* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#include "libavcodec/bitstream.h"
#include "avformat.h"
#include "mpegts.h"
-
-#include <unistd.h>
-#include "network.h"
+#include "internal.h"
+#include "libavutil/random_seed.h"
#include "rtpenc.h"
//#define DEBUG
#define RTCP_SR_SIZE 28
-#define NTP_OFFSET 2208988800ULL
-#define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL)
-static uint64_t ntp_time(void)
+static int is_supported(enum CodecID id)
{
- return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US;
+ 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:
+ return 1;
+ default:
+ return 0;
+ }
}
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
- int payload_type, max_packet_size, n;
+ int max_packet_size, n;
AVStream *st;
if (s1->nb_streams != 1)
return -1;
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);
- payload_type = rtp_get_payload_type(st->codec);
- if (payload_type < 0)
- payload_type = RTP_PT_PRIVATE; /* private payload type */
- s->payload_type = payload_type;
+ return -1;
+ }
-// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
- s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
+ s->payload_type = ff_rtp_get_payload_type(st->codec);
+ if (s->payload_type < 0)
+ s->payload_type = RTP_PT_PRIVATE + (st->codec->codec_type == AVMEDIA_TYPE_AUDIO);
+
+ s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
- s->ssrc = 0; /* FIXME: was random(), what should this be? */
+ s->ssrc = av_get_random_seed();
s->first_packet = 1;
- s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
+ 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 = url_fget_max_packet_size(s1->pb);
+ max_packet_size = s1->pb->max_packet_size;
if (max_packet_size <= 12)
return AVERROR(EIO);
s->buf = av_malloc(max_packet_size);
s->max_frames_per_packet = 0;
if (s1->max_delay) {
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
if (st->codec->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 * st->codec->frame_size, AV_ROUND_DOWN);
}
}
- if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
+ 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);
}
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
+ case 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:
+ 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:
+ /* Due to a historical error, the clock rate for G722 in RTP is
+ * 8000, even if the sample rate is 16000. See RFC 3551. */
+ av_set_pts_info(st, 32, 1, 8000);
+ break;
+ case CODEC_ID_AMR_NB:
+ case 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)
+ 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;
+ }
+ if (st->codec->channels != 1) {
+ av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
+ return -1;
+ }
case CODEC_ID_AAC:
s->num_frames = 0;
default:
- if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
+defaultcase:
+ if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf;
RTPMuxContext *s = s1->priv_data;
uint32_t rtp_ts;
- dprintf(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
+ av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
- if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
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;
- put_byte(s1->pb, (RTP_VERSION << 6));
- put_byte(s1->pb, 200);
- put_be16(s1->pb, 6); /* length in words - 1 */
- put_be32(s1->pb, s->ssrc);
- put_be32(s1->pb, ntp_time / 1000000);
- put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
- put_be32(s1->pb, rtp_ts);
- put_be32(s1->pb, s->packet_count);
- put_be32(s1->pb, s->octet_count);
- put_flush_packet(s1->pb);
+ 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_wb32(s1->pb, rtp_ts);
+ avio_wb32(s1->pb, s->packet_count);
+ avio_wb32(s1->pb, s->octet_count);
+ avio_flush(s1->pb);
}
/* send an rtp packet. sequence number is incremented, but the caller
{
RTPMuxContext *s = s1->priv_data;
- dprintf(s1, "rtp_send_data size=%d\n", len);
+ av_dlog(s1, "rtp_send_data size=%d\n", len);
/* build the RTP header */
- put_byte(s1->pb, (RTP_VERSION << 6));
- put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
- put_be16(s1->pb, s->seq);
- put_be32(s1->pb, s->timestamp);
- put_be32(s1->pb, s->ssrc);
+ 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_wb32(s1->pb, s->ssrc);
- put_buffer(s1->pb, buf1, len);
- put_flush_packet(s1->pb);
+ avio_write(s1->pb, buf1, len);
+ avio_flush(s1->pb);
s->seq++;
s->octet_count += len;
}
}
-/* NOTE: we suppose that exactly one frame is given as argument here */
-/* XXX: test it */
static void rtp_send_mpegaudio(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
}
}
-/* write an RTP packet. 'buf1' must contain a single specific frame. */
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
RTPMuxContext *s = s1->priv_data;
AVStream *st = s1->streams[0];
int rtcp_bytes;
int size= pkt->size;
- uint8_t *buf1= pkt->data;
- dprintf(s1, "%d: write len=%d\n", pkt->stream_index, size);
+ av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size);
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) &&
- (ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
- rtcp_send_sr(s1, ntp_time());
+ (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) {
+ rtcp_send_sr(s1, ff_ntp_time());
s->last_octet_count = s->octet_count;
s->first_packet = 0;
}
case CODEC_ID_PCM_ALAW:
case CODEC_ID_PCM_U8:
case CODEC_ID_PCM_S8:
- rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
+ rtp_send_samples(s1, pkt->data, size, 1 * 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, buf1, size, 2 * st->codec->channels);
+ rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
+ break;
+ case 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 is 1 byte per stream clock. */
+ rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
break;
case CODEC_ID_MP2:
case CODEC_ID_MP3:
- rtp_send_mpegaudio(s1, buf1, size);
+ rtp_send_mpegaudio(s1, pkt->data, size);
break;
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
- ff_rtp_send_mpegvideo(s1, buf1, size);
+ ff_rtp_send_mpegvideo(s1, pkt->data, size);
break;
case CODEC_ID_AAC:
- ff_rtp_send_aac(s1, buf1, size);
+ ff_rtp_send_aac(s1, pkt->data, size);
+ break;
+ case CODEC_ID_AMR_NB:
+ case CODEC_ID_AMR_WB:
+ ff_rtp_send_amr(s1, pkt->data, size);
break;
case CODEC_ID_MPEG2TS:
- rtp_send_mpegts_raw(s1, buf1, size);
+ rtp_send_mpegts_raw(s1, pkt->data, size);
break;
case CODEC_ID_H264:
- ff_rtp_send_h264(s1, buf1, size);
+ ff_rtp_send_h264(s1, pkt->data, size);
+ break;
+ case CODEC_ID_H263:
+ case CODEC_ID_H263P:
+ ff_rtp_send_h263(s1, pkt->data, size);
+ break;
+ case CODEC_ID_VORBIS:
+ case CODEC_ID_THEORA:
+ ff_rtp_send_xiph(s1, pkt->data, size);
+ break;
+ case CODEC_ID_VP8:
+ ff_rtp_send_vp8(s1, pkt->data, size);
break;
default:
/* better than nothing : send the codec raw data */
- rtp_send_raw(s1, buf1, size);
+ rtp_send_raw(s1, pkt->data, size);
break;
}
return 0;
return 0;
}
-AVOutputFormat rtp_muxer = {
+AVOutputFormat ff_rtp_muxer = {
"rtp",
NULL_IF_CONFIG_SMALL("RTP output format"),
NULL,
projects / ffmpeg / blobdiff