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webrtc audio
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211
janus/src/rtpv.c
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211
janus/src/rtpv.c
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/*****************************************************************************
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# #
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# uStreamer - Lightweight and fast MJPEG-HTTP streamer. #
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# #
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# This source file is partially based on this code: #
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# - https://github.com/catid/kvm/blob/master/kvm_pipeline/src #
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# #
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# Copyright (C) 2018-2022 Maxim Devaev <mdevaev@gmail.com> #
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# #
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# This program is free software: you can redistribute it and/or modify #
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# it under the terms of the GNU General Public License as published by #
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# the Free Software Foundation, either version 3 of the License, or #
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# (at your option) any later version. #
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# #
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# This program 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 #
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# GNU General Public License for more details. #
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# #
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# You should have received a copy of the GNU General Public License #
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# along with this program. If not, see <https://www.gnu.org/licenses/>. #
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# #
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*****************************************************************************/
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#include "rtpv.h"
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void _rtpv_process_nalu(rtpv_s *rtpv, const uint8_t *data, size_t size, uint32_t pts, bool marked);
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static ssize_t _find_annexb(const uint8_t *data, size_t size);
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rtpv_s *rtpv_init(rtp_callback_f callback) {
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rtpv_s *rtpv;
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A_CALLOC(rtpv, 1);
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rtpv->rtp = rtp_init(96, true);
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rtpv->callback = callback;
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rtpv->sps = frame_init();
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rtpv->pps = frame_init();
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A_MUTEX_INIT(&rtpv->mutex);
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return rtpv;
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}
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void rtpv_destroy(rtpv_s *rtpv) {
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A_MUTEX_DESTROY(&rtpv->mutex);
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frame_destroy(rtpv->pps);
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frame_destroy(rtpv->sps);
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rtp_destroy(rtpv->rtp);
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free(rtpv);
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}
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char *rtpv_make_sdp(rtpv_s *rtpv) {
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A_MUTEX_LOCK(&rtpv->mutex);
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if (rtpv->sps->used == 0 || rtpv->pps->used == 0) {
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A_MUTEX_UNLOCK(&rtpv->mutex);
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return NULL;
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}
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char *sps = NULL;
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char *pps = NULL;
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base64_encode(rtpv->sps->data, rtpv->sps->used, &sps, NULL);
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base64_encode(rtpv->pps->data, rtpv->pps->used, &pps, NULL);
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A_MUTEX_UNLOCK(&rtpv->mutex);
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# define PAYLOAD rtpv->rtp->payload
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// https://tools.ietf.org/html/rfc6184
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// https://github.com/meetecho/janus-gateway/issues/2443
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char *sdp;
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A_ASPRINTF(sdp,
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"m=video 1 RTP/SAVPF %u" RN
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"c=IN IP4 0.0.0.0" RN
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"a=rtpmap:%u H264/90000" RN
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"a=fmtp:%u profile-level-id=42E01F" RN
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"a=fmtp:%u packetization-mode=1" RN
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"a=fmtp:%u sprop-sps=%s" RN
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"a=fmtp:%u sprop-pps=%s" RN
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"a=rtcp-fb:%u nack" RN
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"a=rtcp-fb:%u nack pli" RN
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"a=rtcp-fb:%u goog-remb" RN
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"a=ssrc:%" PRIu32 " cname:ustreamer" RN
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"a=sendonly" RN,
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PAYLOAD, PAYLOAD, PAYLOAD, PAYLOAD,
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PAYLOAD, sps,
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PAYLOAD, pps,
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PAYLOAD, PAYLOAD, PAYLOAD,
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rtpv->rtp->ssrc
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);
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# undef PAYLOAD
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free(sps);
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free(pps);
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return sdp;
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}
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#define PRE 3 // Annex B prefix length
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void rtpv_wrap(rtpv_s *rtpv, const frame_s *frame) {
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// There is a complicated logic here but everything works as it should:
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// - https://github.com/pikvm/ustreamer/issues/115#issuecomment-893071775
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assert(frame->format == V4L2_PIX_FMT_H264);
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const uint32_t pts = get_now_monotonic_u64() * 9 / 100; // PTS units are in 90 kHz
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ssize_t last_offset = -PRE;
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while (true) { // Find and iterate by nalus
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const size_t next_start = last_offset + PRE;
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ssize_t offset = _find_annexb(frame->data + next_start, frame->used - next_start);
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if (offset < 0) {
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break;
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}
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offset += next_start;
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if (last_offset >= 0) {
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const uint8_t *data = frame->data + last_offset + PRE;
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size_t size = offset - last_offset - PRE;
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if (data[size - 1] == 0) { // Check for extra 00
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--size;
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}
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_rtpv_process_nalu(rtpv, data, size, pts, false);
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}
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last_offset = offset;
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}
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if (last_offset >= 0) {
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const uint8_t *data = frame->data + last_offset + PRE;
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size_t size = frame->used - last_offset - PRE;
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_rtpv_process_nalu(rtpv, data, size, pts, true);
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}
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}
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void _rtpv_process_nalu(rtpv_s *rtpv, const uint8_t *data, size_t size, uint32_t pts, bool marked) {
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const unsigned ref_idc = (data[0] >> 5) & 3;
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const unsigned type = data[0] & 0x1F;
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frame_s *ps = NULL;
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switch (type) {
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case 7: ps = rtpv->sps; break;
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case 8: ps = rtpv->pps; break;
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}
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if (ps) {
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A_MUTEX_LOCK(&rtpv->mutex);
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frame_set_data(ps, data, size);
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A_MUTEX_UNLOCK(&rtpv->mutex);
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}
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# define DG rtpv->rtp->datagram
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if (size + RTP_HEADER_SIZE <= RTP_DATAGRAM_SIZE) {
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rtp_write_header(rtpv->rtp, pts, marked);
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memcpy(DG + RTP_HEADER_SIZE, data, size);
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rtpv->rtp->used = size + RTP_HEADER_SIZE;
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rtpv->callback(rtpv->rtp);
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return;
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}
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const size_t fu_overhead = RTP_HEADER_SIZE + 2; // FU-A overhead
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const uint8_t *src = data + 1;
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ssize_t remaining = size - 1;
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bool first = true;
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while (remaining > 0) {
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ssize_t frag_size = RTP_DATAGRAM_SIZE - fu_overhead;
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const bool last = (remaining <= frag_size);
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if (last) {
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frag_size = remaining;
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}
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rtp_write_header(rtpv->rtp, pts, (marked && last));
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DG[RTP_HEADER_SIZE] = 28 | (ref_idc << 5);
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uint8_t fu = type;
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if (first) {
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fu |= 0x80;
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}
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if (last) {
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fu |= 0x40;
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}
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DG[RTP_HEADER_SIZE + 1] = fu;
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memcpy(DG + fu_overhead, src, frag_size);
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rtpv->rtp->used = fu_overhead + frag_size;
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rtpv->callback(rtpv->rtp);
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src += frag_size;
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remaining -= frag_size;
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first = false;
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}
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# undef DG
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}
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static ssize_t _find_annexb(const uint8_t *data, size_t size) {
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// Parses buffer for 00 00 01 start codes
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if (size >= PRE) {
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for (size_t index = 0; index <= size - PRE; ++index) {
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if (data[index] == 0 && data[index + 1] == 0 && data[index + 2] == 1) {
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return index;
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}
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}
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}
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return -1;
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}
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#undef PRE
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