Adding debug dump tests.

webrtc/modules/audio_processing/test/debug_dump_test.cc

Index: webrtc/modules/audio_processing/test/debug_dump_test.cc
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.cc b/webrtc/modules/audio_processing/test/debug_dump_test.cc
new file mode 100644
index 0000000000000000000000000000000000000000..710320170b8cd27859933b666f0d997770c3d709
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -0,0 +1,655 @@
+/*
+ *  Copyright (c) 2015 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_processing/test/debug_dump_test.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
+#include "webrtc/modules/audio_processing/test/test_utils.h"
+#include "webrtc/test/testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {

[Andrew MacDonald, 2015/10/20 01:22:08]

Blank space after this line.

[minyue-webrtc, 2015/10/23 08:44:45]

Done.

+  const std::string input_file_name =

[Andrew MacDonald, 2015/10/20 01:22:08]

No indent. Use git cl format.

These strings are non-POD, so shouldn't be statics. Use class members instead.

[minyue-webrtc, 2015/10/23 08:44:45]

Done.

+      test::ResourcePath("near32_stereo", "pcm");
+  const int kInputFileRateHz = 32000;
+  const size_t kInputFileChannels = 2;
+  const std::string reverse_file_name =
+      test::ResourcePath("far32_stereo", "pcm");
+  const int kReverseFileRateHz = 32000;
+  const size_t kReverseFileChannels = 2;
+}

[Andrew MacDonald, 2015/10/20 01:22:08]

Blank space before this line.

}  // namespace

[minyue-webrtc, 2015/10/23 08:44:45]

Done.

+
+DebugDumpGenerator::DebugDumpGenerator(std::string input_file_name,
+                                       int input_file_rate_hz,
+                                       size_t input_channels,
+                                       std::string reverse_file_name,
+                                       int reverse_file_rate_hz,
+                                       size_t reverse_channels,
+                                       const Config& config,
+                                       std::string dump_file_name)
+    : input_rate_hz_(input_file_rate_hz),
+      input_mono_(false),
+      reverse_rate_hz_(reverse_file_rate_hz),
+      reverse_mono_(false),
+      output_rate_hz_(input_file_rate_hz),
+      output_channels_(input_channels),
+      input_audio_(new ResampleInputAudioFile(input_file_name,
+                                              input_file_rate_hz,
+                                              input_rate_hz_)),
+      input_channels_(input_channels),
+      reverse_audio_(new ResampleInputAudioFile(reverse_file_name,
+                                                reverse_file_rate_hz,
+                                                reverse_rate_hz_)),
+      reverse_channels_(reverse_channels),
+      // Buffers will be created upon usage.

[Andrew MacDonald, 2015/10/20 01:22:08]

You could trigger the same InitializeFormat or whatever method I suggest below
from the constructor.

[minyue-webrtc, 2015/10/23 08:44:45]

Acknowledged.

+      input_(nullptr),
+      reverse_(nullptr),
+      output_(nullptr),
+      apm_(AudioProcessing::Create(config)),
+      dump_file_name_(dump_file_name) {
+}
+
+void DebugDumpGenerator::SetInputRate(int rate_hz) {
+  RTC_DCHECK(input_audio_.get());

[Andrew MacDonald, 2015/10/20 01:22:08]

This is created in the constructor, so no need for this.

[minyue-webrtc, 2015/10/23 08:44:45]

Acknowledged.

+  input_rate_hz_ = rate_hz;
+  input_audio_->set_output_rate_hz(input_rate_hz_);
+}
+
+void DebugDumpGenerator::ForceInputMono(bool mono) {
+  input_mono_ = mono;
+  if (input_mono_) {
+    // Output channels is set since it should be no bigger than input channels.
+    output_channels_ = 1;

[Andrew MacDonald, 2015/10/20 01:22:08]

What if ForceInputMono(false) is called?

[minyue-webrtc, 2015/10/23 08:44:45]

yes agreed. I think it is better not to change the output channels silently like
this. Caller must make sure that output channel number is smaller than or equal
to input.

+  }
+}
+
+void DebugDumpGenerator::SetReverseRate(int rate_hz) {
+  RTC_DCHECK(reverse_audio_.get());

[Andrew MacDonald, 2015/10/20 01:22:08]

Not needed.

