Implemented the GetSources() in native code.

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index 79e43ef073536d17e58d6f7b8200e0a7d79be743..8f228954fb71a6cefc380237d8b415e338ea5a2d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -15,6 +15,9 @@
 #include <stdlib.h>
 #include <string.h>
 
+#include <set>
+#include <vector>
+
 #include "webrtc/base/logging.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
@@ -23,6 +26,12 @@
 
 namespace webrtc {
 
+// Only return the contribuing sources in the last 10 seconds.
+static const int64_t kContributingSourcesTimeoutMs = 10000;
+
+// The maximum size of the contributing sources lists.
+static const int kMaxContributingSourceListsSize = 100;
+
 using RtpUtility::Payload;
 
 RtpReceiver* RtpReceiver::CreateVideoReceiver(
@@ -53,11 +62,10 @@ RtpReceiver* RtpReceiver::CreateAudioReceiver(
       RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
 }
 
-RtpReceiverImpl::RtpReceiverImpl(
-    Clock* clock,
-    RtpFeedback* incoming_messages_callback,
-    RTPPayloadRegistry* rtp_payload_registry,
-    RTPReceiverStrategy* rtp_media_receiver)
+RtpReceiverImpl::RtpReceiverImpl(Clock* clock,
+                                 RtpFeedback* incoming_messages_callback,
+                                 RTPPayloadRegistry* rtp_payload_registry,
+                                 RTPReceiverStrategy* rtp_media_receiver)
     : clock_(clock),
       rtp_payload_registry_(rtp_payload_registry),
       rtp_media_receiver_(rtp_media_receiver),
@@ -160,6 +168,8 @@ bool RtpReceiverImpl::IncomingRtpPacket(
   webrtc_rtp_header.header = rtp_header;
   CheckCSRC(webrtc_rtp_header);
 
+  UpdateContributingSource();
+
   size_t payload_data_length = payload_length - rtp_header.paddingLength;
 
   bool is_first_packet_in_frame = false;
@@ -203,6 +213,38 @@ TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
   return rtp_media_receiver_->GetTelephoneEventHandler();
 }
 
+std::vector<RtpContributingSource> RtpReceiverImpl::GetContributingSources() {
+  int64_t now = clock_->TimeInMilliseconds();
+  std::vector<RtpContributingSource> contributing_sources;
+
+  {
+    rtc::CritScope lock(&critical_section_rtp_receiver_);
+
+    for (auto rit = csrc_source_list_.rbegin(); rit != csrc_source_list_.rend();
+         ++rit) {
+      if (now - (*rit).timestamp() > kContributingSourcesTimeoutMs) {
+        break;
+      }
+      contributing_sources.push_back(*rit);
+    }
+
+    // Add the contributing sources that use the SSRC.
+    std::set<uint32_t> selected_ssrcs;
+    for (auto rit = ssrc_source_list_.rbegin(); rit != ssrc_source_list_.rend();
+         ++rit) {
+      if (now - (*rit).timestamp() > kContributingSourcesTimeoutMs) {
+        break;
+      }
+      if (selected_ssrcs.find((*rit).ssrc()) == selected_ssrcs.end()) {
+        selected_ssrcs.insert((*rit).ssrc());
+        contributing_sources.push_back(*rit);
+      }
+    }
+  }  // End critsect.
+
+  return contributing_sources;
+}
+
 bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   if (!HaveReceivedFrame())
@@ -461,4 +503,67 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
   }
 }
 
+void RtpReceiverImpl::UpdateContributingSource() {
+  rtc::CritScope lock(&critical_section_rtp_receiver_);
+  int64_t now = clock_->TimeInMilliseconds();
+
+  for (size_t i = 0; i < num_csrcs_; ++i) {
+    auto map_it = iterator_by_csrc_.find(current_remote_csrc_[i]);
+    // If it is a new CSRC, append a new object to the end of the list.
+    if (map_it == iterator_by_csrc_.end()) {
+      RtpContributingSource contributing_source;
+      contributing_source.set_timestamp(now);
+      contributing_source.set_csrc(current_remote_csrc_[i]);
+      csrc_source_list_.push_back(contributing_source);
+    } else {  // Move the object to the end of the list.
+      auto list_it = map_it->second;
+      (*list_it).set_timestamp(now);
+      (*list_it).set_csrc(current_remote_csrc_[i]);
+      csrc_source_list_.splice(csrc_source_list_.end(), csrc_source_list_,
+                               list_it);
+    }
+    // Update the unordered_map.
+    auto new_list_it = std::prev(csrc_source_list_.end());
+    iterator_by_csrc_[current_remote_csrc_[i]] = new_list_it;
+
+    // Remove the out of date objects if the lists are too large.
+    if (csrc_source_list_.size() + ssrc_source_list_.size() >
+        kMaxContributingSourceListsSize) {
+      UpdateSourceLists(now);
+    }
+  }
+
+  // If this is the first packet or the SSRC is changed, insert a new
+  // contributing source that uses the SSRC.
+  if (ssrc_source_list_.size() == 0 ||
+      ssrc_source_list_[ssrc_source_list_.size() - 1].ssrc() != ssrc_) {
+    RtpContributingSource ssrc_source;
+    ssrc_source.set_timestamp(now);
+    ssrc_source.set_ssrc(ssrc_);
+    ssrc_source_list_.push_back(ssrc_source);
+  } else {
+    ssrc_source_list_[ssrc_source_list_.size() - 1].set_timestamp(now);
+  }
+}
+
+// Update the lists and remove the out of date objects.
+void RtpReceiverImpl::UpdateSourceLists(int64_t now) {
+  for (auto it = csrc_source_list_.begin(); it != csrc_source_list_.end();
+       ++it) {
+    if (now - (*it).timestamp() <= kContributingSourcesTimeoutMs) {
+      break;
+    }
+    iterator_by_csrc_.erase((*it).source());
+    csrc_source_list_.erase(it);
+  }
+
+  for (auto it = ssrc_source_list_.begin(); it != ssrc_source_list_.end();
+       ++it) {
+    if (now - (*it).timestamp() <= kContributingSourcesTimeoutMs) {
+      break;
+    }
+    ssrc_source_list_.erase(it);
+  }
+}
+
 }  // namespace webrtc