diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
deleted file mode 100644
index 7536ce9c28..0000000000
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- *  Copyright (c) 2011 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_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
-
-#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include")
-
-#include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
-#include "webrtc/modules/include/module.h"
-#include "webrtc/modules/include/module_common_types.h"
-
-namespace webrtc {
-class AudioMixerOutputReceiver;
-class MixerParticipant;
-class Trace;
-
-class AudioConferenceMixer : public Module
-{
-public:
-    enum {kMaximumAmountOfMixedParticipants = 3};
-    enum Frequency
-    {
-        kNbInHz           = 8000,
-        kWbInHz           = 16000,
-        kSwbInHz          = 32000,
-        kFbInHz           = 48000,
-        kLowestPossible   = -1,
-        kDefaultFrequency = kWbInHz
-    };
-
-    // Factory method. Constructor disabled.
-    static AudioConferenceMixer* Create(int id);
-    virtual ~AudioConferenceMixer() {}
-
-    // Module functions
-    int64_t TimeUntilNextProcess() override = 0;
-    int32_t Process() override = 0;
-
-    // Register/unregister a callback class for receiving the mixed audio.
-    virtual int32_t RegisterMixedStreamCallback(
-        AudioMixerOutputReceiver* receiver) = 0;
-    virtual int32_t UnRegisterMixedStreamCallback() = 0;
-
-    // Add/remove participants as candidates for mixing.
-    virtual int32_t SetMixabilityStatus(MixerParticipant* participant,
-                                        bool mixable) = 0;
-    // Returns true if a participant is a candidate for mixing.
-    virtual bool MixabilityStatus(
-        const MixerParticipant& participant) const = 0;
-
-    // Inform the mixer that the participant should always be mixed and not
-    // count toward the number of mixed participants. Note that a participant
-    // must have been added to the mixer (by calling SetMixabilityStatus())
-    // before this function can be successfully called.
-    virtual int32_t SetAnonymousMixabilityStatus(
-        MixerParticipant* participant, bool mixable) = 0;
-    // Returns true if the participant is mixed anonymously.
-    virtual bool AnonymousMixabilityStatus(
-        const MixerParticipant& participant) const = 0;
-
-    // Set the minimum sampling frequency at which to mix. The mixing algorithm
-    // may still choose to mix at a higher samling frequency to avoid
-    // downsampling of audio contributing to the mixed audio.
-    virtual int32_t SetMinimumMixingFrequency(Frequency freq) = 0;
-
-protected:
-    AudioConferenceMixer() {}
-};
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h b/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
deleted file mode 100644
index 057c37f267..0000000000
--- a/webrtc/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- *  Copyright (c) 2011 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_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-
-#pragma message("WARNING: audio_conference_mixer/interface is DEPRECATED; use include")
-
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-class MixHistory;
-
-// A callback class that all mixer participants must inherit from/implement.
-class MixerParticipant
-{
-public:
-    // The implementation of this function should update audioFrame with new
-    // audio every time it's called.
-    //
-    // If it returns -1, the frame will not be added to the mix.
-    virtual int32_t GetAudioFrame(int32_t id,
-                                  AudioFrame* audioFrame) = 0;
-
-    // Returns true if the participant was mixed this mix iteration.
-    bool IsMixed() const;
-
-    // This function specifies the sampling frequency needed for the AudioFrame
-    // for future GetAudioFrame(..) calls.
-    virtual int32_t NeededFrequency(int32_t id) const = 0;
-
-    MixHistory* _mixHistory;
-protected:
-    MixerParticipant();
-    virtual ~MixerParticipant();
-};
-
-class AudioMixerOutputReceiver
-{
-public:
-    // This callback function provides the mixed audio for this mix iteration.
-    // Note that uniqueAudioFrames is an array of AudioFrame pointers with the
-    // size according to the size parameter.
-    virtual void NewMixedAudio(const int32_t id,
-                               const AudioFrame& generalAudioFrame,
-                               const AudioFrame** uniqueAudioFrames,
-                               const uint32_t size) = 0;
-protected:
-    AudioMixerOutputReceiver() {}
-    virtual ~AudioMixerOutputReceiver() {}
-};
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INCLUDE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/webrtc/modules/interface/module.h b/webrtc/modules/interface/module.h
deleted file mode 100644
index 06054c3d22..0000000000
--- a/webrtc/modules/interface/module.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- *  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_INCLUDE_MODULE_H_
-#define WEBRTC_MODULES_INCLUDE_MODULE_H_
-
-#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include")
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class ProcessThread;
-
-class Module {
- public:
-  // Returns the number of milliseconds until the module wants a worker
-  // thread to call Process.
-  // This method is called on the same worker thread as Process will
-  // be called on.
-  // TODO(tommi): Almost all implementations of this function, need to know
-  // the current tick count.  Consider passing it as an argument.  It could
-  // also improve the accuracy of when the next callback occurs since the
-  // thread that calls Process() will also have it's tick count reference
-  // which might not match with what the implementations use.
-  virtual int64_t TimeUntilNextProcess() = 0;
-
-  // Process any pending tasks such as timeouts.
-  // Called on a worker thread.
-  virtual int32_t Process() = 0;
-
-  // This method is called when the module is attached to a *running* process
-  // thread or detached from one.  In the case of detaching, |process_thread|
-  // will be nullptr.
-  //
-  // This method will be called in the following cases:
-  //
-  // * Non-null process_thread:
-  //   * ProcessThread::RegisterModule() is called while the thread is running.
-  //   * ProcessThread::Start() is called and RegisterModule has previously
-  //     been called.  The thread will be started immediately after notifying
-  //     all modules.
-  //
-  // * Null process_thread:
-  //   * ProcessThread::DeRegisterModule() is called while the thread is
-  //     running.
-  //   * ProcessThread::Stop() was called and the thread has been stopped.
-  //
-  // NOTE: This method is not called from the worker thread itself, but from
-  //       the thread that registers/deregisters the module or calls Start/Stop.
-  virtual void ProcessThreadAttached(ProcessThread* process_thread) {}
-
- protected:
-  virtual ~Module() {}
-};
-
-// Reference counted version of the Module interface.
-class RefCountedModule : public Module {
- public:
-  // Increase the reference count by one.
-  // Returns the incremented reference count.
-  virtual int32_t AddRef() const = 0;
-
-  // Decrease the reference count by one.
-  // Returns the decreased reference count.
-  // Returns 0 if the last reference was just released.
-  // When the reference count reaches 0 the object will self-destruct.
-  virtual int32_t Release() const = 0;
-
- protected:
-  ~RefCountedModule() override = default;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_INCLUDE_MODULE_H_
diff --git a/webrtc/modules/interface/module_common_types.h b/webrtc/modules/interface/module_common_types.h
deleted file mode 100644
index 8d96870703..0000000000
--- a/webrtc/modules/interface/module_common_types.h
+++ /dev/null
@@ -1,812 +0,0 @@
-/*
- *  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_INCLUDE_MODULE_COMMON_TYPES_H_
-#define WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
-
-#pragma message("WARNING: webrtc/modules/include is DEPRECATED; use webrtc/modules/include")
-
-#include <assert.h>
-#include <string.h>  // memcpy
-
-#include <algorithm>
-#include <limits>
-
-#include "webrtc/base/constructormagic.h"
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/rotation.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-struct RTPAudioHeader {
-  uint8_t numEnergy;                  // number of valid entries in arrOfEnergy
-  uint8_t arrOfEnergy[kRtpCsrcSize];  // one energy byte (0-9) per channel
-  bool isCNG;                         // is this CNG
-  uint8_t channel;                    // number of channels 2 = stereo
-};
-
-const int16_t kNoPictureId = -1;
-const int16_t kMaxOneBytePictureId = 0x7F;    // 7 bits
-const int16_t kMaxTwoBytePictureId = 0x7FFF;  // 15 bits
-const int16_t kNoTl0PicIdx = -1;
-const uint8_t kNoTemporalIdx = 0xFF;
-const uint8_t kNoSpatialIdx = 0xFF;
-const uint8_t kNoGofIdx = 0xFF;
-const size_t kMaxVp9RefPics = 3;
-const size_t kMaxVp9FramesInGof = 0xFF;  // 8 bits
-const size_t kMaxVp9NumberOfSpatialLayers = 8;
-const int kNoKeyIdx = -1;
-
-struct RTPVideoHeaderVP8 {
-  void InitRTPVideoHeaderVP8() {
-    nonReference = false;
-    pictureId = kNoPictureId;
-    tl0PicIdx = kNoTl0PicIdx;
-    temporalIdx = kNoTemporalIdx;
-    layerSync = false;
-    keyIdx = kNoKeyIdx;
-    partitionId = 0;
-    beginningOfPartition = false;
-  }
-
-  bool nonReference;          // Frame is discardable.
-  int16_t pictureId;          // Picture ID index, 15 bits;
-                              // kNoPictureId if PictureID does not exist.
-  int16_t tl0PicIdx;          // TL0PIC_IDX, 8 bits;
-                              // kNoTl0PicIdx means no value provided.
-  uint8_t temporalIdx;        // Temporal layer index, or kNoTemporalIdx.
-  bool layerSync;             // This frame is a layer sync frame.
-                              // Disabled if temporalIdx == kNoTemporalIdx.
-  int keyIdx;                 // 5 bits; kNoKeyIdx means not used.
-  int partitionId;            // VP8 partition ID
-  bool beginningOfPartition;  // True if this packet is the first
-                              // in a VP8 partition. Otherwise false
-};
-
-enum TemporalStructureMode {
-  kTemporalStructureMode1,    // 1 temporal layer structure - i.e., IPPP...
-  kTemporalStructureMode2,    // 2 temporal layers 0-1-0-1...
-  kTemporalStructureMode3     // 3 temporal layers 0-2-1-2-0-2-1-2...
-};
-
-struct GofInfoVP9 {
-  void SetGofInfoVP9(TemporalStructureMode tm) {
-    switch (tm) {
-      case kTemporalStructureMode1:
-        num_frames_in_gof = 1;
-        temporal_idx[0] = 0;
-        temporal_up_switch[0] = false;
-        num_ref_pics[0] = 1;
-        pid_diff[0][0] = 1;
-        break;
-      case kTemporalStructureMode2:
-        num_frames_in_gof = 2;
-        temporal_idx[0] = 0;
-        temporal_up_switch[0] = false;
-        num_ref_pics[0] = 1;
-        pid_diff[0][0] = 2;
-
-        temporal_idx[1] = 1;
-        temporal_up_switch[1] = true;
-        num_ref_pics[1] = 1;
-        pid_diff[1][0] = 1;
-        break;
-      case kTemporalStructureMode3:
-        num_frames_in_gof = 4;
-        temporal_idx[0] = 0;
-        temporal_up_switch[0] = false;
-        num_ref_pics[0] = 1;
-        pid_diff[0][0] = 4;
-
-        temporal_idx[1] = 2;
-        temporal_up_switch[1] = true;
-        num_ref_pics[1] = 1;
-        pid_diff[1][0] = 1;
-
-        temporal_idx[2] = 1;
-        temporal_up_switch[2] = true;
-        num_ref_pics[2] = 1;
-        pid_diff[2][0] = 2;
-
-        temporal_idx[3] = 2;
-        temporal_up_switch[3] = false;
-        num_ref_pics[3] = 2;
-        pid_diff[3][0] = 1;
-        pid_diff[3][1] = 2;
-        break;
-      default:
-        assert(false);
-    }
-  }
-
-  void CopyGofInfoVP9(const GofInfoVP9& src) {
-    num_frames_in_gof = src.num_frames_in_gof;
-    for (size_t i = 0; i < num_frames_in_gof; ++i) {
-      temporal_idx[i] = src.temporal_idx[i];
-      temporal_up_switch[i] = src.temporal_up_switch[i];
-      num_ref_pics[i] = src.num_ref_pics[i];
-      for (uint8_t r = 0; r < num_ref_pics[i]; ++r) {
-        pid_diff[i][r] = src.pid_diff[i][r];
-      }
-    }
-  }
-
-  size_t num_frames_in_gof;
-  uint8_t temporal_idx[kMaxVp9FramesInGof];
-  bool temporal_up_switch[kMaxVp9FramesInGof];
-  uint8_t num_ref_pics[kMaxVp9FramesInGof];
-  uint8_t pid_diff[kMaxVp9FramesInGof][kMaxVp9RefPics];
-};
-
-struct RTPVideoHeaderVP9 {
-  void InitRTPVideoHeaderVP9() {
-    inter_pic_predicted = false;
-    flexible_mode = false;
-    beginning_of_frame = false;
-    end_of_frame = false;
-    ss_data_available = false;
-    picture_id = kNoPictureId;
-    max_picture_id = kMaxTwoBytePictureId;
-    tl0_pic_idx = kNoTl0PicIdx;
-    temporal_idx = kNoTemporalIdx;
-    spatial_idx = kNoSpatialIdx;
-    temporal_up_switch = false;
-    inter_layer_predicted = false;
-    gof_idx = kNoGofIdx;
-    num_ref_pics = 0;
-    num_spatial_layers = 1;
-  }
-
-  bool inter_pic_predicted;  // This layer frame is dependent on previously
-                             // coded frame(s).
-  bool flexible_mode;        // This frame is in flexible mode.
-  bool beginning_of_frame;   // True if this packet is the first in a VP9 layer
-                             // frame.
-  bool end_of_frame;  // True if this packet is the last in a VP9 layer frame.
-  bool ss_data_available;  // True if SS data is available in this payload
-                           // descriptor.
-  int16_t picture_id;      // PictureID index, 15 bits;
-                           // kNoPictureId if PictureID does not exist.
-  int16_t max_picture_id;  // Maximum picture ID index; either 0x7F or 0x7FFF;
-  int16_t tl0_pic_idx;     // TL0PIC_IDX, 8 bits;
-                           // kNoTl0PicIdx means no value provided.
-  uint8_t temporal_idx;    // Temporal layer index, or kNoTemporalIdx.
-  uint8_t spatial_idx;     // Spatial layer index, or kNoSpatialIdx.
-  bool temporal_up_switch;  // True if upswitch to higher frame rate is possible
-                            // starting from this frame.
-  bool inter_layer_predicted;  // Frame is dependent on directly lower spatial
-                               // layer frame.
-
-  uint8_t gof_idx;  // Index to predefined temporal frame info in SS data.
-
-  uint8_t num_ref_pics;  // Number of reference pictures used by this layer
-                         // frame.
