Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h |
index 4b5524877c77206c58eb8db145ba0f88202b75c9..ce0169f769e32f6c4143737cd1f89d70b69cb865 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h |
@@ -11,7 +11,10 @@ |
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ |
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ |
+#include <list> |
#include <memory> |
+#include <unordered_map> |
+#include <vector> |
#include "webrtc/base/criticalsection.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
@@ -56,6 +59,8 @@ class RtpReceiverImpl : public RtpReceiver { |
TelephoneEventHandler* GetTelephoneEventHandler() override; |
+ std::vector<RtpContributingSource> GetContributingSources() override; |
+ |
private: |
bool HaveReceivedFrame() const; |
@@ -66,6 +71,9 @@ class RtpReceiverImpl : public RtpReceiver { |
bool* is_red, |
PayloadUnion* payload); |
+ void UpdateContributingSource(); |
+ void UpdateSourceLists(int64_t now); |
+ |
Clock* clock_; |
RTPPayloadRegistry* rtp_payload_registry_; |
std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_; |
@@ -84,6 +92,13 @@ class RtpReceiverImpl : public RtpReceiver { |
uint32_t last_received_timestamp_; |
int64_t last_received_frame_time_ms_; |
uint16_t last_received_sequence_number_; |
+ |
+ std::unordered_map<uint32_t, std::list<RtpContributingSource>::iterator> |
+ iterator_by_csrc_; |
+ // The contributing sources that uses the CSRC. |
+ std::list<RtpContributingSource> csrc_source_list_; |
+ // The contributing sources that uses the SSRC. |
+ std::vector<RtpContributingSource> ssrc_source_list_; |
}; |
} // namespace webrtc |
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ |