#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, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM },
- { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM },
+ { "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 },
};
case AV_CODEC_ID_ADPCM_G722:
case AV_CODEC_ID_ADPCM_G726:
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;
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)
s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
s->num_frames = 0;
goto defaultcase;
- case AV_CODEC_ID_VP8:
- av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
- "incompatible with the latest spec drafts.\n");
- break;
case AV_CODEC_ID_ADPCM_G722:
/* Due to a historical error, the clock rate for G722 in RTP is
* 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");
}
/* 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++;
}
/* 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_bits)
+static int rtp_send_samples(AVFormatContext *s1,
+ const uint8_t *buf1, int size, int sample_size_bits)
{
RTPMuxContext *s = s1->priv_data;
int len, max_packet_size, n;
max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
/* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
- av_abort();
+ return AVERROR(EINVAL);
n = 0;
while (size > 0) {
s->buf_ptr = s->buf;
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
n += (s->buf_ptr - s->buf);
}
+ return 0;
}
static void rtp_send_mpegaudio(AVFormatContext *s1,
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_PCM_ALAW:
case AV_CODEC_ID_PCM_U8:
case AV_CODEC_ID_PCM_S8:
- rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
- break;
+ return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
case AV_CODEC_ID_PCM_U16BE:
case AV_CODEC_ID_PCM_U16LE:
case AV_CODEC_ID_PCM_S16BE:
case AV_CODEC_ID_PCM_S16LE:
- rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
- break;
+ return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
case AV_CODEC_ID_ADPCM_G722:
/* The actual sample size is half a byte per sample, but since the
* stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
* the correct parameter for send_samples_bits is 8 bits per stream
* clock. */
- rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
- break;
+ return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
case AV_CODEC_ID_ADPCM_G726:
- rtp_send_samples(s1, pkt->data, size,
- st->codec->bits_per_coded_sample * st->codec->channels);
- break;
+ return rtp_send_samples(s1, pkt->data, size,
+ st->codec->bits_per_coded_sample * st->codec->channels);
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
rtp_send_mpegaudio(s1, pkt->data, size);
case AV_CODEC_ID_ILBC:
rtp_send_ilbc(s1, pkt->data, size);
break;
+ 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;
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