-  uint8_t pid_diff[kMaxVp9RefPics];  // P_DIFF signaled to derive the PictureID
-                                     // of the reference pictures.
-  int16_t ref_picture_id[kMaxVp9RefPics];  // PictureID of reference pictures.
-
-  // SS data.
-  size_t num_spatial_layers;  // Always populated.
-  bool spatial_layer_resolution_present;
-  uint16_t width[kMaxVp9NumberOfSpatialLayers];
-  uint16_t height[kMaxVp9NumberOfSpatialLayers];
-  GofInfoVP9 gof;
-};
-
-// The packetization types that we support: single, aggregated, and fragmented.
-enum H264PacketizationTypes {
-  kH264SingleNalu,  // This packet contains a single NAL unit.
-  kH264StapA,       // This packet contains STAP-A (single time
-                    // aggregation) packets. If this packet has an
-                    // associated NAL unit type, it'll be for the
-                    // first such aggregated packet.
-  kH264FuA,         // This packet contains a FU-A (fragmentation
-                    // unit) packet, meaning it is a part of a frame
-                    // that was too large to fit into a single packet.
-};
-
-struct RTPVideoHeaderH264 {
-  uint8_t nalu_type;  // The NAL unit type. If this is a header for a
-                      // fragmented packet, it's the NAL unit type of
-                      // the original data. If this is the header for an
-                      // aggregated packet, it's the NAL unit type of
-                      // the first NAL unit in the packet.
-  H264PacketizationTypes packetization_type;
-};
-
-union RTPVideoTypeHeader {
-  RTPVideoHeaderVP8 VP8;
-  RTPVideoHeaderVP9 VP9;
-  RTPVideoHeaderH264 H264;
-};
-
-enum RtpVideoCodecTypes {
-  kRtpVideoNone,
-  kRtpVideoGeneric,
-  kRtpVideoVp8,
-  kRtpVideoVp9,
-  kRtpVideoH264
-};
-// Since RTPVideoHeader is used as a member of a union, it can't have a
-// non-trivial default constructor.
-struct RTPVideoHeader {
-  uint16_t width;  // size
-  uint16_t height;
-  VideoRotation rotation;
-
-  bool isFirstPacket;    // first packet in frame
-  uint8_t simulcastIdx;  // Index if the simulcast encoder creating
-                         // this frame, 0 if not using simulcast.
-  RtpVideoCodecTypes codec;
-  RTPVideoTypeHeader codecHeader;
-};
-union RTPTypeHeader {
-  RTPAudioHeader Audio;
-  RTPVideoHeader Video;
-};
-
-struct WebRtcRTPHeader {
-  RTPHeader header;
-  FrameType frameType;
-  RTPTypeHeader type;
-  // NTP time of the capture time in local timebase in milliseconds.
-  int64_t ntp_time_ms;
-};
-
-class RTPFragmentationHeader {
- public:
-  RTPFragmentationHeader()
-      : fragmentationVectorSize(0),
-        fragmentationOffset(NULL),
-        fragmentationLength(NULL),
-        fragmentationTimeDiff(NULL),
-        fragmentationPlType(NULL) {};
-
-  ~RTPFragmentationHeader() {
-    delete[] fragmentationOffset;
-    delete[] fragmentationLength;
-    delete[] fragmentationTimeDiff;
-    delete[] fragmentationPlType;
-  }
-
-  void CopyFrom(const RTPFragmentationHeader& src) {
-    if (this == &src) {
-      return;
-    }
-
-    if (src.fragmentationVectorSize != fragmentationVectorSize) {
-      // new size of vectors
-
-      // delete old
-      delete[] fragmentationOffset;
-      fragmentationOffset = NULL;
-      delete[] fragmentationLength;
-      fragmentationLength = NULL;
-      delete[] fragmentationTimeDiff;
-      fragmentationTimeDiff = NULL;
-      delete[] fragmentationPlType;
-      fragmentationPlType = NULL;
-
-      if (src.fragmentationVectorSize > 0) {
-        // allocate new
-        if (src.fragmentationOffset) {
-          fragmentationOffset = new size_t[src.fragmentationVectorSize];
-        }
-        if (src.fragmentationLength) {
-          fragmentationLength = new size_t[src.fragmentationVectorSize];
-        }
-        if (src.fragmentationTimeDiff) {
-          fragmentationTimeDiff = new uint16_t[src.fragmentationVectorSize];
-        }
-        if (src.fragmentationPlType) {
-          fragmentationPlType = new uint8_t[src.fragmentationVectorSize];
-        }
-      }
-      // set new size
-      fragmentationVectorSize = src.fragmentationVectorSize;
-    }
-
-    if (src.fragmentationVectorSize > 0) {
-      // copy values
-      if (src.fragmentationOffset) {
-        memcpy(fragmentationOffset, src.fragmentationOffset,
-               src.fragmentationVectorSize * sizeof(size_t));
-      }
-      if (src.fragmentationLength) {
-        memcpy(fragmentationLength, src.fragmentationLength,
-               src.fragmentationVectorSize * sizeof(size_t));
-      }
-      if (src.fragmentationTimeDiff) {
-        memcpy(fragmentationTimeDiff, src.fragmentationTimeDiff,
-               src.fragmentationVectorSize * sizeof(uint16_t));
-      }
-      if (src.fragmentationPlType) {
-        memcpy(fragmentationPlType, src.fragmentationPlType,
-               src.fragmentationVectorSize * sizeof(uint8_t));
-      }
-    }
-  }
-
-  void VerifyAndAllocateFragmentationHeader(const size_t size) {
-    assert(size <= std::numeric_limits<uint16_t>::max());
-    const uint16_t size16 = static_cast<uint16_t>(size);
-    if (fragmentationVectorSize < size16) {
-      uint16_t oldVectorSize = fragmentationVectorSize;
-      {
-        // offset
-        size_t* oldOffsets = fragmentationOffset;
-        fragmentationOffset = new size_t[size16];
-        memset(fragmentationOffset + oldVectorSize, 0,
-               sizeof(size_t) * (size16 - oldVectorSize));
-        // copy old values
-        memcpy(fragmentationOffset, oldOffsets,
-               sizeof(size_t) * oldVectorSize);
-        delete[] oldOffsets;
-      }
-      // length
-      {
-        size_t* oldLengths = fragmentationLength;
-        fragmentationLength = new size_t[size16];
-        memset(fragmentationLength + oldVectorSize, 0,
-               sizeof(size_t) * (size16 - oldVectorSize));
-        memcpy(fragmentationLength, oldLengths,
-               sizeof(size_t) * oldVectorSize);
-        delete[] oldLengths;
-      }
-      // time diff
-      {
-        uint16_t* oldTimeDiffs = fragmentationTimeDiff;
-        fragmentationTimeDiff = new uint16_t[size16];
-        memset(fragmentationTimeDiff + oldVectorSize, 0,
-               sizeof(uint16_t) * (size16 - oldVectorSize));
-        memcpy(fragmentationTimeDiff, oldTimeDiffs,
-               sizeof(uint16_t) * oldVectorSize);
-        delete[] oldTimeDiffs;
-      }
-      // payload type
-      {
-        uint8_t* oldTimePlTypes = fragmentationPlType;
-        fragmentationPlType = new uint8_t[size16];
-        memset(fragmentationPlType + oldVectorSize, 0,
-               sizeof(uint8_t) * (size16 - oldVectorSize));
-        memcpy(fragmentationPlType, oldTimePlTypes,
-               sizeof(uint8_t) * oldVectorSize);
-        delete[] oldTimePlTypes;
-      }
-      fragmentationVectorSize = size16;
-    }
-  }
-
-  uint16_t fragmentationVectorSize;  // Number of fragmentations
-  size_t* fragmentationOffset;       // Offset of pointer to data for each
-                                     // fragmentation
-  size_t* fragmentationLength;       // Data size for each fragmentation
-  uint16_t* fragmentationTimeDiff;   // Timestamp difference relative "now" for
-                                     // each fragmentation
-  uint8_t* fragmentationPlType;      // Payload type of each fragmentation
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(RTPFragmentationHeader);
-};
-
-struct RTCPVoIPMetric {
-  // RFC 3611 4.7
-  uint8_t lossRate;
-  uint8_t discardRate;
-  uint8_t burstDensity;
-  uint8_t gapDensity;
-  uint16_t burstDuration;
-  uint16_t gapDuration;
-  uint16_t roundTripDelay;
-  uint16_t endSystemDelay;
-  uint8_t signalLevel;
-  uint8_t noiseLevel;
-  uint8_t RERL;
-  uint8_t Gmin;
-  uint8_t Rfactor;
-  uint8_t extRfactor;
-  uint8_t MOSLQ;
-  uint8_t MOSCQ;
-  uint8_t RXconfig;
-  uint16_t JBnominal;
-  uint16_t JBmax;
-  uint16_t JBabsMax;
-};
-
-// Types for the FEC packet masks. The type |kFecMaskRandom| is based on a
-// random loss model. The type |kFecMaskBursty| is based on a bursty/consecutive
-// loss model. The packet masks are defined in
-// modules/rtp_rtcp/fec_private_tables_random(bursty).h
-enum FecMaskType {
-  kFecMaskRandom,
-  kFecMaskBursty,
-};
-
-// Struct containing forward error correction settings.
-struct FecProtectionParams {
-  int fec_rate;
-  bool use_uep_protection;
-  int max_fec_frames;
-  FecMaskType fec_mask_type;
-};
-
-// Interface used by the CallStats class to distribute call statistics.
-// Callbacks will be triggered as soon as the class has been registered to a
-// CallStats object using RegisterStatsObserver.
-class CallStatsObserver {
- public:
-  virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) = 0;
-
-  virtual ~CallStatsObserver() {}
-};
-
-struct VideoContentMetrics {
-  VideoContentMetrics()
-      : motion_magnitude(0.0f),
-        spatial_pred_err(0.0f),
-        spatial_pred_err_h(0.0f),
-        spatial_pred_err_v(0.0f) {}
-
-  void Reset() {
-    motion_magnitude = 0.0f;
-    spatial_pred_err = 0.0f;
-    spatial_pred_err_h = 0.0f;
-    spatial_pred_err_v = 0.0f;
-  }
-  float motion_magnitude;
-  float spatial_pred_err;
-  float spatial_pred_err_h;
-  float spatial_pred_err_v;
-};
-
-/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
- * allows for adding and subtracting frames while keeping track of the resulting
- * states.
- *
- * Notes
- * - The total number of samples in |data_| is
- *   samples_per_channel_ * num_channels_
- *
- * - Stereo data is interleaved starting with the left channel.
- *
- * - The +operator assume that you would never add exactly opposite frames when
- *   deciding the resulting state. To do this use the -operator.
- */
-class AudioFrame {
- public:
-  // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
-  static const size_t kMaxDataSizeSamples = 3840;
-
-  enum VADActivity {
-    kVadActive = 0,
-    kVadPassive = 1,
-    kVadUnknown = 2
-  };
-  enum SpeechType {
-    kNormalSpeech = 0,
-    kPLC = 1,
-    kCNG = 2,
-    kPLCCNG = 3,
-    kUndefined = 4
-  };
-
-  AudioFrame();
-  virtual ~AudioFrame() {}
-
-  // Resets all members to their default state (except does not modify the
-  // contents of |data_|).
-  void Reset();
-
-  // |interleaved_| is not changed by this method.
-  void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
-                   size_t samples_per_channel, int sample_rate_hz,
-                   SpeechType speech_type, VADActivity vad_activity,
-                   int num_channels = 1, uint32_t energy = -1);
-
-  AudioFrame& Append(const AudioFrame& rhs);
-
-  void CopyFrom(const AudioFrame& src);
-
-  void Mute();
-
-  AudioFrame& operator>>=(const int rhs);
-  AudioFrame& operator+=(const AudioFrame& rhs);
-  AudioFrame& operator-=(const AudioFrame& rhs);
-
-  int id_;
-  // RTP timestamp of the first sample in the AudioFrame.
-  uint32_t timestamp_;
-  // Time since the first frame in milliseconds.
-  // -1 represents an uninitialized value.
-  int64_t elapsed_time_ms_;
-  // NTP time of the estimated capture time in local timebase in milliseconds.
-  // -1 represents an uninitialized value.
-  int64_t ntp_time_ms_;
-  int16_t data_[kMaxDataSizeSamples];
-  size_t samples_per_channel_;
-  int sample_rate_hz_;
-  int num_channels_;
-  SpeechType speech_type_;
-  VADActivity vad_activity_;
-  // Note that there is no guarantee that |energy_| is correct. Any user of this
-  // member must verify that the value is correct.
-  // TODO(henrike) Remove |energy_|.
-  // See https://code.google.com/p/webrtc/issues/detail?id=3315.
-  uint32_t energy_;
-  bool interleaved_;
-
- private:
-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
-};
-
-inline AudioFrame::AudioFrame()
-    : data_() {
-  Reset();
-}
-
-inline void AudioFrame::Reset() {
-  id_ = -1;
-  // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
-  // to an invalid value, or add a new member to indicate invalidity.
