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Code Issues Releases Wiki Activity

Rename functions for RTC publisher

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This commit is contained in:
winlin 2020-04-30 09:33:21 +08:00
parent f37ffdf740
commit 583ae52df8
4 changed files with 33 additions and 36 deletions
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38
trunk/src/app/srs_app_rtc_conn.cpp
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@ -1573,18 +1573,18 @@ srs_error_t SrsRtcPublisher::on_rtp(SrsUdpMuxSocket* skt, char* buf, int nb_buf)
{
srs_error_t err = srs_success;
SrsRtpSharedPacket* rtp_shared_pkt = new SrsRtpSharedPacket();
SrsAutoFree(SrsRtpSharedPacket, rtp_shared_pkt);
if ((err = rtp_shared_pkt->decode(buf, nb_buf)) != srs_success) {
SrsRtpSharedPacket* pkt = new SrsRtpSharedPacket();
SrsAutoFree(SrsRtpSharedPacket, pkt);
if ((err = pkt->decode(buf, nb_buf)) != srs_success) {
return srs_error_wrap(err, "rtp packet decode failed");
}
uint32_t ssrc = rtp_shared_pkt->rtp_header.get_ssrc();
uint32_t ssrc = pkt->rtp_header.get_ssrc();
if (ssrc == audio_ssrc) {
return on_audio(skt, rtp_shared_pkt);
return on_audio(skt, pkt);
} else if (ssrc == video_ssrc) {
return on_video(skt, rtp_shared_pkt);
return on_video(skt, pkt);
}
return srs_error_new(ERROR_RTC_RTP, "unknown ssrc=%u", ssrc);
@ -1933,20 +1933,19 @@ srs_error_t SrsRtcPublisher::send_rtcp_fb_pli(SrsUdpMuxSocket* skt, uint32_t ssr
return err;
}
srs_error_t SrsRtcPublisher::on_audio(SrsUdpMuxSocket* skt, SrsRtpSharedPacket* rtp_pkt)
srs_error_t SrsRtcPublisher::on_audio(SrsUdpMuxSocket* skt, SrsRtpSharedPacket* pkt)
{
srs_error_t err = srs_success;
rtp_pkt->rtp_payload_header = new SrsRtpOpusHeader();
if ((err = rtp_opus_demuxer->parse(rtp_pkt)) != srs_success) {
pkt->rtp_payload_header = new SrsRtpOpusHeader();
if ((err = rtp_opus_demuxer->parse(pkt)) != srs_success) {
return srs_error_wrap(err, "rtp opus demux failed");
}
// TODO: FIXME: Rename it.
// TODO: FIXME: Error check.
rtp_audio_queue->insert(rtp_pkt);
rtp_audio_queue->consume(pkt);
if (rtp_audio_queue->get_and_clean_if_needed_request_key_frame()) {
if (rtp_audio_queue->should_request_key_frame()) {
// TODO: FIXME: Check error.
send_rtcp_fb_pli(skt, audio_ssrc);
}
@ -1961,7 +1960,7 @@ srs_error_t SrsRtcPublisher::collect_audio_frame()
srs_error_t err = srs_success;
std::vector<std::vector<SrsRtpSharedPacket*> > frames;
rtp_audio_queue->get_and_clean_collected_frames(frames);
rtp_audio_queue->collect_frames(frames);
for (size_t i = 0; i < frames.size(); ++i) {
if (!frames[i].empty()) {
@ -1979,19 +1978,20 @@ srs_error_t SrsRtcPublisher::collect_audio_frame()
return err;
}
srs_error_t SrsRtcPublisher::on_video(SrsUdpMuxSocket* skt, SrsRtpSharedPacket* rtp_pkt)
srs_error_t SrsRtcPublisher::on_video(SrsUdpMuxSocket* skt, SrsRtpSharedPacket* pkt)
{
srs_error_t err = srs_success;
rtp_pkt->rtp_payload_header = new SrsRtpH264Header();
pkt->rtp_payload_header = new SrsRtpH264Header();
if ((err = rtp_h264_demuxer->parse(rtp_pkt)) != srs_success) {
if ((err = rtp_h264_demuxer->parse(pkt)) != srs_success) {
return srs_error_wrap(err, "rtp h264 demux failed");
}
rtp_video_queue->insert(rtp_pkt);
// TODO: FIXME: Error check.
rtp_video_queue->consume(pkt);
if (rtp_video_queue->get_and_clean_if_needed_request_key_frame()) {
if (rtp_video_queue->should_request_key_frame()) {
// TODO: FIXME: Check error.
send_rtcp_fb_pli(skt, video_ssrc);
}
@ -2006,7 +2006,7 @@ srs_error_t SrsRtcPublisher::collect_video_frame()
srs_error_t err = srs_success;
std::vector<std::vector<SrsRtpSharedPacket*> > frames;
rtp_video_queue->get_and_clean_collected_frames(frames);
rtp_video_queue->collect_frames(frames);
for (size_t i = 0; i < frames.size(); ++i) {
if (!frames[i].empty()) {