diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_inst.h b/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
index 373db39..8573b6d 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
@@ -15,7 +15,14 @@
 
 struct WebRtcOpusEncInst {
   OpusEncoder* encoder;
+  int channels;
   int in_dtx_mode;
+  // When Opus is in DTX mode, we use |zero_counts| to count consecutive zeros
+  // to break long zero segment so as to prevent DTX from going wrong. We use
+  // one counter for each channel. After each encoding, |zero_counts| contain
+  // the remaining zeros from the last frame.
+  // TODO(minyue): remove this when Opus gets an internal fix to DTX.
+  size_t* zero_counts;
 };
 
 struct WebRtcOpusDecInst {
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
index 1a63242..9eee89f 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_interface.c
@@ -11,6 +11,7 @@
 #include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
 
+#include <assert.h>
 #include <stdlib.h>
 #include <string.h>
 
@@ -29,48 +30,61 @@
 
   /* Default frame size, 20 ms @ 48 kHz, in samples (for one channel). */
   kWebRtcOpusDefaultFrameSize = 960,
+
+  // Maximum number of consecutive zeros, beyond or equal to which DTX can fail.
+  kZeroBreakCount = 157,
+
+#if defined(OPUS_FIXED_POINT)
+  kZeroBreakValue = 10,
+#else
+  kZeroBreakValue = 1,
+#endif
 };
 
 int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
                                  int32_t channels,
                                  int32_t application) {
-  OpusEncInst* state;
-  if (inst != NULL) {
-    state = (OpusEncInst*) calloc(1, sizeof(OpusEncInst));
-    if (state) {
-      int opus_app;
-      switch (application) {
-        case 0: {
-          opus_app = OPUS_APPLICATION_VOIP;
-          break;
-        }
-        case 1: {
-          opus_app = OPUS_APPLICATION_AUDIO;
-          break;
-        }
-        default: {
-          free(state);
-          return -1;
-        }
-      }
+  int opus_app;
+  if (!inst)
+    return -1;
 
-      int error;
-      state->encoder = opus_encoder_create(48000, channels, opus_app,
-                                           &error);
-      state->in_dtx_mode = 0;
-      if (error == OPUS_OK && state->encoder != NULL) {
-        *inst = state;
-        return 0;
-      }
-      free(state);
-    }
+  switch (application) {
+    case 0:
+      opus_app = OPUS_APPLICATION_VOIP;
+      break;
+    case 1:
+      opus_app = OPUS_APPLICATION_AUDIO;
+      break;
+    default:
+      return -1;
   }
-  return -1;
+
+  OpusEncInst* state = calloc(1, sizeof(OpusEncInst));
+  assert(state);
+
+  // Allocate zero counters.
+  state->zero_counts = calloc(channels, sizeof(size_t));
+  assert(state->zero_counts);
+
+  int error;
+  state->encoder = opus_encoder_create(48000, channels, opus_app,
+                                       &error);
+  if (error != OPUS_OK || !state->encoder) {
+    WebRtcOpus_EncoderFree(state);
+    return -1;
+  }
+
+  state->in_dtx_mode = 0;
+  state->channels = channels;
+
+  *inst = state;
+  return 0;
 }
 
 int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
   if (inst) {
     opus_encoder_destroy(inst->encoder);
+    free(inst->zero_counts);
     free(inst);
     return 0;
   } else {
@@ -84,13 +98,42 @@
                       size_t length_encoded_buffer,
                       uint8_t* encoded) {
   int res;
+  size_t i;
+  int c;
+
+  int16_t buffer[2 * 48 * kWebRtcOpusMaxEncodeFrameSizeMs];
 
   if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
     return -1;
   }
 
+  const int channels = inst->channels;
+  int use_buffer = 0;
+
+  // Break long consecutive zeros by forcing a "1" every |kZeroBreakCount|
+  // samples.
+  if (inst->in_dtx_mode) {
+    for (i = 0; i < samples; ++i) {
+      for (c = 0; c < channels; ++c) {
+        if (audio_in[i * channels + c] == 0) {
+          ++inst->zero_counts[c];
+          if (inst->zero_counts[c] == kZeroBreakCount) {
+            if (!use_buffer) {
+              memcpy(buffer, audio_in, samples * channels * sizeof(int16_t));
+              use_buffer = 1;
+            }
+            buffer[i * channels + c] = kZeroBreakValue;
+            inst->zero_counts[c] = 0;
+          }
+        } else {
+          inst->zero_counts[c] = 0;
+        }
+      }
+    }
+  }
+
   res = opus_encode(inst->encoder,
-                    (const opus_int16*)audio_in,
+                    use_buffer ? buffer : audio_in,
                     (int)samples,
                     encoded,
                     (opus_int32)length_encoded_buffer);
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
index c059fc5..fc5d841 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -36,7 +36,7 @@
  protected:
   OpusTest();
 
