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https://github.com/FFmpeg/FFmpeg.git
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Split the RTP muxer out of rtp.c, to simplify the RTSP demuxer's dependencies
Originally committed as revision 11408 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Luca Abeni
parent
9389e63c83
commit
83a0d3878c
@ -121,9 +121,9 @@ OBJS-$(CONFIG_RM_DEMUXER) += rmdec.o
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OBJS-$(CONFIG_RM_MUXER) += rmenc.o
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OBJS-$(CONFIG_ROQ_DEMUXER) += idroq.o
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OBJS-$(CONFIG_ROQ_MUXER) += raw.o
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OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtp_mpv.o rtp_aac.o
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OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtpenc.o rtp_mpv.o rtp_aac.o
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OBJS-$(CONFIG_RTSP_DEMUXER) += rtsp.o
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OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o rtp.o rtpdec.o rtp_h264.o rtp_mpv.o rtp_aac.o
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OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o rtp.o rtpdec.o rtp_h264.o
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OBJS-$(CONFIG_SEGAFILM_DEMUXER) += segafilm.o
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OBJS-$(CONFIG_SHORTEN_DEMUXER) += raw.o
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OBJS-$(CONFIG_SIFF_DEMUXER) += siff.o
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@ -19,20 +19,15 @@
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avformat.h"
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#include "mpegts.h"
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#include "bitstream.h"
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#include <unistd.h>
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#include "network.h"
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#include "rtp_internal.h"
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#include "rtp_mpv.h"
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#include "rtp_aac.h"
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//#define DEBUG
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#define RTCP_SR_SIZE 28
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/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
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AVRtpPayloadType_t AVRtpPayloadTypes[]=
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{
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@ -225,326 +220,3 @@ enum CodecID ff_rtp_codec_id(const char *buf, enum CodecType codec_type)
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return CODEC_ID_NONE;
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}
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/* rtp output */
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static int rtp_write_header(AVFormatContext *s1)
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{
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RTPDemuxContext *s = s1->priv_data;
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int payload_type, max_packet_size, n;
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AVStream *st;
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if (s1->nb_streams != 1)
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return -1;
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st = s1->streams[0];
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payload_type = rtp_get_payload_type(st->codec);
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if (payload_type < 0)
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payload_type = RTP_PT_PRIVATE; /* private payload type */
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s->payload_type = payload_type;
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// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
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s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
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s->timestamp = s->base_timestamp;
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s->cur_timestamp = 0;
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s->ssrc = 0; /* FIXME: was random(), what should this be? */
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s->first_packet = 1;
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s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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max_packet_size = url_fget_max_packet_size(s1->pb);
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if (max_packet_size <= 12)
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return AVERROR(EIO);
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s->max_payload_size = max_packet_size - 12;
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s->max_frames_per_packet = 0;
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if (s1->max_delay) {
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if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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if (st->codec->frame_size == 0) {
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av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
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} else {
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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);
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}
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}
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if (st->codec->codec_type == CODEC_TYPE_VIDEO) {
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/* FIXME: We should round down here... */
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s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
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}
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}
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av_set_pts_info(st, 32, 1, 90000);
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switch(st->codec->codec_id) {
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case CODEC_ID_MP2:
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case CODEC_ID_MP3:
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s->buf_ptr = s->buf + 4;
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break;
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case CODEC_ID_MPEG1VIDEO:
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case CODEC_ID_MPEG2VIDEO:
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break;
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case CODEC_ID_MPEG2TS:
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n = s->max_payload_size / TS_PACKET_SIZE;
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if (n < 1)
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n = 1;
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s->max_payload_size = n * TS_PACKET_SIZE;
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s->buf_ptr = s->buf;
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break;
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case CODEC_ID_AAC:
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s->read_buf_index = 0;
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default:
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if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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av_set_pts_info(st, 32, 1, st->codec->sample_rate);
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}
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s->buf_ptr = s->buf;
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break;
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}
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return 0;
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}
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/* send an rtcp sender report packet */
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static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
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{
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RTPDemuxContext *s = s1->priv_data;
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uint32_t rtp_ts;
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#if defined(DEBUG)
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printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
