| Index: webrtc/media/engine/fakewebrtcvoiceengine.h
|
| diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| index 5f0be5c977daf6781691c265b98fcb1eb0c9cbb6..8d15f5249e0bbf3c4c0049b0d6a24c3736323f23 100644
|
| --- a/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| +++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
|
| @@ -121,7 +121,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
|
| class FakeWebRtcVoiceEngine
|
| : public webrtc::VoEAudioProcessing,
|
| public webrtc::VoEBase, public webrtc::VoECodec,
|
| - public webrtc::VoEHardware, public webrtc::VoERTP_RTCP,
|
| + public webrtc::VoEHardware,
|
| public webrtc::VoEVolumeControl {
|
| public:
|
| struct Channel {
|
| @@ -136,7 +136,6 @@ class FakeWebRtcVoiceEngine
|
| bool opus_dtx = false;
|
| int cn8_type = 13;
|
| int cn16_type = 105;
|
| - uint32_t send_ssrc = 0;
|
| int associate_send_channel = -1;
|
| std::vector<webrtc::CodecInst> recv_codecs;
|
| webrtc::CodecInst send_codec;
|
| @@ -156,9 +155,6 @@ class FakeWebRtcVoiceEngine
|
| bool IsInited() const { return inited_; }
|
| int GetLastChannel() const { return last_channel_; }
|
| int GetNumChannels() const { return static_cast<int>(channels_.size()); }
|
| - uint32_t GetLocalSSRC(int channel) {
|
| - return channels_[channel]->send_ssrc;
|
| - }
|
| bool GetPlayout(int channel) {
|
| return channels_[channel]->playout;
|
| }
|
| @@ -434,43 +430,6 @@ class FakeWebRtcVoiceEngine
|
| WEBRTC_STUB(EnableBuiltInNS, (bool enable));
|
| bool BuiltInNSIsAvailable() const override { return false; }
|
|
|
| - // webrtc::VoERTP_RTCP
|
| - WEBRTC_FUNC(SetLocalSSRC, (int channel, unsigned int ssrc)) {
|
| - WEBRTC_CHECK_CHANNEL(channel);
|
| - channels_[channel]->send_ssrc = ssrc;
|
| - return 0;
|
| - }
|
| - WEBRTC_STUB(GetLocalSSRC, (int channel, unsigned int& ssrc));
|
| - WEBRTC_STUB(GetRemoteSSRC, (int channel, unsigned int& ssrc));
|
| - WEBRTC_STUB(SetSendAudioLevelIndicationStatus, (int channel, bool enable,
|
| - unsigned char id));
|
| - WEBRTC_STUB(SetReceiveAudioLevelIndicationStatus, (int channel, bool enable,
|
| - unsigned char id));
|
| - WEBRTC_STUB(SetSendAbsoluteSenderTimeStatus, (int channel, bool enable,
|
| - unsigned char id));
|
| - WEBRTC_STUB(SetReceiveAbsoluteSenderTimeStatus, (int channel, bool enable,
|
| - unsigned char id));
|
| - WEBRTC_STUB(SetRTCPStatus, (int channel, bool enable));
|
| - WEBRTC_STUB(GetRTCPStatus, (int channel, bool& enabled));
|
| - WEBRTC_STUB(SetRTCP_CNAME, (int channel, const char cname[256]));
|
| - WEBRTC_STUB(GetRTCP_CNAME, (int channel, char cname[256]));
|
| - WEBRTC_STUB(GetRemoteRTCP_CNAME, (int channel, char* cname));
|
| - WEBRTC_STUB(GetRemoteRTCPData, (int channel, unsigned int& NTPHigh,
|
| - unsigned int& NTPLow,
|
| - unsigned int& timestamp,
|
| - unsigned int& playoutTimestamp,
|
| - unsigned int* jitter,
|
| - unsigned short* fractionLost));
|
| - WEBRTC_STUB(GetRemoteRTCPReportBlocks,
|
| - (int channel, std::vector<webrtc::ReportBlock>* receive_blocks));
|
| - WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs,
|
| - unsigned int& maxJitterMs,
|
| - unsigned int& discardedPackets));
|
| - WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats));
|
| - WEBRTC_STUB(SetREDStatus, (int channel, bool enable, int redPayloadtype));
|
| - WEBRTC_STUB(GetREDStatus, (int channel, bool& enable, int& redPayloadtype));
|
| - WEBRTC_STUB(SetNACKStatus, (int channel, bool enable, int maxNoPackets));
|
| -
|
| // webrtc::VoEVolumeControl
|
| WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
|
| WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));
|
|
|