zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

More minor improvements to BaseChannel/transport code.

Browse Source 
Mostly from late comments on this CL:
https://codereview.webrtc.org/2614263002/

Changes SetTransport to DCHECK instead of returning false.
Renames it to SetTransports.
Fixes some possible transport resource leaks.

BUG=None

Review-Url: https://codereview.webrtc.org/2637503003
Cr-Commit-Position: refs/heads/master@{#16130}
This commit is contained in:
deadbeef
2017-01-17 18:32:35 -08:00
committed by Commit bot
parent b308b037e3
commit bad5dadef3
8 changed files with 179 additions and 82 deletions
Download Patch File Download Diff File Expand all files Collapse all files

38
webrtc/pc/channel.cc
View File

@ -228,11 +228,7 @@ bool BaseChannel::Init_w(TransportChannel* rtp_transport,
bool BaseChannel::InitNetwork_n(TransportChannel* rtp_transport,
TransportChannel* rtcp_transport) {
RTC_DCHECK(network_thread_->IsCurrent());
// const std::string& transport_name =
// (bundle_transport_name ? *bundle_transport_name : content_name());
if (!SetTransport_n(rtp_transport, rtcp_transport)) {
return false;
}
SetTransports_n(rtp_transport, rtcp_transport);
if (!SetDtlsSrtpCryptoSuites_n(rtp_transport_, false)) {
return false;
@ -256,30 +252,27 @@ void BaseChannel::Deinit() {
RTC_FROM_HERE, Bind(&BaseChannel::DisconnectTransportChannels_n, this));
}
bool BaseChannel::SetTransport(TransportChannel* rtp_transport,
TransportChannel* rtcp_transport) {
return network_thread_->Invoke<bool>(
void BaseChannel::SetTransports(TransportChannel* rtp_transport,
TransportChannel* rtcp_transport) {
network_thread_->Invoke<void>(
RTC_FROM_HERE,
Bind(&BaseChannel::SetTransport_n, this, rtp_transport, rtcp_transport));
Bind(&BaseChannel::SetTransports_n, this, rtp_transport, rtcp_transport));
}
bool BaseChannel::SetTransport_n(TransportChannel* rtp_transport,
TransportChannel* rtcp_transport) {
void BaseChannel::SetTransports_n(TransportChannel* rtp_transport,
TransportChannel* rtcp_transport) {
RTC_DCHECK(network_thread_->IsCurrent());
if (!rtp_transport && !rtcp_transport) {
LOG(LS_ERROR) << "Setting nullptr to RTP Transport and RTCP Transport.";
return false;
}
// Verify some assumptions (as described in the comment above SetTransport).
RTC_DCHECK(rtp_transport);
RTC_DCHECK(NeedsRtcpTransport() == (rtcp_transport != nullptr));
if (rtcp_transport) {
RTC_DCHECK(rtp_transport->transport_name() ==
rtcp_transport->transport_name());
RTC_DCHECK(NeedsRtcpTransport());
}
if (rtp_transport->transport_name() == transport_name_) {
// Nothing to do if transport name isn't changing.
return true;
return;
}
transport_name_ = rtp_transport->transport_name();
@ -300,17 +293,13 @@ bool BaseChannel::SetTransport_n(TransportChannel* rtp_transport,
LOG(LS_INFO) << "Setting RTCP Transport for " << content_name() << " on "
<< transport_name() << " transport " << rtcp_transport;
SetTransportChannel_n(true, rtcp_transport);
if (!rtcp_transport_) {
return false;
}
RTC_DCHECK(rtcp_transport_);
}
LOG(LS_INFO) << "Setting non-RTCP Transport for " << content_name() << " on "
<< transport_name() << " transport " << rtp_transport;
SetTransportChannel_n(false, rtp_transport);
if (!rtp_transport_) {
return false;
}
RTC_DCHECK(rtp_transport_);
// Update aggregate writable/ready-to-send state between RTP and RTCP upon
// setting new transport channels.
@ -328,7 +317,6 @@ bool BaseChannel::SetTransport_n(TransportChannel* rtp_transport,
rtp_transport_ && rtp_transport_->writable());
SetTransportChannelReadyToSend(
true, rtcp_transport_ && rtcp_transport_->writable());
return true;
}
void BaseChannel::SetTransportChannel_n(bool rtcp,

