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..3d917ed7c8ff97e94c933cdb6d4ae2a2f21036e5 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 +26,9 @@
 
 namespace webrtc {
 
+// Only return the contribuing sources in the last 10 seconds.
+static const int64_t kContributingSourcesTimeoutMs = 10000;
+
 using RtpUtility::Payload;
 
 RtpReceiver* RtpReceiver::CreateVideoReceiver(
@@ -53,11 +59,10 @@ 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),
@@ -69,7 +74,9 @@ RtpReceiverImpl::RtpReceiverImpl(
       current_remote_csrc_(),
       last_received_timestamp_(0),
       last_received_frame_time_ms_(-1),
-      last_received_sequence_number_(0) {
+      last_received_sequence_number_(0),
+      current_buffer_index_(0),
+      current_buffer_size_(0) {
   assert(incoming_messages_callback);
 
   memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
@@ -160,6 +167,8 @@ 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 +212,42 @@ TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
   return rtp_media_receiver_->GetTelephoneEventHandler();
 }
 
+const std::vector<RtpContributingSource>&
+RtpReceiverImpl::GetContributingSources() {
+  contributing_sources_.clear();
+  std::set<uint32_t> selected_sources_set;
+  int64_t now = clock_->TimeInMilliseconds();
+
+  {
+    rtc::CritScope lock(&critical_section_rtp_receiver_);
+
+    for (size_t i = 1; i <= current_buffer_size_; ++i) {
+      // Iterate the buffer in reverse order.
+      size_t index =
+          (current_buffer_index_ + kContributingSourcesBufferSize - i) %
+          kContributingSourcesBufferSize;
+      RtpContributingSource& contributing_source =
+          contributing_sources_buffer_[index];
+      // Stop iterating when the contributing source object is out of date since
+      // the buffer is ordered by the timestamp.
+      if (now - contributing_source.timestamp > kContributingSourcesTimeoutMs)
+        break;
+      // Return the latest timestamp for a given SSRC and skip the duplicated
+      // ones.
+      if (selected_sources_set.find(contributing_source.source) ==
+          selected_sources_set.end()) {
+        selected_sources_set.insert(contributing_source.source);
+        contributing_sources_.push_back(contributing_source);
+      }
+    }
+
+    // Add the contributing source using the SSRC.
+    contributing_sources_.push_back(ssrc_source_);
+  }  // End critsect.
+
+  return contributing_sources_;
+}
+
 bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   if (!HaveReceivedFrame())
@@ -461,4 +506,32 @@ 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) {
+    contributing_sources_buffer_[current_buffer_index_].timestamp = now;
+    contributing_sources_buffer_[current_buffer_index_].source =
+        current_remote_csrc_[i];
+    current_buffer_index_ =
+        (current_buffer_index_ + 1) % kContributingSourcesBufferSize;
+
+    if (current_buffer_size_ < kContributingSourcesBufferSize) {
+      ++current_buffer_size_;
+    }
+  }
+
+  // If the SSRC is changed and the old SSRC has been kept for more than
+  // |kContributingSourcesTimeoutMs| or this is the first RTP packet, update
+  // both the ssrc and timestamp.
+  // If the SSRC isn't changed, just update the timestamp.
+  if ((now - ssrc_source_.timestamp) > kContributingSourcesTimeoutMs ||
+      ssrc_source_.source == 0) {
+    ssrc_source_.timestamp = now;
+    ssrc_source_.source = ssrc_;
+  } else if (ssrc_ == ssrc_source_.source) {
+    ssrc_source_.timestamp = now;
+  }
+}
+
 }  // namespace webrtc