modules/audio_mixer/frame_combiner_unittest.cc - src - Git at Google

 /*
  *  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 "modules/audio_mixer/frame_combiner.h"

 #include <cstdint>
 #include <initializer_list>
 #include <numeric>
 #include <string>
 #include <type_traits>
 #include <vector>

 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/rtp_packet_info.h"
 #include "api/rtp_packet_infos.h"
 #include "api/units/timestamp.h"
 #include "audio/utility/audio_frame_operations.h"
 #include "modules/audio_mixer/gain_change_calculator.h"
 #include "modules/audio_mixer/sine_wave_generator.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/strings/string_builder.h"
 #include "test/gmock.h"
 #include "test/gtest.h"

 namespace webrtc {

 namespace {

 using ::testing::ElementsAreArray;
 using ::testing::IsEmpty;
 using ::testing::UnorderedElementsAreArray;

 struct FrameCombinerConfig {
   bool use_limiter;
   int sample_rate_hz;
   int number_of_channels;
   float wave_frequency;
 };

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

 std::string ProduceDebugText(const FrameCombinerConfig& config) {
   rtc::StringBuilder ss;
   ss << "Sample rate: " << config.sample_rate_hz << " ,";
   ss << "number of channels: " << config.number_of_channels << " ,";
   ss << "limiter active: " << (config.use_limiter ? "on" : "off") << " ,";
   ss << "wave frequency: " << config.wave_frequency << " ,";
   return ss.Release();
 }

 AudioFrame frame1;
 AudioFrame frame2;

 void SetUpFrames(int sample_rate_hz, int number_of_channels) {
   RtpPacketInfo packet_info1(/*ssrc=*/1001, /*csrcs=*/{},
                              /*rtp_timestamp=*/1000,
                              /*receive_time=*/Timestamp::Millis(1));
   RtpPacketInfo packet_info2(/*ssrc=*/4004, /*csrcs=*/{},
                              /*rtp_timestamp=*/1234,
                              /*receive_time=*/Timestamp::Millis(2));
   RtpPacketInfo packet_info3(/*ssrc=*/7007, /*csrcs=*/{},
                              /*rtp_timestamp=*/1333,
                              /*receive_time=*/Timestamp::Millis(2));

   frame1.packet_infos_ = RtpPacketInfos({packet_info1});
   frame2.packet_infos_ = RtpPacketInfos({packet_info2, packet_info3});

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

 // The limiter requires sample rate divisible by 2000.
 TEST(FrameCombiner, BasicApiCallsLimiter) {
   FrameCombiner combiner(true);
   for (const int rate : {8000, 18000, 34000, 48000}) {
     for (const int number_of_channels : {1, 2, 4, 8}) {
       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);
         AudioFrame audio_frame_for_mixing;
         combiner.Combine(frames_to_combine, number_of_channels, rate,
                          frames_to_combine.size(), &audio_frame_for_mixing);
       }
     }
   }
 }

 // The RtpPacketInfos field of the mixed packet should contain the union of the
 // RtpPacketInfos from the frames that were actually mixed.
 TEST(FrameCombiner, ContainsAllRtpPacketInfos) {
   static constexpr int kSampleRateHz = 48000;
   static constexpr int kNumChannels = 1;
   FrameCombiner combiner(true);
   const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
   SetUpFrames(kSampleRateHz, kNumChannels);

   for (const int number_of_frames : {0, 1, 2}) {
     SCOPED_TRACE(
         ProduceDebugText(kSampleRateHz, kNumChannels, number_of_frames));
     const std::vector<AudioFrame*> frames_to_combine(
         all_frames.begin(), all_frames.begin() + number_of_frames);

     std::vector<RtpPacketInfo> packet_infos;
     for (const auto& frame : frames_to_combine) {
       packet_infos.insert(packet_infos.end(), frame->packet_infos_.begin(),
                           frame->packet_infos_.end());
     }

     AudioFrame audio_frame_for_mixing;
     combiner.Combine(frames_to_combine, kNumChannels, kSampleRateHz,
                      frames_to_combine.size(), &audio_frame_for_mixing);
     EXPECT_THAT(audio_frame_for_mixing.packet_infos_,
                 UnorderedElementsAreArray(packet_infos));
   }
 }

 #if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 // There are CHECKs in place to check for invalid parameters.
 TEST(FrameCombinerDeathTest, BuildCrashesWithManyChannels) {
   FrameCombiner combiner(true);
   for (const int rate : {8000, 18000, 34000, 48000}) {
     for (const int number_of_channels : {10, 20, 21}) {
       if (static_cast<size_t>(rate / 100 * number_of_channels) >
           AudioFrame::kMaxDataSizeSamples) {
         continue;
       }
       const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
       // With an unsupported channel count, this will crash in
       // `AudioFrame::UpdateFrame`.
       EXPECT_DEATH(SetUpFrames(rate, number_of_channels), "");

       const int number_of_frames = 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);
       AudioFrame audio_frame_for_mixing;
       EXPECT_DEATH(
           combiner.Combine(frames_to_combine, number_of_channels, rate,
                            frames_to_combine.size(), &audio_frame_for_mixing),
           "");
     }
   }
 }
 #endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)

 TEST(FrameCombinerDeathTest, DebugBuildCrashesWithHighRate) {
   FrameCombiner combiner(true);
   for (const int rate : {50000, 96000, 128000, 196000}) {
     for (const int number_of_channels : {1, 2, 3}) {
       if (static_cast<size_t>(rate / 100 * number_of_channels) >
           AudioFrame::kMaxDataSizeSamples) {
         continue;
       }
       const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
       SetUpFrames(rate, number_of_channels);

       const int number_of_frames = 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);
       AudioFrame audio_frame_for_mixing;
 #if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
       EXPECT_DEATH(
           combiner.Combine(frames_to_combine, number_of_channels, rate,
                            frames_to_combine.size(), &audio_frame_for_mixing),
           "")
           << "number_of_channels=" << number_of_channels << ", rate=" << rate
           << ", frames to combine=" << frames_to_combine.size();
 #endif
     }
   }
 }

 // With no limiter, the rate has to be divisible by 100 since we use
 // 10 ms frames.
 TEST(FrameCombiner, BasicApiCallsNoLimiter) {
   FrameCombiner combiner(false);
   for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
     for (const int number_of_channels : {1, 2, 4, 8}) {
       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);
         AudioFrame audio_frame_for_mixing;
         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));

       AudioFrame audio_frame_for_mixing;

       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);
       EXPECT_THAT(audio_frame_for_mixing.packet_infos_, IsEmpty());
     }
   }
 }

 TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
   FrameCombiner combiner(false);
   for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
     // kMaxConcurrentChannels is 8.
     for (const int number_of_channels : {1, 2, 4, kMaxConcurrentChannels}) {
       SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1));

       AudioFrame audio_frame_for_mixing;

       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);
       EXPECT_THAT(audio_frame_for_mixing.packet_infos_,
                   ElementsAreArray(frame1.packet_infos_));
     }
   }
 }

 // Send a sine wave through the FrameCombiner, and check that the
 // difference between input and output varies smoothly. Also check
 // that it is inside reasonable bounds. This is to catch issues like
 // chromium:695993 and chromium:816875.
 TEST(FrameCombiner, GainCurveIsSmoothForAlternatingNumberOfStreams) {
   // Rates are divisible by 2000 when limiter is active.
   std::vector<FrameCombinerConfig> configs = {
       {false, 30100, 2, 50.f},  {false, 16500, 1, 3200.f},
       {true, 8000, 1, 3200.f},  {true, 16000, 1, 50.f},
       {true, 18000, 8, 3200.f}, {true, 10000, 2, 50.f},
   };

   for (const auto& config : configs) {
     SCOPED_TRACE(ProduceDebugText(config));

     FrameCombiner combiner(config.use_limiter);

     constexpr int16_t wave_amplitude = 30000;
     SineWaveGenerator wave_generator(config.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(config.sample_rate_hz, config.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_ * config.number_of_channels;

       // Ensures limiter is on if 'use_limiter'.
       constexpr size_t number_of_streams = 2;
       AudioFrame audio_frame_for_mixing;
       combiner.Combine(frames_to_combine, config.number_of_channels,
                        config.sample_rate_hz, 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));
     }

     // Check that the gain doesn't vary too much.
     EXPECT_LT(cumulative_change, 10);

     // Check that the latest gain is within reasonable bounds. It
     // should be slightly less that 1.
     EXPECT_LT(0.9f, change_calculator.LatestGain());
     EXPECT_LT(change_calculator.LatestGain(), 1.01f);
   }
 }
 }  // namespace webrtc
	/*
	* 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 "modules/audio_mixer/frame_combiner.h"

