Optionally disable APM limiter in AudioMixer.

webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc

Index: webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
diff --git a/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc b/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
index e90dac1e70e9839f032da4db4f91b8eb0d002686..f651fe978315fa01dbcad9ab0f0dc873a3129c2d 100644
--- a/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
+++ b/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
@@ -12,10 +12,13 @@
 
 #include <limits>
 #include <memory>
+#include <sstream>
+#include <string>
 #include <utility>
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/base/bind.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/modules/audio_mixer/default_output_rate_calculator.h"
@@ -46,6 +49,16 @@ void ResetFrame(AudioFrame* frame) {
   frame->speech_type_ = AudioFrame::kNormalSpeech;
 }
 
+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();
+}
+
 AudioFrame frame_for_mixing;
 
 }  // namespace
@@ -78,7 +91,8 @@ class MockMixerAudioSource : public AudioMixer::Source {
                                         AudioFrame* audio_frame) {
     audio_frame->CopyFrom(fake_frame_);
     audio_frame->sample_rate_hz_ = sample_rate_hz;
-    audio_frame->samples_per_channel_ = sample_rate_hz / 100;
+    audio_frame->samples_per_channel_ =
+        rtc::CheckedDivExact(sample_rate_hz, 100);
     return fake_info();
   }
 
@@ -89,7 +103,7 @@ class MockMixerAudioSource : public AudioMixer::Source {
 class CustomRateCalculator : public OutputRateCalculator {
  public:
   explicit CustomRateCalculator(int rate) : rate_(rate) {}
-  int CalculateOutputRate(const std::vector<int>& preferred_rates) {
+  int CalculateOutputRate(const std::vector<int>& preferred_rates) override {
     return rate_;
   }
 
@@ -103,19 +117,19 @@ void MixAndCompare(
     const std::vector<AudioFrame>& frames,
     const std::vector<AudioMixer::Source::AudioFrameInfo>& frame_info,
     const std::vector<bool>& expected_status) {
-  int num_audio_sources = frames.size();
+  const size_t num_audio_sources = frames.size();
   RTC_DCHECK(frames.size() == frame_info.size());
   RTC_DCHECK(frame_info.size() == expected_status.size());
 
   const auto mixer = AudioMixerImpl::Create();
   std::vector<MockMixerAudioSource> participants(num_audio_sources);
 
-  for (int i = 0; i < num_audio_sources; i++) {
+  for (size_t i = 0; i < num_audio_sources; ++i) {
     participants[i].fake_frame()->CopyFrom(frames[i]);
     participants[i].set_fake_info(frame_info[i]);
   }
 
-  for (int i = 0; i < num_audio_sources; i++) {
+  for (size_t i = 0; i < num_audio_sources; ++i) {
     EXPECT_TRUE(mixer->AddSource(&participants[i]));
     EXPECT_CALL(participants[i], GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
         .Times(Exactly(1));
@@ -123,7 +137,7 @@ void MixAndCompare(
 
   mixer->Mix(1, &frame_for_mixing);
 
-  for (int i = 0; i < num_audio_sources; i++) {
+  for (size_t i = 0; i < num_audio_sources; ++i) {
     EXPECT_EQ(expected_status[i],
               mixer->GetAudioSourceMixabilityStatusForTest(&participants[i]))
         << "Mixed status of AudioSource #" << i << " wrong.";
@@ -191,11 +205,11 @@ TEST(AudioMixer, FrameNotModifiedForSingleParticipant) {
   MockMixerAudioSource participant;
 
   ResetFrame(participant.fake_frame());
-  const int n_samples = participant.fake_frame()->samples_per_channel_;
+  const size_t n_samples = participant.fake_frame()->samples_per_channel_;
 
   // Modify the frame so that it's not zero.
-  for (int j = 0; j < n_samples; j++) {
-    participant.fake_frame()->data_[j] = j;
+  for (size_t j = 0; j < n_samples; ++j) {
+    participant.fake_frame()->data_[j] = static_cast<int16_t>(j);
   }
 
   EXPECT_TRUE(mixer->AddSource(&participant));
@@ -203,7 +217,7 @@ TEST(AudioMixer, FrameNotModifiedForSingleParticipant) {
 
   AudioFrame audio_frame;
   // Two mix iteration to compare after the ramp-up step.
-  for (int i = 0; i < 2; i++) {
+  for (int i = 0; i < 2; ++i) {
     mixer->Mix(1,  // number of channels
                &audio_frame);
   }
@@ -310,7 +324,7 @@ TEST(AudioMixer, RampedOutSourcesShouldNotBeMarkedMixed) {
   const auto mixer = AudioMixerImpl::Create();
   MockMixerAudioSource participants[kAudioSources];
 
-  for (int i = 0; i < kAudioSources; i++) {
+  for (int i = 0; i < kAudioSources; ++i) {
     ResetFrame(participants[i].fake_frame());
     // Set the participant audio energy to increase with the index
     // |i|.
@@ -318,7 +332,7 @@ TEST(AudioMixer, RampedOutSourcesShouldNotBeMarkedMixed) {
   }
 
