| 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..43965d325cd96daa5a365c1abef01927c7341fe4 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 +59,16 @@
|
|
|
| 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;
|
|
|
| @@ -65,6 +78,9 @@
|
| 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_;
|
| @@ -84,6 +100,12 @@
|
| 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_
|
|
|