| 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 {
|
|
|