In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
270 lines
7.7 KiB
C++
270 lines
7.7 KiB
C++
/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This file contains mock implementations of observers used in PeerConnection.
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#ifndef PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
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#define PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
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#include <memory>
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#include <string>
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#include "api/datachannelinterface.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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class MockCreateSessionDescriptionObserver
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: public webrtc::CreateSessionDescriptionObserver {
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public:
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MockCreateSessionDescriptionObserver()
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: called_(false),
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result_(false) {}
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virtual ~MockCreateSessionDescriptionObserver() {}
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virtual void OnSuccess(SessionDescriptionInterface* desc) {
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called_ = true;
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result_ = true;
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desc_.reset(desc);
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}
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virtual void OnFailure(const std::string& error) {
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called_ = true;
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result_ = false;
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}
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bool called() const { return called_; }
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bool result() const { return result_; }
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std::unique_ptr<SessionDescriptionInterface> MoveDescription() {
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return std::move(desc_);
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}
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private:
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bool called_;
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bool result_;
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std::unique_ptr<SessionDescriptionInterface> desc_;
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};
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class MockSetSessionDescriptionObserver
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: public webrtc::SetSessionDescriptionObserver {
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public:
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MockSetSessionDescriptionObserver()
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: called_(false),
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result_(false) {}
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virtual ~MockSetSessionDescriptionObserver() {}
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virtual void OnSuccess() {
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called_ = true;
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result_ = true;
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}
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virtual void OnFailure(const std::string& error) {
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called_ = true;
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result_ = false;
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}
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bool called() const { return called_; }
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bool result() const { return result_; }
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private:
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bool called_;
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bool result_;
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};
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class MockDataChannelObserver : public webrtc::DataChannelObserver {
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public:
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explicit MockDataChannelObserver(webrtc::DataChannelInterface* channel)
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: channel_(channel) {
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channel_->RegisterObserver(this);
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state_ = channel_->state();
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}
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virtual ~MockDataChannelObserver() {
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channel_->UnregisterObserver();
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}
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void OnBufferedAmountChange(uint64_t previous_amount) override {}
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void OnStateChange() override { state_ = channel_->state(); }
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void OnMessage(const DataBuffer& buffer) override {
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messages_.push_back(
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std::string(buffer.data.data<char>(), buffer.data.size()));
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}
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bool IsOpen() const { return state_ == DataChannelInterface::kOpen; }
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std::vector<std::string> messages() const { return messages_; }
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std::string last_message() const {
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return messages_.empty() ? std::string() : messages_.back();
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}
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size_t received_message_count() const { return messages_.size(); }
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private:
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rtc::scoped_refptr<webrtc::DataChannelInterface> channel_;
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DataChannelInterface::DataState state_;
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std::vector<std::string> messages_;
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};
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class MockStatsObserver : public webrtc::StatsObserver {
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public:
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MockStatsObserver() : called_(false), stats_() {}
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virtual ~MockStatsObserver() {}
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virtual void OnComplete(const StatsReports& reports) {
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RTC_CHECK(!called_);
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called_ = true;
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stats_.Clear();
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stats_.number_of_reports = reports.size();
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for (const auto* r : reports) {
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if (r->type() == StatsReport::kStatsReportTypeSsrc) {
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stats_.timestamp = r->timestamp();
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GetIntValue(r, StatsReport::kStatsValueNameAudioOutputLevel,
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&stats_.audio_output_level);
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GetIntValue(r, StatsReport::kStatsValueNameAudioInputLevel,
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&stats_.audio_input_level);
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GetIntValue(r, StatsReport::kStatsValueNameBytesReceived,
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&stats_.bytes_received);
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GetIntValue(r, StatsReport::kStatsValueNameBytesSent,
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&stats_.bytes_sent);
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GetInt64Value(r, StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
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&stats_.capture_start_ntp_time);
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} else if (r->type() == StatsReport::kStatsReportTypeBwe) {
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stats_.timestamp = r->timestamp();
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GetIntValue(r, StatsReport::kStatsValueNameAvailableReceiveBandwidth,
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&stats_.available_receive_bandwidth);
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} else if (r->type() == StatsReport::kStatsReportTypeComponent) {
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stats_.timestamp = r->timestamp();
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GetStringValue(r, StatsReport::kStatsValueNameDtlsCipher,
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&stats_.dtls_cipher);
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GetStringValue(r, StatsReport::kStatsValueNameSrtpCipher,
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&stats_.srtp_cipher);
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}
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}
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}
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bool called() const { return called_; }
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size_t number_of_reports() const { return stats_.number_of_reports; }
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double timestamp() const { return stats_.timestamp; }
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int AudioOutputLevel() const {
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RTC_CHECK(called_);
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return stats_.audio_output_level;
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}
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int AudioInputLevel() const {
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RTC_CHECK(called_);
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return stats_.audio_input_level;
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}
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int BytesReceived() const {
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RTC_CHECK(called_);
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return stats_.bytes_received;
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}
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int BytesSent() const {
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RTC_CHECK(called_);
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return stats_.bytes_sent;
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}
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int64_t CaptureStartNtpTime() const {
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RTC_CHECK(called_);
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return stats_.capture_start_ntp_time;
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}
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int AvailableReceiveBandwidth() const {
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RTC_CHECK(called_);
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return stats_.available_receive_bandwidth;
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}
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std::string DtlsCipher() const {
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RTC_CHECK(called_);
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return stats_.dtls_cipher;
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}
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std::string SrtpCipher() const {
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RTC_CHECK(called_);
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return stats_.srtp_cipher;
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}
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private:
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bool GetIntValue(const StatsReport* report,
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StatsReport::StatsValueName name,
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int* value) {
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const StatsReport::Value* v = report->FindValue(name);
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if (v) {
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// TODO(tommi): We should really just be using an int here :-/
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*value = rtc::FromString<int>(v->ToString());
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}
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return v != nullptr;
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}
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bool GetInt64Value(const StatsReport* report,
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StatsReport::StatsValueName name,
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int64_t* value) {
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const StatsReport::Value* v = report->FindValue(name);
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if (v) {
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// TODO(tommi): We should really just be using an int here :-/
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*value = rtc::FromString<int64_t>(v->ToString());
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}
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return v != nullptr;
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}
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bool GetStringValue(const StatsReport* report,
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StatsReport::StatsValueName name,
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std::string* value) {
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const StatsReport::Value* v = report->FindValue(name);
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if (v)
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*value = v->ToString();
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return v != nullptr;
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}
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bool called_;
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struct {
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void Clear() {
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number_of_reports = 0;
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timestamp = 0;
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audio_output_level = 0;
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audio_input_level = 0;
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bytes_received = 0;
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bytes_sent = 0;
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capture_start_ntp_time = 0;
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available_receive_bandwidth = 0;
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dtls_cipher.clear();
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srtp_cipher.clear();
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}
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size_t number_of_reports;
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double timestamp;
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int audio_output_level;
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int audio_input_level;
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int bytes_received;
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int bytes_sent;
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int64_t capture_start_ntp_time;
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int available_receive_bandwidth;
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std::string dtls_cipher;
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std::string srtp_cipher;
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} stats_;
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};
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// Helper class that just stores the report from the callback.
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class MockRTCStatsCollectorCallback : public webrtc::RTCStatsCollectorCallback {
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public:
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rtc::scoped_refptr<const RTCStatsReport> report() { return report_; }
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bool called() const { return called_; }
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protected:
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void OnStatsDelivered(
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const rtc::scoped_refptr<const RTCStatsReport>& report) override {
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report_ = report;
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called_ = true;
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}
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private:
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bool called_ = false;
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rtc::scoped_refptr<const RTCStatsReport> report_;
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};
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} // namespace webrtc
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#endif // PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_
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