| 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
|
|
|
|
|