1 files changed, 337 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_mixer/frame_combiner_unittest.cc b/third_party/libwebrtc/modules/audio_mixer/frame_combiner_unittest.cc
new file mode 100644
index 0000000000..6c64d0852a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_mixer/frame_combiner_unittest.cc
@@ -0,0 +1,337 @@
+/*
+ *  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 "modules/audio_mixer/frame_combiner.h"
+
+#include <cstdint>
+#include <initializer_list>
+#include <numeric>
+#include <string>
+#include <type_traits>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/rtp_packet_info.h"
+#include "api/rtp_packet_infos.h"
+#include "api/units/timestamp.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "modules/audio_mixer/gain_change_calculator.h"
+#include "modules/audio_mixer/sine_wave_generator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using ::testing::ElementsAreArray;
+using ::testing::IsEmpty;
+using ::testing::UnorderedElementsAreArray;
+
+struct FrameCombinerConfig {
+  bool use_limiter;
+  int sample_rate_hz;
+  int number_of_channels;
+  float wave_frequency;
+};
+
+std::string ProduceDebugText(int sample_rate_hz,
+                             int number_of_channels,
+                             int number_of_sources) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << sample_rate_hz << " ,";
+  ss << "number of channels: " << number_of_channels << " ,";
+  ss << "number of sources: " << number_of_sources;
+  return ss.Release();
+}
+
+std::string ProduceDebugText(const FrameCombinerConfig& config) {
+  rtc::StringBuilder ss;
+  ss << "Sample rate: " << config.sample_rate_hz << " ,";
+  ss << "number of channels: " << config.number_of_channels << " ,";
+  ss << "limiter active: " << (config.use_limiter ? "on" : "off") << " ,";
+  ss << "wave frequency: " << config.wave_frequency << " ,";
+  return ss.Release();
+}
+
+AudioFrame frame1;
+AudioFrame frame2;
+
+void SetUpFrames(int sample_rate_hz, int number_of_channels) {
+  RtpPacketInfo packet_info1(/*ssrc=*/1001, /*csrcs=*/{},
+                             /*rtp_timestamp=*/1000,
+                             /*receive_time=*/Timestamp::Millis(1));
+  RtpPacketInfo packet_info2(/*ssrc=*/4004, /*csrcs=*/{},
+                             /*rtp_timestamp=*/1234,
+                             /*receive_time=*/Timestamp::Millis(2));
+  RtpPacketInfo packet_info3(/*ssrc=*/7007, /*csrcs=*/{},
+                             /*rtp_timestamp=*/1333,
+                             /*receive_time=*/Timestamp::Millis(2));
+
+  frame1.packet_infos_ = RtpPacketInfos({packet_info1});
+  frame2.packet_infos_ = RtpPacketInfos({packet_info2, packet_info3});
+
+  for (auto* frame : {&frame1, &frame2}) {
+    frame->UpdateFrame(0, nullptr, rtc::CheckedDivExact(sample_rate_hz, 100),
+                       sample_rate_hz, AudioFrame::kNormalSpeech,
+                       AudioFrame::kVadActive, number_of_channels);
+  }
+}
+}  // namespace
+
+// The limiter requires sample rate divisible by 2000.
+TEST(FrameCombiner, BasicApiCallsLimiter) {
+  FrameCombiner combiner(true);
+  for (const int rate : {8000, 18000, 34000, 48000}) {
+    for (const int number_of_channels : {1, 2, 4, 8}) {
+      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);
+        AudioFrame audio_frame_for_mixing;
+        combiner.Combine(frames_to_combine, number_of_channels, rate,
+                         frames_to_combine.size(), &audio_frame_for_mixing);
+      }
+    }
+  }
+}
+
+// The RtpPacketInfos field of the mixed packet should contain the union of the
+// RtpPacketInfos from the frames that were actually mixed.
+TEST(FrameCombiner, ContainsAllRtpPacketInfos) {
+  static constexpr int kSampleRateHz = 48000;
+  static constexpr int kNumChannels = 1;
+  FrameCombiner combiner(true);
+  const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+  SetUpFrames(kSampleRateHz, kNumChannels);
+
+  for (const int number_of_frames : {0, 1, 2}) {
+    SCOPED_TRACE(
+        ProduceDebugText(kSampleRateHz, kNumChannels, number_of_frames));
+    const std::vector<AudioFrame*> frames_to_combine(
+        all_frames.begin(), all_frames.begin() + number_of_frames);
+
+    std::vector<RtpPacketInfo> packet_infos;
+    for (const auto& frame : frames_to_combine) {
+      packet_infos.insert(packet_infos.end(), frame->packet_infos_.begin(),
+                          frame->packet_infos_.end());
+    }
+
+    AudioFrame audio_frame_for_mixing;
+    combiner.Combine(frames_to_combine, kNumChannels, kSampleRateHz,
+                     frames_to_combine.size(), &audio_frame_for_mixing);
+    EXPECT_THAT(audio_frame_for_mixing.packet_infos_,
+                UnorderedElementsAreArray(packet_infos));
+  }
+}
+
+// There are DCHECKs in place to check for invalid parameters.
