webrtc audio

This commit is contained in:
Maxim Devaev
2022-05-29 01:29:01 +03:00
parent 3d4e9fbb1a
commit d4a9862a18
15 changed files with 1072 additions and 310 deletions

211
janus/src/rtpv.c Normal file
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@@ -0,0 +1,211 @@
/*****************************************************************************
# #
# uStreamer - Lightweight and fast MJPEG-HTTP streamer. #
# #
# This source file is partially based on this code: #
# - https://github.com/catid/kvm/blob/master/kvm_pipeline/src #
# #
# Copyright (C) 2018-2022 Maxim Devaev <mdevaev@gmail.com> #
# #
# This program is free software: you can redistribute it and/or modify #
# it under the terms of the GNU General Public License as published by #
# the Free Software Foundation, either version 3 of the License, or #
# (at your option) any later version. #
# #
# This program is distributed in the hope that it will be useful, #
# but WITHOUT ANY WARRANTY; without even the implied warranty of #
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the #
# GNU General Public License for more details. #
# #
# You should have received a copy of the GNU General Public License #
# along with this program. If not, see <https://www.gnu.org/licenses/>. #
# #
*****************************************************************************/
#include "rtpv.h"
void _rtpv_process_nalu(rtpv_s *rtpv, const uint8_t *data, size_t size, uint32_t pts, bool marked);
static ssize_t _find_annexb(const uint8_t *data, size_t size);
rtpv_s *rtpv_init(rtp_callback_f callback) {
rtpv_s *rtpv;
A_CALLOC(rtpv, 1);
rtpv->rtp = rtp_init(96, true);
rtpv->callback = callback;
rtpv->sps = frame_init();
rtpv->pps = frame_init();
A_MUTEX_INIT(&rtpv->mutex);
return rtpv;
}
void rtpv_destroy(rtpv_s *rtpv) {
A_MUTEX_DESTROY(&rtpv->mutex);
frame_destroy(rtpv->pps);
frame_destroy(rtpv->sps);
rtp_destroy(rtpv->rtp);
free(rtpv);
}
char *rtpv_make_sdp(rtpv_s *rtpv) {
A_MUTEX_LOCK(&rtpv->mutex);
if (rtpv->sps->used == 0 || rtpv->pps->used == 0) {
A_MUTEX_UNLOCK(&rtpv->mutex);
return NULL;
}
char *sps = NULL;
char *pps = NULL;
base64_encode(rtpv->sps->data, rtpv->sps->used, &sps, NULL);
base64_encode(rtpv->pps->data, rtpv->pps->used, &pps, NULL);
A_MUTEX_UNLOCK(&rtpv->mutex);
# define PAYLOAD rtpv->rtp->payload
// https://tools.ietf.org/html/rfc6184
// https://github.com/meetecho/janus-gateway/issues/2443
char *sdp;
A_ASPRINTF(sdp,
"m=video 1 RTP/SAVPF %u" RN
"c=IN IP4 0.0.0.0" RN
"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
"a=ssrc:%" PRIu32 " cname:ustreamer" RN
"a=sendonly" RN,
PAYLOAD, PAYLOAD, PAYLOAD, PAYLOAD,
PAYLOAD, sps,
PAYLOAD, pps,
PAYLOAD, PAYLOAD, PAYLOAD,
rtpv->rtp->ssrc
);
# undef PAYLOAD
free(sps);
free(pps);
return sdp;
}
#define PRE 3 // Annex B prefix length
void rtpv_wrap(rtpv_s *rtpv, const frame_s *frame) {
// There is a complicated logic here but everything works as it should:
// - https://github.com/pikvm/ustreamer/issues/115#issuecomment-893071775
assert(frame->format == V4L2_PIX_FMT_H264);
const uint32_t pts = get_now_monotonic_u64() * 9 / 100; // PTS units are in 90 kHz
ssize_t last_offset = -PRE;
while (true) { // Find and iterate by nalus
const size_t next_start = last_offset + PRE;
ssize_t offset = _find_annexb(frame->data + next_start, frame->used - next_start);
if (offset < 0) {
break;
}
offset += next_start;
if (last_offset >= 0) {
const uint8_t *data = frame->data + last_offset + PRE;
size_t size = offset - last_offset - PRE;
if (data[size - 1] == 0) { // Check for extra 00
--size;
}
_rtpv_process_nalu(rtpv, data, size, pts, false);
}
last_offset = offset;
}
if (last_offset >= 0) {
const uint8_t *data = frame->data + last_offset + PRE;
size_t size = frame->used - last_offset - PRE;
_rtpv_process_nalu(rtpv, data, size, pts, true);
}
}
void _rtpv_process_nalu(rtpv_s *rtpv, const uint8_t *data, size_t size, uint32_t pts, bool marked) {
const unsigned ref_idc = (data[0] >> 5) & 3;
const unsigned type = data[0] & 0x1F;
frame_s *ps = NULL;
switch (type) {
case 7: ps = rtpv->sps; break;
case 8: ps = rtpv->pps; break;
}
if (ps) {
A_MUTEX_LOCK(&rtpv->mutex);
frame_set_data(ps, data, size);
A_MUTEX_UNLOCK(&rtpv->mutex);
}
# define DG rtpv->rtp->datagram
if (size + RTP_HEADER_SIZE <= RTP_DATAGRAM_SIZE) {
rtp_write_header(rtpv->rtp, pts, marked);
memcpy(DG + RTP_HEADER_SIZE, data, size);
rtpv->rtp->used = size + RTP_HEADER_SIZE;
rtpv->callback(rtpv->rtp);
return;
}
const size_t fu_overhead = RTP_HEADER_SIZE + 2; // FU-A overhead
const uint8_t *src = data + 1;
ssize_t remaining = size - 1;
bool first = true;
while (remaining > 0) {
ssize_t frag_size = RTP_DATAGRAM_SIZE - fu_overhead;
const bool last = (remaining <= frag_size);
if (last) {
frag_size = remaining;
}
rtp_write_header(rtpv->rtp, pts, (marked && last));
DG[RTP_HEADER_SIZE] = 28 | (ref_idc << 5);
uint8_t fu = type;
if (first) {
fu |= 0x80;
}
if (last) {
fu |= 0x40;
}
DG[RTP_HEADER_SIZE + 1] = fu;
memcpy(DG + fu_overhead, src, frag_size);
rtpv->rtp->used = fu_overhead + frag_size;
rtpv->callback(rtpv->rtp);
src += frag_size;
remaining -= frag_size;
first = false;
}
# undef DG
}
static ssize_t _find_annexb(const uint8_t *data, size_t size) {
// Parses buffer for 00 00 01 start codes
if (size >= PRE) {
for (size_t index = 0; index <= size - PRE; ++index) {
if (data[index] == 0 && data[index + 1] == 0 && data[index + 2] == 1) {
return index;
}
}
}
return -1;
}
#undef PRE