   // Add all participants but the loudest for mixing.
-  for (int i = 0; i < kAudioSources - 1; i++) {
+  for (int i = 0; i < kAudioSources - 1; ++i) {
     EXPECT_TRUE(mixer->AddSource(&participants[i]));
     EXPECT_CALL(participants[i], GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
         .Times(Exactly(1));
@@ -328,14 +342,14 @@ TEST(AudioMixer, RampedOutSourcesShouldNotBeMarkedMixed) {
   mixer->Mix(1, &frame_for_mixing);
 
   // All participants but the loudest should have been mixed.
-  for (int i = 0; i < kAudioSources - 1; i++) {
+  for (int i = 0; i < kAudioSources - 1; ++i) {
     EXPECT_TRUE(mixer->GetAudioSourceMixabilityStatusForTest(&participants[i]))
         << "Mixed status of AudioSource #" << i << " wrong.";
   }
 
   // Add new participant with higher energy.
   EXPECT_TRUE(mixer->AddSource(&participants[kAudioSources - 1]));
-  for (int i = 0; i < kAudioSources; i++) {
+  for (int i = 0; i < kAudioSources; ++i) {
     EXPECT_CALL(participants[i], GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
         .Times(Exactly(1));
   }
@@ -347,7 +361,7 @@ TEST(AudioMixer, RampedOutSourcesShouldNotBeMarkedMixed) {
       << "Mixed status of AudioSource #0 wrong.";
 
   // The loudest participants should have been mixed.
-  for (int i = 1; i < kAudioSources; i++) {
+  for (int i = 1; i < kAudioSources; ++i) {
     EXPECT_EQ(true,
               mixer->GetAudioSourceMixabilityStatusForTest(&participants[i]))
         << "Mixed status of AudioSource #" << i << " wrong.";
@@ -456,9 +470,10 @@ TEST(AudioMixer, UnmutedShouldMixBeforeLoud) {
 
 TEST(AudioMixer, MixingRateShouldBeDecidedByRateCalculator) {
   constexpr int kOutputRate = 22000;
-  const auto mixer = AudioMixerImpl::CreateWithOutputRateCalculator(
+  const auto mixer = AudioMixerImpl::CreateWithOutputRateCalculatorAndLimiter(
       std::unique_ptr<OutputRateCalculator>(
-          new CustomRateCalculator(kOutputRate)));
+          new CustomRateCalculator(kOutputRate)),
+      true);
   MockMixerAudioSource audio_source;
   mixer->AddSource(&audio_source);
   ResetFrame(audio_source.fake_frame());
@@ -471,12 +486,48 @@ TEST(AudioMixer, MixingRateShouldBeDecidedByRateCalculator) {
 
 TEST(AudioMixer, ZeroSourceRateShouldBeDecidedByRateCalculator) {
   constexpr int kOutputRate = 8000;
-  const auto mixer = AudioMixerImpl::CreateWithOutputRateCalculator(
+  const auto mixer = AudioMixerImpl::CreateWithOutputRateCalculatorAndLimiter(
       std::unique_ptr<OutputRateCalculator>(
-          new CustomRateCalculator(kOutputRate)));
+          new CustomRateCalculator(kOutputRate)),
+      true);
 
   mixer->Mix(1, &frame_for_mixing);
 
   EXPECT_EQ(kOutputRate, frame_for_mixing.sample_rate_hz_);
 }
+
+TEST(AudioMixer, NoLimiterBasicApiCalls) {
+  const auto mixer = AudioMixerImpl::CreateWithOutputRateCalculatorAndLimiter(
+      std::unique_ptr<OutputRateCalculator>(new DefaultOutputRateCalculator()),
+      false);
+  mixer->Mix(1, &frame_for_mixing);
+}
+
+TEST(AudioMixer, AnyRateIsPossibleWithNoLimiter) {
+  // No APM limiter means no AudioProcessing::NativeRate restriction
+  // on mixing rate. The rate has to be divisible by 100 since we use
+  // 10 ms frames, though.
+  for (const auto rate : {8000, 20000, 24000, 32000, 44100}) {
+    for (const size_t number_of_channels : {1, 2}) {
+      for (const auto number_of_sources : {0, 1, 2, 3, 4}) {
+        SCOPED_TRACE(
+            ProduceDebugText(rate, number_of_sources, number_of_sources));
+        const auto mixer =
+            AudioMixerImpl::CreateWithOutputRateCalculatorAndLimiter(
+                std::unique_ptr<OutputRateCalculator>(
+                    new CustomRateCalculator(rate)),
+                false);
+
+        std::vector<MockMixerAudioSource> sources(number_of_sources);
+        for (auto& source : sources) {
+          mixer->AddSource(&source);
+        }
+
+        mixer->Mix(number_of_channels, &frame_for_mixing);
+        EXPECT_EQ(rate, frame_for_mixing.sample_rate_hz_);
+        EXPECT_EQ(number_of_channels, frame_for_mixing.num_channels_);
+      }
+    }
+  }
+}
 }  // namespace webrtc