+TEST(FrameCombinerDeathTest, DebugBuildCrashesWithManyChannels) {
+  FrameCombiner combiner(true);
+  for (const int rate : {8000, 18000, 34000, 48000}) {
+    for (const int number_of_channels : {10, 20, 21}) {
+      if (static_cast<size_t>(rate / 100 * number_of_channels) >
+          AudioFrame::kMaxDataSizeSamples) {
+        continue;
+      }
+      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+      SetUpFrames(rate, number_of_channels);
+
+      const int number_of_frames = 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);
+      AudioFrame audio_frame_for_mixing;
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+      EXPECT_DEATH(
+          combiner.Combine(frames_to_combine, number_of_channels, rate,
+                           frames_to_combine.size(), &audio_frame_for_mixing),
+          "");
+#elif !RTC_DCHECK_IS_ON
+      combiner.Combine(frames_to_combine, number_of_channels, rate,
+                       frames_to_combine.size(), &audio_frame_for_mixing);
+#endif
+    }
+  }
+}
+
+TEST(FrameCombinerDeathTest, DebugBuildCrashesWithHighRate) {
+  FrameCombiner combiner(true);
+  for (const int rate : {50000, 96000, 128000, 196000}) {
+    for (const int number_of_channels : {1, 2, 3}) {
+      if (static_cast<size_t>(rate / 100 * number_of_channels) >
+          AudioFrame::kMaxDataSizeSamples) {
+        continue;
+      }
+      const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
+      SetUpFrames(rate, number_of_channels);
+
+      const int number_of_frames = 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);
+      AudioFrame audio_frame_for_mixing;
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+      EXPECT_DEATH(
+          combiner.Combine(frames_to_combine, number_of_channels, rate,
+                           frames_to_combine.size(), &audio_frame_for_mixing),
+          "");
+#elif !RTC_DCHECK_IS_ON
+      combiner.Combine(frames_to_combine, number_of_channels, rate,
+                       frames_to_combine.size(), &audio_frame_for_mixing);
+#endif
+    }
+  }
+}
+
+// With no limiter, the rate has to be divisible by 100 since we use
+// 10 ms frames.
+TEST(FrameCombiner, BasicApiCallsNoLimiter) {
+  FrameCombiner combiner(false);
+  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
+    for (const int number_of_channels : {1, 2, 4, 8}) {
+      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);
+        AudioFrame audio_frame_for_mixing;
+        combiner.Combine(frames_to_combine, number_of_channels, rate,
+                         frames_to_combine.size(), &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));
+
+      AudioFrame audio_frame_for_mixing;
+
+      const std::vector<AudioFrame*> frames_to_combine;
+      combiner.Combine(frames_to_combine, number_of_channels, rate,
+                       frames_to_combine.size(), &audio_frame_for_mixing);
+      const int16_t* audio_frame_for_mixing_data =
+          audio_frame_for_mixing.data();
+      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);
+      EXPECT_THAT(audio_frame_for_mixing.packet_infos_, IsEmpty());
+    }
+  }
+}
+
+TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
+  FrameCombiner combiner(false);
+  for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
+    for (const int number_of_channels : {1, 2, 4, 8, 10}) {
+      SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1));
+
+      AudioFrame audio_frame_for_mixing;
+
+      SetUpFrames(rate, number_of_channels);
+      int16_t* frame1_data = frame1.mutable_data();
+      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,
+                       frames_to_combine.size(), &audio_frame_for_mixing);
+
+      const int16_t* audio_frame_for_mixing_data =
+          audio_frame_for_mixing.data();
+      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);
+      EXPECT_THAT(audio_frame_for_mixing.packet_infos_,
+                  ElementsAreArray(frame1.packet_infos_));
+    }
+  }
+}
+
+// Send a sine wave through the FrameCombiner, and check that the
+// difference between input and output varies smoothly. Also check
+// that it is inside reasonable bounds. This is to catch issues like
+// chromium:695993 and chromium:816875.
+TEST(FrameCombiner, GainCurveIsSmoothForAlternatingNumberOfStreams) {
+  // Rates are divisible by 2000 when limiter is active.
+  std::vector<FrameCombinerConfig> configs = {
+      {false, 30100, 2, 50.f},  {false, 16500, 1, 3200.f},
+      {true, 8000, 1, 3200.f},  {true, 16000, 1, 50.f},
+      {true, 18000, 8, 3200.f}, {true, 10000, 2, 50.f},
+  };
+
+  for (const auto& config : configs) {
+    SCOPED_TRACE(ProduceDebugText(config));
+
+    FrameCombiner combiner(config.use_limiter);
+
+    constexpr int16_t wave_amplitude = 30000;
+    SineWaveGenerator wave_generator(config.wave_frequency, wave_amplitude);
+
+    GainChangeCalculator change_calculator;
+    float cumulative_change = 0.f;
+
+    constexpr size_t iterations = 100;
+
+    for (size_t i = 0; i < iterations; ++i) {
+      SetUpFrames(config.sample_rate_hz, config.number_of_channels);
+      wave_generator.GenerateNextFrame(&frame1);
+      AudioFrameOperations::Mute(&frame2);
+
+      std::vector<AudioFrame*> frames_to_combine = {&frame1};
+      if (i % 2 == 0) {
+        frames_to_combine.push_back(&frame2);
+      }
+      const size_t number_of_samples =
+          frame1.samples_per_channel_ * config.number_of_channels;
+
+      // Ensures limiter is on if 'use_limiter'.
+      constexpr size_t number_of_streams = 2;
+      AudioFrame audio_frame_for_mixing;
+      combiner.Combine(frames_to_combine, config.number_of_channels,
+                       config.sample_rate_hz, number_of_streams,
+                       &audio_frame_for_mixing);
+      cumulative_change += change_calculator.CalculateGainChange(
+          rtc::ArrayView<const int16_t>(frame1.data(), number_of_samples),
+          rtc::ArrayView<const int16_t>(audio_frame_for_mixing.data(),
+                                        number_of_samples));
+    }
+
+    // Check that the gain doesn't vary too much.
+    EXPECT_LT(cumulative_change, 10);
+
+    // Check that the latest gain is within reasonable bounds. It
+    // should be slightly less that 1.
+    EXPECT_LT(0.9f, change_calculator.LatestGain());
+    EXPECT_LT(change_calculator.LatestGain(), 1.01f);
+  }
+}
+}  // namespace webrtc