#include "rtpenc.h"
-//#define DEBUG
-
static const AVOption options[] = {
FF_RTP_FLAG_OPTS(RTPMuxContext, flags),
{ "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
{ "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
+ { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM },
+ { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM },
{ NULL },
};
static int is_supported(enum AVCodecID id)
{
switch(id) {
+ case AV_CODEC_ID_H261:
case AV_CODEC_ID_H263:
case AV_CODEC_ID_H263P:
case AV_CODEC_ID_H264:
+ case AV_CODEC_ID_HEVC:
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
case AV_CODEC_ID_MPEG4:
case AV_CODEC_ID_ILBC:
case AV_CODEC_ID_MJPEG:
case AV_CODEC_ID_SPEEX:
+ case AV_CODEC_ID_OPUS:
return 1;
default:
return 0;
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
- int n;
+ int n, ret = AVERROR(EINVAL);
AVStream *st;
if (s1->nb_streams != 1) {
return -1;
}
- if (s->payload_type < 0)
- s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
+ if (s->payload_type < 0) {
+ /* Re-validate non-dynamic payload types */
+ if (st->id < RTP_PT_PRIVATE)
+ st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
+
+ s->payload_type = st->id;
+ } else {
+ /* private option takes priority */
+ st->id = s->payload_type;
+ }
+
s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
/* Round the NTP time to whole milliseconds. */
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
+ // Pick a random sequence start number, but in the lower end of the
+ // available range, so that any wraparound doesn't happen immediately.
+ // (Immediate wraparound would be an issue for SRTP.)
+ if (s->seq < 0)
+ s->seq = av_get_random_seed() & 0x0fff;
+ else
+ s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
if (s1->packet_size) {
if (s1->pb->max_packet_size)
return AVERROR(EIO);
}
s->buf = av_malloc(s1->packet_size);
- if (s->buf == NULL) {
+ if (!s->buf) {
return AVERROR(ENOMEM);
}
s->max_payload_size = s1->packet_size - 12;
}
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
/* FIXME: We should round down here... */
- s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
+ if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) {
+ s->max_frames_per_packet = av_rescale_q(s1->max_delay,
+ (AVRational){1, 1000000},
+ av_inv_q(st->avg_frame_rate));
+ } else
+ s->max_frames_per_packet = 1;
}
}
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
+ case AV_CODEC_ID_H261:
+ if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
+ av_log(s, AV_LOG_ERROR,
+ "Packetizing H261 is experimental and produces incorrect "
+ "packetization for cases where GOBs don't fit into packets "
+ "(even though most receivers may handle it just fine). "
+ "Please set -f_strict experimental in order to enable it.\n");
+ ret = AVERROR_EXPERIMENTAL;
+ goto fail;
+ }
+ break;
case AV_CODEC_ID_H264:
/* check for H.264 MP4 syntax */
if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
}
break;
+ case AV_CODEC_ID_HEVC:
+ /* Only check for the standardized hvcC version of extradata, keeping
+ * things simple and similar to the avcC/H264 case above, instead
+ * of trying to handle the pre-standardization versions (as in
+ * libavcodec/hevc.c). */
+ if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) {
+ s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1;
+ }
+ break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
* 8000, even if the sample rate is 16000. See RFC 3551. */
avpriv_set_pts_info(st, 32, 1, 8000);
break;
+ case AV_CODEC_ID_OPUS:
+ if (st->codec->channels > 2) {
+ av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
+ goto fail;
+ }
+ /* The opus RTP RFC says that all opus streams should use 48000 Hz
+ * as clock rate, since all opus sample rates can be expressed in
+ * this clock rate, and sample rate changes on the fly are supported. */
+ avpriv_set_pts_info(st, 32, 1, 48000);
+ break;
case AV_CODEC_ID_ILBC:
if (st->codec->block_align != 38 && st->codec->block_align != 50) {
