| /* |
| * 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. |
| // TODO(steveanton): These aren't really mocks and should be renamed. |
| |
| #ifndef PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ |
| #define PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "api/datachannelinterface.h" |
| #include "api/jsepicecandidate.h" |
| #include "pc/streamcollection.h" |
| #include "rtc_base/checks.h" |
| |
| namespace webrtc { |
| |
| class MockPeerConnectionObserver : public PeerConnectionObserver { |
| public: |
| struct AddTrackEvent { |
| explicit AddTrackEvent( |
| rtc::scoped_refptr<RtpReceiverInterface> event_receiver, |
| std::vector<rtc::scoped_refptr<MediaStreamInterface>> event_streams) |
| : receiver(std::move(event_receiver)), |
| streams(std::move(event_streams)) { |
| for (auto stream : streams) { |
| std::vector<rtc::scoped_refptr<MediaStreamTrackInterface>> tracks; |
| for (auto audio_track : stream->GetAudioTracks()) { |
| tracks.push_back(audio_track); |
| } |
| for (auto video_track : stream->GetVideoTracks()) { |
| tracks.push_back(video_track); |
| } |
| snapshotted_stream_tracks[stream] = tracks; |
| } |
| } |
| |
| rtc::scoped_refptr<RtpReceiverInterface> receiver; |
| std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams; |
| // This map records the tracks present in each stream at the time the |
| // OnAddTrack callback was issued. |
| std::map<rtc::scoped_refptr<MediaStreamInterface>, |
| std::vector<rtc::scoped_refptr<MediaStreamTrackInterface>>> |
| snapshotted_stream_tracks; |
| }; |
| |
| MockPeerConnectionObserver() : remote_streams_(StreamCollection::Create()) {} |
| virtual ~MockPeerConnectionObserver() {} |
| void SetPeerConnectionInterface(PeerConnectionInterface* pc) { |
| pc_ = pc; |
| if (pc) { |
| state_ = pc_->signaling_state(); |
| } |
| } |
| void OnSignalingChange( |
| PeerConnectionInterface::SignalingState new_state) override { |
| RTC_DCHECK(pc_->signaling_state() == new_state); |
| state_ = new_state; |
| } |
| |
| MediaStreamInterface* RemoteStream(const std::string& label) { |
| return remote_streams_->find(label); |
| } |
| StreamCollectionInterface* remote_streams() const { return remote_streams_; } |
| void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override { |
| last_added_stream_ = stream; |
| remote_streams_->AddStream(stream); |
| } |
| void OnRemoveStream( |
| rtc::scoped_refptr<MediaStreamInterface> stream) override { |
| last_removed_stream_ = stream; |
| remote_streams_->RemoveStream(stream); |
| } |
| void OnRenegotiationNeeded() override { renegotiation_needed_ = true; } |
| void OnDataChannel( |
| rtc::scoped_refptr<DataChannelInterface> data_channel) override { |
| last_datachannel_ = data_channel; |
| } |
| |
| void OnIceConnectionChange( |
| PeerConnectionInterface::IceConnectionState new_state) override { |
| RTC_DCHECK(pc_->ice_connection_state() == new_state); |
| // When ICE is finished, the caller will get to a kIceConnectionCompleted |
| // state, because it has the ICE controlling role, while the callee |
| // will get to a kIceConnectionConnected state. This means that both ICE |
| // and DTLS are connected. |
| ice_connected_ = |
| (new_state == PeerConnectionInterface::kIceConnectionConnected) || |
| (new_state == PeerConnectionInterface::kIceConnectionCompleted); |
| callback_triggered_ = true; |
| } |
| void OnIceGatheringChange( |
| PeerConnectionInterface::IceGatheringState new_state) override { |
| RTC_DCHECK(pc_->ice_gathering_state() == new_state); |
| ice_gathering_complete_ = |
| new_state == PeerConnectionInterface::kIceGatheringComplete; |
| callback_triggered_ = true; |
| } |
| void OnIceCandidate(const IceCandidateInterface* candidate) override { |
| RTC_DCHECK(PeerConnectionInterface::kIceGatheringNew != |
| pc_->ice_gathering_state()); |
| candidates_.push_back(absl::make_unique<JsepIceCandidate>( |
| candidate->sdp_mid(), candidate->sdp_mline_index(), |
| candidate->candidate())); |
| callback_triggered_ = true; |
| } |
| |
| void OnIceCandidatesRemoved( |
| const std::vector<cricket::Candidate>& candidates) override { |
| num_candidates_removed_++; |
| callback_triggered_ = true; |
| } |
| |
| void OnIceConnectionReceivingChange(bool receiving) override { |
| callback_triggered_ = true; |
| } |
| |
| void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver, |
| const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& |
| streams) override { |
| RTC_DCHECK(receiver); |
| num_added_tracks_++; |
| last_added_track_label_ = receiver->id(); |
| add_track_events_.push_back(AddTrackEvent(receiver, streams)); |
| } |
| |
| void OnTrack( |
| rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override { |
| on_track_transceivers_.push_back(transceiver); |
| } |
| |
| void OnRemoveTrack( |
| rtc::scoped_refptr<RtpReceiverInterface> receiver) override { |
| remove_track_events_.push_back(receiver); |
| } |
| |
| std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetAddTrackReceivers() { |
| std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers; |
| for (const AddTrackEvent& event : add_track_events_) { |
| receivers.push_back(event.receiver); |
| } |
| return receivers; |
| } |
| |
| int CountAddTrackEventsForStream(const std::string& stream_id) { |
| int found_tracks = 0; |
| for (const AddTrackEvent& event : add_track_events_) { |
| bool has_stream_id = false; |
| for (auto stream : event.streams) { |
| if (stream->id() == stream_id) { |
| has_stream_id = true; |
| break; |
| } |
| } |
| if (has_stream_id) { |
| ++found_tracks; |
| } |
| } |
| return found_tracks; |
| } |
| |
| // Returns the id of the last added stream. |
| // Empty string if no stream have been added. |
| std::string GetLastAddedStreamId() { |
| if (last_added_stream_.get()) |
| return last_added_stream_->id(); |
| return ""; |
| } |
| std::string GetLastRemovedStreamId() { |
| if (last_removed_stream_.get()) |
| return last_removed_stream_->id(); |
| return ""; |
| } |
| |
| IceCandidateInterface* last_candidate() { |
| if (candidates_.empty()) { |
| return nullptr; |
| } else { |
| return candidates_.back().get(); |
| } |
| } |
| |
| std::vector<const IceCandidateInterface*> GetAllCandidates() { |
| std::vector<const IceCandidateInterface*> candidates; |
| for (const auto& candidate : candidates_) { |
| candidates.push_back(candidate.get()); |
| } |
| return candidates; |
| } |
| |
| std::vector<IceCandidateInterface*> GetCandidatesByMline(int mline_index) { |
| std::vector<IceCandidateInterface*> candidates; |
| for (const auto& candidate : candidates_) { |
| if (candidate->sdp_mline_index() == mline_index) { |
| candidates.push_back(candidate.get()); |
| } |
| } |
| return candidates; |
| } |
| |
| bool negotiation_needed() const { return renegotiation_needed_; } |
| void clear_negotiation_needed() { renegotiation_needed_ = false; } |
| |
| rtc::scoped_refptr<PeerConnectionInterface> pc_; |
| PeerConnectionInterface::SignalingState state_; |
| std::vector<std::unique_ptr<IceCandidateInterface>> candidates_; |
| rtc::scoped_refptr<DataChannelInterface> last_datachannel_; |
| rtc::scoped_refptr<StreamCollection> remote_streams_; |
| bool renegotiation_needed_ = false; |
| bool ice_gathering_complete_ = false; |
| bool ice_connected_ = false; |
| bool callback_triggered_ = false; |
| int num_added_tracks_ = 0; |
| std::string last_added_track_label_; |
| std::vector<AddTrackEvent> add_track_events_; |
| std::vector<rtc::scoped_refptr<RtpReceiverInterface>> remove_track_events_; |
| std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> |
| on_track_transceivers_; |
| int num_candidates_removed_ = 0; |
| |
| private: |
| rtc::scoped_refptr<MediaStreamInterface> last_added_stream_; |
| rtc::scoped_refptr<MediaStreamInterface> last_removed_stream_; |
| }; |
| |
| class MockCreateSessionDescriptionObserver |
| : public webrtc::CreateSessionDescriptionObserver { |
| public: |
| MockCreateSessionDescriptionObserver() |
| : called_(false), |
| error_("MockCreateSessionDescriptionObserver not called") {} |
| virtual ~MockCreateSessionDescriptionObserver() {} |
| void OnSuccess(SessionDescriptionInterface* desc) override { |
| called_ = true; |
| error_ = ""; |
