* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#include "libavcodec/bitstream.h"
#include "avformat.h"
#include "mpegts.h"
+#include "internal.h"
#include <unistd.h>
-#include "network.h"
-#include "rtp_internal.h"
-#include "rtp_mpv.h"
-#include "rtp_aac.h"
-#include "rtp_h264.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:
+ return 1;
+ default:
+ return 0;
+ }
}
static int rtp_write_header(AVFormatContext *s1)
{
- RTPDemuxContext *s = s1->priv_data;
- int payload_type, max_packet_size, n;
+ RTPMuxContext *s = s1->priv_data;
+ 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);
+
+ return -1;
+ }
- 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;
+ 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 == CODEC_TYPE_AUDIO);
// 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->cur_timestamp = 0;
s->ssrc = 0; /* FIXME: was random(), what should this be? */
s->first_packet = 1;
- s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
+ s->first_rtcp_ntp_time = ff_ntp_time();
max_packet_size = url_fget_max_packet_size(s1->pb);
if (max_packet_size <= 12)
return AVERROR(EIO);
+ s->buf = av_malloc(max_packet_size);
+ if (s->buf == NULL) {
+ return AVERROR(ENOMEM);
+ }
s->max_payload_size = max_packet_size - 12;
s->max_frames_per_packet = 0;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
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->read_buf_index = 0;
+ s->num_frames = 0;
default:
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
uint32_t rtp_ts;
dprintf(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;
must update the timestamp itself */
void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
dprintf(s1, "rtp_send_data size=%d\n", len);
static void rtp_send_samples(AVFormatContext *s1,
const uint8_t *buf1, int size, int sample_size)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
int len, max_packet_size, n;
max_packet_size = (s->max_payload_size / sample_size) * sample_size;
}
}
-/* 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)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
int len, count, max_packet_size;
max_packet_size = s->max_payload_size;
static void rtp_send_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
int len, max_packet_size;
max_packet_size = s->max_payload_size;
static void rtp_send_mpegts_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
- RTPDemuxContext *s = s1->priv_data;
+ RTPMuxContext *s = s1->priv_data;
int len, out_len;
while (size >= TS_PACKET_SIZE) {
}
}
-/* write an RTP packet. 'buf1' must contain a single specific frame. */
static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
- RTPDemuxContext *s = s1->priv_data;
+ 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);
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_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;
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;
}
+static int rtp_write_trailer(AVFormatContext *s1)
+{
+ RTPMuxContext *s = s1->priv_data;
+
+ av_freep(&s->buf);
+
+ return 0;
+}
+
AVOutputFormat rtp_muxer = {
"rtp",
NULL_IF_CONFIG_SMALL("RTP output format"),
NULL,
NULL,
- sizeof(RTPDemuxContext),
+ sizeof(RTPMuxContext),
CODEC_ID_PCM_MULAW,
CODEC_ID_NONE,
rtp_write_header,
rtp_write_packet,
+ rtp_write_trailer,
};
projects / ffmpeg / blobdiff