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diff --git a/third_party/libwebrtc/webrtc/test/mock_audio_encoder.h b/third_party/libwebrtc/webrtc/test/mock_audio_encoder.h
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+++ b/third_party/libwebrtc/webrtc/test/mock_audio_encoder.h
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2014 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.
+ */
+
+#ifndef TEST_MOCK_AUDIO_ENCODER_H_
+#define TEST_MOCK_AUDIO_ENCODER_H_
+
+#include <string>
+
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioEncoder : public AudioEncoder {
+ public:
+  // TODO(nisse): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
+  MockAudioEncoder();
+  ~MockAudioEncoder();
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD1(Mark, void(std::string desc));
+  MOCK_CONST_METHOD0(SampleRateHz, int());
+  MOCK_CONST_METHOD0(NumChannels, size_t());
+  MOCK_CONST_METHOD0(RtpTimestampRateHz, int());
+  MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
+  MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
+  MOCK_CONST_METHOD0(GetTargetBitrate, int());
+  MOCK_METHOD0(Reset, void());
+  MOCK_METHOD1(SetFec, bool(bool enable));
+  MOCK_METHOD1(SetDtx, bool(bool enable));
+  MOCK_METHOD1(SetApplication, bool(Application application));
+  MOCK_METHOD1(SetMaxPlaybackRate, void(int frequency_hz));
+  MOCK_METHOD1(SetMaxBitrate, void(int max_bps));
+  MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
+  MOCK_METHOD2(OnReceivedUplinkBandwidth,
+               void(int target_audio_bitrate_bps,
+                    rtc::Optional<int64_t> probing_interval_ms));
+  MOCK_METHOD1(OnReceivedUplinkPacketLossFraction,
+               void(float uplink_packet_loss_fraction));
+
+  MOCK_METHOD2(EnableAudioNetworkAdaptor,
+               bool(const std::string& config_string, RtcEventLog* event_log));
+
+  // Note, we explicitly chose not to create a mock for the Encode method.
+  MOCK_METHOD3(EncodeImpl,
+               EncodedInfo(uint32_t timestamp,
+                           rtc::ArrayView<const int16_t> audio,
+                           rtc::Buffer* encoded));
+
+  class FakeEncoding {
+   public:
+    // Creates a functor that will return |info| and adjust the rtc::Buffer
+    // given as input to it, so it is info.encoded_bytes larger.
+    explicit FakeEncoding(const AudioEncoder::EncodedInfo& info);
+
+    // Shorthand version of the constructor above, for when only setting
+    // encoded_bytes in the EncodedInfo object matters.
+    explicit FakeEncoding(size_t encoded_bytes);
+
+    AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
+                                         rtc::ArrayView<const int16_t> audio,
+                                         rtc::Buffer* encoded);
+
+   private:
+    AudioEncoder::EncodedInfo info_;
+  };
+
+  class CopyEncoding {
+   public:
+    ~CopyEncoding();
+
+    // Creates a functor that will return |info| and append the data in the
+    // payload to the buffer given as input to it. Up to info.encoded_bytes are
+    // appended - make sure the payload is big enough!  Since it uses an
+    // ArrayView, it _does not_ copy the payload. Make sure it doesn't fall out
+    // of scope!
+    CopyEncoding(AudioEncoder::EncodedInfo info,
+                 rtc::ArrayView<const uint8_t> payload);
+
+    // Shorthand version of the constructor above, for when you wish to append
+    // the whole payload and do not care about any EncodedInfo attribute other
+    // than encoded_bytes.
+    explicit CopyEncoding(rtc::ArrayView<const uint8_t> payload);
+
+    AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
+                                         rtc::ArrayView<const int16_t> audio,
+                                         rtc::Buffer* encoded);
+
+   private:
+    AudioEncoder::EncodedInfo info_;
+    rtc::ArrayView<const uint8_t> payload_;
+  };
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_MOCK_AUDIO_ENCODER_H_