Prevent Opus DTX from generating intermittent noise during silence

webrtc/voice_engine/test/auto_test/voe_output_test.cc

Index: webrtc/voice_engine/test/auto_test/voe_output_test.cc
diff --git a/webrtc/voice_engine/test/auto_test/voe_output_test.cc b/webrtc/voice_engine/test/auto_test/voe_output_test.cc
new file mode 100644
index 0000000000000000000000000000000000000000..521ed15c1036c070d0c69773dccaa570eaca7ef1
--- /dev/null
+++ b/webrtc/voice_engine/test/auto_test/voe_output_test.cc
@@ -0,0 +1,215 @@
+/*
+ *  Copyright (c) 2015 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 "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/system_wrappers/interface/sleep.h"
+#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/random.h"
+#include "webrtc/test/testsupport/fileutils.h"
+#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
+
+namespace {
+
+const char kIp[] = "127.0.0.1";
+const int kPort = 1234;
+const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
+
+}  // namespace
+
+namespace voetest {
+
+using webrtc::test::Random;
+using webrtc::test::VoiceChannelTransport;
+
+// This test allows a check on the output signal in an end-to-end call.
+class OutputTest {
+ public:
+  OutputTest();
+  ~OutputTest();
+
+  void Start();
+
+  void EnableOutputCheck();
+  void DisableOutputCheck();
+  void SetOutputBound(int16_t lower_bound, int16_t upper_bound);
+  size_t GetOutBoundCount();
+  void Mute();
+  void Unmute();
+  void SetBitRate(int rate);
+
+ private:
+  // This class checks all output values and count the number of samples that
+  // go out of a defined range.
+  class VoEOutputCheckMediaProcess : public VoEMediaProcess {
+   public:
+    VoEOutputCheckMediaProcess();
+    void set_enabled(bool enabled) { enabled_ = enabled; }
+    void set_lower_bound(int16_t lower_bound) { lower_bound_ = lower_bound; }
+    void set_upper_bound(int16_t upper_bound) { upper_bound_ = upper_bound; }
+    size_t out_bound_count() const { return out_bound_count_; }
+    void Process(int channel,
+                 ProcessingTypes type,
+                 int16_t audio10ms[],
+                 size_t length,
+                 int samplingFreq,
+                 bool isStereo) override;
+
+   private:
+    bool enabled_;
+    int16_t lower_bound_;
+    int16_t upper_bound_;
+    size_t out_bound_count_;
+  };
+
+  VoETestManager manager_;
+  VoEOutputCheckMediaProcess output_checker_;
+
+  int channel_;
+};
+
+OutputTest::OutputTest() {
+  RTC_CHECK(manager_.Init());
+  manager_.GetInterfaces();
+
+  VoEBase* base = manager_.BasePtr();
+  VoECodec* codec = manager_.CodecPtr();
+  VoENetwork* network = manager_.NetworkPtr();
+  VoEVolumeControl* volume = manager_.VolumeControlPtr();
+  VoEExternalMedia* external = manager_.ExternalMediaPtr();
+
+  RTC_DCHECK_EQ(0, base->Init());
+
+  channel_ = base->CreateChannel();
+
+  // |network| will take care of the life time of |transport|.
+  VoiceChannelTransport* transport =
+      new VoiceChannelTransport(network, channel_);
+
+  RTC_DCHECK_EQ(0, transport->SetSendDestination(kIp, kPort));
+  RTC_DCHECK_EQ(0, transport->SetLocalReceiver(kPort));
+
+  RTC_DCHECK_EQ(0, codec->SetSendCodec(channel_, kCodecInst));
+  RTC_DCHECK_EQ(0, codec->SetOpusDtx(channel_, true));
+
+  RTC_DCHECK_EQ(0, volume->SetSpeakerVolume(255));
+
+  external->RegisterExternalMediaProcessing(
+      channel_, ProcessingTypes::kPlaybackPerChannel, output_checker_);
+}
+
+OutputTest::~OutputTest() {
+  VoENetwork* network = manager_.NetworkPtr();

[kwiberg-webrtc, 2015/10/29 16:35:21]

You don't need to bind this to a local variable.

[minyue-webrtc, 2015/10/30 14:06:46]

you mean "manager_.NetworkPtr()->DeRegisterExternalTransport" in the next line?

