Adding APM configuration in AEC dump.

webrtc/modules/audio_processing/test/audio_processing_unittest.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 1795 matching lines @@
       format_string = "_int";
       break;
     case kFloatFormat:
       format_string = "_float";
       break;
   }
   const std::string ref_filename = test::TempFilename(
       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 the new filed. I might have
accidentally merged it. Please check a following new patch

+#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");
   FILE* out_file = fopen(out_filename.c_str(), "rb");
   ASSERT_TRUE(ref_file != NULL);
   ASSERT_TRUE(out_file != NULL);
   rtc::scoped_ptr<uint8_t[]> ref_bytes;
   rtc::scoped_ptr<uint8_t[]> out_bytes;
 
   size_t ref_size = ReadMessageBytesFromFile(ref_file, &ref_bytes);
@@ skipping 920 matching lines @@
                     std::tr1::make_tuple(16000, 16000, 32000, 16000, 40, 20),
                     std::tr1::make_tuple(16000, 16000, 16000, 16000, 0, 0)));
 #endif
 
 // TODO(henrike): re-implement functionality lost when removing the old main
 //                function. See
 //                https://code.google.com/p/webrtc/issues/detail?id=1981
 
 }  // namespace
 }  // namespace webrtc