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webrtc是否允许发送rtp逻辑移至基类

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ziyue
2021-10-20 16:58:59 +08:00
parent 93c6754fc4
commit 758f1b414e
3 changed files with 5 additions and 17 deletions
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15
webrtc/WebRtcPlayer.cpp
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@@ -33,12 +33,6 @@ void WebRtcPlayer::onStartWebRTC() {
CHECK(_play_src);
WebRtcTransportImp::onStartWebRTC();
if (canSendRtp()) {
//确保该rtp codec类型对方支持
memset(_can_send_rtp, 0, sizeof(_can_send_rtp));
for (auto &m : _answer_sdp->media) {
_can_send_rtp[m.type] = m.direction == RtpDirection::sendonly || m.direction == RtpDirection::sendrecv;
}
_play_src->pause(false);
_reader = _play_src->getRing()->attach(getPoller(), true);
weak_ptr<WebRtcPlayer> weak_self = static_pointer_cast<WebRtcPlayer>(shared_from_this());
@@ -49,7 +43,7 @@ void WebRtcPlayer::onStartWebRTC() {
}
size_t i = 0;
pkt->for_each([&](const RtpPacket::Ptr &rtp) {
strongSelf->beforeSendRtp(rtp, ++i == pkt->size());
strongSelf->onSendRtp(rtp, ++i == pkt->size());
});
});
_reader->setDetachCB([weak_self]() {
@@ -91,10 +85,3 @@ void WebRtcPlayer::onRtcConfigure(RtcConfigure &configure) const {
configure.audio.direction = configure.video.direction = RtpDirection::sendonly;
configure.setPlayRtspInfo(_play_src->getSdp());
}
void WebRtcPlayer::beforeSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx) {
if (!_can_send_rtp[rtp->type]) {
return;
}
onSendRtp(rtp, flush, rtx);
}