Add NetEqPcmuQualityTest

This is virtually the same as NetEq{Isac,Opus}QualityTest but for PCMu.

BUG=2692
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54379004

Cr-Commit-Position: refs/heads/master@{#9176}

webrtc/modules/audio_coding/neteq/neteq_tests.gypi

@@ -220,6 +220,22 @@
       ],
     },
 
+    {
+      'target_name': 'neteq_pcmu_quality_test',
+      'type': 'executable',
+      'dependencies': [
+        'neteq',
+        'neteq_test_support',
+        'G711',
+        '<(DEPTH)/testing/gtest.gyp:gtest',
+        '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
+        '<(webrtc_root)/test/test.gyp:test_support_main',
+      ],
+      'sources': [
+        'test/neteq_pcmu_quality_test.cc',
+      ],
+    },
+
     {
      'target_name': 'neteq_test_tools',
       # Collection of useful functions used in other tests.

webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc

@@ -36,19 +36,6 @@ DEFINE_int32(bit_rate_kbps, 32, "Target bit rate (kbps).");
 static const bool bit_rate_dummy =
     RegisterFlagValidator(&FLAGS_bit_rate_kbps, &ValidateBitRate);
 
-// Define switch for runtime.
-static bool ValidateRuntime(const char* flagname, int32_t value) {
-  if (value > 0)
-    return true;
-  printf("Invalid runtime, should be greater than 0.");
-  return false;
-}
-
-DEFINE_int32(runtime_ms, 10000, "Simulated runtime (milliseconds).");
-
-static const bool runtime_dummy =
-    RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
-
 }  // namespace
 
 class NetEqIsacQualityTest : public NetEqQualityTest {
@@ -110,7 +97,7 @@ int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data,
 }
 
 TEST_F(NetEqIsacQualityTest, Test) {
-  Simulate(FLAGS_runtime_ms);
+  Simulate();
 }
 
 }  // namespace test

webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc

@@ -78,18 +78,6 @@ DEFINE_int32(reported_loss_rate, 10, "Reported percentile of packet loss.");
 static const bool reported_loss_rate_dummy =
     RegisterFlagValidator(&FLAGS_reported_loss_rate, &ValidatePacketLossRate);
 
-// Define switch for runtime.
-static bool ValidateRuntime(const char* flagname, int32_t value) {
-  if (value > 0)
-    return true;
-  printf("Invalid runtime, should be greater than 0.");
-  return false;
-}
-
-DEFINE_int32(runtime_ms, 10000, "Simulated runtime (milliseconds).");
-static const bool runtime_dummy =
-    RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
-
 DEFINE_bool(fec, true, "Whether to enable FEC for encoding.");
 
 DEFINE_bool(dtx, true, "Whether to enable DTX for encoding.");
@@ -197,7 +185,7 @@ int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
 }
 
 TEST_F(NetEqOpusQualityTest, Test) {
-  Simulate(FLAGS_runtime_ms);
+  Simulate();
 }
 
 }  // namespace test

webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc

@@ -0,0 +1,82 @@
+/*
+ *  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 "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+using google::RegisterFlagValidator;
+using google::ParseCommandLineFlags;
+using std::string;
+using testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+
+// Define switch for frame size.
+static bool ValidateFrameSize(const char* flagname, int32_t value) {
+  if (value >= 10 && value <= 60 && (value % 10) == 0)
+    return true;
+  printf("Invalid frame size, should be 10, 20, ..., 60 ms.");
+  return false;
+}
+
+DEFINE_int32(frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+static const bool frame_size_dummy =
+    RegisterFlagValidator(&FLAGS_frame_size_ms, &ValidateFrameSize);
+
+}  // namespace
+
+class NetEqPcmuQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqPcmuQualityTest()
+      : NetEqQualityTest(FLAGS_frame_size_ms,
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         kDecoderPCMu,
+                         1) {
+    AudioEncoderPcmU::Config config;
+    config.frame_size_ms = FLAGS_frame_size_ms;
+    encoder_.reset(new AudioEncoderPcmU(config));
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  int block_size_samples,
+                  uint8_t* payload,
+                  int max_bytes) override {
+    const int kFrameSizeSamples = 80;  // Samples per 10 ms.
+    int encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp, &in_data[encoded_samples],
+                              kFrameSizeSamples, max_bytes, payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  rtc::scoped_ptr<AudioEncoderPcmU> encoder_;
+};
+
+TEST_F(NetEqPcmuQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc

@@ -90,6 +90,19 @@ static bool ValidatePacketLossRate(const char* /* flag_name */, int32_t value) {
   return false;
 }
 
+// Define switch for runtime.
+static bool ValidateRuntime(const char* flagname, int32_t value) {
+  if (value > 0)
+    return true;
+  printf("Invalid runtime, should be greater than 0.");
+  return false;
+}
+
+DEFINE_int32(runtime_ms, 10000, "Simulated runtime (milliseconds).");
+
+static const bool runtime_dummy =
+    RegisterFlagValidator(&FLAGS_runtime_ms, &ValidateRuntime);
+
 DEFINE_int32(packet_loss_rate, 10, "Percentile of packet loss.");
 
 static const bool packet_loss_rate_dummy =
@@ -368,10 +381,10 @@ int NetEqQualityTest::DecodeBlock() {
   }
 }
 
-void NetEqQualityTest::Simulate(int end_time_ms) {
+void NetEqQualityTest::Simulate() {
   int audio_size_samples;
 
-  while (decoded_time_ms_ < end_time_ms) {
+  while (decoded_time_ms_ < FLAGS_runtime_ms) {
     // Assume 10 packets in packets buffer.
     while (decodable_time_ms_ - 10 * block_duration_ms_ < decoded_time_ms_) {
       ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0]));
@@ -386,7 +399,7 @@ void NetEqQualityTest::Simulate(int end_time_ms) {
       decoded_time_ms_ += audio_size_samples / out_sampling_khz_;
     }
   }
-  fprintf(log_file_, "%f", 8.0f * total_payload_size_bytes_ / end_time_ms);
+  fprintf(log_file_, "%f", 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms);
 }
 
 }  // namespace test

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h

@@ -90,10 +90,8 @@ class NetEqQualityTest : public ::testing::Test {
   // |neteq_|.
   int Transmit();
 
-  // Simulate(...) runs encoding / transmitting / decoding up to |end_time_ms|
-  // (miliseconds), the resulted audio is stored in the file with the name of
-  // |out_filename_|.
-  void Simulate(int end_time_ms);
+  // Runs encoding / transmitting / decoding.
+  void Simulate();
 
  private:
   int decoded_time_ms_;