Index: webrtc/audio_send_stream.h |
diff --git a/webrtc/audio_send_stream.h b/webrtc/audio_send_stream.h |
index 18e71253f1158dfe5681c79000d77f2f7faedc46..d5d16f2711eed09f1df5d359196aed0a61f39e28 100644 |
--- a/webrtc/audio_send_stream.h |
+++ b/webrtc/audio_send_stream.h |
@@ -18,7 +18,6 @@ |
#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" |
@@ -29,7 +28,7 @@ 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. |
@@ -90,10 +89,20 @@ class AudioSendStream : public SendStream { |
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 |