Optionally disable APM limiter in AudioMixer.

webrtc/modules/audio_mixer/audio_mixer_impl_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 <string.h>
 
 #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"
 #include "webrtc/test/gmock.h"
 
 using testing::_;
 using testing::Exactly;
 using testing::Invoke;
 using testing::Return;
 
@@ skipping 10 matching lines @@
   frame->id_ = kId;
   frame->sample_rate_hz_ = kDefaultSampleRateHz;
   frame->num_channels_ = 1;
 
   // Frame duration 10ms.
   frame->samples_per_channel_ = kDefaultSampleRateHz / 100;
   frame->vad_activity_ = AudioFrame::kVadActive;
   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
 
 class MockMixerAudioSource : public AudioMixer::Source {
  public:
   MockMixerAudioSource()
       : fake_audio_frame_info_(AudioMixer::Source::AudioFrameInfo::kNormal) {
     ON_CALL(*this, GetAudioFrameWithInfo(_, _))
         .WillByDefault(
@@ skipping 12 matching lines @@
   AudioFrameInfo fake_info() { return fake_audio_frame_info_; }
   void set_fake_info(const AudioFrameInfo audio_frame_info) {
     fake_audio_frame_info_ = audio_frame_info;
   }
 
  private:
   AudioFrameInfo FakeAudioFrameWithInfo(int sample_rate_hz,
                                         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();
   }
 
   AudioFrame fake_frame_;
   AudioFrameInfo fake_audio_frame_info_;
 };
 
 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_;
   }
 
  private:
   const int rate_;
 };
 
 // Creates participants from |frames| and |frame_info| and adds them
 // to the mixer. Compares mixed status with |expected_status|
 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));
   }
 
   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.";
   }
 }
 
 void MixMonoAtGivenNativeRate(int native_sample_rate,
                               AudioFrame* mix_frame,
                               rtc::scoped_refptr<AudioMixer> mixer,
                               MockMixerAudioSource* audio_source) {
@@ skipping 47 matching lines @@
     }
   }
 }
 
 TEST(AudioMixer, FrameNotModifiedForSingleParticipant) {
   const auto mixer = AudioMixerImpl::Create();
 
   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));
   EXPECT_CALL(participant, GetAudioFrameWithInfo(_, _)).Times(Exactly(2));
 
   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);
   }
 
   EXPECT_EQ(
       0, memcmp(participant.fake_frame()->data_, audio_frame.data_, n_samples));
 }
 
 TEST(AudioMixer, SourceAtNativeRateShouldNeverResample) {
   const auto mixer = AudioMixerImpl::Create();
@@ skipping 86 matching lines @@
 
 // Maximal amount of participants are mixed one iteration, then
 // another participant with higher energy is added.
 TEST(AudioMixer, RampedOutSourcesShouldNotBeMarkedMixed) {
   constexpr int kAudioSources =
       AudioMixerImpl::kMaximumAmountOfMixedAudioSources + 1;
 
   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|.
     participants[i].fake_frame()->data_[0] = 100 * i;
   }
 
   // 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));
   }
 
   // First mixer iteration
   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));
   }
 
   mixer->Mix(1, &frame_for_mixing);
 
   // The most quiet participant should not have been mixed.
   EXPECT_FALSE(mixer->GetAudioSourceMixabilityStatusForTest(&participants[0]))
       << "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.";
   }
 }
 
 // This test checks that the initialization and participant addition
 // can be done on a different thread.
 TEST(AudioMixer, ConstructFromOtherThread) {
   std::unique_ptr<rtc::Thread> init_thread = rtc::Thread::Create();
@@ skipping 88 matching lines @@
   std::fill(frames[0].data_, frames[0].data_ + kDefaultSampleRateHz / 100,
             std::numeric_limits<int16_t>::max());
   std::vector<bool> expected_status(kAudioSources, true);
   expected_status[0] = false;
 
   MixAndCompare(frames, frame_info, expected_status);
 }
 
 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());
 
   EXPECT_CALL(audio_source, GetAudioFrameWithInfo(kOutputRate, _))
       .Times(Exactly(1));
 
   mixer->Mix(1, &frame_for_mixing);
 }
 
 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