Move talk/app/webrtc to webrtc/api

talk/app/webrtc/test/mockpeerconnectionobservers.h

-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-// This file contains mock implementations of observers used in PeerConnection.
-
-#ifndef TALK_APP_WEBRTC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
-#define TALK_APP_WEBRTC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
-
-#include <string>
-
-#include "talk/app/webrtc/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_;
-  rtc::scoped_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), received_message_count_(0) {
-    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 {
-    last_message_.assign(buffer.data.data<char>(), buffer.data.size());
-    ++received_message_count_;
-  }
-
-  bool IsOpen() const { return state_ == DataChannelInterface::kOpen; }
-  const std::string& last_message() const { return last_message_; }
-  size_t received_message_count() const { return received_message_count_; }
-
- private:
-  rtc::scoped_refptr<webrtc::DataChannelInterface> channel_;
-  DataChannelInterface::DataState state_;
-  std::string last_message_;
-  size_t received_message_count_;
-};
-
-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  // TALK_APP_WEBRTC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_