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..0234df5cc45cb39a3d37f1f48f62411239d28108 100644
--- a/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
+++ b/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
@@ -16,6 +16,7 @@
 
 #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"
@@ -78,7 +79,8 @@ 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();
   }
 
@@ -456,9 +458,10 @@ 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 +474,46 @@ 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}) {
+        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