	#include <cstdint>
	#include <initializer_list>
	#include <numeric>
	#include <string>
	#include <type_traits>
	#include <vector>

	#include "absl/types/optional.h"
	#include "api/array_view.h"
	#include "api/rtp_packet_info.h"
	#include "api/rtp_packet_infos.h"
	#include "api/units/timestamp.h"
	#include "audio/utility/audio_frame_operations.h"
	#include "modules/audio_mixer/gain_change_calculator.h"
	#include "modules/audio_mixer/sine_wave_generator.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/strings/string_builder.h"
	#include "test/gmock.h"
	#include "test/gtest.h"

	namespace webrtc {

	namespace {

	using ::testing::ElementsAreArray;
	using ::testing::IsEmpty;
	using ::testing::UnorderedElementsAreArray;

	struct FrameCombinerConfig {
	bool use_limiter;
	int sample_rate_hz;
	int number_of_channels;
	float wave_frequency;
	};

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

	std::string ProduceDebugText(const FrameCombinerConfig& config) {
	rtc::StringBuilder ss;
	ss << "Sample rate: " << config.sample_rate_hz << " ,";
	ss << "number of channels: " << config.number_of_channels << " ,";
	ss << "limiter active: " << (config.use_limiter ? "on" : "off") << " ,";
	ss << "wave frequency: " << config.wave_frequency << " ,";
	return ss.Release();
	}

	AudioFrame frame1;
	AudioFrame frame2;

	void SetUpFrames(int sample_rate_hz, int number_of_channels) {
	RtpPacketInfo packet_info1(/ssrc=/1001, /csrcs=/{},
	/rtp_timestamp=/1000,
	/receive_time=/Timestamp::Millis(1));
	RtpPacketInfo packet_info2(/ssrc=/4004, /csrcs=/{},
	/rtp_timestamp=/1234,
	/receive_time=/Timestamp::Millis(2));
	RtpPacketInfo packet_info3(/ssrc=/7007, /csrcs=/{},
	/rtp_timestamp=/1333,
	/receive_time=/Timestamp::Millis(2));

	frame1.packet_infos_ = RtpPacketInfos({packet_info1});
	frame2.packet_infos_ = RtpPacketInfos({packet_info2, packet_info3});

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

	// The limiter requires sample rate divisible by 2000.
	TEST(FrameCombiner, BasicApiCallsLimiter) {
	FrameCombiner combiner(true);
	for (const int rate : {8000, 18000, 34000, 48000}) {
	for (const int number_of_channels : {1, 2, 4, 8}) {
	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);
	AudioFrame audio_frame_for_mixing;
	combiner.Combine(frames_to_combine, number_of_channels, rate,
	frames_to_combine.size(), &audio_frame_for_mixing);
	}
	}
	}
	}

	// The RtpPacketInfos field of the mixed packet should contain the union of the
	// RtpPacketInfos from the frames that were actually mixed.
	TEST(FrameCombiner, ContainsAllRtpPacketInfos) {
	static constexpr int kSampleRateHz = 48000;
	static constexpr int kNumChannels = 1;
	FrameCombiner combiner(true);
	const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
	SetUpFrames(kSampleRateHz, kNumChannels);

	for (const int number_of_frames : {0, 1, 2}) {
	SCOPED_TRACE(
	ProduceDebugText(kSampleRateHz, kNumChannels, number_of_frames));
	const std::vector<AudioFrame*> frames_to_combine(
	all_frames.begin(), all_frames.begin() + number_of_frames);

	std::vector<RtpPacketInfo> packet_infos;
	for (const auto& frame : frames_to_combine) {
	packet_infos.insert(packet_infos.end(), frame->packet_infos_.begin(),
	frame->packet_infos_.end());
	}

	AudioFrame audio_frame_for_mixing;
	combiner.Combine(frames_to_combine, kNumChannels, kSampleRateHz,
	frames_to_combine.size(), &audio_frame_for_mixing);
	EXPECT_THAT(audio_frame_for_mixing.packet_infos_,
	UnorderedElementsAreArray(packet_infos));
	}
	}

