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Unified Diff: webrtc/media/engine/fakewebrtcvoiceengine.h

Issue 2072783002: - Remove use of VoERTP_RTCP::SetLocalSSRC() for receive streams; recreate them instead. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase+added comment Created 4 years, 6 months ago
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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&));
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