[minyue-webrtc, 2015/10/23 08:44:45]

Acknowledged.

+  reverse_rate_hz_ = rate_hz;
+  reverse_audio_->set_output_rate_hz(reverse_rate_hz_);
+}
+
+void DebugDumpGenerator::ForceReverseMono(bool mono) {
+  reverse_mono_ = mono;
+}
+
+void DebugDumpGenerator::StartRecording() {
+  RTC_DCHECK(apm_.get());

[Andrew MacDonald, 2015/10/20 01:22:08]

Not needed.

[minyue-webrtc, 2015/10/23 08:44:45]

Acknowledged.

+  apm_->StartDebugRecording(dump_file_name_.c_str());
+}
+
+void DebugDumpGenerator::Process(size_t num_blocks) {
+  RTC_DCHECK(apm_.get());

[Andrew MacDonald, 2015/10/20 01:22:08]

Not needed.

[minyue-webrtc, 2015/10/23 08:44:45]

Acknowledged.

+
+  const size_t apm_input_channels = input_mono_ ? 1 : input_channels_;
+  const size_t apm_rev_channels = reverse_mono_ ? 1 : reverse_channels_;
+  const size_t apm_input_frames = rtc::CheckedDivExact(
+      AudioProcessing::kChunkSizeMs * input_rate_hz_, 1000);
+  const size_t apm_rev_frames = rtc::CheckedDivExact(
+      AudioProcessing::kChunkSizeMs * reverse_rate_hz_, 1000);
+
+  // The following enlarges buffers when necessary.
+  if (!input_.get() || input_->num_frames() < apm_input_frames ||

[Andrew MacDonald, 2015/10/20 01:22:08]

Since this is a test, we don't care about performance. Perhaps have a
InitializeFormat method instead of the setters and recreate the buffers there?

[minyue-webrtc, 2015/10/23 08:44:45]

Done.

+      input_->num_channels() < static_cast<int>(apm_input_channels)) {
+    input_.reset(
+        new ChannelBuffer<float>(apm_input_frames, apm_input_channels));
+  }
+
+  if (!reverse_.get() || reverse_->num_frames() < apm_rev_frames ||
+      reverse_->num_channels() < static_cast<int>(apm_rev_channels)) {
+    reverse_.reset(
+        new ChannelBuffer<float>(apm_rev_frames, apm_rev_channels));
+  }
+
+  // The following effectively calculates
+  // ceil(apm_input_frames * output_rate_hz_ / input_rate_hz_).
+  const size_t apm_output_frames = (apm_input_frames * output_rate_hz_ +
+      input_rate_hz_ - 1) / input_rate_hz_;
+
+  if (!output_.get() || output_->num_frames() < apm_output_frames ||
+      output_->num_channels() < static_cast<int>(output_channels_)) {
+    output_.reset(
+        new ChannelBuffer<float>(apm_output_frames, output_channels_));
+  }
+
+  for (size_t i = 0; i < num_blocks; ++i) {
+    ReadAndDeinterleave(reverse_audio_.get(), reverse_channels_, apm_rev_frames,
+                        reverse_mono_, reverse_->channels());
+    ReadAndDeinterleave(input_audio_.get(), input_channels_, apm_input_frames,
+                        input_mono_, input_->channels());
+
+    // Set a varying stream delay.
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 apm_->set_stream_delay_ms(100 + i % 10));
+
+    // A key press event is added every 10th block.
+    apm_->set_stream_key_pressed(i % 10 == 9);
+
+    RTC_CHECK_EQ(AudioProcessing::kNoError,
+                 apm_->ProcessStream(input_->channels(),
+                                     apm_input_frames,
+                                     input_rate_hz_,
+                                     LayoutFromChannels(apm_input_channels),
+                                     output_rate_hz_,
+                                     LayoutFromChannels(output_channels_),
+                                     output_->channels()));
+    RTC_CHECK_EQ(
+        AudioProcessing::kNoError,
+        apm_->AnalyzeReverseStream(reverse_->channels(),
+                                   apm_rev_frames,
+                                   reverse_rate_hz_,
+                                   LayoutFromChannels(apm_rev_channels)));
+  }
+}
+
+void DebugDumpGenerator::StopRecording() {
+  apm_->StopDebugRecording();
+}
+
+void DebugDumpGenerator::ReadAndDeinterleave(
+    ResampleInputAudioFile* audio, size_t channels,
+    size_t frames_per_channel, bool force_mono, float* const* buffer) {
+  // Make sure the buffer for reading the file is large enough.
+  if (channels * frames_per_channel > signal_.size()) {
+    signal_.resize(frames_per_channel * channels);
+  }
+
+  audio->Read(frames_per_channel * channels, &signal_[0]);
+
+  const size_t out_channels = force_mono ? 1 : channels;
+  for (size_t channel = 0; channel < out_channels; ++channel) {
+    for (size_t i = 0; i < frames_per_channel; ++i) {
+      buffer[channel][i] = S16ToFloat(signal_[i * channels + channel]);
+    }
+  }
+}
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP

[Andrew MacDonald, 2015/10/20 01:22:08]

This file should only be built when this is defined, right? You shouldn't need
it.

[minyue-webrtc, 2015/10/23 08:44:45]

Ironically, the DebugDumpGenerator can still work, since APM has
WEBRTC_AUDIOPROC_DEBUG_DUMP inside. Then no dump can be generated and
DebugDumpTest won't work.

Therefore I placed the WEBRTC_AUDIOPROC_DEBUG_DUMP around the implementation of
DebugDumpTest.

[Andrew MacDonald, 2015/10/24 00:42:05]

But this file is only compiled when enable_protobuf==1. You _know_ this is going
to be defined.

[minyue-webrtc, 2015/10/26 12:40:31]

Now removed

+
+DebugDumpTest::DebugDumpTest()
+    : input_rate_hz_(-1),
+      input_channels_(0),
+      output_rate_hz_(-1),
+      output_channels_(0),
+      reverse_rate_hz_(-1),
+      reverse_channels_(0),
+      // Buffers will be created upon usage.
+      input_(nullptr),
+      reverse_(nullptr),
+      output_(nullptr),
+      // APM will be created upon usage.
+      apm_(nullptr) {
+}
+
+void DebugDumpTest::VerifyDebugDump(const std::string in_filename) {

[Andrew MacDonald, 2015/10/20 01:22:08]

const std::string&

[minyue-webrtc, 2015/10/23 08:44:45]

Done.

+  FILE* in_file = fopen(in_filename.c_str(), "rb");
+  ASSERT_TRUE(in_file != NULL);

[Andrew MacDonald, 2015/10/20 01:22:08]

nullptr

[minyue-webrtc, 2015/10/23 08:44:45]

maybe better to remove !=NULL at all

+  audioproc::Event event_msg;
+
+  while (ReadMessageFromFile(in_file, &event_msg)) {
+    switch (event_msg.type()) {
+      case audioproc::Event::INIT:
+        OnInitEvent(event_msg.init());
+        break;
+      case audioproc::Event::STREAM:
+        OnStreamEvent(event_msg.stream());
+        break;
+      case audioproc::Event::REVERSE_STREAM:
+        OnReverseStreamEvent(event_msg.reverse_stream());
+        break;
+      case audioproc::Event::CONFIG:
+        OnConfigEvent(event_msg.config());
+        break;
+      case audioproc::Event::UNKNOWN_EVENT:
+        // We do not expect receive UNKNOWN event currently.
+        ASSERT_TRUE(false);
+    }
+  }
+  fclose(in_file);
+}
+
+// OnInitEvent reset the input/output/reserve channel format.
+void DebugDumpTest::OnInitEvent(const audioproc::Init& msg) {
+  input_rate_hz_ = msg.sample_rate();
+
+  ASSERT_TRUE(msg.has_num_input_channels());
+  input_channels_ = msg.num_input_channels();
+
+  ASSERT_TRUE(msg.has_output_sample_rate());
+  output_rate_hz_ = msg.output_sample_rate();
+
+  ASSERT_TRUE(msg.has_num_output_channels());
+  output_channels_ = msg.num_output_channels();
+
+  ASSERT_TRUE(msg.has_reverse_sample_rate());
+  reverse_rate_hz_ = msg.reverse_sample_rate();
+
+  ASSERT_TRUE(msg.has_num_reverse_channels());
+  reverse_channels_ = msg.num_reverse_channels();
+}
+
+// OnStreamEvent replays an input signal and verifies the output.
+void DebugDumpTest::OnStreamEvent(const audioproc::Stream& msg) {
+  // APM should have been created.
+  ASSERT_TRUE(apm_.get());
+
+  EXPECT_NOERR(apm_->gain_control()->set_stream_analog_level(msg.level()));
+  EXPECT_NOERR(apm_->set_stream_delay_ms(msg.delay()));
+  apm_->echo_cancellation()->set_stream_drift_samples(msg.drift());
+  if (msg.has_keypress())
+    apm_->set_stream_key_pressed(msg.keypress());
+  else
+    apm_->set_stream_key_pressed(true);
+
+  ASSERT_EQ(static_cast<int>(input_channels_), msg.input_channel_size());
+
+  const size_t frames_per_input_channel =
+      rtc::CheckedDivExact(msg.input_channel(0).size(), sizeof(float));
+
+  // The following effectively calculates
+  // ceil(frames_per_input_channel * output_rate_hz_ / input_rate_hz_).
+  const size_t frames_per_output_channel = (frames_per_input_channel *
+      output_rate_hz_ + input_rate_hz_ - 1) / input_rate_hz_;
+
+  // Updates the buffers to make sure that the sizes are large enough.
+  if (!input_.get() || input_->num_frames() < frames_per_input_channel ||