-  timestamp_ = 0;
-  elapsed_time_ms_ = -1;
-  ntp_time_ms_ = -1;
-  samples_per_channel_ = 0;
-  sample_rate_hz_ = 0;
-  num_channels_ = 0;
-  speech_type_ = kUndefined;
-  vad_activity_ = kVadUnknown;
-  energy_ = 0xffffffff;
-  interleaved_ = true;
-}
-
-inline void AudioFrame::UpdateFrame(int id,
-                                    uint32_t timestamp,
-                                    const int16_t* data,
-                                    size_t samples_per_channel,
-                                    int sample_rate_hz,
-                                    SpeechType speech_type,
-                                    VADActivity vad_activity,
-                                    int num_channels,
-                                    uint32_t energy) {
-  id_ = id;
-  timestamp_ = timestamp;
-  samples_per_channel_ = samples_per_channel;
-  sample_rate_hz_ = sample_rate_hz;
-  speech_type_ = speech_type;
-  vad_activity_ = vad_activity;
-  num_channels_ = num_channels;
-  energy_ = energy;
-
-  assert(num_channels >= 0);
-  const size_t length = samples_per_channel * num_channels;
-  assert(length <= kMaxDataSizeSamples);
-  if (data != NULL) {
-    memcpy(data_, data, sizeof(int16_t) * length);
-  } else {
-    memset(data_, 0, sizeof(int16_t) * length);
-  }
-}
-
-inline void AudioFrame::CopyFrom(const AudioFrame& src) {
-  if (this == &src) return;
-
-  id_ = src.id_;
-  timestamp_ = src.timestamp_;
-  elapsed_time_ms_ = src.elapsed_time_ms_;
-  ntp_time_ms_ = src.ntp_time_ms_;
-  samples_per_channel_ = src.samples_per_channel_;
-  sample_rate_hz_ = src.sample_rate_hz_;
-  speech_type_ = src.speech_type_;
-  vad_activity_ = src.vad_activity_;
-  num_channels_ = src.num_channels_;
-  energy_ = src.energy_;
-  interleaved_ = src.interleaved_;
-
-  assert(num_channels_ >= 0);
-  const size_t length = samples_per_channel_ * num_channels_;
-  assert(length <= kMaxDataSizeSamples);
-  memcpy(data_, src.data_, sizeof(int16_t) * length);
-}
-
-inline void AudioFrame::Mute() {
-  memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
-}
-
-inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
-  assert((num_channels_ > 0) && (num_channels_ < 3));
-  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
-
-  for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
-    data_[i] = static_cast<int16_t>(data_[i] >> rhs);
-  }
-  return *this;
-}
-
-inline AudioFrame& AudioFrame::Append(const AudioFrame& rhs) {
-  // Sanity check
-  assert((num_channels_ > 0) && (num_channels_ < 3));
-  assert(interleaved_ == rhs.interleaved_);
-  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
-  if (num_channels_ != rhs.num_channels_) return *this;
-
-  if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
-    vad_activity_ = kVadActive;
-  } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
-    vad_activity_ = kVadUnknown;
-  }
-  if (speech_type_ != rhs.speech_type_) {
-    speech_type_ = kUndefined;
-  }
-
-  size_t offset = samples_per_channel_ * num_channels_;
-  for (size_t i = 0; i < rhs.samples_per_channel_ * rhs.num_channels_; i++) {
-    data_[offset + i] = rhs.data_[i];
-  }
-  samples_per_channel_ += rhs.samples_per_channel_;
-  return *this;
-}
-
-namespace {
-inline int16_t ClampToInt16(int32_t input) {
-  if (input < -0x00008000) {
-    return -0x8000;
-  } else if (input > 0x00007FFF) {
-    return 0x7FFF;
-  } else {
-    return static_cast<int16_t>(input);
-  }
-}
-}
-
-inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
-  // Sanity check
-  assert((num_channels_ > 0) && (num_channels_ < 3));
-  assert(interleaved_ == rhs.interleaved_);
-  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
-  if (num_channels_ != rhs.num_channels_) return *this;
-
-  bool noPrevData = false;
-  if (samples_per_channel_ != rhs.samples_per_channel_) {
-    if (samples_per_channel_ == 0) {
-      // special case we have no data to start with
-      samples_per_channel_ = rhs.samples_per_channel_;
-      noPrevData = true;
-    } else {
-      return *this;
-    }
-  }
-
-  if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
-    vad_activity_ = kVadActive;
-  } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
-    vad_activity_ = kVadUnknown;
-  }
-
-  if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
-
-  if (noPrevData) {
-    memcpy(data_, rhs.data_,
-           sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
-  } else {
-    // IMPROVEMENT this can be done very fast in assembly
-    for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
-      int32_t wrap_guard =
-          static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
-      data_[i] = ClampToInt16(wrap_guard);
-    }
-  }
-  energy_ = 0xffffffff;
-  return *this;
-}
-
-inline AudioFrame& AudioFrame::operator-=(const AudioFrame& rhs) {
-  // Sanity check
-  assert((num_channels_ > 0) && (num_channels_ < 3));
-  assert(interleaved_ == rhs.interleaved_);
-  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
-
-  if ((samples_per_channel_ != rhs.samples_per_channel_) ||
-      (num_channels_ != rhs.num_channels_)) {
-    return *this;
-  }
-  if ((vad_activity_ != kVadPassive) || rhs.vad_activity_ != kVadPassive) {
-    vad_activity_ = kVadUnknown;
-  }
-  speech_type_ = kUndefined;
-
-  for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
-    int32_t wrap_guard =
-        static_cast<int32_t>(data_[i]) - static_cast<int32_t>(rhs.data_[i]);
-    data_[i] = ClampToInt16(wrap_guard);
-  }
-  energy_ = 0xffffffff;
-  return *this;
-}
-
-inline bool IsNewerSequenceNumber(uint16_t sequence_number,
-                                  uint16_t prev_sequence_number) {
-  // Distinguish between elements that are exactly 0x8000 apart.
-  // If s1>s2 and |s1-s2| = 0x8000: IsNewer(s1,s2)=true, IsNewer(s2,s1)=false
-  // rather than having IsNewer(s1,s2) = IsNewer(s2,s1) = false.
-  if (static_cast<uint16_t>(sequence_number - prev_sequence_number) == 0x8000) {
-    return sequence_number > prev_sequence_number;
-  }
-  return sequence_number != prev_sequence_number &&
-         static_cast<uint16_t>(sequence_number - prev_sequence_number) < 0x8000;
-}
-
-inline bool IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) {
-  // Distinguish between elements that are exactly 0x80000000 apart.
-  // If t1>t2 and |t1-t2| = 0x80000000: IsNewer(t1,t2)=true,
-  // IsNewer(t2,t1)=false
-  // rather than having IsNewer(t1,t2) = IsNewer(t2,t1) = false.
-  if (static_cast<uint32_t>(timestamp - prev_timestamp) == 0x80000000) {
-    return timestamp > prev_timestamp;
-  }
-  return timestamp != prev_timestamp &&
-         static_cast<uint32_t>(timestamp - prev_timestamp) < 0x80000000;
-}
-
-inline uint16_t LatestSequenceNumber(uint16_t sequence_number1,
-                                     uint16_t sequence_number2) {
-  return IsNewerSequenceNumber(sequence_number1, sequence_number2)
-             ? sequence_number1
-             : sequence_number2;
-}
-
-inline uint32_t LatestTimestamp(uint32_t timestamp1, uint32_t timestamp2) {
-  return IsNewerTimestamp(timestamp1, timestamp2) ? timestamp1 : timestamp2;
-}
-
-// Utility class to unwrap a sequence number to a larger type, for easier
-// handling large ranges. Note that sequence numbers will never be unwrapped
-// to a negative value.
-class SequenceNumberUnwrapper {
- public:
-  SequenceNumberUnwrapper() : last_seq_(-1) {}
-
-  // Get the unwrapped sequence, but don't update the internal state.
-  int64_t UnwrapWithoutUpdate(uint16_t sequence_number) {
-    if (last_seq_ == -1)
-      return sequence_number;
-
-    uint16_t cropped_last = static_cast<uint16_t>(last_seq_);
-    int64_t delta = sequence_number - cropped_last;
-    if (IsNewerSequenceNumber(sequence_number, cropped_last)) {
-      if (delta < 0)
-        delta += (1 << 16);  // Wrap forwards.
-    } else if (delta > 0 && (last_seq_ + delta - (1 << 16)) >= 0) {
-      // If sequence_number is older but delta is positive, this is a backwards
-      // wrap-around. However, don't wrap backwards past 0 (unwrapped).
-      delta -= (1 << 16);
-    }
-
-    return last_seq_ + delta;
-  }
-
-  // Only update the internal state to the specified last (unwrapped) sequence.
-  void UpdateLast(int64_t last_sequence) { last_seq_ = last_sequence; }
-
-  // Unwrap the sequence number and update the internal state.
-  int64_t Unwrap(uint16_t sequence_number) {
-    int64_t unwrapped = UnwrapWithoutUpdate(sequence_number);
-    UpdateLast(unwrapped);
-    return unwrapped;
-  }
-
- private:
-  int64_t last_seq_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
diff --git a/webrtc/modules/media_file/interface/media_file.h b/webrtc/modules/media_file/interface/media_file.h
deleted file mode 100644
index 3be5f08251..0000000000
--- a/webrtc/modules/media_file/interface/media_file.h
+++ /dev/null
@@ -1,182 +0,0 @@
-/*
- *  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_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
-
-#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include")
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/include/module.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/media_file/include/media_file_defines.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-class MediaFile : public Module
-{
-public:
-    // Factory method. Constructor disabled. id is the identifier for the
-    // MediaFile instance.
-    static MediaFile* CreateMediaFile(const int32_t id);
-    static void DestroyMediaFile(MediaFile* module);
-
-    // Put 10-60ms of audio data from file into the audioBuffer depending on
-    // codec frame size. dataLengthInBytes is both an input and output
-    // parameter. As input parameter it indicates the size of audioBuffer.
-    // As output parameter it indicates the number of bytes written to
-    // audioBuffer.
-    // Note: This API only play mono audio but can be used on file containing
-    // audio with more channels (in which case the audio will be converted to
-    // mono).
-    virtual int32_t PlayoutAudioData(
-        int8_t* audioBuffer,
-        size_t& dataLengthInBytes) = 0;
-
-    // Put 10-60ms, depending on codec frame size, of audio data from file into
-    // audioBufferLeft and audioBufferRight. The buffers contain the left and
-    // right channel of played out stereo audio.
-    // dataLengthInBytes is both an input and output parameter. As input
-    // parameter it indicates the size of both audioBufferLeft and
-    // audioBufferRight. As output parameter it indicates the number of bytes
-    // written to both audio buffers.
-    // Note: This API can only be successfully called for WAV files with stereo
-    // audio.
-    virtual int32_t PlayoutStereoData(
-        int8_t* audioBufferLeft,
-        int8_t* audioBufferRight,
-        size_t& dataLengthInBytes) = 0;
-
-    // Open the file specified by fileName (relative path is allowed) for
-    // reading. FileCallback::PlayNotification(..) will be called after
-    // notificationTimeMs of the file has been played if notificationTimeMs is
-    // greater than zero. If loop is true the file will be played until
-    // StopPlaying() is called. When end of file is reached the file is read
-    // from the start. format specifies the type of file fileName refers to.
-    // codecInst specifies the encoding of the audio data. Note that
-    // file formats that contain this information (like WAV files) don't need to
-    // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
-    // specify what part of the file should be read. From startPointMs ms to
-    // stopPointMs ms.
-    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
-    // mono). Stereo audio is only supported for WAV files.
-    virtual int32_t StartPlayingAudioFile(
-        const char* fileName,
-        const uint32_t notificationTimeMs = 0,
-        const bool loop                         = false,
-        const FileFormats format                = kFileFormatPcm16kHzFile,
-        const CodecInst* codecInst              = NULL,
-        const uint32_t startPointMs       = 0,
-        const uint32_t stopPointMs        = 0) = 0;
-
-    // Prepare for playing audio from stream.
-    // FileCallback::PlayNotification(..) will be called after
-    // notificationTimeMs of the file has been played if notificationTimeMs is
-    // greater than zero. format specifies the type of file fileName refers to.
-    // codecInst specifies the encoding of the audio data. Note that
-    // file formats that contain this information (like WAV files) don't need to
-    // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
-    // specify what part of the file should be read. From startPointMs ms to
-    // stopPointMs ms.
-    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
-    // mono). Stereo audio is only supported for WAV files.
-    virtual int32_t StartPlayingAudioStream(
-        InStream& stream,
-        const uint32_t notificationTimeMs = 0,
-        const FileFormats    format             = kFileFormatPcm16kHzFile,
-        const CodecInst*     codecInst          = NULL,
-        const uint32_t startPointMs       = 0,
-        const uint32_t stopPointMs        = 0) = 0;
-
-    // Stop playing from file or stream.
-    virtual int32_t StopPlaying() = 0;
-
-    // Return true if playing.
-    virtual bool IsPlaying() = 0;
-
-
-    // Set durationMs to the number of ms that has been played from file.
-    virtual int32_t PlayoutPositionMs(
-        uint32_t& durationMs) const = 0;
-
-    // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
-    // to file. The audio frame size is determined by the codecInst.pacsize
-    // parameter of the last sucessfull StartRecordingAudioFile(..) call.
-    // Note: bufferLength must be exactly one frame.
-    virtual int32_t IncomingAudioData(
-        const int8_t* audioBuffer,
-        const size_t bufferLength) = 0;
-
-    // Open/creates file specified by fileName for writing (relative path is
-    // allowed). FileCallback::RecordNotification(..) will be called after
-    // notificationTimeMs of audio data has been recorded if
-    // notificationTimeMs is greater than zero.
-    // format specifies the type of file that should be created/opened.
-    // codecInst specifies the encoding of the audio data. maxSizeBytes
-    // specifies the number of bytes allowed to be written to file if it is
-    // greater than zero.
-    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
-    // mono). Stereo is only supported for WAV files.
-    virtual int32_t StartRecordingAudioFile(
-        const char*  fileName,
-        const FileFormats    format,
-        const CodecInst&     codecInst,
-        const uint32_t notificationTimeMs = 0,
-        const uint32_t maxSizeBytes       = 0) = 0;
-
-    // Prepare for recording audio to stream.
-    // FileCallback::RecordNotification(..) will be called after
-    // notificationTimeMs of audio data has been recorded if
-    // notificationTimeMs is greater than zero.
-    // format specifies the type of file that stream should correspond to.
-    // codecInst specifies the encoding of the audio data.
-    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
-    // mono). Stereo is only supported for WAV files.
-    virtual int32_t StartRecordingAudioStream(
-        OutStream&           stream,
-        const FileFormats    format,
-        const CodecInst&     codecInst,
-        const uint32_t notificationTimeMs = 0) = 0;
-
-    // Stop recording to file or stream.
-    virtual int32_t StopRecording() = 0;
-
-    // Return true if recording.
-    virtual bool IsRecording() = 0;
-
-    // Set durationMs to the number of ms that has been recorded to file.
-    virtual int32_t RecordDurationMs(uint32_t& durationMs) = 0;
-
-    // Return true if recording or playing is stereo.
-    virtual bool IsStereo() = 0;
-
-    // Register callback to receive media file related notifications. Disables
-    // callbacks if callback is NULL.
-    virtual int32_t SetModuleFileCallback(FileCallback* callback) = 0;
-
-    // Set durationMs to the size of the file (in ms) specified by fileName.
-    // format specifies the type of file fileName refers to. freqInHz specifies
-    // the sampling frequency of the file.
