Update NetEq network statistics in neteq_unittest.

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 15 matching lines @@
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "webrtc/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+#endif
+
 DEFINE_bool(gen_ref, false, "Generate reference files.");
 
 namespace webrtc {
 
 static bool IsAllZero(const int16_t* buf, size_t buf_length) {
   bool all_zero = true;
   for (size_t n = 0; n < buf_length && all_zero; ++n)
     all_zero = buf[n] == 0;
   return all_zero;
 }
 
 static bool IsAllNonZero(const int16_t* buf, size_t buf_length) {
   bool all_non_zero = true;
   for (size_t n = 0; n < buf_length && all_non_zero; ++n)
     all_non_zero = buf[n] != 0;
   return all_non_zero;
 }
 
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+static void Convert(const NetEqNetworkStatistics& stats_raw,

[hlundin-webrtc, 2015/12/16 10:04:51]

Please, enclose these methods in an unnamed namespace. And then I don't think
the static qualifier gives you any benefits.
(Bonus points if you also include IsAllZero and IsAllNonZero in the unnamed
namespace and drop the static on those too.)

[minyue-webrtc, 2015/12/16 13:51:26]

Done.

+                    neteq_unittest::NetEqNetworkStatistics* stats) {

[hlundin-webrtc, 2015/12/16 10:04:51]

Should you perhaps name the protobuf representation NetworkStatistics isntead,
since the proto namespace already states that it is neteq. WDYT?

[minyue-webrtc, 2015/12/16 13:51:27]

I actually used NetworkStatistics from beginning and then renamed to
NetEqNetworkStatistics. The reason for it is that the protobuf type
NetEqNetworkStatistics is intended to store (cross-platform) the geniune
NetEqNetworkStatistics. The same name makes that intention clearest.

[hlundin-webrtc, 2015/12/16 14:10:50]

Acknowledged.

+  stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
+  stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
+  stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
+  stats->set_packet_loss_rate(stats_raw.packet_loss_rate);
+  stats->set_packet_discard_rate(stats_raw.packet_discard_rate);
+  stats->set_expand_rate(stats_raw.expand_rate);
+  stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
+  stats->set_preemptive_rate(stats_raw.preemptive_rate);
+  stats->set_accelerate_rate(stats_raw.accelerate_rate);
+  stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
+  stats->set_clockdrift_ppm(stats_raw.clockdrift_ppm);
+  stats->set_added_zero_samples(stats_raw.added_zero_samples);
+  stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
+  stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
+  stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
+  stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
+}
+
+static void Convert(const RtcpStatistics& stats_raw,
+                    neteq_unittest::RtcpStatistics* stats) {
+  stats->set_fraction_lost(stats_raw.fraction_lost);
+  stats->set_cumulative_lost(stats_raw.cumulative_lost);
+  stats->set_extended_max_sequence_number(
+      stats_raw.extended_max_sequence_number);
+  stats->set_jitter(stats_raw.jitter);
+}
+
+static void WriteMessage(FILE* file, const std::string& message) {
+  int32_t size = message.length();
+  ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
+  if (size <= 0)
+    return;
+  ASSERT_EQ(static_cast<size_t>(size),
+            fwrite(message.data(), sizeof(char), size, file));
+}
+
+static void ReadMessage(FILE* file, std::string* message) {
+  int32_t size;
+  ASSERT_EQ(1u, fread(&size, sizeof(size), 1, file));
+  if (size <= 0)
+    return;
+  rtc::scoped_ptr<char[]> buffer(new char[size]);
+  ASSERT_EQ(static_cast<size_t>(size),
+            fread(buffer.get(), sizeof(char), size, file));
+  message->assign(buffer.get(), size);
+}
+

[hlundin-webrtc, 2015/12/16 10:04:51]

Close the #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT here.

