Merge 663413002 to M39:

content/renderer/media/media_stream_audio_processor_options.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 "content/renderer/media/media_stream_audio_processor_options.h"
 
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
 #include "base/logging.h"
 #include "base/metrics/field_trial.h"
@@ skipping 233 matching lines @@
       webrtc::VoiceDetection::kVeryLowLikelihood);
   CHECK_EQ(err, 0);
 
   // Configure the update period to 1s (100 * 10ms) in the typing detector.
   typing_detector->SetParameters(0, 0, 0, 0, 0, 100);
 }
 
 void StartEchoCancellationDump(AudioProcessing* audio_processing,
                                base::File aec_dump_file) {
   DCHECK(aec_dump_file.IsValid());
-
-  FILE* stream = base::FileToFILE(aec_dump_file.Pass(), "w");
-  if (!stream) {
-    LOG(ERROR) << "Failed to open AEC dump file";
-    return;
+  if (audio_processing->StartDebugRecordingForPlatformFile(
+      aec_dump_file.TakePlatformFile())) {
+    DLOG(ERROR) << "Fail to start AEC debug recording";
   }
-
-  if (audio_processing->StartDebugRecording(stream))
-    DLOG(ERROR) << "Fail to start AEC debug recording";
 }
 
 void StopEchoCancellationDump(AudioProcessing* audio_processing) {
   if (audio_processing->StopDebugRecording())
     DLOG(ERROR) << "Fail to stop AEC debug recording";
 }
 
 void EnableAutomaticGainControl(AudioProcessing* audio_processing) {
 #if defined(OS_ANDROID) || defined(OS_IOS)
   const webrtc::GainControl::Mode mode = webrtc::GainControl::kFixedDigital;
@@ skipping 38 matching lines @@
   }
 
   int median = 0, std = 0;
   if (!audio_processing->echo_cancellation()->GetDelayMetrics(&median, &std)) {
     stats->echo_delay_median_ms = median;
     stats->echo_delay_std_ms = std;
   }
 }
 
 }  // namespace content