[FFmpeg-cvslog] rtpdec: Split mpegts parsing to a normal depacketizer
Martin Storsjö
git at videolan.org
Mon Jan 21 13:57:09 CET 2013
ffmpeg | branch: master | Martin Storsjö <martin at martin.st> | Thu Jan 17 14:34:59 2013 +0200| [2326558d5277ec87ba6d607a01ec6acfc51c694c] | committer: Martin Storsjö
rtpdec: Split mpegts parsing to a normal depacketizer
This gets rid of a number of special cases from the common rtpdec
code.
Signed-off-by: Martin Storsjö <martin at martin.st>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=2326558d5277ec87ba6d607a01ec6acfc51c694c
---
libavformat/Makefile | 1 +
libavformat/rtpdec.c | 56 ++++------------------
libavformat/rtpdec.h | 6 ---
libavformat/rtpdec_formats.h | 1 +
libavformat/rtpdec_mpegts.c | 106 ++++++++++++++++++++++++++++++++++++++++++
libavformat/rtsp.c | 16 +++++--
6 files changed, 131 insertions(+), 55 deletions(-)
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 111380a..a8a7530 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -35,6 +35,7 @@ OBJS-$(CONFIG_RTPDEC) += rdt.o \
rtpdec_jpeg.o \
rtpdec_latm.o \
rtpdec_mpeg4.o \
+ rtpdec_mpegts.o \
rtpdec_qcelp.o \
rtpdec_qdm2.o \
rtpdec_qt.o \
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 28b4033..7fa0a93 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -24,7 +24,6 @@
#include "libavutil/time.h"
#include "libavcodec/get_bits.h"
#include "avformat.h"
-#include "mpegts.h"
#include "network.h"
#include "srtp.h"
#include "url.h"
@@ -76,6 +75,7 @@ void av_register_rtp_dynamic_payload_handlers(void)
ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
@@ -481,8 +481,7 @@ int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
- * MPEG2-TS streams to indicate that they should be demuxed inside the
- * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
+ * MPEG2-TS streams.
*/
RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
int payload_type, int queue_size)
@@ -499,13 +498,7 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
s->st = st;
s->queue_size = queue_size;
rtp_init_statistics(&s->statistics, 0);
- if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
- s->ts = ff_mpegts_parse_open(s->ic);
- if (s->ts == NULL) {
- av_free(s);
- return NULL;
- }
- } else if (st) {
+ if (st) {
switch (st->codec->codec_id) {
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
@@ -591,7 +584,7 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len)
{
unsigned int ssrc, h;
- int payload_type, seq, ret, flags = 0;
+ int payload_type, seq, flags = 0;
int ext;
AVStream *st;
uint32_t timestamp;
@@ -645,26 +638,11 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
buf += ext;
}
- if (!st) {
- /* specific MPEG2-TS demux support */
- ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
- /* The only error that can be returned from ff_mpegts_parse_packet
- * is "no more data to return from the provided buffer", so return
- * AVERROR(EAGAIN) for all errors */
- if (ret < 0)
- return AVERROR(EAGAIN);
- if (ret < len) {
- s->read_buf_size = FFMIN(len - ret, sizeof(s->buf));
- memcpy(s->buf, buf + ret, s->read_buf_size);
- s->read_buf_index = 0;
- return 1;
- }
- return 0;
- } else if (s->handler && s->handler->parse_packet) {
+ if (s->handler && s->handler->parse_packet) {
rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
s->st, pkt, ×tamp, buf, len, seq,
flags);
- } else {
+ } else if (st) {
/* At this point, the RTP header has been stripped;
* This is ASSUMING that there is only 1 CSRC, which isn't wise. */
switch (st->codec->codec_id) {
@@ -704,6 +682,8 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
}
pkt->stream_index = st->index;
+ } else {
+ return AVERROR(EINVAL);
}
// now perform timestamp things....
