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 43965d325cd96daa5a365c1abef01927c7341fe4..4b5524877c77206c58eb8db145ba0f88202b75c9 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h |
@@ -11,10 +11,7 @@ |
#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" |
@@ -59,16 +56,6 @@ |
TelephoneEventHandler* GetTelephoneEventHandler() override; |
- std::vector<RtpSource> GetSources() const override; |
- |
- const std::vector<RtpSource>& ssrc_sources_for_testing() const { |
- return ssrc_sources_; |
- } |
- |
- const std::list<RtpSource>& csrc_sources_for_testing() const { |
- return csrc_sources_; |
- } |
- |
private: |
bool HaveReceivedFrame() const; |
@@ -78,9 +65,6 @@ |
const int8_t first_payload_byte, |
bool* is_red, |
PayloadUnion* payload); |
- |
- void UpdateSources(); |
- void RemoveOutdatedSources(int64_t now_ms); |
Clock* clock_; |
RTPPayloadRegistry* rtp_payload_registry_; |
@@ -100,12 +84,6 @@ |
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<RtpSource>::iterator> |
- iterator_by_csrc_; |
- // The RtpSource objects are sorted chronologically. |
- std::list<RtpSource> csrc_sources_; |
- std::vector<RtpSource> ssrc_sources_; |
}; |
} // namespace webrtc |
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_ |