| desc_.reset(desc); |
| } |
| void OnFailure(webrtc::RTCError error) override { |
| called_ = true; |
| error_ = error.message(); |
| } |
| bool called() const { return called_; } |
| bool result() const { return error_.empty(); } |
| const std::string& error() const { return error_; } |
| std::unique_ptr<SessionDescriptionInterface> MoveDescription() { |
| return std::move(desc_); |
| } |
| |
| private: |
| bool called_; |
| std::string error_; |
| std::unique_ptr<SessionDescriptionInterface> desc_; |
| }; |
| |
| class MockSetSessionDescriptionObserver |
| : public webrtc::SetSessionDescriptionObserver { |
| public: |
| MockSetSessionDescriptionObserver() |
| : called_(false), |
| error_("MockSetSessionDescriptionObserver not called") {} |
| ~MockSetSessionDescriptionObserver() override {} |
| void OnSuccess() override { |
| called_ = true; |
| error_ = ""; |
| } |
| void OnFailure(webrtc::RTCError error) override { |
| called_ = true; |
| error_ = error.message(); |
| } |
| |
| bool called() const { return called_; } |
| bool result() const { return error_.empty(); } |
| const std::string& error() const { return error_; } |
| |
| private: |
| bool called_; |
| std::string error_; |
| }; |
| |
| class MockSetRemoteDescriptionObserver |
| : public rtc::RefCountedObject<SetRemoteDescriptionObserverInterface> { |
| public: |
| bool called() const { return error_.has_value(); } |
| RTCError& error() { |
| RTC_DCHECK(error_.has_value()); |
| return *error_; |
| } |
| |
| // SetRemoteDescriptionObserverInterface implementation. |
| void OnSetRemoteDescriptionComplete(RTCError error) override { |
| error_ = std::move(error); |
| } |
| |
| private: |
| // Set on complete, on success this is set to an RTCError::OK() error. |
| absl::optional<RTCError> error_; |
| }; |
| |
| 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) { |
| RTC_CHECK(!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); |
| GetInt64Value(r, StatsReport::kStatsValueNameCaptureStartNtpTimeMs, |
| &stats_.capture_start_ntp_time); |
| } 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 { |
| RTC_CHECK(called_); |
| return stats_.audio_output_level; |
| } |
| |
| int AudioInputLevel() const { |
| RTC_CHECK(called_); |
| return stats_.audio_input_level; |
| } |
| |
| int BytesReceived() const { |
| RTC_CHECK(called_); |
| return stats_.bytes_received; |
| } |
| |
| int BytesSent() const { |
| RTC_CHECK(called_); |
| return stats_.bytes_sent; |
| } |
| |
| int64_t CaptureStartNtpTime() const { |
| RTC_CHECK(called_); |
| return stats_.capture_start_ntp_time; |
| } |
| |
| int AvailableReceiveBandwidth() const { |
| RTC_CHECK(called_); |
| return stats_.available_receive_bandwidth; |
| } |
| |
| std::string DtlsCipher() const { |
| RTC_CHECK(called_); |
| return stats_.dtls_cipher; |
| } |
| |
| std::string SrtpCipher() const { |
| RTC_CHECK(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 GetInt64Value(const StatsReport* report, |
| StatsReport::StatsValueName name, |
| int64_t* value) { |
| const StatsReport::Value* v = report->FindValue(name); |
| if (v) { |
| // TODO(tommi): We should really just be using an int here :-/ |
| *value = rtc::FromString<int64_t>(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; |
| capture_start_ntp_time = 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; |
| int64_t capture_start_ntp_time; |
| int available_receive_bandwidth; |
| std::string dtls_cipher; |
| std::string srtp_cipher; |
| } stats_; |
| }; |
| |
| // Helper class that just stores the report from the callback. |
| class MockRTCStatsCollectorCallback : public webrtc::RTCStatsCollectorCallback { |
| public: |
| rtc::scoped_refptr<const RTCStatsReport> report() { return report_; } |
| |
| bool called() const { return called_; } |
| |
| protected: |
| void OnStatsDelivered( |
| const rtc::scoped_refptr<const RTCStatsReport>& report) override { |
| report_ = report; |
| called_ = true; |
| } |
| |
| private: |
| bool called_ = false; |
| rtc::scoped_refptr<const RTCStatsReport> report_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_ |