#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 },
+ { "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 },
};
#define RTCP_SR_SIZE 28
-static int is_supported(enum CodecID id)
+static int is_supported(enum AVCodecID id)
{
switch(id) {
- case CODEC_ID_H263:
- case CODEC_ID_H263P:
- case CODEC_ID_H264:
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
- case CODEC_ID_MPEG4:
- case CODEC_ID_AAC:
- case CODEC_ID_MP2:
- case CODEC_ID_MP3:
- case CODEC_ID_PCM_ALAW:
- case CODEC_ID_PCM_MULAW:
- case CODEC_ID_PCM_S8:
- case CODEC_ID_PCM_S16BE:
- case CODEC_ID_PCM_S16LE:
- case CODEC_ID_PCM_U16BE:
- case CODEC_ID_PCM_U16LE:
- case CODEC_ID_PCM_U8:
- case CODEC_ID_MPEG2TS:
- case CODEC_ID_AMR_NB:
- case CODEC_ID_AMR_WB:
- case CODEC_ID_VORBIS:
- case CODEC_ID_THEORA:
- case CODEC_ID_VP8:
- case CODEC_ID_ADPCM_G722:
- case CODEC_ID_ADPCM_G726:
+ case AV_CODEC_ID_H263:
+ case AV_CODEC_ID_H263P:
+ case AV_CODEC_ID_H264:
+ case AV_CODEC_ID_MPEG1VIDEO:
+ case AV_CODEC_ID_MPEG2VIDEO:
+ case AV_CODEC_ID_MPEG4:
+ case AV_CODEC_ID_AAC:
+ case AV_CODEC_ID_MP2:
+ case AV_CODEC_ID_MP3:
+ case AV_CODEC_ID_PCM_ALAW:
+ case AV_CODEC_ID_PCM_MULAW:
+ case AV_CODEC_ID_PCM_S8:
+ case AV_CODEC_ID_PCM_S16BE:
+ case AV_CODEC_ID_PCM_S16LE:
+ case AV_CODEC_ID_PCM_U16BE:
+ case AV_CODEC_ID_PCM_U16LE:
+ case AV_CODEC_ID_PCM_U8:
+ case AV_CODEC_ID_MPEG2TS:
+ case AV_CODEC_ID_AMR_NB:
+ case AV_CODEC_ID_AMR_WB:
+ case AV_CODEC_ID_VORBIS:
+ case AV_CODEC_ID_THEORA:
+ case AV_CODEC_ID_VP8:
+ 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;
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
- int max_packet_size, n;
+ int n;
AVStream *st;
- if (s1->nb_streams != 1)
- return -1;
+ if (s1->nb_streams != 1) {
+ av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
+ return AVERROR(EINVAL);
+ }
st = s1->streams[0];
if (!is_supported(st->codec->codec_id)) {
av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id);
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;
- s->ssrc = av_get_random_seed();
+ if (!s->ssrc)
+ s->ssrc = av_get_random_seed();
s->first_packet = 1;
s->first_rtcp_ntp_time = ff_ntp_time();
if (s1->start_time_realtime)
/* Round the NTP time to whole milliseconds. */
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
-
- max_packet_size = s1->pb->max_packet_size;
- if (max_packet_size <= 12)
+ // 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)
+ s1->packet_size = FFMIN(s1->packet_size,
+ s1->pb->max_packet_size);
+ } else
+ s1->packet_size = s1->pb->max_packet_size;
+ if (s1->packet_size <= 12) {
+ av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
return AVERROR(EIO);
- s->buf = av_malloc(max_packet_size);
+ }
+ s->buf = av_malloc(s1->packet_size);
if (s->buf == NULL) {
return AVERROR(ENOMEM);
}
- s->max_payload_size = max_packet_size - 12;
+ s->max_payload_size = s1->packet_size - 12;
s->max_frames_per_packet = 0;
- if (s1->max_delay) {
+ if (s1->max_delay > 0) {
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
- if (st->codec->frame_size == 0) {
+ int frame_size = av_get_audio_frame_duration(st->codec, 0);
+ if (!frame_size)
+ frame_size = st->codec->frame_size;
+ if (frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
- s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * (int64_t)st->codec->frame_size, AV_ROUND_DOWN);
+ s->max_frames_per_packet =
+ av_rescale_q_rnd(s1->max_delay,
+ AV_TIME_BASE_Q,
+ (AVRational){ frame_size, st->codec->sample_rate },
+ AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
avpriv_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
- case CODEC_ID_MP2:
- case CODEC_ID_MP3:
+ case AV_CODEC_ID_MP2:
+ case AV_CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
break;
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
+ case AV_CODEC_ID_MPEG1VIDEO:
+ case AV_CODEC_ID_MPEG2VIDEO:
break;
- case CODEC_ID_MPEG2TS:
+ case AV_CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
- case CODEC_ID_H264:
+ 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 CODEC_ID_VORBIS:
- case CODEC_ID_THEORA:
+ case AV_CODEC_ID_VORBIS:
+ case AV_CODEC_ID_THEORA:
if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
s->num_frames = 0;
goto defaultcase;
- case CODEC_ID_VP8:
- av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
- "incompatible with the latest spec drafts.\n");
- break;
- case CODEC_ID_ADPCM_G722:
+ 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 CODEC_ID_AMR_NB:
- case CODEC_ID_AMR_WB:
+ 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");
+ goto fail;
+ }
+ if (!s->max_frames_per_packet)
+ s->max_frames_per_packet = 1;
+ s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
+ s->max_payload_size / st->codec->block_align);
+ goto defaultcase;
+ case AV_CODEC_ID_AMR_NB:
+ case AV_CODEC_ID_AMR_WB:
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 12;