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#endif
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if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time;
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s->last_rtcp_ntp_time = ntp_time;
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rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q,
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s1->streams[0]->time_base) + s->base_timestamp;
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put_byte(s1->pb, (RTP_VERSION << 6));
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put_byte(s1->pb, 200);
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put_be16(s1->pb, 6); /* length in words - 1 */
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put_be32(s1->pb, s->ssrc);
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put_be32(s1->pb, ntp_time / 1000000);
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put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000);
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put_be32(s1->pb, rtp_ts);
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put_be32(s1->pb, s->packet_count);
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put_be32(s1->pb, s->octet_count);
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put_flush_packet(s1->pb);
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}
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/* send an rtp packet. sequence number is incremented, but the caller
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must update the timestamp itself */
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void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
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{
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RTPDemuxContext *s = s1->priv_data;
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#ifdef DEBUG
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printf("rtp_send_data size=%d\n", len);
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#endif
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/* build the RTP header */
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put_byte(s1->pb, (RTP_VERSION << 6));
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put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
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put_be16(s1->pb, s->seq);
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put_be32(s1->pb, s->timestamp);
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put_be32(s1->pb, s->ssrc);
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put_buffer(s1->pb, buf1, len);
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put_flush_packet(s1->pb);
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s->seq++;
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s->octet_count += len;
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s->packet_count++;
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}
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/* send an integer number of samples and compute time stamp and fill
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the rtp send buffer before sending. */
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static void rtp_send_samples(AVFormatContext *s1,
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const uint8_t *buf1, int size, int sample_size)
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{
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RTPDemuxContext *s = s1->priv_data;
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int len, max_packet_size, n;
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max_packet_size = (s->max_payload_size / sample_size) * sample_size;
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/* not needed, but who nows */
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if ((size % sample_size) != 0)
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av_abort();
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n = 0;
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while (size > 0) {
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s->buf_ptr = s->buf;
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len = FFMIN(max_packet_size, size);
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/* copy data */
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memcpy(s->buf_ptr, buf1, len);
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s->buf_ptr += len;
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buf1 += len;
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size -= len;
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s->timestamp = s->cur_timestamp + n / sample_size;
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ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
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n += (s->buf_ptr - s->buf);
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}
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}
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/* NOTE: we suppose that exactly one frame is given as argument here */
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/* XXX: test it */
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static void rtp_send_mpegaudio(AVFormatContext *s1,
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const uint8_t *buf1, int size)
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{
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RTPDemuxContext *s = s1->priv_data;
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int len, count, max_packet_size;
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max_packet_size = s->max_payload_size;
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/* test if we must flush because not enough space */
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len = (s->buf_ptr - s->buf);
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if ((len + size) > max_packet_size) {
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if (len > 4) {
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ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
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s->buf_ptr = s->buf + 4;
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}
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}
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if (s->buf_ptr == s->buf + 4) {
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s->timestamp = s->cur_timestamp;
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}
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/* add the packet */
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if (size > max_packet_size) {
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/* big packet: fragment */
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count = 0;
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while (size > 0) {
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len = max_packet_size - 4;
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if (len > size)
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len = size;
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/* build fragmented packet */
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s->buf[0] = 0;
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s->buf[1] = 0;
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s->buf[2] = count >> 8;
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s->buf[3] = count;
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memcpy(s->buf + 4, buf1, len);
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ff_rtp_send_data(s1, s->buf, len + 4, 0);
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size -= len;
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buf1 += len;
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count += len;
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}
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} else {
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if (s->buf_ptr == s->buf + 4) {
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/* no fragmentation possible */
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s->buf[0] = 0;
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s->buf[1] = 0;
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s->buf[2] = 0;
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s->buf[3] = 0;
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}
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memcpy(s->buf_ptr, buf1, size);