@@ -786,7 +766,7 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
uint8_t **bufptr, int len)
{
uint8_t *buf = bufptr ? *bufptr : NULL;
- int ret, flags = 0;
+ int flags = 0;
uint32_t timestamp;
int rv = 0;
@@ -797,7 +777,7 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
if (s->prev_ret <= 0)
return rtp_parse_queued_packet(s, pkt);
/* return the next packets, if any */
- if (s->st && s->handler && s->handler->parse_packet) {
+ if (s->handler && s->handler->parse_packet) {
/* timestamp should be overwritten by parse_packet, if not,
* the packet is left with pts == AV_NOPTS_VALUE */
timestamp = RTP_NOTS_VALUE;
@@ -806,19 +786,6 @@ static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
flags);
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 AVERROR(EAGAIN);
- ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
- s->read_buf_size - s->read_buf_index);
- if (ret < 0)
- return AVERROR(EAGAIN);
- s->read_buf_index += ret;
- if (s->read_buf_index < s->read_buf_size)
- return 1;
- else
- return 0;
}
}
@@ -894,9 +861,6 @@ int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
void ff_rtp_parse_close(RTPDemuxContext *s)
{
ff_rtp_reset_packet_queue(s);
- if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
- ff_mpegts_parse_close(s->ts);
- }
ff_srtp_free(&s->srtp);
av_free(s);
}
diff --git a/libavformat/rtpdec.h b/libavformat/rtpdec.h
index eaef993..a93963a 100644
--- a/libavformat/rtpdec.h
+++ b/libavformat/rtpdec.h
@@ -160,9 +160,6 @@ struct RTPDemuxContext {
int64_t unwrapped_timestamp;
int64_t range_start_offset;
int max_payload_size;
- struct MpegTSContext *ts; /* only used for MP2T payloads */
- int read_buf_index;
- int read_buf_size;
/* used to send back RTCP RR */
char hostname[256];
@@ -192,9 +189,6 @@ struct RTPDemuxContext {
unsigned int last_octet_count;
int64_t last_feedback_time;
- /* buffer for partially parsed packets */
- uint8_t buf[RTP_MAX_PACKET_LENGTH];
-
/* dynamic payload stuff */
const RTPDynamicProtocolHandler *handler;
PayloadContext *dynamic_protocol_context;
diff --git a/libavformat/rtpdec_formats.h b/libavformat/rtpdec_formats.h
index e449ddf..6c07791 100644
--- a/libavformat/rtpdec_formats.h
+++ b/libavformat/rtpdec_formats.h
@@ -50,6 +50,7 @@ extern RTPDynamicProtocolHandler ff_jpeg_dynamic_handler;
extern RTPDynamicProtocolHandler ff_mp4a_latm_dynamic_handler;
extern RTPDynamicProtocolHandler ff_mp4v_es_dynamic_handler;
extern RTPDynamicProtocolHandler ff_mpeg4_generic_dynamic_handler;
+extern RTPDynamicProtocolHandler ff_mpegts_dynamic_handler;
extern RTPDynamicProtocolHandler ff_ms_rtp_asf_pfa_handler;
extern RTPDynamicProtocolHandler ff_ms_rtp_asf_pfv_handler;
extern RTPDynamicProtocolHandler ff_qcelp_dynamic_handler;
diff --git a/libavformat/rtpdec_mpegts.c b/libavformat/rtpdec_mpegts.c
new file mode 100644
index 0000000..e2c8977
--- /dev/null
+++ b/libavformat/rtpdec_mpegts.c
@@ -0,0 +1,106 @@
+/*
+ * RTP MPEG2TS depacketizer
+ * Copyright (c) 2003 Fabrice Bellard
+ *
+ * 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 "mpegts.h"
+#include "rtpdec_formats.h"
+
+struct PayloadContext {
+ struct MpegTSContext *ts;
+ int read_buf_index;
+ int read_buf_size;
+ uint8_t buf[RTP_MAX_PACKET_LENGTH];
+};
+
+static PayloadContext *mpegts_new_context(void)
+{
+ return av_mallocz(sizeof(PayloadContext));
+}
+
+static void mpegts_free_context(PayloadContext *data)
+{
+ if (!data)
+ return;
+ if (data->ts)
+ ff_mpegts_parse_close(data->ts);
+ av_free(data);
+}
+
+static int mpegts_init(AVFormatContext *ctx, int st_index, PayloadContext *data)
+{
+ data->ts = ff_mpegts_parse_open(ctx);
+ if (!data->ts)
+ return AVERROR(ENOMEM);
+ return 0;
+}
+
+static int mpegts_handle_packet(AVFormatContext *ctx, PayloadContext *data,
+ AVStream *st, AVPacket *pkt, uint32_t *timestamp,
+ const uint8_t *buf, int len, uint16_t seq,
+ int flags)
+{
+ int ret;
+
+ // We don't want to use the RTP timestamps at all. If the mpegts demuxer
+ // doesn't set any pts/dts, the generic rtpdec code shouldn't try to
+ // fill it in either, since the mpegts and RTP timestamps are in totally
+ // different ranges.