+  RTC_DCHECK_EQ(0, network->DeRegisterExternalTransport(channel_));
+  RTC_DCHECK_EQ(0, manager_.ReleaseInterfaces());
+}
+
+void OutputTest::Start() {
+  const std::string file_name =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
+
+  VoEFile* file = manager_.FilePtr();
+  RTC_DCHECK_EQ(
+      0, file->StartPlayingFileAsMicrophone(channel_, file_name.c_str(), true,
+                                            false, kInputFormat, 1.0));
+
+  VoEBase* base = manager_.BasePtr();
+  RTC_DCHECK_EQ(0, base->StartPlayout(channel_));
+  RTC_DCHECK_EQ(0, base->StartSend(channel_));
+}
+
+void OutputTest::EnableOutputCheck() {
+  output_checker_.set_enabled(true);
+}
+
+void OutputTest::DisableOutputCheck() {
+  output_checker_.set_enabled(false);
+}
+
+void OutputTest::SetOutputBound(int16_t lower_bound, int16_t upper_bound) {
+  output_checker_.set_lower_bound(lower_bound);
+  output_checker_.set_upper_bound(upper_bound);
+}
+
+size_t OutputTest::GetOutBoundCount() {
+  return output_checker_.out_bound_count();
+}
+
+void OutputTest::Mute() {
+  VoEVolumeControl* volume = manager_.VolumeControlPtr();
+  volume->SetInputMute(channel_, true);
+}
+
+void OutputTest::Unmute() {
+  VoEVolumeControl* volume = manager_.VolumeControlPtr();
+  volume->SetInputMute(channel_, false);
+}
+
+void OutputTest::SetBitRate(int rate) {
+  VoECodec* codec = manager_.CodecPtr();
+  codec->SetBitRate(channel_, rate);
+}
+
+OutputTest::VoEOutputCheckMediaProcess::VoEOutputCheckMediaProcess()
+    : enabled_(false),
+      lower_bound_(-32768),
+      upper_bound_(32767),
+      out_bound_count_(0) {}
+
+void OutputTest::VoEOutputCheckMediaProcess::Process(int channel,
+                                                     ProcessingTypes type,
+                                                     int16_t* audio10ms,
+                                                     size_t length,
+                                                     int samplingFreq,
+                                                     bool isStereo) {
+  if (!enabled_)
+    return;
+  const int num_channels = isStereo ? 2 : 1;
+  for (size_t i = 0; i < length; ++i) {
+    for (int c = 0; c < num_channels; ++c, ++audio10ms) {
+      if (*audio10ms < lower_bound_ || *audio10ms > upper_bound_) {
+        ++out_bound_count_;
+      }
+    }
+  }
+}
+
+TEST(OutputTest, OpusDtxHasNoNoisePump) {
+  const int kRuntimeMs = 20000;
+  const int kUnmuteTimeMs = 1000;
+  const int kCheckAfterMute = 2000;
+  const int kCheckTimeMs = 2000;
+  const int kMinOpusRate = 6000;
+  const int kMaxOpusRate = 64000;
+  const int16_t kDtxBoundForSilence = 2;
+
+  OutputTest test;
+  Random random(1234ull);
+
+  uint32_t start_time = rtc::Time();
+  test.Start();
+  test.SetOutputBound(-kDtxBoundForSilence, kDtxBoundForSilence);
+  while (rtc::TimeSince(start_time) < kRuntimeMs) {
+    webrtc::SleepMs(kUnmuteTimeMs + random.Gaussian(0, 100));

[kwiberg-webrtc, 2015/10/29 16:35:21]

Replace 100 with e.g. kUnmuteTimeMs/10 here, to make it easier to read?

And in principle check that the value is positive, although I guess we'd have to
be *really* unlucky...

[minyue-webrtc, 2015/10/30 14:06:46]

To avoid unlucky, we can use a bounded distribution

+    test.Mute();
+    webrtc::SleepMs(kCheckAfterMute);
+    test.EnableOutputCheck();
+    webrtc::SleepMs(kCheckTimeMs + random.Gaussian(0, 100));
+    test.DisableOutputCheck();
+    test.SetBitRate(random.Rand(kMinOpusRate, kMaxOpusRate));
+    test.Unmute();
+  }
+  EXPECT_EQ(0u, test.GetOutBoundCount());
+}
+
+}  // namespace voetest