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..d1e47c7416396a548c7d2a57e2c5f563b64ab9ec
--- /dev/null
+++ b/webrtc/voice_engine/test/auto_test/voe_output_test.cc
@@ -0,0 +1,204 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.

[the sun, 2015/11/09 12:41:01]

Why do we need to test at this level? Shouldn't this be covered in specific
tests for the actual codec?

VoE shouldn't need to know about codec specifics. It should just be a mediator,
forwarding config information to codec instances, and piping data into/out of
them.

I'm really not fond of adding more tests that use the soon-to-be obsolete VoE
interfaces.

Also, I'm really not fond of adding 20s of test time for this one particular
case (20s, to be added to each try job, on each bot, from hereon until the end
of days...)

[minyue-webrtc, 2015/11/09 15:13:00]

I see. There are two reasons that I wanted to check this piece of code in
1. It gives a tool to check the output value range of the whole
encoding/decoding chain.

2. It makes sure that NetEq etc. does not bring any problems in the DTX
particular situation.

3. This is a fuzz test and can be run arbitrary time (20s can be changed). I
used it for local testing. But I think others may also use it to debug similar
problems.

Regarding old VoE APIs, this test relies on VoETestManager to talk to VoE, which
is used in other tests. If the plan is to refactor VoETestManager, no particular
cares may need to be taken in this test.

But if you insist, I can remove the test.

[the sun, 2015/11/09 15:29:56]

Ah, I see, sorry I didn't notice it was a fuzzer test. Very well then.

+ *
+ *  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/scoped_ptr.h"
+#include "webrtc/base/timeutils.h"
+#include "webrtc/system_wrappers/include/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);

[the sun, 2015/11/09 12:41:01]

Make this settable in c-tor instead - the bounds don't change in your test.

[minyue-webrtc, 2015/11/09 15:13:00]

ok. I just wanted to make this class a bit more generic.

+  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; }

[the sun, 2015/11/09 12:41:00]

set in c-tor

[minyue-webrtc, 2015/11/09 15:13:00]

Done.

+    void set_upper_bound(int16_t upper_bound) { upper_bound_ = upper_bound; }
+    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_;
+  };
+
+  VoETestManager manager_;
+  VoEOutputCheckMediaProcess output_checker_;
+
+  int channel_;
+};
+
+OutputTest::OutputTest() {
+  EXPECT_TRUE(manager_.Init());
+  manager_.GetInterfaces();
+
+  VoEBase* base = manager_.BasePtr();
+  VoECodec* codec = manager_.CodecPtr();
+  VoENetwork* network = manager_.NetworkPtr();
+
+  EXPECT_EQ(0, base->Init());
+
+  channel_ = base->CreateChannel();
+
+  // |network| will take care of the life time of |transport|.
+  VoiceChannelTransport* transport =
+      new VoiceChannelTransport(network, channel_);
+
+  EXPECT_EQ(0, transport->SetSendDestination(kIp, kPort));
+  EXPECT_EQ(0, transport->SetLocalReceiver(kPort));
+
+  EXPECT_EQ(0, codec->SetSendCodec(channel_, kCodecInst));
+  EXPECT_EQ(0, codec->SetOpusDtx(channel_, true));
+
+  EXPECT_EQ(0, manager_.VolumeControlPtr()->SetSpeakerVolume(255));
+
+  manager_.ExternalMediaPtr()->RegisterExternalMediaProcessing(
+      channel_, ProcessingTypes::kPlaybackPerChannel, output_checker_);
+}
+
+OutputTest::~OutputTest() {
+  EXPECT_EQ(0, manager_.NetworkPtr()->DeRegisterExternalTransport(channel_));
+  EXPECT_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;
+
+  ASSERT_EQ(0,
+            manager_.FilePtr()->StartPlayingFileAsMicrophone(
+             channel_, file_name.c_str(), true, false, kInputFormat, 1.0));

[the sun, 2015/11/09 12:41:01]

nit: formatting

[minyue-webrtc, 2015/11/09 15:13:00]

Done.

+
+  VoEBase* base = manager_.BasePtr();
+  ASSERT_EQ(0, base->StartPlayout(channel_));
+  ASSERT_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);
+}
+
+void OutputTest::Mute() {
+  manager_.VolumeControlPtr()->SetInputMute(channel_, true);
+}
+
+void OutputTest::Unmute() {
+  manager_.VolumeControlPtr()->SetInputMute(channel_, false);
+}
+
+void OutputTest::SetBitRate(int rate) {
+  manager_.CodecPtr()->SetBitRate(channel_, rate);
+}
+
+OutputTest::VoEOutputCheckMediaProcess::VoEOutputCheckMediaProcess()
+    : enabled_(false),
+      lower_bound_(-32768),
+      upper_bound_(32767) {}
+
+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) {
+      ASSERT_GE(audio10ms[i * num_channels + c], lower_bound_);
+      ASSERT_LE(audio10ms[i * num_channels + c], upper_bound_);
+    }
+  }
+}
+
+TEST(OutputTest, OpusDtxHasNoNoisePump) {
+  const int kRuntimeMs = 20000;
+  const uint32_t kUnmuteTimeMs = 1000;
+  const int kCheckAfterMute = 2000;
+  const uint32_t kCheckTimeMs = 2000;
+  const int kMinOpusRate = 6000;
+  const int kMaxOpusRate = 64000;
+
+#if defined(OPUS_FIXED_POINT)
+  const int16_t kDtxBoundForSilence = 20;
+#else
+  const int16_t kDtxBoundForSilence = 2;
+#endif
+
+  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(random.Rand(kUnmuteTimeMs - kUnmuteTimeMs / 10,
+                                kUnmuteTimeMs + kUnmuteTimeMs / 10));
+    test.Mute();
+    webrtc::SleepMs(kCheckAfterMute);
+    test.EnableOutputCheck();
+    webrtc::SleepMs(random.Rand(kCheckTimeMs - kCheckTimeMs / 10,
+                                kCheckTimeMs + kCheckTimeMs / 10));
+    test.DisableOutputCheck();
+    test.SetBitRate(random.Rand(kMinOpusRate, kMaxOpusRate));
+    test.Unmute();
+  }
+}
+
+}  // namespace voetest