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Remove unnecessary calls to BaseChannel::SetRtpTransport

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Also updating SocketOptionsMergedOnSetTransport test code to make the
call to SetRtpTransport from the right context.

Bug: webrtc:12636
Change-Id: I343851bcf8ac663d7559128d12447a9a742786f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33633}
This commit is contained in:
Tomas Gunnarsson 2021-04-02 17:42:02 +02:00 committed by Commit Bot
parent fe041643b4
commit d9a51b05da
4 changed files with 1 additions and 17 deletions
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9
pc/channel.cc
View File

@ -228,11 +228,6 @@ void BaseChannel::Deinit() {
}
bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
if (!network_thread_->IsCurrent()) {
return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this, rtp_transport] {
return SetRtpTransport(rtp_transport);
});
}
RTC_DCHECK_RUN_ON(network_thread());
if (rtp_transport == rtp_transport_) {
return true;
@ -881,10 +876,6 @@ VoiceChannel::~VoiceChannel() {
Deinit();
}
void VoiceChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
BaseChannel::Init_w(rtp_transport);
}
void VoiceChannel::UpdateMediaSendRecvState_w() {
// Render incoming data if we're the active call, and we have the local
// content. We receive data on the default channel and multiplexed streams.

1
pc/channel.h
View File

@ -434,7 +434,6 @@ class VoiceChannel : public BaseChannel {
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
void Init_w(webrtc::RtpTransportInternal* rtp_transport) override;
private:
// overrides from BaseChannel

2
pc/channel_unittest.cc
View File

@ -1276,11 +1276,11 @@ class ChannelTest : public ::testing::Test, public sigslot::has_slots<> {
new_rtp_transport_ = CreateDtlsSrtpTransport(
fake_rtp_dtls_transport2_.get(), fake_rtcp_dtls_transport2_.get());
channel1_->SetRtpTransport(new_rtp_transport_.get());
bool rcv_success, send_success;
int rcv_buf, send_buf;
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
channel1_->SetRtpTransport(new_rtp_transport_.get());
send_success = fake_rtp_dtls_transport2_->GetOption(
rtc::Socket::Option::OPT_SNDBUF, &send_buf);
rcv_success = fake_rtp_dtls_transport2_->GetOption(

6
pc/sdp_offer_answer.cc
View File

@ -4629,8 +4629,6 @@ cricket::VoiceChannel* SdpOfferAnswerHandler::CreateVoiceChannel(
}
voice_channel->SignalSentPacket().connect(pc_,
&PeerConnection::OnSentPacket_w);
voice_channel->SetRtpTransport(rtp_transport);
return voice_channel;
}
@ -4654,8 +4652,6 @@ cricket::VideoChannel* SdpOfferAnswerHandler::CreateVideoChannel(
}
video_channel->SignalSentPacket().connect(pc_,
&PeerConnection::OnSentPacket_w);
video_channel->SetRtpTransport(rtp_transport);
return video_channel;
}
@ -4688,8 +4684,6 @@ bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
}
data_channel_controller()->rtp_data_channel()->SignalSentPacket().connect(
pc_, &PeerConnection::OnSentPacket_w);
data_channel_controller()->rtp_data_channel()->SetRtpTransport(
rtp_transport);
have_pending_rtp_data_channel_ = true;
return true;
}