- if (st->codec->codec_id == CODEC_ID_AMR_NB)
+ if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
n = 31;
else
n = 61;
/* max_header_toc_size + the largest AMR payload must fit */
if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
- return -1;
+ goto fail;
}
if (st->codec->channels != 1) {
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
- return -1;
+ goto fail;
}
- case CODEC_ID_AAC:
+ case AV_CODEC_ID_AAC:
s->num_frames = 0;
default:
defaultcase:
}
return 0;
+
+fail:
+ av_freep(&s->buf);
+ return AVERROR(EINVAL);
}
/* send an rtcp sender report packet */
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);
+ }
+
avio_flush(s1->pb);
}
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,
}
}
+static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size)
+{
+ RTPMuxContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
+ int frame_duration = av_get_audio_frame_duration(st->codec, 0);
+ int frame_size = st->codec->block_align;
+ int frames = size / frame_size;
+
+ while (frames > 0) {
+ int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames);
+
+ if (!s->num_frames) {
+ s->buf_ptr = s->buf;
+ s->timestamp = s->cur_timestamp;
+ }
+ memcpy(s->buf_ptr, buf, n * frame_size);
+ frames -= n;
+ s->num_frames += n;
+ s->buf_ptr += n * frame_size;
+ buf += n * frame_size;
+ s->cur_timestamp += n * frame_duration;
+
+ if (s->num_frames == s->max_frames_per_packet) {
+ ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1);
+ s->num_frames = 0;
+ }
+ }
+ return 0;
+}
+
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
RTPMuxContext *s = s1->priv_data;
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
RTCP_TX_RATIO_DEN;
- if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
- (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
+ 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());
s->last_octet_count = s->octet_count;
s->first_packet = 0;
s->cur_timestamp = s->base_timestamp + pkt->pts;
switch(st->codec->codec_id) {
- case CODEC_ID_PCM_MULAW:
- case CODEC_ID_PCM_ALAW:
- case CODEC_ID_PCM_U8:
- case CODEC_ID_PCM_S8:
- rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
- break;
- case CODEC_ID_PCM_U16BE:
- case CODEC_ID_PCM_U16LE:
- case CODEC_ID_PCM_S16BE:
- case CODEC_ID_PCM_S16LE:
- rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
- break;
- case CODEC_ID_ADPCM_G722:
+ case AV_CODEC_ID_PCM_MULAW:
+ case AV_CODEC_ID_PCM_ALAW:
+ case AV_CODEC_ID_PCM_U8:
+ case AV_CODEC_ID_PCM_S8:
+ 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:
+ 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;
- case CODEC_ID_ADPCM_G726:
- rtp_send_samples(s1, pkt->data, size,
- st->codec->bits_per_coded_sample * st->codec->channels);
- break;
- case CODEC_ID_MP2:
- case CODEC_ID_MP3:
+ return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
+ case AV_CODEC_ID_ADPCM_G726:
+ 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);
break;
- case CODEC_ID_MPEG1VIDEO:
- case CODEC_ID_MPEG2VIDEO:
+ case AV_CODEC_ID_MPEG1VIDEO:
+ case AV_CODEC_ID_MPEG2VIDEO:
ff_rtp_send_mpegvideo(s1, pkt->data, size);
break;
- case CODEC_ID_AAC:
+ case AV_CODEC_ID_AAC:
if (s->flags & FF_RTP_FLAG_MP4A_LATM)
ff_rtp_send_latm(s1, pkt->data, size);
else
ff_rtp_send_aac(s1, pkt->data, size);
break;
- case CODEC_ID_AMR_NB:
- case CODEC_ID_AMR_WB:
+ case AV_CODEC_ID_AMR_NB:
+ case AV_CODEC_ID_AMR_WB:
ff_rtp_send_amr(s1, pkt->data, size);
break;
- case CODEC_ID_MPEG2TS:
+ case AV_CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, pkt->data, size);
break;
- case CODEC_ID_H264:
+ case AV_CODEC_ID_H264:
ff_rtp_send_h264(s1, pkt->data, size);
break;
- case CODEC_ID_H263:
- case CODEC_ID_H263P:
+ case AV_CODEC_ID_H263:
+ if (s->flags & FF_RTP_FLAG_RFC2190) {
+ int mb_info_size = 0;
+ const uint8_t *mb_info =
+ av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO,
+ &mb_info_size);
+ ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size);
+ break;
+ }
+ /* Fallthrough */
+ case AV_CODEC_ID_H263P:
ff_rtp_send_h263(s1, pkt->data, size);
break;
- case CODEC_ID_VORBIS:
- case CODEC_ID_THEORA:
+ case AV_CODEC_ID_VORBIS:
+ case AV_CODEC_ID_THEORA:
ff_rtp_send_xiph(s1, pkt->data, size);
break;
- case CODEC_ID_VP8:
+ case AV_CODEC_ID_VP8:
ff_rtp_send_vp8(s1, pkt->data, size);
break;
+ 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);
AVOutputFormat ff_rtp_muxer = {
.name = "rtp",
- .long_name = NULL_IF_CONFIG_SMALL("RTP output format"),
+ .long_name = NULL_IF_CONFIG_SMALL("RTP output"),
.priv_data_size = sizeof(RTPMuxContext),
- .audio_codec = CODEC_ID_PCM_MULAW,
- .video_codec = CODEC_ID_MPEG4,
+ .audio_codec = AV_CODEC_ID_PCM_MULAW,
+ .video_codec = AV_CODEC_ID_MPEG4,
.write_header = rtp_write_header,
.write_packet = rtp_write_packet,
.write_trailer = rtp_write_trailer,
- .priv_class = &rtp_muxer_class,
+ .priv_class = &rtp_muxer_class,
};
projects / ffmpeg / blobdiff