mirror of https://git.ffmpeg.org/ffmpeg.git
rtp: remove disabled code
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be7bd626c4
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4cc843facd
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@ -218,22 +218,6 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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return 1;
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}
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#if 0
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/**
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* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
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* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
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* never change. I left this in in case someone else can see a way. (rdm)
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*/
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static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
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{
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uint32_t transit= arrival_timestamp - sent_timestamp;
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int d;
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s->transit= transit;
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d= FFABS(transit - s->transit);
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s->jitter += d - ((s->jitter + 8)>>4);
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}
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#endif
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int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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{
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AVIOContext *pb;
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@ -225,20 +225,6 @@ static int rtp_read(URLContext *h, uint8_t *buf, int size)
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int len, n;
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struct pollfd p[2] = {{s->rtp_fd, POLLIN, 0}, {s->rtcp_fd, POLLIN, 0}};
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#if 0
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for(;;) {
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from_len = sizeof(from);
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len = recvfrom (s->rtp_fd, buf, size, 0,
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(struct sockaddr *)&from, &from_len);
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if (len < 0) {
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if (ff_neterrno() == AVERROR(EAGAIN) ||
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ff_neterrno() == AVERROR(EINTR))
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continue;
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return AVERROR(EIO);
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}
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break;
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}
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#else
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for(;;) {
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if (url_interrupt_cb())
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return AVERROR_EXIT;
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@ -277,7 +263,6 @@ static int rtp_read(URLContext *h, uint8_t *buf, int size)
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return AVERROR(EIO);
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}
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}
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#endif
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return len;
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}
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@ -296,14 +281,6 @@ static int rtp_write(URLContext *h, const uint8_t *buf, int size)
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}
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ret = ffurl_write(hd, buf, size);
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#if 0
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{
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struct timespec ts;
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ts.tv_sec = 0;
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ts.tv_nsec = 10 * 1000000;
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nanosleep(&ts, NULL);
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}
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#endif
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return ret;
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}
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