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Move SelectMediaType from RampUpTester to BaseTest.

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This provides a better default for audio-only tests.

BUG=None

Review-Url: https://codereview.webrtc.org/2794193003
Cr-Commit-Position: refs/heads/master@{#17536}
This commit is contained in:
nisse
2017-04-05 00:33:40 -07:00
committed by Commit bot
parent 6eca98b7e5
commit d8ce1e172e
4 changed files with 19 additions and 25 deletions
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20
webrtc/call/rampup_tests.cc
View File

@ -89,19 +89,6 @@ void RampUpTester::OnVideoStreamsCreated(
send_stream_ = send_stream;
}
MediaType RampUpTester::SelectMediaType() {
if (num_video_streams_ > 0) {
if (num_audio_streams_ > 0) {
// Rely on call to set media type from payload type.
return MediaType::ANY;
} else {
return MediaType::VIDEO;
}
} else {
return MediaType::AUDIO;
}
}
test::PacketTransport* RampUpTester::CreateSendTransport(Call* sender_call) {
send_transport_ = new test::PacketTransport(sender_call, this,
test::PacketTransport::kSender,
@ -110,13 +97,6 @@ test::PacketTransport* RampUpTester::CreateSendTransport(Call* sender_call) {
return send_transport_;
}
test::PacketTransport* RampUpTester::CreateReceiveTransport() {
return new test::PacketTransport(nullptr, this,
test::PacketTransport::kReceiver,
SelectMediaType(),
FakeNetworkPipe::Config());
}
size_t RampUpTester::GetNumVideoStreams() const {
return num_video_streams_;
}

2
webrtc/call/rampup_tests.h
View File

@ -84,9 +84,7 @@ class RampUpTester : public test::EndToEndTest {
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override;
MediaType SelectMediaType();
test::PacketTransport* CreateSendTransport(Call* sender_call) override;
test::PacketTransport* CreateReceiveTransport() override;
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,

17
webrtc/test/call_test.cc
View File

@ -492,15 +492,28 @@ Call::Config BaseTest::GetReceiverCallConfig() {
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
}
MediaType BaseTest::SelectMediaType() {
if (GetNumVideoStreams() > 0) {
if (GetNumAudioStreams() > 0) {
// Relies on PayloadDemuxer to set media type from payload type.
return MediaType::ANY;
} else {
return MediaType::VIDEO;
}
} else {
return MediaType::AUDIO;
}
}
test::PacketTransport* BaseTest::CreateSendTransport(Call* sender_call) {
return new PacketTransport(sender_call, this, test::PacketTransport::kSender,
MediaType::VIDEO,
SelectMediaType(),
FakeNetworkPipe::Config());
}
test::PacketTransport* BaseTest::CreateReceiveTransport() {
return new PacketTransport(nullptr, this, test::PacketTransport::kReceiver,
MediaType::VIDEO,
SelectMediaType(),
FakeNetworkPipe::Config());
}

5
webrtc/test/call_test.h
View File

@ -192,7 +192,10 @@ class BaseTest : public RtpRtcpObserver {
virtual Call::Config GetReceiverCallConfig();
virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
// The default implementation creates MediaType::VIDEO transports.
// Returns VIDEO for video-only tests, AUDIO for audio-only tests,
// and ANY for tests sending audio and video over the same
// transport.
virtual MediaType SelectMediaType();
virtual test::PacketTransport* CreateSendTransport(Call* sender_call);
virtual test::PacketTransport* CreateReceiveTransport();