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 {
 
+// Only return the contribuing sources in the last 10 seconds.
+static const int64_t kContributingSourcesTimeoutMs = 10000;
+
 using RtpUtility::Payload;
 
 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)
@@ skipping 10 matching lines @@
     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),
       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_));
 }
 
 RtpReceiverImpl::~RtpReceiverImpl() {
   for (int i = 0; i < num_csrcs_; ++i) {
     cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false);
   }
 }
@@ skipping 70 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);
 
+  UpdateContributingSource();
+
   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();
 }
 
+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())
     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::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