Optionally disable APM limiter in AudioMixer.

webrtc/modules/audio_mixer/frame_combiner_unittest.cc

+/*
+ *  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) {

[hlundin-webrtc, 2017/02/20 14:16:31]

I cannot see that the actual data_ in the AudioFrames is ever set (to zero or
anything else). Won't this mean that your tests are reading and acting upon
undefined data? You could have this method call AudioFrame::UpdateFrame to
resolve that.

[aleloi, 2017/02/20 15:21:37]

Good point! Done.

+  for (auto* frame : {&frame1, &frame2}) {
+    frame->sample_rate_hz_ = sample_rate_hz;
+    frame->num_channels_ = number_of_channels;
+    frame->samples_per_channel_ = rtc::CheckedDivExact(sample_rate_hz, 100);
+  }
+}
+}  // 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}) {
+        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}) {
+        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}) {
+      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}) {
+      ProduceDebugText(rate, number_of_channels, 1);
+
+      SetUpFrames(rate, number_of_channels);
+      std::iota(frame1.data_, frame1.data_ + number_of_channels * rate / 100,

[hlundin-webrtc, 2017/02/20 14:16:32]

Didn't know about std::iota. Nice.

+                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