-    virtual int32_t FileDurationMs(
-        const char*  fileName,
-        uint32_t&      durationMs,
-        const FileFormats    format,
-        const uint32_t freqInHz = 16000) = 0;
-
-    // Update codecInst according to the current audio codec being used for
-    // reading or writing.
-    virtual int32_t codec_info(CodecInst& codecInst) const = 0;
-
-protected:
-    MediaFile() {}
-    virtual ~MediaFile() {}
-};
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_H_
diff --git a/webrtc/modules/media_file/interface/media_file_defines.h b/webrtc/modules/media_file/interface/media_file_defines.h
deleted file mode 100644
index 345fe547b2..0000000000
--- a/webrtc/modules/media_file/interface/media_file_defines.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  Copyright (c) 2011 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_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
-
-#pragma message("WARNING: media_file/interface is DEPRECATED; use media_file/include")
-
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-// Callback class for the MediaFile class.
-class FileCallback
-{
-public:
-    virtual ~FileCallback(){}
-
-    // This function is called by MediaFile when a file has been playing for
-    // durationMs ms. id is the identifier for the MediaFile instance calling
-    // the callback.
-    virtual void PlayNotification(const int32_t id,
-                                  const uint32_t durationMs) = 0;
-
-    // This function is called by MediaFile when a file has been recording for
-    // durationMs ms. id is the identifier for the MediaFile instance calling
-    // the callback.
-    virtual void RecordNotification(const int32_t id,
-                                    const uint32_t durationMs) = 0;
-
-    // This function is called by MediaFile when a file has been stopped
-    // playing. id is the identifier for the MediaFile instance calling the
-    // callback.
-    virtual void PlayFileEnded(const int32_t id) = 0;
-
-    // This function is called by MediaFile when a file has been stopped
-    // recording. id is the identifier for the MediaFile instance calling the
-    // callback.
-    virtual void RecordFileEnded(const int32_t id) = 0;
-
-protected:
-    FileCallback() {}
-};
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INCLUDE_MEDIA_FILE_DEFINES_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h b/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
deleted file mode 100644
index ec7ec399d9..0000000000
--- a/webrtc/modules/rtp_rtcp/interface/fec_receiver.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- *  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_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include")
-
-#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-struct FecPacketCounter {
-  FecPacketCounter()
-      : num_packets(0),
-        num_fec_packets(0),
-        num_recovered_packets(0) {}
-
-  size_t num_packets;            // Number of received packets.
-  size_t num_fec_packets;        // Number of received FEC packets.
-  size_t num_recovered_packets;  // Number of recovered media packets using FEC.
-};
-
-class FecReceiver {
- public:
-  static FecReceiver* Create(RtpData* callback);
-
-  virtual ~FecReceiver() {}
-
-  virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
-                                       const uint8_t* incoming_rtp_packet,
-                                       size_t packet_length,
-                                       uint8_t ulpfec_payload_type) = 0;
-
-  virtual int32_t ProcessReceivedFec() = 0;
-
-  virtual FecPacketCounter GetPacketCounter() const = 0;
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_FEC_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h b/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
deleted file mode 100644
index 183eb2e98a..0000000000
--- a/webrtc/modules/rtp_rtcp/interface/receive_statistics.h
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- *  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_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include <map>
-
-#include "webrtc/modules/include/module.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class Clock;
-
-class StreamStatistician {
- public:
-  virtual ~StreamStatistician();
-
-  virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0;
-  virtual void GetDataCounters(size_t* bytes_received,
-                               uint32_t* packets_received) const = 0;
-
-  // Gets received stream data counters (includes reset counter values).
-  virtual void GetReceiveStreamDataCounters(
-      StreamDataCounters* data_counters) const = 0;
-
-  virtual uint32_t BitrateReceived() const = 0;
-
-  // Returns true if the packet with RTP header |header| is likely to be a
-  // retransmitted packet, false otherwise.
-  virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
-                                       int64_t min_rtt) const = 0;
-
-  // Returns true if |sequence_number| is received in order, false otherwise.
-  virtual bool IsPacketInOrder(uint16_t sequence_number) const = 0;
-};
-
-typedef std::map<uint32_t, StreamStatistician*> StatisticianMap;
-
-class ReceiveStatistics : public Module {
- public:
-  virtual ~ReceiveStatistics() {}
-
-  static ReceiveStatistics* Create(Clock* clock);
-
-  // Updates the receive statistics with this packet.
-  virtual void IncomingPacket(const RTPHeader& rtp_header,
-                              size_t packet_length,
-                              bool retransmitted) = 0;
-
-  // Increment counter for number of FEC packets received.
-  virtual void FecPacketReceived(const RTPHeader& header,
-                                 size_t packet_length) = 0;
-
-  // Returns a map of all statisticians which have seen an incoming packet
-  // during the last two seconds.
-  virtual StatisticianMap GetActiveStatisticians() const = 0;
-
-  // Returns a pointer to the statistician of an ssrc.
-  virtual StreamStatistician* GetStatistician(uint32_t ssrc) const = 0;
-
-  // Sets the max reordering threshold in number of packets.
-  virtual void SetMaxReorderingThreshold(int max_reordering_threshold) = 0;
-
-  // Called on new RTCP stats creation.
-  virtual void RegisterRtcpStatisticsCallback(
-      RtcpStatisticsCallback* callback) = 0;
-
-  // Called on new RTP stats creation.
-  virtual void RegisterRtpStatisticsCallback(
-      StreamDataCountersCallback* callback) = 0;
-};
-
-class NullReceiveStatistics : public ReceiveStatistics {
- public:
-  void IncomingPacket(const RTPHeader& rtp_header,
-                      size_t packet_length,
-                      bool retransmitted) override;
-  void FecPacketReceived(const RTPHeader& header,
-                         size_t packet_length) override;
-  StatisticianMap GetActiveStatisticians() const override;
-  StreamStatistician* GetStatistician(uint32_t ssrc) const override;
-  int64_t TimeUntilNextProcess() override;
-  int32_t Process() override;
-  void SetMaxReorderingThreshold(int max_reordering_threshold) override;
-  void RegisterRtcpStatisticsCallback(
-      RtcpStatisticsCallback* callback) override;
-  void RegisterRtpStatisticsCallback(
-      StreamDataCountersCallback* callback) override;
-};
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RECEIVE_STATISTICS_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h b/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
deleted file mode 100644
index d271c9bbd5..0000000000
--- a/webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  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_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
-
-namespace webrtc {
-
-class Clock;
-class TimestampExtrapolator;
-
-// RemoteNtpTimeEstimator can be used to estimate a given RTP timestamp's NTP
-// time in local timebase.
-// Note that it needs to be trained with at least 2 RTCP SR (by calling
-// |UpdateRtcpTimestamp|) before it can be used.
-class RemoteNtpTimeEstimator {
- public:
-  explicit RemoteNtpTimeEstimator(Clock* clock);
-
-  ~RemoteNtpTimeEstimator();
-
-  // Updates the estimator with round trip time |rtt|, NTP seconds |ntp_secs|,
-  // NTP fraction |ntp_frac| and RTP timestamp |rtcp_timestamp|.
-  bool UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac,
-                           uint32_t rtp_timestamp);
-
-  // Estimates the NTP timestamp in local timebase from |rtp_timestamp|.
-  // Returns the NTP timestamp in ms when success. -1 if failed.
-  int64_t Estimate(uint32_t rtp_timestamp);
-
- private:
-  Clock* clock_;
-  rtc::scoped_ptr<TimestampExtrapolator> ts_extrapolator_;
-  RtcpList rtcp_list_;
-  int64_t last_timing_log_ms_;
-  RTC_DISALLOW_COPY_AND_ASSIGN(RemoteNtpTimeEstimator);
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_REMOTE_NTP_TIME_ESTIMATOR_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h b/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
deleted file mode 100644
index 6d942cbc92..0000000000
--- a/webrtc/modules/rtp_rtcp/interface/rtp_cvo.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- *  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.
- */
-#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include "webrtc/common_video/rotation.h"
-
-namespace webrtc {
-
-// Please refer to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/
-// 12.07.00_60/ts_126114v120700p.pdf Section 7.4.5. The rotation of a frame is
-// the clockwise angle the frames must be rotated in order to display the frames
-// correctly if the display is rotated in its natural orientation.
-inline uint8_t ConvertVideoRotationToCVOByte(VideoRotation rotation) {
-  switch (rotation) {
-    case kVideoRotation_0:
-      return 0;
-    case kVideoRotation_90:
-      return 1;
-    case kVideoRotation_180:
-      return 2;
-    case kVideoRotation_270:
-      return 3;
-  }
-  assert(false);
-  return 0;
-}
-
-inline VideoRotation ConvertCVOByteToVideoRotation(uint8_t rotation) {
-  switch (rotation) {
-    case 0:
-      return kVideoRotation_0;
-    case 1:
-      return kVideoRotation_90;
-    case 2:
-      return kVideoRotation_180;
-      break;
-    case 3:
-      return kVideoRotation_270;
-    default:
-      assert(false);
-      return kVideoRotation_0;
-  }
-}
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_CVO_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h b/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
deleted file mode 100644
index 9b2cabc546..0000000000
--- a/webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- *  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_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-struct RTPHeader;
-
-class RtpHeaderParser {
- public:
-  static RtpHeaderParser* Create();
-  virtual ~RtpHeaderParser() {}
-
-  // Returns true if the packet is an RTCP packet, false otherwise.
-  static bool IsRtcp(const uint8_t* packet, size_t length);
-
-  // Parses the packet and stores the parsed packet in |header|. Returns true on
-  // success, false otherwise.
-  // This method is thread-safe in the sense that it can parse multiple packets
-  // at once.
-  virtual bool Parse(const uint8_t* packet,
-                     size_t length,
-                     RTPHeader* header) const = 0;
-
-  // Registers an RTP header extension and binds it to |id|.
-  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type,
-                                          uint8_t id) = 0;
-
-  // De-registers an RTP header extension.
-  virtual bool DeregisterRtpHeaderExtension(RTPExtensionType type) = 0;
-};
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h b/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
deleted file mode 100644
index 4994eb50a2..0000000000
--- a/webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- *  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_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-
-namespace webrtc {
-
-// This strategy deals with the audio/video-specific aspects
-// of payload handling.
-class RTPPayloadStrategy {
- public:
-  virtual ~RTPPayloadStrategy() {}
-
-  virtual bool CodecsMustBeUnique() const = 0;
-
-  virtual bool PayloadIsCompatible(const RtpUtility::Payload& payload,
-                                   const uint32_t frequency,
-                                   const uint8_t channels,
-                                   const uint32_t rate) const = 0;
-
-  virtual void UpdatePayloadRate(RtpUtility::Payload* payload,
-                                 const uint32_t rate) const = 0;
-
-  virtual RtpUtility::Payload* CreatePayloadType(
-      const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-      const int8_t payloadType,
-      const uint32_t frequency,
-      const uint8_t channels,
-      const uint32_t rate) const = 0;
-
-  virtual int GetPayloadTypeFrequency(
-      const RtpUtility::Payload& payload) const = 0;
-
-  static RTPPayloadStrategy* CreateStrategy(const bool handling_audio);
-
- protected:
-  RTPPayloadStrategy() {}
-};
-
-class RTPPayloadRegistry {
- public:
-  // The registry takes ownership of the strategy.
-  RTPPayloadRegistry(RTPPayloadStrategy* rtp_payload_strategy);
-  ~RTPPayloadRegistry();
-
-  int32_t RegisterReceivePayload(
-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-      const int8_t payload_type,
-      const uint32_t frequency,
-      const uint8_t channels,
-      const uint32_t rate,
-      bool* created_new_payload_type);
-
-  int32_t DeRegisterReceivePayload(
-      const int8_t payload_type);
-
-  int32_t ReceivePayloadType(
-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-      const uint32_t frequency,
-      const uint8_t channels,
-      const uint32_t rate,
-      int8_t* payload_type) const;
-
-  bool RtxEnabled() const;
-
-  void SetRtxSsrc(uint32_t ssrc);
-
-  bool GetRtxSsrc(uint32_t* ssrc) const;
-
-  void SetRtxPayloadType(int payload_type, int associated_payload_type);
-
-  bool IsRtx(const RTPHeader& header) const;
-
-  // DEPRECATED. Use RestoreOriginalPacket below that takes a uint8_t*
-  // restored_packet, instead of a uint8_t**.
-  // TODO(noahric): Remove this when all callers have been updated.
-  bool RestoreOriginalPacket(uint8_t** restored_packet,
-                             const uint8_t* packet,
-                             size_t* packet_length,
-                             uint32_t original_ssrc,
-                             const RTPHeader& header) const;
-
-  bool RestoreOriginalPacket(uint8_t* restored_packet,
-                             const uint8_t* packet,
-                             size_t* packet_length,
-                             uint32_t original_ssrc,
-                             const RTPHeader& header) const;
-
-  bool IsRed(const RTPHeader& header) const;
-
-  // Returns true if the media of this RTP packet is encapsulated within an
-  // extra header, such as RTX or RED.
-  bool IsEncapsulated(const RTPHeader& header) const;
-
-  bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
-
-  int GetPayloadTypeFrequency(uint8_t payload_type) const;
-
-  bool PayloadTypeToPayload(const uint8_t payload_type,
-                            RtpUtility::Payload*& payload) const;
-
-  void ResetLastReceivedPayloadTypes() {
-    CriticalSectionScoped cs(crit_sect_.get());
-    last_received_payload_type_ = -1;
-    last_received_media_payload_type_ = -1;
-  }
-
-  // This sets the payload type of the packets being received from the network
-  // on the media SSRC. For instance if packets are encapsulated with RED, this
-  // payload type will be the RED payload type.
-  void SetIncomingPayloadType(const RTPHeader& header);
-
-  // Returns true if the new media payload type has not changed.
