode properly requires parsing
the full H263 bitstream to find macroblock boundaries (and knowing
their macroblock and gob numbers and motion vector predictors).

This implementation tries to look for GOB headers (which
can be inserted by using -ps <small number>), but if the GOBs
aren't small enough to fit into the MTU, the packetizer blindly
splits packets at any offset and claims it to be a GOB boundary
(by using Mode A from the RFC). While not correct, this seems
to work with some receivers.

Signed-off-by: Martin Storsjö <martin@martin.st>
pull/3/merge
Martin Storsjö 13 years ago
parent c2ff63e3ac
commit c4584f3c1f
  1. 1
      libavformat/Makefile
  2. 4
      libavformat/rtp.c
  3. 5
      libavformat/rtpenc.c
  4. 6
      libavformat/rtpenc.h
  5. 7
      libavformat/rtpenc_h263.c
  6. 104
      libavformat/rtpenc_h263_rfc2190.c
  7. 3
      libavformat/sdp.c

@ -242,6 +242,7 @@ OBJS-$(CONFIG_RTP_MUXER) += rtp.o \
rtpenc_latm.o \ rtpenc_latm.o \
rtpenc_amr.o \ rtpenc_amr.o \
rtpenc_h263.o \ rtpenc_h263.o \
rtpenc_h263_rfc2190.o \
rtpenc_mpv.o \ rtpenc_mpv.o \
rtpenc.o \ rtpenc.o \
rtpenc_h264.o \ rtpenc_h264.o \

@ -106,7 +106,9 @@ int ff_rtp_get_payload_type(AVFormatContext *fmt, AVCodecContext *codec)
/* static payload type */ /* static payload type */
for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) for (i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) { if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
if (codec->codec_id == CODEC_ID_H263) if (codec->codec_id == CODEC_ID_H263 && (!fmt ||
!fmt->oformat->priv_class ||
!av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190")))
continue; continue;
if (codec->codec_id == CODEC_ID_PCM_S16BE) if (codec->codec_id == CODEC_ID_PCM_S16BE)
if (codec->channels != AVRtpPayloadTypes[i].audio_channels) if (codec->channels != AVRtpPayloadTypes[i].audio_channels)

@ -443,6 +443,11 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
ff_rtp_send_h264(s1, pkt->data, size); ff_rtp_send_h264(s1, pkt->data, size);
break; break;
case CODEC_ID_H263: case CODEC_ID_H263:
if (s->flags & FF_RTP_FLAG_RFC2190) {
ff_rtp_send_h263_rfc2190(s1, pkt->data, size);
break;
}
/* Fallthrough */
case CODEC_ID_H263P: case CODEC_ID_H263P:
ff_rtp_send_h263(s1, pkt->data, size); ff_rtp_send_h263(s1, pkt->data, size);
break; break;

@ -64,15 +64,18 @@ struct RTPMuxContext {
typedef struct RTPMuxContext RTPMuxContext; typedef struct RTPMuxContext RTPMuxContext;
#define FF_RTP_FLAG_MP4A_LATM 1 #define FF_RTP_FLAG_MP4A_LATM 1
#define FF_RTP_FLAG_RFC2190 2
#define FF_RTP_FLAG_OPTS(ctx, fieldname) \ #define FF_RTP_FLAG_OPTS(ctx, fieldname) \
{ "rtpflags", "RTP muxer flags", offsetof(ctx, fieldname), AV_OPT_TYPE_FLAGS, {.dbl = 0}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \ { "rtpflags", "RTP muxer flags", offsetof(ctx, fieldname), AV_OPT_TYPE_FLAGS, {.dbl = 0}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
{ "latm", "Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_MP4A_LATM}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \ { "latm", "Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_MP4A_LATM}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
{ "rfc2190", "Use RFC 2190 packetization instead of RFC 4629 for H.263", 0, AV_OPT_TYPE_CONST, {.dbl = FF_RTP_FLAG_RFC2190}, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM, "rtpflags" }, \
void ff_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);
void ff_rtp_send_h264(AVFormatContext *s1, const uint8_t *buf1, int size); void ff_rtp_send_h264(AVFormatContext *s1, const uint8_t *buf1, int size);
void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size); void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size);
void ff_rtp_send_h263_rfc2190(AVFormatContext *s1, const uint8_t *buf1, int size);
void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size); void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size);
void ff_rtp_send_latm(AVFormatContext *s1, const uint8_t *buff, int size); void ff_rtp_send_latm(AVFormatContext *s1, const uint8_t *buff, int size);
void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size); void ff_rtp_send_amr(AVFormatContext *s1, const uint8_t *buff, int size);
@ -80,4 +83,7 @@ void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size);
void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size); void ff_rtp_send_xiph(AVFormatContext *s1, const uint8_t *buff, int size);
void ff_rtp_send_vp8(AVFormatContext *s1, const uint8_t *buff, int size); void ff_rtp_send_vp8(AVFormatContext *s1, const uint8_t *buff, int size);
const uint8_t *ff_h263_find_resync_marker_reverse(const uint8_t *restrict start,
const uint8_t *restrict end);
#endif /* AVFORMAT_RTPENC_H */ #endif /* AVFORMAT_RTPENC_H */

