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&)); |