Re-enable Opus complexity tests on Android

webrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc

 /*
  *  Copyright (c) 2016 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/format_macros.h"
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 namespace {
 int64_t RunComplexityTest(const AudioEncoderOpus::Config& config) {
   // Create encoder.
   AudioEncoderOpus encoder(config);
   // Open speech file.
   const std::string kInputFileName =
       webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");

[hlundin-webrtc, 2016/12/19 12:54:15]

This is where the resource file is used.

   test::AudioLoop audio_loop;
   constexpr int kSampleRateHz = 48000;
   EXPECT_EQ(kSampleRateHz, encoder.SampleRateHz());
   constexpr size_t kMaxLoopLengthSamples =
       kSampleRateHz * 10;  // 10 second loop.
   constexpr size_t kInputBlockSizeSamples =
       10 * kSampleRateHz / 1000;  // 60 ms.
   EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
                               kInputBlockSizeSamples));
   // Encode.
   webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
   const int64_t start_time_ms = clock->TimeInMilliseconds();
   AudioEncoder::EncodedInfo info;
   rtc::Buffer encoded(500);
   uint32_t rtp_timestamp = 0u;
   for (size_t i = 0; i < 10000; ++i) {
     encoded.Clear();
     info = encoder.Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
     rtp_timestamp += kInputBlockSizeSamples;
   }
   return clock->TimeInMilliseconds() - start_time_ms;
 }
 }  // namespace
 
-#if defined(WEBRTC_ANDROID)
-#define MAYBE_AudioEncoderOpusComplexityAdaptationTest \
-  DISABLED_AudioEncoderOpusComplexityAdaptationTest
-#else
-#define MAYBE_AudioEncoderOpusComplexityAdaptationTest \
-  AudioEncoderOpusComplexityAdaptationTest
-#endif
-
 // This test encodes an audio file using Opus twice with different bitrates
 // (12.5 kbps and 15.5 kbps). The runtime for each is measured, and the ratio
 // between the two is calculated and tracked. This test explicitly sets the
 // low_rate_complexity to 9. When running on desktop platforms, this is the same
 // as the regular complexity, and the expectation is that the resulting ratio
 // should be less than 100% (since the encoder runs faster at lower bitrates,
 // given a fixed complexity setting). On the other hand, when running on
 // mobiles, the regular complexity is 5, and we expect the resulting ratio to
 // be higher, since we have explicitly asked for a higher complexity setting at
 // the lower rate.
-TEST(MAYBE_AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) {
+TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) {
   // Create config.
   AudioEncoderOpus::Config config;
-  config.bitrate_bps = rtc::Optional<int>(12500);
+  // The limit -- including the hysteresis window -- at which the complexity
+  // shuold be increased.
+  config.bitrate_bps = rtc::Optional<int>(11000 - 1);
   config.low_rate_complexity = 9;
   int64_t runtime_12500bps = RunComplexityTest(config);
 
   config.bitrate_bps = rtc::Optional<int>(15500);
   int64_t runtime_15500bps = RunComplexityTest(config);
 
   test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_on",
                     100.0 * runtime_12500bps / runtime_15500bps, "percent",
                     true);
 }
 
 // This test is identical to the one above, but without the complexity
 // adaptation enabled (neither on desktop, nor on mobile). The expectation is
 // that the resulting ratio is less than 100% at all times.
-TEST(MAYBE_AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) {
+TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) {
   // Create config.
   AudioEncoderOpus::Config config;
-  config.bitrate_bps = rtc::Optional<int>(12500);
+  // The limit -- including the hysteresis window -- at which the complexity
+  // shuold be increased (but not in this test since complexity adaptation is
+  // disabled).
+  config.bitrate_bps = rtc::Optional<int>(11000 - 1);
   int64_t runtime_12500bps = RunComplexityTest(config);

[minyue-webrtc, 2016/12/19 13:50:35]

still 12500?

[hlundin-webrtc, 2016/12/19 13:55:16]

No. My bad. Changed the variable name to runtime_10999bps instead.

 
   config.bitrate_bps = rtc::Optional<int>(15500);
   int64_t runtime_15500bps = RunComplexityTest(config);
 
   test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off",
-                    100.0 * runtime_12500bps / runtime_15500bps, "", true);
+                    100.0 * runtime_12500bps / runtime_15500bps, "percent",

[hlundin-webrtc, 2016/12/19 12:54:15]

This was missing; added now to match line 75.

+                    true);
 }
 }  // namespace webrtc