Remove the 'audioDebugRecording' media constraint and the aec_dump AudioOptions flag.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 74 matching lines @@
 
 // Opus bitrate should be in the range between 6000 and 510000.
 const int kOpusMinBitrate = 6000;
 const int kOpusMaxBitrate = 510000;
 
 // Default audio dscp value.
 // See http://tools.ietf.org/html/rfc2474 for details.
 // See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
 const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
 
-// Ensure we open the file in a writeable path on ChromeOS and Android. This
-// workaround can be removed when it's possible to specify a filename for audio
-// option based AEC dumps.
-//
-// TODO(grunell): Use a string in the options instead of hardcoding it here
-// and let the embedder choose the filename (crbug.com/264223).
-//
-// NOTE(ajm): Don't use hardcoded paths on platforms not explicitly specified
-// below.
-#if defined(CHROMEOS)
-const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
-#elif defined(ANDROID)
-const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
-#else
-const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
-#endif
-
 // Constants from voice_engine_defines.h.
 const int kMinTelephoneEventCode = 0;           // RFC4733 (Section 2.3.1)
 const int kMaxTelephoneEventCode = 255;
 const int kMinTelephoneEventDuration = 100;
 const int kMaxTelephoneEventDuration = 60000;   // Actual limit is 2^16
 
 class ProxySink : public webrtc::AudioSinkInterface {
  public:
   ProxySink(AudioSinkInterface* sink) : sink_(sink) { RTC_DCHECK(sink); }
 
@@ skipping 486 matching lines @@
     options.highpass_filter = rtc::Optional<bool>(true);
     options.stereo_swapping = rtc::Optional<bool>(false);
     options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
     options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
     options.typing_detection = rtc::Optional<bool>(true);
     options.adjust_agc_delta = rtc::Optional<int>(0);
     options.experimental_agc = rtc::Optional<bool>(false);
     options.extended_filter_aec = rtc::Optional<bool>(false);
     options.delay_agnostic_aec = rtc::Optional<bool>(false);
     options.experimental_ns = rtc::Optional<bool>(false);
-    options.aec_dump = rtc::Optional<bool>(false);
     if (!ApplyOptions(options)) {
       return false;
     }
   }
 
   // Print our codec list again for the call diagnostic log
   LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
   for (const AudioCodec& codec : codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
@@ skipping 232 matching lines @@
     }
   }
 
   if (options.adjust_agc_delta) {
     LOG(LS_INFO) << "Adjust agc delta is " << *options.adjust_agc_delta;
     if (!AdjustAgcLevel(*options.adjust_agc_delta)) {
       return false;
     }
   }
 
-  if (options.aec_dump) {
-    LOG(LS_INFO) << "Aec dump is enabled? " << *options.aec_dump;
-    if (*options.aec_dump)
-      StartAecDump(kAecDumpByAudioOptionFilename);
-    else
-      StopAecDump();
-  }
-
   webrtc::Config config;
 
   if (options.delay_agnostic_aec)
     delay_agnostic_aec_ = options.delay_agnostic_aec;
   if (delay_agnostic_aec_) {
     LOG(LS_INFO) << "Delay agnostic aec is enabled? " << *delay_agnostic_aec_;
     config.Set<webrtc::DelayAgnostic>(
         new webrtc::DelayAgnostic(*delay_agnostic_aec_));
   }
 
@@ skipping 1657 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE