Adds more unit tests for WebRTC.

content/renderer/media/webrtc_audio_device_unittest.cc

 // Copyright (c) 2011 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/environment.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/test/webrtc_audio_device_test.h"
 #include "media/audio/audio_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_base.h"
+#include "third_party/webrtc/voice_engine/main/interface/voe_external_media.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_file.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_network.h"
 
 using testing::_;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
 
 namespace {
 
@@ skipping 34 matching lines @@
   DISALLOW_COPY_AND_ASSIGN(AudioUtilNoHardware);
 };
 
 bool IsRunningHeadless() {
   scoped_ptr<base::Environment> env(base::Environment::Create());
   if (env->HasVar("CHROME_HEADLESS"))
     return true;
   return false;
 }
 
+class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
+ public:
+  explicit WebRTCMediaProcessImpl(base::WaitableEvent* event)
+      : event_(event),
+        channel_id_(-1),
+        type_(webrtc::kPlaybackPerChannel),
+        packet_size_(0),
+        sample_rate_(0),
+        channels_(0) {
+  }
+  virtual ~WebRTCMediaProcessImpl() {}
+
+  // TODO(henrika): Refactor in WebRTC and convert to Chrome coding style.
+  virtual void Process(const int channel,
+                       const webrtc::ProcessingTypes type,
+                       WebRtc_Word16 audio_10ms[],
+                       const int length,
+                       const int sampling_freq,
+                       const bool is_stereo) {
+    channel_id_ = channel;
+    type_ = type;
+    packet_size_ = length;
+    sample_rate_ = sampling_freq;
+    channels_ = (is_stereo ? 2 : 1);
+    if (event_) {
+      // Signal that a new callback has been received.
+      event_->Signal();
+    }
+  }
+
+  int channel_id() const { return channel_id_; }
+  int type() const { return type_; }
+  int packet_size() const { return packet_size_; }
+  int sample_rate() const { return sample_rate_; }
+  int channels() const { return channels_; }
+
+ private:
+  base::WaitableEvent* event_;
+  int channel_id_;
+  webrtc::ProcessingTypes type_;
+  int packet_size_;
+  int sample_rate_;
+  int channels_;
+  DISALLOW_COPY_AND_ASSIGN(WebRTCMediaProcessImpl);
+};
+
 }  // end namespace
 
 // Basic test that instantiates and initializes an instance of
 // WebRtcAudioDeviceImpl.
 TEST_F(WebRTCAudioDeviceTest, Construct) {
   AudioUtilNoHardware audio_util(48000.0, 48000.0);
   set_audio_util_callback(&audio_util);
   scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
       new WebRtcAudioDeviceImpl());
   audio_device->SetSessionId(1);
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   int err = base->Init(audio_device);
   EXPECT_EQ(0, err);
   EXPECT_EQ(0, base->Terminate());
 }
 
