| /* |
| * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "media/base/media_engine.h" |
| |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| using ::testing::ElementsAre; |
| using ::testing::Field; |
| using ::testing::Return; |
| using ::testing::StrEq; |
| using ::webrtc::RtpExtension; |
| using ::webrtc::RtpHeaderExtensionCapability; |
| using ::webrtc::RtpTransceiverDirection; |
| |
| namespace cricket { |
| namespace { |
| |
| class MockRtpHeaderExtensionQueryInterface |
| : public RtpHeaderExtensionQueryInterface { |
| public: |
| MOCK_METHOD(std::vector<RtpHeaderExtensionCapability>, |
| GetRtpHeaderExtensions, |
| (), |
| (const, override)); |
| }; |
| |
| } // namespace |
| |
| TEST(MediaEngineTest, ReturnsNotStoppedHeaderExtensions) { |
| MockRtpHeaderExtensionQueryInterface mock; |
| std::vector<RtpHeaderExtensionCapability> extensions( |
| {RtpHeaderExtensionCapability("uri1", 1, |
| RtpTransceiverDirection::kInactive), |
| RtpHeaderExtensionCapability("uri2", 2, |
| RtpTransceiverDirection::kSendRecv), |
| RtpHeaderExtensionCapability("uri3", 3, |
| RtpTransceiverDirection::kStopped), |
| RtpHeaderExtensionCapability("uri4", 4, |
| RtpTransceiverDirection::kSendOnly), |
| RtpHeaderExtensionCapability("uri5", 5, |
| RtpTransceiverDirection::kRecvOnly)}); |
| EXPECT_CALL(mock, GetRtpHeaderExtensions).WillOnce(Return(extensions)); |
| EXPECT_THAT(GetDefaultEnabledRtpHeaderExtensions(mock), |
| ElementsAre(Field(&RtpExtension::uri, StrEq("uri1")), |
| Field(&RtpExtension::uri, StrEq("uri2")), |
| Field(&RtpExtension::uri, StrEq("uri4")), |
| Field(&RtpExtension::uri, StrEq("uri5")))); |
| } |
| |
| // This class mocks methods declared as pure virtual in the interface. |
| // Since the tests are aiming to check the patterns of overrides, the |
| // functions with default implementations are not mocked. |
| class MostlyMockVoiceEngineInterface : public VoiceEngineInterface { |
| public: |
| MOCK_METHOD(std::vector<webrtc::RtpHeaderExtensionCapability>, |
| GetRtpHeaderExtensions, |
| (), |
| (const, override)); |
| MOCK_METHOD(void, Init, (), (override)); |
| MOCK_METHOD(rtc::scoped_refptr<webrtc::AudioState>, |
| GetAudioState, |
| (), |
| (const, override)); |
| MOCK_METHOD(std::vector<AudioCodec>&, send_codecs, (), (const, override)); |
| MOCK_METHOD(std::vector<AudioCodec>&, recv_codecs, (), (const, override)); |
| MOCK_METHOD(bool, |
| StartAecDump, |
| (webrtc::FileWrapper file, int64_t max_size_bytes), |
| (override)); |
| MOCK_METHOD(void, StopAecDump, (), (override)); |
| MOCK_METHOD(absl::optional<webrtc::AudioDeviceModule::Stats>, |
| GetAudioDeviceStats, |
| (), |
| (override)); |
| }; |
| |
| class OldStyleVoiceEngineInterface : public MostlyMockVoiceEngineInterface { |
| public: |
| using MostlyMockVoiceEngineInterface::CreateMediaChannel; |
| // Old style overrides the deprecated API only. |
| VoiceMediaChannel* CreateMediaChannel( |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const AudioOptions& options, |
| const webrtc::CryptoOptions& crypto_options) override { |
| ++call_count; |
| return nullptr; |
| } |
| int call_count = 0; |
| }; |
| |
| class NewStyleVoiceEngineInterface : public MostlyMockVoiceEngineInterface { |
| // New style overrides the non-deprecated API. |
| VoiceMediaChannel* CreateMediaChannel( |
| MediaChannel::Role role, |
| webrtc::Call* call, |
| const MediaConfig& config, |
| const AudioOptions& options, |
| const webrtc::CryptoOptions& crypto_options, |
| webrtc::AudioCodecPairId codec_pair_id) override { |
| return nullptr; |
| } |
| }; |
| |
| TEST(MediaEngineTest, NewStyleApiCallsOldIfOverridden) { |
| OldStyleVoiceEngineInterface implementation_under_test; |
| MediaConfig config; |
| AudioOptions options; |
| webrtc::CryptoOptions crypto_options; |
| // Calling the old-style interface. |
| implementation_under_test.CreateMediaChannel(nullptr, config, options, |
| crypto_options); |
| EXPECT_EQ(implementation_under_test.call_count, 1); |
| // Calling the new-style interface redirects to the old-style interface. |
| implementation_under_test.CreateMediaChannel( |
| MediaChannel::Role::kBoth, nullptr, config, options, crypto_options, |
| webrtc::AudioCodecPairId::Create()); |
| EXPECT_EQ(implementation_under_test.call_count, 2); |
| } |
| |
| #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) |
| TEST(MediaEngineTest, NoOverrideOfCreateCausesCrash) { |
| MostlyMockVoiceEngineInterface implementation_under_test; |
| MediaConfig config; |
| AudioOptions options; |
| webrtc::CryptoOptions crypto_options; |
| #pragma clang diagnostic push |
| #pragma clang diagnostic ignored "-Wdeprecated-declarations" |
| |
| EXPECT_DEATH(implementation_under_test.CreateMediaChannel( |
| nullptr, config, options, crypto_options), |
| "Check failed: !recursion_guard_"); |
| #pragma clang diagnostic pop |
| EXPECT_DEATH(implementation_under_test.CreateMediaChannel( |
| MediaChannel::Role::kBoth, nullptr, config, options, |
| crypto_options, webrtc::AudioCodecPairId::Create()), |
| "Check failed: !recursion_guard_"); |
| } |
| #endif |
| |
| } // namespace cricket |