Make ~webrtc::AudioSendStream public, and s/config()/GetConfig(), as well as make public.

webrtc/call/audio_send_stream.h

Index: webrtc/call/audio_send_stream.h
diff --git a/webrtc/call/audio_send_stream.h b/webrtc/call/audio_send_stream.h
index 26729e426cb835613ad61d610a19a9b0bb26f5d0..a2236a747538f0718668a77a2007b99dd505fc02 100644
--- a/webrtc/call/audio_send_stream.h
+++ b/webrtc/call/audio_send_stream.h
@@ -18,7 +18,9 @@
 #include "webrtc/api/audio_codecs/audio_encoder_factory.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/call/transport.h"
+#include "webrtc/audio/time_interval.h"
 #include "webrtc/config.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/rtc_base/optional.h"
 #include "webrtc/typedefs.h"
 
@@ -31,6 +33,8 @@ namespace webrtc {
 
 class AudioSendStream {
  public:
+  virtual ~AudioSendStream() = default;
+
   struct Stats {
     Stats();
     ~Stats();
@@ -147,8 +151,11 @@ class AudioSendStream {
 
   virtual Stats GetStats() const = 0;
 
- protected:
-  virtual ~AudioSendStream() {}
+  virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0;
+
+  virtual RtpState GetRtpState() const = 0;
+
+  virtual const TimeInterval& GetActiveLifetime() const = 0;
 };
 }  // namespace webrtc