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 <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 42 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) {}
@@ skipping 357 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}) {
+        const auto mixer =

[hlundin-webrtc, 2017/02/17 11:04:12]

Please, use a SCOPED_TRACE to annotate which lap in the for loop is being
tested. This helps provide context in the error printout when something fails.

See e.g., webrtc/modules/audio_processing/aec3/block_framer_unittest.cc for
examples of how to use SCOPED_TRACE.

[aleloi, 2017/02/20 10:41:33]

Done.

+            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