Add LoudestFilter in ConferenceTransport

webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h

+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
+#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
+
+#include <map>
+#include "webrtc/base/timeutils.h"
+#include "webrtc/common_types.h"
+
+namespace voetest {
+
+class LoudestFilter {
+ public:
+  /* ForwardThisPacket()
+   * Decide whether to forward a RTP packet, given its header.
+   *
+   * Input:
+   *   rtp_header : Header of the RTP packet of interest.
+   */
+  bool ForwardThisPacket(const webrtc::RTPHeader& rtp_header);
+
+ private:
+  struct Status {
+    void Set(int audio_level, uint32 last_time_ms) {
+      this->audio_level = audio_level;
+      this->last_time_ms = last_time_ms;
+    }
+    int audio_level;
+    uint32 last_time_ms;
+  };
+
+  void RemoveTimeoutStreams(uint32 time_ms);
+  unsigned int FindQuietestStream();
+
+  // Keeps the streams being forwarded in pair<SSRC, Status>.
+  std::map<unsigned int, Status> stream_levels_;
+
+  const int32 kStreamTimeOutMs = 5000;
+  const size_t kMaxMixSize = 3;
+  const int kInvalidAudioLevel = 128;
+};
+
+
+}  // namespace voetest
+
+#endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_