1 files changed, 13 insertions, 4 deletions
diff --git a/chromium/third_party/webrtc/audio_send_stream.h b/chromium/third_party/webrtc/audio_send_stream.h
index 24c3d77ab27..d8e98bb0ec9 100644
--- a/chromium/third_party/webrtc/audio_send_stream.h
+++ b/chromium/third_party/webrtc/audio_send_stream.h
@@ -11,13 +11,12 @@
 #ifndef WEBRTC_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_SEND_STREAM_H_
 
+#include <memory>
 #include <string>
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/stream.h"
 #include "webrtc/transport.h"
 #include "webrtc/typedefs.h"
 
@@ -28,7 +27,7 @@ namespace webrtc {
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
-class AudioSendStream : public SendStream {
+class AudioSendStream {
  public:
   struct Stats {
     // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
@@ -84,15 +83,25 @@ class AudioSendStream : public SendStream {
     // Ownership of the encoder object is transferred to Call when the config is
     // passed to Call::CreateAudioSendStream().
     // TODO(solenberg): Implement, once we configure codecs through the new API.
-    // rtc::scoped_ptr<AudioEncoder> encoder;
+    // std::unique_ptr<AudioEncoder> encoder;
     int cng_payload_type = -1;  // pt, or -1 to disable Comfort Noise Generator.
     int red_payload_type = -1;  // pt, or -1 to disable REDundant coding.
   };
 
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+  // Stops stream activity.
+  // When a stream is stopped, it can't receive, process or deliver packets.
+  virtual void Stop() = 0;
+
   // TODO(solenberg): Make payload_type a config property instead.
   virtual bool SendTelephoneEvent(int payload_type, int event,
                                   int duration_ms) = 0;
   virtual Stats GetStats() const = 0;
+
+ protected:
+  virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc