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273d0296a4
This enables PeerConnection tests to use LoopbackMediaTransport to test data-channel-over-media-transport code. Also changes LoopbackMediaTransport to invoke callbacks asynchronously. This is more accurate, as these callbacks are triggered by network events. The caller should not block while the callback executes. Since LoopbackMediaTransport is used for testing, it provides a FlushAsyncInvokes() method which may be used to ensure that callbacks occur deterministically (eg. before checking that data has been received). Bug: webrtc:9719 Change-Id: Ib8ea9aebf4a0ad3d5934a6fe4ab33432c68523fd Tbr: stefan@webrtc.org Reviewed-on: https://webrtc-review.googlesource.com/c/109060 Commit-Queue: Bjorn Mellem <mellem@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25489}
125 lines
4.1 KiB
C++
125 lines
4.1 KiB
C++
/*
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* Copyright 2018 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include <vector>
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#include "api/test/loopback_media_transport.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace {
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class MockMediaTransportAudioSinkInterface
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: public MediaTransportAudioSinkInterface {
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public:
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MOCK_METHOD2(OnData, void(uint64_t, MediaTransportEncodedAudioFrame));
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};
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class MockDataChannelSink : public DataChannelSink {
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public:
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MOCK_METHOD3(OnDataReceived,
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void(int, DataMessageType, const rtc::CopyOnWriteBuffer&));
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MOCK_METHOD1(OnChannelClosing, void(int));
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MOCK_METHOD1(OnChannelClosed, void(int));
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};
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// Test only uses the sequence number.
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MediaTransportEncodedAudioFrame CreateAudioFrame(int sequence_number) {
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static constexpr int kSamplingRateHz = 48000;
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static constexpr int kStartingSampleIndex = 0;
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static constexpr int kSamplesPerChannel = 480;
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static constexpr uint8_t kPayloadType = 17;
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return MediaTransportEncodedAudioFrame(
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kSamplingRateHz, kStartingSampleIndex, kSamplesPerChannel,
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sequence_number, MediaTransportEncodedAudioFrame::FrameType::kSpeech,
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kPayloadType, std::vector<uint8_t>(kSamplesPerChannel));
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}
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} // namespace
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TEST(LoopbackMediaTransport, AudioWithNoSinkSilentlyIgnored) {
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std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
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thread->Start();
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MediaTransportPair transport_pair(thread.get());
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transport_pair.first()->SendAudioFrame(1, CreateAudioFrame(0));
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transport_pair.second()->SendAudioFrame(2, CreateAudioFrame(0));
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transport_pair.FlushAsyncInvokes();
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}
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TEST(LoopbackMediaTransport, AudioDeliveredToSink) {
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std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
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thread->Start();
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MediaTransportPair transport_pair(thread.get());
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testing::StrictMock<MockMediaTransportAudioSinkInterface> sink;
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EXPECT_CALL(sink,
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OnData(1, testing::Property(
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&MediaTransportEncodedAudioFrame::sequence_number,
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testing::Eq(10))));
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transport_pair.second()->SetReceiveAudioSink(&sink);
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transport_pair.first()->SendAudioFrame(1, CreateAudioFrame(10));
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transport_pair.FlushAsyncInvokes();
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transport_pair.second()->SetReceiveAudioSink(nullptr);
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}
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TEST(LoopbackMediaTransport, DataDeliveredToSink) {
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std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
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thread->Start();
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MediaTransportPair transport_pair(thread.get());
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MockDataChannelSink sink;
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transport_pair.first()->SetDataSink(&sink);
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const int channel_id = 1;
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EXPECT_CALL(sink,
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OnDataReceived(
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channel_id, DataMessageType::kText,
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testing::Property<rtc::CopyOnWriteBuffer, const char*>(
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&rtc::CopyOnWriteBuffer::cdata, testing::StrEq("foo"))));
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SendDataParams params;
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params.type = DataMessageType::kText;
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rtc::CopyOnWriteBuffer buffer("foo");
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transport_pair.second()->SendData(channel_id, params, buffer);
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transport_pair.FlushAsyncInvokes();
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transport_pair.first()->SetDataSink(nullptr);
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}
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TEST(LoopbackMediaTransport, CloseDeliveredToSink) {
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std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
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thread->Start();
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MediaTransportPair transport_pair(thread.get());
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MockDataChannelSink first_sink;
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transport_pair.first()->SetDataSink(&first_sink);
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MockDataChannelSink second_sink;
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transport_pair.second()->SetDataSink(&second_sink);
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const int channel_id = 1;
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{
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testing::InSequence s;
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EXPECT_CALL(second_sink, OnChannelClosing(channel_id));
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EXPECT_CALL(second_sink, OnChannelClosed(channel_id));
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EXPECT_CALL(first_sink, OnChannelClosed(channel_id));
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}
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transport_pair.first()->CloseChannel(channel_id);
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transport_pair.FlushAsyncInvokes();
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transport_pair.first()->SetDataSink(nullptr);
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transport_pair.second()->SetDataSink(nullptr);
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}
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} // namespace webrtc
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