-  void TestDtxEffect(bool dtx);
+  void TestDtxEffect(bool dtx, int block_length_ms);
 
   // Prepare |speech_data_| for encoding, read from a hard-coded file.
   // After preparation, |speech_data_.GetNextBlock()| returns a pointer to a
@@ -53,6 +53,9 @@
   void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
                           opus_int32 expect, int32_t set);
 
+  void CheckAudioBounded(const int16_t* audio, size_t samples, int channels,
+                         uint16_t bound) const;
+
   WebRtcOpusEncInst* opus_encoder_;
   WebRtcOpusDecInst* opus_decoder_;
 
@@ -95,6 +98,16 @@
   EXPECT_EQ(expect, bandwidth);
 }
 
+void OpusTest::CheckAudioBounded(const int16_t* audio, size_t samples,
+                                 int channels, uint16_t bound) const {
+  for (size_t i = 0; i < samples; ++i) {
+    for (int c = 0; c < channels; ++c) {
+      ASSERT_GE(audio[i * channels + c], -bound);
+      ASSERT_LE(audio[i * channels + c], bound);
+    }
+  }
+}
+
 int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
                            rtc::ArrayView<const int16_t> input_audio,
                            WebRtcOpusDecInst* decoder,
@@ -116,8 +129,9 @@
 
 // Test if encoder/decoder can enter DTX mode properly and do not enter DTX when
 // they should not. This test is signal dependent.
-void OpusTest::TestDtxEffect(bool dtx) {
-  PrepareSpeechData(channels_, 20, 2000);
+void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
+  PrepareSpeechData(channels_, block_length_ms, 2000);
+  const size_t samples = kOpusRateKhz * block_length_ms;
 
   // Create encoder memory.
   EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
@@ -130,17 +144,17 @@
                                      channels_ == 1 ? 32000 : 64000));
 
   // Set input audio as silence.
-  std::vector<int16_t> silence(kOpus20msFrameSamples * channels_, 0);
+  std::vector<int16_t> silence(samples * channels_, 0);
 
   // Setting DTX.
   EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_) :
       WebRtcOpus_DisableDtx(opus_encoder_));
 
   int16_t audio_type;
-  int16_t* output_data_decode = new int16_t[kOpus20msFrameSamples * channels_];
+  int16_t* output_data_decode = new int16_t[samples * channels_];
 
   for (int i = 0; i < 100; ++i) {
-    EXPECT_EQ(kOpus20msFrameSamples,
+    EXPECT_EQ(samples,
               static_cast<size_t>(EncodeDecode(
                   opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
                   output_data_decode, &audio_type)));
@@ -157,9 +171,10 @@
   // We input some silent segments. In DTX mode, the encoder will stop sending.
   // However, DTX may happen after a while.
   for (int i = 0; i < 30; ++i) {
-    EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(EncodeDecode(
-                                         opus_encoder_, silence, opus_decoder_,
-                                         output_data_decode, &audio_type)));
+    EXPECT_EQ(samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, silence, opus_decoder_, output_data_decode,
+                  &audio_type)));
     if (!dtx) {
       EXPECT_GT(encoded_bytes_, 1U);
       EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
@@ -175,21 +190,47 @@
 
