Add LoudestFilter in ConferenceTransport

webrtc/voice_engine/test/auto_test/voe_conference_test.cc

Index: webrtc/voice_engine/test/auto_test/voe_conference_test.cc
diff --git a/webrtc/voice_engine/test/auto_test/voe_conference_test.cc b/webrtc/voice_engine/test/auto_test/voe_conference_test.cc
index 20a74b46b0880147edae791ffc36161b5be8fae6..ce6332a82b6a7712114e86adf9612e1e136eaef2 100644
--- a/webrtc/voice_engine/test/auto_test/voe_conference_test.cc
+++ b/webrtc/voice_engine/test/auto_test/voe_conference_test.cc
@@ -14,6 +14,7 @@
 #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 {
@@ -22,6 +23,10 @@ namespace {
   static bool IsNear(int ref, int comp, int error) {
     return (ref - comp <= error) && (comp - ref >= -error);
   }
+  static const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  static const webrtc::FileFormats kInputFormat =
+      webrtc::kFileFormatPcm32kHzFile;
 }
 
 namespace voetest {
@@ -42,8 +47,8 @@ TEST(VoeConferenceTest, RttAndStartNtpTime) {
   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
@@ -105,4 +110,58 @@ TEST(VoeConferenceTest, RttAndStartNtpTime) {
     }
   }
 }
+
+
+TEST(VoeConferenceTest, ReceivedPackets) {
+  const int kPackets = 50;
+  const int kPacketDurationMs = 20;  // Correspond to Opus.
+
+  const std::string silence_file =

[phoglund, 2015/08/10 08:43:28]

Extract this to a helper method CreateTempSilenceFile.

[minyue-webrtc, 2015/08/10 09:30:22]

ok, will do.

+      webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence");
+  FILE* fid = fopen(silence_file.c_str(), "wb");
+  int temp = 0;
+  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);
+
+  {
+    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