+ *timestamp = RTP_NOTS_VALUE;
+
+ if (!data->ts)
+ return AVERROR(EINVAL);
+
+ if (!buf) {
+ if (data->read_buf_index >= data->read_buf_size)
+ return AVERROR(EAGAIN);
+ ret = ff_mpegts_parse_packet(data->ts, pkt, data->buf + data->read_buf_index,
+ data->read_buf_size - data->read_buf_index);
+ if (ret < 0)
+ return AVERROR(EAGAIN);
+ data->read_buf_index += ret;
+ if (data->read_buf_index < data->read_buf_size)
+ return 1;
+ else
+ return 0;
+ }
+
+ ret = ff_mpegts_parse_packet(data->ts, pkt, buf, len);
+ /* The only error that can be returned from ff_mpegts_parse_packet
+ * is "no more data to return from the provided buffer", so return
+ * AVERROR(EAGAIN) for all errors */
+ if (ret < 0)
+ return AVERROR(EAGAIN);
+ if (ret < len) {
+ data->read_buf_size = FFMIN(len - ret, sizeof(data->buf));
+ memcpy(data->buf, buf + ret, data->read_buf_size);
+ data->read_buf_index = 0;
+ return 1;
+ }
+ return 0;
+}
+
+RTPDynamicProtocolHandler ff_mpegts_dynamic_handler = {
+ .codec_type = AVMEDIA_TYPE_DATA,
+ .parse_packet = mpegts_handle_packet,
+ .alloc = mpegts_new_context,
+ .init = mpegts_init,
+ .free = mpegts_free_context,
+ .static_payload_id = 33,
+};
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 9a657fb..1ce28d6 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -180,7 +180,8 @@ static void init_rtp_handler(RTPDynamicProtocolHandler *handler,
{
if (!handler)
return;
- codec->codec_id = handler->codec_id;
+ if (codec)
+ codec->codec_id = handler->codec_id;
rtsp_st->dynamic_handler = handler;
if (handler->alloc) {
rtsp_st->dynamic_protocol_context = handler->alloc();
@@ -382,8 +383,17 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
/* no corresponding stream */
- if (rt->transport == RTSP_TRANSPORT_RAW && !rt->ts && CONFIG_RTPDEC)
- rt->ts = ff_mpegts_parse_open(s);
+ if (rt->transport == RTSP_TRANSPORT_RAW) {
+ if (!rt->ts && CONFIG_RTPDEC)
+ rt->ts = ff_mpegts_parse_open(s);
+ } else {
+ RTPDynamicProtocolHandler *handler;
+ handler = ff_rtp_handler_find_by_id(
+ rtsp_st->sdp_payload_type, AVMEDIA_TYPE_DATA);
+ init_rtp_handler(handler, rtsp_st, NULL);
+ if (handler && handler->init)
+ handler->init(s, -1, rtsp_st->dynamic_protocol_context);
+ }
} else if (rt->server_type == RTSP_SERVER_WMS &&
codec_type == AVMEDIA_TYPE_DATA) {
/* RTX stream, a stream that carries all the other actual
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