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);
   }
 }
 
 namespace voetest {
 
 TEST(VoeConferenceTest, RttAndStartNtpTime) {
+  const std::string kInputFileName =

[Andrew MacDonald, 2015/08/05 16:37:07]

nit: not a compile-time const.

[minyue-webrtc, 2015/08/06 13:31:22]

Done.

[minyue-webrtc, 2015/08/06 14:57:14]

Hi Andrew,

Per offline talk with Tina, I think I have not addressed your comment here
correctly. Do you suggest the that the name should not have a k-?

[Andrew MacDonald, 2015/08/13 19:48:11]

Right, it's not a compile-time const (i.e. can only be evaluated at run-time),
so should use input_file_name.

+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
+
   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 std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
+  const int kPackets = 50;
+  const int kPacketDurationMs = 20;  // Correspond to Opus.
+
+  ConferenceTransport trans;
+  // Add silence to stream 0, so that it will be filtered out.
+  unsigned int id_0 = trans.AddStream(
+      webrtc::test::ResourcePath("audio_coding/silence", "pcm"),
+      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));
+
+  // Cannot be accurate since stream 0 started the earliest.

[tlegrand-webrtc, 2015/08/05 13:32:44]

Can you explain this comment? I'm not following.

[minyue-webrtc, 2015/08/06 13:31:22]

I have updated the comments.

[tlegrand-webrtc, 2015/08/06 14:51:55]

Acknowledged.

+  EXPECT_NEAR(stats_0.packetsReceived, 0, 2);
+  // Cannot be accurate since it replies on the sleep timer.
+  EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2);
+  EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2);
+  EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2);
+}
+
 }  // namespace voetest