-  bool ReportMediaPayloadType(uint8_t media_payload_type);
-
-  int8_t red_payload_type() const {
-    CriticalSectionScoped cs(crit_sect_.get());
-    return red_payload_type_;
-  }
-  int8_t ulpfec_payload_type() const {
-    CriticalSectionScoped cs(crit_sect_.get());
-    return ulpfec_payload_type_;
-  }
-  int8_t last_received_payload_type() const {
-    CriticalSectionScoped cs(crit_sect_.get());
-    return last_received_payload_type_;
-  }
-  void set_last_received_payload_type(int8_t last_received_payload_type) {
-    CriticalSectionScoped cs(crit_sect_.get());
-    last_received_payload_type_ = last_received_payload_type;
-  }
-
-  int8_t last_received_media_payload_type() const {
-    CriticalSectionScoped cs(crit_sect_.get());
-    return last_received_media_payload_type_;
-  };
-
-  bool use_rtx_payload_mapping_on_restore() const {
-    CriticalSectionScoped cs(crit_sect_.get());
-    return use_rtx_payload_mapping_on_restore_;
-  }
-
-  void set_use_rtx_payload_mapping_on_restore(bool val) {
-    CriticalSectionScoped cs(crit_sect_.get());
-    use_rtx_payload_mapping_on_restore_ = val;
-  }
-
- private:
-  // Prunes the payload type map of the specific payload type, if it exists.
-  void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-      const size_t payload_name_length,
-      const uint32_t frequency,
-      const uint8_t channels,
-      const uint32_t rate);
-
-  bool IsRtxInternal(const RTPHeader& header) const;
-
-  rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
-  RtpUtility::PayloadTypeMap payload_type_map_;
-  rtc::scoped_ptr<RTPPayloadStrategy> rtp_payload_strategy_;
-  int8_t  red_payload_type_;
-  int8_t ulpfec_payload_type_;
-  int8_t incoming_payload_type_;
-  int8_t  last_received_payload_type_;
-  int8_t  last_received_media_payload_type_;
-  bool rtx_;
-  // TODO(changbin): Remove rtx_payload_type_ once interop with old clients that
-  // only understand one RTX PT is no longer needed.
-  int rtx_payload_type_;
-  // Mapping rtx_payload_type_map_[rtx] = associated.
-  std::map<int, int> rtx_payload_type_map_;
-  // When true, use rtx_payload_type_map_ when restoring RTX packets to get the
-  // correct payload type.
-  bool use_rtx_payload_mapping_on_restore_;
-  uint32_t ssrc_rtx_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h b/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
deleted file mode 100644
index db5f46e430..0000000000
--- a/webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- *  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_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class RTPPayloadRegistry;
-
-class TelephoneEventHandler {
- public:
-  virtual ~TelephoneEventHandler() {}
-
-  // The following three methods implement the TelephoneEventHandler interface.
-  // Forward DTMFs to decoder for playout.
-  virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
-
-  // Is forwarding of outband telephone events turned on/off?
-  virtual bool TelephoneEventForwardToDecoder() const = 0;
-
-  // Is TelephoneEvent configured with payload type payload_type
-  virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
-};
-
-class RtpReceiver {
- public:
-  // Creates a video-enabled RTP receiver.
-  static RtpReceiver* CreateVideoReceiver(
-      Clock* clock,
-      RtpData* incoming_payload_callback,
-      RtpFeedback* incoming_messages_callback,
-      RTPPayloadRegistry* rtp_payload_registry);
-
-  // Creates an audio-enabled RTP receiver.
-  static RtpReceiver* CreateAudioReceiver(
-      Clock* clock,
-      RtpAudioFeedback* incoming_audio_feedback,
-      RtpData* incoming_payload_callback,
-      RtpFeedback* incoming_messages_callback,
-      RTPPayloadRegistry* rtp_payload_registry);
-
-  virtual ~RtpReceiver() {}
-
-  // Returns a TelephoneEventHandler if available.
-  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
-
-  // Registers a receive payload in the payload registry and notifies the media
-  // receiver strategy.
-  virtual int32_t RegisterReceivePayload(
-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
-      const int8_t payload_type,
-      const uint32_t frequency,
-      const uint8_t channels,
-      const uint32_t rate) = 0;
-
-  // De-registers |payload_type| from the payload registry.
-  virtual int32_t DeRegisterReceivePayload(const int8_t payload_type) = 0;
-
-  // Parses the media specific parts of an RTP packet and updates the receiver
-  // state. This for instance means that any changes in SSRC and payload type is
-  // detected and acted upon.
-  virtual bool IncomingRtpPacket(const RTPHeader& rtp_header,
-                                 const uint8_t* payload,
-                                 size_t payload_length,
-                                 PayloadUnion payload_specific,
-                                 bool in_order) = 0;
-
-  // Returns the currently configured NACK method.
-  virtual NACKMethod NACK() const = 0;
-
-  // Turn negative acknowledgement (NACK) requests on/off.
-  virtual void SetNACKStatus(const NACKMethod method) = 0;
-
-  // Gets the last received timestamp. Returns true if a packet has been
-  // received, false otherwise.
-  virtual bool Timestamp(uint32_t* timestamp) const = 0;
-  // Gets the time in milliseconds when the last timestamp was received.
-  // Returns true if a packet has been received, false otherwise.
-  virtual bool LastReceivedTimeMs(int64_t* receive_time_ms) const = 0;
-
-  // Returns the remote SSRC of the currently received RTP stream.
-  virtual uint32_t SSRC() const = 0;
-
-  // Returns the current remote CSRCs.
-  virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
-
-  // Returns the current energy of the RTP stream received.
-  virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RECEIVER_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
deleted file mode 100644
index d004f67498..0000000000
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ /dev/null
@@ -1,643 +0,0 @@
-/*
- *  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_RTP_RTCP_INCLUDE_RTP_RTCP_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include <set>
-#include <vector>
-
-#include "webrtc/modules/include/module.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-
-namespace webrtc {
-// Forward declarations.
-class ReceiveStatistics;
-class RemoteBitrateEstimator;
-class RtpReceiver;
-class Transport;
-namespace rtcp {
-class TransportFeedback;
-}
-
-class RtpRtcp : public Module {
- public:
-  struct Configuration {
-    Configuration();
-
-   /*  id                   - Unique identifier of this RTP/RTCP module object
-    *  audio                - True for a audio version of the RTP/RTCP module
-    *                         object false will create a video version
-    *  clock                - The clock to use to read time. If NULL object
-    *                         will be using the system clock.
-    *  incoming_data        - Callback object that will receive the incoming
-    *                         data. May not be NULL; default callback will do
-    *                         nothing.
-    *  incoming_messages    - Callback object that will receive the incoming
-    *                         RTP messages. May not be NULL; default callback
-    *                         will do nothing.
-    *  outgoing_transport   - Transport object that will be called when packets
-    *                         are ready to be sent out on the network
-    *  intra_frame_callback - Called when the receiver request a intra frame.
-    *  bandwidth_callback   - Called when we receive a changed estimate from
-    *                         the receiver of out stream.
-    *  audio_messages       - Telephone events. May not be NULL; default
-    *                         callback will do nothing.
-    *  remote_bitrate_estimator - Estimates the bandwidth available for a set of
-    *                             streams from the same client.
-    *  paced_sender             - Spread any bursts of packets into smaller
-    *                             bursts to minimize packet loss.
-    */
-    bool audio;
-    bool receiver_only;
-    Clock* clock;
-    ReceiveStatistics* receive_statistics;
-    Transport* outgoing_transport;
-    RtcpIntraFrameObserver* intra_frame_callback;
-    RtcpBandwidthObserver* bandwidth_callback;
-    TransportFeedbackObserver* transport_feedback_callback;
-    RtcpRttStats* rtt_stats;
-    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
-    RtpAudioFeedback* audio_messages;
-    RemoteBitrateEstimator* remote_bitrate_estimator;
-    RtpPacketSender* paced_sender;
-    TransportSequenceNumberAllocator* transport_sequence_number_allocator;
-    BitrateStatisticsObserver* send_bitrate_observer;
-    FrameCountObserver* send_frame_count_observer;
-    SendSideDelayObserver* send_side_delay_observer;
-  };
-
-  /*
-   *   Create a RTP/RTCP module object using the system clock.
-   *
-   *   configuration  - Configuration of the RTP/RTCP module.
-   */
-  static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration);
-
-  /**************************************************************************
-   *
-   *   Receiver functions
-   *
-   ***************************************************************************/
-
-    virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet,
-                                       size_t incoming_packet_length) = 0;
-
-    virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
-
-    /**************************************************************************
-    *
-    *   Sender
-    *
-    ***************************************************************************/
-
-    /*
-    *   set MTU
-    *
-    *   size    -  Max transfer unit in bytes, default is 1500
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetMaxTransferUnit(uint16_t size) = 0;
-
-    /*
-    *   set transtport overhead
-    *   default is IPv4 and UDP with no encryption
-    *
-    *   TCP                     - true for TCP false UDP
-    *   IPv6                    - true for IP version 6 false for version 4
-    *   authenticationOverhead  - number of bytes to leave for an
-    *                             authentication header
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetTransportOverhead(
-        bool TCP,
-        bool IPV6,
-        uint8_t authenticationOverhead = 0) = 0;
-
-    /*
-    *   Get max payload length
-    *
-    *   A combination of the configuration MaxTransferUnit and
-    *   TransportOverhead.
-    *   Does not account FEC/ULP/RED overhead if FEC is enabled.
-    *   Does not account for RTP headers
-    */
-    virtual uint16_t MaxPayloadLength() const = 0;
-
-    /*
-    *   Get max data payload length
-    *
-    *   A combination of the configuration MaxTransferUnit, headers and
-    *   TransportOverhead.
-    *   Takes into account FEC/ULP/RED overhead if FEC is enabled.
-    *   Takes into account RTP headers
-    */
-    virtual uint16_t MaxDataPayloadLength() const = 0;
-
-    /*
-    *   set codec name and payload type
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RegisterSendPayload(
-        const CodecInst& voiceCodec) = 0;
-
-    /*
-    *   set codec name and payload type
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RegisterSendPayload(
-        const VideoCodec& videoCodec) = 0;
-
-    /*
-    *   Unregister a send payload
-    *
-    *   payloadType - payload type of codec
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0;
-
-   /*
-    *   (De)register RTP header extension type and id.
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
-                                                   uint8_t id) = 0;
-
-    virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;
-
-    /*
-    *   get start timestamp
-    */
-    virtual uint32_t StartTimestamp() const = 0;
-
-    /*
-    *   configure start timestamp, default is a random number
-    *
-    *   timestamp   - start timestamp
-    */
-    virtual void SetStartTimestamp(uint32_t timestamp) = 0;
-
-    /*
-    *   Get SequenceNumber
-    */
-    virtual uint16_t SequenceNumber() const = 0;
-
-    /*
-    *   Set SequenceNumber, default is a random number
-    */
-    virtual void SetSequenceNumber(uint16_t seq) = 0;
-
-    // Returns true if the ssrc matched this module, false otherwise.
-    virtual bool SetRtpStateForSsrc(uint32_t ssrc,
-                                    const RtpState& rtp_state) = 0;
-    virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
-
-    /*
-    *   Get SSRC
-    */
-    virtual uint32_t SSRC() const = 0;
-
-    /*
-    *   configure SSRC, default is a random number
-    */
-    virtual void SetSSRC(uint32_t ssrc) = 0;
-
-    /*
-    *   Set CSRC
-    *
-    *   csrcs   - vector of CSRCs
-    */
-    virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
-
-    /*
-    * Turn on/off sending RTX (RFC 4588). The modes can be set as a combination
-    * of values of the enumerator RtxMode.
-    */
-    virtual void SetRtxSendStatus(int modes) = 0;
-
-    /*
-    * Get status of sending RTX (RFC 4588). The returned value can be
-    * a combination of values of the enumerator RtxMode.
-    */
-    virtual int RtxSendStatus() const = 0;
-
-    // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX,
-    // only the SSRC is set.
-    virtual void SetRtxSsrc(uint32_t ssrc) = 0;
-
-    // Sets the payload type to use when sending RTX packets. Note that this
-    // doesn't enable RTX, only the payload type is set.
-    virtual void SetRtxSendPayloadType(int payload_type,
-                                       int associated_payload_type) = 0;
-
-    // Gets the payload type pair of (RTX, associated) to use when sending RTX
-    // packets.
-    virtual std::pair<int, int> RtxSendPayloadType() const = 0;
-
-    /*
-    *   sends kRtcpByeCode when going from true to false
-    *
-    *   sending - on/off
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetSendingStatus(bool sending) = 0;
-
-    /*
-    *   get send status
-    */
-    virtual bool Sending() const = 0;
-
-    /*
-    *   Starts/Stops media packets, on by default
-    *
-    *   sending - on/off
-    */
-    virtual void SetSendingMediaStatus(bool sending) = 0;
-
-    /*
-    *   get send status
-    */
-    virtual bool SendingMedia() const = 0;
-
-    /*
-    *   get sent bitrate in Kbit/s
-    */
-    virtual void BitrateSent(uint32_t* totalRate,
-                             uint32_t* videoRate,
-                             uint32_t* fecRate,
-                             uint32_t* nackRate) const = 0;
-
-    /*
-    *   Used by the codec module to deliver a video or audio frame for
-    *   packetization.
-    *
-    *   frameType       - type of frame to send
-    *   payloadType     - payload type of frame to send
-    *   timestamp       - timestamp of frame to send
-    *   payloadData     - payload buffer of frame to send
-    *   payloadSize     - size of payload buffer to send
-    *   fragmentation   - fragmentation offset data for fragmented frames such
-    *                     as layers or RED
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendOutgoingData(
-        FrameType frameType,
-        int8_t payloadType,
-        uint32_t timeStamp,
-        int64_t capture_time_ms,
-        const uint8_t* payloadData,
-        size_t payloadSize,
-        const RTPFragmentationHeader* fragmentation = NULL,
-        const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
-
-    virtual bool TimeToSendPacket(uint32_t ssrc,
-                                  uint16_t sequence_number,
-                                  int64_t capture_time_ms,
-                                  bool retransmission) = 0;
-
-    virtual size_t TimeToSendPadding(size_t bytes) = 0;
-
-    // Called on generation of new statistics after an RTP send.
