#include <audio_send_stream.h>
◆ ~AudioSendStream() [1/2]
virtual webrtc::AudioSendStream::~AudioSendStream |
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inlineprotectedvirtual |
◆ ~AudioSendStream() [2/2]
virtual webrtc::AudioSendStream::~AudioSendStream |
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inlineprotectedvirtual |
◆ GetStats() [1/2]
virtual Stats webrtc::AudioSendStream::GetStats |
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const |
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pure virtual |
◆ GetStats() [2/2]
virtual Stats webrtc::AudioSendStream::GetStats |
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const |
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pure virtual |
◆ SendTelephoneEvent() [1/2]
virtual bool webrtc::AudioSendStream::SendTelephoneEvent |
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int |
payload_type, |
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int |
payload_frequency, |
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int |
event, |
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int |
duration_ms |
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) |
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pure virtual |
◆ SendTelephoneEvent() [2/2]
virtual bool webrtc::AudioSendStream::SendTelephoneEvent |
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int |
payload_type, |
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int |
payload_frequency, |
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int |
event, |
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int |
duration_ms |
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) |
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pure virtual |
◆ SetMuted() [1/2]
virtual void webrtc::AudioSendStream::SetMuted |
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bool |
muted | ) |
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pure virtual |
◆ SetMuted() [2/2]
virtual void webrtc::AudioSendStream::SetMuted |
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bool |
muted | ) |
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pure virtual |
◆ Start() [1/2]
virtual void webrtc::AudioSendStream::Start |
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pure virtual |
◆ Start() [2/2]
virtual void webrtc::AudioSendStream::Start |
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pure virtual |
◆ Stop() [1/2]
virtual void webrtc::AudioSendStream::Stop |
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pure virtual |
◆ Stop() [2/2]
virtual void webrtc::AudioSendStream::Stop |
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pure virtual |
The documentation for this class was generated from the following file: