Remove Soundclip handling from libjingle.

BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51009004

Cr-Commit-Position: refs/heads/master@{#9216}

talk/app/webrtc/peerconnectionfactory.cc

@@ -276,7 +276,7 @@ rtc::Thread* PeerConnectionFactory::worker_thread() {
 cricket::MediaEngineInterface* PeerConnectionFactory::CreateMediaEngine_w() {
   ASSERT(worker_thread_ == rtc::Thread::Current());
   return cricket::WebRtcMediaEngineFactory::Create(
-      default_adm_.get(), NULL, video_encoder_factory_.get(),
+      default_adm_.get(), video_encoder_factory_.get(),
       video_decoder_factory_.get());
 }
 

talk/libjingle.gyp

@@ -673,8 +673,6 @@
         'session/media/mediasink.h',
         'session/media/rtcpmuxfilter.cc',
         'session/media/rtcpmuxfilter.h',
-        'session/media/soundclip.cc',
-        'session/media/soundclip.h',
         'session/media/srtpfilter.cc',
         'session/media/srtpfilter.h',
         'session/media/typingmonitor.cc',

talk/media/base/fakemediaengine.h

@@ -630,11 +630,6 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
   int max_bps_;
 };
 
-class FakeSoundclipMedia : public SoundclipMedia {
- public:
-  virtual bool PlaySound(const char* buf, int len, int flags) { return true; }
-};
-
 class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
  public:
   explicit FakeDataMediaChannel(void* unused)
@@ -784,7 +779,6 @@ class FakeVoiceEngine : public FakeBaseEngine {
   void UnregisterChannel(VoiceMediaChannel* channel) {
     channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
   }
-  SoundclipMedia* CreateSoundclip() { return new FakeSoundclipMedia(); }
 
   const std::vector<AudioCodec>& codecs() { return codecs_; }
   void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }

talk/media/base/filemediaengine.h

@@ -88,7 +88,6 @@ class FileMediaEngine : public MediaEngineInterface {
   virtual VoiceMediaChannel* CreateChannel();
   virtual VideoMediaChannel* CreateVideoChannel(const VideoOptions& options,
                                                 VoiceMediaChannel* voice_ch);
-  virtual SoundclipMedia* CreateSoundclip() { return NULL; }
   virtual AudioOptions GetAudioOptions() const { return AudioOptions(); }
   virtual bool SetAudioOptions(const AudioOptions& options) { return true; }
   virtual bool SetAudioDelayOffset(int offset) { return true; }

talk/media/base/filemediaengine_unittest.cc

@@ -215,7 +215,6 @@ TEST_F(FileMediaEngineTest, TestDefaultImplementation) {
   EXPECT_EQ(0, engine_->GetCapabilities());
   EXPECT_TRUE(NULL == voice_channel_.get());
   EXPECT_TRUE(NULL == video_channel_.get());
-  EXPECT_TRUE(NULL == engine_->CreateSoundclip());
   cricket::AudioOptions audio_options;
   EXPECT_TRUE(engine_->SetAudioOptions(audio_options));
   VideoEncoderConfig video_encoder_config;

talk/media/base/mediachannel.h

@@ -446,22 +446,6 @@ struct VideoOptions {
   Settable<int> screencast_min_bitrate;
 };
 
-// A class for playing out soundclips.
-class SoundclipMedia {
- public:
-  enum SoundclipFlags {
-    SF_LOOP = 1,
-  };
-
-  virtual ~SoundclipMedia() {}
-
-  // Plays a sound out to the speakers with the given audio stream. The stream
-  // must be 16-bit little-endian 16 kHz PCM. If a stream is already playing
-  // on this SoundclipMedia, it is stopped. If clip is NULL, nothing is played.
-  // Returns whether it was successful.
-  virtual bool PlaySound(const char *clip, int len, int flags) = 0;
-};
-
 struct RtpHeaderExtension {
   RtpHeaderExtension() : id(0) {}
   RtpHeaderExtension(const std::string& u, int i) : uri(u), id(i) {}

talk/media/base/mediaengine.h

@@ -84,9 +84,6 @@ class MediaEngineInterface {
       const VideoOptions& options,
       VoiceMediaChannel* voice_media_channel) = 0;
 
-  // Creates a soundclip object for playing sounds on. Returns NULL on failure.
-  virtual SoundclipMedia *CreateSoundclip() = 0;
-
   // Configuration
   // Gets global audio options.
   virtual AudioOptions GetAudioOptions() const = 0;
@@ -101,7 +98,6 @@ class MediaEngineInterface {
       = 0;
 
   // Device selection
-  // TODO(tschmelcher): Add method for selecting the soundclip device.
   virtual bool SetSoundDevices(const Device* in_device,
                                const Device* out_device) = 0;
 