+// Verify that a call to webrtc::VoEBase::StartPlayout() starts audio output
+// with the correct set of parameters. A WebRtcAudioDeviceImpl instance will
+// be utilized to implement the actual audio path. The test registers a
+// webrtc::VoEExternalMedia implementation to hijack the output audio and
+// verify that streaming starts correctly.
+// Disabled when running headless since the bots don't have the required config.
+TEST_F(WebRTCAudioDeviceTest, StartPlayout) {
+  if (IsRunningHeadless())
+    return;
+
+  AudioUtil audio_util;
+  set_audio_util_callback(&audio_util);
+
+  EXPECT_CALL(media_observer(),
+      OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
+  EXPECT_CALL(media_observer(),
+      OnSetAudioStreamPlaying(_, 1, true)).Times(1);
+  EXPECT_CALL(media_observer(),
+      OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
+  EXPECT_CALL(media_observer(),
+      OnDeleteAudioStream(_, 1)).Times(1);
+
+  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+      new WebRtcAudioDeviceImpl());
+  audio_device->SetSessionId(1);
+  WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
+  ASSERT_TRUE(engine.valid());
+
+  ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
+  ASSERT_TRUE(base.valid());
+  int err = base->Init(audio_device);
+  ASSERT_EQ(0, err);
+
+  int ch = base->CreateChannel();
+  EXPECT_NE(-1, ch);
+
+  ScopedWebRTCPtr<webrtc::VoEExternalMedia> external_media(engine.get());
+  ASSERT_TRUE(external_media.valid());
+
+  base::WaitableEvent event(false, false);
+  scoped_ptr<WebRTCMediaProcessImpl> media_process(
+      new WebRTCMediaProcessImpl(&event));
+  EXPECT_EQ(0, external_media->RegisterExternalMediaProcessing(
+      ch, webrtc::kPlaybackPerChannel, *media_process.get()));
+
+  EXPECT_EQ(0, base->StartPlayout(ch));
+
+  EXPECT_TRUE(event.TimedWait(
+      base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));
+  WaitForIOThreadCompletion();
+
+  EXPECT_TRUE(audio_device->playing());
+  EXPECT_FALSE(audio_device->recording());
+  EXPECT_EQ(ch, media_process->channel_id());
+  EXPECT_EQ(webrtc::kPlaybackPerChannel, media_process->type());
+  EXPECT_EQ(80, media_process->packet_size());
+  EXPECT_EQ(8000, media_process->sample_rate());
+
+  EXPECT_EQ(0, external_media->DeRegisterExternalMediaProcessing(
+      ch, webrtc::kPlaybackPerChannel));
+  EXPECT_EQ(0, base->StopPlayout(ch));
+
+  EXPECT_EQ(0, base->DeleteChannel(ch));
+  EXPECT_EQ(0, base->Terminate());
+}
+
+// Verify that a call to webrtc::VoEBase::StartRecording() starts audio input
+// with the correct set of parameters. A WebRtcAudioDeviceImpl instance will
+// be utilized to implement the actual audio path. The test registers a
+// webrtc::VoEExternalMedia implementation to hijack the input audio and
+// verify that streaming starts correctly. An external transport implementation
+// is also required to ensure that "sending" can start without actually trying
+// to send encoded packets to the network. Our main interest here is to ensure
+// that the audio capturing starts as it should.
+// Disabled when running headless since the bots don't have the required config.
+TEST_F(WebRTCAudioDeviceTest, StartRecording) {
+  if (IsRunningHeadless())
+    return;
+
+  AudioUtil audio_util;
+  set_audio_util_callback(&audio_util);
+
+  // TODO(tommi): extend MediaObserver and MockMediaObserver with support
+  // for new interfaces, like OnSetAudioStreamRecording(). When done, add
+  // EXPECT_CALL() macros here.
+
+  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+      new WebRtcAudioDeviceImpl());
+  audio_device->SetSessionId(1);
+  WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
+  ASSERT_TRUE(engine.valid());
+
+  ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
+  ASSERT_TRUE(base.valid());
+  int err = base->Init(audio_device);
+  ASSERT_EQ(0, err);
+
+  int ch = base->CreateChannel();
+  EXPECT_NE(-1, ch);
+
+  ScopedWebRTCPtr<webrtc::VoEExternalMedia> external_media(engine.get());
+  ASSERT_TRUE(external_media.valid());
+
+  base::WaitableEvent event(false, false);
+  scoped_ptr<WebRTCMediaProcessImpl> media_process(
+      new WebRTCMediaProcessImpl(&event));
+  EXPECT_EQ(0, external_media->RegisterExternalMediaProcessing(
+      ch, webrtc::kRecordingPerChannel, *media_process.get()));
+
+  // We must add an external transport implementation to be able to start
+  // recording without actually sending encoded packets to the network. All
+  // we want to do here is to verify that audio capturing starts as it should.
+  ScopedWebRTCPtr<webrtc::VoENetwork> network(engine.get());
+  scoped_ptr<WebRTCTransportImpl> transport(
+      new WebRTCTransportImpl(network.get()));
+  EXPECT_EQ(0, network->RegisterExternalTransport(ch, *transport.get()));
+  EXPECT_EQ(0, base->StartSend(ch));
+
+  EXPECT_TRUE(event.TimedWait(
+      base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));
+  WaitForIOThreadCompletion();
+
+  EXPECT_FALSE(audio_device->playing());
+  EXPECT_TRUE(audio_device->recording());
+  EXPECT_EQ(ch, media_process->channel_id());
+  EXPECT_EQ(webrtc::kRecordingPerChannel, media_process->type());
+  EXPECT_EQ(80, media_process->packet_size());
+  EXPECT_EQ(8000, media_process->sample_rate());
+
+  EXPECT_EQ(0, external_media->DeRegisterExternalMediaProcessing(
+      ch, webrtc::kRecordingPerChannel));
+  EXPECT_EQ(0, base->StopSend(ch));
+
+  EXPECT_EQ(0, base->DeleteChannel(ch));
+  EXPECT_EQ(0, base->Terminate());
+}
+
 // Uses WebRtcAudioDeviceImpl to play a local wave file.
 // Disabled when running headless since the bots don't have the required config.
 TEST_F(WebRTCAudioDeviceTest, PlayLocalFile) {
   if (IsRunningHeadless())
     return;
 
