Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_coding/neteq/mock/mock_buffer_level_filter.h

@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+
+#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockBufferLevelFilter : public BufferLevelFilter {
+ public:
+  virtual ~MockBufferLevelFilter() { Die(); }
+  MOCK_METHOD0(Die,
+      void());
+  MOCK_METHOD0(Reset,
+      void());
+  MOCK_METHOD3(Update,
+      void(size_t buffer_size_packets, int time_stretched_samples,
+           size_t packet_len_samples));
+  MOCK_METHOD1(SetTargetBufferLevel,
+      void(int target_buffer_level));
+  MOCK_CONST_METHOD0(filtered_current_level,
+      int());
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_

modules/audio_coding/neteq/mock/mock_decoder_database.h

@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
+
+#include <string>
+
+#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockDecoderDatabase : public DecoderDatabase {
+ public:
+  explicit MockDecoderDatabase(
+      rtc::scoped_refptr<AudioDecoderFactory> factory = nullptr)
+      : DecoderDatabase(factory) {}
+  virtual ~MockDecoderDatabase() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_CONST_METHOD0(Empty,
+      bool());
+  MOCK_CONST_METHOD0(Size,
+      int());
+  MOCK_METHOD0(Reset,
+      void());
+  MOCK_METHOD3(RegisterPayload,
+      int(uint8_t rtp_payload_type, NetEqDecoder codec_type,
+          const std::string& name));
+  MOCK_METHOD2(RegisterPayload,
+               int(int rtp_payload_type, const SdpAudioFormat& audio_format));
+  MOCK_METHOD4(InsertExternal,
+               int(uint8_t rtp_payload_type,
+                   NetEqDecoder codec_type,
+                   const std::string& codec_name,
+                   AudioDecoder* decoder));
+  MOCK_METHOD1(Remove,
+      int(uint8_t rtp_payload_type));
+  MOCK_METHOD0(RemoveAll, void());
+  MOCK_CONST_METHOD1(GetDecoderInfo,
+      const DecoderInfo*(uint8_t rtp_payload_type));
+  MOCK_METHOD2(SetActiveDecoder,
+      int(uint8_t rtp_payload_type, bool* new_decoder));
+  MOCK_CONST_METHOD0(GetActiveDecoder,
+      AudioDecoder*());
+  MOCK_METHOD1(SetActiveCngDecoder,
+      int(uint8_t rtp_payload_type));
+  MOCK_CONST_METHOD0(GetActiveCngDecoder,
+      ComfortNoiseDecoder*());
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_

modules/audio_coding/neteq/mock/mock_delay_manager.h

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+
+#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockDelayManager : public DelayManager {
+ public:
+  MockDelayManager(size_t max_packets_in_buffer,
+                   DelayPeakDetector* peak_detector,
+                   const TickTimer* tick_timer)
+      : DelayManager(max_packets_in_buffer, peak_detector, tick_timer) {}
+  virtual ~MockDelayManager() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_CONST_METHOD0(iat_vector,
+      const IATVector&());
+  MOCK_METHOD3(Update,
+      int(uint16_t sequence_number, uint32_t timestamp, int sample_rate_hz));
+  MOCK_METHOD1(CalculateTargetLevel,
+      int(int iat_packets));
+  MOCK_METHOD1(SetPacketAudioLength,
+      int(int length_ms));
+  MOCK_METHOD0(Reset,
+      void());
+  MOCK_CONST_METHOD0(PeakFound,
+      bool());
+  MOCK_METHOD1(UpdateCounters,
+      void(int elapsed_time_ms));
+  MOCK_METHOD0(ResetPacketIatCount,
+      void());
+  MOCK_CONST_METHOD2(BufferLimits,
+      void(int* lower_limit, int* higher_limit));
+  MOCK_CONST_METHOD0(TargetLevel,
+      int());
+  MOCK_METHOD0(RegisterEmptyPacket, void());
+  MOCK_METHOD1(set_extra_delay_ms,
+      void(int16_t delay));
+  MOCK_CONST_METHOD0(base_target_level,
+      int());
+  MOCK_METHOD1(set_streaming_mode,
+      void(bool value));
+  MOCK_CONST_METHOD0(last_pack_cng_or_dtmf,
+      int());
+  MOCK_METHOD1(set_last_pack_cng_or_dtmf,
+      void(int value));
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_

