Index: webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
index d82ea31c24b1713f6f87ac0e76e00f19df33dc3e..0e4fa99bd2ea27f300f9f07d94894e6f96f3b4f4 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
+++ b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc |
@@ -1813,8 +1813,52 @@ void ApmTest::VerifyDebugDumpTest(Format format) { |
test::OutputPath(), std::string("ref") + format_string + "_aecdump"); |
const std::string out_filename = test::TempFilename( |
test::OutputPath(), std::string("out") + format_string + "_aecdump"); |
- EnableAllComponents(); |
+ |
+#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) |
+ EXPECT_NOERR(apm_->echo_control_mobile()->Enable(true)); |
+ |
+ EXPECT_NOERR(apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital)); |
+ EXPECT_NOERR(apm_->gain_control()->Enable(true)); |
+#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
+ EXPECT_NOERR(apm_->echo_cancellation()->enable_drift_compensation(true)); |
+ EXPECT_NOERR(apm_->echo_cancellation()->enable_metrics(true)); |
+ EXPECT_NOERR(apm_->echo_cancellation()->enable_delay_logging(true)); |
+ EXPECT_NOERR(apm_->echo_cancellation()->Enable(true)); |
+ |
+ EXPECT_NOERR(apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog)); |
+ EXPECT_NOERR(apm_->gain_control()->set_analog_level_limits(0, 255)); |
+ EXPECT_NOERR(apm_->gain_control()->Enable(true)); |
+#endif |
+ |
+ EXPECT_NOERR(apm_->high_pass_filter()->Enable(true)); |
+ EXPECT_NOERR(apm_->level_estimator()->Enable(true)); |
+ EXPECT_NOERR(apm_->noise_suppression()->Enable(true)); |
+ |
+ EXPECT_NOERR(apm_->voice_detection()->Enable(true)); |
ProcessDebugDump(in_filename, ref_filename, format); |
+ |
+// dsadadasaadasdfewafscdeeds |
peah-webrtc
2015/10/02 08:34:27
This comment should be updated to be more proper :
minyue-webrtc
2015/10/02 09:01:42
sorry sorry, this is my local branch for testing t
|
+#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE) |
+ EXPECT_NOERR(apm_->echo_control_mobile()->Enable(false)); |
+ |
+ EXPECT_NOERR(apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital)); |
+ EXPECT_NOERR(apm_->gain_control()->Enable(false)); |
+#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE) |
+ EXPECT_NOERR(apm_->echo_cancellation()->enable_drift_compensation(false)); |
+ EXPECT_NOERR(apm_->echo_cancellation()->enable_metrics(false)); |
+ EXPECT_NOERR(apm_->echo_cancellation()->enable_delay_logging(false)); |
+ EXPECT_NOERR(apm_->echo_cancellation()->Enable(false)); |
+ |
+ EXPECT_NOERR(apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog)); |
+ EXPECT_NOERR(apm_->gain_control()->set_analog_level_limits(0, 255)); |
+ EXPECT_NOERR(apm_->gain_control()->Enable(false)); |
+#endif |
+ |
+ EXPECT_NOERR(apm_->high_pass_filter()->Enable(false)); |
+ EXPECT_NOERR(apm_->level_estimator()->Enable(false)); |
+ EXPECT_NOERR(apm_->noise_suppression()->Enable(false)); |
+ |
+ EXPECT_NOERR(apm_->voice_detection()->Enable(false)); |
ProcessDebugDump(ref_filename, out_filename, format); |
FILE* ref_file = fopen(ref_filename.c_str(), "rb"); |