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..9dd43c416abd02112087364eb925f5600d8e4cae 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"
@@ -25,6 +28,9 @@ namespace webrtc {
 
 using RtpUtility::Payload;
 
+// Only return the sources in the last 10 seconds.
+const int64_t kGetSourcesTimeoutMs = 10000;
+
 RtpReceiver* RtpReceiver::CreateVideoReceiver(
     Clock* clock,
     RtpData* incoming_payload_callback,
@@ -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),
@@ -160,6 +165,8 @@ bool RtpReceiverImpl::IncomingRtpPacket(
   webrtc_rtp_header.header = rtp_header;
   CheckCSRC(webrtc_rtp_header);
 
+  UpdateSources();
+
   size_t payload_data_length = payload_length - rtp_header.paddingLength;
 
   bool is_first_packet_in_frame = false;
@@ -203,6 +210,45 @@ TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
   return rtp_media_receiver_->GetTelephoneEventHandler();
 }
 
+std::vector<RtpSource> RtpReceiverImpl::GetSources() const {
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  std::vector<RtpSource> sources;
+
+  {

[danilchap, 2017/04/10 12:14:06]

taking current time and constructing an empty vector and returning vector by
value can be done inside lock too, and this extra indentation can be removed.

+    rtc::CritScope lock(&critical_section_rtp_receiver_);
+
+    RTC_DCHECK(std::is_sorted(ssrc_sources_.begin(), ssrc_sources_.end(),
+                              [](const RtpSource& lhs, const RtpSource& rhs) {
+                                return lhs.timestamp_ms() < rhs.timestamp_ms();
+                              }));
+    RTC_DCHECK(std::is_sorted(csrc_sources_.begin(), csrc_sources_.end(),
+                              [](const RtpSource& lhs, const RtpSource& rhs) {
+                                return lhs.timestamp_ms() < rhs.timestamp_ms();
+                              }));
+
+    std::set<uint32_t> selected_ssrcs;
+    for (auto rit = ssrc_sources_.rbegin(); rit != ssrc_sources_.rend();
+         ++rit) {
+      if ((now_ms - rit->timestamp_ms()) > kGetSourcesTimeoutMs) {
+        break;
+      }
+      if (selected_ssrcs.insert(rit->source_id()).second) {
+        sources.push_back(*rit);
+      }
+    }
+
+    for (auto rit = csrc_sources_.rbegin(); rit != csrc_sources_.rend();
+         ++rit) {
+      if ((now_ms - rit->timestamp_ms()) > kGetSourcesTimeoutMs) {
+        break;
+      }
+      sources.push_back(*rit);
+    }
+  }  // End critsect.
+
+  return sources;
+}
+
 bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   if (!HaveReceivedFrame())
@@ -461,4 +507,54 @@ void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
   }
 }
 
+void RtpReceiverImpl::UpdateSources() {
+  rtc::CritScope lock(&critical_section_rtp_receiver_);
+  int64_t now_ms = clock_->TimeInMilliseconds();
+
+  for (size_t i = 0; i < num_csrcs_; ++i) {
+    auto map_it = iterator_by_csrc_.find(current_remote_csrc_[i]);
+    if (map_it == iterator_by_csrc_.end()) {
+      // If it is a new CSRC, append a new object to the end of the list.
+      csrc_sources_.emplace_back(now_ms, current_remote_csrc_[i],
+                                 RtpSourceType::CSRC);
+    } else {
+      // If it is an existing CSRC, move the object to the end of the list.
+      map_it->second->update_timestamp_ms(now_ms);
+      csrc_sources_.splice(csrc_sources_.end(), csrc_sources_, map_it->second);
+    }
+    // Update the unordered_map.
+    iterator_by_csrc_[current_remote_csrc_[i]] = std::prev(csrc_sources_.end());
+  }
+
+  // If this is the first packet or the SSRC is changed, insert a new
+  // contributing source that uses the SSRC.
+  if (ssrc_sources_.empty() || ssrc_sources_.rbegin()->source_id() != ssrc_) {
+    ssrc_sources_.emplace_back(now_ms, ssrc_, RtpSourceType::SSRC);
+  } else {
+    ssrc_sources_.rbegin()->update_timestamp_ms(now_ms);
+  }
+
+  RemoveOutdatedSources(now_ms);
+}
+
+void RtpReceiverImpl::RemoveOutdatedSources(int64_t now_ms) {
+  std::list<RtpSource>::iterator it;
+  for (it = csrc_sources_.begin(); it != csrc_sources_.end(); ++it) {
+    if ((now_ms - it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
+      break;
+    }
+    iterator_by_csrc_.erase(it->source_id());
+  }
+  csrc_sources_.erase(csrc_sources_.begin(), it);
+
+  std::vector<RtpSource>::iterator vec_it;
+  for (vec_it = ssrc_sources_.begin(); vec_it != ssrc_sources_.end();
+       ++vec_it) {
+    if ((now_ms - vec_it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
+      break;
+    }
+  }
+  ssrc_sources_.erase(ssrc_sources_.begin(), vec_it);
+}
+
 }  // namespace webrtc