modules/audio_coding/neteq/mock/mock_delay_peak_detector.h

@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_
+
+#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockDelayPeakDetector : public DelayPeakDetector {
+ public:
+  MockDelayPeakDetector(const TickTimer* tick_timer)
+      : DelayPeakDetector(tick_timer) {}
+  virtual ~MockDelayPeakDetector() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD0(Reset, void());
+  MOCK_METHOD1(SetPacketAudioLength, void(int length_ms));
+  MOCK_METHOD0(peak_found, bool());
+  MOCK_CONST_METHOD0(MaxPeakHeight, int());
+  MOCK_CONST_METHOD0(MaxPeakPeriod, uint64_t());
+  MOCK_METHOD2(Update, bool(int inter_arrival_time, int target_level));
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_

modules/audio_coding/neteq/mock/mock_dtmf_buffer.h

@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
+
+#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockDtmfBuffer : public DtmfBuffer {
+ public:
+  MockDtmfBuffer(int fs) : DtmfBuffer(fs) {}
+  virtual ~MockDtmfBuffer() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD0(Flush,
+      void());
+  MOCK_METHOD1(InsertEvent,
+      int(const DtmfEvent& event));
+  MOCK_METHOD2(GetEvent,
+      bool(uint32_t current_timestamp, DtmfEvent* event));
+  MOCK_CONST_METHOD0(Length,
+      size_t());
+  MOCK_CONST_METHOD0(Empty,
+      bool());
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_

modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h

@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
+
+#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockDtmfToneGenerator : public DtmfToneGenerator {
+ public:
+  virtual ~MockDtmfToneGenerator() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD3(Init,
+      int(int fs, int event, int attenuation));
+  MOCK_METHOD0(Reset,
+      void());
+  MOCK_METHOD2(Generate,
+      int(size_t num_samples, AudioMultiVector* output));
+  MOCK_CONST_METHOD0(initialized,
+      bool());
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_

modules/audio_coding/neteq/mock/mock_expand.h

@@ -0,0 +1,64 @@
+/*
+ *  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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
+
+#include "webrtc/modules/audio_coding/neteq/expand.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockExpand : public Expand {
+ public:
+  MockExpand(BackgroundNoise* background_noise,
+             SyncBuffer* sync_buffer,
+             RandomVector* random_vector,
+             StatisticsCalculator* statistics,
+             int fs,
+             size_t num_channels)
+      : Expand(background_noise,
+               sync_buffer,
+               random_vector,
+               statistics,
+               fs,
+               num_channels) {}
+  virtual ~MockExpand() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD0(Reset,
+      void());
+  MOCK_METHOD1(Process,
+      int(AudioMultiVector* output));
+  MOCK_METHOD0(SetParametersForNormalAfterExpand,
+      void());
+  MOCK_METHOD0(SetParametersForMergeAfterExpand,
+      void());
+  MOCK_CONST_METHOD0(overlap_length,
+      size_t());
+};
+
+}  // namespace webrtc
+
+namespace webrtc {
+
+class MockExpandFactory : public ExpandFactory {
+ public:
+  MOCK_CONST_METHOD6(Create,
+                     Expand*(BackgroundNoise* background_noise,
+                             SyncBuffer* sync_buffer,
+                             RandomVector* random_vector,
+                             StatisticsCalculator* statistics,
+                             int fs,
+                             size_t num_channels));
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_

modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h

@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2013 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
+
+#include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::Invoke;
+
+// Implement an external version of the PCM16b decoder.
+class ExternalPcm16B : public AudioDecoder {
+ public:
+  explicit ExternalPcm16B(int sample_rate_hz)
+      : sample_rate_hz_(sample_rate_hz) {}
+  void Reset() override {}
+
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override {
+    EXPECT_EQ(sample_rate_hz_, sample_rate_hz);
+    size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len, decoded);
+    *speech_type = ConvertSpeechType(1);
+    return static_cast<int>(ret);
+  }
+  int SampleRateHz() const override { return sample_rate_hz_; }
+  size_t Channels() const override { return 1; }
+
+ private:
+  const int sample_rate_hz_;
+  RTC_DISALLOW_COPY_AND_ASSIGN(ExternalPcm16B);
+};
+
+// Create a mock of ExternalPcm16B which delegates all calls to the real object.
+// The reason is that we can then track that the correct calls are being made.
+class MockExternalPcm16B : public AudioDecoder {
+ public:
+  explicit MockExternalPcm16B(int sample_rate_hz) : real_(sample_rate_hz) {
+    // By default, all calls are delegated to the real object.
+    ON_CALL(*this, DecodeInternal(_, _, _, _, _))
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::DecodeInternal));
+    ON_CALL(*this, HasDecodePlc())
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::HasDecodePlc));
+    ON_CALL(*this, DecodePlc(_, _))
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::DecodePlc));
+    ON_CALL(*this, Reset())
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::Reset));
+    ON_CALL(*this, IncomingPacket(_, _, _, _, _))
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::IncomingPacket));
+    ON_CALL(*this, ErrorCode())
+        .WillByDefault(Invoke(&real_, &ExternalPcm16B::ErrorCode));
+  }
+  virtual ~MockExternalPcm16B() { Die(); }
+
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD5(DecodeInternal,
+               int(const uint8_t* encoded,
+                   size_t encoded_len,
+                   int sample_rate_hz,
+                   int16_t* decoded,
+                   SpeechType* speech_type));
+  MOCK_CONST_METHOD0(HasDecodePlc,
+      bool());
+  MOCK_METHOD2(DecodePlc,
+      size_t(size_t num_frames, int16_t* decoded));
+  MOCK_METHOD0(Reset, void());
+  MOCK_METHOD5(IncomingPacket,
+      int(const uint8_t* payload, size_t payload_len,
+          uint16_t rtp_sequence_number, uint32_t rtp_timestamp,
+          uint32_t arrival_timestamp));
+  MOCK_METHOD0(ErrorCode,
+      int());
+
+  int SampleRateHz() const /* override */ { return real_.SampleRateHz(); }
+  size_t Channels() const /* override */ { return real_.Channels(); }
+
+ private:
+  ExternalPcm16B real_;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_