   // When Opus is in DTX, it wakes up in a regular basis. It sends two packets,
   // one with an arbitrary size and the other of 1-byte, then stops sending for
-  // 19 frames.
-  const int cycles = 5;
-  for (int j = 0; j < cycles; ++j) {
-    // DTX mode is maintained 19 frames.
-    for (int i = 0; i < 19; ++i) {
-      EXPECT_EQ(kOpus20msFrameSamples,
-                static_cast<size_t>(
-                    EncodeDecode(opus_encoder_, silence, opus_decoder_,
-                                 output_data_decode, &audio_type)));
+  // a certain number of frames.
+
+  // |max_dtx_frames| is the maximum number of frames Opus can stay in DTX.
+  const int max_dtx_frames = 400 / block_length_ms + 1;
+
+  // We run |kRunTimeMs| milliseconds of pure silence.
+  const int kRunTimeMs = 2000;
+
+  // We check that, after a |kCheckTimeMs| milliseconds (given that the CNG in
+  // Opus needs time to adapt), the absolute values of DTX decoded signal are
+  // bounded by |kOutputValueBound|.
+  const int kCheckTimeMs = 1500;
+
+#if defined(OPUS_FIXED_POINT)
+  const uint16_t kOutputValueBound = 20;
+#else
+  const uint16_t kOutputValueBound = 2;
+#endif
+
+  int time = 0;
+  while (time < kRunTimeMs) {
+    // DTX mode is maintained for maximum |max_dtx_frames| frames.
+    int i = 0;
+    for (; i < max_dtx_frames; ++i) {
+      time += block_length_ms;
+      EXPECT_EQ(samples,
+                static_cast<size_t>(EncodeDecode(
+                    opus_encoder_, silence, opus_decoder_, output_data_decode,
+                    &audio_type)));
       if (dtx) {
+        if (encoded_bytes_ > 1)
+          break;
         EXPECT_EQ(0U, encoded_bytes_)  // Send 0 byte.
             << "Opus should have entered DTX mode.";
         EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
         EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
         EXPECT_EQ(2, audio_type);  // Comfort noise.
+        if (time >= kCheckTimeMs) {
+          CheckAudioBounded(output_data_decode, samples, channels_,
+                            kOutputValueBound);
+        }
       } else {
         EXPECT_GT(encoded_bytes_, 1U);
         EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
@@ -198,25 +239,31 @@
       }
     }
 
-    // Quit DTX after 19 frames.
-    EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(EncodeDecode(
-                                         opus_encoder_, silence, opus_decoder_,
-                                         output_data_decode, &audio_type)));
+    if (dtx) {
+      // With DTX, Opus must stop transmission for some time.
+      EXPECT_GT(i, 1);
+    }
 
-    EXPECT_GT(encoded_bytes_, 1U);
+    // We expect a normal payload.
     EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
     EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
     EXPECT_EQ(0, audio_type);  // Speech.
 
     // Enters DTX again immediately.
-    EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(EncodeDecode(
-                                         opus_encoder_, silence, opus_decoder_,
-                                         output_data_decode, &audio_type)));
+    time += block_length_ms;
+    EXPECT_EQ(samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, silence, opus_decoder_, output_data_decode,
+                  &audio_type)));
     if (dtx) {
       EXPECT_EQ(1U, encoded_bytes_);  // Send 1 byte.
       EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
       EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
       EXPECT_EQ(2, audio_type);  // Comfort noise.
+      if (time >= kCheckTimeMs) {
+        CheckAudioBounded(output_data_decode, samples, channels_,
+                          kOutputValueBound);
+      }
     } else {
       EXPECT_GT(encoded_bytes_, 1U);
       EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
@@ -228,9 +275,10 @@
   silence[0] = 10000;
   if (dtx) {
     // Verify that encoder/decoder can jump out from DTX mode.
-    EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(EncodeDecode(
-                                         opus_encoder_, silence, opus_decoder_,
-                                         output_data_decode, &audio_type)));
+    EXPECT_EQ(samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, silence, opus_decoder_, output_data_decode,
+                  &audio_type)));
     EXPECT_GT(encoded_bytes_, 1U);
     EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
     EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
@@ -436,11 +484,15 @@
 }
 
 TEST_P(OpusTest, OpusDtxOff) {
-  TestDtxEffect(false);
+  TestDtxEffect(false, 10);
+  TestDtxEffect(false, 20);
+  TestDtxEffect(false, 40);
 }
 
 TEST_P(OpusTest, OpusDtxOn) {
-  TestDtxEffect(true);
+  TestDtxEffect(true, 10);
+  TestDtxEffect(true, 20);
+  TestDtxEffect(true, 40);
 }
 