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s->buf_ptr += size;
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}
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}
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static void rtp_send_raw(AVFormatContext *s1,
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const uint8_t *buf1, int size)
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{
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RTPDemuxContext *s = s1->priv_data;
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int len, max_packet_size;
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max_packet_size = s->max_payload_size;
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while (size > 0) {
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len = max_packet_size;
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if (len > size)
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len = size;
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s->timestamp = s->cur_timestamp;
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ff_rtp_send_data(s1, buf1, len, (len == size));
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buf1 += len;
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size -= len;
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}
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}
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/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
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static void rtp_send_mpegts_raw(AVFormatContext *s1,
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const uint8_t *buf1, int size)
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{
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RTPDemuxContext *s = s1->priv_data;
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int len, out_len;
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while (size >= TS_PACKET_SIZE) {
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len = s->max_payload_size - (s->buf_ptr - s->buf);
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if (len > size)
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len = size;
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memcpy(s->buf_ptr, buf1, len);
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buf1 += len;
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size -= len;
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s->buf_ptr += len;
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out_len = s->buf_ptr - s->buf;
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if (out_len >= s->max_payload_size) {
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ff_rtp_send_data(s1, s->buf, out_len, 0);
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s->buf_ptr = s->buf;
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}
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}
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}
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/* write an RTP packet. 'buf1' must contain a single specific frame. */
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static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
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{
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RTPDemuxContext *s = s1->priv_data;
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AVStream *st = s1->streams[0];
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int rtcp_bytes;
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int size= pkt->size;
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uint8_t *buf1= pkt->data;
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#ifdef DEBUG
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printf("%d: write len=%d\n", pkt->stream_index, size);
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#endif
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/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
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rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
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RTCP_TX_RATIO_DEN;
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if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) &&
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(av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
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rtcp_send_sr(s1, av_gettime());
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s->last_octet_count = s->octet_count;
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s->first_packet = 0;
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}
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s->cur_timestamp = s->base_timestamp + pkt->pts;
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switch(st->codec->codec_id) {
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case CODEC_ID_PCM_MULAW:
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case CODEC_ID_PCM_ALAW:
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case CODEC_ID_PCM_U8:
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case CODEC_ID_PCM_S8:
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rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
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break;
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case CODEC_ID_PCM_U16BE:
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case CODEC_ID_PCM_U16LE:
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case CODEC_ID_PCM_S16BE:
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case CODEC_ID_PCM_S16LE:
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rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
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break;
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case CODEC_ID_MP2:
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case CODEC_ID_MP3:
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rtp_send_mpegaudio(s1, buf1, size);
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break;
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case CODEC_ID_MPEG1VIDEO:
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case CODEC_ID_MPEG2VIDEO:
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ff_rtp_send_mpegvideo(s1, buf1, size);
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break;
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case CODEC_ID_AAC:
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ff_rtp_send_aac(s1, buf1, size);
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break;
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case CODEC_ID_MPEG2TS:
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rtp_send_mpegts_raw(s1, buf1, size);
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break;
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default:
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/* better than nothing : send the codec raw data */
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rtp_send_raw(s1, buf1, size);
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break;
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}
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return 0;
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}
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AVOutputFormat rtp_muxer = {
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"rtp",
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"RTP output format",
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NULL,
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NULL,
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sizeof(RTPDemuxContext),
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CODEC_ID_PCM_MULAW,
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CODEC_ID_NONE,
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rtp_write_header,
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rtp_write_packet,
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};
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355
libavformat/rtpenc.c
Normal file
355
libavformat/rtpenc.c
Normal file
@ -0,0 +1,355 @@
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/*
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* RTP output format
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* Copyright (c) 2002 Fabrice Bellard.
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
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* Lesser General Public License for more details.
