* RTP input format
* Copyright (c) 2002 Fabrice Bellard
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * 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.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/* needed for gethostname() */
-#define _XOPEN_SOURCE 600
-
+#include "libavutil/mathematics.h"
+#include "libavutil/avstring.h"
+#include "libavutil/time.h"
#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
+#include "url.h"
-#include <unistd.h>
-#include <strings.h>
#include "network.h"
#include "rtpdec.h"
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')
+ 'ffio_open_dyn_packet_buf')
*/
-static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
+static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
.enc_name = "X-MP3-draft-00",
.codec_type = AVMEDIA_TYPE_AUDIO,
- .codec_id = CODEC_ID_MP3ADU,
+ .codec_id = AV_CODEC_ID_MP3ADU,
+};
+
+static RTPDynamicProtocolHandler speex_dynamic_handler = {
+ .enc_name = "speex",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_SPEEX,
+};
+
+static RTPDynamicProtocolHandler opus_dynamic_handler = {
+ .enc_name = "opus",
+ .codec_type = AVMEDIA_TYPE_AUDIO,
+ .codec_id = AV_CODEC_ID_OPUS,
};
/* statistics functions */
ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
- ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
+ ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
+ ff_register_dynamic_payload_handler(&speex_dynamic_handler);
+ ff_register_dynamic_payload_handler(&opus_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_qt_rtp_vid_handler);
ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
+
+ ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
+ ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
}
RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
RTPDynamicProtocolHandler *handler;
for (handler = RTPFirstDynamicPayloadHandler;
handler; handler = handler->next)
- if (!strcasecmp(name, handler->enc_name) &&
+ if (!av_strcasecmp(name, handler->enc_name) &&
codec_type == handler->codec_type)
return handler;
return NULL;
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
{
int payload_len;
- while (len >= 2) {
+ while (len >= 4) {
+ payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
+
switch (buf[1]) {
case RTCP_SR:
- if (len < 16) {
+ if (payload_len < 20) {
av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
return AVERROR_INVALIDDATA;
}
- payload_len = (AV_RB16(buf + 2) + 1) * 4;
s->last_rtcp_ntp_time = AV_RB64(buf + 8);
s->last_rtcp_timestamp = AV_RB32(buf + 16);
s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
}
- buf += payload_len;
- len -= payload_len;
break;
case RTCP_BYE:
return -RTCP_BYE;
- default:
- return -1;
}
+
+ buf += payload_len;
+ len -= payload_len;
}
return -1;
}
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)
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
AVIOContext *pb;
uint8_t *buf;
return -1;
s->last_octet_count = s->octet_count;
- if (url_open_dyn_buf(&pb) < 0)
+ if (avio_open_dyn_buf(&pb) < 0)
return -1;
// Receiver Report
- put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
- put_byte(pb, RTCP_RR);
- put_be16(pb, 7); /* length in words - 1 */
+ avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
+ avio_w8(pb, RTCP_RR);
+ avio_wb16(pb, 7); /* length in words - 1 */
// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
- put_be32(pb, s->ssrc + 1);
- put_be32(pb, s->ssrc); // server SSRC
+ avio_wb32(pb, s->ssrc + 1);
+ avio_wb32(pb, s->ssrc); // server SSRC
// some placeholders we should really fill...
// RFC 1889/p64
extended_max= stats->cycles + stats->max_seq;
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 */
+ avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
+ avio_wb32(pb, extended_max); /* max sequence received */
+ avio_wb32(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 */
+ avio_wb32(pb, 0); /* last SR timestamp */
+ avio_wb32(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 */
+ avio_wb32(pb, middle_32_bits); /* last SR timestamp */
+ avio_wb32(pb, delay_since_last); /* delay since last SR */
}
// CNAME
- put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
- put_byte(pb, RTCP_SDES);
+ avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
+ avio_w8(pb, RTCP_SDES);
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);
+ avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
+ avio_wb32(pb, s->ssrc + 1);
+ avio_w8(pb, 0x01);
+ avio_w8(pb, len);
+ avio_write(pb, s->hostname, len);
// padding
for (len = (6 + len) % 4; len % 4; len++) {
- put_byte(pb, 0);
+ avio_w8(pb, 0);
}
- put_flush_packet(pb);
- len = url_close_dyn_buf(pb, &buf);
+ avio_flush(pb);
+ len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf) {
- int result;
+ int av_unused result;
av_dlog(s->ic, "sending %d bytes of RR\n", len);
