Index: webrtc/audio/audio_send_stream.h |
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h |
index 70cf1c874d46b41651b67451bad58227ce6e400a..42a04aee0932b3d06ea0508d1b40373934c02c73 100644 |
--- a/webrtc/audio/audio_send_stream.h |
+++ b/webrtc/audio/audio_send_stream.h |
@@ -14,6 +14,7 @@ |
#include <memory> |
#include <vector> |
+#include "webrtc/audio/time_interval.h" |
#include "webrtc/call/audio_send_stream.h" |
#include "webrtc/call/audio_state.h" |
#include "webrtc/call/bitrate_allocator.h" |
@@ -76,8 +77,11 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
void SetTransportOverhead(int transport_overhead_per_packet); |
RtpState GetRtpState() const; |
+ const TimeInterval& GetActiveLifetime() const; |
private: |
+ class TimedTransport; |
+ |
VoiceEngine* voice_engine() const; |
// These are all static to make it less likely that (the old) config_ is |
@@ -117,6 +121,9 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
RtpRtcp* rtp_rtcp_module_; |
rtc::Optional<RtpState> const suspended_rtp_state_; |
+ std::unique_ptr<TimedTransport> timed_send_transport_adapter_; |
+ TimeInterval active_lifetime_; |
+ |
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
}; |
} // namespace internal |