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/*
* 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_
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