Optionally disable APM limiter in AudioMixer.

webrtc/modules/audio_mixer/frame_combiner_unittest.cc

Index: webrtc/modules/audio_mixer/frame_combiner_unittest.cc
diff --git a/webrtc/modules/audio_mixer/frame_combiner_unittest.cc b/webrtc/modules/audio_mixer/frame_combiner_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..13c66012f99f27c4fd30d820e19c0166fcf3d1f6
--- /dev/null
+++ b/webrtc/modules/audio_mixer/frame_combiner_unittest.cc
@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2017 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_mixer/frame_combiner.h"
+
+#include <numeric>
+#include <sstream>
+#include <string>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+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 frame1;
+AudioFrame frame2;
+AudioFrame audio_frame_for_mixing;
+
+void SetUpFrames(int sample_rate_hz, int number_of_channels) {
+  for (auto* frame : {&frame1, &frame2}) {
+    frame->UpdateFrame(-1, 0, nullptr,
+                       rtc::CheckedDivExact(sample_rate_hz, 100),
+                       sample_rate_hz, AudioFrame::kNormalSpeech,
+                       AudioFrame::kVadActive, number_of_channels);
+  }
+}
+}  // namespace
+
+TEST(FrameCombiner, BasicApiCallsLimiter) {
+  FrameCombiner combiner(true);
+  for (const int rate : {8000, 16000, 32000, 48000}) {
+    for (const int number_of_channels : {1, 2}) {
+      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+      SetUpFrames(rate, number_of_channels);
+
+      for (const int number_of_frames : {0, 1, 2}) {
+        SCOPED_TRACE(
+            ProduceDebugText(rate, number_of_channels, number_of_frames));
+        const std::vector<AudioFrame*> frames_to_combine(
+            all_frames.begin(), all_frames.begin() + number_of_frames);
+        combiner.Combine(frames_to_combine, number_of_channels, rate,
+                         &audio_frame_for_mixing);
+      }
+    }
+  }
+}
+
+// No APM limiter means no AudioProcessing::NativeRate restriction
+// on rate. The rate has to be divisible by 100 since we use
+// 10 ms frames, though.
+TEST(FrameCombiner, BasicApiCallsNoLimiter) {
+  FrameCombiner combiner(false);
+  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
+    for (const int number_of_channels : {1, 2}) {
+      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+      SetUpFrames(rate, number_of_channels);
+
+      for (const int number_of_frames : {0, 1, 2}) {
+        SCOPED_TRACE(
+            ProduceDebugText(rate, number_of_channels, number_of_frames));
+        const std::vector<AudioFrame*> frames_to_combine(
+            all_frames.begin(), all_frames.begin() + number_of_frames);
+        combiner.Combine(frames_to_combine, number_of_channels, rate,
+                         &audio_frame_for_mixing);
+      }
+    }
+  }
+}
+
+TEST(FrameCombiner, CombiningZeroFramesShouldProduceSilence) {
+  FrameCombiner combiner(false);
+  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
+    for (const int number_of_channels : {1, 2}) {
+      SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 0));
+
+      const std::vector<AudioFrame*> frames_to_combine;
+      combiner.Combine(frames_to_combine, number_of_channels, rate,
+                       &audio_frame_for_mixing);
+
+      const std::vector<int16_t> mixed_data(
+          audio_frame_for_mixing.data_,
+          audio_frame_for_mixing.data_ + number_of_channels * rate / 100);
+
+      const std::vector<int16_t> expected(number_of_channels * rate / 100, 0);
+      EXPECT_EQ(mixed_data, expected);
+    }
+  }
+}
+
+TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
+  FrameCombiner combiner(false);
+  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
+    for (const int number_of_channels : {1, 2}) {
+      SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1));
+
+      SetUpFrames(rate, number_of_channels);
+      std::iota(frame1.data_, frame1.data_ + number_of_channels * rate / 100,
+                0);
+      const std::vector<AudioFrame*> frames_to_combine = {&frame1};
+      combiner.Combine(frames_to_combine, number_of_channels, rate,
+                       &audio_frame_for_mixing);
+
+      const std::vector<int16_t> mixed_data(
+          audio_frame_for_mixing.data_,
+          audio_frame_for_mixing.data_ + number_of_channels * rate / 100);
+
+      std::vector<int16_t> expected(number_of_channels * rate / 100);
+      std::iota(expected.begin(), expected.end(), 0);
+      EXPECT_EQ(mixed_data, expected);
+    }
+  }
+}
+
+}  // namespace webrtc