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..9d7d049cf882e5afc13dd1fc6b936b04c02e0d48 100644 |
--- a/webrtc/audio/audio_send_stream.h |
+++ b/webrtc/audio/audio_send_stream.h |
@@ -20,6 +20,7 @@ |
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
#include "webrtc/rtc_base/constructormagic.h" |
#include "webrtc/rtc_base/thread_checker.h" |
+#include "webrtc/rtc_base/time_interval.h" |
#include "webrtc/voice_engine/transport_feedback_packet_loss_tracker.h" |
namespace webrtc { |
@@ -76,8 +77,11 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
void SetTransportOverhead(int transport_overhead_per_packet); |
RtpState GetRtpState() const; |
+ const rtc::TimeInterval* GetActiveLifetime() const; |
ossu
2017/07/14 11:13:15
Why not return as a const& instead? Can it be null
saza WebRTC
2017/07/17 14:27:29
Done.
|
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_; |
+ rtc::TimeInterval active_lifetime_; |
+ |
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |
}; |
} // namespace internal |