Chromium Code Reviews| Index: webrtc/video/rtp_video_stream_receiver.h |
| diff --git a/webrtc/video/rtp_video_stream_receiver.h b/webrtc/video/rtp_video_stream_receiver.h |
| index afcc696da16e78abdfe3a0729cca448d25c5aee7..0fede1e3a82aac9bff7fe42097bf0e3cb240bcd4 100644 |
| --- a/webrtc/video/rtp_video_stream_receiver.h |
| +++ b/webrtc/video/rtp_video_stream_receiver.h |
| @@ -18,6 +18,7 @@ |
| #include <vector> |
| #include "webrtc/call/rtp_packet_sink_interface.h" |
| +#include "webrtc/call/secondary_rtp_sinks_container.h" |
| #include "webrtc/modules/include/module_common_types.h" |
| #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
| #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
| @@ -60,6 +61,7 @@ class RtpVideoStreamReceiver : public RtpData, |
| public RecoveredPacketReceiver, |
| public RtpFeedback, |
| public RtpPacketSinkInterface, |
| + public SecondaryRtpSinksContainer, |
| public VCMFrameTypeCallback, |
| public VCMPacketRequestCallback, |
| public video_coding::OnReceivedFrameCallback, |
| @@ -101,6 +103,10 @@ class RtpVideoStreamReceiver : public RtpData, |
| // Implements RtpPacketSinkInterface. |
| void OnRtpPacket(const RtpPacketReceived& packet) override; |
| + // Implements SecondaryRtpSinksContainer. |
| + void AddSecondarySink(RtpPacketSinkInterface* sink) override; |
| + void RemoveSecondarySink(const RtpPacketSinkInterface* sink) override; |
| + |
| // Implements RtpData. |
| int32_t OnReceivedPayloadData(const uint8_t* payload_data, |
| size_t payload_size, |
| @@ -201,6 +207,10 @@ class RtpVideoStreamReceiver : public RtpData, |
| int16_t last_payload_type_ = -1; |
| bool has_received_frame_; |
| + |
| + // TODO(eladalon): !!! Unit-tests, here and elsewhere, after initial approach |
| + // gets the green light. |
| + std::vector<RtpPacketSinkInterface*> secondary_sinks_; |
|
danilchap
2017/07/24 09:03:46
How do you plan to support protecting audio with f
eladalon
2017/07/24 13:15:48
1.
For audio - yes, its own vector of sinks. I con
|
| }; |
| } // namespace webrtc |