-    virtual void RegisterSendChannelRtpStatisticsCallback(
-        StreamDataCountersCallback* callback) = 0;
-    virtual StreamDataCountersCallback*
-        GetSendChannelRtpStatisticsCallback() const = 0;
-
-    /**************************************************************************
-    *
-    *   RTCP
-    *
-    ***************************************************************************/
-
-    /*
-    *    Get RTCP status
-    */
-    virtual RtcpMode RTCP() const = 0;
-
-    /*
-    *   configure RTCP status i.e on(compound or non- compound)/off
-    *
-    *   method  - RTCP method to use
-    */
-    virtual void SetRTCPStatus(RtcpMode method) = 0;
-
-    /*
-    *   Set RTCP CName (i.e unique identifier)
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetCNAME(const char* c_name) = 0;
-
-    /*
-    *   Get remote CName
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoteCNAME(uint32_t remoteSSRC,
-                                char cName[RTCP_CNAME_SIZE]) const = 0;
-
-    /*
-    *   Get remote NTP
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoteNTP(
-        uint32_t *ReceivedNTPsecs,
-        uint32_t *ReceivedNTPfrac,
-        uint32_t *RTCPArrivalTimeSecs,
-        uint32_t *RTCPArrivalTimeFrac,
-        uint32_t *rtcp_timestamp) const  = 0;
-
-    /*
-    *   AddMixedCNAME
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0;
-
-    /*
-    *   RemoveMixedCNAME
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0;
-
-    /*
-    *   Get RoundTripTime
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RTT(uint32_t remoteSSRC,
-                        int64_t* RTT,
-                        int64_t* avgRTT,
-                        int64_t* minRTT,
-                        int64_t* maxRTT) const = 0;
-
-    /*
-    *   Force a send of a RTCP packet
-    *   periodic SR and RR are triggered via the process function
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0;
-
-    /*
-    *   Force a send of a RTCP packet with more than one packet type.
-    *   periodic SR and RR are triggered via the process function
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendCompoundRTCP(
-        const std::set<RTCPPacketType>& rtcpPacketTypes) = 0;
-
-    /*
-    *    Good state of RTP receiver inform sender
-    */
-    virtual int32_t SendRTCPReferencePictureSelection(
-        const uint64_t pictureID) = 0;
-
-    /*
-    *    Send a RTCP Slice Loss Indication (SLI)
-    *    6 least significant bits of pictureID
-    */
-    virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
-
-    /*
-    *   Statistics of the amount of data sent
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t DataCountersRTP(
-        size_t* bytesSent,
-        uint32_t* packetsSent) const = 0;
-
-    /*
-    *   Get send statistics for the RTP and RTX stream.
-    */
-    virtual void GetSendStreamDataCounters(
-        StreamDataCounters* rtp_counters,
-        StreamDataCounters* rtx_counters) const = 0;
-
-    /*
-     *  Get packet loss statistics for the RTP stream.
-     */
-    virtual void GetRtpPacketLossStats(
-        bool outgoing,
-        uint32_t ssrc,
-        struct RtpPacketLossStats* loss_stats) const = 0;
-
-    /*
-    *   Get received RTCP sender info
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
-
-    /*
-    *   Get received RTCP report block
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RemoteRTCPStat(
-        std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
-
-    /*
-    *   (APP) Application specific data
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType,
-                                                   uint32_t name,
-                                                   const uint8_t* data,
-                                                   uint16_t length) = 0;
-    /*
-    *   (XR) VOIP metric
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetRTCPVoIPMetrics(
-        const RTCPVoIPMetric* VoIPMetric) = 0;
-
-    /*
-    *   (XR) Receiver Reference Time Report
-    */
-    virtual void SetRtcpXrRrtrStatus(bool enable) = 0;
-
-    virtual bool RtcpXrRrtrStatus() const = 0;
-
-    /*
-    *  (REMB) Receiver Estimated Max Bitrate
-    */
-    virtual bool REMB() const = 0;
-
-    virtual void SetREMBStatus(bool enable) = 0;
-
-    virtual void SetREMBData(uint32_t bitrate,
-                             const std::vector<uint32_t>& ssrcs) = 0;
-
-    /*
-    *   (TMMBR) Temporary Max Media Bit Rate
-    */
-    virtual bool TMMBR() const = 0;
-
-    virtual void SetTMMBRStatus(bool enable) = 0;
-
-    /*
-    *   (NACK)
-    */
-
-    /*
-     *  TODO(holmer): Propagate this API to VideoEngine.
-     *  Returns the currently configured selective retransmission settings.
-     */
-    virtual int SelectiveRetransmissions() const = 0;
-
-    /*
-     *  TODO(holmer): Propagate this API to VideoEngine.
-     *  Sets the selective retransmission settings, which will decide which
-     *  packets will be retransmitted if NACKed. Settings are constructed by
-     *  combining the constants in enum RetransmissionMode with bitwise OR.
-     *  All packets are retransmitted if kRetransmitAllPackets is set, while no
-     *  packets are retransmitted if kRetransmitOff is set.
-     *  By default all packets except FEC packets are retransmitted. For VP8
-     *  with temporal scalability only base layer packets are retransmitted.
-     *
-     *  Returns -1 on failure, otherwise 0.
-     */
-    virtual int SetSelectiveRetransmissions(uint8_t settings) = 0;
-
-    /*
-    *   Send a Negative acknowledgement packet
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendNACK(const uint16_t* nackList, uint16_t size) = 0;
-
-    /*
-    *   Store the sent packets, needed to answer to a Negative acknowledgement
-    *   requests
-    */
-    virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
-
-    // Returns true if the module is configured to store packets.
-    virtual bool StorePackets() const = 0;
-
-    // Called on receipt of RTCP report block from remote side.
-    virtual void RegisterRtcpStatisticsCallback(
-        RtcpStatisticsCallback* callback) = 0;
-    virtual RtcpStatisticsCallback*
-        GetRtcpStatisticsCallback() = 0;
-    // BWE feedback packets.
-    virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0;
-
-    /**************************************************************************
-    *
-    *   Audio
-    *
-    ***************************************************************************/
-
-    /*
-    *   set audio packet size, used to determine when it's time to send a DTMF
-    *   packet in silence (CNG)
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0;
-
-    /*
-    *   Send a TelephoneEvent tone using RFC 2833 (4733)
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SendTelephoneEventOutband(uint8_t key,
-                                              uint16_t time_ms,
-                                              uint8_t level) = 0;
-
-    /*
-    *   Set payload type for Redundant Audio Data RFC 2198
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0;
-
-    /*
-    *   Get payload type for Redundant Audio Data RFC 2198
-    *
-    *   return -1 on failure else 0
-    */
-     virtual int32_t SendREDPayloadType(
-         int8_t& payloadType) const = 0;
-
-     /*
-     * Store the audio level in dBov for header-extension-for-audio-level-
-     * indication.
-     * This API shall be called before transmision of an RTP packet to ensure
-     * that the |level| part of the extended RTP header is updated.
-     *
-     * return -1 on failure else 0.
-     */
-     virtual int32_t SetAudioLevel(uint8_t level_dBov) = 0;
-
-    /**************************************************************************
-    *
-    *   Video
-    *
-    ***************************************************************************/
-
-    /*
-    *   Set the target send bitrate
-    */
-    virtual void SetTargetSendBitrate(uint32_t bitrate_bps) = 0;
-
-    /*
-    *   Turn on/off generic FEC
-    */
-    virtual void SetGenericFECStatus(bool enable,
-                                     uint8_t payload_type_red,
-                                     uint8_t payload_type_fec) = 0;
-
-    /*
-    *   Get generic FEC setting
-    */
-    virtual void GenericFECStatus(bool& enable,
-                                     uint8_t& payloadTypeRED,
-                                     uint8_t& payloadTypeFEC) = 0;
-
-
-    virtual int32_t SetFecParameters(
-        const FecProtectionParams* delta_params,
-        const FecProtectionParams* key_params) = 0;
-
-    /*
-    *   Set method for requestion a new key frame
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
-
-    /*
-    *   send a request for a keyframe
-    *
-    *   return -1 on failure else 0
-    */
-    virtual int32_t RequestKeyFrame() = 0;
-};
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
deleted file mode 100644
index ac3954ba9b..0000000000
--- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ /dev/null
@@ -1,442 +0,0 @@
-/*
- *  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_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
-
-#pragma message("WARNING: rtp_rtcp/interface is DEPRECATED; use include dir.")
-
-#include <stddef.h>
-#include <list>
-
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/typedefs.h"
-
-#define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
-#define IP_PACKET_SIZE 1500    // we assume ethernet
-#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
-#define TIMEOUT_SEI_MESSAGES_MS 30000   // in milliseconds
-
-namespace webrtc {
-namespace rtcp {
-class TransportFeedback;
-}
-
-const int kVideoPayloadTypeFrequency = 90000;
-
-// Minimum RTP header size in bytes.
-const uint8_t kRtpHeaderSize = 12;
-
-struct AudioPayload
-{
-    uint32_t    frequency;
-    uint8_t     channels;
-    uint32_t    rate;
-};
-
-struct VideoPayload
-{
-    RtpVideoCodecTypes   videoCodecType;
-    uint32_t       maxRate;
-};
-
-union PayloadUnion
-{
-    AudioPayload Audio;
-    VideoPayload Video;
-};
-
-enum RTPAliveType
-{
-    kRtpDead   = 0,
-    kRtpNoRtp = 1,
-    kRtpAlive  = 2
-};
-
-enum ProtectionType {
-  kUnprotectedPacket,
-  kProtectedPacket
-};
-
-enum StorageType {
-  kDontRetransmit,
-  kAllowRetransmission
-};
-
-enum RTPExtensionType {
-  kRtpExtensionNone,
-  kRtpExtensionTransmissionTimeOffset,
-  kRtpExtensionAudioLevel,
-  kRtpExtensionAbsoluteSendTime,
-  kRtpExtensionVideoRotation,
-  kRtpExtensionTransportSequenceNumber,
-};
-
-enum RTCPAppSubTypes
-{
-    kAppSubtypeBwe     = 0x00
-};
-
-// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
-enum RTCPPacketType : uint32_t {
-  kRtcpReport = 0x0001,
-  kRtcpSr = 0x0002,
-  kRtcpRr = 0x0004,
-  kRtcpSdes = 0x0008,
-  kRtcpBye = 0x0010,
-  kRtcpPli = 0x0020,
-  kRtcpNack = 0x0040,
-  kRtcpFir = 0x0080,
-  kRtcpTmmbr = 0x0100,
-  kRtcpTmmbn = 0x0200,
-  kRtcpSrReq = 0x0400,
-  kRtcpXrVoipMetric = 0x0800,
-  kRtcpApp = 0x1000,
-  kRtcpSli = 0x4000,
-  kRtcpRpsi = 0x8000,
-  kRtcpRemb = 0x10000,
-  kRtcpTransmissionTimeOffset = 0x20000,
-  kRtcpXrReceiverReferenceTime = 0x40000,
-  kRtcpXrDlrrReportBlock = 0x80000,
-  kRtcpTransportFeedback = 0x100000,
-};
-
-enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };
-
-enum RtpRtcpPacketType
-{
-    kPacketRtp        = 0,
-    kPacketKeepAlive = 1
-};
-
-enum NACKMethod
-{
-    kNackOff      = 0,
-    kNackRtcp     = 2
-};
-
-enum RetransmissionMode : uint8_t {
-  kRetransmitOff = 0x0,
-  kRetransmitFECPackets = 0x1,
-  kRetransmitBaseLayer = 0x2,
-  kRetransmitHigherLayers = 0x4,
-  kRetransmitAllPackets = 0xFF
-};
-
-enum RtxMode {
-  kRtxOff                 = 0x0,
-  kRtxRetransmitted       = 0x1,  // Only send retransmissions over RTX.
-  kRtxRedundantPayloads   = 0x2   // Preventively send redundant payloads
-                                  // instead of padding.
-};
-
-const size_t kRtxHeaderSize = 2;
-
-struct RTCPSenderInfo
-{
-    uint32_t NTPseconds;
-    uint32_t NTPfraction;
-    uint32_t RTPtimeStamp;
-    uint32_t sendPacketCount;
-    uint32_t sendOctetCount;
-};
-
-struct RTCPReportBlock {
-  RTCPReportBlock()
-      : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0),
-        extendedHighSeqNum(0), jitter(0), lastSR(0),
-        delaySinceLastSR(0) {}
-
-  RTCPReportBlock(uint32_t remote_ssrc,
-                  uint32_t source_ssrc,
-                  uint8_t fraction_lost,
-                  uint32_t cumulative_lost,
-                  uint32_t extended_high_sequence_number,
-                  uint32_t jitter,
-                  uint32_t last_sender_report,
-                  uint32_t delay_since_last_sender_report)
-      : remoteSSRC(remote_ssrc),
-        sourceSSRC(source_ssrc),
-        fractionLost(fraction_lost),
-        cumulativeLost(cumulative_lost),
-        extendedHighSeqNum(extended_high_sequence_number),
-        jitter(jitter),
-        lastSR(last_sender_report),
-        delaySinceLastSR(delay_since_last_sender_report) {}
-
-  // Fields as described by RFC 3550 6.4.2.
-  uint32_t remoteSSRC;  // SSRC of sender of this report.
-  uint32_t sourceSSRC;  // SSRC of the RTP packet sender.
-  uint8_t fractionLost;
-  uint32_t cumulativeLost;  // 24 bits valid.
-  uint32_t extendedHighSeqNum;
-  uint32_t jitter;
-  uint32_t lastSR;
-  uint32_t delaySinceLastSR;
-};
-
-struct RtcpReceiveTimeInfo {
-  // Fields as described by RFC 3611 4.5.
-  uint32_t sourceSSRC;
-  uint32_t lastRR;
-  uint32_t delaySinceLastRR;
-};
-
-typedef std::list<RTCPReportBlock> ReportBlockList;
-
-struct RtpState {
-  RtpState()
-      : sequence_number(0),
-        start_timestamp(0),
-        timestamp(0),
-        capture_time_ms(-1),
-        last_timestamp_time_ms(-1),
-        media_has_been_sent(false) {}
-  uint16_t sequence_number;
-  uint32_t start_timestamp;
-  uint32_t timestamp;
-  int64_t capture_time_ms;
-  int64_t last_timestamp_time_ms;
-  bool media_has_been_sent;
-};
-
-class RtpData
-{
-public:
-    virtual ~RtpData() {}
-
-    virtual int32_t OnReceivedPayloadData(
-        const uint8_t* payloadData,
-        const size_t payloadSize,
-        const WebRtcRTPHeader* rtpHeader) = 0;
-
-    virtual bool OnRecoveredPacket(const uint8_t* packet,
-                                   size_t packet_length) = 0;
-};
-
-class RtpFeedback
-{
-public:
-    virtual ~RtpFeedback() {}
-
-    // Receiving payload change or SSRC change. (return success!)
