| Index: webrtc/api/test/mockpeerconnectionobservers.h | 
| diff --git a/webrtc/api/test/mockpeerconnectionobservers.h b/webrtc/api/test/mockpeerconnectionobservers.h | 
| deleted file mode 100644 | 
| index 23647f6de3b5cbf3d5c8e6fda845103c77a04441..0000000000000000000000000000000000000000 | 
| --- a/webrtc/api/test/mockpeerconnectionobservers.h | 
| +++ /dev/null | 
| @@ -1,229 +0,0 @@ | 
| -/* | 
| - *  Copyright 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. | 
| - */ | 
| - | 
| -// This file contains mock implementations of observers used in PeerConnection. | 
| - | 
| -#ifndef WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ | 
| -#define WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ | 
| - | 
| -#include <memory> | 
| -#include <string> | 
| - | 
| -#include "webrtc/api/datachannelinterface.h" | 
| - | 
| -namespace webrtc { | 
| - | 
| -class MockCreateSessionDescriptionObserver | 
| -    : public webrtc::CreateSessionDescriptionObserver { | 
| - public: | 
| -  MockCreateSessionDescriptionObserver() | 
| -      : called_(false), | 
| -        result_(false) {} | 
| -  virtual ~MockCreateSessionDescriptionObserver() {} | 
| -  virtual void OnSuccess(SessionDescriptionInterface* desc) { | 
| -    called_ = true; | 
| -    result_ = true; | 
| -    desc_.reset(desc); | 
| -  } | 
| -  virtual void OnFailure(const std::string& error) { | 
| -    called_ = true; | 
| -    result_ = false; | 
| -  } | 
| -  bool called() const { return called_; } | 
| -  bool result() const { return result_; } | 
| -  SessionDescriptionInterface* release_desc() { | 
| -    return desc_.release(); | 
| -  } | 
| - | 
| - private: | 
| -  bool called_; | 
| -  bool result_; | 
| -  std::unique_ptr<SessionDescriptionInterface> desc_; | 
| -}; | 
| - | 
| -class MockSetSessionDescriptionObserver | 
| -    : public webrtc::SetSessionDescriptionObserver { | 
| - public: | 
| -  MockSetSessionDescriptionObserver() | 
| -      : called_(false), | 
| -        result_(false) {} | 
| -  virtual ~MockSetSessionDescriptionObserver() {} | 
| -  virtual void OnSuccess() { | 
| -    called_ = true; | 
| -    result_ = true; | 
| -  } | 
| -  virtual void OnFailure(const std::string& error) { | 
| -    called_ = true; | 
| -    result_ = false; | 
| -  } | 
| -  bool called() const { return called_; } | 
| -  bool result() const { return result_; } | 
| - | 
| - private: | 
| -  bool called_; | 
| -  bool result_; | 
| -}; | 
| - | 
| -class MockDataChannelObserver : public webrtc::DataChannelObserver { | 
| - public: | 
| -  explicit MockDataChannelObserver(webrtc::DataChannelInterface* channel) | 
| -      : channel_(channel) { | 
| -    channel_->RegisterObserver(this); | 
| -    state_ = channel_->state(); | 
| -  } | 
| -  virtual ~MockDataChannelObserver() { | 
| -    channel_->UnregisterObserver(); | 
| -  } | 
| - | 
| -  void OnBufferedAmountChange(uint64_t previous_amount) override {} | 
| - | 
| -  void OnStateChange() override { state_ = channel_->state(); } | 
| -  void OnMessage(const DataBuffer& buffer) override { | 
| -    messages_.push_back( | 
| -        std::string(buffer.data.data<char>(), buffer.data.size())); | 
| -  } | 
| - | 
| -  bool IsOpen() const { return state_ == DataChannelInterface::kOpen; } | 
| -  std::vector<std::string> messages() const { return messages_; } | 
| -  std::string last_message() const { | 
| -    return messages_.empty() ? std::string() : messages_.back(); | 
| -  } | 
| -  size_t received_message_count() const { return messages_.size(); } | 
| - | 
| - private: | 
| -  rtc::scoped_refptr<webrtc::DataChannelInterface> channel_; | 
| -  DataChannelInterface::DataState state_; | 
| -  std::vector<std::string> messages_; | 
| -}; | 
| - | 
| -class MockStatsObserver : public webrtc::StatsObserver { | 
| - public: | 
| -  MockStatsObserver() : called_(false), stats_() {} | 
| -  virtual ~MockStatsObserver() {} | 
| - | 
| -  virtual void OnComplete(const StatsReports& reports) { | 
| -    ASSERT(!called_); | 
| -    called_ = true; | 