@@ -193,9 +189,6 @@ class CompositeMediaEngine : public MediaEngineInterface {
                                                 VoiceMediaChannel* channel) {
     return video_.CreateChannel(options, channel);
   }
-  virtual SoundclipMedia *CreateSoundclip() {
-    return voice_.CreateSoundclip();
-  }
 
   virtual AudioOptions GetAudioOptions() const {
     return voice_.GetOptions();
@@ -279,9 +272,6 @@ class NullVoiceEngine {
   VoiceMediaChannel* CreateChannel() {
     return NULL;
   }
-  SoundclipMedia* CreateSoundclip() {
-    return NULL;
-  }
   bool SetDelayOffset(int offset) { return true; }
   AudioOptions GetOptions() const { return AudioOptions(); }
   bool SetOptions(const AudioOptions& options) { return true; }

talk/media/other/linphonemediaengine.h

@@ -68,7 +68,6 @@ class LinphoneMediaEngine : public MediaEngineInterface {
   virtual int GetCapabilities();
   virtual VoiceMediaChannel* CreateChannel();
   virtual VideoMediaChannel* CreateVideoChannel(VoiceMediaChannel* voice_ch);
-  virtual SoundclipMedia* CreateSoundclip() { return NULL; }
   virtual bool SetAudioOptions(int options) { return true; }
   virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
     return true;

talk/media/webrtc/webrtcmediaengine.cc

@@ -37,10 +37,9 @@ class WebRtcMediaEngine2
     : public CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine2> {
  public:
   WebRtcMediaEngine2(webrtc::AudioDeviceModule* adm,
-                     webrtc::AudioDeviceModule* adm_sc,
                      WebRtcVideoEncoderFactory* encoder_factory,
                      WebRtcVideoDecoderFactory* decoder_factory) {
-    voice_.SetAudioDeviceModule(adm, adm_sc);
+    voice_.SetAudioDeviceModule(adm);
     video_.SetExternalDecoderFactory(decoder_factory);
     video_.SetExternalEncoderFactory(encoder_factory);
   }
@@ -51,10 +50,9 @@ class WebRtcMediaEngine2
 WRME_EXPORT
 cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
     webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory) {
-  return new cricket::WebRtcMediaEngine2(adm, adm_sc, encoder_factory,
+  return new cricket::WebRtcMediaEngine2(adm, encoder_factory,
                                          decoder_factory);
 }
 
@@ -69,10 +67,9 @@ namespace cricket {
 // ChannelManager.
 MediaEngineInterface* WebRtcMediaEngineFactory::Create(
     webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc,
     WebRtcVideoEncoderFactory* encoder_factory,
     WebRtcVideoDecoderFactory* decoder_factory) {
-  return CreateWebRtcMediaEngine(adm, adm_sc, encoder_factory, decoder_factory);
+  return CreateWebRtcMediaEngine(adm, encoder_factory, decoder_factory);
 }
 
 }  // namespace cricket

talk/media/webrtc/webrtcmediaengine.h

@@ -45,7 +45,7 @@ class WebRtcVideoEncoderFactory;
 
 WRME_EXPORT
 cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
-    webrtc::AudioDeviceModule* adm, webrtc::AudioDeviceModule* adm_sc,
+    webrtc::AudioDeviceModule* adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
@@ -69,7 +69,6 @@ class WebRtcMediaEngineFactory {
         // !defined(LIBPEERCONNECTION_IMPLEMENTATION)
   static MediaEngineInterface* Create(
       webrtc::AudioDeviceModule* adm,
-      webrtc::AudioDeviceModule* adm_sc,
       WebRtcVideoEncoderFactory* encoder_factory,
       WebRtcVideoDecoderFactory* decoder_factory);
 };
@@ -88,11 +87,10 @@ class DelegatingWebRtcMediaEngine : public cricket::MediaEngineInterface {
  public:
   DelegatingWebRtcMediaEngine(
       webrtc::AudioDeviceModule* adm,
-      webrtc::AudioDeviceModule* adm_sc,
       WebRtcVideoEncoderFactory* encoder_factory,
       WebRtcVideoDecoderFactory* decoder_factory)
       : delegate_(CreateWebRtcMediaEngine(
-          adm, adm_sc, encoder_factory, decoder_factory)) {
+          adm, encoder_factory, decoder_factory)) {
   }
   virtual ~DelegatingWebRtcMediaEngine() {
     DestroyWebRtcMediaEngine(delegate_);
@@ -110,9 +108,6 @@ class DelegatingWebRtcMediaEngine : public cricket::MediaEngineInterface {
       VoiceMediaChannel* voice_media_channel) override {
     return delegate_->CreateVideoChannel(options, voice_media_channel);
   }
-  SoundclipMedia* CreateSoundclip() override {
-    return delegate_->CreateSoundclip();
-  }
   AudioOptions GetAudioOptions() const override {
     return delegate_->GetAudioOptions();
   }
@@ -186,11 +181,10 @@ class DelegatingWebRtcMediaEngine : public cricket::MediaEngineInterface {
 // ChannelManager.
 MediaEngineInterface* WebRtcMediaEngineFactory::Create(
     webrtc::AudioDeviceModule* adm,
-    webrtc::AudioDeviceModule* adm_sc,
     WebRtcVideoEncoderFactory* encoder_factory,
     WebRtcVideoDecoderFactory* decoder_factory) {
   return new cricket::DelegatingWebRtcMediaEngine(
-      adm, adm_sc, encoder_factory, decoder_factory);
+      adm, encoder_factory, decoder_factory);
 }
 