17
webrtc/pc/channel.h
View File

@ -107,8 +107,14 @@ class BaseChannel
bool writable() const { return writable_; }
bool SetTransport(TransportChannel* rtp_transport,
TransportChannel* rtcp_transport);
// Set the transport(s), and update writability and "ready-to-send" state.
// |rtp_transport| must be non-null.
// |rtcp_transport| must be supplied if NeedsRtcpTransport() is true (meaning
// RTCP muxing is not fully active yet).
// |rtp_transport| and |rtcp_transport| must share the same transport name as
// well.
void SetTransports(TransportChannel* rtp_transport,
TransportChannel* rtcp_transport);
bool PushdownLocalDescription(const SessionDescription* local_desc,
ContentAction action,
std::string* error_desc);
@ -194,11 +200,8 @@ class BaseChannel
protected:
virtual MediaChannel* media_channel() const { return media_channel_; }
// Sets the |rtp_transport_| (and |rtcp_transport_|, if
// |rtcp_enabled_| is true).
// This method also updates writability and "ready-to-send" state.
bool SetTransport_n(TransportChannel* rtp_transport,
TransportChannel* rtcp_transport);
void SetTransports_n(TransportChannel* rtp_transport,
TransportChannel* rtcp_transport);
// This does not update writability or "ready-to-send" state; it just
// disconnects from the old channel and connects to the new one.

81
webrtc/pc/channel_unittest.cc
View File

@ -3577,4 +3577,85 @@ TEST_F(RtpDataChannelDoubleThreadTest, TestSendData) {
EXPECT_EQ("foo", media_channel1_->last_sent_data());
}
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// Verifies some DCHECKs are in place.
// Uses VoiceChannel, but any BaseChannel subclass would work.
class BaseChannelDeathTest : public testing::Test {
public:
BaseChannelDeathTest()
: // RTCP mux not required, SRTP required.
voice_channel_(
rtc::Thread::Current(),
rtc::Thread::Current(),
rtc::Thread::Current(),
&fake_media_engine_,
new cricket::FakeVoiceMediaChannel(nullptr,
cricket::AudioOptions()),
cricket::CN_AUDIO,
false,
true) {
rtp_transport_ = fake_transport_controller_.CreateTransportChannel(
"foo", cricket::ICE_CANDIDATE_COMPONENT_RTP);
rtcp_transport_ = fake_transport_controller_.CreateTransportChannel(
"foo", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
EXPECT_TRUE(voice_channel_.Init_w(rtp_transport_, rtcp_transport_));
}
protected:
cricket::FakeTransportController fake_transport_controller_;
cricket::FakeMediaEngine fake_media_engine_;
cricket::VoiceChannel voice_channel_;
// Will be cleaned up by FakeTransportController, don't need to worry about
// deleting them in this test.
cricket::TransportChannel* rtp_transport_;
cricket::TransportChannel* rtcp_transport_;
};
TEST_F(BaseChannelDeathTest, SetTransportWithNullRtpTransport) {
cricket::TransportChannel* new_rtcp_transport =
fake_transport_controller_.CreateTransportChannel(
"bar", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
EXPECT_DEATH(voice_channel_.SetTransports(nullptr, new_rtcp_transport), "");
}
TEST_F(BaseChannelDeathTest, SetTransportWithMissingRtcpTransport) {
cricket::TransportChannel* new_rtp_transport =
fake_transport_controller_.CreateTransportChannel(
"bar", cricket::ICE_CANDIDATE_COMPONENT_RTP);
EXPECT_DEATH(voice_channel_.SetTransports(new_rtp_transport, nullptr), "");
}
TEST_F(BaseChannelDeathTest, SetTransportWithUnneededRtcpTransport) {
// Activate RTCP muxing, simulating offer/answer negotiation.
cricket::AudioContentDescription content;
content.set_rtcp_mux(true);
ASSERT_TRUE(voice_channel_.SetLocalContent(&content, CA_OFFER, nullptr));
ASSERT_TRUE(voice_channel_.SetRemoteContent(&content, CA_ANSWER, nullptr));
cricket::TransportChannel* new_rtp_transport =
fake_transport_controller_.CreateTransportChannel(
"bar", cricket::ICE_CANDIDATE_COMPONENT_RTP);
cricket::TransportChannel* new_rtcp_transport =
fake_transport_controller_.CreateTransportChannel(
"bar", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
// After muxing is enabled, no RTCP transport should be passed in here.
EXPECT_DEATH(
voice_channel_.SetTransports(new_rtp_transport, new_rtcp_transport), "");
}
// This test will probably go away if/when we move the transport name out of
// the transport classes and into their parent classes.
TEST_F(BaseChannelDeathTest, SetTransportWithMismatchingTransportNames) {
cricket::TransportChannel* new_rtp_transport =
fake_transport_controller_.CreateTransportChannel(
"bar", cricket::ICE_CANDIDATE_COMPONENT_RTP);
cricket::TransportChannel* new_rtcp_transport =
fake_transport_controller_.CreateTransportChannel(
"baz", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
EXPECT_DEATH(
voice_channel_.SetTransports(new_rtp_transport, new_rtcp_transport), "");
}
#endif // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
// TODO(pthatcher): TestSetReceiver?