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 "api/audio_codecs/opus/audio_encoder_opus.h"

#include "api/environment/environment_factory.h"

#include "api/test/metrics/global_metrics_logger_and_exporter.h"

#include "api/test/metrics/metric.h"

#include "modules/audio_coding/neteq/tools/audio_loop.h"

#include "rtc_base/time_utils.h"

#include "test/gtest.h"

#include "test/testsupport/file_utils.h"

namespace webrtc {

namespace {

using ::webrtc::test::GetGlobalMetricsLogger;

using ::webrtc::test::ImprovementDirection;

using ::webrtc::test::Unit;

int64_t RunComplexityTest(const Environment& env,

                          const AudioEncoderOpusConfig& config) {

  // Create encoder.

  const auto encoder =

      AudioEncoderOpus::MakeAudioEncoder(env, config, {.payload_type = 17});

  // 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.

  EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,

                              kInputBlockSizeSamples));

  // Encode.

  const int64_t start_time_ms = rtc::TimeMillis();

  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 rtc::TimeMillis() - start_time_ms;

// This test encodes an audio file using Opus twice with different bitrates

// (~11 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(AudioEncoderOpusComplexityAdaptationTest, Adaptation_On) {

  const Environment env = CreateEnvironment();

  // Create config.

  AudioEncoderOpusConfig config;

  // The limit -- including the hysteresis window -- at which the complexity

  // shuold be increased.

  config.bitrate_bps = 11000 - 1;

  config.low_rate_complexity = 9;

  int64_t runtime_10999bps = RunComplexityTest(env, config);

  config.bitrate_bps = 15500;

  int64_t runtime_15500bps = RunComplexityTest(env, config);

  GetGlobalMetricsLogger()->LogSingleValueMetric(

      "opus_encoding_complexity_ratio", "adaptation_on",

      100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,

      ImprovementDirection::kNeitherIsBetter);

// 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, Adaptation_Off) {

  const Environment env = CreateEnvironment();

  // Create 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 = 11000 - 1;

  int64_t runtime_10999bps = RunComplexityTest(env, config);

  config.bitrate_bps = 15500;

  int64_t runtime_15500bps = RunComplexityTest(env, config);

  GetGlobalMetricsLogger()->LogSingleValueMetric(

      "opus_encoding_complexity_ratio", "adaptation_off",

      100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,

      ImprovementDirection::kNeitherIsBetter);

}  // namespace

}  // namespace webrtc