 TEST_P(OpusTest, OpusSetPacketLossRate) {
diff --git a/webrtc/modules/audio_coding/main/audio_coding_module.gypi b/webrtc/modules/audio_coding/main/audio_coding_module.gypi
index 088a319..09ac0a8 100644
--- a/webrtc/modules/audio_coding/main/audio_coding_module.gypi
+++ b/webrtc/modules/audio_coding/main/audio_coding_module.gypi
@@ -91,6 +91,11 @@
           '<(webrtc_root)',
         ],
       },
+      'conditions': [
+        ['include_opus==1', {
+          'export_dependent_settings': ['webrtc_opus'],
+        }],
+      ],
       'sources': [
         'acm2/acm_common_defs.h',
         'acm2/acm_receiver.cc',
diff --git a/webrtc/voice_engine/test/auto_test/voe_output_test.cc b/webrtc/voice_engine/test/auto_test/voe_output_test.cc
new file mode 100644
index 0000000..54bfe80
--- /dev/null
+++ b/webrtc/voice_engine/test/auto_test/voe_output_test.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/random.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
+
+namespace {
+
+const char kIp[] = "127.0.0.1";
+const int kPort = 1234;
+const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
+
+}  // namespace
+
+namespace voetest {
+
+using webrtc::test::Random;
+using webrtc::test::VoiceChannelTransport;
+
+// This test allows a check on the output signal in an end-to-end call.
+class OutputTest {
+ public:
+  OutputTest(int16_t lower_bound, int16_t upper_bound);
+  ~OutputTest();
+
+  void Start();
+
+  void EnableOutputCheck();
+  void DisableOutputCheck();
+  void SetOutputBound(int16_t lower_bound, int16_t upper_bound);
+  void Mute();
+  void Unmute();
+  void SetBitRate(int rate);
+
+ private:
+  // This class checks all output values and count the number of samples that
+  // go out of a defined range.
+  class VoEOutputCheckMediaProcess : public VoEMediaProcess {
+   public:
+    VoEOutputCheckMediaProcess(int16_t lower_bound, int16_t upper_bound);
+
+    void set_enabled(bool enabled) { enabled_ = enabled; }
+    void Process(int channel,
+                 ProcessingTypes type,
+                 int16_t audio10ms[],
+                 size_t length,
+                 int samplingFreq,
+                 bool isStereo) override;
+
+   private:
+    bool enabled_;
+    int16_t lower_bound_;
+    int16_t upper_bound_;
+  };
+
+  VoETestManager manager_;
+  VoEOutputCheckMediaProcess output_checker_;
+
+  int channel_;
+};
+
+OutputTest::OutputTest(int16_t lower_bound, int16_t upper_bound)
+    : output_checker_(lower_bound, upper_bound) {
+  EXPECT_TRUE(manager_.Init());
+  manager_.GetInterfaces();
+
+  VoEBase* base = manager_.BasePtr();
+  VoECodec* codec = manager_.CodecPtr();
+  VoENetwork* network = manager_.NetworkPtr();
+
+  EXPECT_EQ(0, base->Init());
+
+  channel_ = base->CreateChannel();
+
+  // |network| will take care of the life time of |transport|.
+  VoiceChannelTransport* transport =
+      new VoiceChannelTransport(network, channel_);
+
+  EXPECT_EQ(0, transport->SetSendDestination(kIp, kPort));
+  EXPECT_EQ(0, transport->SetLocalReceiver(kPort));
+
+  EXPECT_EQ(0, codec->SetSendCodec(channel_, kCodecInst));
+  EXPECT_EQ(0, codec->SetOpusDtx(channel_, true));
+
+  EXPECT_EQ(0, manager_.VolumeControlPtr()->SetSpeakerVolume(255));
+
+  manager_.ExternalMediaPtr()->RegisterExternalMediaProcessing(
+      channel_, ProcessingTypes::kPlaybackPerChannel, output_checker_);
+}
+
+OutputTest::~OutputTest() {
+  EXPECT_EQ(0, manager_.NetworkPtr()->DeRegisterExternalTransport(channel_));
+  EXPECT_EQ(0, manager_.ReleaseInterfaces());
+}
+
+void OutputTest::Start() {
+  const std::string file_name =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
+
+  ASSERT_EQ(0, manager_.FilePtr()->StartPlayingFileAsMicrophone(
+      channel_, file_name.c_str(), true, false, kInputFormat, 1.0));
+
+  VoEBase* base = manager_.BasePtr();
+  ASSERT_EQ(0, base->StartPlayout(channel_));
+  ASSERT_EQ(0, base->StartSend(channel_));
+}
+
+void OutputTest::EnableOutputCheck() {
+  output_checker_.set_enabled(true);
+}
+
+void OutputTest::DisableOutputCheck() {