 }  // namespace cricket

talk/media/webrtc/webrtcvoiceengine.cc

@@ -103,7 +103,6 @@ static const CodecPref kCodecPrefs[] = {
 
 #ifdef WIN32
 static const int kDefaultAudioDeviceId = -1;
-static const int kDefaultSoundclipDeviceId = -2;
 #else
 static const int kDefaultAudioDeviceId = 0;
 #endif
@@ -368,103 +367,10 @@ static std::string GetEnableString(bool enable) {
   return enable ? "enable" : "disable";
 }
 
-class WebRtcSoundclipMedia : public SoundclipMedia {
- public:
-  explicit WebRtcSoundclipMedia(WebRtcVoiceEngine *engine)
-      : engine_(engine), webrtc_channel_(-1) {
-    engine_->RegisterSoundclip(this);
-  }
-
-  ~WebRtcSoundclipMedia() override {
-    engine_->UnregisterSoundclip(this);
-    if (webrtc_channel_ != -1) {
-      // We shouldn't have to call Disable() here. DeleteChannel() should call
-      // StopPlayout() while deleting the channel.  We should fix the bug
-      // inside WebRTC and remove the Disable() call bellow.  This work is
-      // tracked by bug http://b/issue?id=5382855.
-      PlaySound(NULL, 0, 0);
-      Disable();
-      if (engine_->voe_sc()->base()->DeleteChannel(webrtc_channel_)
-          == -1) {
-        LOG_RTCERR1(DeleteChannel, webrtc_channel_);
-      }
-    }
-  }
-
-  bool Init() {
-    if (!engine_->voe_sc()) {
-      return false;
-    }
-    webrtc_channel_ = engine_->CreateSoundclipVoiceChannel();
-    if (webrtc_channel_ == -1) {
-      LOG_RTCERR0(CreateChannel);
-      return false;
-    }
-    return true;
-  }
-
-  bool Enable() {
-    if (engine_->voe_sc()->base()->StartPlayout(webrtc_channel_) == -1) {
-      LOG_RTCERR1(StartPlayout, webrtc_channel_);
-      return false;
-    }
-    return true;
-  }
-
-  bool Disable() {
-    if (engine_->voe_sc()->base()->StopPlayout(webrtc_channel_) == -1) {
-      LOG_RTCERR1(StopPlayout, webrtc_channel_);
-      return false;
-    }
-    return true;
-  }
-
-  bool PlaySound(const char* buf, int len, int flags) override {
-    // The voe file api is not available in chrome.
-    if (!engine_->voe_sc()->file()) {
-      return false;
-    }
-    // Must stop playing the current sound (if any), because we are about to
-    // modify the stream.
-    if (engine_->voe_sc()->file()->StopPlayingFileLocally(webrtc_channel_)
-        == -1) {
-      LOG_RTCERR1(StopPlayingFileLocally, webrtc_channel_);
-      return false;
-    }
-
-    if (buf) {
-      stream_.reset(new WebRtcSoundclipStream(buf, len));
-      stream_->set_loop((flags & SF_LOOP) != 0);
-      stream_->Rewind();
-
-      // Play it.
-      if (engine_->voe_sc()->file()->StartPlayingFileLocally(
-          webrtc_channel_, stream_.get()) == -1) {
-        LOG_RTCERR2(StartPlayingFileLocally, webrtc_channel_, stream_.get());
-        LOG(LS_ERROR) << "Unable to start soundclip";
-        return false;
-      }
-    } else {
-      stream_.reset();
-    }
-    return true;
-  }
-
-  int GetLastEngineError() const { return engine_->voe_sc()->error(); }
-
- private:
-  WebRtcVoiceEngine *engine_;
-  int webrtc_channel_;
-  rtc::scoped_ptr<WebRtcSoundclipStream> stream_;
-};
-
 WebRtcVoiceEngine::WebRtcVoiceEngine()
     : voe_wrapper_(new VoEWrapper()),
-      voe_wrapper_sc_(new VoEWrapper()),
-      voe_wrapper_sc_initialized_(false),
       tracing_(new VoETraceWrapper()),
       adm_(NULL),
-      adm_sc_(NULL),
       log_filter_(SeverityToFilter(kDefaultLogSeverity)),
       is_dumping_aec_(false),
       desired_local_monitor_enable_(false),
@@ -474,14 +380,10 @@ WebRtcVoiceEngine::WebRtcVoiceEngine()
 }
 
 WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
-                                     VoEWrapper* voe_wrapper_sc,
                                      VoETraceWrapper* tracing)
     : voe_wrapper_(voe_wrapper),
-      voe_wrapper_sc_(voe_wrapper_sc),
-      voe_wrapper_sc_initialized_(false),
       tracing_(tracing),
       adm_(NULL),
-      adm_sc_(NULL),
       log_filter_(SeverityToFilter(kDefaultLogSeverity)),
       is_dumping_aec_(false),
       desired_local_monitor_enable_(false),
@@ -593,11 +495,6 @@ WebRtcVoiceEngine::~WebRtcVoiceEngine() {
     adm_->Release();
     adm_ = NULL;
   }
-  if (adm_sc_) {
-    voe_wrapper_sc_.reset();
-    adm_sc_->Release();
-    adm_sc_ = NULL;
-  }
 
   // Test to see if the media processor was deregistered properly
   DCHECK(SignalRxMediaFrame.is_empty());
@@ -673,61 +570,12 @@ bool WebRtcVoiceEngine::InitInternal() {
   return true;
 }
 
-bool WebRtcVoiceEngine::EnsureSoundclipEngineInit() {
-  if (voe_wrapper_sc_initialized_) {
-    return true;
-  }
-  // Note that, if initialization fails, voe_wrapper_sc_initialized_ will still
-  // be false, so subsequent calls to EnsureSoundclipEngineInit will
-  // probably just fail again. That's acceptable behavior.
-#if defined(LINUX) && !defined(HAVE_LIBPULSE)
-  voe_wrapper_sc_->hw()->SetAudioDeviceLayer(webrtc::kAudioLinuxAlsa);
-#endif
-
-  // Initialize the VoiceEngine instance that we'll use to play out sound clips.
-  if (voe_wrapper_sc_->base()->Init(adm_sc_) == -1) {
-    LOG_RTCERR0_EX(Init, voe_wrapper_sc_->error());
-    return false;
-  }
-
-  // On Windows, tell it to use the default sound (not communication) devices.
-  // First check whether there is a valid sound device for playback.
-  // TODO(juberti): Clean this up when we support setting the soundclip device.
-#ifdef WIN32
-  // The SetPlayoutDevice may not be implemented in the case of external ADM.
-  // TODO(ronghuawu): We should only check the adm_sc_ here, but current
-  // PeerConnection interface never set the adm_sc_, so need to check both
-  // in order to determine if the external adm is used.
-  if (!adm_ && !adm_sc_) {
-    int num_of_devices = 0;
-    if (voe_wrapper_sc_->hw()->GetNumOfPlayoutDevices(num_of_devices) != -1 &&
-        num_of_devices > 0) {
-      if (voe_wrapper_sc_->hw()->SetPlayoutDevice(kDefaultSoundclipDeviceId)
-          == -1) {
-        LOG_RTCERR1_EX(SetPlayoutDevice, kDefaultSoundclipDeviceId,
-                       voe_wrapper_sc_->error());
-        return false;
-      }
-    } else {
-      LOG(LS_WARNING) << "No valid sound playout device found.";
-    }
-  }
-#endif
-  voe_wrapper_sc_initialized_ = true;
-  LOG(LS_INFO) << "Initialized WebRtc soundclip engine.";
-  return true;
-}
-
 void WebRtcVoiceEngine::Terminate() {
   LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
   initialized_ = false;
 
   StopAecDump();
 
-  if (voe_wrapper_sc_) {
-    voe_wrapper_sc_initialized_ = false;
-    voe_wrapper_sc_->base()->Terminate();
-  }
   voe_wrapper_->base()->Terminate();
   desired_local_monitor_enable_ = false;
 }
@@ -745,20 +593,6 @@ VoiceMediaChannel *WebRtcVoiceEngine::CreateChannel() {
   return ch;
 }
 