[Andrew MacDonald, 2015/10/20 01:22:08]

An init event has to occur for these values to change, right? Just recreate the
ChannelBuffers immediately there?

[minyue-webrtc, 2015/10/23 08:44:45]

Done.

+      input_->num_channels() < static_cast<int>(input_channels_)) {
+    input_.reset(
+        new ChannelBuffer<float>(frames_per_input_channel, input_channels_));
+  }
+
+  if (!output_.get() || output_->num_frames() < frames_per_output_channel ||
+      output_->num_channels() < static_cast<int>(output_channels_)) {
+    output_.reset(
+        new ChannelBuffer<float>(frames_per_output_channel, output_channels_));
+  }
+
+  for (int i = 0; i < msg.input_channel_size(); ++i) {
+     memcpy(input_->channels()[i], msg.input_channel(i).data(),
+            msg.input_channel(i).size());
+  }
+  ASSERT_EQ(AudioProcessing::kNoError,
+            apm_->ProcessStream(input_->channels(),

[Andrew MacDonald, 2015/10/20 01:22:08]

Can you use the ProcessStream overload with StreamConfigs? This one is
deprecated.

[minyue-webrtc, 2015/10/23 08:44:45]

ok

+                                frames_per_input_channel,
+                                input_rate_hz_,
+                                LayoutFromChannels(input_channels_),
+                                output_rate_hz_,
+                                LayoutFromChannels(output_channels_),
+                                output_->channels()));
+
+  // Check that output of APM is bit exact identical to the output in the dump.

[Andrew MacDonald, 2015/10/20 01:22:08]

s/bit exact identical/bit-exact

[minyue-webrtc, 2015/10/23 08:44:45]

Done.

+  ASSERT_EQ(static_cast<int>(output_channels_), msg.output_channel_size());
+  ASSERT_EQ(msg.output_channel(0).size(),
+            frames_per_output_channel * sizeof(float));
+  for (int i = 0; i < msg.output_channel_size(); ++i) {
+    ASSERT_EQ(0, memcmp(output_->channels()[i], msg.output_channel(i).data(),
+                        msg.output_channel(i).size()));
+  }
+}
+
+void DebugDumpTest::OnReverseStreamEvent(const audioproc::ReverseStream& msg) {
+  // APM should have been created.
+  ASSERT_TRUE(apm_.get());
+
+  ASSERT_GT(msg.channel_size(), 0);
+  ASSERT_EQ(static_cast<int>(reverse_channels_), msg.channel_size());
+
+  const size_t frames_per_channel =
+      rtc::CheckedDivExact(msg.channel(0).size(), sizeof(float));
+  if (!reverse_.get() || reverse_->num_frames() < frames_per_channel ||

[Andrew MacDonald, 2015/10/20 01:22:08]

Same thing: recreate this directly upon init event?

[minyue-webrtc, 2015/10/23 08:44:45]

Done.

