Feed the WebRTC APM with empty far-end frames for the number of frames skipped by OS.

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 <vector>
 
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
 #include "base/logging.h"
 #include "base/memory/aligned_memory.h"
@@ skipping 113 matching lines @@
       const bool is_aec_enabled = ap && ap->echo_cancellation()->is_enabled();
 #endif
       if (is_aec_enabled) {
         if (params.channels() > kMaxNumberOfPlayoutDataChannels) {
           for (int i = 0; i < kMaxNumberOfPlayoutDataChannels; ++i) {
             data_bus_playout->SetChannelData(
                 i, const_cast<float*>(data_bus->channel(i)));
           }
         }
         audio_processor->OnPlayoutData(data_bus_playout_to_use,
-                                       params.sample_rate(), 10);
+                                       params.sample_rate(), 10, 0);
       }
 
       media::AudioBus* processed_data = nullptr;
       base::TimeDelta capture_delay;
       int new_volume = 0;
       while (audio_processor->ProcessAndConsumeData(
                  255, false, &processed_data, &capture_delay, &new_volume)) {
         EXPECT_TRUE(processed_data);
         EXPECT_NEAR(input_capture_delay.InMillisecondsF(),
                     capture_delay.InMillisecondsF(),
@@ skipping 443 matching lines @@
   ProcessDataAndVerifyFormat(audio_processor.get(),
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 }  // namespace content