-    /*
-    *   channels    - number of channels in codec (1 = mono, 2 = stereo)
-    */
-    virtual int32_t OnInitializeDecoder(
-        const int8_t payloadType,
-        const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-        const int frequency,
-        const uint8_t channels,
-        const uint32_t rate) = 0;
-
-    virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0;
-
-    virtual void OnIncomingCSRCChanged(const uint32_t CSRC,
-                                       const bool added) = 0;
-};
-
-class RtpAudioFeedback {
- public:
-  virtual void OnPlayTelephoneEvent(const uint8_t event,
-                                    const uint16_t lengthMs,
-                                    const uint8_t volume) = 0;
-
- protected:
-  virtual ~RtpAudioFeedback() {}
-};
-
-class RtcpIntraFrameObserver {
- public:
-  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
-
-  virtual void OnReceivedSLI(uint32_t ssrc,
-                             uint8_t picture_id) = 0;
-
-  virtual void OnReceivedRPSI(uint32_t ssrc,
-                              uint64_t picture_id) = 0;
-
-  virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) = 0;
-
-  virtual ~RtcpIntraFrameObserver() {}
-};
-
-class RtcpBandwidthObserver {
- public:
-  // REMB or TMMBR
-  virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
-
-  virtual void OnReceivedRtcpReceiverReport(
-      const ReportBlockList& report_blocks,
-      int64_t rtt,
-      int64_t now_ms) = 0;
-
-  virtual ~RtcpBandwidthObserver() {}
-};
-
-struct PacketInfo {
-  PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number)
-      : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0, false) {}
-
-  PacketInfo(int64_t arrival_time_ms,
-             int64_t send_time_ms,
-             uint16_t sequence_number,
-             size_t payload_size,
-             bool was_paced)
-      : PacketInfo(-1,
-                   arrival_time_ms,
-                   send_time_ms,
-                   sequence_number,
-                   payload_size,
-                   was_paced) {}
-
-  PacketInfo(int64_t creation_time_ms,
-             int64_t arrival_time_ms,
-             int64_t send_time_ms,
-             uint16_t sequence_number,
-             size_t payload_size,
-             bool was_paced)
-      : creation_time_ms(creation_time_ms),
-        arrival_time_ms(arrival_time_ms),
-        send_time_ms(send_time_ms),
-        sequence_number(sequence_number),
-        payload_size(payload_size),
-        was_paced(was_paced) {}
-
-  // Time corresponding to when this object was created.
-  int64_t creation_time_ms;
-  // Time corresponding to when the packet was received. Timestamped with the
-  // receiver's clock.
-  int64_t arrival_time_ms;
-  // Time corresponding to when the packet was sent, timestamped with the
-  // sender's clock.
-  int64_t send_time_ms;
-  // Packet identifier, incremented with 1 for every packet generated by the
-  // sender.
-  uint16_t sequence_number;
-  // Size of the packet excluding RTP headers.
-  size_t payload_size;
-  // True if the packet was paced out by the pacer.
-  bool was_paced;
-};
-
-class TransportFeedbackObserver {
- public:
-  TransportFeedbackObserver() {}
-  virtual ~TransportFeedbackObserver() {}
-
-  // Note: Transport-wide sequence number as sequence number. Arrival time
-  // must be set to 0.
-  virtual void AddPacket(uint16_t sequence_number,
-                         size_t length,
-                         bool was_paced) = 0;
-
-  virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
-};
-
-class RtcpRttStats {
- public:
-  virtual void OnRttUpdate(int64_t rtt) = 0;
-
-  virtual int64_t LastProcessedRtt() const = 0;
-
-  virtual ~RtcpRttStats() {};
-};
-
-// Null object version of RtpFeedback.
-class NullRtpFeedback : public RtpFeedback {
- public:
-  virtual ~NullRtpFeedback() {}
-
-  int32_t OnInitializeDecoder(const int8_t payloadType,
-                              const char payloadName[RTP_PAYLOAD_NAME_SIZE],
-                              const int frequency,
-                              const uint8_t channels,
-                              const uint32_t rate) override {
-    return 0;
-  }
-
-  void OnIncomingSSRCChanged(const uint32_t ssrc) override {}
-  void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
-};
-
-// Null object version of RtpData.
-class NullRtpData : public RtpData {
- public:
-  virtual ~NullRtpData() {}
-
-  int32_t OnReceivedPayloadData(const uint8_t* payloadData,
-                                const size_t payloadSize,
-                                const WebRtcRTPHeader* rtpHeader) override {
-    return 0;
-  }
-
-  bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override {
-    return true;
-  }
-};
-
-// Null object version of RtpAudioFeedback.
-class NullRtpAudioFeedback : public RtpAudioFeedback {
- public:
-  virtual ~NullRtpAudioFeedback() {}
-
-  void OnPlayTelephoneEvent(const uint8_t event,
-                            const uint16_t lengthMs,
-                            const uint8_t volume) override {}
-};
-
-// Statistics about packet loss for a single directional connection. All values
-// are totals since the connection initiated.
-struct RtpPacketLossStats {
-  // The number of packets lost in events where no adjacent packets were also
-  // lost.
-  uint64_t single_packet_loss_count;
-  // The number of events in which more than one adjacent packet was lost.
-  uint64_t multiple_packet_loss_event_count;
-  // The number of packets lost in events where more than one adjacent packet
-  // was lost.
-  uint64_t multiple_packet_loss_packet_count;
-};
-
-class RtpPacketSender {
- public:
-  RtpPacketSender() {}
-  virtual ~RtpPacketSender() {}
-
-  enum Priority {
-    kHighPriority = 0,    // Pass through; will be sent immediately.
-    kNormalPriority = 2,  // Put in back of the line.
-    kLowPriority = 3,     // Put in back of the low priority line.
-  };
-  // Low priority packets are mixed with the normal priority packets
-  // while we are paused.
-
-  // Returns true if we send the packet now, else it will add the packet
-  // information to the queue and call TimeToSendPacket when it's time to send.
-  virtual void InsertPacket(Priority priority,
-                            uint32_t ssrc,
-                            uint16_t sequence_number,
-                            int64_t capture_time_ms,
-                            size_t bytes,
-                            bool retransmission) = 0;
-};
-
-class TransportSequenceNumberAllocator {
- public:
-  TransportSequenceNumberAllocator() {}
-  virtual ~TransportSequenceNumberAllocator() {}
-
-  virtual uint16_t AllocateSequenceNumber() = 0;
-};
-
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
diff --git a/webrtc/modules/utility/interface/audio_frame_operations.h b/webrtc/modules/utility/interface/audio_frame_operations.h
deleted file mode 100644
index 017352afc6..0000000000
--- a/webrtc/modules/utility/interface/audio_frame_operations.h
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- *  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_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
-
-#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
-
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-class AudioFrame;
-
-// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
-// Change reference parameters to pointers. Consider using a namespace rather
-// than a class.
-class AudioFrameOperations {
- public:
-  // Upmixes mono |src_audio| to stereo |dst_audio|. This is an out-of-place
-  // operation, meaning src_audio and dst_audio must point to different
-  // buffers. It is the caller's responsibility to ensure that |dst_audio| is
-  // sufficiently large.
-  static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel,
-                           int16_t* dst_audio);
-  // |frame.num_channels_| will be updated. This version checks for sufficient
-  // buffer size and that |num_channels_| is mono.
-  static int MonoToStereo(AudioFrame* frame);
-
-  // Downmixes stereo |src_audio| to mono |dst_audio|. This is an in-place
-  // operation, meaning |src_audio| and |dst_audio| may point to the same
-  // buffer.
-  static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel,
-                           int16_t* dst_audio);
-  // |frame.num_channels_| will be updated. This version checks that
-  // |num_channels_| is stereo.
-  static int StereoToMono(AudioFrame* frame);
-
-  // Swap the left and right channels of |frame|. Fails silently if |frame| is
-  // not stereo.
-  static void SwapStereoChannels(AudioFrame* frame);
-
-  // Zeros out the audio and sets |frame.energy| to zero.
-  static void Mute(AudioFrame& frame);
-
-  static int Scale(float left, float right, AudioFrame& frame);
-
-  static int ScaleWithSat(float scale, AudioFrame& frame);
-};
-
-}  // namespace webrtc
-
-#endif  // #ifndef WEBRTC_MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
diff --git a/webrtc/modules/utility/interface/file_player.h b/webrtc/modules/utility/interface/file_player.h
deleted file mode 100644
index d6737e1918..0000000000
--- a/webrtc/modules/utility/interface/file_player.h
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- *  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_UTILITY_INCLUDE_FILE_PLAYER_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
-
-#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
-
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_frame.h"
-
-namespace webrtc {
-class FileCallback;
-
-class FilePlayer
-{
-public:
-    // The largest decoded frame size in samples (60ms with 32kHz sample rate).
-    enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
-    enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
-
-    // Note: will return NULL for unsupported formats.
-    static FilePlayer* CreateFilePlayer(const uint32_t instanceID,
-                                        const FileFormats fileFormat);
-
-    static void DestroyFilePlayer(FilePlayer* player);
-
-    // Read 10 ms of audio at |frequencyInHz| to |outBuffer|. |lengthInSamples|
-    // will be set to the number of samples read (not the number of samples per
-    // channel).
-    virtual int Get10msAudioFromFile(
-        int16_t* outBuffer,
-        size_t& lengthInSamples,
-        int frequencyInHz) = 0;
-
-    // Register callback for receiving file playing notifications.
-    virtual int32_t RegisterModuleFileCallback(
-        FileCallback* callback) = 0;
-
-    // API for playing audio from fileName to channel.
-    // Note: codecInst is used for pre-encoded files.
-    virtual int32_t StartPlayingFile(
-        const char* fileName,
-        bool loop,
-        uint32_t startPosition,
-        float volumeScaling,
-        uint32_t notification,
-        uint32_t stopPosition = 0,
-        const CodecInst* codecInst = NULL) = 0;
-
-    // Note: codecInst is used for pre-encoded files.
-    virtual int32_t StartPlayingFile(
-        InStream& sourceStream,
-        uint32_t startPosition,
-        float volumeScaling,
-        uint32_t notification,
-        uint32_t stopPosition = 0,
-        const CodecInst* codecInst = NULL) = 0;
-
-    virtual int32_t StopPlayingFile() = 0;
-
-    virtual bool IsPlayingFile() const = 0;
-
-    virtual int32_t GetPlayoutPosition(uint32_t& durationMs) = 0;
-
-    // Set audioCodec to the currently used audio codec.
-    virtual int32_t AudioCodec(CodecInst& audioCodec) const = 0;
-
-    virtual int32_t Frequency() const = 0;
-
-    // Note: scaleFactor is in the range [0.0 - 2.0]
-    virtual int32_t SetAudioScaling(float scaleFactor) = 0;
-
-    // Return the time in ms until next video frame should be pulled (by
-    // calling GetVideoFromFile(..)).
-    // Note: this API reads one video frame from file. This means that it should
-    //       be called exactly once per GetVideoFromFile(..) API call.
-    virtual int32_t TimeUntilNextVideoFrame() { return -1;}
-
-    virtual int32_t StartPlayingVideoFile(
-        const char* /*fileName*/,
-        bool /*loop*/,
-        bool /*videoOnly*/) { return -1;}
-
-    virtual int32_t video_codec_info(VideoCodec& /*videoCodec*/) const
-    {return -1;}
-
-    virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/) { return -1; }
-
-    // Same as GetVideoFromFile(). videoFrame will have the resolution specified
-    // by the width outWidth and height outHeight in pixels.
-    virtual int32_t GetVideoFromFile(VideoFrame& /*videoFrame*/,
-                                     const uint32_t /*outWidth*/,
-                                     const uint32_t /*outHeight*/) {
-      return -1;
-    }
-
-protected:
-    virtual ~FilePlayer() {}
-
-};
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_PLAYER_H_
diff --git a/webrtc/modules/utility/interface/file_recorder.h b/webrtc/modules/utility/interface/file_recorder.h
deleted file mode 100644
index c7e26f6b7f..0000000000
--- a/webrtc/modules/utility/interface/file_recorder.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- *  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_UTILITY_INCLUDE_FILE_RECORDER_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
-
-#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
-
-#include "webrtc/common_types.h"
-#include "webrtc/engine_configurations.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/media_file/include/media_file_defines.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/video_frame.h"
-
-namespace webrtc {
-
-class FileRecorder
-{
-public:
-
-    // Note: will return NULL for unsupported formats.
-    static FileRecorder* CreateFileRecorder(const uint32_t instanceID,
-                                            const FileFormats fileFormat);
-
-    static void DestroyFileRecorder(FileRecorder* recorder);
-
-    virtual int32_t RegisterModuleFileCallback(
-        FileCallback* callback) = 0;
-
-    virtual FileFormats RecordingFileFormat() const = 0;
-
-    virtual int32_t StartRecordingAudioFile(
-        const char* fileName,
-        const CodecInst& codecInst,
-        uint32_t notification) = 0;
-
-    virtual int32_t StartRecordingAudioFile(
-        OutStream& destStream,
-        const CodecInst& codecInst,
-        uint32_t notification) = 0;
-
-    // Stop recording.
-    // Note: this API is for both audio and video.
-    virtual int32_t StopRecording() = 0;
-
-    // Return true if recording.
-    // Note: this API is for both audio and video.
-    virtual bool IsRecording() const = 0;
-
-    virtual int32_t codec_info(CodecInst& codecInst) const = 0;
-
-    // Write frame to file. Frame should contain 10ms of un-ecoded audio data.
-    virtual int32_t RecordAudioToFile(
-        const AudioFrame& frame,
-        const TickTime* playoutTS = NULL) = 0;
-
-    // Open/create the file specified by fileName for writing audio/video data
-    // (relative path is allowed). audioCodecInst specifies the encoding of the
-    // audio data. videoCodecInst specifies the encoding of the video data.
-    // Only video data will be recorded if videoOnly is true. amrFormat
-    // specifies the amr/amrwb storage format.
-    // Note: the file format is AVI.
-    virtual int32_t StartRecordingVideoFile(
-        const char* fileName,
-        const CodecInst& audioCodecInst,
-        const VideoCodec& videoCodecInst,
-        bool videoOnly = false) = 0;
-
-    // Record the video frame in videoFrame to AVI file.
