Replace voe_conference_test.

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
deleted file mode 100644
index 9e466afdaa7ac45c7633a8026e3265ce2444a22e..0000000000000000000000000000000000000000
--- a/webrtc/voice_engine/test/auto_test/voe_conference_test.cc
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- *  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 "webrtc/rtc_base/format_macros.h"
-#include "webrtc/rtc_base/timeutils.h"
-#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
-
-namespace webrtc {
-namespace {
-
-const int kRttMs = 25;
-
-bool IsNear(int ref, int comp, int error) {
-  return (ref - comp <= error) && (comp - ref >= -error);
-}
-
-void CreateSilenceFile(const std::string& silence_file, int sample_rate_hz) {
-  FILE* fid = fopen(silence_file.c_str(), "wb");
-  int16_t zero = 0;
-  for (int i = 0; i < sample_rate_hz; ++i) {
-    // Write 1 second, but it does not matter since the file will be looped.
-    fwrite(&zero, sizeof(int16_t), 1, fid);
-  }
-  fclose(fid);
-}
-
-}  // namespace
-
-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 std::string input_file =
-      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
-  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
-
-  const int kDelayMs = 987;
-  ConferenceTransport trans;
-  trans.SetRtt(kRttMs);
-
-  unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
-  unsigned int id_2 = trans.AddStream(input_file, 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;
-  const int kStatsBufferSize = 3;
-
-  int64_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
-  // Run the following up to |kMaxRunTimeMs| milliseconds.
-  int successive_pass = 0;
-  webrtc::CallStatistics stats_1;
-  webrtc::CallStatistics stats_2;
-  std::queue<Stats> stats_buffer;
-
-  while (rtc::TimeMillis() < deadline &&
-         successive_pass < kNeedSuccessivePass) {
-    webrtc::SleepMs(kStatsRequestIntervalMs);
-
-    EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
-    EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
-
-    // It is not easy to verify the NTP time directly. We verify it by testing
-    // the difference of two start NTP times.
-    int64_t captured_start_ntp_delay = stats_2.capture_start_ntp_time_ms_ -
-        stats_1.capture_start_ntp_time_ms_;
-
-    // For the checks of RTT and start NTP time, We allow 10% accuracy.
-    if (IsNear(kRttMs, stats_1.rttMs, kRttMs / 10 + 1) &&
-        IsNear(kRttMs, stats_2.rttMs, kRttMs / 10 + 1) &&
-        IsNear(kDelayMs, captured_start_ntp_delay, kDelayMs / 10 + 1)) {
-      successive_pass++;
-    } else {
-      successive_pass = 0;
-    }
-    if (stats_buffer.size() >= kStatsBufferSize) {
-      stats_buffer.pop();
-    }
-    stats_buffer.push(Stats(stats_1.rttMs, stats_2.rttMs,
-                            captured_start_ntp_delay));
-  }
-
-  EXPECT_GE(successive_pass, kNeedSuccessivePass) << "Expected to get RTT and"
-      " start NTP time estimate within 10% of the correct value over "
-      << kStatsRequestIntervalMs * kNeedSuccessivePass / 1000
-      << " seconds.";
-  if (successive_pass < kNeedSuccessivePass) {
-    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 input_file =
-      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
-  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
-
-  const std::string silence_file =
-      webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence");
-  CreateSilenceFile(silence_file, 32000);
-
-  {
-    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(input_file, kInputFormat);
-    unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
-    unsigned int id_3 = trans.AddStream(input_file, 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
-}  // namespace webrtc