modules/audio_coding/neteq/mock/mock_packet_buffer.h

@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+
+#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockPacketBuffer : public PacketBuffer {
+ public:
+  MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
+      : PacketBuffer(max_number_of_packets, tick_timer) {}
+  virtual ~MockPacketBuffer() { Die(); }
+  MOCK_METHOD0(Die, void());
+  MOCK_METHOD0(Flush,
+      void());
+  MOCK_CONST_METHOD0(Empty,
+      bool());
+  int InsertPacket(Packet&& packet, StatisticsCalculator* stats) {
+    return InsertPacketWrapped(&packet, stats);
+  }
+  // Since gtest does not properly support move-only types, InsertPacket is
+  // implemented as a wrapper. You'll have to implement InsertPacketWrapped
+  // instead and move from |*packet|.
+  MOCK_METHOD2(InsertPacketWrapped,
+               int(Packet* packet, StatisticsCalculator* stats));
+  MOCK_METHOD5(InsertPacketList,
+               int(PacketList* packet_list,
+                   const DecoderDatabase& decoder_database,
+                   rtc::Optional<uint8_t>* current_rtp_payload_type,
+                   rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
+                   StatisticsCalculator* stats));
+  MOCK_CONST_METHOD1(NextTimestamp,
+      int(uint32_t* next_timestamp));
+  MOCK_CONST_METHOD2(NextHigherTimestamp,
+      int(uint32_t timestamp, uint32_t* next_timestamp));
+  MOCK_CONST_METHOD0(PeekNextPacket,
+      const Packet*());
+  MOCK_METHOD0(GetNextPacket,
+      rtc::Optional<Packet>());
+  MOCK_METHOD1(DiscardNextPacket, int(StatisticsCalculator* stats));
+  MOCK_METHOD3(DiscardOldPackets,
+               void(uint32_t timestamp_limit,
+                    uint32_t horizon_samples,
+                    StatisticsCalculator* stats));
+  MOCK_METHOD2(DiscardAllOldPackets,
+               void(uint32_t timestamp_limit, StatisticsCalculator* stats));
+  MOCK_CONST_METHOD0(NumPacketsInBuffer,
+      size_t());
+  MOCK_METHOD1(IncrementWaitingTimes,
+      void(int));
+  MOCK_CONST_METHOD0(current_memory_bytes,
+      int());
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_

modules/audio_coding/neteq/mock/mock_red_payload_splitter.h

@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
+
+#include "webrtc/modules/audio_coding/neteq/red_payload_splitter.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockRedPayloadSplitter : public RedPayloadSplitter {
+ public:
+  MOCK_METHOD1(SplitRed, bool(PacketList* packet_list));
+  MOCK_METHOD2(CheckRedPayloads,
+               int(PacketList* packet_list,
+                   const DecoderDatabase& decoder_database));
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_

modules/audio_coding/neteq/mock/mock_statistics_calculator.h

@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2017 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
+
+#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
+
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+class MockStatisticsCalculator : public StatisticsCalculator {
+ public:
+  MOCK_METHOD1(PacketsDiscarded, void(size_t num_packets));
+  MOCK_METHOD1(SecondaryPacketsDiscarded, void(size_t num_packets));
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_