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..fa5b5eecc8d20c5fa57116b7b1aba8c5bc7bd887 100644 |
--- a/webrtc/call/audio_send_stream.h |
+++ b/webrtc/call/audio_send_stream.h |
@@ -129,6 +129,10 @@ class AudioSendStream { |
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory; |
}; |
+ virtual ~AudioSendStream() = default; |
+ |
+ virtual const webrtc::AudioSendStream::Config& GetConfig() const = 0; |
+ |
// Reconfigure the stream according to the Configuration. |
virtual void Reconfigure(const Config& config) = 0; |
@@ -146,9 +150,6 @@ class AudioSendStream { |
virtual void SetMuted(bool muted) = 0; |
virtual Stats GetStats() const = 0; |
- |
- protected: |
- virtual ~AudioSendStream() {} |
}; |
} // namespace webrtc |