@ -23,8 +23,8 @@
#include "avformat.h" #include "avformat.h"
#include "rtpenc.h" #include "rtpenc.h"
static const uint8_t *find_resync_marker_reverse(const uint8_t *restrict start, const uint8_t *ff_h263_find_resync_marker_reverse(const uint8_t *restrict start,
const uint8_t *restrict end) const uint8_t *restrict end)
{ {
const uint8_t *p = end - 1; const uint8_t *p = end - 1;
start += 1; /* Make sure we never return the original start. */ start += 1; /* Make sure we never return the original start. */
@ -63,7 +63,8 @@ void ff_rtp_send_h263(AVFormatContext *s1, const uint8_t *buf1, int size)
/* Look for a better place to split the frame into packets. */ /* Look for a better place to split the frame into packets. */
if (len < size) { if (len < size) {
const uint8_t *end = find_resync_marker_reverse(buf1, buf1 + len); const uint8_t *end = ff_h263_find_resync_marker_reverse(buf1,
buf1 + len);
len = end - buf1; len = end - buf1;
} }

@ -0,0 +1,104 @@
/*
* RTP packetization for H.263 video
* Copyright (c) 2012 Martin Storsjo
*
* 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 2.1 of the License, or (at your option) any later version.
*
* 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avformat.h"
#include "rtpenc.h"
#include "libavcodec/put_bits.h"
#include "libavcodec/get_bits.h"
struct H263Info {
int src;
int i;
int u;
int s;
int a;
int pb;
int tr;
};
static void send_mode_a(AVFormatContext *s1, const struct H263Info *info,
const uint8_t *buf, int len, int m)
{
RTPMuxContext *s = s1->priv_data;
PutBitContext pb;
init_put_bits(&pb, s->buf, 32);
put_bits(&pb, 1, 0); /* F - 0, mode A */
put_bits(&pb, 1, 0); /* P - 0, normal I/P */
put_bits(&pb, 3, 0); /* SBIT - 0 bits */
put_bits(&pb, 3, 0); /* EBIT - 0 bits */
put_bits(&pb, 3, info->src); /* SRC - source format */
put_bits(&pb, 1, info->i); /* I - inter/intra */
put_bits(&pb, 1, info->u); /* U - unrestricted motion vector */
put_bits(&pb, 1, info->s); /* S - syntax-baesd arithmetic coding */
put_bits(&pb, 1, info->a); /* A - advanced prediction */
put_bits(&pb, 4, 0); /* R - reserved */
put_bits(&pb, 2, 0); /* DBQ - 0 */
put_bits(&pb, 3, 0); /* TRB - 0 */
put_bits(&pb, 8, info->tr); /* TR */
flush_put_bits(&pb);
memcpy(s->buf + 4, buf, len);
ff_rtp_send_data(s1, s->buf, len + 4, m);
}
void ff_rtp_send_h263_rfc2190(AVFormatContext *s1, const uint8_t *buf, int size)
{
RTPMuxContext *s = s1->priv_data;
int len;
GetBitContext gb;
struct H263Info info = { 0 };
s->timestamp = s->cur_timestamp;
init_get_bits(&gb, buf, size*8);
if (get_bits(&gb, 22) == 0x20) { /* Picture Start Code */
info.tr = get_bits(&gb, 8);
skip_bits(&gb, 2); /* PTYPE start, H261 disambiguation */
skip_bits(&gb, 3); /* Split screen, document camera, freeze picture release */
info.src = get_bits(&gb, 3);
info.i = get_bits(&gb, 1);
info.u = get_bits(&gb, 1);
info.s = get_bits(&gb, 1);
info.a = get_bits(&gb, 1);
info.pb = get_bits(&gb, 1);
}
while (size > 0) {
len = FFMIN(s->max_payload_size - 4, size);
/* Look for a better place to split the frame into packets. */
if (len < size) {
const uint8_t *end = ff_h263_find_resync_marker_reverse(buf,
buf + len);
len = end - buf;
if (len == s->max_payload_size - 4)
av_log(s1, AV_LOG_WARNING,
"No GOB boundary found within MTU size, splitting at "
"a random boundary\n");
}
send_mode_a(s1, &info, buf, len, len == size);
buf += len;
size -= len;
}
}

@ -404,6 +404,9 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
* actually specifies the maximum video size, but we only know * actually specifies the maximum video size, but we only know
* the current size. This is required for playback on Android * the current size. This is required for playback on Android
* stagefright and on Samsung bada. */ * stagefright and on Samsung bada. */
if (!fmt || !fmt->oformat->priv_class ||
!av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190") ||
c->codec_id == CODEC_ID_H263P)
av_strlcatf(buff, size, "a=rtpmap:%d H263-2000/90000\r\n" av_strlcatf(buff, size, "a=rtpmap:%d H263-2000/90000\r\n"
"a=framesize:%d %d-%d\r\n", "a=framesize:%d %d-%d\r\n",
payload_type, payload_type,

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