78410ad413
This change corrects a potential race condition when updating a FrameEncryptor for the audio send channel. If a FrameEncryptor is set on an active audio stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the channel. To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor> to only allow deallocation when it is actually set on the encoder queue. ChannelSend is unique in this respect as the Audio Receiver a long with the Video Sender and Video Receiver streams all recreate themselves when they have a configuration change. ChannelSend instead reconfigures itself using the existing channel object. Added Seth as TBR as this only introduces mocks. TBR=shampson@webrtc.org Bug: webrtc:9907 Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4 Reviewed-on: https://webrtc-review.googlesource.com/c/107664 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Qingsi Wang <qingsi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25374}
117 lines
5.3 KiB
C++
117 lines
5.3 KiB
C++
/*
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* Copyright (c) 2015 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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#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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#include <map>
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/test/mock_frame_encryptor.h"
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#include "audio/channel_receive_proxy.h"
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#include "audio/channel_send_proxy.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockChannelReceiveProxy : public voe::ChannelReceiveProxy {
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public:
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MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
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MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
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MOCK_METHOD1(RegisterReceiverCongestionControlObjects,
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void(PacketRouter* packet_router));
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MOCK_METHOD0(ResetReceiverCongestionControlObjects, void());
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MOCK_CONST_METHOD0(GetRTCPStatistics, CallReceiveStatistics());
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MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
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MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
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MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int());
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MOCK_CONST_METHOD0(GetTotalOutputEnergy, double());
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MOCK_CONST_METHOD0(GetTotalOutputDuration, double());
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MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
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MOCK_METHOD1(SetSink, void(AudioSinkInterface* sink));
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MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
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MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
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MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
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MOCK_METHOD2(GetAudioFrameWithInfo,
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AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
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AudioFrame* audio_frame));
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MOCK_CONST_METHOD0(PreferredSampleRate, int());
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MOCK_METHOD1(AssociateSendChannel,
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void(const voe::ChannelSendProxy& send_channel_proxy));
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MOCK_METHOD0(DisassociateSendChannel, void());
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MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
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MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
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MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));
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MOCK_METHOD1(SetReceiveCodecs,
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void(const std::map<int, SdpAudioFormat>& codecs));
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MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());
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MOCK_METHOD0(StartPlayout, void());
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MOCK_METHOD0(StopPlayout, void());
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};
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class MockChannelSendProxy : public voe::ChannelSendProxy {
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public:
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// GMock doesn't like move-only types, like std::unique_ptr.
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virtual bool SetEncoder(int payload_type,
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std::unique_ptr<AudioEncoder> encoder) {
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return SetEncoderForMock(payload_type, &encoder);
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}
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MOCK_METHOD2(SetEncoderForMock,
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bool(int payload_type, std::unique_ptr<AudioEncoder>* encoder));
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MOCK_METHOD1(
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ModifyEncoder,
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void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier));
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MOCK_METHOD1(SetRTCPStatus, void(bool enable));
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MOCK_METHOD1(SetLocalSSRC, void(uint32_t ssrc));
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MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
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MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
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MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
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MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
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MOCK_METHOD2(RegisterSenderCongestionControlObjects,
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void(RtpTransportControllerSendInterface* transport,
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RtcpBandwidthObserver* bandwidth_observer));
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MOCK_METHOD0(ResetSenderCongestionControlObjects, void());
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MOCK_CONST_METHOD0(GetRTCPStatistics, CallSendStatistics());
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MOCK_CONST_METHOD0(GetRemoteRTCPReportBlocks, std::vector<ReportBlock>());
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MOCK_CONST_METHOD0(GetANAStatistics, ANAStats());
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MOCK_METHOD2(SetSendTelephoneEventPayloadType,
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bool(int payload_type, int payload_frequency));
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MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
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MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms));
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MOCK_METHOD1(SetInputMute, void(bool muted));
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MOCK_METHOD1(RegisterTransport, void(Transport* transport));
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MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
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// GMock doesn't like move-only types, like std::unique_ptr.
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virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) {
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ProcessAndEncodeAudioForMock(&audio_frame);
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}
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MOCK_METHOD1(ProcessAndEncodeAudioForMock,
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void(std::unique_ptr<AudioFrame>* audio_frame));
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MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
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MOCK_CONST_METHOD0(GetRtpRtcp, RtpRtcp*());
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MOCK_CONST_METHOD0(GetBitrate, int());
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MOCK_METHOD1(OnTwccBasedUplinkPacketLossRate, void(float packet_loss_rate));
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MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
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void(float recoverable_packet_loss_rate));
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MOCK_METHOD0(StartSend, void());
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MOCK_METHOD0(StopSend, void());
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MOCK_METHOD1(
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SetFrameEncryptor,
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void(rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor));
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};
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} // namespace test
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} // namespace webrtc
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#endif // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
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