| -    stats_.Clear(); | 
| -    stats_.number_of_reports = reports.size(); | 
| -    for (const auto* r : reports) { | 
| -      if (r->type() == StatsReport::kStatsReportTypeSsrc) { | 
| -        stats_.timestamp = r->timestamp(); | 
| -        GetIntValue(r, StatsReport::kStatsValueNameAudioOutputLevel, | 
| -            &stats_.audio_output_level); | 
| -        GetIntValue(r, StatsReport::kStatsValueNameAudioInputLevel, | 
| -            &stats_.audio_input_level); | 
| -        GetIntValue(r, StatsReport::kStatsValueNameBytesReceived, | 
| -            &stats_.bytes_received); | 
| -        GetIntValue(r, StatsReport::kStatsValueNameBytesSent, | 
| -            &stats_.bytes_sent); | 
| -      } else if (r->type() == StatsReport::kStatsReportTypeBwe) { | 
| -        stats_.timestamp = r->timestamp(); | 
| -        GetIntValue(r, StatsReport::kStatsValueNameAvailableReceiveBandwidth, | 
| -            &stats_.available_receive_bandwidth); | 
| -      } else if (r->type() == StatsReport::kStatsReportTypeComponent) { | 
| -        stats_.timestamp = r->timestamp(); | 
| -        GetStringValue(r, StatsReport::kStatsValueNameDtlsCipher, | 
| -            &stats_.dtls_cipher); | 
| -        GetStringValue(r, StatsReport::kStatsValueNameSrtpCipher, | 
| -            &stats_.srtp_cipher); | 
| -      } | 
| -    } | 
| -  } | 
| - | 
| -  bool called() const { return called_; } | 
| -  size_t number_of_reports() const { return stats_.number_of_reports; } | 
| -  double timestamp() const { return stats_.timestamp; } | 
| - | 
| -  int AudioOutputLevel() const { | 
| -    ASSERT(called_); | 
| -    return stats_.audio_output_level; | 
| -  } | 
| - | 
| -  int AudioInputLevel() const { | 
| -    ASSERT(called_); | 
| -    return stats_.audio_input_level; | 
| -  } | 
| - | 
| -  int BytesReceived() const { | 
| -    ASSERT(called_); | 
| -    return stats_.bytes_received; | 
| -  } | 
| - | 
| -  int BytesSent() const { | 
| -    ASSERT(called_); | 
| -    return stats_.bytes_sent; | 
| -  } | 
| - | 
| -  int AvailableReceiveBandwidth() const { | 
| -    ASSERT(called_); | 
| -    return stats_.available_receive_bandwidth; | 
| -  } | 
| - | 
| -  std::string DtlsCipher() const { | 
| -    ASSERT(called_); | 
| -    return stats_.dtls_cipher; | 
| -  } | 
| - | 
| -  std::string SrtpCipher() const { | 
| -    ASSERT(called_); | 
| -    return stats_.srtp_cipher; | 
| -  } | 
| - | 
| - private: | 
| -  bool GetIntValue(const StatsReport* report, | 
| -                   StatsReport::StatsValueName name, | 
| -                   int* value) { | 
| -    const StatsReport::Value* v = report->FindValue(name); | 
| -    if (v) { | 
| -      // TODO(tommi): We should really just be using an int here :-/ | 
| -      *value = rtc::FromString<int>(v->ToString()); | 
| -    } | 
| -    return v != nullptr; | 
| -  } | 
| - | 
| -  bool GetStringValue(const StatsReport* report, | 
| -                      StatsReport::StatsValueName name, | 
| -                      std::string* value) { | 
| -    const StatsReport::Value* v = report->FindValue(name); | 
| -    if (v) | 
| -      *value = v->ToString(); | 
| -    return v != nullptr; | 
| -  } | 
| - | 
| -  bool called_; | 
| -  struct { | 
| -    void Clear() { | 
| -      number_of_reports = 0; | 
| -      timestamp = 0; | 
| -      audio_output_level = 0; | 
| -      audio_input_level = 0; | 
| -      bytes_received = 0; | 
| -      bytes_sent = 0; | 
| -      available_receive_bandwidth = 0; | 
| -      dtls_cipher.clear(); | 
| -      srtp_cipher.clear(); | 
| -    } | 
| - | 
| -    size_t number_of_reports; | 
| -    double timestamp; | 
| -    int audio_output_level; | 
| -    int audio_input_level; | 
| -    int bytes_received; | 
| -    int bytes_sent; | 
| -    int available_receive_bandwidth; | 
| -    std::string dtls_cipher; | 
| -    std::string srtp_cipher; | 
| -  } stats_; | 
| -}; | 
| - | 
| -}  // namespace webrtc | 
| - | 
| -#endif  // WEBRTC_API_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ | 
|  |