diff --git a/webrtc/api/localaudiosource.cc b/webrtc/api/localaudiosource.cc
index 4b961cf2a9..a118655516 100644
--- a/webrtc/api/localaudiosource.cc
+++ b/webrtc/api/localaudiosource.cc
@@ -50,8 +50,7 @@ void FromConstraints(const MediaConstraintsInterface::Constraints& constraints,
       {MediaConstraintsInterface::kHighpassFilter, options->highpass_filter},
       {MediaConstraintsInterface::kTypingNoiseDetection,
        options->typing_detection},
-      {MediaConstraintsInterface::kAudioMirroring, options->stereo_swapping},
-      {MediaConstraintsInterface::kAecDump, options->aec_dump}
+      {MediaConstraintsInterface::kAudioMirroring, options->stereo_swapping}
   };
 
   for (const auto& constraint : constraints) {
diff --git a/webrtc/api/localaudiosource_unittest.cc b/webrtc/api/localaudiosource_unittest.cc
index fad78d9a91..1abb940414 100644
--- a/webrtc/api/localaudiosource_unittest.cc
+++ b/webrtc/api/localaudiosource_unittest.cc
@@ -35,7 +35,6 @@ TEST(LocalAudioSourceTest, SetValidOptions) {
       MediaConstraintsInterface::kExperimentalAutoGainControl, true);
   constraints.AddMandatory(MediaConstraintsInterface::kNoiseSuppression, false);
   constraints.AddOptional(MediaConstraintsInterface::kHighpassFilter, true);
-  constraints.AddOptional(MediaConstraintsInterface::kAecDump, true);
 
   rtc::scoped_refptr<LocalAudioSource> source =
       LocalAudioSource::Create(PeerConnectionFactoryInterface::Options(),
@@ -48,7 +47,6 @@ TEST(LocalAudioSourceTest, SetValidOptions) {
   EXPECT_EQ(rtc::Optional<bool>(true), source->options().experimental_agc);
   EXPECT_EQ(rtc::Optional<bool>(false), source->options().noise_suppression);
   EXPECT_EQ(rtc::Optional<bool>(true), source->options().highpass_filter);
-  EXPECT_EQ(rtc::Optional<bool>(true), source->options().aec_dump);
 }
 
 TEST(LocalAudioSourceTest, OptionNotSet) {
diff --git a/webrtc/api/mediaconstraintsinterface.cc b/webrtc/api/mediaconstraintsinterface.cc
index 51521946f6..b0a68b15b0 100644
--- a/webrtc/api/mediaconstraintsinterface.cc
+++ b/webrtc/api/mediaconstraintsinterface.cc
@@ -50,7 +50,6 @@ const char MediaConstraintsInterface::kHighpassFilter[] =
 const char MediaConstraintsInterface::kTypingNoiseDetection[] =
     "googTypingNoiseDetection";
 const char MediaConstraintsInterface::kAudioMirroring[] = "googAudioMirroring";
-const char MediaConstraintsInterface::kAecDump[] = "audioDebugRecording";
 
 // Google-specific constraint keys for a local video source (getUserMedia).
 const char MediaConstraintsInterface::kNoiseReduction[] = "googNoiseReduction";
diff --git a/webrtc/api/mediaconstraintsinterface.h b/webrtc/api/mediaconstraintsinterface.h
index ed5d84364d..0c251f8793 100644
--- a/webrtc/api/mediaconstraintsinterface.h
+++ b/webrtc/api/mediaconstraintsinterface.h
@@ -69,7 +69,6 @@ class MediaConstraintsInterface {
   static const char kHighpassFilter[];  // googHighpassFilter
   static const char kTypingNoiseDetection[];  // googTypingNoiseDetection
   static const char kAudioMirroring[];  // googAudioMirroring
-  static const char kAecDump[];               // audioDebugRecording
 
   // Google-specific constraint keys for a local video source
   static const char kNoiseReduction[];  // googNoiseReduction
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 8f15878ecd..bb322ed97d 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -134,7 +134,6 @@ struct AudioOptions {
     SetFrom(&extended_filter_aec, change.extended_filter_aec);
     SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
     SetFrom(&experimental_ns, change.experimental_ns);
-    SetFrom(&aec_dump, change.aec_dump);
     SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
     SetFrom(&tx_agc_digital_compression_gain,
             change.tx_agc_digital_compression_gain);
@@ -160,7 +159,6 @@ struct AudioOptions {
         delay_agnostic_aec == o.delay_agnostic_aec &&
         experimental_ns == o.experimental_ns &&
         adjust_agc_delta == o.adjust_agc_delta &&
-        aec_dump == o.aec_dump &&
         tx_agc_target_dbov == o.tx_agc_target_dbov &&
         tx_agc_digital_compression_gain == o.tx_agc_digital_compression_gain &&
         tx_agc_limiter == o.tx_agc_limiter &&
@@ -188,7 +186,6 @@ struct AudioOptions {
     ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
     ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
     ost << ToStringIfSet("experimental_ns", experimental_ns);
-    ost << ToStringIfSet("aec_dump", aec_dump);
     ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
     ost << ToStringIfSet("tx_agc_digital_compression_gain",
         tx_agc_digital_compression_gain);
@@ -223,7 +220,6 @@ struct AudioOptions {
   rtc::Optional<bool> extended_filter_aec;
   rtc::Optional<bool> delay_agnostic_aec;
   rtc::Optional<bool> experimental_ns;
-  rtc::Optional<bool> aec_dump;
   // Note that tx_agc_* only applies to non-experimental AGC.
   rtc::Optional<uint16_t> tx_agc_target_dbov;
   rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 70f7f3a4b4..3709e807d3 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -92,23 +92,6 @@ const int kOpusMaxBitrate = 510000;
 // 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;
@@ -615,7 +598,6 @@ bool WebRtcVoiceEngine::InitInternal() {
     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;
     }
@@ -868,14 +850,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     }
   }
 
-  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)