-    virtual int32_t RecordVideoToFile(const VideoFrame& videoFrame) = 0;
-
-protected:
-    virtual ~FileRecorder() {}
-
-};
-}  // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INCLUDE_FILE_RECORDER_H_
diff --git a/webrtc/modules/utility/interface/helpers_android.h b/webrtc/modules/utility/interface/helpers_android.h
deleted file mode 100644
index 80dd676faa..0000000000
--- a/webrtc/modules/utility/interface/helpers_android.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- *  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_UTILITY_INCLUDE_HELPERS_ANDROID_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
-
-#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
-
-#include <jni.h>
-#include <string>
-
-// Abort the process if |jni| has a Java exception pending.
-// TODO(henrika): merge with CHECK_JNI_EXCEPTION() in jni_helpers.h.
-#define CHECK_EXCEPTION(jni)        \
-  RTC_CHECK(!jni->ExceptionCheck()) \
-      << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
-
-namespace webrtc {
-
-// Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
-JNIEnv* GetEnv(JavaVM* jvm);
-
-// Return a |jlong| that will correctly convert back to |ptr|.  This is needed
-// because the alternative (of silently passing a 32-bit pointer to a vararg
-// function expecting a 64-bit param) picks up garbage in the high 32 bits.
-jlong PointerTojlong(void* ptr);
-
-// JNIEnv-helper methods that wraps the API which uses the JNI interface
-// pointer (JNIEnv*). It allows us to RTC_CHECK success and that no Java
-// exception is thrown while calling the method.
-jmethodID GetMethodID(
-    JNIEnv* jni, jclass c, const char* name, const char* signature);
-
-jmethodID GetStaticMethodID(
-    JNIEnv* jni, jclass c, const char* name, const char* signature);
-
-jclass FindClass(JNIEnv* jni, const char* name);
-
-jobject NewGlobalRef(JNIEnv* jni, jobject o);
-
-void DeleteGlobalRef(JNIEnv* jni, jobject o);
-
-// Return thread ID as a string.
-std::string GetThreadId();
-
-// Return thread ID as string suitable for debug logging.
-std::string GetThreadInfo();
-
-// Attach thread to JVM if necessary and detach at scope end if originally
-// attached.
-class AttachThreadScoped {
- public:
-  explicit AttachThreadScoped(JavaVM* jvm);
-  ~AttachThreadScoped();
-  JNIEnv* env();
-
- private:
-  bool attached_;
-  JavaVM* jvm_;
-  JNIEnv* env_;
-};
-
-// Scoped holder for global Java refs.
-template<class T>  // T is jclass, jobject, jintArray, etc.
-class ScopedGlobalRef {
- public:
-  ScopedGlobalRef(JNIEnv* jni, T obj)
-      : jni_(jni), obj_(static_cast<T>(NewGlobalRef(jni, obj))) {}
-  ~ScopedGlobalRef() {
-    DeleteGlobalRef(jni_, obj_);
-  }
-  T operator*() const {
-    return obj_;
-  }
- private:
-  JNIEnv* jni_;
-  T obj_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_ANDROID_H_
diff --git a/webrtc/modules/utility/interface/helpers_ios.h b/webrtc/modules/utility/interface/helpers_ios.h
deleted file mode 100644
index 6013ee9c14..0000000000
--- a/webrtc/modules/utility/interface/helpers_ios.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- *  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.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
-
-#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
-
-#if defined(WEBRTC_IOS)
-
-#include <string>
-
-namespace webrtc {
-namespace ios {
-
-bool CheckAndLogError(BOOL success, NSError* error);
-
-std::string StdStringFromNSString(NSString* nsString);
-
-// Return thread ID as a string.
-std::string GetThreadId();
-
-// Return thread ID as string suitable for debug logging.
-std::string GetThreadInfo();
-
-// Returns [NSThread currentThread] description as string.
-// Example: <NSThread: 0x170066d80>{number = 1, name = main}
-std::string GetCurrentThreadDescription();
-
-std::string GetAudioSessionCategory();
-
-// Returns the current name of the operating system.
-std::string GetSystemName();
-
-// Returns the current version of the operating system.
-std::string GetSystemVersion();
-
-// Returns the version of the operating system as a floating point value.
-float GetSystemVersionAsFloat();
-
-// Returns the device type.
-// Examples: ”iPhone” and ”iPod touch”.
-std::string GetDeviceType();
-
-// Returns a more detailed device name.
-// Examples: "iPhone 5s (GSM)" and "iPhone 6 Plus".
-std::string GetDeviceName();
-
-}  // namespace ios
-}  // namespace webrtc
-
-#endif  // defined(WEBRTC_IOS)
-
-#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_HELPERS_IOS_H_
diff --git a/webrtc/modules/utility/interface/jvm_android.h b/webrtc/modules/utility/interface/jvm_android.h
deleted file mode 100644
index a417c1b942..0000000000
--- a/webrtc/modules/utility/interface/jvm_android.h
+++ /dev/null
@@ -1,187 +0,0 @@
-/*
- *  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.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
-
-#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
-
-#include <jni.h>
-#include <string>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/base/thread_checker.h"
-#include "webrtc/modules/utility/include/helpers_android.h"
-
-namespace webrtc {
-
-// The JNI interface pointer (JNIEnv) is valid only in the current thread.
-// Should another thread need to access the Java VM, it must first call
-// AttachCurrentThread() to attach itself to the VM and obtain a JNI interface
-// pointer. The native thread remains attached to the VM until it calls
-// DetachCurrentThread() to detach.
-class AttachCurrentThreadIfNeeded {
- public:
-  AttachCurrentThreadIfNeeded();
-  ~AttachCurrentThreadIfNeeded();
-
- private:
-  rtc::ThreadChecker thread_checker_;
-  bool attached_;
-};
-
-// This class is created by the NativeRegistration class and is used to wrap
-// the actual Java object handle (jobject) on which we can call methods from
-// C++ in to Java. See example in JVM for more details.
-// TODO(henrika): extend support for type of function calls.
-class GlobalRef {
- public:
-  GlobalRef(JNIEnv* jni, jobject object);
-  ~GlobalRef();
-
-  jboolean CallBooleanMethod(jmethodID methodID, ...);
-  jint CallIntMethod(jmethodID methodID, ...);
-  void CallVoidMethod(jmethodID methodID, ...);
-
- private:
-  JNIEnv* const jni_;
-  const jobject j_object_;
-};
-
-// Wraps the jclass object on which we can call GetMethodId() functions to
-// query method IDs.
-class JavaClass {
- public:
-  JavaClass(JNIEnv* jni, jclass clazz) : jni_(jni), j_class_(clazz) {}
-  ~JavaClass() {}
-
-  jmethodID GetMethodId(const char* name, const char* signature);
-  jmethodID GetStaticMethodId(const char* name, const char* signature);
-  jobject CallStaticObjectMethod(jmethodID methodID, ...);
-
- protected:
-  JNIEnv* const jni_;
-  jclass const j_class_;
-};
-
-// Adds support of the NewObject factory method to the JavaClass class.
-// See example in JVM for more details on how to use it.
-class NativeRegistration : public JavaClass {
- public:
-  NativeRegistration(JNIEnv* jni, jclass clazz);
-  ~NativeRegistration();
-
-  rtc::scoped_ptr<GlobalRef> NewObject(
-      const char* name, const char* signature, ...);
-
- private:
-  JNIEnv* const jni_;
-};
-
-// This class is created by the JVM class and is used to expose methods that
-// needs the JNI interface pointer but its main purpose is to create a
-// NativeRegistration object given name of a Java class and a list of native
-// methods. See example in JVM for more details.
-class JNIEnvironment {
- public:
-  explicit JNIEnvironment(JNIEnv* jni);
-  ~JNIEnvironment();
-
-  // Registers native methods with the Java class specified by |name|.
-  // Note that the class name must be one of the names in the static
-  // |loaded_classes| array defined in jvm_android.cc.
-  // This method must be called on the construction thread.
-  rtc::scoped_ptr<NativeRegistration> RegisterNatives(
-      const char* name, const JNINativeMethod *methods, int num_methods);
-
-  // Converts from Java string to std::string.
-  // This method must be called on the construction thread.
-  std::string JavaToStdString(const jstring& j_string);
-
- private:
-  rtc::ThreadChecker thread_checker_;
-  JNIEnv* const jni_;
-};
-
-// Main class for working with Java from C++ using JNI in WebRTC.
-//
-// Example usage:
-//
-//   // At initialization (e.g. in JNI_OnLoad), call JVM::Initialize.
-//   JNIEnv* jni = ::base::android::AttachCurrentThread();
-//   JavaVM* jvm = NULL;
-//   jni->GetJavaVM(&jvm);
-//   jobject context = ::base::android::GetApplicationContext();
-//   webrtc::JVM::Initialize(jvm, context);
-//
-//   // Header (.h) file of example class called User.
-//   rtc::scoped_ptr<JNIEnvironment> env;
-//   rtc::scoped_ptr<NativeRegistration> reg;
-//   rtc::scoped_ptr<GlobalRef> obj;
-//
-//   // Construction (in .cc file) of User class.
-//   User::User() {
-//     // Calling thread must be attached to the JVM.
-//     env = JVM::GetInstance()->environment();
-//     reg = env->RegisterNatives("org/webrtc/WebRtcTest", ,);
-//     obj = reg->NewObject("<init>", ,);
-//   }
-//
-//   // Each User method can now use |reg| and |obj| and call Java functions
-//   // in WebRtcTest.java, e.g. boolean init() {}.
-//   bool User::Foo() {
-//     jmethodID id = reg->GetMethodId("init", "()Z");
-//     return obj->CallBooleanMethod(id);
-//   }
-//
-//   // And finally, e.g. in JNI_OnUnLoad, call JVM::Uninitialize.
-//   JVM::Uninitialize();
-class JVM {
- public:
-  // Stores global handles to the Java VM interface and the application context.
-  // Should be called once on a thread that is attached to the JVM.
-  static void Initialize(JavaVM* jvm, jobject context);
-  // Clears handles stored in Initialize(). Must be called on same thread as
-  // Initialize().
-  static void Uninitialize();
-  // Gives access to the global Java VM interface pointer, which then can be
-  // used to create a valid JNIEnvironment object or to get a JavaClass object.
-  static JVM* GetInstance();
-
-  // Creates a JNIEnvironment object.
-  // This method returns a NULL pointer if AttachCurrentThread() has not been
-  // called successfully. Use the AttachCurrentThreadIfNeeded class if needed.
-  rtc::scoped_ptr<JNIEnvironment> environment();
-
-  // Returns a JavaClass object given class |name|.
-  // Note that the class name must be one of the names in the static
-  // |loaded_classes| array defined in jvm_android.cc.
-  // This method must be called on the construction thread.
-  JavaClass GetClass(const char* name);
-
-  // TODO(henrika): can we make these private?
-  JavaVM* jvm() const { return jvm_; }
-  jobject context() const { return context_; }
-
- protected:
-  JVM(JavaVM* jvm, jobject context);
-  ~JVM();
-
- private:
-  JNIEnv* jni() const { return GetEnv(jvm_); }
-
-  rtc::ThreadChecker thread_checker_;
-  JavaVM* const jvm_;
-  jobject context_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_JVM_ANDROID_H_
diff --git a/webrtc/modules/utility/interface/mock/mock_process_thread.h b/webrtc/modules/utility/interface/mock/mock_process_thread.h
deleted file mode 100644
index c494d4c0ba..0000000000
--- a/webrtc/modules/utility/interface/mock/mock_process_thread.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- *  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_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
-
-#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
-
-#include "webrtc/modules/utility/include/process_thread.h"
-
-#include "testing/gmock/include/gmock/gmock.h"
-
-namespace webrtc {
-
-class MockProcessThread : public ProcessThread {
- public:
-  MOCK_METHOD0(Start, void());
-  MOCK_METHOD0(Stop, void());
-  MOCK_METHOD1(WakeUp, void(Module* module));
-  MOCK_METHOD1(PostTask, void(ProcessTask* task));
-  MOCK_METHOD1(RegisterModule, void(Module* module));
-  MOCK_METHOD1(DeRegisterModule, void(Module* module));
-
-  // MOCK_METHOD1 gets confused with mocking this method, so we work around it
-  // by overriding the method from the interface and forwarding the call to a
-  // mocked, simpler method.
-  void PostTask(rtc::scoped_ptr<ProcessTask> task) override {
-    PostTask(task.get());
-  }
-};
-
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_UTILITY_INCLUDE_MOCK_MOCK_PROCESS_THREAD_H_
diff --git a/webrtc/modules/utility/interface/process_thread.h b/webrtc/modules/utility/interface/process_thread.h
deleted file mode 100644
index 1a7a8750cd..0000000000
--- a/webrtc/modules/utility/interface/process_thread.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  Copyright (c) 2011 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_UTILITY_INCLUDE_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
-
-#pragma message("WARNING: utility/interface is DEPRECATED; use utility/include")
-
-#include "webrtc/typedefs.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace webrtc {
-class Module;
-
-class ProcessTask {
- public:
-  ProcessTask() {}
-  virtual ~ProcessTask() {}
-
-  virtual void Run() = 0;
-};
-
-class ProcessThread {
- public:
-  virtual ~ProcessThread();
-
-  static rtc::scoped_ptr<ProcessThread> Create(const char* thread_name);
-
-  // Starts the worker thread.  Must be called from the construction thread.
-  virtual void Start() = 0;
-
-  // Stops the worker thread.  Must be called from the construction thread.
-  virtual void Stop() = 0;
-
-  // Wakes the thread up to give a module a chance to do processing right
-  // away.  This causes the worker thread to wake up and requery the specified
-  // module for when it should be called back. (Typically the module should
-  // return 0 from TimeUntilNextProcess on the worker thread at that point).
-  // Can be called on any thread.
-  virtual void WakeUp(Module* module) = 0;
-
-  // Queues a task object to run on the worker thread.  Ownership of the
-  // task object is transferred to the ProcessThread and the object will
-  // either be deleted after running on the worker thread, or on the
-  // construction thread of the ProcessThread instance, if the task did not
-  // get a chance to run (e.g. posting the task while shutting down or when
-  // the thread never runs).
-  virtual void PostTask(rtc::scoped_ptr<ProcessTask> task) = 0;
-
-  // Adds a module that will start to receive callbacks on the worker thread.
-  // Can be called from any thread.
-  virtual void RegisterModule(Module* module) = 0;
-
-  // Removes a previously registered module.
-  // Can be called from any thread.
-  virtual void DeRegisterModule(Module* module) = 0;
-};
-
-}  // namespace webrtc
-
-#endif // WEBRTC_MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_