- result= url_write(s->rtp_ctx, buf, len);
- av_dlog(s->ic, "result from url_write: %d\n", result);
+ result= ffurl_write(s->rtp_ctx, buf, len);
+ av_dlog(s->ic, "result from ffurl_write: %d\n", result);
av_free(buf);
}
return 0;
}
-void rtp_send_punch_packets(URLContext* rtp_handle)
+void ff_rtp_send_punch_packets(URLContext* rtp_handle)
{
AVIOContext *pb;
uint8_t *buf;
int len;
/* Send a small RTP packet */
- if (url_open_dyn_buf(&pb) < 0)
+ if (avio_open_dyn_buf(&pb) < 0)
return;
- put_byte(pb, (RTP_VERSION << 6));
- put_byte(pb, 0); /* Payload type */
- put_be16(pb, 0); /* Seq */
- put_be32(pb, 0); /* Timestamp */
- put_be32(pb, 0); /* SSRC */
+ avio_w8(pb, (RTP_VERSION << 6));
+ avio_w8(pb, 0); /* Payload type */
+ avio_wb16(pb, 0); /* Seq */
+ avio_wb32(pb, 0); /* Timestamp */
+ avio_wb32(pb, 0); /* SSRC */
- put_flush_packet(pb);
- len = url_close_dyn_buf(pb, &buf);
+ avio_flush(pb);
+ len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
- url_write(rtp_handle, buf, len);
+ ffurl_write(rtp_handle, buf, len);
av_free(buf);
/* Send a minimal RTCP RR */
- if (url_open_dyn_buf(&pb) < 0)
+ if (avio_open_dyn_buf(&pb) < 0)
return;
- put_byte(pb, (RTP_VERSION << 6));
- put_byte(pb, RTCP_RR); /* receiver report */
- put_be16(pb, 1); /* length in words - 1 */
- put_be32(pb, 0); /* our own SSRC */
+ avio_w8(pb, (RTP_VERSION << 6));
+ avio_w8(pb, RTCP_RR); /* receiver report */
+ avio_wb16(pb, 1); /* length in words - 1 */
+ avio_wb32(pb, 0); /* our own SSRC */
- put_flush_packet(pb);
- len = url_close_dyn_buf(pb, &buf);
+ avio_flush(pb);
+ len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
- url_write(rtp_handle, buf, len);
+ ffurl_write(rtp_handle, buf, len);
av_free(buf);
}
/**
* 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)
+ * rtp demux (otherwise AV_CODEC_ID_MPEG2TS packets are returned)
*/
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
{
RTPDemuxContext *s;
av_free(s);
return NULL;
}
- } else {
+ } else if (st) {
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_H263:
- case CODEC_ID_H264:
+ case AV_CODEC_ID_MPEG1VIDEO:
+ case AV_CODEC_ID_MPEG2VIDEO:
+ case AV_CODEC_ID_MP2:
+ case AV_CODEC_ID_MP3:
+ case AV_CODEC_ID_MPEG4:
+ case AV_CODEC_ID_H263:
+ case AV_CODEC_ID_H264:
st->need_parsing = AVSTREAM_PARSE_FULL;
break;
- case CODEC_ID_ADPCM_G722:
+ case AV_CODEC_ID_VORBIS:
+ st->need_parsing = AVSTREAM_PARSE_HEADERS;
+ break;
+ case AV_CODEC_ID_ADPCM_G722:
/* According to RFC 3551, the stream clock rate is 8000
* even if the sample rate is 16000. */
if (st->codec->sample_rate == 8000)
}
void
-rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
- RTPDynamicProtocolHandler *handler)
+ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+ RTPDynamicProtocolHandler *handler)
{
s->dynamic_protocol_context = ctx;
s->parse_packet = handler->parse_packet;
{
if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
return; /* Timestamp already set by depacketizer */
- if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
+ if (timestamp == RTP_NOTS_VALUE)
+ return;
+
+ if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
int64_t addend;
int delta_timestamp;
delta_timestamp;
return;
}
- if (timestamp == RTP_NOTS_VALUE)
- return;
+
if (!s->base_timestamp)
s->base_timestamp = timestamp;
- pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
+ /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
+ if (!s->timestamp)
+ s->unwrapped_timestamp += timestamp;
+ else
+ s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
+ s->timestamp = timestamp;
+ pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
}
static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
} 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:
- case CODEC_ID_MP3:
+ case AV_CODEC_ID_MP2:
+ case AV_CODEC_ID_MP3:
/* better than nothing: skip mpeg audio RTP header */
if (len <= 4)
return -1;
av_new_packet(pkt, len);
memcpy(pkt->data, buf, len);
break;
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
+ case AV_CODEC_ID_MPEG1VIDEO:
+ case AV_CODEC_ID_MPEG2VIDEO:
/* better than nothing: skip mpeg video RTP header */
if (len <= 4)
return -1;
if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
return -1;
- if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
+ if (RTP_PT_IS_RTCP(buf[1])) {
return rtcp_parse_packet(s, buf, 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,
- uint8_t **bufptr, int len)
+int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
+ uint8_t **bufptr, int len)
{
int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
s->prev_ret = rv;
return rv ? rv : has_next_packet(s);
}
-void rtp_parse_close(RTPDemuxContext *s)
+void ff_rtp_parse_close(RTPDemuxContext *s)
{
ff_rtp_reset_packet_queue(s);
if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
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