Support the Aec dump for the APM in chrome

content/renderer/media/media_stream_audio_processor_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/command_line.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/logging.h"
 #include "base/path_service.h"
 #include "base/time/time.h"
@@ skipping 138 matching lines @@
   media::AudioParameters params_;
 };
 
 TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) {
   // Setup the audio processor without enabling the flag.
   blink::WebMediaConstraints constraints;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
-          params_, constraints, 0, webrtc_audio_device.get()));
+          constraints, 0, webrtc_audio_device.get()));
   EXPECT_FALSE(audio_processor->has_audio_processing());
+  audio_processor->OnCaptureFormatChanged(params_);
 
   ProcessDataAndVerifyFormat(audio_processor,
                              params_.sample_rate(),
                              params_.channels(),
                              params_.sample_rate() / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) {
   // Setup the audio processor with enabling the flag.
   CommandLine::ForCurrentProcess()->AppendSwitch(
       switches::kEnableAudioTrackProcessing);
   blink::WebMediaConstraints constraints;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
-          params_, constraints, 0, webrtc_audio_device.get()));
+          constraints, 0, webrtc_audio_device.get()));
   EXPECT_TRUE(audio_processor->has_audio_processing());
+  audio_processor->OnCaptureFormatChanged(params_);
   VerifyDefaultComponents(audio_processor);
 
   ProcessDataAndVerifyFormat(audio_processor,
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 }  // namespace content