NetEq: Convert AverageIAT from int to float calculations

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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.
  */
 
@@ skipping 430 matching lines @@
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
 
   const std::string output_checksum = PlatformChecksum(
       "acd33f5c73625c1529c412ad59b5565132826f1b",
       "1a2e82a0410421c1d1d3eb0615334db5e2c63784",
       "acd33f5c73625c1529c412ad59b5565132826f1b",
       "52797b781758a1d2303140b80b9c5030c9093d6b");
 
   const std::string network_stats_checksum = PlatformChecksum(
-      "9c5bb9e74a583be89313b158a19ea10d41bf9de6",
-      "e948ec65cf18852ba2a197189a3186635db34c3b",
-      "9c5bb9e74a583be89313b158a19ea10d41bf9de6",
-      "9c5bb9e74a583be89313b158a19ea10d41bf9de6");
+      "f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f",
+      "c8b2a93842e48d014f7e6efe10ae96cb3892b129",
+      "f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f",
+      "f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f");
 
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
       "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
 
   DecodeAndCompare(input_rtp_file,
                    output_checksum,
                    network_stats_checksum,
@@ skipping 12 matching lines @@
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
 
   const std::string output_checksum = PlatformChecksum(
       "9d7d52bc94e941d106aa518f324f16a58d231586",
       "9d7d52bc94e941d106aa518f324f16a58d231586",
       "9d7d52bc94e941d106aa518f324f16a58d231586",
       "9d7d52bc94e941d106aa518f324f16a58d231586");
 
   const std::string network_stats_checksum = PlatformChecksum(
-      "191af29ed3b8b6dd4c4cc94dc3f33bdf48f055ef",
-      "191af29ed3b8b6dd4c4cc94dc3f33bdf48f055ef",
-      "191af29ed3b8b6dd4c4cc94dc3f33bdf48f055ef",
-      "191af29ed3b8b6dd4c4cc94dc3f33bdf48f055ef");
+      "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
+      "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
+      "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
+      "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8");
 
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
 
   DecodeAndCompare(input_rtp_file,
                    output_checksum,
                    network_stats_checksum,
@@ skipping 69 matching lines @@
     }
 
     // Pull out data once.
     bool muted;
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
   NetEqNetworkStatistics network_stats;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
-  EXPECT_EQ(-103196, network_stats.clockdrift_ppm);
+  EXPECT_EQ(-103192, network_stats.clockdrift_ppm);
 }
 
 TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
   const int kNumFrames = 5000;  // Needed for convergence.
   int frame_index = 0;
   const size_t kSamples = 10 * 16;
   const size_t kPayloadBytes = kSamples * 2;
   for (int i = 0; i < kNumFrames; ++i) {
     // Insert one packet each time, except every 10th time where we don't insert
     // any packet. This will create a positive clock-drift of approx. 11%.
     int num_packets = (i % 10 == 9 ? 0 : 1);
     for (int n = 0; n < num_packets; ++n) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
       ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
     bool muted;
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
 
   NetEqNetworkStatistics network_stats;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
-  EXPECT_EQ(110946, network_stats.clockdrift_ppm);
+  EXPECT_EQ(110953, network_stats.clockdrift_ppm);
 }
 
 void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
                                               double network_freeze_ms,
                                               bool pull_audio_during_freeze,
                                               int delay_tolerance_ms,
                                               int max_time_to_speech_ms) {
   uint16_t seq_no = 0;
   uint32_t timestamp = 0;
   const int kFrameSizeMs = 30;
@@ skipping 962 matching lines @@
     if (muted) {
       EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
     } else {
       EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
     }
   }
   EXPECT_FALSE(muted);
 }
 
 }  // namespace webrtc