removed sps/pps from sdp

This commit is contained in:
Maxim Devaev
2022-11-03 22:54:06 +03:00
parent c8600e62c2
commit 22e5c8627b
5 changed files with 3 additions and 54 deletions

View File

@@ -429,10 +429,6 @@ static struct janus_plugin_result *_plugin_handle_message(
{
// atomic_store(&_g_key_required, true);
char *const video_sdp = us_rtpv_make_sdp(_g_rtpv);
if (video_sdp == NULL) {
PUSH_ERROR(503, "Haven't received SPS/PPS from memsink yet");
goto ok_wait;
}
char *const audio_sdp = (_g_rtpa ? us_rtpa_make_sdp(_g_rtpa) : us_strdup(""));
US_ASPRINTF(sdp,
"v=0" RN

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@@ -36,35 +36,15 @@ us_rtpv_s *us_rtpv_init(us_rtp_callback_f callback, bool zero_playout_delay) {
US_CALLOC(rtpv, 1);
rtpv->rtp = us_rtp_init(96, true, zero_playout_delay);
rtpv->callback = callback;
rtpv->sps = us_frame_init();
rtpv->pps = us_frame_init();
US_MUTEX_INIT(rtpv->mutex);
return rtpv;
}
void us_rtpv_destroy(us_rtpv_s *rtpv) {
US_MUTEX_DESTROY(rtpv->mutex);
us_frame_destroy(rtpv->pps);
us_frame_destroy(rtpv->sps);
us_rtp_destroy(rtpv->rtp);
free(rtpv);
}
char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
US_MUTEX_LOCK(rtpv->mutex);
if (rtpv->sps->used == 0 || rtpv->pps->used == 0) {
US_MUTEX_UNLOCK(rtpv->mutex);
return NULL;
}
char *sps = NULL;
char *pps = NULL;
us_base64_encode(rtpv->sps->data, rtpv->sps->used, &sps, NULL);
us_base64_encode(rtpv->pps->data, rtpv->pps->used, &pps, NULL);
US_MUTEX_UNLOCK(rtpv->mutex);
# define PAYLOAD rtpv->rtp->payload
// https://tools.ietf.org/html/rfc6184
// https://github.com/meetecho/janus-gateway/issues/2443
@@ -75,8 +55,6 @@ char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
"a=rtpmap:%u H264/90000" RN
"a=fmtp:%u profile-level-id=42E01F" RN
"a=fmtp:%u packetization-mode=1" RN
"a=fmtp:%u sprop-sps=%s" RN
"a=fmtp:%u sprop-pps=%s" RN
"a=rtcp-fb:%u nack" RN
"a=rtcp-fb:%u nack pli" RN
"a=rtcp-fb:%u goog-remb" RN
@@ -84,17 +62,12 @@ char *us_rtpv_make_sdp(us_rtpv_s *rtpv) {
"%s" // playout-delay
"a=sendonly" RN,
PAYLOAD, PAYLOAD, PAYLOAD, PAYLOAD,
PAYLOAD, sps,
PAYLOAD, pps,
PAYLOAD, PAYLOAD, PAYLOAD,
rtpv->rtp->ssrc,
(rtpv->rtp->zero_playout_delay ? "a=extmap:1 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay" RN : "")
);
# undef PAYLOAD
free(sps);
free(pps);
return sdp;
# undef PAYLOAD
}
#define _PRE 3 // Annex B prefix length
@@ -136,22 +109,11 @@ void us_rtpv_wrap(us_rtpv_s *rtpv, const us_frame_s *frame) {
}
void _rtpv_process_nalu(us_rtpv_s *rtpv, const uint8_t *data, size_t size, uint32_t pts, bool marked) {
# define DG rtpv->rtp->datagram
const unsigned ref_idc = (data[0] >> 5) & 3;
const unsigned type = data[0] & 0x1F;
us_frame_s *ps = NULL;
switch (type) {
case 7: ps = rtpv->sps; break;
case 8: ps = rtpv->pps; break;
}
if (ps != NULL) {
US_MUTEX_LOCK(rtpv->mutex);
us_frame_set_data(ps, data, size);
US_MUTEX_UNLOCK(rtpv->mutex);
}
# define DG rtpv->rtp->datagram
if (size + US_RTP_HEADER_SIZE <= US_RTP_DATAGRAM_SIZE) {
us_rtp_write_header(rtpv->rtp, pts, marked);
memcpy(DG + US_RTP_HEADER_SIZE, data, size);

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@@ -31,12 +31,8 @@
#include <sys/types.h>
#include <linux/videodev2.h>
#include <pthread.h>
#include "uslibs/tools.h"
#include "uslibs/threading.h"
#include "uslibs/frame.h"
#include "uslibs/base64.h"
#include "rtp.h"
@@ -44,9 +40,6 @@
typedef struct {
us_rtp_s *rtp;
us_rtp_callback_f callback;
us_frame_s *sps; // Actually not a frame, just a bytes storage
us_frame_s *pps;
pthread_mutex_t mutex;
} us_rtpv_s;

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@@ -1 +0,0 @@
../../../src/libs/base64.c

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@@ -1 +0,0 @@
../../../src/libs/base64.h