diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc index 502d79fb7b..c9051bb209 100644 --- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc +++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc @@ -113,6 +113,7 @@ class AudioCodingModuleTest : public ::testing::Test { rtp_utility_->Populate(&rtp_header_); input_frame_.sample_rate_hz_ = kSampleRateHz; + input_frame_.num_channels_ = 1; input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000; // 10 ms. COMPILE_ASSERT(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples, audio_frame_too_small); @@ -246,8 +247,8 @@ TEST_F(AudioCodingModuleTest, FailOnZeroDesiredFrequency) { class AudioCodingModuleMtTest : public AudioCodingModuleTest { protected: - static const int kNumPackets = 10000; - static const int kNumPullCalls = 10000; + static const int kNumPackets = 5000; + static const int kNumPullCalls = 5000; AudioCodingModuleMtTest() : AudioCodingModuleTest(), @@ -290,7 +291,7 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest { insert_packet_thread_->Stop(); } - EventTypeWrapper RunTest() { return test_complete_->Wait(60000); } + EventTypeWrapper RunTest() { return test_complete_->Wait(120000); } private: static bool CbSendThread(void* context) { @@ -300,6 +301,10 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest { // The send thread doesn't have to care about the current simulated time, // since only the AcmReceiver is using the clock. bool CbSendImpl() { + if (HasFatalFailure()) { + // End the test early if a fatal failure (ASSERT_*) has occurred. + test_complete_->Set(); + } ++send_count_; InsertAudio(); Encode(); @@ -364,7 +369,7 @@ class AudioCodingModuleMtTest : public AudioCodingModuleTest { SimulatedClock* fake_clock_; }; -TEST_F(AudioCodingModuleMtTest, DISABLED_DoTest) { +TEST_F(AudioCodingModuleMtTest, DoTest) { EXPECT_EQ(kEventSignaled, RunTest()); }