|
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*
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* You should have received a copy of the GNU Lesser General Public
|
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* License along with FFmpeg; if not, write to the Free Software
|
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avformat.h"
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#include "mpegts.h"
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#include "bitstream.h"
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#include <unistd.h>
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#include "network.h"
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#include "rtp_internal.h"
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#include "rtp_mpv.h"
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#include "rtp_aac.h"
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//#define DEBUG
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#define RTCP_SR_SIZE 28
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static int rtp_write_header(AVFormatContext *s1)
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{
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RTPDemuxContext *s = s1->priv_data;
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int payload_type, max_packet_size, n;
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AVStream *st;
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if (s1->nb_streams != 1)
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return -1;
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st = s1->streams[0];
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payload_type = rtp_get_payload_type(st->codec);
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if (payload_type < 0)
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payload_type = RTP_PT_PRIVATE; /* private payload type */
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s->payload_type = payload_type;
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// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately
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s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
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s->timestamp = s->base_timestamp;
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s->cur_timestamp = 0;
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s->ssrc = 0; /* FIXME: was random(), what should this be? */
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s->first_packet = 1;
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s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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max_packet_size = url_fget_max_packet_size(s1->pb);
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if (max_packet_size <= 12)
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return AVERROR(EIO);
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s->max_payload_size = max_packet_size - 12;
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s->max_frames_per_packet = 0;
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if (s1->max_delay) {
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if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
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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) {
|
||||
/* FIXME: We should round down here... */
|
||||
s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base);
|
||||
}
|
||||
}
|
||||
|
||||
av_set_pts_info(st, 32, 1, 90000);
|
||||
switch(st->codec->codec_id) {
|
||||
case CODEC_ID_MP2:
|
||||
case CODEC_ID_MP3:
|
||||
s->buf_ptr = s->buf + 4;
|
||||
break;
|
||||
case CODEC_ID_MPEG1VIDEO:
|
||||
case CODEC_ID_MPEG2VIDEO:
|
||||
break;
|
||||
case 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_AAC:
|
||||
s->read_buf_index = 0;
|
||||
default:
|
||||
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
|
||||
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
|
||||
}
|
||||
s->buf_ptr = s->buf;
|
||||
break;
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* send an rtcp sender report packet */
|
||||
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
|
||||
{
|
||||
RTPDemuxContext *s = s1->priv_data;
|
||||
uint32_t rtp_ts;
|
||||
|
||||
#if defined(DEBUG)
|
||||
printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
|
||||
#endif
|
||||
|
||||
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, AV_TIME_BASE_Q,
|
||||
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);
|
||||
}
|
||||
|
||||
/* send an rtp packet. sequence number is incremented, but the caller
|
||||
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;
|
||||
|
||||
#ifdef DEBUG
|
||||
printf("rtp_send_data size=%d\n", len);
|
||||
#endif
|
||||
|
||||
/* 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);