-SoundclipMedia *WebRtcVoiceEngine::CreateSoundclip() {
-  if (!EnsureSoundclipEngineInit()) {
-    LOG(LS_ERROR) << "Unable to create soundclip: soundclip engine failed to "
-                  << "initialize.";
-    return NULL;
-  }
-  WebRtcSoundclipMedia *soundclip = new WebRtcSoundclipMedia(this);
-  if (!soundclip->Init() || !soundclip->Enable()) {
-    delete soundclip;
-    return NULL;
-  }
-  return soundclip;
-}
-
 bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) {
   if (!ApplyOptions(options)) {
     return false;
@@ -1532,19 +1366,6 @@ void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel *channel) {
   }
 }
 
-void WebRtcVoiceEngine::RegisterSoundclip(WebRtcSoundclipMedia *soundclip) {
-  soundclips_.push_back(soundclip);
-}
-
-void WebRtcVoiceEngine::UnregisterSoundclip(WebRtcSoundclipMedia *soundclip) {
-  SoundclipList::iterator i = std::find(soundclips_.begin(),
-                                        soundclips_.end(),
-                                        soundclip);
-  if (i != soundclips_.end()) {
-    soundclips_.erase(i);
-  }
-}
-
 // Adjusts the default AGC target level by the specified delta.
 // NB: If we start messing with other config fields, we'll want
 // to save the current webrtc::AgcConfig as well.
@@ -1563,8 +1384,7 @@ bool WebRtcVoiceEngine::AdjustAgcLevel(int delta) {
   return true;
 }
 
-bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
+bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) {
-    webrtc::AudioDeviceModule* adm_sc) {
   if (initialized_) {
     LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init.";
     return false;
@@ -1577,15 +1397,6 @@ bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
     adm_ = adm;
     adm_->AddRef();
   }
-
-  if (adm_sc_) {
-    adm_sc_->Release();
-    adm_sc_ = NULL;
-  }
-  if (adm_sc) {
-    adm_sc_ = adm_sc;
-    adm_sc_->AddRef();
-  }
   return true;
 }
 
@@ -1791,10 +1602,6 @@ int WebRtcVoiceEngine::CreateMediaVoiceChannel() {
   return CreateVoiceChannel(voe_wrapper_.get());
 }
 
-int WebRtcVoiceEngine::CreateSoundclipVoiceChannel() {
-  return CreateVoiceChannel(voe_wrapper_sc_.get());
-}
-
 class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
     : public AudioRenderer::Sink {
  public:

talk/media/webrtc/webrtcvoiceengine.h

@@ -91,7 +91,6 @@ class AudioRenderer;
 class VoETraceWrapper;
 class VoEWrapper;
 class VoiceProcessor;
-class WebRtcSoundclipMedia;
 class WebRtcVoiceMediaChannel;
 
 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine.
@@ -103,9 +102,7 @@ class WebRtcVoiceEngine
  public:
   WebRtcVoiceEngine();
   // Dependency injection for testing.
-  WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
+  WebRtcVoiceEngine(VoEWrapper* voe_wrapper, VoETraceWrapper* tracing);
-                    VoEWrapper* voe_wrapper_sc,
-                    VoETraceWrapper* tracing);
   ~WebRtcVoiceEngine();
   bool Init(rtc::Thread* worker_thread);
   void Terminate();
@@ -113,8 +110,6 @@ class WebRtcVoiceEngine
   int GetCapabilities();
   VoiceMediaChannel* CreateChannel();
 
-  SoundclipMedia* CreateSoundclip();
-
   AudioOptions GetOptions() const { return options_; }
   bool SetOptions(const AudioOptions& options);
   // Overrides, when set, take precedence over the options on a
@@ -166,21 +161,15 @@ class WebRtcVoiceEngine
   void RegisterChannel(WebRtcVoiceMediaChannel *channel);
   void UnregisterChannel(WebRtcVoiceMediaChannel *channel);
 
-  // May only be called by WebRtcSoundclipMedia.
-  void RegisterSoundclip(WebRtcSoundclipMedia *channel);
-  void UnregisterSoundclip(WebRtcSoundclipMedia *channel);
-
   // Called by WebRtcVoiceMediaChannel to set a gain offset from
   // the default AGC target level.
   bool AdjustAgcLevel(int delta);
 
   VoEWrapper* voe() { return voe_wrapper_.get(); }
-  VoEWrapper* voe_sc() { return voe_wrapper_sc_.get(); }
   int GetLastEngineError();
 
-  // Set the external ADMs. This can only be called before Init.
+  // Set the external ADM. This can only be called before Init.
-  bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
+  bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm);
-                            webrtc::AudioDeviceModule* adm_sc);
 
   // Starts AEC dump using existing file.
   bool StartAecDump(rtc::PlatformFile file);
@@ -190,10 +179,8 @@ class WebRtcVoiceEngine
 
   // Create a VoiceEngine Channel.
   int CreateMediaVoiceChannel();
-  int CreateSoundclipVoiceChannel();
 
  private:
-  typedef std::vector<WebRtcSoundclipMedia *> SoundclipList;
   typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
   typedef sigslot::
       signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal;
@@ -202,7 +189,6 @@ class WebRtcVoiceEngine
   void ConstructCodecs();
   bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
-  bool EnsureSoundclipEngineInit();
   void SetTraceFilter(int filter);
   void SetTraceOptions(const std::string& options);
   // Applies either options or overrides.  Every option that is "set"
@@ -250,13 +236,9 @@ class WebRtcVoiceEngine
 
   // The primary instance of WebRtc VoiceEngine.
   rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
-  // A secondary instance, for playing out soundclips (on the 'ring' device).
-  rtc::scoped_ptr<VoEWrapper> voe_wrapper_sc_;
-  bool voe_wrapper_sc_initialized_;
   rtc::scoped_ptr<VoETraceWrapper> tracing_;
   // The external audio device manager
   webrtc::AudioDeviceModule* adm_;
-  webrtc::AudioDeviceModule* adm_sc_;
   int log_filter_;
   std::string log_options_;
   bool is_dumping_aec_;
@@ -264,7 +246,6 @@ class WebRtcVoiceEngine
   std::vector<RtpHeaderExtension> rtp_header_extensions_;
   bool desired_local_monitor_enable_;
   rtc::scoped_ptr<WebRtcMonitorStream> monitor_;
-  SoundclipList soundclips_;
   ChannelList channels_;
   // channels_ can be read from WebRtc callback thread. We need a lock on that
   // callback as well as the RegisterChannel/UnregisterChannel.

talk/media/webrtc/webrtcvoiceengine_unittest.cc

@@ -128,12 +128,10 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
 
   WebRtcVoiceEngineTestFake()
       : voe_(kAudioCodecs, ARRAY_SIZE(kAudioCodecs)),
-        voe_sc_(kAudioCodecs, ARRAY_SIZE(kAudioCodecs)),
         trace_wrapper_(new FakeVoETraceWrapper()),
         engine_(new FakeVoEWrapper(&voe_),
-                new FakeVoEWrapper(&voe_sc_),
                 trace_wrapper_),
-        channel_(NULL), soundclip_(NULL) {
+        channel_(NULL) {
     options_conference_.conference_mode.Set(true);
     options_adjust_agc_.adjust_agc_delta.Set(-10);
   }
@@ -168,7 +166,6 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     channel_->OnPacketReceived(&packet, rtc::PacketTime());
   }
   void TearDown() override {
-    delete soundclip_;
     delete channel_;
     engine_.Terminate();
   }
@@ -335,11 +332,9 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
 
  protected:
   cricket::FakeWebRtcVoiceEngine voe_;
-  cricket::FakeWebRtcVoiceEngine voe_sc_;
   FakeVoETraceWrapper* trace_wrapper_;
   cricket::WebRtcVoiceEngine engine_;
   cricket::VoiceMediaChannel* channel_;
-  cricket::SoundclipMedia* soundclip_;
 
   cricket::AudioOptions options_conference_;
   cricket::AudioOptions options_adjust_agc_;
@@ -348,14 +343,10 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
 // Tests that our stub library "works".
 TEST_F(WebRtcVoiceEngineTestFake, StartupShutdown) {
   EXPECT_FALSE(voe_.IsInited());
-  EXPECT_FALSE(voe_sc_.IsInited());
   EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
   EXPECT_TRUE(voe_.IsInited());
-  // The soundclip engine is lazily initialized.
-  EXPECT_FALSE(voe_sc_.IsInited());
   engine_.Terminate();
   EXPECT_FALSE(voe_.IsInited());
-  EXPECT_FALSE(voe_sc_.IsInited());
 }
 
 // Tests that we can create and destroy a channel.
@@ -2652,35 +2643,6 @@ TEST_F(WebRtcVoiceEngineTestFake, PlayRingbackWithMultipleStreams) {
   EXPECT_EQ(0, voe_.IsPlayingFileLocally(channel_num));
 }
 
-// Tests creating soundclips, and make sure they come from the right engine.
-TEST_F(WebRtcVoiceEngineTestFake, CreateSoundclip) {
-  EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  EXPECT_FALSE(voe_sc_.IsInited());
-  soundclip_ = engine_.CreateSoundclip();
-  EXPECT_TRUE(voe_sc_.IsInited());
-  ASSERT_TRUE(soundclip_ != NULL);
-  EXPECT_EQ(0, voe_.GetNumChannels());
-  EXPECT_EQ(1, voe_sc_.GetNumChannels());
-  int channel_num = voe_sc_.GetLastChannel();
-  EXPECT_TRUE(voe_sc_.GetPlayout(channel_num));
-  delete soundclip_;
-  soundclip_ = NULL;
-  EXPECT_EQ(0, voe_sc_.GetNumChannels());
-  // Make sure the soundclip engine is uninitialized on shutdown, now that
-  // we've initialized it by creating a soundclip.
-  engine_.Terminate();
-  EXPECT_FALSE(voe_sc_.IsInited());
-}
-
-// Tests playing out a fake sound.
-TEST_F(WebRtcVoiceEngineTestFake, PlaySoundclip) {
-  static const char kZeroes[16000] = {};
-  EXPECT_TRUE(engine_.Init(rtc::Thread::Current()));
-  soundclip_ = engine_.CreateSoundclip();
-  ASSERT_TRUE(soundclip_ != NULL);
-  EXPECT_TRUE(soundclip_->PlaySound(kZeroes, sizeof(kZeroes), 0));
-}
-
 TEST_F(WebRtcVoiceEngineTestFake, MediaEngineCallbackOnError) {
   rtc::scoped_ptr<ChannelErrorListener> listener;
   cricket::WebRtcVoiceMediaChannel* media_channel;

talk/session/media/channelmanager.cc

@@ -41,7 +41,6 @@
 #ifdef HAVE_SCTP
 #include "talk/media/sctp/sctpdataengine.h"
 #endif
-#include "talk/session/media/soundclip.h"
 #include "talk/session/media/srtpfilter.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/common.h"
@@ -311,9 +310,6 @@ void ChannelManager::Terminate_w() {
   while (!voice_channels_.empty()) {
     DestroyVoiceChannel_w(voice_channels_.back(), nullptr);
   }
-  while (!soundclips_.empty()) {
-    DestroySoundclip_w(soundclips_.back());
-  }
   if (!SetCaptureDevice_w(NULL)) {
     LOG(LS_WARNING) << "failed to delete video capturer";
   }
@@ -504,45 +500,6 @@ void ChannelManager::DestroyDataChannel_w(DataChannel* data_channel) {
   delete data_channel;
 }
 
-Soundclip* ChannelManager::CreateSoundclip() {
-  return worker_thread_->Invoke<Soundclip*>(
-      Bind(&ChannelManager::CreateSoundclip_w, this));
-}
-
-Soundclip* ChannelManager::CreateSoundclip_w() {
-  ASSERT(initialized_);
-  ASSERT(worker_thread_ == rtc::Thread::Current());
-
-  SoundclipMedia* soundclip_media = media_engine_->CreateSoundclip();
-  if (!soundclip_media) {
-    return NULL;
-  }
-
-  Soundclip* soundclip = new Soundclip(worker_thread_, soundclip_media);
-  soundclips_.push_back(soundclip);
-  return soundclip;
-}
-
-void ChannelManager::DestroySoundclip(Soundclip* soundclip) {
-  if (soundclip) {
-    worker_thread_->Invoke<void>(
-        Bind(&ChannelManager::DestroySoundclip_w, this, soundclip));
-  }
-}
-
-void ChannelManager::DestroySoundclip_w(Soundclip* soundclip) {
-  // Destroy soundclip.
-  ASSERT(initialized_);
-  Soundclips::iterator it = std::find(soundclips_.begin(),
-      soundclips_.end(), soundclip);
-  ASSERT(it != soundclips_.end());
-  if (it == soundclips_.end())
-    return;
-
-  soundclips_.erase(it);
-  delete soundclip;
-}
-
 bool ChannelManager::GetAudioOptions(std::string* in_name,
                                      std::string* out_name,
                                      AudioOptions* options) {

talk/session/media/channelmanager.h

@@ -44,7 +44,6 @@ namespace cricket {
 
 const int kDefaultAudioDelayOffset = 0;
 
-class Soundclip;
 class VideoProcessor;
 class VoiceChannel;
 class VoiceProcessor;
@@ -129,15 +128,9 @@ class ChannelManager : public rtc::MessageHandler,
   // Destroys a data channel created with the Create API.
   void DestroyDataChannel(DataChannel* data_channel);
 
-  // Creates a soundclip.
-  Soundclip* CreateSoundclip();
-  // Destroys a soundclip created with the Create API.
-  void DestroySoundclip(Soundclip* soundclip);
-
   // Indicates whether any channels exist.
   bool has_channels() const {
-    return (!voice_channels_.empty() || !video_channels_.empty() ||
+    return (!voice_channels_.empty() || !video_channels_.empty());
-            !soundclips_.empty());
   }
 
   // Configures the audio and video devices. A null pointer can be passed to
@@ -253,7 +246,6 @@ class ChannelManager : public rtc::MessageHandler,
   typedef std::vector<VoiceChannel*> VoiceChannels;
   typedef std::vector<VideoChannel*> VideoChannels;
   typedef std::vector<DataChannel*> DataChannels;
-  typedef std::vector<Soundclip*> Soundclips;
 
   void Construct(MediaEngineInterface* me,
                  DataEngineInterface* dme,
@@ -277,8 +269,6 @@ class ChannelManager : public rtc::MessageHandler,
       BaseSession* session, const std::string& content_name,
       bool rtcp, DataChannelType data_channel_type);
   void DestroyDataChannel_w(DataChannel* data_channel);
-  Soundclip* CreateSoundclip_w();
-  void DestroySoundclip_w(Soundclip* soundclip);
   bool SetAudioOptions_w(const AudioOptions& options, int delay_offset,
                          const Device* in_dev, const Device* out_dev);
   bool SetEngineAudioOptions_w(const AudioOptions& options);
@@ -306,7 +296,6 @@ class ChannelManager : public rtc::MessageHandler,
   VoiceChannels voice_channels_;
   VideoChannels video_channels_;
   DataChannels data_channels_;
-  Soundclips soundclips_;
 
   std::string audio_in_device_;
   std::string audio_out_device_;

talk/session/media/soundclip.cc

@@ -1,82 +1 @@
-/*
+// TODO(solenberg): Remove this file when it's no longer built in Chromium.
- * libjingle
- * Copyright 2004 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/session/media/soundclip.h"
-
-namespace cricket {
-
-enum {
-  MSG_PLAYSOUND = 1,
-};
-
-struct PlaySoundMessageData : rtc::MessageData {
-  PlaySoundMessageData(const void *c,
-                       int l,
-                       SoundclipMedia::SoundclipFlags f)
-      : clip(c),
-        len(l),
-        flags(f),
-        result(false) {
-  }
-
-  const void *clip;
-  int len;
-  SoundclipMedia::SoundclipFlags flags;
-  bool result;
-};
-
-Soundclip::Soundclip(rtc::Thread *thread, SoundclipMedia *soundclip_media)
-    : worker_thread_(thread),
-      soundclip_media_(soundclip_media) {
-}
-
-bool Soundclip::PlaySound(const void *clip,
-                          int len,
-                          SoundclipMedia::SoundclipFlags flags) {
-  PlaySoundMessageData data(clip, len, flags);
-  worker_thread_->Send(this, MSG_PLAYSOUND, &data);
-  return data.result;
-}
-
-bool Soundclip::PlaySound_w(const void *clip,
-                            int len,
-                            SoundclipMedia::SoundclipFlags flags) {
-  return soundclip_media_->PlaySound(static_cast<const char *>(clip),
-                                     len,
-                                     flags);
-}
-
-void Soundclip::OnMessage(rtc::Message *message) {
-  ASSERT(message->message_id == MSG_PLAYSOUND);
-  PlaySoundMessageData *data =
-      static_cast<PlaySoundMessageData *>(message->pdata);
-  data->result = PlaySound_w(data->clip,
-                             data->len,
-                             data->flags);
-}
-
-}  // namespace cricket

talk/session/media/soundclip.h

@@ -1,70 +1,2 @@
-/*
+// TODO(solenberg): Remove this file when it's no longer built in Chromium.
- * libjingle
- * Copyright 2004 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
 
-#ifndef TALK_SESSION_MEDIA_SOUNDCLIP_H_
-#define TALK_SESSION_MEDIA_SOUNDCLIP_H_
-
-#include "talk/media/base/mediaengine.h"
-#include "webrtc/base/scoped_ptr.h"
-
-namespace rtc {
-
-class Thread;
-
-}
-
-namespace cricket {
-
-// Soundclip wraps SoundclipMedia to support marshalling calls to the proper
-// thread.
-class Soundclip : private rtc::MessageHandler {
- public:
-  Soundclip(rtc::Thread* thread, SoundclipMedia* soundclip_media);
-
-  // Plays a sound out to the speakers with the given audio stream. The stream
-  // must be 16-bit little-endian 16 kHz PCM. If a stream is already playing
-  // on this Soundclip, it is stopped. If clip is NULL, nothing is played.
-  // Returns whether it was successful.
-  bool PlaySound(const void* clip,
-                 int len,
-                 SoundclipMedia::SoundclipFlags flags);
-
- private:
-  bool PlaySound_w(const void* clip,
-                   int len,
-                   SoundclipMedia::SoundclipFlags flags);
-
-  // From MessageHandler
-  virtual void OnMessage(rtc::Message* message);
-
-  rtc::Thread* worker_thread_;
-  rtc::scoped_ptr<SoundclipMedia> soundclip_media_;
-};
-
-}  // namespace cricket
-
-#endif  // TALK_SESSION_MEDIA_SOUNDCLIP_H_