Add LoudestFilter in ConferenceTransport

webrtc/voice_engine/test/auto_test/voe_conference_test.cc

 /*
  *  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.
  */
 
 #include <queue>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
 
 namespace {
   static const int kRttMs = 25;
 
   static bool IsNear(int ref, int comp, int error) {
     return (ref - comp <= error) && (comp - ref >= -error);
   }
+
+  static void CreateSilenceFile(const std::string& silence_file) {
+    FILE* fid = fopen(silence_file.c_str(), "wb");
+    int temp = 0;

[hlundin-webrtc, 2015/08/13 07:11:12]

I don't think it matters in practice, but I think this should be an int16_t,
since it represents one sample.

[minyue-webrtc, 2015/08/13 11:31:15]

Done.

+    for (int i = 0; i < 32000; i++) {
+      // Write 1 second, but it does not matter since the file will be looped.
+      fwrite(&temp, 2, 1, fid);
+    }
+    fclose(fid);
+  }
+
+  static const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  static const webrtc::FileFormats kInputFormat =
+      webrtc::kFileFormatPcm32kHzFile;
 }
 
 namespace voetest {
 
 TEST(VoeConferenceTest, RttAndStartNtpTime) {
   struct Stats {
     Stats(int64_t rtt_receiver_1, int64_t rtt_receiver_2, int64_t ntp_delay)
         : rtt_receiver_1_(rtt_receiver_1),
           rtt_receiver_2_(rtt_receiver_2),
           ntp_delay_(ntp_delay) {
     }
     int64_t rtt_receiver_1_;
     int64_t rtt_receiver_2_;
     int64_t ntp_delay_;
   };
 
   const int kDelayMs = 987;
   ConferenceTransport trans;
   trans.SetRtt(kRttMs);
 
-  unsigned int id_1 = trans.AddStream();
-  unsigned int id_2 = trans.AddStream();
+  unsigned int id_1 = trans.AddStream(kInputFileName, kInputFormat);
+  unsigned int id_2 = trans.AddStream(kInputFileName, kInputFormat);
 
   EXPECT_TRUE(trans.StartPlayout(id_1));
   // Start NTP time is the time when a stream is played out, rather than
   // when it is added.
   webrtc::SleepMs(kDelayMs);
   EXPECT_TRUE(trans.StartPlayout(id_2));
 
   const int kMaxRunTimeMs = 25000;
   const int kNeedSuccessivePass = 3;
   const int kStatsRequestIntervalMs = 1000;
@@ skipping 41 matching lines @@
     printf("The most recent values (RTT for receiver 1, RTT for receiver 2, "
         "NTP delay between receiver 1 and 2) are (from oldest):\n");
     while (!stats_buffer.empty()) {
       Stats stats = stats_buffer.front();
       printf("(%" PRId64 ", %" PRId64 ", %" PRId64 ")\n", stats.rtt_receiver_1_,
              stats.rtt_receiver_2_, stats.ntp_delay_);
       stats_buffer.pop();
     }
   }
 }
+
+
+TEST(VoeConferenceTest, ReceivedPackets) {
+  const int kPackets = 50;
+  const int kPacketDurationMs = 20;  // Correspond to Opus.
+  const std::string silence_file =
+      webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence");
+  CreateSilenceFile(silence_file);
+
+  {
+    ConferenceTransport trans;
+    // Add silence to stream 0, so that it will be filtered out.
+    unsigned int id_0 = trans.AddStream(silence_file, kInputFormat);
+    unsigned int id_1 = trans.AddStream(kInputFileName, kInputFormat);
+    unsigned int id_2 = trans.AddStream(kInputFileName, kInputFormat);
+    unsigned int id_3 = trans.AddStream(kInputFileName, kInputFormat);
+
+    EXPECT_TRUE(trans.StartPlayout(id_0));
+    EXPECT_TRUE(trans.StartPlayout(id_1));
+    EXPECT_TRUE(trans.StartPlayout(id_2));
+    EXPECT_TRUE(trans.StartPlayout(id_3));
+
+    webrtc::SleepMs(kPacketDurationMs * kPackets);
+
+    webrtc::CallStatistics stats_0;
+    webrtc::CallStatistics stats_1;
+    webrtc::CallStatistics stats_2;
+    webrtc::CallStatistics stats_3;
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_0, &stats_0));
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
+    EXPECT_TRUE(trans.GetReceiverStatistics(id_3, &stats_3));
+
+    // We expect stream 0 to be filtered out totally, but since it may join the
+    // call earlier than other streams and the beginning packets might have got
+    // through. So we only expect |packetsReceived| to be close to zero.
+    EXPECT_NEAR(stats_0.packetsReceived, 0, 2);
+    // We expect |packetsReceived| to match |kPackets|, but the actual value
+    // depends on the sleep timer. So we allow a small off from |kPackets|.
+    EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2);
+    EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2);
+    EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2);
+  }
+
+  remove(silence_file.c_str());
+}
+
 }  // namespace voetest