Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
index 9acef79f383ce13d2632643525ac55ed29773ddc..2367a68d89ec5fcc3c347820b0a04e9f3ff5c5e3 100644 |
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h |
@@ -217,6 +217,16 @@ |
virtual void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) = 0; |
}; |
+class RtpAudioFeedback { |
+ public: |
+ virtual void OnPlayTelephoneEvent(const uint8_t event, |
+ const uint16_t lengthMs, |
+ const uint8_t volume) = 0; |
+ |
+ protected: |
+ virtual ~RtpAudioFeedback() {} |
+}; |
+ |
class RtcpIntraFrameObserver { |
public: |
virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0; |
@@ -347,6 +357,16 @@ |
} |
}; |
+// Null object version of RtpAudioFeedback. |
+class NullRtpAudioFeedback : public RtpAudioFeedback { |
+ public: |
+ virtual ~NullRtpAudioFeedback() {} |
+ |
+ void OnPlayTelephoneEvent(const uint8_t event, |
+ const uint16_t lengthMs, |
+ const uint8_t volume) override {} |
+}; |
+ |
// Statistics about packet loss for a single directional connection. All values |
// are totals since the connection initiated. |
struct RtpPacketLossStats { |