| 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_
|
|
|