+      reverse_->num_channels() < static_cast<int>(reverse_channels_)) {
+    reverse_.reset(
+        new ChannelBuffer<float>(frames_per_channel, reverse_channels_));
+  }
+
+  for (int i = 0; i < msg.channel_size(); ++i) {
+     memcpy(reverse_->channels()[i], msg.channel(i).data(),
+            msg.channel(i).size());
+  }
+
+  ASSERT_EQ(AudioProcessing::kNoError,
+            apm_->AnalyzeReverseStream(

[Andrew MacDonald, 2015/10/20 01:22:08]

Can you use ProcessReverseStream instead? This is deprecated.

[minyue-webrtc, 2015/10/23 08:44:45]

ok

+                reverse_->channels(),
+                frames_per_channel,
+                reverse_rate_hz_,
+                LayoutFromChannels(reverse_channels_)));
+}
+
+void DebugDumpTest::OnConfigEvent(const audioproc::Config& msg) {
+  MaybeRecreateApm(msg);
+  ConfigurateApm(msg);
+}
+
+void DebugDumpTest::MaybeRecreateApm(const audioproc::Config& msg) {
+  if (apm_.get()) {
+    // We only create APM once, since changes on these fields should not
+    // happen in current implementation.
+    return;
+  }
+
+  Config config;
+  ASSERT_TRUE(msg.has_aec_delay_agnostic_enabled());
+  config.Set<DelayAgnostic>(
+      new DelayAgnostic(msg.aec_delay_agnostic_enabled()));
+
+  ASSERT_TRUE(msg.has_noise_robust_agc_enabled());
+  config.Set<ExperimentalAgc>(
+       new ExperimentalAgc(msg.noise_robust_agc_enabled()));
+
+  ASSERT_TRUE(msg.has_transient_suppression_enabled());
+  config.Set<ExperimentalNs>(
+      new ExperimentalNs(msg.transient_suppression_enabled()));
+
+  ASSERT_TRUE(msg.has_aec_extended_filter_enabled());
+  config.Set<ExtendedFilter>(new ExtendedFilter(
+      msg.aec_extended_filter_enabled()));
+
+  apm_.reset(AudioProcessing::Create(config));
+}
+
+void DebugDumpTest::ConfigurateApm(const audioproc::Config& msg) {

[Andrew MacDonald, 2015/10/20 01:22:08]

ConfigureApm

[minyue-webrtc, 2015/10/23 08:44:45]

Done.

+  // AEC configs.
+  ASSERT_TRUE(msg.has_aec_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_cancellation()->Enable(msg.aec_enabled()));
+
+  ASSERT_TRUE(msg.has_aec_drift_compensation_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_cancellation()->enable_drift_compensation(
+                msg.aec_drift_compensation_enabled()));
+
+  ASSERT_TRUE(msg.has_aec_suppression_level());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_cancellation()->set_suppression_level(
+                static_cast<webrtc::EchoCancellation::SuppressionLevel>(
+                    msg.aec_suppression_level())));
+
+  // AECM configs.
+  ASSERT_TRUE(msg.has_aecm_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_control_mobile()->Enable(msg.aecm_enabled()));
+
+  ASSERT_TRUE(msg.has_aecm_comfort_noise_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_control_mobile()->enable_comfort_noise(
+                msg.aecm_comfort_noise_enabled()));
+
+  ASSERT_TRUE(msg.has_aecm_routing_mode());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->echo_control_mobile()->set_routing_mode(
+                static_cast<webrtc::EchoControlMobile::RoutingMode>(
+                    msg.aecm_routing_mode())));
+
+  // AGC configs.
+  ASSERT_TRUE(msg.has_agc_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->gain_control()->Enable(msg.agc_enabled()));
+
+  ASSERT_TRUE(msg.has_agc_mode());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->gain_control()->set_mode(
+                static_cast<webrtc::GainControl::Mode>(msg.agc_mode())));
+
+  ASSERT_TRUE(msg.has_agc_limiter_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->gain_control()->enable_limiter(msg.agc_limiter_enabled()));
+
+  // HPF configs.
+  ASSERT_TRUE(msg.has_hpf_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->high_pass_filter()->Enable(msg.hpf_enabled()));
+
+  // NS configs.
+  ASSERT_TRUE(msg.has_ns_enabled());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->noise_suppression()->Enable(msg.ns_enabled()));
+
+  ASSERT_TRUE(msg.has_ns_level());
+  EXPECT_EQ(AudioProcessing::kNoError,
+            apm_->noise_suppression()->set_level(
+                static_cast<webrtc::NoiseSuppression::Level>(msg.ns_level())));
+}
+
+#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
+
+TEST_F(DebugDumpTest, SimpleCase) {
+  Config config;

[Andrew MacDonald, 2015/10/20 01:22:08]

It looks like you never do anything with config in every test. Is that right?
Just remove it in that case.