[minyue-webrtc, 2015/12/16 13:51:27]

Done.

 class RefFiles {
  public:
   RefFiles(const std::string& input_file, const std::string& output_file);
   ~RefFiles();
   template<class T> void ProcessReference(const T& test_results);
   template<typename T, size_t n> void ProcessReference(
       const T (&test_results)[n],
       size_t length);
   template<typename T, size_t n> void WriteToFile(
       const T (&test_results)[n],
@@ skipping 57 matching lines @@
   if (input_fp_) {
     // Read from ref file.
     T* ref = new T[length];
     ASSERT_EQ(length, fread(ref, sizeof(T), length, input_fp_));
     // Compare
     ASSERT_EQ(0, memcmp(&test_results, ref, sizeof(T) * length));
     delete [] ref;
   }
 }
 
-void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats) {
-  if (output_fp_) {
-    ASSERT_EQ(1u, fwrite(&stats, sizeof(NetEqNetworkStatistics), 1,
-                         output_fp_));
-  }
+void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats_raw) {

[hlundin-webrtc, 2015/12/16 10:04:51]

Provide dummy implementations of WriteToFile and ReadFromFileAndCompare when
WEBRTC_NETEQ_UNITTEST_BITEXACT is not defined.

void RefFiles::WriteToFile(const NetEqNetworkStatistics& stats_raw) {
  FAIL();
}

[minyue-webrtc, 2015/12/16 13:51:26]

Done.

+  if (!output_fp_)
+    return;
+  neteq_unittest::NetEqNetworkStatistics stats;
+  Convert(stats_raw, &stats);
+
+  std::string stats_string;
+  ASSERT_TRUE(stats.SerializeToString(&stats_string));
+  WriteMessage(output_fp_, stats_string);
 }
 
 void RefFiles::ReadFromFileAndCompare(
     const NetEqNetworkStatistics& stats) {
-  // TODO(minyue): Update resource/audio_coding/neteq_network_stats.dat and
-  // resource/audio_coding/neteq_network_stats_win32.dat.
-  struct NetEqNetworkStatisticsOld {
-    uint16_t current_buffer_size_ms;  // Current jitter buffer size in ms.
-    uint16_t preferred_buffer_size_ms;  // Target buffer size in ms.
-    uint16_t jitter_peaks_found;  // 1 if adding extra delay due to peaky
-                                  // jitter; 0 otherwise.
-    uint16_t packet_loss_rate;  // Loss rate (network + late) in Q14.
-    uint16_t packet_discard_rate;  // Late loss rate in Q14.
-    uint16_t expand_rate;  // Fraction (of original stream) of synthesized
-                           // audio inserted through expansion (in Q14).
-    uint16_t preemptive_rate;  // Fraction of data inserted through pre-emptive
-                               // expansion (in Q14).
-    uint16_t accelerate_rate;  // Fraction of data removed through acceleration
-                               // (in Q14).
-    int32_t clockdrift_ppm;  // Average clock-drift in parts-per-million
-                             // (positive or negative).
-    int added_zero_samples;  // Number of zero samples added in "off" mode.
-  };
-  if (input_fp_) {
-    // Read from ref file.
-    size_t stat_size = sizeof(NetEqNetworkStatisticsOld);
-    NetEqNetworkStatisticsOld ref_stats;
-    ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
-    // Compare
-    ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms);
-    ASSERT_EQ(stats.preferred_buffer_size_ms,
-              ref_stats.preferred_buffer_size_ms);
-    ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found);
-    ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate);
-    ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate);
-    ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate);
-    ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate);
-    ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate);
-    ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm);
-    ASSERT_EQ(stats.added_zero_samples,
-              static_cast<size_t>(ref_stats.added_zero_samples));
-    ASSERT_EQ(stats.secondary_decoded_rate, 0);
-    ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate);
-  }
+  if (!input_fp_)
+    return;
+
+  std::string stats_string;
+  ReadMessage(input_fp_, &stats_string);
+  neteq_unittest::NetEqNetworkStatistics ref_stats;
+  ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
+
+  // Compare
+  ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms());
+  ASSERT_EQ(stats.preferred_buffer_size_ms,
+            ref_stats.preferred_buffer_size_ms());
+  ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found());
+  ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate());
+  ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate());
+  ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate());
+  ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate());
+  ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate());
+  ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm());
+  ASSERT_EQ(stats.added_zero_samples, ref_stats.added_zero_samples());
+  ASSERT_EQ(stats.secondary_decoded_rate, 0);
+  ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate());
 }
 
-void RefFiles::WriteToFile(const RtcpStatistics& stats) {
-  if (output_fp_) {
-    ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
-                         output_fp_));
-    ASSERT_EQ(1u, fwrite(&(stats.cumulative_lost),
-                         sizeof(stats.cumulative_lost), 1, output_fp_));
-    ASSERT_EQ(1u, fwrite(&(stats.extended_max_sequence_number),
-                         sizeof(stats.extended_max_sequence_number), 1,
-                         output_fp_));
-    ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
-                         output_fp_));
-  }
+void RefFiles::WriteToFile(const RtcpStatistics& stats_raw) {
+  if (!output_fp_)
+    return;
+  neteq_unittest::RtcpStatistics stats;
+  Convert(stats_raw, &stats);
+
+  std::string stats_string;
+  ASSERT_TRUE(stats.SerializeToString(&stats_string));
+  WriteMessage(output_fp_, stats_string);
 }
 
 void RefFiles::ReadFromFileAndCompare(
     const RtcpStatistics& stats) {
-  if (input_fp_) {
-    // Read from ref file.
-    RtcpStatistics ref_stats;
-    ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
-                        sizeof(ref_stats.fraction_lost), 1, input_fp_));
-    ASSERT_EQ(1u, fread(&(ref_stats.cumulative_lost),
-                        sizeof(ref_stats.cumulative_lost), 1, input_fp_));
-    ASSERT_EQ(1u, fread(&(ref_stats.extended_max_sequence_number),
-                        sizeof(ref_stats.extended_max_sequence_number), 1,
-                        input_fp_));
-    ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1,
-                        input_fp_));
-    // Compare
-    ASSERT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
-    ASSERT_EQ(ref_stats.cumulative_lost, stats.cumulative_lost);
-    ASSERT_EQ(ref_stats.extended_max_sequence_number,
-              stats.extended_max_sequence_number);
-    ASSERT_EQ(ref_stats.jitter, stats.jitter);
-  }
+  if (!input_fp_)
+    return;
+
+  std::string stats_string;
+  ReadMessage(input_fp_, &stats_string);
+  neteq_unittest::RtcpStatistics ref_stats;
+  ASSERT_TRUE(ref_stats.ParseFromString(stats_string));
+
+  // Compare
+  ASSERT_EQ(stats.fraction_lost, ref_stats.fraction_lost());
+  ASSERT_EQ(stats.cumulative_lost, ref_stats.cumulative_lost());
+  ASSERT_EQ(stats.extended_max_sequence_number,
+            ref_stats.extended_max_sequence_number());
+  ASSERT_EQ(stats.jitter, ref_stats.jitter());
 }
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 
 class NetEqDecodingTest : public ::testing::Test {
  protected:
   // NetEQ must be polled for data once every 10 ms. Thus, neither of the
   // constants below can be changed.
   static const int kTimeStepMs = 10;
   static const size_t kBlockSize8kHz = kTimeStepMs * 8;
   static const size_t kBlockSize16kHz = kTimeStepMs * 16;
   static const size_t kBlockSize32kHz = kTimeStepMs * 32;
   static const size_t kMaxBlockSize = kBlockSize32kHz;
   static const int kInitSampleRateHz = 8000;
 
   NetEqDecodingTest();
   virtual void SetUp();
   virtual void TearDown();
   void SelectDecoders(NetEqDecoder* used_codec);
   void LoadDecoders();
   void OpenInputFile(const std::string &rtp_file);
   void Process(size_t* out_len);
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT

[hlundin-webrtc, 2015/12/16 10:04:51]

Remove this #ifdef.

[minyue-webrtc, 2015/12/16 13:51:27]

Done.

   void DecodeAndCompare(const std::string& rtp_file,
                         const std::string& ref_file,
                         const std::string& stat_ref_file,
                         const std::string& rtcp_ref_file);
+#endif
+
   static void PopulateRtpInfo(int frame_index,
                               int timestamp,
                               WebRtcRTPHeader* rtp_info);
   static void PopulateCng(int frame_index,
                           int timestamp,
                           WebRtcRTPHeader* rtp_info,
                           uint8_t* payload,
                           size_t* payload_len);
 
   void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp,
@@ skipping 119 matching lines @@
   ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
               (*out_len == kBlockSize16kHz) ||
               (*out_len == kBlockSize32kHz));
   output_sample_rate_ = static_cast<int>(*out_len / 10 * 1000);
   EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
 
   // Increase time.
   sim_clock_ += kTimeStepMs;
 }
 
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT

[hlundin-webrtc, 2015/12/16 10:04:51]

Remove this #ifdef.

[minyue-webrtc, 2015/12/16 13:51:26]

Done.

 void NetEqDecodingTest::DecodeAndCompare(const std::string& rtp_file,
                                          const std::string& ref_file,
                                          const std::string& stat_ref_file,
                                          const std::string& rtcp_ref_file) {
   OpenInputFile(rtp_file);
 
   std::string ref_out_file = "";
   if (ref_file.empty()) {
     ref_out_file = webrtc::test::OutputPath() + "neteq_universal_ref.pcm";
   }
@@ skipping 31 matching lines @@
       // Compare with CurrentDelay, which should be identical.
       EXPECT_EQ(network_stats.current_buffer_size_ms, neteq_->CurrentDelayMs());
 
       // Process RTCPstat.
       RtcpStatistics rtcp_stats;
       neteq_->GetRtcpStatistics(&rtcp_stats);
       ASSERT_NO_FATAL_FAILURE(rtcp_stat_files.ProcessReference(rtcp_stats));
     }
   }
 }
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+
 
 void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
                                         int timestamp,
                                         WebRtcRTPHeader* rtp_info) {
   rtp_info->header.sequenceNumber = frame_index;
   rtp_info->header.timestamp = timestamp;
   rtp_info->header.ssrc = 0x1234;  // Just an arbitrary SSRC.
   rtp_info->header.payloadType = 94;  // PCM16b WB codec.
   rtp_info->header.markerBit = 0;
 }
@@ skipping 12 matching lines @@
   *payload_len = 1;  // Only noise level, no spectral parameters.
 }
 
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
 #define IF_ALL_CODECS(x) x
 #else
 #define IF_ALL_CODECS(x) DISABLED_##x
 #endif
 
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT

[hlundin-webrtc, 2015/12/16 10:04:51]

The preferred way of disabling tests is to do the MAYBE_TestName thing. These
tests don't follow that, and instead use the DIABLED_ON_ANDROID/IOS macros.
However, those are deprecated and error prone, and should be deleted; see
https://bugs.chromium.org/p/webrtc/issues/detail?id=5333.

I suggest the following:
#if defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && !defined(WEBRTC_ANDROID) &&
!defined(WEBRTC_IOS)
#define MAYBE_TestBitExactness TestBitExactness
#else
#define MAYBE_TestBitExactness DISABLED_TestBitExactness
#endif

Then the test will always compile, but will not run. And, if someone runs it
anyway (e.g., --gtest_also_run_disabled_tests), then the dummy implementations
suggested above will FAIL().

[minyue-webrtc, 2015/12/16 13:51:26]

Done.

 TEST_F(NetEqDecodingTest,
        DISABLED_ON_IOS(DISABLED_ON_ANDROID(IF_ALL_CODECS(TestBitExactness)))) {
   const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
       "resources/audio_coding/neteq_universal_new.rtp";
   // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
   // are identical. The latter could have been removed, but if clients still
   // have a copy of the file, the test will fail.
   const std::string input_ref_file =
       webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
 #if defined(_MSC_VER) && (_MSC_VER >= 1700)
@@ skipping 10 matching lines @@
 
   if (FLAGS_gen_ref) {
     DecodeAndCompare(input_rtp_file, "", "", "");
   } else {
     DecodeAndCompare(input_rtp_file,
                      input_ref_file,
                      network_stat_ref_file,
                      rtcp_stat_ref_file);
   }
 }
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
 
 // Use fax mode to avoid time-scaling. This is to simplify the testing of
 // packet waiting times in the packet buffer.
 class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
  protected:
   NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
     config_.playout_mode = kPlayoutFax;
   }
 };
 
@@ skipping 1043 matching lines @@
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
   // Verify speech output.
   EXPECT_EQ(kOutputNormal, type);
 }
 
 }  // namespace webrtc