Opus implementation of the AudioEncoderFactoryTemplate API

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/base/timeutils.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"
 
 namespace webrtc {
 
 namespace {
-int64_t RunComplexityTest(const AudioEncoderOpus::Config& config) {
+int64_t RunComplexityTest(const AudioEncoderOpusConfig& config) {
   // Create encoder.
-  AudioEncoderOpus encoder(config);
+  constexpr int payload_type = 17;
+  AudioEncoderOpus encoder(config, payload_type);
   // Open speech file.
   const std::string kInputFileName =
       webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
   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.
@@ skipping 18 matching lines @@
 // 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(AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) {
   // Create config.
-  AudioEncoderOpus::Config config;
+  AudioEncoderOpusConfig config;
   // 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_10999bps = 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_10999bps / 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(AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) {
   // Create config.
-  AudioEncoderOpus::Config config;
+  AudioEncoderOpusConfig config;
   // 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_10999bps = RunComplexityTest(config);
 
   config.bitrate_bps = rtc::Optional<int>(15500);
   int64_t runtime_15500bps = RunComplexityTest(config);
 
   test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off",
                     100.0 * runtime_10999bps / runtime_15500bps, "percent",
                     true);
 }
 }  // namespace webrtc