[minyue-webrtc, 2015/10/23 08:44:45]

There are tests with config modified. see Line 536 e,g.

[Andrew MacDonald, 2015/10/24 00:42:05]

Ah, OK.

+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");

[Andrew MacDonald, 2015/10/20 01:22:08]

You use the same name in every test. I'd make dump_file_name_ a member of
DebugDumpTest, initialized in the constructor.

You can then also remove() the file in the destructor.

[minyue-webrtc, 2015/10/23 08:44:45]

I made dump_file_name_ a member in the generator class. That reads more nicely I
think

+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+TEST_F(DebugDumpTest, ChangeInputFormat) {
+  Config config;
+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetInputRate(48000);
+  generator.ForceInputMono(true);
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+TEST_F(DebugDumpTest, ChangeReverseFormat) {
+  Config config;
+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.SetReverseRate(48000);
+  generator.ForceReverseMono(true);
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+TEST_F(DebugDumpTest, ChangeOutputFormat) {
+  Config config;
+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.set_output_rate_hz(48000);
+  generator.set_output_channels(1);
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+TEST_F(DebugDumpTest, ToggleAec) {
+  Config config;
+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  generator.StartRecording();
+  generator.Process(100);
+
+  EchoCancellation* aec = generator.apm()->echo_cancellation();
+  EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(!aec->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+TEST_F(DebugDumpTest, ToggleDelayAgnosticAec) {
+  Config config;
+  config.Set<DelayAgnostic>(new DelayAgnostic(true));
+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  generator.StartRecording();
+  generator.Process(100);
+
+  EchoCancellation* aec = generator.apm()->echo_cancellation();
+  EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(!aec->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+TEST_F(DebugDumpTest, ToggleAecLevel) {
+  Config config;
+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  EchoCancellation* aec = generator.apm()->echo_cancellation();
+  EXPECT_EQ(AudioProcessing::kNoError, aec->Enable(true));
+  EXPECT_EQ(AudioProcessing::kNoError,
+            aec->set_suppression_level(EchoCancellation::kLowSuppression));
+  generator.StartRecording();
+  generator.Process(100);
+
+  EXPECT_EQ(AudioProcessing::kNoError,
+            aec->set_suppression_level(EchoCancellation::kHighSuppression));
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+TEST_F(DebugDumpTest, ToggleAgc) {
+  Config config;
+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  generator.StartRecording();
+  generator.Process(100);
+
+  GainControl* agc = generator.apm()->gain_control();
+  EXPECT_EQ(AudioProcessing::kNoError, agc->Enable(!agc->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+TEST_F(DebugDumpTest, ToggleNs) {
+  Config config;
+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  generator.StartRecording();
+  generator.Process(100);
+
+  NoiseSuppression* ns = generator.apm()->noise_suppression();
+  EXPECT_EQ(AudioProcessing::kNoError, ns->Enable(!ns->is_enabled()));
+
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+TEST_F(DebugDumpTest, TransientSuppressionOn) {
+  Config config;
+  config.Set<ExperimentalNs>(new ExperimentalNs(true));
+  const std::string dump_file_name =
+      test::TempFilename(test::OutputPath(), "debug_aec");
+  DebugDumpGenerator generator(input_file_name,
+                               kInputFileRateHz,
+                               kInputFileChannels,
+                               reverse_file_name,
+                               kReverseFileRateHz,
+                               kReverseFileChannels,
+                               config,
+                               dump_file_name);
+  generator.StartRecording();
+  generator.Process(100);
+  generator.StopRecording();
+  VerifyDebugDump(dump_file_name);
+  remove(dump_file_name.c_str());
+}
+
+}  // namespace test
+}  // namespace webrtc