   std::string file_path(
       GetTestDataPath(FILE_PATH_LITERAL("speechmusic_mono_16kHz.pcm")));
 
   AudioUtil audio_util;
@@ skipping 24 matching lines @@
   EXPECT_NE(-1, ch);
   EXPECT_EQ(0, base->StartPlayout(ch));
 
   ScopedWebRTCPtr<webrtc::VoEFile> file(engine.get());
   int duration = 0;
   EXPECT_EQ(0, file->GetFileDuration(file_path.c_str(), duration,
                                      webrtc::kFileFormatPcm16kHzFile));
   EXPECT_NE(0, duration);
 
   EXPECT_EQ(0, file->StartPlayingFileLocally(ch, file_path.c_str(), false,
-      webrtc::kFileFormatPcm16kHzFile));
+                                             webrtc::kFileFormatPcm16kHzFile));
 
   message_loop_.PostDelayedTask(FROM_HERE,
                                 new MessageLoop::QuitTask(),
                                 TestTimeouts::action_timeout_ms());
   message_loop_.Run();
 
   EXPECT_EQ(0, base->Terminate());
 }
+
+// Uses WebRtcAudioDeviceImpl to play out recorded audio in loopback.
+// An external transport implementation is utilized to feed back RTP packets
+// which are recorded, encoded, packetized into RTP packets and finally
+// "transmitted". The RTP packets are then fed back into the VoiceEngine
+// where they are decoded and played out on the default audio output device.
+// Disabled when running headless since the bots don't have the required config.
+// TODO(henrika): improve quality by using a wideband codec, enabling noise-
+// suppressions and perhaps also the digital AGC.
+TEST_F(WebRTCAudioDeviceTest, FullDuplexAudio) {
+  if (IsRunningHeadless())
+    return;
+
+  AudioUtil audio_util;
+  set_audio_util_callback(&audio_util);
+
+  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+      new WebRtcAudioDeviceImpl());
+  audio_device->SetSessionId(1);
+  WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
+  ASSERT_TRUE(engine.valid());
+
+  ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
+  ASSERT_TRUE(base.valid());
+  int err = base->Init(audio_device);
+  ASSERT_EQ(0, err);
+
+  int ch = base->CreateChannel();
+  EXPECT_NE(-1, ch);
+
+  ScopedWebRTCPtr<webrtc::VoENetwork> network(engine.get());
+  scoped_ptr<WebRTCTransportImpl> transport(
+      new WebRTCTransportImpl(network.get()));
+  EXPECT_EQ(0, network->RegisterExternalTransport(ch, *transport.get()));
+  EXPECT_EQ(0, base->StartPlayout(ch));
+  EXPECT_EQ(0, base->StartSend(ch));
+
+  LOG(INFO) << ">> You should now be able to hear yourself in loopback...";
+  message_loop_.PostDelayedTask(FROM_HERE,
+                                new MessageLoop::QuitTask(),
+                                TestTimeouts::action_timeout_ms());
+  message_loop_.Run();
+
+  EXPECT_EQ(0, base->StopSend(ch));
+  EXPECT_EQ(0, base->StopPlayout(ch));
+
+  EXPECT_EQ(0, base->DeleteChannel(ch));
+  EXPECT_EQ(0, base->Terminate());
+}