|
||||
|
||||
put_buffer(s1->pb, buf1, len);
|
||||
put_flush_packet(s1->pb);
|
||||
|
||||
s->seq++;
|
||||
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)
|
||||
{
|
||||
RTPDemuxContext *s = s1->priv_data;
|
||||
int len, max_packet_size, n;
|
||||
|
||||
max_packet_size = (s->max_payload_size / sample_size) * sample_size;
|
||||
/* not needed, but who nows */
|
||||
if ((size % sample_size) != 0)
|
||||
av_abort();
|
||||
n = 0;
|
||||
while (size > 0) {
|
||||
s->buf_ptr = s->buf;
|
||||
len = FFMIN(max_packet_size, size);
|
||||
|
||||
/* copy data */
|
||||
memcpy(s->buf_ptr, buf1, len);
|
||||
s->buf_ptr += len;
|
||||
buf1 += len;
|
||||
size -= len;
|
||||
s->timestamp = s->cur_timestamp + n / sample_size;
|
||||
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
|
||||
n += (s->buf_ptr - s->buf);
|
||||
}
|
||||
}
|
||||
|
||||
/* 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;
|
||||
int len, count, max_packet_size;
|
||||
|
||||
max_packet_size = s->max_payload_size;
|
||||
|
||||
/* test if we must flush because not enough space */
|
||||
len = (s->buf_ptr - s->buf);
|
||||
if ((len + size) > max_packet_size) {
|
||||
if (len > 4) {
|
||||
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
|
||||
s->buf_ptr = s->buf + 4;
|
||||
}
|
||||
}
|
||||
if (s->buf_ptr == s->buf + 4) {
|
||||
s->timestamp = s->cur_timestamp;
|
||||
}
|
||||
|
||||
/* add the packet */
|
||||
if (size > max_packet_size) {
|
||||
/* big packet: fragment */
|
||||
count = 0;
|
||||
while (size > 0) {
|
||||
len = max_packet_size - 4;
|
||||
if (len > size)
|
||||
len = size;
|
||||
/* build fragmented packet */
|
||||
s->buf[0] = 0;
|
||||
s->buf[1] = 0;
|
||||
s->buf[2] = count >> 8;
|
||||
s->buf[3] = count;
|
||||
memcpy(s->buf + 4, buf1, len);
|
||||
ff_rtp_send_data(s1, s->buf, len + 4, 0);
|
||||
size -= len;
|
||||
buf1 += len;
|
||||
count += len;
|
||||
}
|
||||
} else {
|
||||
if (s->buf_ptr == s->buf + 4) {
|
||||
/* no fragmentation possible */
|
||||
s->buf[0] = 0;
|
||||
s->buf[1] = 0;
|
||||
s->buf[2] = 0;
|
||||
s->buf[3] = 0;
|
||||
}
|
||||
memcpy(s->buf_ptr, buf1, size);
|
||||
s->buf_ptr += size;
|
||||
}
|
||||
}
|
||||
|
||||
static void rtp_send_raw(AVFormatContext *s1,
|
||||
const uint8_t *buf1, int size)
|
||||
{
|
||||
RTPDemuxContext *s = s1->priv_data;
|
||||
int len, max_packet_size;
|
||||
|
||||
max_packet_size = s->max_payload_size;
|
||||
|
||||
while (size > 0) {
|
||||
len = max_packet_size;
|
||||
if (len > size)
|
||||
len = size;
|
||||
|
||||
s->timestamp = s->cur_timestamp;
|
||||
ff_rtp_send_data(s1, buf1, len, (len == size));
|
||||
|
||||
buf1 += len;
|
||||
size -= len;
|
||||
}
|
||||
}
|
||||
|
||||
/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
|
||||
static void rtp_send_mpegts_raw(AVFormatContext *s1,
|
||||
const uint8_t *buf1, int size)
|
||||
{
|
||||
RTPDemuxContext *s = s1->priv_data;
|
||||
int len, out_len;
|
||||
|
||||
while (size >= TS_PACKET_SIZE) {
|
||||
len = s->max_payload_size - (s->buf_ptr - s->buf);
|
||||
if (len > size)
|
||||
len = size;
|
||||
memcpy(s->buf_ptr, buf1, len);
|
||||
buf1 += len;
|
||||
size -= len;
|
||||
s->buf_ptr += len;
|
||||
|
||||
out_len = s->buf_ptr - s->buf;
|
||||
if (out_len >= s->max_payload_size) {
|
||||
ff_rtp_send_data(s1, s->buf, out_len, 0);
|
||||
s->buf_ptr = s->buf;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/* 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;
|
||||
AVStream *st = s1->streams[0];
|
||||
int rtcp_bytes;
|
||||
int size= pkt->size;
|
||||
uint8_t *buf1= pkt->data;
|
||||
|
||||
#ifdef DEBUG
|
||||
printf("%d: write len=%d\n", pkt->stream_index, size);
|
||||
#endif
|
||||
|
||||
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
|
||||
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) &&
|
||||
(av_gettime() - s->last_rtcp_ntp_time > 5000000))) {
|
||||
rtcp_send_sr(s1, av_gettime());
|
||||
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, buf1, 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);
|
||||
break;
|
||||
case CODEC_ID_MP2:
|
||||
case CODEC_ID_MP3:
|
||||
rtp_send_mpegaudio(s1, buf1, size);
|
||||
break;
|
||||
case CODEC_ID_MPEG1VIDEO:
|
||||
case CODEC_ID_MPEG2VIDEO:
|
||||
ff_rtp_send_mpegvideo(s1, buf1, size);
|
||||
break;
|
||||
case CODEC_ID_AAC:
|
||||
ff_rtp_send_aac(s1, buf1, size);
|
||||
break;
|
||||
case CODEC_ID_MPEG2TS:
|
||||
rtp_send_mpegts_raw(s1, buf1, size);
|
||||
break;
|
||||
default:
|
||||
/* better than nothing : send the codec raw data */
|
||||
rtp_send_raw(s1, buf1, size);
|
||||
break;
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
AVOutputFormat rtp_muxer = {
|
||||
"rtp",
|
||||
"RTP output format",
|
||||
NULL,
|
||||
NULL,
|
||||
sizeof(RTPDemuxContext),
|
||||
CODEC_ID_PCM_MULAW,
|
||||
CODEC_ID_NONE,
|
||||
rtp_write_header,
|
||||
rtp_write_packet,
|
||||
};
|
||||
Loading…
Reference in New Issue
Block a user