modules/audio_mixer/frame_combiner_unittest.cc

/*
 *  Copyright (c) 2017 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/modules/audio_mixer/frame_combiner.h"

#include <numeric>
#include <sstream>
#include <string>

#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/modules/audio_mixer/gain_change_calculator.h"
#include "webrtc/modules/audio_mixer/sine_wave_generator.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/test/gtest.h"

namespace webrtc {

namespace {
std::string ProduceDebugText(int sample_rate_hz,
                             int number_of_channels,
                             int number_of_sources) {
  std::ostringstream ss;
  ss << "Sample rate: " << sample_rate_hz << " ,";
  ss << "number of channels: " << number_of_channels << " ,";
  ss << "number of sources: " << number_of_sources;
  return ss.str();
}

std::string ProduceDebugText(int sample_rate_hz,
                             int number_of_channels,
                             int number_of_sources,
                             bool limiter_active,
                             float wave_frequency) {
  std::ostringstream ss;
  ss << "Sample rate: " << sample_rate_hz << " ,";
  ss << "number of channels: " << number_of_channels << " ,";
  ss << "number of sources: " << number_of_sources << " ,";
  ss << "limiter active: " << (limiter_active ? "true" : "false") << " ,";
  ss << "wave frequency: " << wave_frequency << " ,";
  return ss.str();
}

AudioFrame frame1;
AudioFrame frame2;
AudioFrame audio_frame_for_mixing;

void SetUpFrames(int sample_rate_hz, int number_of_channels) {
  for (auto* frame : {&frame1, &frame2}) {
    frame->UpdateFrame(-1, 0, nullptr,
                       rtc::CheckedDivExact(sample_rate_hz, 100),
                       sample_rate_hz, AudioFrame::kNormalSpeech,
                       AudioFrame::kVadActive, number_of_channels);
  }
}
}  // namespace

TEST(FrameCombiner, BasicApiCallsLimiter) {
  FrameCombiner combiner(true);
  for (const int rate : {8000, 16000, 32000, 48000}) {
    for (const int number_of_channels : {1, 2}) {
      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
      SetUpFrames(rate, number_of_channels);

      for (const int number_of_frames : {0, 1, 2}) {
        SCOPED_TRACE(
            ProduceDebugText(rate, number_of_channels, number_of_frames));
        const std::vector<AudioFrame*> frames_to_combine(
            all_frames.begin(), all_frames.begin() + number_of_frames);
        combiner.Combine(frames_to_combine, number_of_channels, rate,
                         frames_to_combine.size(), &audio_frame_for_mixing);
      }
    }
  }
}

// No APM limiter means no AudioProcessing::NativeRate restriction
// on rate. The rate has to be divisible by 100 since we use
// 10 ms frames, though.
TEST(FrameCombiner, BasicApiCallsNoLimiter) {
  FrameCombiner combiner(false);
  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
    for (const int number_of_channels : {1, 2}) {
      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
      SetUpFrames(rate, number_of_channels);

      for (const int number_of_frames : {0, 1, 2}) {
        SCOPED_TRACE(
            ProduceDebugText(rate, number_of_channels, number_of_frames));
        const std::vector<AudioFrame*> frames_to_combine(
            all_frames.begin(), all_frames.begin() + number_of_frames);
        combiner.Combine(frames_to_combine, number_of_channels, rate,
                         frames_to_combine.size(), &audio_frame_for_mixing);
      }
    }
  }
}

TEST(FrameCombiner, CombiningZeroFramesShouldProduceSilence) {
  FrameCombiner combiner(false);
  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
    for (const int number_of_channels : {1, 2}) {
      SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 0));

      const std::vector<AudioFrame*> frames_to_combine;
      combiner.Combine(frames_to_combine, number_of_channels, rate,
                       frames_to_combine.size(), &audio_frame_for_mixing);

      const int16_t* audio_frame_for_mixing_data =
          audio_frame_for_mixing.data();
      const std::vector<int16_t> mixed_data(
          audio_frame_for_mixing_data,
          audio_frame_for_mixing_data + number_of_channels * rate / 100);

      const std::vector<int16_t> expected(number_of_channels * rate / 100, 0);
      EXPECT_EQ(mixed_data, expected);
    }
  }
}

TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
  FrameCombiner combiner(false);
  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
    for (const int number_of_channels : {1, 2}) {
      SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1));

      SetUpFrames(rate, number_of_channels);
      int16_t* frame1_data = frame1.mutable_data();
      std::iota(frame1_data, frame1_data + number_of_channels * rate / 100, 0);
      const std::vector<AudioFrame*> frames_to_combine = {&frame1};
      combiner.Combine(frames_to_combine, number_of_channels, rate,
                       frames_to_combine.size(), &audio_frame_for_mixing);

      const int16_t* audio_frame_for_mixing_data =
          audio_frame_for_mixing.data();
      const std::vector<int16_t> mixed_data(
          audio_frame_for_mixing_data,
          audio_frame_for_mixing_data + number_of_channels * rate / 100);

      std::vector<int16_t> expected(number_of_channels * rate / 100);
      std::iota(expected.begin(), expected.end(), 0);
      EXPECT_EQ(mixed_data, expected);
    }
  }
}

// Send a sine wave through the FrameCombiner, and check that the
// difference between input and output varies smoothly. This is to
// catch issues like chromium:695993.
TEST(FrameCombiner, GainCurveIsSmoothForAlternatingNumberOfStreams) {
  // Test doesn't work with rates requiring a band split, because it
  // introduces a small delay measured in single samples, and this
  // test cannot handle it.
  //
  // TODO(aleloi): Add more rates when APM limiter doesn't use band
  // split.
  for (const bool use_limiter : {true, false}) {
    for (const int rate : {8000, 16000}) {
      constexpr int number_of_channels = 2;
      for (const float wave_frequency : {50, 400, 3200}) {
        SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1, use_limiter,
                                      wave_frequency));

        FrameCombiner combiner(use_limiter);

        constexpr int16_t wave_amplitude = 30000;
        SineWaveGenerator wave_generator(wave_frequency, wave_amplitude);

        GainChangeCalculator change_calculator;
        float cumulative_change = 0.f;

        constexpr size_t iterations = 100;

        for (size_t i = 0; i < iterations; ++i) {
          SetUpFrames(rate, number_of_channels);
          wave_generator.GenerateNextFrame(&frame1);
          AudioFrameOperations::Mute(&frame2);

          std::vector<AudioFrame*> frames_to_combine = {&frame1};
          if (i % 2 == 0) {
            frames_to_combine.push_back(&frame2);
          }
          const size_t number_of_samples =
              frame1.samples_per_channel_ * number_of_channels;

          // Ensures limiter is on if 'use_limiter'.
          constexpr size_t number_of_streams = 2;
          combiner.Combine(frames_to_combine, number_of_channels, rate,
                           number_of_streams, &audio_frame_for_mixing);
          cumulative_change += change_calculator.CalculateGainChange(
              rtc::ArrayView<const int16_t>(frame1.data(), number_of_samples),
              rtc::ArrayView<const int16_t>(audio_frame_for_mixing.data(),
                                            number_of_samples));
        }
        RTC_DCHECK_LT(cumulative_change, 10);
      }
    }
  }
}
}  // namespace webrtc