	#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
	// There are CHECKs in place to check for invalid parameters.
	TEST(FrameCombinerDeathTest, BuildCrashesWithManyChannels) {
	FrameCombiner combiner(true);
	for (const int rate : {8000, 18000, 34000, 48000}) {
	for (const int number_of_channels : {10, 20, 21}) {
	if (static_cast<size_t>(rate / 100 * number_of_channels) >
	AudioFrame::kMaxDataSizeSamples) {
	continue;
	}
	const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
	// With an unsupported channel count, this will crash in
	// `AudioFrame::UpdateFrame`.
	EXPECT_DEATH(SetUpFrames(rate, number_of_channels), "");

	const int number_of_frames = 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);
	AudioFrame audio_frame_for_mixing;
	EXPECT_DEATH(
	combiner.Combine(frames_to_combine, number_of_channels, rate,
	frames_to_combine.size(), &audio_frame_for_mixing),
	"");
	}
	}
	}
	#endif // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)

	TEST(FrameCombinerDeathTest, DebugBuildCrashesWithHighRate) {
	FrameCombiner combiner(true);
	for (const int rate : {50000, 96000, 128000, 196000}) {
	for (const int number_of_channels : {1, 2, 3}) {
	if (static_cast<size_t>(rate / 100 * number_of_channels) >
	AudioFrame::kMaxDataSizeSamples) {
	continue;
	}
	const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
	SetUpFrames(rate, number_of_channels);

	const int number_of_frames = 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);
	AudioFrame audio_frame_for_mixing;
	#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
	EXPECT_DEATH(
	combiner.Combine(frames_to_combine, number_of_channels, rate,
	frames_to_combine.size(), &audio_frame_for_mixing),
	"")
	<< "number_of_channels=" << number_of_channels << ", rate=" << rate
	<< ", frames to combine=" << frames_to_combine.size();
	#endif
	}
	}
	}

	// With no limiter, the rate has to be divisible by 100 since we use
	// 10 ms frames.
	TEST(FrameCombiner, BasicApiCallsNoLimiter) {
	FrameCombiner combiner(false);
	for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
	for (const int number_of_channels : {1, 2, 4, 8}) {
	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);
	AudioFrame audio_frame_for_mixing;
	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));

	AudioFrame audio_frame_for_mixing;

	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);
	EXPECT_THAT(audio_frame_for_mixing.packet_infos_, IsEmpty());
	}
	}
	}

	TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
	FrameCombiner combiner(false);
	for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
	// kMaxConcurrentChannels is 8.
	for (const int number_of_channels : {1, 2, 4, kMaxConcurrentChannels}) {
	SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1));

	AudioFrame audio_frame_for_mixing;

	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);
	EXPECT_THAT(audio_frame_for_mixing.packet_infos_,
	ElementsAreArray(frame1.packet_infos_));
	}
	}
	}

	// Send a sine wave through the FrameCombiner, and check that the
	// difference between input and output varies smoothly. Also check
	// that it is inside reasonable bounds. This is to catch issues like
	// chromium:695993 and chromium:816875.
	TEST(FrameCombiner, GainCurveIsSmoothForAlternatingNumberOfStreams) {
	// Rates are divisible by 2000 when limiter is active.
	std::vector<FrameCombinerConfig> configs = {
	{false, 30100, 2, 50.f}, {false, 16500, 1, 3200.f},
	{true, 8000, 1, 3200.f}, {true, 16000, 1, 50.f},
	{true, 18000, 8, 3200.f}, {true, 10000, 2, 50.f},
	};

	for (const auto& config : configs) {
	SCOPED_TRACE(ProduceDebugText(config));

	FrameCombiner combiner(config.use_limiter);

	constexpr int16_t wave_amplitude = 30000;
	SineWaveGenerator wave_generator(config.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(config.sample_rate_hz, config.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_ * config.number_of_channels;

	// Ensures limiter is on if 'use_limiter'.
	constexpr size_t number_of_streams = 2;
	AudioFrame audio_frame_for_mixing;
	combiner.Combine(frames_to_combine, config.number_of_channels,
	config.sample_rate_hz, 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));
	}

	// Check that the gain doesn't vary too much.
	EXPECT_LT(cumulative_change, 10);

	// Check that the latest gain is within reasonable bounds. It
	// should be slightly less that 1.
	EXPECT_LT(0.9f, change_calculator.LatestGain());
	EXPECT_LT(change_calculator.LatestGain(), 1.01f);
	}
	}
	} // namespace webrtc