+  output_checker_.set_enabled(false);
+}
+
+void OutputTest::Mute() {
+  manager_.VolumeControlPtr()->SetInputMute(channel_, true);
+}
+
+void OutputTest::Unmute() {
+  manager_.VolumeControlPtr()->SetInputMute(channel_, false);
+}
+
+void OutputTest::SetBitRate(int rate) {
+  manager_.CodecPtr()->SetBitRate(channel_, rate);
+}
+
+OutputTest::VoEOutputCheckMediaProcess::VoEOutputCheckMediaProcess(
+    int16_t lower_bound, int16_t upper_bound)
+    : enabled_(false),
+      lower_bound_(lower_bound),
+      upper_bound_(upper_bound) {}
+
+void OutputTest::VoEOutputCheckMediaProcess::Process(int channel,
+                                                     ProcessingTypes type,
+                                                     int16_t* audio10ms,
+                                                     size_t length,
+                                                     int samplingFreq,
+                                                     bool isStereo) {
+  if (!enabled_)
+    return;
+  const int num_channels = isStereo ? 2 : 1;
+  for (size_t i = 0; i < length; ++i) {
+    for (int c = 0; c < num_channels; ++c) {
+      ASSERT_GE(audio10ms[i * num_channels + c], lower_bound_);
+      ASSERT_LE(audio10ms[i * num_channels + c], upper_bound_);
+    }
+  }
+}
+
+// This test checks if the Opus does not produce high noise (noise pump) when
+// DTX is enabled. The microphone is toggled on and off, and values of the
+// output signal during muting should be bounded.
+// We do not run this test on bots. Developers that want to see the result
+// and/or listen to sound quality can run this test manually.
+TEST(OutputTest, DISABLED_OpusDtxHasNoNoisePump) {
+  const int kRuntimeMs = 20000;
+  const uint32_t kUnmuteTimeMs = 1000;
+  const int kCheckAfterMute = 2000;
+  const uint32_t kCheckTimeMs = 2000;
+  const int kMinOpusRate = 6000;
+  const int kMaxOpusRate = 64000;
+
+#if defined(OPUS_FIXED_POINT)
+  const int16_t kDtxBoundForSilence = 20;
+#else
+  const int16_t kDtxBoundForSilence = 2;
+#endif
+
+  OutputTest test(-kDtxBoundForSilence, kDtxBoundForSilence);
+  Random random(1234ull);
+
+  uint32_t start_time = rtc::Time();
+  test.Start();
+  while (rtc::TimeSince(start_time) < kRuntimeMs) {
+    webrtc::SleepMs(random.Rand(kUnmuteTimeMs - kUnmuteTimeMs / 10,
+                                kUnmuteTimeMs + kUnmuteTimeMs / 10));
+    test.Mute();
+    webrtc::SleepMs(kCheckAfterMute);
+    test.EnableOutputCheck();
+    webrtc::SleepMs(random.Rand(kCheckTimeMs - kCheckTimeMs / 10,
+                                kCheckTimeMs + kCheckTimeMs / 10));
+    test.DisableOutputCheck();
+    test.SetBitRate(random.Rand(kMinOpusRate, kMaxOpusRate));
+    test.Unmute();
+  }
+}
+
+}  // namespace voetest
diff --git a/webrtc/voice_engine/voice_engine.gyp b/webrtc/voice_engine/voice_engine.gyp
index 221b2aa..265ad01 100644
--- a/webrtc/voice_engine/voice_engine.gyp
+++ b/webrtc/voice_engine/voice_engine.gyp
@@ -28,6 +28,9 @@
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
         '<(webrtc_root)/webrtc.gyp:rtc_event_log',
       ],
+      'export_dependent_settings': [
+        '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
+      ],
       'sources': [
         'include/voe_audio_processing.h',
         'include/voe_base.h',
@@ -154,6 +157,7 @@
             '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
             '<(webrtc_root)/test/test.gyp:channel_transport',
             '<(webrtc_root)/test/test.gyp:test_support',
+            '<(webrtc_root)/test/webrtc_test_common.gyp:webrtc_test_common',
             '<(webrtc_root)/webrtc.gyp:rtc_event_log',
            ],
           'sources': [
@@ -194,6 +198,7 @@
             'test/auto_test/voe_conference_test.cc',
             'test/auto_test/voe_cpu_test.cc',
             'test/auto_test/voe_cpu_test.h',
+            'test/auto_test/voe_output_test.cc',
             'test/auto_test/voe_standard_test.cc',
             'test/auto_test/voe_standard_test.h',
             'test/auto_test/voe_stress_test.cc',