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
goto fail;
}
+ s->num_frames = 0;
+ goto defaultcase;
case AV_CODEC_ID_AAC:
s->num_frames = 0;
+ goto defaultcase;
default:
defaultcase:
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
fail:
av_freep(&s->buf);
- return AVERROR(EINVAL);
+ return ret;
}
/* send an rtcp sender report packet */
-static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
+static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye)
{
RTPMuxContext *s = s1->priv_data;
uint32_t rtp_ts;
s->last_rtcp_ntp_time = ntp_time;
rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000},
s1->streams[0]->time_base) + s->base_timestamp;
- avio_w8(s1->pb, (RTP_VERSION << 6));
+ avio_w8(s1->pb, RTP_VERSION << 6);
avio_w8(s1->pb, RTCP_SR);
avio_wb16(s1->pb, 6); /* length in words - 1 */
avio_wb32(s1->pb, s->ssrc);
- avio_wb32(s1->pb, ntp_time / 1000000);
- avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
+ avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time));
avio_wb32(s1->pb, rtp_ts);
avio_wb32(s1->pb, s->packet_count);
avio_wb32(s1->pb, s->octet_count);
+
+ if (s->cname) {
+ int len = FFMIN(strlen(s->cname), 255);
+ avio_w8(s1->pb, (RTP_VERSION << 6) + 1);
+ avio_w8(s1->pb, RTCP_SDES);
+ avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */
+
+ avio_wb32(s1->pb, s->ssrc);
+ avio_w8(s1->pb, 0x01); /* CNAME */
+ avio_w8(s1->pb, len);
+ avio_write(s1->pb, s->cname, len);
+ avio_w8(s1->pb, 0); /* END */
+ for (len = (7 + len) % 4; len % 4; len++)
+ avio_w8(s1->pb, 0);
+ }
+
+ if (bye) {
+ avio_w8(s1->pb, (RTP_VERSION << 6) | 1);
+ avio_w8(s1->pb, RTCP_BYE);
+ avio_wb16(s1->pb, 1); /* length in words - 1 */
+ avio_wb32(s1->pb, s->ssrc);
+ }
+
avio_flush(s1->pb);
}
av_dlog(s1, "rtp_send_data size=%d\n", len);
/* build the RTP header */
- avio_w8(s1->pb, (RTP_VERSION << 6));
+ avio_w8(s1->pb, RTP_VERSION << 6);
avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
avio_wb16(s1->pb, s->seq);
avio_wb32(s1->pb, s->timestamp);
avio_write(s1->pb, buf1, len);
avio_flush(s1->pb);
- s->seq++;
+ s->seq = (s->seq + 1) & 0xffff;
s->octet_count += len;
s->packet_count++;
}
RTPMuxContext *s = s1->priv_data;
int len, out_len;
+ s->timestamp = s->cur_timestamp;
while (size >= TS_PACKET_SIZE) {
len = s->max_payload_size - (s->buf_ptr - s->buf);
if (len > size)
if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
(ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) &&
!(s->flags & FF_RTP_FLAG_SKIP_RTCP)) {
- rtcp_send_sr(s1, ff_ntp_time());
+ rtcp_send_sr(s1, ff_ntp_time(), 0);
s->last_octet_count = s->octet_count;
s->first_packet = 0;
}
case AV_CODEC_ID_H264:
ff_rtp_send_h264(s1, pkt->data, size);
break;
+ case AV_CODEC_ID_H261:
+ ff_rtp_send_h261(s1, pkt->data, size);
+ break;
case AV_CODEC_ID_H263:
if (s->flags & FF_RTP_FLAG_RFC2190) {
int mb_info_size = 0;
case AV_CODEC_ID_H263P:
ff_rtp_send_h263(s1, pkt->data, size);
break;
+ case AV_CODEC_ID_HEVC:
+ ff_rtp_send_hevc(s1, pkt->data, size);
+ break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
ff_rtp_send_xiph(s1, pkt->data, size);
case AV_CODEC_ID_MJPEG:
ff_rtp_send_jpeg(s1, pkt->data, size);
break;
+ case AV_CODEC_ID_OPUS:
+ if (size > s->max_payload_size) {
+ av_log(s1, AV_LOG_ERROR,
+ "Packet size %d too large for max RTP payload size %d\n",
+ size, s->max_payload_size);
+ return AVERROR(EINVAL);
+ }
+ /* Intentional fallthrough */
default:
/* better than nothing : send the codec raw data */
rtp_send_raw(s1, pkt->data, size);
{
RTPMuxContext *s = s1->priv_data;
+ /* If the caller closes and recreates ->pb, this might actually
+ * be NULL here even if it was successfully allocated at the start. */
+ if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE))
+ rtcp_send_sr(s1, ff_ntp_time(), 1);
av_freep(&s->buf);
return 0;
.write_packet = rtp_write_packet,
.write_trailer = rtp_write_trailer,
.priv_class = &rtp_muxer_class,
+ .flags = AVFMT_TS_NONSTRICT,
};
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