Reland of Implemented the GetSources() in native code.

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"
 
 #include <assert.h>
 #include <math.h>
 #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"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 
 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,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
   if (!incoming_payload_callback)
     incoming_payload_callback = NullObjectRtpData();
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
       clock, incoming_messages_callback, rtp_payload_registry,
       RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
 }
 
 RtpReceiver* RtpReceiver::CreateAudioReceiver(
     Clock* clock,
     RtpData* incoming_payload_callback,
     RtpFeedback* incoming_messages_callback,
     RTPPayloadRegistry* rtp_payload_registry) {
   if (!incoming_payload_callback)
     incoming_payload_callback = NullObjectRtpData();
   if (!incoming_messages_callback)
     incoming_messages_callback = NullObjectRtpFeedback();
   return new RtpReceiverImpl(
       clock, incoming_messages_callback, rtp_payload_registry,
       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),
       cb_rtp_feedback_(incoming_messages_callback),
       last_receive_time_(0),
       last_received_payload_length_(0),
       ssrc_(0),
       num_csrcs_(0),
       current_remote_csrc_(),
       last_received_timestamp_(0),
@@ skipping 82 matching lines @@
     }
     LOG(LS_WARNING) << "Receiving invalid payload type.";
     return false;
   }
 
   WebRtcRTPHeader webrtc_rtp_header;
   memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
   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;
   {
     rtc::CritScope lock(&critical_section_rtp_receiver_);
     if (HaveReceivedFrame()) {
       is_first_packet_in_frame =
           last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
           last_received_timestamp_ != rtp_header.timestamp;
     } else {
@@ skipping 23 matching lines @@
       last_received_sequence_number_ = rtp_header.sequenceNumber;
     }
   }
   return true;
 }
 
 TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
   return rtp_media_receiver_->GetTelephoneEventHandler();
 }
 
+std::vector<RtpSource> RtpReceiverImpl::GetSources() const {
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  std::vector<RtpSource> sources;
+
+  {
+    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())
     return false;
   *timestamp = last_received_timestamp_;
   return true;
 }
 
 bool RtpReceiverImpl::LastReceivedTimeMs(int64_t* receive_time_ms) const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
@@ skipping 238 matching lines @@
     // Using CSRC 0 to signal this event, not interop safe, other
     // implementations might have CSRC 0 as a valid value.
     if (num_csrcs_diff > 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
     } else if (num_csrcs_diff < 0) {
       cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
     }
   }
 }
 
+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